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[alsa-devel] [PATCH v2 1/3] ASoC: fsl_sai: add sai master mode support
by Zidan Wang 13 May '15

13 May '15

When sai works on master mode, set its bit clock and frame clock. SAI has 4 MCLK source, bus clock, MCLK1, MCLK2 and MCLK3. fsl_sai_set_bclk will select proper MCLK source, then calculate and set the bit clock divider. After fsl_sai_set_bclk, enable the selected mclk in hw_params(), and add hw_free() to disable the mclk. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_sai.c | 117 ++++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 9 +++- 2 files changed, 121 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ee2671b..1ccc10d1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1,7 +1,7 @@ /* * Freescale ALSA SoC Digital Audio Interface (SAI) driver. * - * Copyright 2012-2013 Freescale Semiconductor, Inc. + * Copyright 2012-2015 Freescale Semiconductor, 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 @@ -251,12 +251,14 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, val_cr4 |= FSL_SAI_CR4_FSD_MSTR; break; case SND_SOC_DAIFMT_CBM_CFM: + sai->is_slave_mode = true; break; case SND_SOC_DAIFMT_CBS_CFM: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; break; case SND_SOC_DAIFMT_CBM_CFS: val_cr4 |= FSL_SAI_CR4_FSD_MSTR; + sai->is_slave_mode = true; break; default: return -EINVAL; @@ -288,6 +290,79 @@ static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) return ret; } +static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(dai); + unsigned long clk_rate; + u32 savediv = 0, ratio, savesub = freq; + u32 id; + int ret = 0; + + /* Don't apply to slave mode */ + if (sai->is_slave_mode) + return 0; + + for (id = 0; id < FSL_SAI_MCLK_MAX; id++) { + clk_rate = clk_get_rate(sai->mclk_clk[id]); + if (!clk_rate) + continue; + + ratio = clk_rate / freq; + + ret = clk_rate - ratio * freq; + + /* + * Drop the source that can not be + * divided into the required rate. + */ + if (ret != 0 && clk_rate / ret < 1000) + continue; + + dev_dbg(dai->dev, + "ratio %d for freq %dHz based on clock %ldHz\n", + ratio, freq, clk_rate); + + if (ratio % 2 == 0 && ratio >= 2 && ratio <= 512) + ratio /= 2; + else + continue; + + if (ret < savesub) { + savediv = ratio; + sai->mclk_id[tx] = id; + savesub = ret; + } + + if (ret == 0) + break; + } + + if (savediv == 0) { + dev_err(dai->dev, "failed to derive required %cx rate: %d\n", + tx ? 'T' : 'R', freq); + return -EINVAL; + } + + if ((tx && sai->synchronous[TX]) || (!tx && !sai->synchronous[RX])) { + regmap_update_bits(sai->regmap, FSL_SAI_RCR2, + FSL_SAI_CR2_MSEL_MASK, + FSL_SAI_CR2_MSEL(sai->mclk_id[tx])); + regmap_update_bits(sai->regmap, FSL_SAI_RCR2, + FSL_SAI_CR2_DIV_MASK, savediv - 1); + } else { + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, + FSL_SAI_CR2_MSEL_MASK, + FSL_SAI_CR2_MSEL(sai->mclk_id[tx])); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, + FSL_SAI_CR2_DIV_MASK, savediv - 1); + } + + dev_dbg(dai->dev, "best fit: clock id=%d, div=%d, deviation =%d\n", + sai->mclk_id[tx], savediv, savesub); + + return 0; +} + static int fsl_sai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -297,6 +372,24 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, unsigned int channels = params_channels(params); u32 word_width = snd_pcm_format_width(params_format(params)); u32 val_cr4 = 0, val_cr5 = 0; + int ret; + + if (!sai->is_slave_mode) { + ret = fsl_sai_set_bclk(cpu_dai, tx, + 2 * word_width * params_rate(params)); + if (ret) + return ret; + + /* Do not enable the clock if it is already enabled */ + if (!(sai->mclk_streams & BIT(substream->stream))) { + ret = clk_prepare_enable(sai->mclk_clk[sai->mclk_id[tx]]); + if (ret) + return ret; + + sai->mclk_streams |= BIT(substream->stream); + } + + } if (!sai->is_dsp_mode) val_cr4 |= FSL_SAI_CR4_SYWD(word_width); @@ -322,6 +415,22 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, return 0; } +static int fsl_sai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + if (!sai->is_slave_mode && + sai->mclk_streams & BIT(substream->stream)) { + clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[tx]]); + sai->mclk_streams &= ~BIT(substream->stream); + } + + return 0; +} + + static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { @@ -428,6 +537,7 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { .set_sysclk = fsl_sai_set_dai_sysclk, .set_fmt = fsl_sai_set_dai_fmt, .hw_params = fsl_sai_hw_params, + .hw_free = fsl_sai_hw_free, .trigger = fsl_sai_trigger, .startup = fsl_sai_startup, .shutdown = fsl_sai_shutdown, @@ -600,8 +710,9 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->bus_clk = NULL; } - for (i = 0; i < FSL_SAI_MCLK_MAX; i++) { - sprintf(tmp, "mclk%d", i + 1); + sai->mclk_clk[0] = sai->bus_clk; + for (i = 1; i < FSL_SAI_MCLK_MAX; i++) { + sprintf(tmp, "mclk%d", i); sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 3466720..0662809 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -72,13 +72,15 @@ /* SAI Transmit and Recieve Configuration 2 Register */ #define FSL_SAI_CR2_SYNC BIT(30) -#define FSL_SAI_CR2_MSEL_MASK (0xff << 26) +#define FSL_SAI_CR2_MSEL_MASK (0x3 << 26) #define FSL_SAI_CR2_MSEL_BUS 0 #define FSL_SAI_CR2_MSEL_MCLK1 BIT(26) #define FSL_SAI_CR2_MSEL_MCLK2 BIT(27) #define FSL_SAI_CR2_MSEL_MCLK3 (BIT(26) | BIT(27)) +#define FSL_SAI_CR2_MSEL(ID) ((ID) << 26) #define FSL_SAI_CR2_BCP BIT(25) #define FSL_SAI_CR2_BCD_MSTR BIT(24) +#define FSL_SAI_CR2_DIV_MASK 0xff /* SAI Transmit and Recieve Configuration 3 Register */ #define FSL_SAI_CR3_TRCE BIT(16) @@ -120,7 +122,7 @@ #define FSL_SAI_CLK_MAST2 2 #define FSL_SAI_CLK_MAST3 3 -#define FSL_SAI_MCLK_MAX 3 +#define FSL_SAI_MCLK_MAX 4 /* SAI data transfer numbers per DMA request */ #define FSL_SAI_MAXBURST_TX 6 @@ -132,11 +134,14 @@ struct fsl_sai { struct clk *bus_clk; struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; + bool is_slave_mode; bool is_lsb_first; bool is_dsp_mode; bool sai_on_imx; bool synchronous[2]; + unsigned int mclk_id[2]; + unsigned int mclk_streams; struct snd_dmaengine_dai_dma_data dma_params_rx; struct snd_dmaengine_dai_dma_data dma_params_tx; }; -- 1.9.1

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[alsa-devel] [PATCH v2 00/17] new locomo driver set
by Dmitry Eremin-Solenikov 13 May '15

13 May '15

Sharp Zaurus SL-5500 and SL-5600 use special companion Gate Array. With this patchset I tried to modernise and restructure the LoCoMo drivers. * Use platform bus and mfd core code * Use GPIO API instead of custom locomo GPIO code * Use irqdomains to manage IRQs * Split mixed locomolcd driver to lcd and backlight parts * Implement IRQ handling in GPIO driver * Implement SPI driver used as a host for mmc_spi device to control SD cards V1 was submitted about a half of a year ago. Major changes: * Use gpiod_* in most of places * Use regmap for accessing registers instead of manual RMW cycles * Split dac driver from core module to i2c bus and separate IIO driver * Replaced mdelay() calls with usleep_range or msleep where it was suitable * Rewrote SPI driver to use core transfer_one() interface

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Re: [alsa-devel] [PATCH 3/4] ASoC: rockchip: add rockchip machine driver for rt5650/rt5645
by Mark Brown 12 May '15

12 May '15

On Tue, May 12, 2015 at 05:26:36PM +0800, Xing Zheng wrote: > From: zhengxing <zhengxing(a)rock-chips.com> > > The driver is used for rockchip board using a rt5650/rt5645. > Test on RK3288 with rt5650. Similar comments to the Maxim driver plus... > +/* Jack detect via rt5645 driver. */ > +extern int rt5645_set_jack_detect(struct snd_soc_codec *codec, > + struct snd_soc_jack *hp_jack, > + struct snd_soc_jack *mic_jack, > + struct snd_soc_jack *btn_jack); > + > +/* For compatible. */ > +int __weak rt5645_set_jack_detect(struct snd_soc_codec *codec, > + struct snd_soc_jack *hp_jack, > + struct snd_soc_jack *mic_jack, > + struct snd_soc_jack *btn_jack) > +{ > + dev_warn(codec->dev, "Jack detect dummy, have no implement it?\n"); > + > + return 0; > +} Don't do this - if the functionality isn't there in the CODEC driver get it implemented before you use it. Make sure you're using the API the CODEC actually ends up exporting. > + /* Enable Headset and 4 Buttons Jack detection */ > + ret = snd_soc_card_jack_new(card, "Headphone Jack", > + SND_JACK_HEADPHONE | SND_JACK_MICROPHONE, > + ret = snd_soc_card_jack_new(card, "Mic Jack", > + > + ret = snd_soc_card_jack_new(card, "Button Jack", No, don't do this - as discussed in another Chromebook review (I'm guessing this is a Chromebook) the jacks should represent the physical jacks in the system. It's possible you have separate headphone and microphone jacks but I really expect microphone and button to be on the same socket.

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Re: [alsa-devel] [PATCH 2/4] ASoC: rockchip: add rockchip machine driver for max98090
by Mark Brown 12 May '15

12 May '15

On Tue, May 12, 2015 at 05:26:35PM +0800, Xing Zheng wrote: > sound/soc/rockchip/Kconfig | 10 ++ > sound/soc/rockchip/Makefile | 2 + > sound/soc/rockchip/rockchip_machine_driver.c | 6 + > sound/soc/rockchip/rockchip_machine_driver.h | 6 + > sound/soc/rockchip/rockchip_max98090.c | 185 ++++++++++++++++++++++++++ This looks more like a normal and reasonable machine driver but then why have you created the generic rockchip machine driver? It seems like this should just be a regular machine driver like other platforms have, were it not for that this would be mostly fine apart from a couple of nitpicks below. > +config SND_SOC_ROCKCHIP_MAX98090 > + tristate "ASoC support for Rockchip boards using a MAX98090 codec" > + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB > + select SND_SOC_ROCKCHIP_I2S > + select SND_SOC_MAX98090 > + select SND_SOC_TS3A227E This looks like it's a driver specific to Chromebooks (possibly even specific Chromeooks) and so should be named as such. > + card->dapm.idle_bias_off = true; Just set this when declaring the card, don't do it at runtime. > + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); > + snd_soc_dapm_enable_pin(dapm, "Headphone"); > + snd_soc_dapm_enable_pin(dapm, "Speaker"); > + snd_soc_dapm_enable_pin(dapm, "Int Mic"); No need to do this, all pins are enabled by default. > + snd_soc_dapm_sync(dapm); This has no effect during initialization.

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Re: [alsa-devel] [PATCH 1/4] ASoC: rockchip: add rockchip machine driver
by Mark Brown 12 May '15

12 May '15

On Tue, May 12, 2015 at 05:26:34PM +0800, Xing Zheng wrote: > From: zhengxing <zhengxing(a)rock-chips.com> > > In most cases, we maybe use simple-card as generic machine driver > on next kernel, but there are some issue that jack detection and > widgets extandable on simple-card. > > Own machine drvier can select one valid codec from supported codecs, > and add needful function from dts on rockchip platform. Is it not possible to extend simple card to handle your use cases? Given the very generic naming and the fact that things like jack detection and so on should depend on the CODEC and board usually rather than on the SoC it doesn't sound like this is Rockchip specific. This also looks like you're reimplementing some device model enumeration stuff which probably shouldn't be happening but let's understand the problem you're trying to solve here before going too far into the code.

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[alsa-devel] [PATCH 1/4] ASoC: wm8960: Let wm8960 driver configure its bit clock and frame clock
by Zidan Wang 12 May '15

12 May '15

wm8960 codec driver missing configure its bit clock and frame clock for codec master mode, so add support for it. It will calculate a appropriate frequency dividing ratio according to the system clock, bit clock and frame clock, then set the corresponding registers. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 101 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 101 insertions(+) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 6fa832b..b072501 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -127,6 +127,8 @@ struct wm8960_priv { struct snd_soc_dapm_widget *out3; bool deemph; int playback_fs; + int bclk; + int sysclk; struct wm8960_data pdata; }; @@ -563,6 +565,72 @@ static struct { { 8000, 5 }, }; +/* Multiply 256 for internal 256 div */ +static const int dac_divs[] = { 256, 384, 512, 768, 1024, 1408, 1536 }; + +/* Multiply 10 to eliminate decimials */ +static const int bclk_divs[] = { + 10, 15, 20, 30, 40, 55, 60, 80, 110, + 120, 160, 220, 240, 320, 320, 320 +}; + +static void wm8960_configure_clocking(struct snd_soc_codec *codec, + bool tx, int lrclk) +{ + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + u16 iface1 = snd_soc_read(codec, WM8960_IFACE1); + u16 iface2 = snd_soc_read(codec, WM8960_IFACE2); + u32 sysclk; + int i, j; + + if (!(iface1 & (1<<6))) { + dev_dbg(codec->dev, + "Codec is slave mode, no need to configure clock\n"); + return; + } + + if (!wm8960->sysclk) { + dev_dbg(codec->dev, "No SYSCLK configured\n"); + return; + } + + if (!wm8960->bclk || !lrclk) { + dev_dbg(codec->dev, "No audio clocks configured\n"); + return; + } + + for (i = 0; i < ARRAY_SIZE(dac_divs); ++i) { + if (wm8960->sysclk == lrclk * dac_divs[i]) { + for (j = 0; j < ARRAY_SIZE(bclk_divs); ++j) { + sysclk = wm8960->bclk * bclk_divs[j] / 10; + if (wm8960->sysclk == sysclk) + break; + } + if(j != ARRAY_SIZE(bclk_divs)) + break; + } + } + + if (i == ARRAY_SIZE(dac_divs)) { + dev_err(codec->dev, "Unsupported sysclk %d\n", wm8960->sysclk); + return; + } + + /* + * configure frame clock. If ADCLRC configure as GPIO pin, DACLRC + * pin is used as a frame clock for ADCs and DACs. + */ + if (iface2 & (1<<6)) + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); + else if (tx) + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); + else if (!tx) + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, i << 6); + + /* configure bit clock */ + snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, j); +} + static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -570,8 +638,13 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int i; + wm8960->bclk = snd_soc_params_to_bclk(params); + if (params_channels(params) == 1) + wm8960->bclk *= 2; + /* bit size */ switch (params_width(params)) { case 16: @@ -602,6 +675,9 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, /* set iface */ snd_soc_write(codec, WM8960_IFACE1, iface); + + wm8960_configure_clocking(codec, tx, params_rate(params)); + return 0; } @@ -946,6 +1022,30 @@ static int wm8960_set_bias_level(struct snd_soc_codec *codec, return wm8960->set_bias_level(codec, level); } +static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case WM8960_SYSCLK_MCLK: + snd_soc_update_bits(codec, WM8960_CLOCK1, + 0x1, WM8960_SYSCLK_MCLK); + break; + case WM8960_SYSCLK_PLL: + snd_soc_update_bits(codec, WM8960_CLOCK1, + 0x1, WM8960_SYSCLK_PLL); + break; + default: + return -EINVAL; + } + + wm8960->sysclk = freq; + + return 0; +} + #define WM8960_RATES SNDRV_PCM_RATE_8000_48000 #define WM8960_FORMATS \ @@ -958,6 +1058,7 @@ static const struct snd_soc_dai_ops wm8960_dai_ops = { .set_fmt = wm8960_set_dai_fmt, .set_clkdiv = wm8960_set_dai_clkdiv, .set_pll = wm8960_set_dai_pll, + .set_sysclk = wm8960_set_dai_sysclk, }; static struct snd_soc_dai_driver wm8960_dai = { -- 1.9.1

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Re: [alsa-devel] [PATCH v2] ASoC: bt-sco: Add devicetree support for bt-sco codec
by Mark Brown 12 May '15

12 May '15

On Fri, May 08, 2015 at 09:02:34PM +0200, Marek Belisko wrote: > Add devicetree support for bluetooth SCO link codec. Applied with a bit of effort, you seem to have sent this against some non-mainline tree - please check that in future.

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Re: [alsa-devel] [PATCH 1/4] ASoC: rt5645: improve headphone depop function
by Mark Brown 12 May '15

12 May '15

On Tue, May 12, 2015 at 08:43:02PM +0800, John Lin wrote: > We add a calibration function and call it at the beginning of i2c_probe. > The calibration value will be kept until codec is shutdown. We will reset > the codec after the calibration is finished. So, we set cache_bypass in > the calibration function. The benefit is we can shorter the delay time > in headphone depop. Applied all, thanks.

1 0

[alsa-devel] [PATCH 1/5] ASoC: wm8996: Pass the IRQF_ONESHOT flag
by Fabio Estevam 12 May '15

12 May '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Since commit 1c6c69525b40eb76de8adf039409722015927dc3 ("genirq: Reject bogus threaded irq requests") threaded IRQs without a primary handler need to be requested with IRQF_ONESHOT, otherwise the request will fail. So pass the IRQF_ONESHOT flag in this case. The semantic patch that makes this change is available in scripts/coccinelle/misc/irqf_oneshot.cocci. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm8996.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 3dce507..232f234 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2644,10 +2644,12 @@ static int wm8996_probe(struct snd_soc_codec *codec) if (irq_flags & (IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING)) ret = request_threaded_irq(i2c->irq, NULL, wm8996_edge_irq, - irq_flags, "wm8996", codec); + irq_flags | IRQF_ONESHOT, + "wm8996", codec); else ret = request_threaded_irq(i2c->irq, NULL, wm8996_irq, - irq_flags, "wm8996", codec); + irq_flags | IRQF_ONESHOT, + "wm8996", codec); if (ret == 0) { /* Unmask the interrupt */ -- 1.9.1

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[alsa-devel] [PATCH 0/3] Add sai master mode, tdm slot operation and add some sample rate support
by Zidan Wang 12 May '15

12 May '15

Add SAI master mode support. Add tdm slots operation for SAI master mode. Add 12kHz, 24kHz, 176.4kHz and 192kHz sample rate support. Zidan Wang (3): SoC: fsl_sai: add sai master mode support ASoC: fsl_sai: Add tdm slots operation for SAI master mode ASoC: fsl_sai: add 12kHz, 24kHz, 176.4kHz and 192kHz sample rate support sound/soc/fsl/fsl_sai.c | 165 +++++++++++++++++++++++++++++++++++++++++++++--- sound/soc/fsl/fsl_sai.h | 12 +++- 2 files changed, 165 insertions(+), 12 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: dapm: Modify widget stream name according to prefix
by Koro Chen 12 May '15

12 May '15

When there is prefix specified, currently we will add this prefix in widget->name, but not in widget->sname. it causes failure at snd_soc_dapm_link_dai_widgets: if (!w->sname || !strstr(w->sname, dai_w->name)) because dai_w->name has prefix added, but w->sname does not. We should also add prefix for stream name Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/soc-dapm.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index defe0f0..158204d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3100,11 +3100,16 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, } prefix = soc_dapm_prefix(dapm); - if (prefix) + if (prefix) { w->name = kasprintf(GFP_KERNEL, "%s %s", prefix, widget->name); - else + if (widget->sname) + w->sname = kasprintf(GFP_KERNEL, "%s %s", prefix, + widget->sname); + } else { w->name = kasprintf(GFP_KERNEL, "%s", widget->name); - + if (widget->sname) + w->sname = kasprintf(GFP_KERNEL, "%s", widget->sname); + } if (w->name == NULL) { kfree(w); return NULL; -- 1.8.1.1.dirty

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[alsa-devel] [PATCH] ASoC: codecs-ac97: Remove rate constraints
by Maciej S. Szmigiero 12 May '15

12 May '15

Remove rate constraints from generic ASoC AC'97 CODEC. Supported rates should be detected and constrained anyway by AC'97 generic code - was tested with VT1613 CODEC and iMX6 SSI controller. This way this driver can be used for platforms which don't need specialized AC'97 CODEC drivers while at the same avoiding code duplication from implementing equivalent functionality in a controller driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -44,10 +44,6 @@ static int ac97_prepare(struct snd_pcm_substream *substream, return snd_ac97_set_rate(ac97, reg, substream->runtime->rate); } -#define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000) - static const struct snd_soc_dai_ops ac97_dai_ops = { .prepare = ac97_prepare, }; @@ -58,13 +54,13 @@ static struct snd_soc_dai_driver ac97_dai = { .stream_name = "AC97 Playback", .channels_min = 1, .channels_max = 2, - .rates = STD_AC97_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SND_SOC_STD_AC97_FMTS,}, .capture = { .stream_name = "AC97 Capture", .channels_min = 1, .channels_max = 2, - .rates = STD_AC97_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &ac97_dai_ops, };

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[alsa-devel] [PATCH v2 1/2] ASoC: sta32x: Use devm_gpiod_get_optional at appropriate place
by Axel Lin 12 May '15

12 May '15

devm_gpiod_get_optional() is equivalent to devm_gpiod_get(), except that when no GPIO was assigned to the requested function it will return NULL. This is convenient for drivers that need to handle optional GPIOs. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- v2: Fixup subject line sound/soc/codecs/sta32x.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 033b7d9..847fb20 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -1095,16 +1095,10 @@ static int sta32x_i2c_probe(struct i2c_client *i2c, #endif /* GPIOs */ - sta32x->gpiod_nreset = devm_gpiod_get(dev, "reset"); - if (IS_ERR(sta32x->gpiod_nreset)) { - ret = PTR_ERR(sta32x->gpiod_nreset); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta32x->gpiod_nreset = NULL; - } else { - gpiod_direction_output(sta32x->gpiod_nreset, 0); - } + sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(sta32x->gpiod_nreset)) + return PTR_ERR(sta32x->gpiod_nreset); /* regulators */ for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) -- 2.1.0

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Re: [alsa-devel] commit b1ef29725865 (ACPI _REV=2) causes sound regression on Dell XPS 13 [Was: Discussion around quirking the _REV behavior for the XPS 13 (2015) until 4.2]
by Mario Limonciello 12 May '15

12 May '15

On 05/12/2015 12:43 PM, Matthew Garrett wrote: > On Tue, May 12, 2015 at 10:29 AM, Mario Limonciello > <mario_limonciello(a)dell.com> wrote: >> Just to be clear, from the Dell side only the XPS 13 9343 has this _REV >> behavior. > The Inspiron 7437 queries _REV and uses it to modify its EC behaviour, > and apparently breaks on Linux without that. Thanks for bringing this to my attention. I'll check with my BIOS architecture team on this. XPS 13 9343 was supposed to be the only system doing this. >> What's wrong with a DMI quirk until I2S is mature on the kernel side and >> userspace is new enough in distros? Is it too early in boot for DMI quirks? >> That would enforce that no other machines can use _REV value of 5 to detect >> Linux without breaking the 9343. > DMI quirking is fine, but it has to be behind a config option. We're > going to have to carry the quirk for several years because we have to > support old userspace for an arbitrarily long time, and if there's no > config option then Linux will be stuck in HDA mode for that entire > time. > OK, that makes more sense to me. Thanks.

1 0

Re: [alsa-devel] commit b1ef29725865 (ACPI _REV=2) causes sound regression on Dell XPS 13 [Was: Discussion around quirking the _REV behavior for the XPS 13 (2015) until 4.2]
by Mario Limonciello 12 May '15

12 May '15

On 05/12/2015 12:21 PM, Matthew Garrett wrote: > On Tue, May 12, 2015 at 9:18 AM, Mark Brown <broonie(a)kernel.org> wrote: >> You need PulseAudio 6.0 but otherwise yes, it works for me. > Since there's no way to get this working without using new userspace, > we're basically stuck with having to support _REV 5. And since we're > supposed to support old userspace basically forever, this isn't going > to be something that we can just remove in a few months. But since the > hardware is intended to be used in I2S mode, we don't want this quirk > to be there forever. So we're basically left with adding > CONFIG_ACPI_WORKAROUND_BROKEN_DELLS which enables the quirk, default > it to Y and then let distributions set it to N when they've got > sufficiently new userspace. Unless someone has a better plan? Just to be clear, from the Dell side only the XPS 13 9343 has this _REV behavior. What's wrong with a DMI quirk until I2S is mature on the kernel side and userspace is new enough in distros? Is it too early in boot for DMI quirks? That would enforce that no other machines can use _REV value of 5 to detect Linux without breaking the 9343.

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Re: [alsa-devel] commit b1ef29725865 (ACPI _REV=2) causes sound regression on Dell XPS 13 [Was: Discussion around quirking the _REV behavior for the XPS 13 (2015) until 4.2]
by Mark Brown 12 May '15

12 May '15

On Tue, May 12, 2015 at 10:21:41AM -0700, Matthew Garrett wrote: > Since there's no way to get this working without using new userspace, > we're basically stuck with having to support _REV 5. And since we're > supposed to support old userspace basically forever, this isn't going > to be something that we can just remove in a few months. But since the > hardware is intended to be used in I2S mode, we don't want this quirk > to be there forever. So we're basically left with adding > CONFIG_ACPI_WORKAROUND_BROKEN_DELLS which enables the quirk, default > it to Y and then let distributions set it to N when they've got > sufficiently new userspace. Unless someone has a better plan? That makes sense to me given that the firmware doesn't seem to do anything problematic in Linux mode - hopefully we won't have too many more of these systems to contend with.

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Re: [alsa-devel] [PATCH v2 02/17] leds: port locomo leds driver to new locomo core
by Dmitry Eremin-Solenikov 12 May '15

12 May '15

2015-05-06 18:05 GMT+03:00 Jacek Anaszewski <j.anaszewski(a)samsung.com>: > On 04/28/2015 01:55 AM, Dmitry Eremin-Solenikov wrote: >> >> Adapt locomo leds driver to new locomo core setup. >> >> Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> >> --- >> drivers/leds/Kconfig | 1 - >> drivers/leds/leds-locomo.c | 119 >> +++++++++++++++++++++++---------------------- >> 2 files changed, 61 insertions(+), 59 deletions(-) >> >> diff --git a/drivers/leds/Kconfig b/drivers/leds/Kconfig >> index 966b960..4b4650b 100644 >> --- a/drivers/leds/Kconfig >> +++ b/drivers/leds/Kconfig >> @@ -79,7 +79,6 @@ config LEDS_LM3642 >> config LEDS_LOCOMO >> tristate "LED Support for Locomo device" >> depends on LEDS_CLASS >> - depends on SHARP_LOCOMO > > > Why do you remove this dependency? Because SHARP_LOCOMO is a Kconfig symbol for the old driver. New driver uses MFD_LOCOMO Kconfig entry. Also the driver now uses generic platform device and regmap interfaces, so there is no direct dependency on main LoCoMo driver. And the policy (IIRC) was not to have such dependencies. > >> help >> This option enables support for the LEDs on Sharp Locomo. >> Zaurus models SL-5500 and SL-5600. >> diff --git a/drivers/leds/leds-locomo.c b/drivers/leds/leds-locomo.c >> index 80ba048..7fde812 100644 >> --- a/drivers/leds/leds-locomo.c >> +++ b/drivers/leds/leds-locomo.c >> @@ -9,89 +9,92 @@ >> */ >> >> #include <linux/kernel.h> >> -#include <linux/init.h> >> -#include <linux/module.h> >> -#include <linux/device.h> >> #include <linux/leds.h> >> - >> -#include <mach/hardware.h> >> -#include <asm/hardware/locomo.h> >> +#include <linux/module.h> >> +#include <linux/platform_device.h> >> +#include <linux/regmap.h> >> +#include <linux/mfd/locomo.h> > > > Please keep alphabetical order. Ack > >> + >> +struct locomo_led { >> + struct led_classdev led; >> + struct regmap *regmap; >> + unsigned int reg; >> +}; >> >> static void locomoled_brightness_set(struct led_classdev *led_cdev, >> - enum led_brightness value, int offset) >> + enum led_brightness value) >> { >> - struct locomo_dev *locomo_dev = LOCOMO_DEV(led_cdev->dev->parent); >> - unsigned long flags; >> + struct locomo_led *led = container_of(led_cdev, struct locomo_led, >> led); >> >> - local_irq_save(flags); >> - if (value) >> - locomo_writel(LOCOMO_LPT_TOFH, locomo_dev->mapbase + >> offset); >> - else >> - locomo_writel(LOCOMO_LPT_TOFL, locomo_dev->mapbase + >> offset); >> - local_irq_restore(flags); >> + regmap_write(led->regmap, led->reg, >> + value ? LOCOMO_LPT_TOFH : LOCOMO_LPT_TOFL); >> } > > > Please use work queue for setting brightness. This is required for the > driver to be compatible with led triggers. You can refer to the > existing LED drivers on how to implement this. Hmm. Why? The regmap here uses MMIO access, so it is atomic operation and doesn't need to be wrapped into work queue, does it? [skipped] -- With best wishes Dmitry

1 0

[alsa-devel] ALC298 headset mode support
by Kailang 12 May '15

12 May '15

Hi Takashi, I finished the alc298 headset mode supported. But it only tested via our develop system. BR, Kailang

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[alsa-devel] [PATCH] Revert "ASoC: rsnd: make sure it uses lock when it calls rsnd_dai_call"
by Kuninori Morimoto 12 May '15

12 May '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This reverts commit 'e9c390df671f ("ASoC: rsnd: make sure it uses lock when it calls rsnd_dai_call)' The additional locks make 1") lock issue when boot 2) lock issue when unbind/rmmod. And there is no problem without these locks. This patch revert it. Reported-by: Geert Uytterhoeven <geert+renesas(a)glider.be> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 27 +++++---------------------- 1 file changed, 5 insertions(+), 22 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 99eb109..405cacd 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -731,15 +731,10 @@ static int rsnd_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_dai *dai = rsnd_substream_to_dai(substream); struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); - struct rsnd_priv *priv = rsnd_dai_to_priv(dai); struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); - unsigned long flags; int ret; - rsnd_lock(priv, flags); ret = rsnd_dai_call(hw_params, io, substream, hw_params); - rsnd_unlock(priv, flags); - if (ret) return ret; @@ -926,16 +921,14 @@ int rsnd_kctrl_new_e(struct rsnd_mod *mod, static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *dai = rtd->cpu_dai; - struct rsnd_priv *priv = rsnd_dai_to_priv(dai); struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); - unsigned long flags; - int ret = 0; + int ret; - rsnd_lock(priv, flags); - ret |= rsnd_dai_call(pcm_new, &rdai->playback, rtd); - ret |= rsnd_dai_call(pcm_new, &rdai->capture, rtd); - rsnd_unlock(priv, flags); + ret = rsnd_dai_call(pcm_new, &rdai->playback, rtd); + if (ret) + return ret; + ret = rsnd_dai_call(pcm_new, &rdai->capture, rtd); if (ret) return ret; @@ -958,11 +951,8 @@ static const struct snd_soc_component_driver rsnd_soc_component = { static int rsnd_rdai_continuance_probe(struct rsnd_priv *priv, struct rsnd_dai_stream *io) { - unsigned long flags; int ret; - rsnd_lock(priv, flags); - ret = rsnd_dai_call(probe, io, priv); if (ret == -EAGAIN) { /* @@ -995,7 +985,6 @@ static int rsnd_rdai_continuance_probe(struct rsnd_priv *priv, */ ret = rsnd_dai_call(probe, io, priv); } - rsnd_unlock(priv, flags); return ret; } @@ -1011,7 +1000,6 @@ static int rsnd_probe(struct platform_device *pdev) struct rsnd_dai *rdai; const struct of_device_id *of_id = of_match_device(rsnd_of_match, dev); const struct rsnd_of_data *of_data; - unsigned long flags; int (*probe_func[])(struct platform_device *pdev, const struct rsnd_of_data *of_data, struct rsnd_priv *priv) = { @@ -1098,12 +1086,10 @@ static int rsnd_probe(struct platform_device *pdev) exit_snd_soc: snd_soc_unregister_platform(dev); exit_snd_probe: - rsnd_lock(priv, flags); for_each_rsnd_dai(rdai, priv, i) { rsnd_dai_call(remove, &rdai->playback, priv); rsnd_dai_call(remove, &rdai->capture, priv); } - rsnd_unlock(priv, flags); return ret; } @@ -1112,7 +1098,6 @@ static int rsnd_remove(struct platform_device *pdev) { struct rsnd_priv *priv = dev_get_drvdata(&pdev->dev); struct rsnd_dai *rdai; - unsigned long flags; void (*remove_func[])(struct platform_device *pdev, struct rsnd_priv *priv) = { rsnd_ssi_remove, @@ -1123,12 +1108,10 @@ static int rsnd_remove(struct platform_device *pdev) pm_runtime_disable(&pdev->dev); - rsnd_lock(priv, flags); for_each_rsnd_dai(rdai, priv, i) { ret |= rsnd_dai_call(remove, &rdai->playback, priv); ret |= rsnd_dai_call(remove, &rdai->capture, priv); } - rsnd_unlock(priv, flags); for (i = 0; i < ARRAY_SIZE(remove_func); i++) remove_func[i](pdev, priv); -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ASoC: dapm: Add cache to speed up adding of routes
by Charles Keepax 12 May '15

12 May '15

Some CODECs have a significant number of DAPM routes and for each route, when it is added to the card, the entire card widget list must be searched. When adding routes it is very likely, however, that adjacent routes will require adjacent widgets. For example all the routes for a mux are likely added in a block and the sink widget will be the same each time and it is also quite likely that the source widgets are sequential located in the widget list. This patch adds a cache to the DAPM context, this cache will hold the source and sink widgets from the last call to snd_soc_dapm_add_route for that context. A small search of the widget list will be made from those points for both the sink and source. Currently this search only checks both the last widget and the one adjacent to it. On wm8280 which has approximately 500 widgets and 30000 routes (one of the largest CODECs in mainline), the number of paths that hit the cache is 24000, which significantly improves probe time. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v1: - Provide better encapsulation of the cache, to make swapping out the implementation simpler - Put the cache in the DAPM context instead of passing explicitly to snd_soc_dapm_add_route. I went with the context over the card as it is likely that requests on the same context will hit the cache in the case we have multiple context's interleaving requests. I hope you don't mind Lars but I didn't pull your reviewed-by forward as the code has changed quite a bit for what you reviewed. Thanks, Charles include/sound/soc-dapm.h | 7 +++++++ sound/soc/soc-dapm.c | 39 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 46 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index bfc415e..2f66e1c 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -593,6 +593,10 @@ struct snd_soc_dapm_update { int val; }; +struct snd_soc_dapm_wcache { + struct snd_soc_dapm_widget *widget; +}; + /* DAPM context */ struct snd_soc_dapm_context { enum snd_soc_bias_level bias_level; @@ -614,6 +618,9 @@ struct snd_soc_dapm_context { int (*set_bias_level)(struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level); + struct snd_soc_dapm_wcache path_sink_cache; + struct snd_soc_dapm_wcache path_source_cache; + #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dapm; #endif diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7654161..327f9357 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -572,6 +572,35 @@ static void soc_dapm_async_complete(struct snd_soc_dapm_context *dapm) snd_soc_component_async_complete(dapm->component); } +static struct snd_soc_dapm_widget * +dapm_wcache_lookup(struct snd_soc_dapm_wcache *wcache, const char *name) +{ + struct snd_soc_dapm_widget *w = wcache->widget; + struct list_head *wlist; + const int depth = 2; + int i = 0; + + if (w) { + wlist = &w->dapm->card->widgets; + + list_for_each_entry_from(w, wlist, list) { + if (!strcmp(name, w->name)) + return w; + + if (++i == depth) + break; + } + } + + return NULL; +} + +static inline void dapm_wcache_update(struct snd_soc_dapm_wcache *wcache, + struct snd_soc_dapm_widget *w) +{ + wcache->widget = w; +} + /** * snd_soc_dapm_force_bias_level() - Sets the DAPM bias level * @dapm: The DAPM context for which to set the level @@ -2610,6 +2639,12 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, source = route->source; } + wsource = dapm_wcache_lookup(&dapm->path_source_cache, source); + wsink = dapm_wcache_lookup(&dapm->path_sink_cache, sink); + + if (wsink && wsource) + goto skip_search; + /* * find src and dest widgets over all widgets but favor a widget from * current DAPM context @@ -2644,6 +2679,10 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, return -ENODEV; } +skip_search: + dapm_wcache_update(&dapm->path_sink_cache, wsink); + dapm_wcache_update(&dapm->path_source_cache, wsource); + ret = snd_soc_dapm_add_path(dapm, wsource, wsink, route->control, route->connected); if (ret) -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Remove unused call to "get_pin" in patch_hdmi.c
by David Henningsson 12 May '15

12 May '15

Janitorial patch (no functional change) Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e8d8478..fd11ad7 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2049,9 +2049,7 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hda_pcm *info; struct hda_pcm_stream *pstr; - struct hdmi_spec_per_pin *per_pin; - per_pin = get_pin(spec, pin_idx); info = snd_hda_codec_pcm_new(codec, "HDMI %d", pin_idx); if (!info) return -ENOMEM; -- 1.9.1

2 2

[alsa-devel] [PATCH 0/5] ASoC: rsnd: tidyup patches
by Kuninori Morimoto 12 May '15

12 May '15

Hi Mark These are not so very important, but tidyup patches for Renesas R-Car driver. Kuninori Morimoto (5): ASoC: rsnd: add rsnd_dai_to_priv() macro ASoC: rsnd: make sure it uses lock when it calls rsnd_dai_call ASoC: rsnd: tidyup SSI parent related function/macro names ASoC: rsnd: make sure SSI parent/child uses same number of sound channel. ASoC: rsnd: care snd_kcontrol's index sound/soc/sh/rcar/core.c | 34 +++++++++++++++++++++++++++------- sound/soc/sh/rcar/ssi.c | 52 +++++++++++++++++++++++++++++++++++++++++++++------- 2 files changed, 72 insertions(+), 14 deletions(-) Best regards --- Kuninori Morimoto

3 11

[alsa-devel] [PATCH v2] ASoC: wm8741: Add differential mono mode support
by Sergej Sawazki 11 May '15

11 May '15

The WM8741 DAC supports several differential output modes (stereo, stereo reversed, mono left, mono right). Add platform data and DT bindings to configure it. Signed-off-by: Sergej Sawazki <ce3a(a)gmx.de> --- Documentation/devicetree/bindings/sound/wm8741.txt | 11 +++ sound/soc/codecs/wm8741.c | 108 ++++++++++++++++++--- sound/soc/codecs/wm8741.h | 10 ++ 3 files changed, 117 insertions(+), 12 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/wm8741.txt b/Documentation/devicetree/bindings/sound/wm8741.txt index 74bda58..a133154 100644 --- a/Documentation/devicetree/bindings/sound/wm8741.txt +++ b/Documentation/devicetree/bindings/sound/wm8741.txt @@ -10,9 +10,20 @@ Required properties: - reg : the I2C address of the device for I2C, the chip select number for SPI. +Optional properties: + + - diff-mode: Differential output mode configuration. Default value for field + DIFF in register R8 (MODE_CONTROL_2). If absent, the default is 0, shall be: + 0 = stereo + 1 = mono left + 2 = stereo reversed + 3 = mono right + Example: codec: wm8741@1a { compatible = "wlf,wm8741"; reg = <0x1a>; + + diff-mode = <3>; }; diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 9e71c76..cbf90ab 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -41,6 +41,7 @@ static const char *wm8741_supply_names[WM8741_NUM_SUPPLIES] = { /* codec private data */ struct wm8741_priv { + struct wm8741_platform_data pdata; struct regmap *regmap; struct regulator_bulk_data supplies[WM8741_NUM_SUPPLIES]; unsigned int sysclk; @@ -398,7 +399,7 @@ static struct snd_soc_dai_driver wm8741_dai = { .name = "wm8741", .playback = { .stream_name = "Playback", - .channels_min = 2, /* Mono modes not yet supported */ + .channels_min = 2, .channels_max = 2, .rates = WM8741_RATES, .formats = WM8741_FORMATS, @@ -416,6 +417,60 @@ static int wm8741_resume(struct snd_soc_codec *codec) #define wm8741_resume NULL #endif +static int wm8741_configure(struct snd_soc_codec *codec) +{ + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + + /* Configure differential mode */ + switch (wm8741->pdata.diff_mode) { + case WM8741_DIFF_MODE_STEREO: + case WM8741_DIFF_MODE_STEREO_REVERSED: + case WM8741_DIFF_MODE_MONO_LEFT: + case WM8741_DIFF_MODE_MONO_RIGHT: + snd_soc_update_bits(codec, WM8741_MODE_CONTROL_2, + WM8741_DIFF_MASK, + wm8741->pdata.diff_mode << WM8741_DIFF_SHIFT); + break; + default: + return -EINVAL; + } + + /* Change some default settings - latch VU */ + snd_soc_update_bits(codec, WM8741_DACLLSB_ATTENUATION, + WM8741_UPDATELL, WM8741_UPDATELL); + snd_soc_update_bits(codec, WM8741_DACLMSB_ATTENUATION, + WM8741_UPDATELM, WM8741_UPDATELM); + snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, + WM8741_UPDATERL, WM8741_UPDATERL); + snd_soc_update_bits(codec, WM8741_DACRMSB_ATTENUATION, + WM8741_UPDATERM, WM8741_UPDATERM); + + return 0; +} + +static int wm8741_add_controls(struct snd_soc_codec *codec) +{ + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + + switch (wm8741->pdata.diff_mode) { + case WM8741_DIFF_MODE_STEREO: + case WM8741_DIFF_MODE_STEREO_REVERSED: + snd_soc_add_codec_controls(codec, wm8741_snd_controls, + ARRAY_SIZE(wm8741_snd_controls)); + break; + case WM8741_DIFF_MODE_MONO_LEFT: + case WM8741_DIFF_MODE_MONO_RIGHT: + /* The machine driver is responsible for mixer controls + * if the codec is configured in differential mono mode. + */ + break; + default: + return -EINVAL; + } + + return 0; +} + static int wm8741_probe(struct snd_soc_codec *codec) { struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); @@ -434,15 +489,17 @@ static int wm8741_probe(struct snd_soc_codec *codec) goto err_enable; } - /* Change some default settings - latch VU */ - snd_soc_update_bits(codec, WM8741_DACLLSB_ATTENUATION, - WM8741_UPDATELL, WM8741_UPDATELL); - snd_soc_update_bits(codec, WM8741_DACLMSB_ATTENUATION, - WM8741_UPDATELM, WM8741_UPDATELM); - snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, - WM8741_UPDATERL, WM8741_UPDATERL); - snd_soc_update_bits(codec, WM8741_DACRMSB_ATTENUATION, - WM8741_UPDATERM, WM8741_UPDATERM); + ret = wm8741_configure(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to change default settings\n"); + goto err_enable; + } + + ret = wm8741_add_controls(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to add controls\n"); + goto err_enable; + } dev_dbg(codec->dev, "Successful registration\n"); return ret; @@ -467,8 +524,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8741 = { .remove = wm8741_remove, .resume = wm8741_resume, - .controls = wm8741_snd_controls, - .num_controls = ARRAY_SIZE(wm8741_snd_controls), .dapm_widgets = wm8741_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8741_dapm_widgets), .dapm_routes = wm8741_dapm_routes, @@ -493,6 +548,23 @@ static const struct regmap_config wm8741_regmap = { .readable_reg = wm8741_readable, }; +static int wm8741_set_pdata(struct device *dev, struct wm8741_priv *wm8741) +{ + const struct wm8741_platform_data *pdata = dev_get_platdata(dev); + u32 diff_mode; + + if (dev->of_node) { + if (of_property_read_u32(dev->of_node, "diff-mode", &diff_mode) + >= 0) + wm8741->pdata.diff_mode = diff_mode; + } else { + if (pdata != NULL) + memcpy(&wm8741->pdata, pdata, sizeof(wm8741->pdata)); + } + + return 0; +} + #if IS_ENABLED(CONFIG_I2C) static int wm8741_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -522,6 +594,12 @@ static int wm8741_i2c_probe(struct i2c_client *i2c, return ret; } + wm8741_set_pdata(&i2c->dev, wm8741); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to set pdata: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, wm8741); ret = snd_soc_register_codec(&i2c->dev, @@ -582,6 +660,12 @@ static int wm8741_spi_probe(struct spi_device *spi) return ret; } + wm8741_set_pdata(&spi->dev, wm8741); + if (ret != 0) { + dev_err(&spi->dev, "Failed to set pdata: %d\n", ret); + return ret; + } + spi_set_drvdata(spi, wm8741); ret = snd_soc_register_codec(&spi->dev, diff --git a/sound/soc/codecs/wm8741.h b/sound/soc/codecs/wm8741.h index 56c1b1d..c8835f6 100644 --- a/sound/soc/codecs/wm8741.h +++ b/sound/soc/codecs/wm8741.h @@ -194,6 +194,12 @@ #define WM8741_DITHER_SHIFT 0 /* DITHER - [1:0] */ #define WM8741_DITHER_WIDTH 2 /* DITHER - [1:0] */ +/* DIFF field values */ +#define WM8741_DIFF_MODE_STEREO 0 /* stereo normal */ +#define WM8741_DIFF_MODE_STEREO_REVERSED 2 /* stereo reversed */ +#define WM8741_DIFF_MODE_MONO_LEFT 1 /* mono left */ +#define WM8741_DIFF_MODE_MONO_RIGHT 3 /* mono right */ + /* * R32 (0x20) - ADDITONAL_CONTROL_1 */ @@ -208,4 +214,8 @@ #define WM8741_SYSCLK 0 +struct wm8741_platform_data { + u32 diff_mode; /* Differential Output Mode */ +}; + #endif -- 1.9.1

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[alsa-devel] Atmel_usba_udc: Kernel panic in usba_ep_enable
by Angelo Compagnucci 11 May '15

11 May '15

Hello List, I'm having troubles with usb audio on an SAMA5D3 on 3.19.8 with a strange kernel panic. The function f_audio_set_alt (f_uac1.c) is passing a NULL out_ep and so the usba_ep_enable crashes with a null pointer dereference as you can see from the debugging output: [ 24.210000] ep1: ep_enable: desc= (null) Unfortunately, there is nothing strange with my configuration, I'm trying to use the soundcard (which plays files flawlessly) as a usb soundcard, so the only thing I did was to compile g_audio as a module, enable UAC1 and did a modprode. Can you share any advice? Following kernel panic output: [ 23.500000] g_audio gadget: adding config #1 'Linux USB Audio Gadget'/bf0232c8 [ 23.510000] g_audio gadget: adding 'g_audio'/cf9c6400 to config 'Linux USB Audio Gadget'/bf0232c8 [ 23.600000] g_audio gadget: Hardware params: access 3, format 2, channels 2, rate 48000 [ 23.630000] g_audio gadget: cfg 1/bf0232c8 speeds: high full [ 23.630000] g_audio gadget: interface 0 = g_audio/cf9c6400 [ 23.640000] g_audio gadget: interface 1 = g_audio/cf9c6400 [ 23.640000] g_audio gadget: Linux USB Audio Gadget, version: Feb 2, 2012 [ 23.650000] g_audio gadget: g_audio ready [ 24.200000] g_audio gadget: high-speed config #1: Linux USB Audio Gadget [ 24.200000] g_audio gadget: intf 0, alt 0 [ 24.200000] g_audio gadget: intf 1, alt 0 [ 24.210000] g_audio gadget: intf 1, alt 1 [ 24.210000] ep1: ep_enable: desc= (null) [ 24.210000] Unable to handle kernel NULL pointer dereference at virtual address 00000002 [ 24.210000] pgd = c0004000 [ 24.210000] [00000002] *pgd=00000000 [ 24.210000] Internal error: Oops: 17 [#1] ARM [ 24.210000] Modules linked in: usb_f_uac1 g_audio libcomposite [ 24.210000] CPU: 0 PID: 0 Comm: swapper Not tainted 3.19.8+ #69 [ 24.210000] Hardware name: Atmel SAMA5 (Device Tree) [ 24.210000] task: c052c7b8 ti: c0522000 task.ti: c0522000 [ 24.210000] PC is at usba_ep_enable+0x28/0x200 [ 24.210000] LR is at usba_ep_enable+0x24/0x200 [ 24.210000] pc : [<c02815bc>] lr : [<c02815b8>] psr: 200f0193 [ 24.210000] sp : c0523e18 ip : 00000007 fp : 0000bb80 [ 24.210000] r10: 00000100 r9 : cfb17c00 r8 : 00000001 [ 24.210000] r7 : 000000c8 r6 : cf99aa10 r5 : 00000000 r4 : cf99d868 [ 24.210000] r3 : 00000006 r2 : 00000007 r1 : 00000000 r0 : 00000001 [ 24.210000] Flags: nzCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment kernel [ 24.210000] Control: 10c53c7d Table: 2fb28059 DAC: 00000015 [ 24.210000] Process swapper (pid: 0, stack limit = 0xc0522230) [ 24.210000] Stack: (0xc0523e18 to 0xc0524000) [ 24.210000] 3e00: 00000001 cf99d868 [ 24.210000] 3e20: cf9c6400 bf028694 00000001 c0530008 cfa13ac0 cfb17c00 cfb17a00 00000001 [ 24.210000] 3e40: cf9c6400 00000001 00000001 cf99aa18 00000000 bf015104 c0523e74 c0034d9c [ 24.210000] 3e60: cf9471c0 600f0193 cfb05f6c c0530704 c00398b8 c00398c4 c05306f8 c0039400 [ 24.210000] 3e80: 00000000 200f0193 00000007 00000006 00000000 cf99d810 0000000b cf99d848 [ 24.210000] 3ea0: cf99aa10 00000000 00000000 00000000 00000800 c0282f0c 0000001c ffffffff [ 24.210000] 3ec0: 7fffffff c0547d80 00000000 00000125 00000001 c054e23e 00010b01 00000001 [ 24.210000] 3ee0: 00000000 cf998240 cf802540 00000000 00000000 0000002d cf901dc0 c054e23e [ 24.210000] 3f00: 00000000 c0040af0 cf802200 c054eb00 cf901dc0 cf802540 00000000 00000000 [ 24.210000] 3f20: cf802200 c051ac50 cfffc480 c0040be0 cf901dc0 c0042e00 0000002d 0000002d [ 24.210000] 3f40: 00000000 c0040314 c0534ca0 c0040538 c000eec4 600f0013 ffffffff c0523f9c [ 24.210000] 3f60: c054e340 c0011b00 00000000 c0530918 00000000 c0017880 c0522000 c052a0c4 [ 24.210000] 3f80: c054e23c c054e23c c054e340 c051ac50 cfffc480 00000000 01000000 c0523fb0 [ 24.210000] 3fa0: c000eec0 c000eec4 600f0013 ffffffff c0522000 c0039b58 ffffffff c04f5c30 [ 24.210000] 3fc0: ffffffff ffffffff c04f5678 00000000 00000000 c051ac50 c054e514 c052a070 [ 24.210000] 3fe0: c051ac4c c052d860 20004059 410fc051 00000000 20008070 00000000 00000000 [ 24.210000] [<c02815bc>] (usba_ep_enable) from [<bf028694>] (f_audio_set_alt+0xc4/0x19c [usb_f_uac1]) [ 24.210000] [<bf028694>] (f_audio_set_alt [usb_f_uac1]) from [<bf015104>] (composite_setup+0x4e0/0x1640 [libcomposite]) [ 24.210000] [<bf015104>] (composite_setup [libcomposite]) from [<c0282f0c>] (usba_udc_irq+0x75c/0xc74) [ 24.210000] [<c0282f0c>] (usba_udc_irq) from [<c0040af0>] (handle_irq_event_percpu+0x78/0x140) [ 24.210000] [<c0040af0>] (handle_irq_event_percpu) from [<c0040be0>] (handle_irq_event+0x28/0x38) [ 24.210000] [<c0040be0>] (handle_irq_event) from [<c0042e00>] (handle_fasteoi_irq+0x9c/0x164) [ 24.210000] [<c0042e00>] (handle_fasteoi_irq) from [<c0040314>] (generic_handle_irq+0x2c/0x3c) [ 24.210000] [<c0040314>] (generic_handle_irq) from [<c0040538>] (__handle_domain_irq+0x54/0xa8) [ 24.210000] [<c0040538>] (__handle_domain_irq) from [<c0011b00>] (__irq_svc+0x40/0x54) [ 24.210000] [<c0011b00>] (__irq_svc) from [<c000eec4>] (arch_cpu_idle+0x38/0x3c) [ 24.210000] [<c000eec4>] (arch_cpu_idle) from [<c0039b58>] (cpu_startup_entry+0xa8/0xf4) [ 24.210000] [<c0039b58>] (cpu_startup_entry) from [<c04f5c30>] (start_kernel+0x374/0x380) [ 24.210000] Code: e5941004 e5946024 eb04d705 e5d40033 (e5d51002) [ 24.210000] ---[ end trace d5ac053970e7d2d1 ]--- [ 24.210000] Kernel panic - not syncing: Fatal exception in interrupt [ 24.210000] ---[ end Kernel panic - not syncing: Fatal exception in interrupt -- Profile: http://it.linkedin.com/in/compagnucciangelo

2 1

[alsa-devel] [PATCH 01/11] ASoC: 88pm860x: Replace direct snd_soc_codec dapm field access
by Lars-Peter Clausen 11 May '15

11 May '15

The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm.bias_level with snd_soc_codec_get_bias_level(). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- All patches in this series have a dependency on topic/dapm --- sound/soc/codecs/88pm860x-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index c0b2686..ee31fa7 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1140,7 +1140,7 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { /* Enable Audio PLL & Audio section */ data = AUDIO_PLL | AUDIO_SECTION_ON; pm860x_reg_write(pm860x->i2c, REG_MISC2, data); -- 1.8.0

2 11

[alsa-devel] [PATCH 0/3] ASoC: DPCM can care FE/BE merged format
by Kuninori Morimoto 11 May '15

11 May '15

Hi Mark These patches add new .dpcm_merged_foramt to merge FE/BE formats. It will be no sound if DPCM were below style, and user selects S24_LE. FE: S16_LE/S24_LE BE: S16_LE DPCM will selects FE/BE merged formats (= S16_LE here) if FE has .dpcm_merged_foramt. Kuninori Morimoto (3): ASoC: soc.h: tidyup struct snd_soc_dai_link definition order ASoC: soc-pcm: DPCM cares BE format ASoC: rsnd: rsrc-card uses FE/BE merged format when DPCM include/sound/soc.h | 39 +++++++++++++++++++++------------------ sound/soc/sh/rcar/rsrc-card.c | 1 + sound/soc/soc-pcm.c | 47 ++++++++++++++++++++++++++++++++++++++++++----- 3 files changed, 64 insertions(+), 23 deletions(-) Best regards --- Kuninori Morimoto

3 12

Re: [alsa-devel] [PATCH 2/2] sound/usb: Update ALSA driver to use media controller API
by Shuah Khan 11 May '15

11 May '15

On 05/08/2015 10:51 PM, Eric Wong wrote: > Shuah Khan <shuahkh(a)osg.samsung.com> wrote: >> @@ -541,13 +591,19 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, >> [QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk, >> [QUIRK_AUDIO_STANDARD_MIXER] = create_standard_mixer_quirk, >> }; >> + int ret; >> >> + if (quirk->media_device) { >> + /* don't want to fail when media_device_init() doesn't work */ >> + ret = media_device_init(iface); >> + } >> if (quirk->type < QUIRK_TYPE_COUNT) { >> - return quirk_funcs[quirk->type](chip, iface, driver, quirk); >> + ret = quirk_funcs[quirk->type](chip, iface, driver, quirk); >> } else { >> usb_audio_err(chip, "invalid quirk type %d\n", quirk->type); >> return -ENXIO; >> } >> + return ret; >> } > > What is the point of saving 'ret' of media_device_init if it'll > only be clobbered or ignored for ENXIO? > Agreed. There is no point in saving it. thanks, -- Shuah -- Shuah Khan Sr. Linux Kernel Developer Open Source Innovation Group Samsung Research America (Silicon Valley) shuahkh(a)osg.samsung.com | (970) 217-8978

1 0

Re: [alsa-devel] [PATCH 1/2] ASoC: rt5645: fix jack type detect error
by Mark Brown 11 May '15

11 May '15

On Mon, May 11, 2015 at 02:33:36AM +0000, John Lin wrote: > > -----Original Message----- > > From: Mark Brown [mailto:broonie@kernel.org] > > Sent: Saturday, May 09, 2015 1:14 AM > > To: John Lin > > Cc: lgirdwood(a)gmail.com; alsa-devel(a)alsa-project.org; Bard Liao; Flove; > > Oder Chiou; koro.chen(a)mediatek.com; yang.a.fang(a)intel.com; > > zhengxing(a)rock-chips.com > > Subject: Re: [PATCH 1/2] ASoC: rt5645: fix jack type detect error > > On Fri, May 08, 2015 at 01:53:33PM +0800, John Lin wrote: > > > rt5645_jack_detect doesn't report the correct jack type consistently. > > > This patch can make it more stable. > > Can you be more specific about what the problem is and how this fixes it > > pleased? It's a bit hard to review based on your changelog. > It mistakes OMTP type headset to CTIA type if "LDO2" power is disable in > jack detection function. So, we keep "LDO2" power enabled until jack out. > We also change the register sequence to make it more stable and remove > unnecessary power such as "micbias1" and "micbias2" for jack detection. This stuff needs to be in the changelog. It also sounds like there's at least two probably three patches here - one to keep the LDO power enabled for lonnger and one or more changes to the register sequence.

1 0

[alsa-devel] stereo uac device control over mixer
by Arijit De 11 May '15

11 May '15

Hi, I am working on a USB audio MIC device(capture device), implemented UAC1.0 spec inside the firmware for that, my device is a stereo uac capture? device. I am trying to enable left and write channel separate control(volume, mute etc.) over mixer interface. my understanding about how to enable channel specific control is in UAC Feature Unit Descriptor if I add bmaControls(0) and bmaControls(1) with correct bitmap for Mute & Volume support amixer is able to identify my device as a two MIC device, amixer output for my usb audio capture device ********************************** Simple mixer control 'Mic',0 Capabilities: cvolumecaps joined 780 cvolume-joined cswitch cswitch-joined Capture channels: Mono Limits: Capture 0 - 15 Mono: Capture 9 [60%] [27.00dB] [on] Simple mixer control 'Mic',1 Capabilities: cvolumecaps joined 780 cvolume-joined cswitch cswitch-joined Capture channels: Mono Limits: Capture 0 - 15 Mono: Capture 9 [60%] [27.00dB] [on] ********************************** is the correct way of achieving it OR ideally there should be a single Mic with "Front Left - Front Right" support ? Thanks Arijit

2 4

[alsa-devel] [PATCH 1/2] ASoC: fsl: fsl_dma: Use true/false for boolean init
by Fabio Estevam 11 May '15

11 May '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Bool initializations should use true and false. Bool tests don't need comparisons. Based on contributions from Joe Perches, Rusty Russell and Bruce W Allan. The semantic patch that makes this change is available in scripts/coccinelle/misc/boolinit.cocci. More information about semantic patching is available at http://coccinelle.lip6.fr/ Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 93d7e56..ccadefc 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -445,7 +445,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) return ret; } - dma->assigned = 1; + dma->assigned = true; snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware); @@ -814,7 +814,7 @@ static int fsl_dma_close(struct snd_pcm_substream *substream) substream->runtime->private_data = NULL; } - dma->assigned = 0; + dma->assigned = false; return 0; } -- 1.9.1

3 3

Re: [alsa-devel] Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 11 May '15

11 May '15

> I started MS Windows and without externals speakers plugged in I could test > 5.1 configuration and it seemed that sound was coming from different directions. > > What I managed to determine: > I've got 3 stacks: > 1 ST: Headphone, Line-out (FL and FR), SPDIF > 2 ST: Microphone, Line-out (CLFE) > 3 ST: Line-in, Line-out (RL and RR) > > I've got 5 responsive output pins > PIN 0x14: Internal FL FR > PIN 0x15: External FL FR > PIN 0x17: Internal LFE > PIN 0x18: External CLFE > PIN 0x1a: External RL RR > > How can we make it work with these information? https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/plain/Document… seem chnmap type _FIXED , _VAR and _PAIRED does not fit when LFE and FC can be swapped for internal speaker and external speaker https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… How do the driver implement channel map when codec has swap center/lfe switch ? Do chnmap contain both {FL FR RL RR FC LFE} and {FL FR RL RR LFE FC} ? https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… atihdmi_paired_swap_fc_lfe

1 0

[alsa-devel] [PATCH] ASoC: codecs-ac97: Remove rate constraints
by Maciej S. Szmigiero 10 May '15

10 May '15

Remove rate constraints from generic ASoC AC'97 CODEC and make it selectable in config. Supported rates should be detected and constrained anyway by AC'97 generic code - was tested with VT1613 CODEC and iMX6 SSI controller. This way this driver can be used for platforms which don't need specialized AC'97 CODEC drivers while at the same avoiding code duplication from implementing equivalent functionality in a controller driver. Resending due to no response received. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index d0ac723..5b3224c 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -44,10 +44,6 @@ static int ac97_prepare(struct snd_pcm_substream *substream, return snd_ac97_set_rate(ac97, reg, substream->runtime->rate); } -#define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000) - static const struct snd_soc_dai_ops ac97_dai_ops = { .prepare = ac97_prepare, }; @@ -58,13 +54,13 @@ static struct snd_soc_dai_driver ac97_dai = { .stream_name = "AC97 Playback", .channels_min = 1, .channels_max = 2, - .rates = STD_AC97_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SND_SOC_STD_AC97_FMTS,}, .capture = { .stream_name = "AC97 Capture", .channels_min = 1, .channels_max = 2, - .rates = STD_AC97_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &ac97_dai_ops, }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..84cad9a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,7 +16,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 select SND_SOC_AB8500_CODEC if ABX500_CORE - select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS + select SND_SOC_AC97_CODEC select SND_SOC_AD1836 if SPI_MASTER select SND_SOC_AD193X_SPI if SPI_MASTER select SND_SOC_AD193X_I2C if I2C @@ -211,8 +211,9 @@ config SND_SOC_AB8500_CODEC tristate config SND_SOC_AC97_CODEC - tristate + tristate "Build generic ASoC AC97 CODEC driver" select SND_AC97_CODEC + select SND_SOC_AC97_BUS config SND_SOC_AD1836 tristate

2 4

[alsa-devel] [PATCH 1/2] ASoC: sta32x: Use devm_gpiod_get_optional appropriate place
by Axel Lin 09 May '15

09 May '15

devm_gpiod_get_optional() is equivalent to devm_gpiod_get(), except that when no GPIO was assigned to the requested function it will return NULL. This is convenient for drivers that need to handle optional GPIOs. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/sta32x.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 033b7d9..847fb20 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -1095,16 +1095,10 @@ static int sta32x_i2c_probe(struct i2c_client *i2c, #endif /* GPIOs */ - sta32x->gpiod_nreset = devm_gpiod_get(dev, "reset"); - if (IS_ERR(sta32x->gpiod_nreset)) { - ret = PTR_ERR(sta32x->gpiod_nreset); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta32x->gpiod_nreset = NULL; - } else { - gpiod_direction_output(sta32x->gpiod_nreset, 0); - } + sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(sta32x->gpiod_nreset)) + return PTR_ERR(sta32x->gpiod_nreset); /* regulators */ for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) -- 2.1.0

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[alsa-devel] PreSonus Information For Developers
by Karl Swisher 09 May '15

09 May '15

Hello all, I have what might be useful information to some developers but I do not sure who or how to contact the right people. I recently purchased a PreSonus StudioLive AI 32.4.2 mixer. Have it half way working under Linux. This mixer has a network connection in addition to the firewire interface. I've installed their software under VirtualBox and ran tcpflow capturing packet content between the computer and the mixer. I have started creating text files of the packets of which below is an example of their software performing functions like the mixer booting, backing up, restore, and selecting the digital output channels 33 through 48. 55 43 00 01 06 00 4b 41 65 00 64 00 55 43 00 01 06 00 4b 41 65 00 64 00 55 43 00 01 06 00 4b 41 65 00 64 00 55 43 00 01 06 00 4b 41 65 00 64 00 55 43 00 01 06 00 4b 41 65 00 64 00 55 43 00 01 06 00 4b 41 65 00 64 00 55 43 00 01 06 00 4b 41 65 00 64 00 55 43 00 01 1d 00 46 52 65 00 64 00 01 00 Listpresets/scene 00 00 55 43 00 01 1b 00 46 52 65 00 64 00 01 00 Listpresets/channel 00 00 55 43 00 01 19 00 46 52 65 00 64 00 01 00 Listpresets/geq 00 00 55 43 00 01 19 00 46 52 65 00 64 00 01 00 Listpresets/reverb 00 00 I'm trying to separate out the bin code from the text. Does anyone here know if there is value in this information or anyone that might want it? Have a few functions decoded in the format above. Thanks Karl

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Re: [alsa-devel] Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 09 May '15

09 May '15

> > > > http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixe > > > > r/paths/analog-output-headphones.conf > > > > > > > > pulseaudio mute external center/lfe controls when headphone jack is > > > > plugged > > That would be unwanted. > This is common to all notebook using headphone jack and two other jacks to support surround51 or surround71 since pulseaudio mute the other retasked mio jack when headphone jack is plugged > > > > I used hdajackretask to override: > > > PIN ID: 0x15 to "Line out (front)" > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/patch/sound/pc… When the static model were used, the jack was configured as line out instead of headphone jack and there is no headphone playback volume -static const struct hda_verb alc888_acer_aspire_4930g_verbs[] = { -/* Enable unsolicited event for HP jack */ - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, -/* Connect HP out to front */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, -static const struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), The jack is configured as headphone jack after the auto parser is used > > Pulseaudio don't know node 0x15 is headphone / Line Out (External > speaker) > > > > select speaker profile mute the headphone playback switch > > Doesn't HP mean HeadPhone? Isn't solution to my problem obvious, namely add > Line out capability to 0x15? > > Are there any side effect if you change the headphone jack to Line Out jack as the jack no longer enable HP amp ? use add_out_jack_modes for create Headphone/Line Out jack mode control to change Headphone to Line Out and vice versa diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 3d2597b..f25cd23 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -180,10 +180,10 @@ static void parse_user_hints(struct hda_codec *codec) /* the following two are just for compatibility */ val = snd_hda_get_bool_hint(codec, "add_out_jack_modes"); if (val >= 0) - spec->add_jack_modes = !!val; + spec->add_out_jack_modes = !!val; val = snd_hda_get_bool_hint(codec, "add_in_jack_modes"); if (val >= 0) spec->add_jack_modes = !!val; @@ -2640,7 +2642,7 @@ static void get_jack_mode_name(struct hda_codec *codec, hda_nid_t pin, static int get_out_jack_num_items(struct hda_codec *codec, hda_nid_t pin) { struct hda_gen_spec *spec = codec->spec; - if (spec->add_jack_modes) { + if (spec->add_out_jack_modes || spec->add_jack_modes) { unsigned int pincap = snd_hda_query_pin_caps(codec, pin); if ((pincap & AC_PINCAP_OUT) && (pincap & AC_PINCAP_HP_DRV)) return 2; @@ -4892,7 +4895,7 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, return err; } - if (spec->add_jack_modes) { + if (spec->add_out_jack_modes || spec->add_jack_modes) { if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { err = create_out_jack_modes(codec, cfg->line_outs, cfg->line_out_pins); diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 56e4139..8d7568b 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -227,6 +227,8 @@ struct hda_gen_spec { unsigned int prefer_hp_amp:1; /* enable HP amp for speaker if any */ unsigned int add_stereo_mix_input:2; /* add aamix as a capture src */ unsigned int add_jack_modes:1; /* add i/o jack mode enum ctls */ + unsigned int add_out_jack_modes:1; unsigned int power_down_unused:1; /* power down unused widgets */ unsigned int dac_min_mute:1; /* minimal = mute for DACs */

1 0

[alsa-devel] [PATCH 1/3] ASoC: rt5645: fix add missing widget
by Bard Liao 08 May '15

08 May '15

"IF1 DAC0" and "IF1 DAC3" are used in rt5645_dapm_routes but missing in rt5645_dapm_widgets. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 7996c9c..a72d989 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -1697,8 +1697,10 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { /* Digital Interface */ SND_SOC_DAPM_SUPPLY("I2S1", RT5645_PWR_DIG1, RT5645_PWR_I2S1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC0", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 DAC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC3", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("RT5645 IF1 DAC1 L Mux", SND_SOC_NOPM, 0, 0, &rt5645_if1_dac0_tdm_sel_mux), SND_SOC_DAPM_MUX("RT5645 IF1 DAC1 R Mux", SND_SOC_NOPM, 0, 0, -- 1.8.1.1.439.g50a6b54

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Re: [alsa-devel] [PATCH] ASoC: bt-sco: Add devicetree support for bt-sco codec
by Mark Brown 08 May '15

08 May '15

On Tue, May 05, 2015 at 10:15:41PM +0200, Marek Belisko wrote: > +static const struct of_device_id bt_sco_codec_of_match[] = { > + { .compatible = "delta,dfbmcs320", }, > + { .compatible = "linux,bt-sco", }, Is there any burning reason for adding the linux,bt-sco compatible?

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Re: [alsa-devel] [PATCH 2/2] ASoC: rt5645: improve headphone depop function
by Mark Brown 08 May '15

08 May '15

On Fri, May 08, 2015 at 01:53:34PM +0800, John Lin wrote: > We add a calibration function and call it at the beginning of > i2c_probe. The calibration value will be kept until codec is > shutdown. We will reset the codec after the calibration is finished. > So, we set cache_bypass in the calibration function. The benefit is > we can shorter the delay time in headphone depop. This looks fine but seems to depend on the previous patch.

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Re: [alsa-devel] [PATCH 1/2] ASoC: rt5645: fix jack type detect error
by Mark Brown 08 May '15

08 May '15

On Fri, May 08, 2015 at 01:53:33PM +0800, John Lin wrote: > rt5645_jack_detect doesn't report the correct jack type consistently. > This patch can make it more stable. Can you be more specific about what the problem is and how this fixes it pleased? It's a bit hard to review based on your changelog.

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[alsa-devel] [PATCH] ASoC: rt5677: Add DMIC ASRC detect function
by Oder Chiou 08 May '15

08 May '15

The patch adds DMIC ASRC detect function to dominate whether the DMIC ASRC enable or not. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 81 +++++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 75 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 685e268..8bb7be8 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1230,6 +1230,75 @@ int rt5677_sel_asrc_clk_src(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(rt5677_sel_asrc_clk_src); +static int rt5677_dmic_use_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int asrc_setting; + + switch (source->shift) { + case 11: + regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); + asrc_setting = (asrc_setting & RT5677_AD_STO1_CLK_SEL_MASK) >> + RT5677_AD_STO1_CLK_SEL_SFT; + if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && + asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) + return 1; + break; + + case 10: + regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); + asrc_setting = (asrc_setting & RT5677_AD_STO2_CLK_SEL_MASK) >> + RT5677_AD_STO2_CLK_SEL_SFT; + if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && + asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) + return 1; + break; + + case 9: + regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); + asrc_setting = (asrc_setting & RT5677_AD_STO3_CLK_SEL_MASK) >> + RT5677_AD_STO3_CLK_SEL_SFT; + if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && + asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) + return 1; + break; + + case 8: + regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); + asrc_setting = (asrc_setting & RT5677_AD_STO4_CLK_SEL_MASK) >> + RT5677_AD_STO4_CLK_SEL_SFT; + if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && + asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) + return 1; + break; + + case 7: + regmap_read(rt5677->regmap, RT5677_ASRC_6, &asrc_setting); + asrc_setting = (asrc_setting & RT5677_AD_MONOL_CLK_SEL_MASK) >> + RT5677_AD_MONOL_CLK_SEL_SFT; + if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && + asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) + return 1; + break; + + case 6: + regmap_read(rt5677->regmap, RT5677_ASRC_6, &asrc_setting); + asrc_setting = (asrc_setting & RT5677_AD_MONOR_CLK_SEL_MASK) >> + RT5677_AD_MONOR_CLK_SEL_SFT; + if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && + asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) + return 1; + break; + + default: + break; + } + + return 0; +} + /* Digital Mixer */ static const struct snd_kcontrol_new rt5677_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5677_STO1_ADC_MIXER, @@ -3087,12 +3156,12 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { }; static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { - { "Stereo1 DMIC Mux", NULL, "DMIC STO1 ASRC", can_use_asrc }, - { "Stereo2 DMIC Mux", NULL, "DMIC STO2 ASRC", can_use_asrc }, - { "Stereo3 DMIC Mux", NULL, "DMIC STO3 ASRC", can_use_asrc }, - { "Stereo4 DMIC Mux", NULL, "DMIC STO4 ASRC", can_use_asrc }, - { "Mono DMIC L Mux", NULL, "DMIC MONO L ASRC", can_use_asrc }, - { "Mono DMIC R Mux", NULL, "DMIC MONO R ASRC", can_use_asrc }, + { "Stereo1 DMIC Mux", NULL, "DMIC STO1 ASRC", rt5677_dmic_use_asrc }, + { "Stereo2 DMIC Mux", NULL, "DMIC STO2 ASRC", rt5677_dmic_use_asrc }, + { "Stereo3 DMIC Mux", NULL, "DMIC STO3 ASRC", rt5677_dmic_use_asrc }, + { "Stereo4 DMIC Mux", NULL, "DMIC STO4 ASRC", rt5677_dmic_use_asrc }, + { "Mono DMIC L Mux", NULL, "DMIC MONO L ASRC", rt5677_dmic_use_asrc }, + { "Mono DMIC R Mux", NULL, "DMIC MONO R ASRC", rt5677_dmic_use_asrc }, { "I2S1", NULL, "I2S1 ASRC", can_use_asrc}, { "I2S2", NULL, "I2S2 ASRC", can_use_asrc}, { "I2S3", NULL, "I2S3 ASRC", can_use_asrc}, -- 1.8.1.1.439.g50a6b54

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Re: [alsa-devel] [PATCH RFC v2 10/13] sound/core: add DRM ELD helper
by Russell King - ARM Linux 08 May '15

08 May '15

On Fri, May 08, 2015 at 04:16:08PM +0300, Jyri Sarha wrote: > On 2015-04-02 12:22, Russell King wrote: > >+static const unsigned int eld_rates[] = { > >+ 32000, > >+ 44100, > >+ 48000, > >+ 88200, > >+ 96000, > >+ 176400, > >+ 192000, > >+}; > >+ > > For what I can see, you are setting the cross dependency between > the sample rate and channels incorrectly. There is a dependency between sample rate and channels. > You should look for the > currently tested channel count directly from hw params. I'm not so sure that's right after looking at snd_ac97_pcm_double_rate_rules(). That's pretty much the same problem - AC'97 codecs can operate at a sample rate of either the bus frame frequency or twice the bus frame frequency. They achieve twice the bus frame frequency by reallocating a couple of the PCM data slots which would otherwise be used for >2 channels to the first two channels, thus allowing two front channel samples per frame. AC'97 limits the number of channels to two if: if (rate->min > 48000) { and limits the sample rate to 1-48kHz if: if (channels->min > 2) { > From this side the dependency is missing all together. Yes, it I initially couldn't work out how to do that bit... but I have a solution now which seems to sort-of work. Testing with aplay, I find that it seems to mostly work, and I guess that: aplay -D hw:0,0 -v -f S24_LE -c 6 -r 192000 /dev/zero resulting in: Playing raw data '/dev/zero' : Signed 24 bit Little Endian, Rate 192000 Hz, Channels 6 Warning: rate is not accurate (requested = 192000Hz, got = 48000Hz) please, try the plug plugin Hardware PCM card 0 'DW-HDMI' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S24_LE subformat : STD channels : 6 rate : 48000 exact rate : 48000 (48000/1) is acceptable. Also looking at AC'97, I don't need to list the same parameter as a dependent of the rule... something I'll try when I'm back from the hospital later this afternoon... -- FTTC broadband for 0.8mile line: currently at 10.5Mbps down 400kbps up according to speedtest.net.

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Re: [alsa-devel] [PATCH RFC v2 10/13] sound/core: add DRM ELD helper
by Russell King - ARM Linux 08 May '15

08 May '15

On Fri, May 08, 2015 at 01:56:02PM +0300, Jyri Sarha wrote: > On 2015-04-05 20:26, Russell King - ARM Linux wrote: > >On Sun, Apr 05, 2015 at 06:46:13PM +0200, Takashi Iwai wrote: > >>At Sun, 5 Apr 2015 17:20:34 +0100, > >>Russell King - ARM Linux wrote: > >>> Since (afaik) ALSA has a lack of support for dynamic reconfiguration > >>> according to the attached device changing, the best we can do without > >>> a huge amount of re-work of HDMI support across all adapters is to > >>> process the capabilities of the attached device at prepare time > >>> according to the current capabilities. > >> > >>Yeah, reconfiguration is tricky. BTW, how is the HDMI unplug handled > >>during playback? > > > >We don't handle it right now - and we don't have any notification to > >the audio drivers that that has happened. Even if we did have such a > >notification, I'm not sure what the audio driver could do with it at > >the moment. > > > >>> Implementing dynamic reconfiguration in ALSA is not something I want to > >>> get involved with, and as we /already/ have HDMI support merged in the > >>> kernel, this is not a blocker for at least trying to get some semblence > >>> of sanity, rather than having every driver re-implementing stuff like > >>> this. > >> > >>Well, I didn't mean about the dynamic reconfiguration. I thought of > >>rather min/max pairs, but it was just a wrong assumption. Scratch > >>it. > >> > >>One another question: don't we need to deal with the sample bits in > >>sad[2]? > > > >It should, but I'm very wary about doing that without seeing more > >examples of real SADs. > > If the same constraint setting helpers are to be used also with > external HDMI encoders with i2s interface, there should be an option > to leave out the constraints for the sample bits. There is no direct > connection between the number of bits on I2S and HDMI link. The CPU > DAI may apply its own constraints on the available sample formats and > too strict constraints at the encoder end may lead to zero available > formats. Possibly, but then how do we know which SAD to apply to limit the number of channels? I suspect for the use case you're suggesting above, it's better left to the user rather than generic code to work it out otherwise things get far too complex. We'd need to have some way for the driver to report back to this generic code the actual number of bits on the HDMI link. However, as we currently know, ALSA appears to have a problem here which pretty much makes properly restricting the channels and sample rates impossible to do. That issue is the one I'm much more keen to solve right now, because without that being solved, this set of patches is pretty much dead in the water. -- FTTC broadband for 0.8mile line: currently at 10.5Mbps down 400kbps up according to speedtest.net.

1 0

[alsa-devel] [PATCH v12 0/6] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 08 May '15

08 May '15

Based on 4.1-rc1 As Russell King's DRM ELD helper is not yet in any repository, it is included here with a little patch. v12: - use Russell King's DRM ELD helper (Mark Brown) v11: - reduce the patch series to adding the tda998x codec only v10: - add the generic dt-card - define the audio ports from a DT graph of ports (Russell King) - reuse HDMI constants (Andrew Jackson - Jyri Sarha) - alloc rate_constraints in codec (Jyri Sarha) - fix bad number of channels (Jyri Sarha) - correct codec generation from config (Russell King - Jyri Sarha) - no module init/exit (Russell King) v9: - back to a TDA998x specific CODEC - more comments - change magic values to constants v8: - change some comments about the patches v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (6): drm/edid: add function to help find SADs sound/core: add DRM ELD helper sound/core: remove the trace from the DRM ELD helper drm/i2c: tda998x: Add support of a DT graph of ports drm/i2c: tda998x: Change drvdata for audio extension ASoC: tda998x: add a codec to the HDMI transmitter .../devicetree/bindings/drm/i2c/tda998x.txt | 51 ++++++ drivers/gpu/drm/i2c/tda998x_drv.c | 178 +++++++++++++++++++-- include/drm/drm_edid.h | 19 +++ include/sound/pcm_drm_eld.h | 6 + include/sound/tda998x.h | 23 +++ sound/core/Kconfig | 3 + sound/core/Makefile | 1 + sound/core/pcm_drm_eld.c | 91 +++++++++++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 132 +++++++++++++++ 11 files changed, 501 insertions(+), 11 deletions(-) create mode 100644 include/sound/pcm_drm_eld.h create mode 100644 include/sound/tda998x.h create mode 100644 sound/core/pcm_drm_eld.c create mode 100644 sound/soc/codecs/tda998x.c -- 2.1.4

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[alsa-devel] [PATCH] ASoC: fsl_ssi: No need call of_device_is_available()
by Fabio Estevam 08 May '15

08 May '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> The comment and the call to of_device_is_available() are not really needed. It is the expected behaviour to probe only the ssi nodes that are enabled in the device tree. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0d48804..c7647e0 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1292,13 +1292,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) void __iomem *iomem; char name[64]; - /* SSIs that are not connected on the board should have a - * status = "disabled" - * property in their device tree nodes. - */ - if (!of_device_is_available(np)) - return -ENODEV; - of_id = of_match_device(fsl_ssi_ids, &pdev->dev); if (!of_id || !of_id->data) return -EINVAL; -- 1.9.1

3 2

[alsa-devel] [PATCH 1/4] ASoC: brownstone: Automatically disconnect non-connected pins
by Lars-Peter Clausen 08 May '15

08 May '15

Most DAPM input and output pins of the wm8994 are either used in the card's DAPM routing table or are marked as not connected. The only exception is DMIC2DAT input. Given that DMIC1DAT is explicitly mentioned in the DAPM routes lets assume that DMIC2DAT simply has been overlooked and should be marked as not connected. Set the fully_routed flag of the card instead of manually marking the unused inputs and outputs as not connected. This makes the code a bit shorter and cleaner. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/pxa/brownstone.c | 25 +------------------------ 1 file changed, 1 insertion(+), 24 deletions(-) diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 79936e3..2b26318 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -45,29 +45,6 @@ static const struct snd_soc_dapm_route brownstone_audio_map[] = { {"MICBIAS1", NULL, "Main Mic"}, }; -static int brownstone_wm8994_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* set endpoints to not connected */ - snd_soc_dapm_nc_pin(dapm, "HPOUT2P"); - snd_soc_dapm_nc_pin(dapm, "HPOUT2N"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT1N"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT1P"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT2N"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT2P"); - snd_soc_dapm_nc_pin(dapm, "IN1LN"); - snd_soc_dapm_nc_pin(dapm, "IN1LP"); - snd_soc_dapm_nc_pin(dapm, "IN1RP"); - snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN"); - snd_soc_dapm_nc_pin(dapm, "IN2RN"); - snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); - snd_soc_dapm_nc_pin(dapm, "IN2LN"); - - return 0; -} - static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -115,7 +92,6 @@ static struct snd_soc_dai_link brownstone_wm8994_dai[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ops = &brownstone_ops, - .init = brownstone_wm8994_init, }, }; @@ -132,6 +108,7 @@ static struct snd_soc_card brownstone = { .num_dapm_widgets = ARRAY_SIZE(brownstone_dapm_widgets), .dapm_routes = brownstone_audio_map, .num_dapm_routes = ARRAY_SIZE(brownstone_audio_map), + .fully_routed = true, }; static int brownstone_probe(struct platform_device *pdev) -- 1.8.0

2 4

[alsa-devel] [TINYCOMPRESS][PATCH v3 0/7] Convert build system to autotools
by Qais Yousef 08 May '15

08 May '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. Changes in v3: - fix a typo in PATCH 1 where we compile cplay.c instead of crec.c for crecord app. - Add mising SOB lines. Qais Yousef (7): 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 tinycompress.h: explicitly include required headers .gitignore | 39 ++++ AUTHORS | 0 Android.mk | 6 +- COPYING | 51 +++++ ChangeLog | 0 INSTALL | 365 ++++++++++++++++++++++++++++++++++++ Makefile.am | 3 + NEWS | 0 autogen.sh | 6 + configure.ac | 32 ++++ include/Makefile.am | 6 + include/tinycompress/tinycompress.h | 3 + m4/.place_holder | 0 makefile.linux | 41 ---- src/Makefile.am | 1 + src/lib/Makefile.am | 5 + compress.c => src/lib/compress.c | 22 ++- src/utils/Makefile.am | 11 ++ cplay.c => src/utils/cplay.c | 0 crec.c => src/utils/crec.c | 0 utils.c | 62 ------ 21 files changed, 537 insertions(+), 116 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 create mode 100644 configure.ac create mode 100644 include/Makefile.am create mode 100644 m4/.place_holder delete mode 100644 makefile.linux create mode 100644 src/Makefile.am create mode 100644 src/lib/Makefile.am rename compress.c => src/lib/compress.c (98%) create mode 100644 src/utils/Makefile.am rename cplay.c => src/utils/cplay.c (100%) rename crec.c => src/utils/crec.c (100%) delete mode 100644 utils.c -- 2.1.0

2 8

Re: [alsa-devel] [PATCH 3/3] ASoC: rt5645: add device tree support
by Mark Brown 08 May '15

08 May '15

On Fri, May 08, 2015 at 11:13:39AM +0300, Mika Westerberg wrote: > Looks like some of the context is lost but if I decode the above > right, you want to know if there is anything that corresponds to > of_get_named_gpio() in the generic property API? > You can use fwnode_get_named_gpiod() if you need to extract a GPIO > directly from firmware node. Otherwise you should be using > devm_gpiod_get() and friends. Right, almost certainly devm_gpiod_get() in this case as I've said several times now :(

1 0

[alsa-devel] [PATCH] ASoC: dapm: Add missing mutex unlock
by Charles Keepax 08 May '15

08 May '15

The is a missing mutex unlock on the error path in snd_soc_dapm_get_enum_double. This was introduced in commit 561ed680b764 ("ASoC: dapm: Add support for autodisable mux controls"). This patch adds the missing unlock. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ea3348e..6374852 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3075,8 +3075,10 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); if (e->reg != SND_SOC_NOPM && dapm_kcontrol_is_powered(kcontrol)) { int ret = soc_dapm_read(dapm, e->reg, &reg_val); - if (ret) + if (ret) { + mutex_unlock(&card->dapm_mutex); return ret; + } } else { reg_val = dapm_kcontrol_get_value(kcontrol); } -- 1.7.2.5

2 1

Re: [alsa-devel] [PATCH v3 3/8] mfd: arizona: Add support for WM8998 and WM1814
by Richard Fitzgerald 08 May '15

08 May '15

On Thu, May 07, 2015 at 09:52:38AM +0100, Lee Jones wrote: > On Fri, 01 May 2015, Richard Fitzgerald wrote: > > > Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> > > --- > > drivers/mfd/Kconfig | 6 + > > drivers/mfd/Makefile | 3 + > > drivers/mfd/arizona-core.c | 105 ++- > > drivers/mfd/arizona-i2c.c | 8 + > > drivers/mfd/arizona-irq.c | 9 + > > drivers/mfd/arizona.h | 5 + > > drivers/mfd/wm8998-tables.c | 1592 +++++++++++++++++++++++++++++++++ > > include/linux/mfd/arizona/core.h | 3 + > > include/linux/mfd/arizona/pdata.h | 2 + > > include/linux/mfd/arizona/registers.h | 220 +++++ > > 10 files changed, 1945 insertions(+), 8 deletions(-) > > create mode 100644 drivers/mfd/wm8998-tables.c > > How orthogonal is this patch? Can I just take it? > > For my own reference: > Acked-by: Lee Jones <lee.jones(a)linaro.org> This patches code added by patch 1 "mfd: arizona: Split INx_MODE into two fields" so you'll need to take that one as well. But if you are ok with that change too then yes it's ok to take both of them into mfd.

1 0

Re: [alsa-devel] Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 08 May '15

08 May '15

> > > My laptop has built-in front left, front right, rear left, rear right and > > > subwoofer and something is strange. > > > When I set channel output to 5.1 or 4.1 and external speakers are > > > disconnected, then front left and front right work as expected but > > > > built-in > > > > > subwoofer is heard only if I test rear left channel (wrong channel > > > > mapping?) > > > > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pc > > i/hda/hda_codec.c?id=cd4035e814b04c009ece9939396d06370c53ed44 > > > > Seem only two pairs of stereo speakers and internal subwoofer at left > > channel before this patch > > > > ALSA: hda - Enable surround speakers (when line out is also present) > > On my laptop case "Virtual SurroundSound" is written . When I test front left > and rear left, I hear distinct sound, so it would mean it's from two different > sources of sound. The driver use more than two channels only after David 's patch But all internal speakers of the notebook are at front of the user, The 3D sound effect is quite different from your external 5.1 speakers Do you mean your 5930g need champ {FL FR RL RR LFE }? This can be easily implemented by using "swap center/lfe" switch for those codec support R/L swap http://git.alsa-project.org/?p=alsa-lib.git;a=blob_plain;f=test/chmap.c;hb=… http://git.alsa-project.org/?p=alsa-lib.git;a=tree;f=src/conf/pcm;hb=HEAD If codec does not support R/L swap, your 4.1 swap chmap need 5.0 routing Do you expect pulseaudio show 4.1 swap(5.0) profile for internal speakers and 4.1 and 5.1 profiles for external speakers ?

1 0

[alsa-devel] Focusrite 6i6 invalid PCM device "front"
by Ramon Marco Navarro 08 May '15

08 May '15

# Background I can't seem to use any of the PCM device list defined automatically for my [Focusrite 6i6 device][1]. I tested this using `speaker-test`. # Tests: ## `speaker-test` output comparing between using [`sysdefault`][2] and [`front:USB`][3]. The `sysdefault` device emits sound for all of the formats, except for FLOAT_LE 44100Hz and 48000Hz. While front:USB doesn't emit anything at all. [1]: http://www.alsa-project.org/db/?f=b1c10accc89cd34dfdeb0d61b087a3cf413ea1a0 [2]: https://gist.githubusercontent.com/ramonmaruko/deb1918efa27eb860aa1/raw/f7a… [3]: https://gist.githubusercontent.com/ramonmaruko/1377846e39a7a6acd72f/raw/4d6…

1 0

Re: [alsa-devel] [PATCH v2 1/4] ASoC: Add gtm601 codec driver
by Mark Brown 07 May '15

07 May '15

On Wed, May 06, 2015 at 10:38:47PM +0200, Marek Belisko wrote: > This driver add PCM interface to a GTM601 UMTS modem chip. > There is no configuration interface. Applied, thanks. Please remember to CC maintainers on patches, you've missed Liam for some reason.

2 1

Re: [alsa-devel] [PATCH v2 2/4] Documentation: devicetree: sound: Add documentation for gtm601 codec
by Mark Brown 07 May '15

07 May '15

On Wed, May 06, 2015 at 10:38:48PM +0200, Marek Belisko wrote: > Add small documentation for gtm601 UMTS modem audio interface. git am complains that this patch is corrupt so I can't apply it. Please also use subject lines reflecting the style for the subsystem.

1 0

[alsa-devel] [PATCH 1/2] ASoC: dapm: Break out of widget search when source and sink are located
by Charles Keepax 07 May '15

07 May '15

Currently snd_soc_dapm_add_route will continue to search the widget list even after both the source and sink for the route have been located. This patch breaks out of the search when both are located giving a small improvement in probe time for drivers. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Note these patches depend on: commit 92fa12426741d52b39ec92ad77c9843d3fc2b3d6 Author: Lars-Peter Clausen <lars(a)metafoo.de> Date: Fri May 1 18:02:42 2015 +0200 ASoC: dapm: Add new widgets to the end of the widget list Thanks, Charles sound/soc/soc-dapm.c | 10 ++++++++-- 1 files changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a2e5f22..ea3348e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2617,14 +2617,20 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, list_for_each_entry(w, &dapm->card->widgets, list) { if (!wsink && !(strcmp(w->name, sink))) { wtsink = w; - if (w->dapm == dapm) + if (w->dapm == dapm) { wsink = w; + if (wsource) + break; + } continue; } if (!wsource && !(strcmp(w->name, source))) { wtsource = w; - if (w->dapm == dapm) + if (w->dapm == dapm) { wsource = w; + if (wsink) + break; + } } } /* use widget from another DAPM context if not found from this */ -- 1.7.2.5

4 12

[alsa-devel] Alsa triggering methods
by Andy Ng 07 May '15

07 May '15

Hi, I am working on a ALSA card driver that handles a DSP instance. The problem I have is that the triggering mechanism in ALSA to start a PCM has the interrupts disabled, and my dsp framework is actually acknowledging the start trigger with an event after 50 microseconds time that I can't handle because the IRQs are disabled. Is there any way when the trigger mechanism goes down to the card to be able to handle any acknowledgements what will happen within a sort period of time. I have seen bespoke triggering method but not sure if everyone is using it. Any ideas would be helpful. Best regards, Andreas

3 2

Re: [alsa-devel] [PATCH v2 RESEND] ARM: dts: Support audio on Exynos5422-odroidxu3 using simple-audio-card
by Inha Song 07 May '15

07 May '15

Hi Kukjin, Any comments about this patch? Best Regards, Inha Song. On Fri, 10 Apr 2015 14:48:13 +0530 Anand Moon <linux.amoon(a)gmail.com> wrote: > Hi Inha, > > Tested-by: Anand Moon <linux.amoon(a)gmail.com> > > -Anand Moon > > On 10 April 2015 at 14:32, Markus Reichl <m.reichl(a)fivetechno.de> wrote: > > Hi Inha, > > > > I have tested this on XU3 (Headphone output). > > > > Tested-by: Markus Reichl <m.reichl(a)fivetechno.de> > > > > Best Regards > > -- > > Markus Reichl > > > > Am 10.04.2015 um 09:32 schrieb Inha Song: > >> 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 ac6f860..73886d6 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 { > >> > > > -- > To unsubscribe from this list: send the line "unsubscribe linux-samsung-soc" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html

3 2

[alsa-devel] [PATCH 0/5] ASoC: Intel: let machines configure SSP configuration
by Vinod Koul 06 May '15

06 May '15

This series first add frame and data polarity to ssp config and then starts to use local values for ssp config and then eventually lets that be loaded by hw_params and updates by pcm format and tdm calls to allow machine t specify these parameters Vinod Koul (5): ASoC: Intel: add frame and data polarity to ssp config ASoC: Intel: use local values for ssp configuration ASoC: Intel: load hw_defaults in hw_params of ssp be ASoC: intel: add support for specifying PCM format ASoC: Intel: add support for configuring TDM slots for SSP sound/soc/intel/atom/sst-atom-controls.c | 167 +++++++++++++++++++++++---- sound/soc/intel/atom/sst-atom-controls.h | 7 ++ sound/soc/intel/atom/sst-mfld-platform-pcm.c | 47 +++++++- sound/soc/intel/atom/sst-mfld-platform.h | 2 + 4 files changed, 197 insertions(+), 26 deletions(-) -- 1.9.1

2 6

Re: [alsa-devel] Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 06 May '15

06 May '15

> > > > > > My problem is that when I set channel output to "Analog output > > > > surround > > > > > > > 5.1" > > > > > (or 2.1 or 4.1), I hear no sound in front speakers, but in case of > > > > 5.1 or > > > > > > > 4.1 > > > > You need pulseaudio lfe-filter for 2.1 > > > > http://cgit.freedesktop.org/pulseaudio/pulseaudio/log/?qt=grep&q=lfe-filter > > External 5.1 and 4.1 worked fine without lfe-filter. Fedora doesn't package lfe- > filter and I'm more interested to get my 5.1 configuration back. > > > do you mean surround21 need 6 channel instead of channels with 5.1 internal > > /external speaker ? > > > > http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=48f1b308cc66152eb > > 6db66742dd0d08d888cda8d;hp=5c4cd46810cef8850b037fca9e38ffd43b0bff22 > > No. When in KDE channel output is set to 2.1 it makes no difference what is set > in alsamixer. When my external speakers are disconnected built-in front left > and front right works on my laptop, but built-in subwoofer doesn't work There > is no difference what channel output is set in alsamixer. > > My laptop has built-in front left, front right, rear left, rear right and > subwoofer and something is strange. > When I set channel output to 5.1 or 4.1 and external speakers are > disconnected, then front left and front right work as expected but built-in > subwoofer is heard only if I test rear left channel (wrong channel mapping?) https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… Seem only two pairs of stereo speakers and internal subwoofer at left channel before this patch ALSA: hda - Enable surround speakers (when line out is also present)

1 0

[alsa-devel] [PATCH] ASoC: omap-twl4030: Use card DAPM context to access widgets
by Lars-Peter Clausen 06 May '15

06 May '15

Use the card DAPM context instead of the CODEC DAPM context since only card level widgets are accessed here. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/omap/omap-twl4030.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index 3673ada..7431314 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -159,9 +159,8 @@ static inline void twl4030_disconnect_pin(struct snd_soc_dapm_context *dapm, static int omap_twl4030_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = rtd->card; - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_context *dapm = &card->dapm; struct omap_tw4030_pdata *pdata = dev_get_platdata(card->dev); struct omap_twl4030 *priv = snd_soc_card_get_drvdata(card); int ret = 0; -- 1.8.0

2 1

[alsa-devel] [PATCH v3 0/7] ASoC: intel - add skylake PCM driver
by Vinod Koul 06 May '15

06 May '15

This patch series adds the hda codec wrapper first followed by asoc hda controller library, then SKL PCM driver and last decouples the controller for splitting the links Jeeja KP (7): ASoC: hda - add soc hda codec driver wrapper ALSA: hda - add new HDA registers ASoC: hda - adds SoC controller and stream operations ASoC: intel - add Skylake HDA platform driver ASoC: intel - add Skylake HDA audio driver ASoC: intel - add makefile support for SKL driver ASoC: intel - adds support for decoupled mode in skl driver include/sound/hda_register.h | 87 +++ include/sound/soc-hdaudio.h | 343 ++++++++++++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/hda/Kconfig | 3 + sound/soc/hda/Makefile | 3 + sound/soc/hda/soc-hda-codec.c | 85 +++ sound/soc/hda/soc-hda-codec.h | 44 ++ sound/soc/hda/soc-hdac-controller.c | 289 ++++++++++ sound/soc/hda/soc-hdac-stream.c | 390 ++++++++++++++ sound/soc/intel/Kconfig | 20 + sound/soc/intel/Makefile | 1 + sound/soc/intel/skylake/Makefile | 3 + sound/soc/intel/skylake/hda-skl-pcm.c | 933 +++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/hda-skl.c | 743 ++++++++++++++++++++++++++ sound/soc/intel/skylake/hda-skl.h | 74 +++ 16 files changed, 3020 insertions(+) create mode 100644 include/sound/soc-hdaudio.h create mode 100644 sound/soc/hda/Kconfig create mode 100644 sound/soc/hda/Makefile create mode 100644 sound/soc/hda/soc-hda-codec.c create mode 100644 sound/soc/hda/soc-hda-codec.h create mode 100644 sound/soc/hda/soc-hdac-controller.c create mode 100644 sound/soc/hda/soc-hdac-stream.c create mode 100644 sound/soc/intel/skylake/Makefile create mode 100644 sound/soc/intel/skylake/hda-skl-pcm.c create mode 100644 sound/soc/intel/skylake/hda-skl.c create mode 100644 sound/soc/intel/skylake/hda-skl.h -- 1.7.9.5

3 25

[alsa-devel] [PATCH 1/4] ASoC: dapm: Add new widgets to the end of the widget list
by Lars-Peter Clausen 06 May '15

06 May '15

Currently new widgets are appended to the beginning of the cards widget list. This has the effect that widgets that are created while iterating over the widget list in snd_soc_dapm_new_widgets() (like e.g. the auto-disable widgets) are not covered during that invocation of the function. If no further invocations of snd_soc_dapm_new_widgets() happen these widgets will not be fully initialized and e.g. no debugfs entries are created for them. By adding new widgets to the end of the widget list we make sure that widgets that are created in snd_soc_dapm_new_widgets() will still be handled during the same snd_soc_dapm_new_widgets() invocation and are always fully initialized. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- This series depends on Charles' mux auto-disable series. --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 79e6cf4..5c159f4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3252,7 +3252,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); INIT_LIST_HEAD(&w->dirty); - list_add(&w->list, &dapm->card->widgets); + list_add_tail(&w->list, &dapm->card->widgets); w->inputs = -1; w->outputs = -1; -- 1.8.0

2 5

[alsa-devel] [PATCH v3 1/5] ASoC: Correct typo in SOC_VALUE_ENUM_SINGLE macro
by Charles Keepax 06 May '15

06 May '15

xnitmes is clearly intended to be xnitems, but all other macros just refer to this as xitems, so change it to that. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v2: - Use snd_soc_enum_item_to_val to get zero value from mux to account for non-value enums - Improve handling in snd_soc_dapm_put_enum_double of the situation where the register and widget views of the control values don't match Thanks, Charles include/sound/soc.h | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a73d855..45cfd69 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -190,8 +190,8 @@ #define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xitems, xtexts, xvalues) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .mask = xmask, .items = xitems, .texts = xtexts, .values = xvalues} -#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xnitmes, xtexts, xvalues) \ - SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xnitmes, xtexts, xvalues) +#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xitems, xtexts, xvalues) \ + SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xitems, xtexts, xvalues) #define SOC_ENUM_SINGLE_VIRT(xitems, xtexts) \ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, xitems, xtexts) #define SOC_ENUM(xname, xenum) \ -- 1.7.2.5

3 11

[alsa-devel] [PATCH V4 0/5] tas571x amplifier driver
by Kevin Cernekee 06 May '15

06 May '15

V3->V4: Eliminate unnecessary dev->of_node NULL check. Move the "should I sync this register?" test into a common helper function. Clear map->no_sync_defaults at the end of the regcache_sync functions. Kevin Cernekee (5): regmap: Add a helper function for regcache sync test regmap: Use regcache_mark_dirty() to indicate power loss or reset ASoC: tas571x: Add DT binding document ASoC: tas571x: New driver for TI TAS571x power amplifiers MAINTAINERS: Add entry for tas571x ASoC codec driver .../devicetree/bindings/sound/tas571x.txt | 41 ++ MAINTAINERS | 6 + drivers/base/regmap/internal.h | 3 + drivers/base/regmap/regcache.c | 45 +- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas571x.c | 515 +++++++++++++++++++++ sound/soc/codecs/tas571x.h | 33 ++ 8 files changed, 635 insertions(+), 15 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tas571x.txt create mode 100644 sound/soc/codecs/tas571x.c create mode 100644 sound/soc/codecs/tas571x.h -- 2.2.0.rc0.207.ga3a616c

2 10

[alsa-devel] [PATCH] ASoC: Intel: Enabled button jack for BSW platform with rt5650 codec
by yang.a.fang＠intel.com 06 May '15

06 May '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> rt5650 codec supports 4 buttons detections so enabled it Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 23 ++++++++++++++++++++++- 1 file changed, 22 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 8f96c21..1468c3d 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -43,6 +43,7 @@ struct cht_acpi_card { struct cht_mc_private { struct snd_soc_jack hp_jack; struct snd_soc_jack mic_jack; + struct snd_soc_jack btn_jack; struct cht_acpi_card *acpi_card; }; @@ -182,6 +183,7 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) struct snd_soc_codec *codec = runtime->codec; struct snd_soc_dai *codec_dai = runtime->codec_dai; struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); + struct snd_soc_jack *btn_jack; /* Select clk_i2s1_asrc as ASRC clock source */ rt5645_sel_asrc_clk_src(codec, @@ -214,7 +216,26 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) return ret; } - rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack, NULL); + if (ctx->acpi_card->codec_type == CODEC_TYPE_RT5650) { + + ret = snd_soc_card_jack_new(runtime->card, "Button Jack", + SND_JACK_BTN_0 | + SND_JACK_BTN_1 | + SND_JACK_BTN_2 | + SND_JACK_BTN_3, &ctx->btn_jack, + NULL, 0); + if (ret) { + dev_err(runtime->dev, + "Btn jack creation failed %d\n", ret); + return ret; + } + + btn_jack = &ctx->btn_jack; + } else { + btn_jack = NULL; + } + + rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack, btn_jack); return ret; } -- 1.7.9.5

5 12

[alsa-devel] [PATCH] ASoC: tas571x: Eliminate redundant dev->of_node NULL check
by Kevin Cernekee 06 May '15

06 May '15

of_match_device() checks if dev->of_node is NULL, so we don't need to do it again in the probe function. Signed-off-by: Kevin Cernekee <cernekee(a)chromium.org> --- This applies to the broonie/sound.git for-next branch. sound/soc/codecs/tas571x.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index b187ea53a7f9..85bcc374c8e8 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -377,6 +377,7 @@ static int tas571x_i2c_probe(struct i2c_client *client, { struct tas571x_private *priv; struct device *dev = &client->dev; + const struct of_device_id *of_id; int i, ret; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); @@ -384,18 +385,12 @@ static int tas571x_i2c_probe(struct i2c_client *client, return -ENOMEM; i2c_set_clientdata(client, priv); - if (dev->of_node) { - const struct of_device_id *of_id; - - of_id = of_match_device(tas571x_of_match, dev); - if (of_id) - priv->chip = of_id->data; - } - - if (!priv->chip) { + of_id = of_match_device(tas571x_of_match, dev); + if (!of_id) { dev_err(dev, "Unknown device type\n"); return -EINVAL; } + priv->chip = of_id->data; priv->mclk = devm_clk_get(dev, "mclk"); if (IS_ERR(priv->mclk) && PTR_ERR(priv->mclk) != -ENOENT) { -- 2.2.0.rc0.207.ga3a616c

2 1

[alsa-devel] [PATCH] ALSA: hda - Add headset mic quirk for Dell Inspiron 5548
by David Henningsson 06 May '15

06 May '15

This enables the headset microphone on Dell Inspiron 5548, or at least some variants of it. Cc: stable(a)kernel.org BugLink: https://bugs.launchpad.net/bugs/1452175 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 06199e4..c5413e9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5337,6 +5337,13 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x17, 0x40000000}, {0x1d, 0x40700001}, {0x21, 0x02211050}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell Inspiron 5548", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC255_STANDARD_PINS, + {0x12, 0x90a60180}, + {0x14, 0x90170130}, + {0x17, 0x40000000}, + {0x1d, 0x40700001}, + {0x21, 0x02211040}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC256_STANDARD_PINS, {0x13, 0x40000000}), -- 1.9.1

1 0

[alsa-devel] [PATCH v11 0/3] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 06 May '15

06 May '15

Based on 4.1-rc1 v11: - reduce the patch series to adding the tda998x codec only v10: - add the generic dt-card - define the audio ports from a DT graph of ports (Russell King) - reuse HDMI constants (Andrew Jackson - Jyri Sarha) - alloc rate_constraints in codec (Jyri Sarha) - fix bad number of channels (Jyri Sarha) - correct codec generation from config (Russell King - Jyri Sarha) - no module init/exit (Russell King) v9: - back to a TDA998x specific CODEC - more comments - change magic values to constants v8: - change some comments about the patches v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (3): drm/i2c: tda998x: Add support of a DT graph of ports drm/i2c: tda998x: Change drvdata for audio extension ASoC: tda998x: add a codec to the HDMI transmitter .../devicetree/bindings/drm/i2c/tda998x.txt | 51 +++++ drivers/gpu/drm/i2c/tda998x_drv.c | 220 +++++++++++++++++++-- include/sound/tda998x.h | 30 +++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 174 ++++++++++++++++ 6 files changed, 471 insertions(+), 11 deletions(-) create mode 100644 include/sound/tda998x.h create mode 100644 sound/soc/codecs/tda998x.c -- 2.1.4

2 4

Re: [alsa-devel] [PATCH 2/2] ASoC: zx: Add zx296702 SPDIF support
by Mark Brown 06 May '15

06 May '15

On Tue, May 05, 2015 at 09:43:56AM +0800, Jun Nie wrote: > 2015-05-04 21:54 GMT+08:00 Mark Brown <broonie(a)kernel.org>: > > So this special HDMI S/PDIF controller is directly integrated with the > > HDMI IP in the SoC and not usable as a generic S/PDIF controller? That > > seems to mirror system designs which have an external HDMI encoder which > > use S/PDIF or I2S to connect the audio portion to an external device. > The first SPDIF controller on ZTE SOC is dedicated for HDMI and the > 2nd one is for external output only via physical pad. The two > controller IPs are identical, also independent with each other and > HDMI in register address region. OK, so this does sound like the software should like one of the systems with an external HDMI encoder and the HDMI handled separately rather than in this driver anyway. > > I'd suggest for future versions reorganizing things so that you add the > > S/PDIF driver with only S/PDIF support and then have a separate patch > > that layers on the HDMI support. > Reasonable suggestion. Then I can have more time to think about HDMI > part. More comments is welcome if any. This sounds like the software should look like a TDA998x (which is having patches posted still) so just have the S/PDIF controllers as pure S/PDIF controllers and then a separate driver for the HDMI part that it's hooked up to which deals with the audio configuration for the HDMI side.

1 0

Re: [alsa-devel] Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 05 May '15

05 May '15

> > > seem internal subwoofer only use one channel and don't have any stereo to > > mono downmix according to your patch > > > > It is not easy to assign different name to internal 4.1 and external 5.1 > > speaker when alc888 only has five DAC > > > > Some playback volume control must be shared > > If you know what's missing, you could send a few lines of code and I will test > it, otherwise I'll have to experiment on my own. https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/patch/sound/pc… c3e837bbcc03e39991da5dd1f791c0 6cde9f7a45 As your patch indicate internal subwoofer use external center speaker channel and your codec does not support L/R swap You may need to reserve Audio mixer 0x26 for internal subwoofer playback volume by restrict the connection of the internal rear speaker excluding 0x26 so that last DAC can be assigned to the internal subwoofer similar to alc889_fixup_dac_route void alc888_fixup_dac_route(struct hda_codec *codec, const struct alc_fixup *fix, int action) { if (action == ALC_FIXUP_ACT_PRE_PROBE) { hda_nid_t conn1[4] = { 0x0c, 0x0d, 0x0e, 0x0f}; snd_hda_override_conn_list(codec, 0x17, 4, conn1); } else if (action == ALC_FIXUP_ACT_PROBE) { conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 }; snd_hda_override_conn_list(codec, 0x17, 5, conn); } } > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/patch/sound/pc… c3e837bbcc03e39991da5dd1f791c0 6cde9f7a45 > > > > Do you mean internal subwoofer has low pass filter > > I don't know. How to check it? > > > > what do you mean when you say 4.1 worked fine ? > > External 5.1 speakers working as 4.1 speakers because such profile was set. > > > > The channel map of internal 4.1 speaker are different from external 5.1 > > speaker > > Recently I also made such conclusion. > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=2d3… Seem 4.1 is not defined in snd_pcm_std_chmap and you need to define and add your 4.1 chmap similar to 2.1 https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=ee8…

1 0

[alsa-devel] [RFC PATCH] ASoC: wm8741: Add differential mono mode support
by Sergej Sawazki 05 May '15

05 May '15

The WM8741 DAC supports several differential output modes (stereo, stereo reversed, mono left, mono right). Add platform data and DT bindings to configure it. Signed-off-by: Sergej Sawazki <ce3a(a)gmx.de> --- Documentation/devicetree/bindings/sound/wm8741.txt | 11 ++ sound/soc/codecs/wm8741.c | 129 ++++++++++++++++++--- sound/soc/codecs/wm8741.h | 10 ++ 3 files changed, 137 insertions(+), 13 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/wm8741.txt b/Documentation/devicetree/bindings/sound/wm8741.txt index 74bda58..a133154 100644 --- a/Documentation/devicetree/bindings/sound/wm8741.txt +++ b/Documentation/devicetree/bindings/sound/wm8741.txt @@ -10,9 +10,20 @@ Required properties: - reg : the I2C address of the device for I2C, the chip select number for SPI. +Optional properties: + + - diff-mode: Differential output mode configuration. Default value for field + DIFF in register R8 (MODE_CONTROL_2). If absent, the default is 0, shall be: + 0 = stereo + 1 = mono left + 2 = stereo reversed + 3 = mono right + Example: codec: wm8741@1a { compatible = "wlf,wm8741"; reg = <0x1a>; + + diff-mode = <3>; }; diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 9e71c76..9b777a6 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -41,6 +41,7 @@ static const char *wm8741_supply_names[WM8741_NUM_SUPPLIES] = { /* codec private data */ struct wm8741_priv { + struct wm8741_platform_data pdata; struct regmap *regmap; struct regulator_bulk_data supplies[WM8741_NUM_SUPPLIES]; unsigned int sysclk; @@ -87,13 +88,27 @@ static int wm8741_reset(struct snd_soc_codec *codec) static const DECLARE_TLV_DB_SCALE(dac_tlv_fine, -12700, 13, 0); static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 400, 0); -static const struct snd_kcontrol_new wm8741_snd_controls[] = { +static const struct snd_kcontrol_new wm8741_snd_controls_stereo[] = { SOC_DOUBLE_R_TLV("Fine Playback Volume", WM8741_DACLLSB_ATTENUATION, WM8741_DACRLSB_ATTENUATION, 1, 255, 1, dac_tlv_fine), SOC_DOUBLE_R_TLV("Playback Volume", WM8741_DACLMSB_ATTENUATION, WM8741_DACRMSB_ATTENUATION, 0, 511, 1, dac_tlv), }; +static const struct snd_kcontrol_new wm8741_snd_controls_mono_left[] = { +SOC_SINGLE_TLV("Fine Playback Volume Left", WM8741_DACLLSB_ATTENUATION, + 1, 255, 1, dac_tlv_fine), +SOC_SINGLE_TLV("Playback Volume Left", WM8741_DACLMSB_ATTENUATION, + 0, 511, 1, dac_tlv), +}; + +static const struct snd_kcontrol_new wm8741_snd_controls_mono_right[] = { + SOC_SINGLE_TLV("Fine Playback Volume Right", WM8741_DACRLSB_ATTENUATION, + 1, 255, 1, dac_tlv_fine), + SOC_SINGLE_TLV("Playback Volume Right", WM8741_DACRMSB_ATTENUATION, + 0, 511, 1, dac_tlv), +}; + static const struct snd_soc_dapm_widget wm8741_dapm_widgets[] = { SND_SOC_DAPM_DAC("DACL", "Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DACR", "Playback", SND_SOC_NOPM, 0, 0), @@ -398,7 +413,7 @@ static struct snd_soc_dai_driver wm8741_dai = { .name = "wm8741", .playback = { .stream_name = "Playback", - .channels_min = 2, /* Mono modes not yet supported */ + .channels_min = 1, .channels_max = 2, .rates = WM8741_RATES, .formats = WM8741_FORMATS, @@ -416,6 +431,65 @@ static int wm8741_resume(struct snd_soc_codec *codec) #define wm8741_resume NULL #endif +static int wm8741_configure(struct snd_soc_codec *codec) +{ + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + + /* Configure differential mode */ + switch (wm8741->pdata.diff_mode) { + case WM8741_DIFF_MODE_STEREO: + case WM8741_DIFF_MODE_STEREO_REVERSED: + case WM8741_DIFF_MODE_MONO_LEFT: + case WM8741_DIFF_MODE_MONO_RIGHT: + snd_soc_update_bits(codec, WM8741_MODE_CONTROL_2, + WM8741_DIFF_MASK, + wm8741->pdata.diff_mode << WM8741_DIFF_SHIFT); + break; + default: + return -EINVAL; + } + + /* Change some default settings - latch VU */ + snd_soc_update_bits(codec, WM8741_DACLLSB_ATTENUATION, + WM8741_UPDATELL, WM8741_UPDATELL); + snd_soc_update_bits(codec, WM8741_DACLMSB_ATTENUATION, + WM8741_UPDATELM, WM8741_UPDATELM); + snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, + WM8741_UPDATERL, WM8741_UPDATERL); + snd_soc_update_bits(codec, WM8741_DACRMSB_ATTENUATION, + WM8741_UPDATERM, WM8741_UPDATERM); + + return 0; +} + +static int wm8741_add_controls(struct snd_soc_codec *codec) +{ + struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); + + switch (wm8741->pdata.diff_mode) { + case WM8741_DIFF_MODE_STEREO: + case WM8741_DIFF_MODE_STEREO_REVERSED: + snd_soc_add_codec_controls(codec, + wm8741_snd_controls_stereo, + ARRAY_SIZE(wm8741_snd_controls_stereo)); + break; + case WM8741_DIFF_MODE_MONO_LEFT: + snd_soc_add_codec_controls(codec, + wm8741_snd_controls_mono_left, + ARRAY_SIZE(wm8741_snd_controls_mono_left)); + break; + case WM8741_DIFF_MODE_MONO_RIGHT: + snd_soc_add_codec_controls(codec, + wm8741_snd_controls_mono_right, + ARRAY_SIZE(wm8741_snd_controls_mono_right)); + break; + default: + return -EINVAL; + } + + return 0; +} + static int wm8741_probe(struct snd_soc_codec *codec) { struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); @@ -434,15 +508,17 @@ static int wm8741_probe(struct snd_soc_codec *codec) goto err_enable; } - /* Change some default settings - latch VU */ - snd_soc_update_bits(codec, WM8741_DACLLSB_ATTENUATION, - WM8741_UPDATELL, WM8741_UPDATELL); - snd_soc_update_bits(codec, WM8741_DACLMSB_ATTENUATION, - WM8741_UPDATELM, WM8741_UPDATELM); - snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, - WM8741_UPDATERL, WM8741_UPDATERL); - snd_soc_update_bits(codec, WM8741_DACRMSB_ATTENUATION, - WM8741_UPDATERM, WM8741_UPDATERM); + ret = wm8741_configure(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to change default settings\n"); + goto err_enable; + } + + ret = wm8741_add_controls(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to add controls\n"); + goto err_enable; + } dev_dbg(codec->dev, "Successful registration\n"); return ret; @@ -467,8 +543,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8741 = { .remove = wm8741_remove, .resume = wm8741_resume, - .controls = wm8741_snd_controls, - .num_controls = ARRAY_SIZE(wm8741_snd_controls), .dapm_widgets = wm8741_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8741_dapm_widgets), .dapm_routes = wm8741_dapm_routes, @@ -493,6 +567,23 @@ static const struct regmap_config wm8741_regmap = { .readable_reg = wm8741_readable, }; +static int wm8741_set_pdata(struct device *dev, struct wm8741_priv *wm8741) +{ + const struct wm8741_platform_data *pdata = dev_get_platdata(dev); + u32 diff_mode; + + if (dev->of_node) { + if (of_property_read_u32(dev->of_node, "diff-mode", &diff_mode) + >= 0) + wm8741->pdata.diff_mode = diff_mode; + } else { + if (pdata != NULL) + memcpy(&wm8741->pdata, pdata, sizeof(wm8741->pdata)); + } + + return 0; +} + #if IS_ENABLED(CONFIG_I2C) static int wm8741_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -522,6 +613,12 @@ static int wm8741_i2c_probe(struct i2c_client *i2c, return ret; } + wm8741_set_pdata(&i2c->dev, wm8741); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to set pdata: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, wm8741); ret = snd_soc_register_codec(&i2c->dev, @@ -582,6 +679,12 @@ static int wm8741_spi_probe(struct spi_device *spi) return ret; } + wm8741_set_pdata(&spi->dev, wm8741); + if (ret != 0) { + dev_err(&spi->dev, "Failed to set pdata: %d\n", ret); + return ret; + } + spi_set_drvdata(spi, wm8741); ret = snd_soc_register_codec(&spi->dev, diff --git a/sound/soc/codecs/wm8741.h b/sound/soc/codecs/wm8741.h index 56c1b1d..c8835f6 100644 --- a/sound/soc/codecs/wm8741.h +++ b/sound/soc/codecs/wm8741.h @@ -194,6 +194,12 @@ #define WM8741_DITHER_SHIFT 0 /* DITHER - [1:0] */ #define WM8741_DITHER_WIDTH 2 /* DITHER - [1:0] */ +/* DIFF field values */ +#define WM8741_DIFF_MODE_STEREO 0 /* stereo normal */ +#define WM8741_DIFF_MODE_STEREO_REVERSED 2 /* stereo reversed */ +#define WM8741_DIFF_MODE_MONO_LEFT 1 /* mono left */ +#define WM8741_DIFF_MODE_MONO_RIGHT 3 /* mono right */ + /* * R32 (0x20) - ADDITONAL_CONTROL_1 */ @@ -208,4 +214,8 @@ #define WM8741_SYSCLK 0 +struct wm8741_platform_data { + u32 diff_mode; /* Differential Output Mode */ +}; + #endif -- 1.9.1

3 7

[alsa-devel] [PATCH] ALSA: hda/tegra - Fix oops
by Thierry Reding 05 May '15

05 May '15

From: Thierry Reding <treding(a)nvidia.com> Commit a41d122449be ("ALSA: hda - Embed bus into controller object") introduced a regression in the Tegra HDA driver that causes the following oops during boot: [ 2.333458] Unable to handle kernel NULL pointer dereference at virtual address 000004c4 [ 2.341537] pgd = c0004000 [ 2.344312] [000004c4] *pgd=00000000 [ 2.347898] Internal error: Oops: 5 [#1] PREEMPT SMP ARM [ 2.353200] Modules linked in: [ 2.356264] CPU: 0 PID: 1 Comm: swapper/0 Tainted: G W 4.1.0-rc2-next-20150505-00344-g8577890defbf #79 [ 2.366682] Hardware name: NVIDIA Tegra SoC (Flattened Device Tree) [ 2.372939] task: ee0d8b40 ti: ee0da000 task.ti: ee0da000 [ 2.378336] PC is at azx_bus_init+0x18/0xf4 [ 2.382516] LR is at hda_tegra_probe+0x6c/0x478 [ 2.387043] pc : [<c06156c4>] lr : [<c061cf00>] psr: 60000113 [ 2.387043] sp : ee0dbe38 ip : 00000000 fp : 00000000 [ 2.398501] r10: ed874c00 r9 : 000000fd r8 : 00000000 [ 2.403717] r7 : ed874c10 r6 : 00000000 r5 : 00000000 r4 : ed016810 [ 2.410232] r3 : c08a2ad4 r2 : c08a1ea0 r1 : 00000000 r0 : ed016810 [ 2.416750] Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel [ 2.424046] Control: 10c5387d Table: 8000406a DAC: 00000015 [ 2.429783] Process swapper/0 (pid: 1, stack limit = 0xee0da210) [ 2.435778] Stack: (0xee0dbe38 to 0xee0dc000) [ 2.440129] be20: 00000000 ed016810 [ 2.448297] be40: 00000000 c061cf00 00000000 ee0dbe5c ed8735d0 c0a7bc48 ed02fd50 ed016000 [ 2.456462] be60: c1250164 ed874c10 c0c66bf8 fffffdfb 00000000 000000fd c0b8dc98 c046664c [ 2.464628] be80: c0466608 c1250164 ed874c10 00000000 c0c66bf8 c0464eb4 ed874c10 c0c66bf8 [ 2.472793] bea0: ed874c44 c0c43458 00000000 c04650d0 00000000 c0c66bf8 c046503c c04633b4 [ 2.480959] bec0: ee11bea4 ed85f390 c0c66bf8 ed017ac0 00000000 c0464634 c0ab2b7c c0c66bf8 [ 2.489125] bee0: c0bfde20 c0c66bf8 c0bfde20 ed01ce40 c0b7b414 c04656e8 c04665b0 c0bfde20 [ 2.497291] bf00: c0bfde20 c0009770 ee0d8b40 c0c02488 60000113 00000000 00000000 00000003 [ 2.505458] bf20: 00000000 c0c02488 60000113 00000000 c0b54598 c0b16a90 ef7fcc57 c0041228 [ 2.513624] bf40: c0a9150c ef7fcc5f 00000006 00000006 00000000 c0bf1fa8 c0bf2354 00000006 [ 2.521790] bf60: c0b8dc90 c0c7c000 c0c7c000 c0b8dc98 00000000 c0b54dd8 00000006 00000006 [ 2.529956] bf80: c0b54598 00000000 00000000 c07ff08c 00000000 00000000 00000000 00000000 [ 2.538122] bfa0: 00000000 c07ff094 00000000 c000f5a0 00000000 00000000 00000000 00000000 [ 2.546286] bfc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 [ 2.554451] bfe0: 00000000 00000000 00000000 00000000 00000013 00000000 fffff7ff c013f264 [ 2.562624] [<c06156c4>] (azx_bus_init) from [<c061cf00>] (hda_tegra_probe+0x6c/0x478) [ 2.570535] [<c061cf00>] (hda_tegra_probe) from [<c046664c>] (platform_drv_probe+0x44/0xa4) [ 2.578879] [<c046664c>] (platform_drv_probe) from [<c0464eb4>] (driver_probe_device+0x174/0x2b8) [ 2.587739] [<c0464eb4>] (driver_probe_device) from [<c04650d0>] (__driver_attach+0x94/0x98) [ 2.596172] [<c04650d0>] (__driver_attach) from [<c04633b4>] (bus_for_each_dev+0x6c/0xa0) [ 2.604342] [<c04633b4>] (bus_for_each_dev) from [<c0464634>] (bus_add_driver+0x148/0x1f0) [ 2.612597] [<c0464634>] (bus_add_driver) from [<c04656e8>] (driver_register+0x78/0xf8) [ 2.620593] [<c04656e8>] (driver_register) from [<c0009770>] (do_one_initcall+0x8c/0x1d4) [ 2.628765] [<c0009770>] (do_one_initcall) from [<c0b54dd8>] (kernel_init_freeable+0x144/0x1e4) [ 2.637459] [<c0b54dd8>] (kernel_init_freeable) from [<c07ff094>] (kernel_init+0x8/0xe8) [ 2.645543] [<c07ff094>] (kernel_init) from [<c000f5a0>] (ret_from_fork+0x14/0x34) This is caused by azx_bus_init() trying to dereference chip->card, which for the Tegra driver doesn't get initialized until sometime later during the call to hda_tegra_create(). Fix this by mimicking the behaviour of the Intel driver and defer HDA bus initialization until right before the call to snd_device_new(). Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 801e9fb4a467..db0bb50fa5b9 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -439,6 +439,10 @@ static int hda_tegra_create(struct snd_card *card, chip->single_cmd = false; chip->snoop = true; + err = azx_bus_init(chip, NULL, &hda_tegra_io_ops); + if (err < 0) + return err; + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { dev_err(card->dev, "Error creating device\n"); @@ -475,10 +479,6 @@ static int hda_tegra_probe(struct platform_device *pdev) return err; } - err = azx_bus_init(chip, NULL, &hda_tegra_io_ops); - if (err < 0) - goto out_free; - err = hda_tegra_create(card, driver_flags, hda); if (err < 0) goto out_free; -- 2.3.5

2 1

Re: [alsa-devel] Add ALC298 alias name for Dell
by Takashi Iwai 05 May '15

05 May '15

At Tue, 5 May 2015 07:19:04 +0000, Kailang wrote: > > Hi Takashi, > > Dell has new platform use ALC298 codec. > ALC298 for Dell is call ALC3266. Thanks, applied. Takashi > > BR, > Kailang > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 0000-add-alc298-alias-name-for-Dell.patch <application/octet-stream (base64)>] >

1 0

[alsa-devel] [PATCH] ASoC: Intel: Fixed kbuild warnings
by yang.a.fang＠intel.com 05 May '15

05 May '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> fix following sparse warnings: (new ones prefixed by >>) >> sound/soc/intel/boards/cht_bsw_max98090_ti.c:168:37: sparse: >> incorrect type in argument 2 (different base types) sound/soc/intel/boards/cht_bsw_max98090_ti.c:168:37: expected unsigned int [unsigned] val sound/soc/intel/boards/cht_bsw_max98090_ti.c:168:37: got restricted snd_pcm_format_t [usertype] <noident> Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 3c518b1..1be0794 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -163,9 +163,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP2 to 24-bit */ - snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S24_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/6] ASoC: rt5645: remove RT5645_I2S_BCLK_MS1 control
by Bard Liao 05 May '15

05 May '15

RT5645_I2S_BCLK_MS1 (reg 0x73 [5]) is reserverd in rt5645 and rt5650. This function is move to TDM control. We can configure it by snd_soc_dai_set_tdm_slot's slot_width parameter. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 5 ++--- sound/soc/codecs/rt5645.h | 4 ---- 2 files changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 8bad9fe..33ebe88 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2104,9 +2104,8 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, switch (dai->id) { case RT5645_AIF1: - mask_clk = RT5645_I2S_BCLK_MS1_MASK | RT5645_I2S_PD1_MASK; - val_clk = bclk_ms << RT5645_I2S_BCLK_MS1_SFT | - pre_div << RT5645_I2S_PD1_SFT; + mask_clk = RT5645_I2S_PD1_MASK; + val_clk = pre_div << RT5645_I2S_PD1_SFT; snd_soc_update_bits(codec, RT5645_I2S1_SDP, (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 4473636..fa5c560 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -942,10 +942,6 @@ #define RT5645_I2S2_SDI_I2S2 (0x1 << 6) /* ADC/DAC Clock Control 1 (0x73) */ -#define RT5645_I2S_BCLK_MS1_MASK (0x1 << 15) -#define RT5645_I2S_BCLK_MS1_SFT 15 -#define RT5645_I2S_BCLK_MS1_32 (0x0 << 15) -#define RT5645_I2S_BCLK_MS1_64 (0x1 << 15) #define RT5645_I2S_PD1_MASK (0x7 << 12) #define RT5645_I2S_PD1_SFT 12 #define RT5645_I2S_PD1_1 (0x0 << 12) -- 1.8.1.1.439.g50a6b54

2 11

[alsa-devel] need information about dapm events, bias and mute management
by gpe 05 May '15

05 May '15

Hi, I'm working on an asoc codec but I'm facing some problems with the DAPM events the bias and the mute management. The kernel is a 3.4.91. Globally my codec is functionnal but I've some glitch. By using DAPM events (PRE_PMU, POST_PMU) and bias management I've suppressed the glitchs at the start but I've always some glitchs at the end of a playback. I'm trying to suppress them with PRE_PMD but it not works. When I test my code I see that some power off are done before the call of the SND_SOC_DAPM_PRE_PMD event. Is it normal ? Another question is regarding the mute management (snd_soc_dai_digital_mute) which is done after all the power off ... I expected it to be done just at the end of the playback. Last question about the bias management (set_bias_level in snd_soc_codec_driver). During the start I see the bias level incrementing but during the stop there is no action on the bias level. Why ? Regards. PS: I'm a newbie on codec development so don't hit to hard on my head if my questions are stupid. I've read many docs but I don't found a doc which describes the interactions between the APIs. PS2: sorry for my bad english. -- +-------+ | gpe92 | +-------+

1 0

[alsa-devel] Software based mmap support and dmix
by Yogesh Soni 05 May '15

05 May '15

Hi all, I am trying to have dmix plugin work ( I succeeded) with the audio card that does not have hardware mmap support and uses pcm_indirect method to map software buffer for mmap feature. Most of the plugins were working quite well with mmap/pcm_indirect combo. However, dmix plugin did not work (no sound). I debugged and realized that appl_ptr in pcm_direct.h(snd_pcm_indirect_playback_transfer()) was always 0 and hence nothing got copied to slave pcm buffer. I then discovered that dmix operates in freewheeling-mode.. I have made a small workaround patch (I believe it can be cleaned up) in case of free-wheeling mode, basically using sw_ready to hold bytes that need to be copied. I used rec->sw_ready because it looked safest to use for this purpose. As a result of this patch, normally period_size bytes are copied to pcm buffer every period and sound is playing smooth. I am a noob in alsa and linux in general. So, I would like your opinion on the patch. I would much prefer if I could write a patch that is generic and clean enough, so that it could be used by others. New pcm-indirect.h attached. Thanks a lot for your help. Regards Yogesh

1 0

[alsa-devel] [PATCH 1/7] ASoC: ad1836: Replace direct snd_soc_codec dapm field access
by Lars-Peter Clausen 04 May '15

04 May '15

The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm with snd_soc_codec_get_dapm(). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ad1836.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 685998d..95f0bec 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -251,7 +251,7 @@ static int ad1836_resume(struct snd_soc_codec *codec) static int ad1836_probe(struct snd_soc_codec *codec) { struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); int num_dacs, num_adcs; int ret = 0; int i; -- 1.8.0

2 7

[alsa-devel] [PATCH V3 0/4] tas571x amplifier driver
by Kevin Cernekee 04 May '15

04 May '15

V2->V3: Revert earlier changes to the regcache APIs. Change regcache_sync() so that regcache_mark_dirty() is used to indicate that the HW was reset. This allows tas571x (which does NOT call regcache_mark_dirty()) to use regcache_sync() to write back any register changes that occurred while in cache_only (tas571x pdn) mode. One assumption here is that regcache_sync() will be called once on resume, before any other writes. If it fails, the no_sync_defaults flag will be cleared and the next sync attempt, if any, will try to write out all contents of the cache. Kevin Cernekee (4): regmap: Use regcache_mark_dirty() to indicate power loss or reset ASoC: tas571x: Add DT binding document ASoC: tas571x: New driver for TI TAS571x power amplifiers MAINTAINERS: Add entry for tas571x ASoC codec driver .../devicetree/bindings/sound/tas571x.txt | 41 ++ MAINTAINERS | 6 + drivers/base/regmap/internal.h | 3 + drivers/base/regmap/regcache.c | 61 ++- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas571x.c | 520 +++++++++++++++++++++ sound/soc/codecs/tas571x.h | 33 ++ 8 files changed, 651 insertions(+), 20 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tas571x.txt create mode 100644 sound/soc/codecs/tas571x.c create mode 100644 sound/soc/codecs/tas571x.h -- 2.2.0.rc0.207.ga3a616c

3 10

[alsa-devel] [PATCH 1/4] ASoC: Intel: Add support max98090 in sst driver
by yang.a.fang＠intel.com 04 May '15

04 May '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> Added entry in sst driver to support max98090 codec for intel Braswell platform. Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/intel/atom/sst/sst_acpi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index fc02a48..bb19b58 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -356,6 +356,8 @@ static struct sst_machines sst_acpi_chv[] = { &chv_platform_data }, {"10EC5650", "cht-bsw", "cht-bsw-rt5645", NULL, "intel/fw_sst_22a8.bin", &chv_platform_data }, + {"193C9890", "cht-bsw", "cht-bsw-max98090", NULL, + "intel/fw_sst_22a8.bin", &chv_platform_data }, {}, }; -- 1.7.9.5

7 23

[alsa-devel] [PATCH] ASoC: uda1380: Avoid accessing i2c bus when codec is disabled
by Vasily Khoruzhick 04 May '15

04 May '15

set_dai_fmt_both() callback is called from snd_soc_runtime_set_dai_fmt() which is called from snd_soc_register_card(), but at this time codec is not powered on yet. Replace direct i2c write with regcache write. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- sound/soc/codecs/uda1380.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index dc7778b..c3c33bd 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -437,7 +437,7 @@ static int uda1380_set_dai_fmt_both(struct snd_soc_dai *codec_dai, if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) return -EINVAL; - uda1380_write(codec, UDA1380_IFACE, iface); + uda1380_write_reg_cache(codec, UDA1380_IFACE, iface); return 0; } -- 2.3.5

2 1

Re: [alsa-devel] [PATCH] ASoC: simple-card: support platform in dts parse
by Mark Brown 04 May '15

04 May '15

On Wed, Apr 29, 2015 at 06:11:07PM +0800, Jun Nie wrote: > Support platform in dts parse so that dma pcm component > can be added in dts. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH 2/2] ASoC: zx: Add zx296702 SPDIF support
by Mark Brown 04 May '15

04 May '15

On Mon, May 04, 2015 at 09:13:48PM +0800, Jun Nie wrote: > 2015-05-01 18:55 GMT+08:00 Mark Brown <broonie(a)kernel.org>: > > This is adding a new device with DT bindings but there is no bindings > > document. All new DT bindings must be documented. > Plan to add a separate patch to device tree. Or you prefer to add > device tree patch in this patch set and loop device tree maintainer? You *always* need to document new device tree bindings at the same time as adding new code. > There are more than one SPDIF controller on ZTE SOC. One of them > support HDMI output, while the other SPDIF controller support direct > output. So I want to separate SPDIF and HDMI and add SPDIF id 0 test So this special HDMI S/PDIF controller is directly integrated with the HDMI IP in the SoC and not usable as a generic S/PDIF controller? That seems to mirror system designs which have an external HDMI encoder which use S/PDIF or I2S to connect the audio portion to an external device. > Do you have any suggestion on what driver to look at for the interface > between audio and video? Half the problem right now is that the people working on HDMI don't appear to be talking to each other very much so everything is very inconsistent with no real code sharing. I don't really feel I've got a good enough picture of what the hardware looks like in general to be 100% clear on what the best way is forward beyond the fact that we should be sharing the EDID parsing code. I'd suggest for future versions reorganizing things so that you add the S/PDIF driver with only S/PDIF support and then have a separate patch that layers on the HDMI support.

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Re: [alsa-devel] Fix typo for ALC286/ALC288
by Takashi Iwai 04 May '15

04 May '15

At Mon, 4 May 2015 07:59:07 +0000, Kailang wrote: > > Hi Takashi, > > Sorry. > It had typo for my prior patch. OK, applied now. Thanks. Takashi > > BR, > Kailang > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 0000-fix-typo-for-ALC286.patch <application/octet-stream (base64)>] >

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[alsa-devel] [PATCH] ASoC: rsnd: Use generic names for device nodes
by Geert Uytterhoeven 04 May '15

04 May '15

rcar_sound -> sound Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index f316ce1f214a03e3..62ece4c59da759a4 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -47,7 +47,7 @@ DAI subnode properties: Example: -rcar_sound: rcar_sound@ec500000 { +rcar_sound: sound@ec500000 { #sound-dai-cells = <1>; compatible = "renesas,rcar_sound-r8a7791", "renesas,rcar_sound-gen2"; reg = <0 0xec500000 0 0x1000>, /* SCU */ -- 1.9.1

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[alsa-devel] [PATCH 1/5] ASoC: lowland: Use card DAPM context to access widgets
by Lars-Peter Clausen 04 May '15

04 May '15

The dapm field of the snd_soc_codec struct will eventually be removed (replaced with the DAPM context from the component embedded inside the CODEC). Replace its usage with the card's DAPM context. The idea is that DAPM is hierarchical and with the card at the root it is possible to access widgets from other contexts through the card context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/samsung/lowland.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index 5f15609..0d0f582 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -72,7 +72,7 @@ static int lowland_wm9081_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; - snd_soc_dapm_nc_pin(&codec->dapm, "LINEOUT"); + snd_soc_dapm_nc_pin(&rtd->card->dapm, "LINEOUT"); /* At any time the WM9081 is active it will have this clock */ return snd_soc_codec_set_sysclk(codec, WM9081_SYSCLK_MCLK, 0, -- 1.8.0

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[alsa-devel] [PATCH] ASoC: da7210: Fix dependency to allow build with SND_SOC_I2C_AND_SPI
by Axel Lin 04 May '15

04 May '15

Since commit aa0e25caafb7 ("ASoC: da7210: Add support for spi regmap"), the da7210 codec driver supports both I2C and SPI buses. Thus update the dependency accordingly. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..ac84ac4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -54,7 +54,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C select SND_SOC_CX20442 if TTY - select SND_SOC_DA7210 if I2C + select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C -- 2.1.0

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[alsa-devel] [PATCH] ASoC: samsung: wolfson: Improve compile test coverage
by Lars-Peter Clausen 04 May '15

04 May '15

While the the Wolfson machine drivers have a runtime dependency on a specific machine there is no compile time dependency. Allow to lets those drivers to be selected when COMPILE_TEST is selected to improve the compile time test coverage. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/samsung/Kconfig | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 0632a36..3744c9e 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -174,7 +174,8 @@ config SND_SOC_SMDK_WM8994_PCM config SND_SOC_SPEYSIDE tristate "Audio support for Wolfson Speyside" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C && SPI_MASTER + depends on SND_SOC_SAMSUNG && I2C && SPI_MASTER + depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select SND_SOC_WM8996 select SND_SOC_WM9081 @@ -183,13 +184,15 @@ config SND_SOC_SPEYSIDE config SND_SOC_TOBERMORY tristate "Audio support for Wolfson Tobermory" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && INPUT && I2C + depends on SND_SOC_SAMSUNG && INPUT && I2C + depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select SND_SOC_WM8962 config SND_SOC_BELLS tristate "Audio support for Wolfson Bells" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA && I2C && SPI_MASTER + depends on SND_SOC_SAMSUNG && MFD_ARIZONA && I2C && SPI_MASTER + depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select SND_SOC_WM5102 select SND_SOC_WM5110 @@ -199,14 +202,16 @@ config SND_SOC_BELLS config SND_SOC_LOWLAND tristate "Audio support for Wolfson Lowland" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C + depends on SND_SOC_SAMSUNG && I2C + depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select SND_SOC_WM5100 select SND_SOC_WM9081 config SND_SOC_LITTLEMILL tristate "Audio support for Wolfson Littlemill" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C + depends on SND_SOC_SAMSUNG && I2C + depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select MFD_WM8994 select SND_SOC_WM8994 -- 1.8.0

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Re: [alsa-devel] [PATCH v3 3/8] mfd: arizona: Add support for WM8998 and WM1814
by Paul Bolle 04 May '15

04 May '15

Something I didn't notice when v2 came along. On Fri, 2015-05-01 at 16:15 +0100, Richard Fitzgerald wrote: > --- a/drivers/mfd/Kconfig > +++ b/drivers/mfd/Kconfig > +config MFD_WM8998 > + bool "Wolfson Microelectronics WM8998" > + depends on MFD_ARIZONA > + help > + Support for Wolfson Microelectronics WM8998 low power audio SoC > + > config MFD_WM8400 > bool "Wolfson Microelectronics WM8400" > select MFD_CORE > --- a/drivers/mfd/Makefile > +++ b/drivers/mfd/Makefile > ifneq ($(CONFIG_MFD_WM8997),n) > obj-$(CONFIG_MFD_ARIZONA) += wm8997-tables.o > endif > +ifneq ($(CONFIG_MFD_WM8998),n) > +obj-$(CONFIG_MFD_ARIZONA) += wm8998-tables.o > +endif Playing a bit with the current version of drivers/mfd/Makefile it seems these ifneq ($(CONFIG_MFD_WM[...]),n) tests will always be true. Because, as far as I know, the Kconfig macros used in those tests will either be "y" or the empty string, but never "n". (I don't speak Makefilese fluent enough to know how to add simple debugging prints for the values of the CONFIG_MFD_WM[...] macros to actually test this, so correct me if I'm wrong here.) So I think that, effectively, these wm[...]-tables.o objects will always be built if CONFIG_MFD_ARIZONA is set. Is that the intention? If not, perhaps these test should read ifeq ($(CONFIG_MFD_WM[...]),y) Thanks, Paul Bolle

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[alsa-devel] [PATCH v5 0/2] Add support for select accessory detect mode to HPDETL or HPDETR
by Inha Song 04 May '15

04 May '15

This set of patches adds support for select accessory detect mode to HPDETL or HPDETR. Changes in v5: - Rebased on extcon-next branch. Changes in v4: - Set the hpdet_channel default value to HPDETL in variable declaration. - Fix the indentation in binding documentation. Changes in v3: - Set the hpdet_channel default value to HPDETL If the value is unknown or invalid. Changes in v2: - Use the value in pdata instead of hpdet_channel in extcon_info. - Wrap arizona_extcon_of_get in IS_ENABLED(CONFIG_OF). - Change hpdet_channel type to unsigned from signed in pdata. - Move ARIZONA_ACCDET_MODE_* define to dt-binding header and directly set it to pdata. Inha Song (2): extcon: arizona: Add support for select accessory detect mode when headphone detection mfd: arizona: Update DT binding to support hpdet channel Documentation/devicetree/bindings/mfd/arizona.txt | 6 ++++ drivers/extcon/extcon-arizona.c | 38 ++++++++++++++++++----- include/dt-bindings/mfd/arizona.h | 4 +++ include/linux/mfd/arizona/pdata.h | 3 ++ 4 files changed, 43 insertions(+), 8 deletions(-) -- 2.0.0.390.gcb682f8

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[alsa-devel] [RESEND RFC PATCH 0/3] ASoC: Mediatek: Add support for MT8173 SOC
by Koro Chen 04 May '15

04 May '15

Refactor and resend this series because of the 60KB limitation. This adds basic support for the Mediatek AFE unit for MT8173 SoC. This patch is based on Linux 4.0-rc1, Sascha's clk patch [1], and his SCPSYS power patch [2]. The AFE unit comprises several memory interfaces that communicate with CPU, a multi input multi output digital audio interconnect, and several external interfaces (e.g. I2S). The AFE unit is a DMA master, so no external DMA engine is needed. Each external interface is presented as a DAI to ASoC. A memory interface must be connected via the interconnect to an external interface. The connection pathes are configured through the device tree. [1] http://lists.infradead.org/pipermail/linux-mediatek/2015-March/000079.html [2] http://lists.infradead.org/pipermail/linux-mediatek/2015-March/000132.html Koro Chen (3): ASoC: mediatek: Add binding support for AFE driver ASoC: mediatek: Add AFE connection control ASoC: mediatek: Add AFE platform driver .../devicetree/bindings/sound/mtk-afe-pcm.txt | 105 ++ include/dt-bindings/sound/mtk-afe.h | 36 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/mediatek/Kconfig | 8 + sound/soc/mediatek/Makefile | 2 + sound/soc/mediatek/mtk-afe-common.h | 105 ++ sound/soc/mediatek/mtk-afe-connection.c | 416 ++++++ sound/soc/mediatek/mtk-afe-connection.h | 30 + sound/soc/mediatek/mtk-afe-pcm.c | 1497 ++++++++++++++++++++ 10 files changed, 2201 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mtk-afe-pcm.txt create mode 100644 include/dt-bindings/sound/mtk-afe.h create mode 100644 sound/soc/mediatek/Kconfig create mode 100644 sound/soc/mediatek/Makefile create mode 100644 sound/soc/mediatek/mtk-afe-common.h create mode 100644 sound/soc/mediatek/mtk-afe-connection.c create mode 100644 sound/soc/mediatek/mtk-afe-connection.h create mode 100644 sound/soc/mediatek/mtk-afe-pcm.c --

3 23

Re: [alsa-devel] Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 03 May '15

03 May '15

> > > My problem is that when I set channel output to "Analog output surround > > > 5.1" > > > (or 2.1 or 4.1), I hear no sound in front speakers, but in case of 5.1 or > > > 4.1 You need pulseaudio lfe-filter for 2.1 http://cgit.freedesktop.org/pulseaudio/pulseaudio/log/?qt=grep&q=lfe-filter do you mean surround21 need 6 channel instead of channels with 5.1 internal /external speaker ? http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=48f1b308cc66152e… > > > I can hear sound in rear speakers. I use kernel 3.19.5 and it didn't > > > happen in > > > earlier kernels that is 3.18 and in 3.17 for sure. > > > > > > My configuration is: > > > OS: Fedora 21 64 bit > > > SoundCard: ALC888 > > > Model name: acer-aspire-4930g > > > Laptop: Acer Aspire 5930g > > > Speakers: external 5.1 with 3 jacks > > > Sound channel configuration module is standard configuration module from > > > KDE. > > > > > > Additional info: > > > 1) When channel output is 5.1 and I unplug green jack and leave two others > > > jacks connected then I can hear sound in laptop internal speakers. auto mute only check headphone jack or line out jack but not those retasked mio jacks > > > 2) Earlier (kernel 3.18 and 3.17) I didn't have possibility to choose 2.1 > > > configuration and additionally to 4.1 configuration I could also choose > > > 4.0 > > > configuration. > > > > > > You need to switch "Channel mode" to "6 ch" if you need to use external > > > > 5.1 speakers since you have 5.1 internal speakers > > > > this change pin-ctl of headphone jack, mic jack and line in to OUT > > > > post the output of alsa-info.sh > > I always switched to "6 ch" through alsamixer. It always worked as expected, > but recently I've got problems with it so it regressed somewhere. > > > My alsa-info.sh with "6 ch" set through alsamixer and "Analog output surround > 5.1 + Analog input stereo" set through KDE configuration module. > http://www.alsa-project.org/db/?f=ff3f5a1f7f3e4662c3fa9bcf0158474a076b4cc4 > > With that I hear all channels except front left and front right. autoconfig for ALC888: line_outs=3 (0x14/0x17/0x16/0x0/0x0) type:speaker speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) hp_outs=1 (0x15/0x0/0x0/0x0/0x0) mono: mono_out=0x0 dig-out=0x1e/0x0 inputs: Mic=0x18 Internal Mic=0x12 Line=0x1a ==> Best config: lo_type=2, wired=1, mio=1 multi_outs = 15/0/0/0 : 2/3/4/0 (type HP) out path: depth=3 '02:0c:15' multi_ios(2) = 1a/18 : 3/4 mio path: depth=3 '03:0d:1a' mio path: depth=3 '04:0e:18' spk_outs = 14/17/16/0 : 5/25/4/0 spk path: depth=3 '05:0f:14' spk path: depth=3 '25:26:17' spk path: depth=3 '04:0e:16' Node 0x15 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Front Headphone Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001003e: IN OUT HP EAPD Detect Trigger EAPD 0x2: EAPD Pin Default 0x0221101f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 it is still headphone jack with pin-ctl : 0xc0 OUT HP while mic jack and line in jack 's pin ctl are changed to 0x40 OUT channel mode only change two mio jacks from IN to OUT May need to use add_out_jack_mode=1 instead of add_jack_modes=1 when there are Mic and line in jacks Use Headphone Jack Mode control to switch HP to Line Out for pin-ctl and EAPD https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=f81… > To exclude that my speakers are broken or wrong connected: I get sound from > front left and front right if i switch in KDE configuration module to "Analog > duplex stereo", but then I get no sound from rest of my speakers. http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… pulseaudio mute external center/lfe controls when headphone jack is plugged since pulseaudio not aware that the role of headphone has been changed [Element Center] switch = off volume = off [Element LFE] switch = off volume = off > > > KDE configuration module lets me choose "connector" configuration but only in > stereo mode. > In stereo mode I can choose between "headphones" and "speakers". > If it is set to "headphones" I hear sound in front left and front right > speakers. > If it is set to "speakers" I hear sound in rear left and rear right speakers. > you need to switch "channel mode" to "2 ch" if you want to use your 5.1 internal speaker https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=786… There are difference between static model and generic parser

1 0

Re: [alsa-devel] [PATCH] ALSA: hda - Add dock support for ThinkPad X250 (17aa:2226)
by Takashi Iwai 02 May '15

02 May '15

At Sat, 11 Apr 2015 09:31:35 +0200, Yves-Alexis Perez wrote: > > This model uses the same dock port as the previous generation. > > Signed-off-by: Yves-Alexis Perez <corsac(a)debian.org> Thanks, applied with Cc to stable. But, at the next time, please put maintainers to Cc. Takashi > --- > sound/pci/hda/patch_realtek.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 7438213..2087728 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -5030,6 +5030,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { > 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, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), > + SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), > SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), > SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), > SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), > -- > 2.1.4 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ >

3 2

[alsa-devel] [PATCH] ALSA: jack: fix a randconfig build issue
by Jie Yang 02 May '15

02 May '15

Building errors reported such as below when 'CONFIG_INPUT=m': ...undefined reference to `input_xxx'... Here change to enable SND_JACK selectively to fix the issue. Also remove the config 'SND_HDA_INPUT_JACK' which won't be used anymore. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/pci/hda/Kconfig | 10 +--------- sound/pci/hda/hda_codec.h | 2 -- 2 files changed, 1 insertion(+), 11 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 5c296d3..117bf5c 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -4,7 +4,7 @@ config SND_HDA tristate select SND_PCM select SND_VMASTER - select SND_JACK + select SND_JACK if INPUT=y || INPUT=SND select SND_HDA_CORE config SND_HDA_INTEL @@ -84,14 +84,6 @@ config SND_HDA_INPUT_BEEP_MODE Set 1 to always enable the digital beep interface for HD-audio by default. -config SND_HDA_INPUT_JACK - bool "Support jack plugging notification via input layer" - depends on INPUT=y || INPUT=SND - select SND_JACK - help - Say Y here to enable the jack plugging notification via - input layer. - config SND_HDA_PATCH_LOADER bool "Support initialization patch loading for HD-audio" select FW_LOADER diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 0f8b6b5..ed7e9cf 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -274,10 +274,8 @@ struct hda_codec { unsigned long jackpoll_interval; /* In jiffies. Zero means no poll, rely on unsol events */ struct delayed_work jackpoll_work; -#ifdef CONFIG_SND_HDA_INPUT_JACK /* jack detection */ struct snd_array jacks; -#endif int depop_delay; /* depop delay in ms, -1 for default delay time */ -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: jack: fix a randconfig build issue
by Jie Yang 02 May '15

02 May '15

There may be no input_dev for jack(e.g. phantom jack), when CONFIG_INPUT is not selected, building errors such as: undefined reference to `input_xxx'... Here add #ifdef judgement to fix the issue. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- include/sound/jack.h | 2 ++ sound/core/jack.c | 40 ++++++++++++++++++++++++++-------------- 2 files changed, 28 insertions(+), 14 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 23bede1..c1010f9 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -72,7 +72,9 @@ enum snd_jack_types { #define SND_JACK_SWITCH_TYPES 6 struct snd_jack { +#ifdef CONFIG_INPUT struct input_dev *input_dev; +#endif struct list_head kctl_list; struct snd_card *card; int registered; diff --git a/sound/core/jack.c b/sound/core/jack.c index eb66327..3a1901e 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -32,6 +32,7 @@ struct snd_jack_kctl { unsigned int mask_bits; /* only masked status bits are reported via kctl */ }; +#ifdef CONFIG_INPUT static int jack_switch_types[SND_JACK_SWITCH_TYPES] = { SW_HEADPHONE_INSERT, SW_MICROPHONE_INSERT, @@ -40,9 +41,11 @@ static int jack_switch_types[SND_JACK_SWITCH_TYPES] = { SW_VIDEOOUT_INSERT, SW_LINEIN_INSERT, }; +#endif static int snd_jack_dev_disconnect(struct snd_device *device) { +#ifdef CONFIG_INPUT struct snd_jack *jack = device->device_data; if (!jack->input_dev) @@ -54,7 +57,9 @@ static int snd_jack_dev_disconnect(struct snd_device *device) input_unregister_device(jack->input_dev); else input_free_device(jack->input_dev); + jack->input_dev = NULL; +#endif return 0; } @@ -83,11 +88,12 @@ static int snd_jack_dev_register(struct snd_device *device) { struct snd_jack *jack = device->device_data; struct snd_card *card = device->card; - int err, i; + int err = 0; snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); +#ifdef CONFIG_INPUT if (!jack->input_dev) return 0; @@ -98,7 +104,7 @@ static int snd_jack_dev_register(struct snd_device *device) jack->input_dev->dev.parent = snd_card_get_device_link(card); /* Add capabilities for any keys that are enabled */ - for (i = 0; i < ARRAY_SIZE(jack->key); i++) { + for (int i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = SND_JACK_BTN_0 >> i; if (!(jack->type & testbit)) @@ -113,6 +119,7 @@ static int snd_jack_dev_register(struct snd_device *device) err = input_register_device(jack->input_dev); if (err == 0) jack->registered = 1; +#endif return err; } @@ -209,7 +216,6 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, struct snd_jack *jack; struct snd_jack_kctl *jack_kctl = NULL; int err; - int i; static struct snd_device_ops ops = { .dev_free = snd_jack_dev_free, .dev_register = snd_jack_dev_register, @@ -230,26 +236,31 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, /* don't creat input device for phantom jack */ if (!phantom_jack) { +#ifdef CONFIG_INPUT jack->input_dev = input_allocate_device(); if (jack->input_dev == NULL) { err = -ENOMEM; - goto fail_input; + goto error; } jack->input_dev->phys = "ALSA"; - jack->type = type; - - for (i = 0; i < SND_JACK_SWITCH_TYPES; i++) + for (int i = 0; i < SND_JACK_SWITCH_TYPES; i++) if (type & (1 << i)) input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); +#endif + jack->type = type; } err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); - if (err < 0) - goto fail_input; + if (err < 0) { +#ifdef CONFIG_INPUT + input_free_device(jack->input_dev); +#endif + goto error; + } jack->card = card; INIT_LIST_HEAD(&jack->kctl_list); @@ -261,8 +272,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return 0; -fail_input: - input_free_device(jack->input_dev); +error: kfree(jack->id); kfree(jack); return err; @@ -281,11 +291,13 @@ EXPORT_SYMBOL(snd_jack_new); */ void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { +#ifdef CONFIG_INPUT WARN_ON(jack->registered); if (!jack->input_dev) return; jack->input_dev->dev.parent = parent; +#endif } EXPORT_SYMBOL(snd_jack_set_parent); @@ -340,7 +352,6 @@ EXPORT_SYMBOL(snd_jack_set_key); void snd_jack_report(struct snd_jack *jack, int status) { struct snd_jack_kctl *jack_kctl; - int i; if (!jack) return; @@ -349,10 +360,11 @@ void snd_jack_report(struct snd_jack *jack, int status) snd_kctl_jack_report(jack->card, jack_kctl->kctl, status & jack_kctl->mask_bits); +#ifdef CONFIG_INPUT if (!jack->input_dev) return; - for (i = 0; i < ARRAY_SIZE(jack->key); i++) { + for (int i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = SND_JACK_BTN_0 >> i; if (jack->type & testbit) @@ -369,7 +381,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); - +#endif } EXPORT_SYMBOL(snd_jack_report); -- 1.9.1

3 6

Re: [alsa-devel] [PATCH 1/3] ASoC: Add gtm601 codec driver
by Mark Brown 01 May '15

01 May '15

On Thu, Apr 30, 2015 at 11:28:01PM +0200, Marek Belisko wrote: > select SND_SOC_WM9705 if SND_SOC_AC97_BUS > select SND_SOC_WM9712 if SND_SOC_AC97_BUS > select SND_SOC_WM9713 if SND_SOC_AC97_BUS > + select SND_SOC_GTM601 > help > +config SND_SOC_GTM601 > + tristate 'GTM601 UMTS modem audio codec' > + Keep Makefile and Kconfig sorted. > +/* > + * Note this is a simple chip with no configuration interface, sample rate is > + * determined automatically by examining the Master clock and Bit clock ratios > + */ > +#define GTM601_RATES (SNDRV_PCM_RATE_8000) Does it or doesn't it support multiple sample rates?

2 3

Re: [alsa-devel] [PATCH 4/4] ASoC: samsung: Constify platform_device_id
by Mark Brown 01 May '15

01 May '15

On Sat, May 02, 2015 at 01:00:14AM +0900, Krzysztof Kozlowski wrote: > The platform_device_id is not modified by the driver and core uses it as > const. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH 3/4] ASoC: fsl: Constify platform_device_id
by Mark Brown 01 May '15

01 May '15

On Sat, May 02, 2015 at 01:00:13AM +0900, Krzysztof Kozlowski wrote: > The platform_device_id is not modified by the driver and core uses it as > const. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH 2/4] ASoC: bt-sco: Constify platform_device_id
by Mark Brown 01 May '15

01 May '15

On Sat, May 02, 2015 at 01:00:12AM +0900, Krzysztof Kozlowski wrote: > The platform_device_id is not modified by the driver and core uses it as > const. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH 1/4] ASoC: au1x: Constify platform_device_id
by Mark Brown 01 May '15

01 May '15

On Sat, May 02, 2015 at 01:00:11AM +0900, Krzysztof Kozlowski wrote: > The platform_device_id is not modified by the driver and core uses it as > const. Applied, thanks.

1 0

[alsa-devel] [PATCH RFT] ASoC: mc13783: Fix wrong mask value used in mc13xxx_reg_rmw() calls
by Axel Lin 01 May '15

01 May '15

mc13xxx_reg_rmw() won't change any bit if passing 0 to the mask field. Pass AUDIO_SSI_SEL instead of 0 for the mask field to set AUDIO_SSI_SEL bit. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi, I don't have this h/w for test, so please help review and test this patch. Thanks. sound/soc/codecs/mc13783.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 2ffb9a0..3d44fc5 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -623,14 +623,14 @@ static int mc13783_probe(struct snd_soc_codec *codec) AUDIO_SSI_SEL, 0); else mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC, - 0, AUDIO_SSI_SEL); + AUDIO_SSI_SEL, AUDIO_SSI_SEL); if (priv->dac_ssi_port == MC13783_SSI1_PORT) mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC, AUDIO_SSI_SEL, 0); else mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC, - 0, AUDIO_SSI_SEL); + AUDIO_SSI_SEL, AUDIO_SSI_SEL); return 0; } -- 2.1.0

3 2

[alsa-devel] [PATCH 1/3] ALSA: hda: add hdac_adsp_enable module flag
by Vinod Koul 01 May '15

01 May '15

Some Intel HDA controllers sport a DSP. These systems can also be enabled with ASoC HDA driver as well. So add a flag in hda-core to enable/disable aDSP This flag for now is false, and should be true once the ASoC based systems mature. The integrators/OS vendors can configure this flag based on system preference. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/hdaudio.h | 11 +++++++++++ sound/hda/hda_bus_type.c | 4 ++++ 2 files changed, 15 insertions(+) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index b97c59eab7ab..015bec1079f9 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -12,6 +12,17 @@ #include <sound/memalloc.h> #include <sound/hda_verbs.h> +/* + * hdac_adsp_enable: exported HD-A aDSP enable configuration. + * + * Some Intel HDA controllers sport a DSP, for these platform we can bypass + * aDSP and use as regular HDA controller or enable aDSP and use aDSP + * along with I2S codecs etc. + * hdac_adsp_enable would enable the aDSP based HDA controller if the + * platform supports it + */ +extern bool hdac_adsp_enable; + /* codec node id */ typedef u16 hda_nid_t; diff --git a/sound/hda/hda_bus_type.c b/sound/hda/hda_bus_type.c index 519914a12e8a..80e0570ffbf4 100644 --- a/sound/hda/hda_bus_type.c +++ b/sound/hda/hda_bus_type.c @@ -10,6 +10,10 @@ MODULE_DESCRIPTION("HD-audio bus"); MODULE_LICENSE("GPL"); +bool hdac_adsp_enable = false; +module_param(hdac_adsp_enable, bool, 0444); +MODULE_PARM_DESC(hdac_adsp_enable, "Enable aDSP on Intel HDA based systems"); + static int hda_bus_match(struct device *dev, struct device_driver *drv) { struct hdac_device *hdev = dev_to_hdac_dev(dev); -- 1.9.1

4 14

Re: [alsa-devel] [PATCH 1/3] ASoC: Add gtm601 codec driver
by Paul Bolle 01 May '15

01 May '15

Just a nit: a license mismatch. On Thu, 2015-04-30 at 23:28 +0200, Marek Belisko wrote: > --- /dev/null > +++ b/sound/soc/codecs/gtm601.c > + * 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 states the license is GPL v2 or later. > +MODULE_LICENSE("GPL v2"); And, according to include/linux/module.h, this states the license is (just) GPL v2. So I think either the comment at the top of this file or the ident used in the MODULE_LICESE() macro should change. Paul Bolle

2 1

Re: [alsa-devel] [PATCH 1/2] ASoC: zx: Add ZTE zx296702 pcm support
by Mark Brown 01 May '15

01 May '15

On Thu, Apr 30, 2015 at 12:07:29PM +0800, Jun Nie wrote: > +static const struct snd_pcm_hardware snd_zx_hardware = { > + .info = SNDRV_PCM_INFO_MMAP | > + SNDRV_PCM_INFO_MMAP_VALID | > + SNDRV_PCM_INFO_INTERLEAVED | > + SNDRV_PCM_INFO_PAUSE | > + SNDRV_PCM_INFO_RESUME | > + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, > + .period_bytes_min = 32, > + .period_bytes_max = 16 * 1024, > + .periods_min = 2, > + .periods_max = 32, > + .buffer_bytes_max = 60 * 1024, > +}; Can the generic DMA code not work this out by querying the dmaengine driver? > +int zx_pcm_platform_register(struct device *dev) > +{ > + return devm_snd_dmaengine_pcm_register(dev, &zx_dmaengine_pcm_config, > + SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME | > + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); _NO_RESIDUE has been removed in current code, the core figures it out automatically. Like I said looking at the driver code I'm not clear why this is specifying _CUSTOM_CHANNEL_NAME either, it looked like the driver just used the one standard channel name.

1 0

Re: [alsa-devel] [PATCH 2/2] ASoC: zx: Add zx296702 SPDIF support
by Mark Brown 01 May '15

01 May '15

On Thu, Apr 30, 2015 at 12:07:30PM +0800, Jun Nie wrote: > Add zx296702 SPDIF support This is adding a new device with DT bindings but there is no bindings document. All new DT bindings must be documented. > --- /dev/null > +++ b/include/sound/zx_hdmi_audio.h > @@ -0,0 +1,7 @@ > +#ifndef __ZX_HDMI_AUDIO_H__ > +#define __ZX_HDMI_AUDIO_H__ > + > +int zx_hdmi_audio_cfg(int audio_codec, int audio_way, > + u32 sample_rate, u32 sample_len); > +void zx_hdmi_audio_en(int on); > +#endif /* __ZX_HDMI_AUDIO_H__ */ So this isn't a S/PDIF controller but rather a HDMI controller and from the looks of it one with yet another custom interface between the video part and the audio part. As I keep saying I'd really like to see more code sharing between the various controllers doing this. Please take a look at some of the other controllers and see what can be shared, and also at this: http://thread.gmane.org/gmane.comp.video.dri.devel/126588 series from Russell King. > +config ZX296702_SPDIF > + bool "ZX296702 spdif" > + select ZX_PCM_DMA > + help > + Say Y or M if you want to add support for codecs attached to the > + zx296702 spdif interface The indentation above looks inconsistent - probably a mix of tabs and spaces. > +static struct snd_dmaengine_dai_dma_data zx_spdif_pcm_stereo_out = { > + .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, > + .chan_name = "tx", > + .maxburst = 8, > +}; The DMA driver said it needed custom channel names but this looks like the standard "tx" for transmit? > +static int zx_spdif_dai_probe(struct snd_soc_dai *dai) > +{ > + struct zx_spdif_info *zx_spdif = dev_get_drvdata(dai->dev); > + > + snd_soc_dai_set_drvdata(dai, zx_spdif); > + zx_spdif_pcm_stereo_out.addr = zx_spdif->mapbase + ZX_DATA; > + snd_soc_dai_init_dma_data(dai, &zx_spdif_pcm_stereo_out, NULL); > + return 0; > +} This appears to be modifying a global variable rather than some driver data, please don't do that - store anything discovered per device at runtime in the driver datta. > +static void zx_spdif_chanstats(void __iomem *base, unsigned int rate) > +{ > + u32 cstas1; > + > + switch (rate) { > + case 22050: > + default: > + return; > + } This should handle and report errrors, not ignore them. > + return -EINVAL; > + } > + if (2 == params_channels(params)) > + val |= ZX_CTRL_DOUBLE_TRACK; > + else > + val |= ZX_CTRL_LEFT_TRACK; > + writel_relaxed(val, zx_spdif->reg_base + ZX_CTRL); Missing blank line between this and the above block and please reverse the arguments for the == - it's the more normal kernel coding style. > +#if defined(CONFIG_ZX_HDMI_SND_SPDIF) > + sample_len = params_physical_width(params); > + zx_hdmi_audio_cfg(1, 0, params_rate(params), sample_len); > +#endif But this is the S/PDIF dirver... in any case, this looks like something that depends on the hardware rather than something that should be a compile time option. We should be able to boot the same kernel on many systems. > + val = readl_relaxed(base + ZX_CTRL); > + val &= ~(ZX_CTRL_ENB_MASK | ZX_CTRL_TX_MASK); > + val |= on ? (ZX_CTRL_TX_OPEN | ZX_CTRL_ENB) > + : (ZX_CTRL_TX_CLOSE | ZX_CTRL_DNB); > + writel_relaxed(val, base + ZX_CTRL); Please don't use the ternery operator like this, it does nothing for legibility. > + if (clk_prepare_enable(zx_spdif->dai_clk)) > + return -EIO; Don't discard return values, pass them back. You should probably also be managing this clock at runtime to save power. > + &zx_spdif_dai, 1); > + if (ret) { > + dev_err(&pdev->dev, "Register DAI failed\n"); Print error codes as well.

1 0

[alsa-devel] [PATCH v2 1/5] ASoC: Correct typo in SOC_VALUE_ENUM_SINGLE macro
by Charles Keepax 01 May '15

01 May '15

xnitmes is clearly intended to be xnitems, but all other macros just refer to this as xitems, so change it to that. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v1: - Use first entry of values array as off value for the mux Thanks, Charles include/sound/soc.h | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a73d855..45cfd69 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -190,8 +190,8 @@ #define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xitems, xtexts, xvalues) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .mask = xmask, .items = xitems, .texts = xtexts, .values = xvalues} -#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xnitmes, xtexts, xvalues) \ - SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xnitmes, xtexts, xvalues) +#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xitems, xtexts, xvalues) \ + SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xitems, xtexts, xvalues) #define SOC_ENUM_SINGLE_VIRT(xitems, xtexts) \ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, xitems, xtexts) #define SOC_ENUM(xname, xenum) \ -- 1.7.2.5

2 7

[alsa-devel] [PATCH] ALSA: hda - Add headphone quirk for Lifebook E752
by Takashi Iwai 01 May '15

01 May '15

Fujitsu Lifebook E752 laptop needs a similar quirk done for Lifebook T731. Otherwise the headphone is always muted. Reported-and-tested-by: Christian Weber <we_chris(a)hotmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e2afd53cc14c..eb2fc79cbd32 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5119,6 +5119,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9099, "Sony VAIO S13", ALC275_FIXUP_SONY_DISABLE_AAMIX), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x10cf, 0x15dc, "Lifebook T731", ALC269_FIXUP_LIFEBOOK_HP_PIN), + SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_BXBT2807_MIC), -- 2.3.6

1 0

[alsa-devel] [PATCH] ASoC: fix simple_return.cocci warnings
by kbuild test robot 30 Apr '15

30 Apr '15

sound/soc/codecs/adau1977.c:496:5-8: WARNING: end returns can be simpified Simplify a trivial if-return sequence. Possibly combine with a preceding function call. Generated by: scripts/coccinelle/misc/simple_return.cocci CC: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- adau1977.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) --- a/sound/soc/codecs/adau1977.c +++ b/sound/soc/codecs/adau1977.c @@ -493,10 +493,7 @@ static int adau1977_set_bias_level(struc break; } - if (ret) - return ret; - - return 0; + return ret; } static int adau1977_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,

2 1

[alsa-devel] [PATCH] ASoC: fix simple_return.cocci warnings
by kbuild test robot 30 Apr '15

30 Apr '15

sound/soc/codecs/ssm2518.c:521:5-8: WARNING: end returns can be simpified Simplify a trivial if-return sequence. Possibly combine with a preceding function call. Generated by: scripts/coccinelle/misc/simple_return.cocci CC: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- ssm2518.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -518,10 +518,7 @@ static int ssm2518_set_bias_level(struct break; } - if (ret) - return ret; - - return 0; + return ret; } static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,

2 2

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Correct pm status check in suspend callback
by Peter Ujfalusi 30 Apr '15

30 Apr '15

pm_runtime_enabled() will only tell if the pm runtime has been enabled for the device, which is done at probe time but will not tell the actual power state of the device. pm_runtime_active() provides this information. This patch fixes a kernel crash when doing suspend when McASP is not active. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Mark, can you schedule this for the next -rc? Thank you, Peter sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index ddf7e97d8f0b..1ab3fcdc6bba 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1247,7 +1247,7 @@ static int davinci_mcasp_suspend(struct snd_soc_dai *dai) u32 reg; int i; - context->pm_state = pm_runtime_enabled(mcasp->dev); + context->pm_state = pm_runtime_active(mcasp->dev); if (!context->pm_state) pm_runtime_get_sync(mcasp->dev); -- 2.3.5

2 1

[alsa-devel] [PATCH v2 0/3] Calculate BCLK using TDM slots and remove channels rule
by Jyri Sarha 30 Apr '15

30 Apr '15

The first and the last patch are excatly the same as in the first series, but in the middle thee is a new patch to constraint out unsupported channel counts in multiple serializers case. - The first patch is a bug fix and also fixes Dan Carpenter's concern about stack usage in davinci_mcasp_hw_rule_channels() - The second is also a bug fix, even thou the bug has existed in the dirver for a very long time. It is just that nobody has used multiple serializers before. - The last patch is a simple optimization All the patches have been tested are internally reviewed. Jyri Sarha (3): ASoC: davinci-mcasp: Calculate BCLK using TDM slots and remove channels rule ASoC: davinci-mcasp: Channel count constraints for multi-serializer case ASoC: davinci-macsp: Optimize implicit BLCK sample-rate rule sound/soc/davinci/davinci-mcasp.c | 167 ++++++++++++++++++++------------------ 1 file changed, 90 insertions(+), 77 deletions(-) -- 1.9.1

2 4

[alsa-devel] ASoC fixes for v4.1
by Mark Brown 30 Apr '15

30 Apr '15

The following changes since commit b787f68c36d49bb1d9236f403813641efa74a031: Linux 4.1-rc1 (2015-04-26 17:59:10 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.1-rc1 for you to fetch changes up to d839c98f98826f0c13e46d5a4cecb46dd357b50e: Merge remote-tracking branches 'asoc/fix/rt5677', 'asoc/fix/samsung' and 'asoc/fix/tfa9879' into asoc-linus (2015-04-29 13:37:31 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.1 A few fixes for v4.1, none earth shattering and mostly driver related except for one change to fix !PM builds for Intel platforms which is done by adding stubs in the core so other platforms don't run into the same issue. ---------------------------------------------------------------- Axel Lin (1): ASoC: rt5645: Fix mask for setting RT5645_DMIC_2_DP_GPIO12 bit Bard Liao (2): ASoC: rt5677: add register patch for PLL ASoC: rt5677: fixed wrong DMIC ref clock Charles Keepax (1): ASoC: dapm: Enable autodisable on SOC_DAPM_SINGLE_TLV_AUTODISABLE Fabio Estevam (1): ASoC: fsl_ssi: Fix platform_get_irq() error handling Fang, Yang A (1): ASoC: rt5645: Add ACPI match ID Jie Yang (1): ASoC: add static inline funcs to fix a compiling issue Jin Yao (1): ASoC: Intel: Remove invalid kfree of devm allocated data Kuninori Morimoto (1): ASoC: rsnd: set dmaen->chan = NULL when error case Mark Brown (5): Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/email', 'asoc/fix/fsl-ssi', 'asoc/fix/pm', 'asoc/fix/qcom' and 'asoc/fix/rcar' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5677', 'asoc/fix/samsung' and 'asoc/fix/tfa9879' into asoc-linus Rajeev Kumar (1): ASoC: Update email-id of Rajeev Kumar Takashi Iwai (1): ASoC: qcom: Return an error for invalid PCM trigger command Vinod Koul (1): ASoC: Intel: fix the makefile for atom code Wei Yongjun (3): ASoC: tfa9879: Fix return value check in tfa9879_i2c_probe() ASoC: samsung: s3c24xx-i2s: Fix return value check in s3c24xx_iis_dev_probe() ASoC: Intel: sst_byt: remove kfree for memory allocated with devm_kzalloc include/sound/designware_i2s.h | 2 +- include/sound/soc-dapm.h | 2 +- include/sound/soc.h | 12 ++++++++++++ include/sound/spear_dma.h | 2 +- sound/soc/codecs/rt5645.c | 13 ++++++++++++- sound/soc/codecs/rt5677.c | 5 ++++- sound/soc/codecs/tfa9879.c | 4 ++-- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/intel/Makefile | 2 +- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 1 - sound/soc/intel/haswell/sst-haswell-ipc.c | 1 - sound/soc/qcom/lpass-cpu.c | 2 +- sound/soc/samsung/s3c24xx-i2s.c | 4 ++-- sound/soc/sh/rcar/dma.c | 1 + 14 files changed, 39 insertions(+), 14 deletions(-)

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[alsa-devel] [PATCH 1/5] ASoC: Correct typo in SOC_VALUE_ENUM_SINGLE macro
by Charles Keepax 30 Apr '15

30 Apr '15

xnitmes is clearly intended to be xnitems, but all other macros just refer to this as xitems, so change it to that. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a73d855..45cfd69 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -190,8 +190,8 @@ #define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xitems, xtexts, xvalues) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .mask = xmask, .items = xitems, .texts = xtexts, .values = xvalues} -#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xnitmes, xtexts, xvalues) \ - SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xnitmes, xtexts, xvalues) +#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xitems, xtexts, xvalues) \ + SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xitems, xtexts, xvalues) #define SOC_ENUM_SINGLE_VIRT(xitems, xtexts) \ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, xitems, xtexts) #define SOC_ENUM(xname, xenum) \ -- 1.7.2.5

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[alsa-devel] [RFC 0/7] ASoC: hda - ASoC DSP widget event handlers
by Vinod Koul 30 Apr '15

30 Apr '15

The aDSP is triggered by DAPM. The DAPM code is atm dependent upon the DFW aka tplg RFC send by Liam. We will post missing tplg to driver links once that series is accepted This series is dependent upon a) Takashi topic/hda which has core hdac splitup b) SKL PCM driver series, which adds the SKL HDA Platform driver c) SKL IPC series which sends IPCs to aDSP This last series adds the missing link between b) and c). Comments most welcome -- ~Vinod Jeeja KP (7): ASoC: hda: Add skl dsp init and registering with SST IPC lib ASoC: hda: add hepler to configure module params ASoC: hda: add FW module init/bind IPC ASoC: hda: add FW pipe create/delete/set_pipe_state IPC ASOC: hda: add DSP platfrom controls widget event handlers ASoC: hda: Add support for SSP register settings ASoC: hda: Apply dai params_fixup for DSP widgets include/sound/soc-hda-sst-dsp.h | 3 + sound/soc/hda/Kconfig | 1 + sound/soc/hda/Makefile | 3 +- sound/soc/hda/hda_dsp_controls.c | 913 ++++++++++++++++++++++++++++++++++++ sound/soc/hda/hda_dsp_controls.h | 338 +++++++++++++ sound/soc/hda/hda_dsp_ssp_config.c | 290 ++++++++++++ sound/soc/hda/hda_dsp_ssp_config.h | 423 +++++++++++++++++ sound/soc/hda/hda_skl.h | 108 ++++- sound/soc/hda/hda_soc_dsp.c | 702 +++++++++++++++++++++++++++ 9 files changed, 2767 insertions(+), 14 deletions(-) create mode 100644 sound/soc/hda/hda_dsp_controls.c create mode 100644 sound/soc/hda/hda_dsp_controls.h create mode 100644 sound/soc/hda/hda_dsp_ssp_config.c create mode 100644 sound/soc/hda/hda_dsp_ssp_config.h create mode 100644 sound/soc/hda/hda_soc_dsp.c -- 1.7.9.5

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[alsa-devel] Simple Mixer help
by Bram Bos 30 Apr '15

30 Apr '15

I want to get access to the ALSA Simple Mixer interface, but rather than the 'default' sound device I want to target a specific USB sound device. I'm at a loss how to refer to it... I am using a modified version of the code below but I have no idea what to replace the "card" and "selem_name" strings with. The device name? The 'hwX.Y' string? And do I need to change the index#? Too many variables to solve this by trial and error. Any hints appreciated! void SetAlsaMasterVolume(long volume){ long min, max; snd_mixer_t *handle; snd_mixer_selem_id_t *sid; const char *card = "default"; const char *selem_name = "Master"; snd_mixer_open(&handle, 0); snd_mixer_attach(handle, card); snd_mixer_selem_register(handle, NULL, NULL); snd_mixer_load(handle); snd_mixer_selem_id_alloca(&sid); snd_mixer_selem_id_set_index(sid, 0); snd_mixer_selem_id_set_name(sid, selem_name); snd_mixer_elem_t* elem = snd_mixer_find_selem(handle, sid); snd_mixer_selem_get_playback_volume_range(elem, &min, &max); snd_mixer_selem_set_playback_volume_all(elem, volume * max / 100); snd_mixer_close(handle);}

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[alsa-devel] [for-next] NULL pointer dereference in codec probe
by Jani Nikula 30 Apr '15

30 Apr '15

Hi all, our graphics QA found an issue [1] with the for-next branch of [2] merged on top of v4.1-rc1. Backtrace in the bug. I didn't really have time to look into the details, but just letting you know. BR, Jani. [1] https://bugs.freedesktop.org/show_bug.cgi?id=90229 [2] git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git -- Jani Nikula, Intel Open Source Technology Center

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[alsa-devel] [PATCH lib 0/5] alsa-lib namehint fixes
by Takashi Iwai 30 Apr '15

30 Apr '15

Hi, here is a series of fixes for alsa-lib namehint stuff. Currently snd_device_name_hint() doesn't return proper lists except for PCM. Also it may do bogus free() or segfault for wrong operations. This patchset addresses such issues and also adds the missing hint descriptions to ctl, seq, hwdep, etc. Takashi === Takashi Iwai (5): namehint: Fix invalid list access in snd_device_name_hint() namehint: Fix the listing without device number namehint: Fix bad free with invalid iface name Allow hint for ctl, hwdep, timer and seq conf: Add hint descriptions to ctl, hwdep, seq and timer devices include/local.h | 2 ++ src/conf/alsa.conf | 15 ++++++++++++++- src/confmisc.c | 13 +++++++++++++ src/control/control_hw.c | 4 +--- src/control/control_shm.c | 7 ++----- src/control/namehint.c | 26 ++++++++++++-------------- src/hwdep/hwdep_hw.c | 4 +--- src/pcm/pcm.c | 12 ------------ src/pcm/pcm_local.h | 5 ++--- src/rawmidi/rawmidi.c | 18 ------------------ src/rawmidi/rawmidi_local.h | 2 +- src/seq/seq_hw.c | 4 +--- src/timer/timer_hw.c | 4 +--- src/timer/timer_query_hw.c | 4 +--- 14 files changed, 51 insertions(+), 69 deletions(-) -- 2.3.6

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[alsa-devel] [PATCH v4 0/2] Add support for select accessory detect mode to HPDETL or HPDETR
by Inha Song 30 Apr '15

30 Apr '15

This set of patches adds support for select accessory detect mode to HPDETL or HPDETR. Changes in v4: - Set the hpdet_channel default value to HPDETL in variable declaration. - Fix the indentation in binding documentation. Changes in v3: - Set the hpdet_channel default value to HPDETL If the value is unknown or invalid. Changes in v2: - Use the value in pdata instead of hpdet_channel in extcon_info. - Wrap arizona_extcon_of_get in IS_ENABLED(CONFIG_OF). - Change hpdet_channel type to unsigned from signed in pdata. - Move ARIZONA_ACCDET_MODE_* define to dt-binding header and directly set it to pdata. Inha Song (2): extcon: arizona: Add support for select accessory detect mode when headphone detection mfd: arizona: Update DT binding to support hpdet channel Documentation/devicetree/bindings/mfd/arizona.txt | 6 ++++ drivers/extcon/extcon-arizona.c | 38 ++++++++++++++++++----- include/dt-bindings/mfd/arizona.h | 8 +++++ include/linux/mfd/arizona/pdata.h | 3 ++ 4 files changed, 47 insertions(+), 8 deletions(-) create mode 100644 include/dt-bindings/mfd/arizona.h -- 2.0.0.390.gcb682f8

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[alsa-devel] [PATCH] ASoC: rt5645: fix implicit declaration error
by Bard Liao 30 Apr '15

30 Apr '15

kbuild robot reports a implicit declaration of function 'rt5645_irq_detection' error. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index a81a24f..8bad9fe 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2535,6 +2535,8 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) return rt5645->jack_type; } +static int rt5645_irq_detection(struct rt5645_priv *rt5645); + int rt5645_set_jack_detect(struct snd_soc_codec *codec, struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack, struct snd_soc_jack *btn_jack) -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ALSA: hda - remove check on CONFIG_SND_HDA_I915 in azx_probe_continue()
by mengdong.lin＠intel.com 30 Apr '15

30 Apr '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This is to avoid a build warning that label 'i915_power_fail' is defined but not used. There are already fake hda_i915 functions when CONFIG_SND_HDA_I915 is not defined, and error message for missing CONFIG_SND_HDA_I915 for the HD-A controller or codec that depends on i915 power well. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5aa5cfa..87df90d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1907,8 +1907,6 @@ static int azx_probe_continue(struct azx *chip) if (pci->device != 0x0f04 && pci->device != 0x2284) hda->need_i915_power = 1; - -#ifdef CONFIG_SND_HDA_I915 err = hda_i915_init(hda); if (err < 0) goto i915_power_fail; @@ -1919,7 +1917,6 @@ static int azx_probe_continue(struct azx *chip) "Cannot turn on display power on i915\n"); goto i915_power_fail; } -#endif } err = azx_first_init(chip); -- 1.9.1

2 1

[alsa-devel] [PATCH v3 0/5] ALSA: hda - remove unnecessary controller dependency on i915 gfx power
by mengdong.lin＠intel.com 30 Apr '15

30 Apr '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Mengdong Lin (5): ALSA: hda - implement a refcount for i915 power well switch ALSA: hda - allow a codec to control the link power ALSA: hda - implement link_power ops for i915 display power control ALSA: hda - divide controller and codec dependency on i915 gfx power well ALSA: hda - remove controller dependency on i915 power well for Baytrail/Braswell include/sound/hdaudio.h | 4 ++++ sound/hda/hdac_device.c | 15 ++++++++++++++ sound/pci/hda/hda_codec.c | 4 ++++ sound/pci/hda/hda_controller.c | 11 ++++++++++ sound/pci/hda/hda_controller.h | 2 ++ sound/pci/hda/hda_i915.c | 18 +++++++++++++---- sound/pci/hda/hda_intel.c | 46 ++++++++++++++++++++++++++++++++++-------- sound/pci/hda/hda_intel.h | 2 ++ sound/pci/hda/patch_hdmi.c | 9 +++++++++ 9 files changed, 99 insertions(+), 12 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v2 0/8] Add support for Wolfson WM8998 and WM1814 codecs
by Richard Fitzgerald 30 Apr '15

30 Apr '15

This set of patches adds support for the Wolfson Microelectronics WM8998 and WM1814 codecs. Richard Fitzgerald (8): mfd: arizona: Split INx_MODE into two fields ASoC: arizona: add defines for single-input gain control mfd: arizona: Add support for WM8998 and WM1814 gpio: arizona: add support for WM8998 and WM1814 regulator: arizona: add support for WM8998 and WM1814 extcon: arizona: Add support for WM8998 and WM1814 ASoC: wm8998: Initial WM8998 codec driver Documentation: Add WM8998/WM1814 device tree bindings Documentation/devicetree/bindings/mfd/arizona.txt | 13 +- drivers/extcon/extcon-arizona.c | 33 +- drivers/gpio/gpio-arizona.c | 2 + drivers/mfd/Kconfig | 6 + drivers/mfd/Makefile | 3 + drivers/mfd/arizona-core.c | 105 ++- drivers/mfd/arizona-i2c.c | 8 + drivers/mfd/arizona-irq.c | 9 + drivers/mfd/arizona.h | 5 + drivers/mfd/wm8998-tables.c | 1592 +++++++++++++++++++++ drivers/regulator/arizona-ldo1.c | 2 + include/linux/mfd/arizona/core.h | 3 + include/linux/mfd/arizona/pdata.h | 7 +- include/linux/mfd/arizona/registers.h | 247 +++- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 7 + sound/soc/codecs/arizona.h | 7 +- sound/soc/codecs/wm8998.c | 1394 ++++++++++++++++++ sound/soc/codecs/wm8998.h | 23 + 20 files changed, 3442 insertions(+), 32 deletions(-) create mode 100644 drivers/mfd/wm8998-tables.c create mode 100644 sound/soc/codecs/wm8998.c create mode 100644 sound/soc/codecs/wm8998.h -- 1.7.2.5

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[alsa-devel] [PATCH V2 0/4] tas571x amplifier driver
by Kevin Cernekee 29 Apr '15

29 Apr '15

V1->V2: - Incorporate changes from review feedback - Change GPIOs to active low - Create a tas571x_chip struct to capture the growing list of differences between 5711 and 5717/5719 - Add register defaults for each chip - Extend regcache_sync_region() to allow it to sync with hardware registers that don't contain their power-on-reset values Kevin Cernekee (4): regmap: cache: Add "was_reset" argument to regcache_sync_region() ASoC: tas571x: Add DT binding document ASoC: tas571x: New driver for TI TAS571x power amplifiers MAINTAINERS: Add entry for tas571x ASoC codec driver .../devicetree/bindings/sound/tas571x.txt | 41 ++ MAINTAINERS | 6 + drivers/base/regmap/internal.h | 5 +- drivers/base/regmap/regcache-lzo.c | 2 +- drivers/base/regmap/regcache-rbtree.c | 5 +- drivers/base/regmap/regcache.c | 75 +-- drivers/media/radio/radio-si476x.c | 18 +- drivers/mfd/wm8994-core.c | 5 +- include/linux/regmap.h | 4 +- include/sound/hda_regmap.h | 3 +- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas571x.c | 521 +++++++++++++++++++++ sound/soc/codecs/tas571x.h | 33 ++ sound/soc/codecs/wm8962.c | 3 +- 15 files changed, 680 insertions(+), 48 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tas571x.txt create mode 100644 sound/soc/codecs/tas571x.c create mode 100644 sound/soc/codecs/tas571x.h -- 2.2.0.rc0.207.ga3a616c

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[alsa-devel] [PATCH utils 1/2] aplay: Fix type for signal flag
by Takashi Iwai 29 Apr '15

29 Apr '15

A flag used in signal handlers has to be a special atomic type, volatile sig_atomic_t. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- aplay/aplay.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index e58e1bcbdd7e..dfa1a0a95867 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -107,7 +107,7 @@ static snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK; static int mmap_flag = 0; static int interleaved = 1; static int nonblock = 0; -static int in_aborting = 0; +static volatile sig_atomic_t in_aborting = 0; static u_char *audiobuf = NULL; static snd_pcm_uframes_t chunk_size = 0; static unsigned period_time = 0; -- 2.3.6

1 1

[alsa-devel] [TINYCOMPRESS][v2 PATCH 0/7] Convert build system to autotools
by Qais Yousef 29 Apr '15

29 Apr '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. Changes in v2: - Use -M option when generating the patches - Add Makefile.am to include directory that exports tinycompress.h - New patch (7) that fixes include dependency in tinycompress.h Qais Yousef (7): 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 tinycompress.h: explicitly include required headers .gitignore | 39 ++++ AUTHORS | 0 Android.mk | 6 +- COPYING | 51 +++++ ChangeLog | 0 INSTALL | 365 ++++++++++++++++++++++++++++++++++++ Makefile.am | 3 + NEWS | 0 autogen.sh | 6 + configure.ac | 32 ++++ include/Makefile.am | 6 + include/tinycompress/tinycompress.h | 3 + m4/.place_holder | 0 makefile.linux | 41 ---- src/Makefile.am | 1 + src/lib/Makefile.am | 5 + compress.c => src/lib/compress.c | 22 ++- src/utils/Makefile.am | 11 ++ cplay.c => src/utils/cplay.c | 0 crec.c => src/utils/crec.c | 0 utils.c | 62 ------ 21 files changed, 537 insertions(+), 116 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 create mode 100644 configure.ac create mode 100644 include/Makefile.am create mode 100644 m4/.place_holder delete mode 100644 makefile.linux create mode 100644 src/Makefile.am create mode 100644 src/lib/Makefile.am rename compress.c => src/lib/compress.c (98%) create mode 100644 src/utils/Makefile.am rename cplay.c => src/utils/cplay.c (100%) rename crec.c => src/utils/crec.c (100%) delete mode 100644 utils.c -- 2.1.0

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[alsa-devel] [PATCH v2 0/2] Add support for select accessory detect mode to HPDETL or HPDETR
by Inha Song 29 Apr '15

29 Apr '15

This set of patches adds support for select accessory detect mode to HPDETL or HPDETR. Changes in v2: - Use the value in pdata instead of hpdet_channel in extcon_info. - Wrap arizona_extcon_of_get in IS_ENABLED(CONFIG_OF). - Change hpdet_channel type to unsigned from signed in pdata. - Move ARIZONA_ACCDET_MODE_* define to dt-binding header and directly set it to pdata. Inha Song (2): extcon: arizona: Add support for select accessory detect mode when headphone detection mfd: arizona: Update DT binding to support hpdet channel Documentation/devicetree/bindings/mfd/arizona.txt | 6 +++++ drivers/extcon/extcon-arizona.c | 28 ++++++++++++++++------- include/dt-bindings/mfd/arizona.h | 8 +++++++ include/linux/mfd/arizona/pdata.h | 3 +++ 4 files changed, 37 insertions(+), 8 deletions(-) create mode 100644 include/dt-bindings/mfd/arizona.h -- 2.0.0.390.gcb682f8

3 6

[alsa-devel] [PATCH 1/2] ASoC: rt5677: fixed wrong DMIC ref clock
by Bard Liao 29 Apr '15

29 Apr '15

DMIC clock source is not from codec system clock directly. it is generated from the division of system clock. And it should be 256 * sample rate of AIF1. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5677.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index fbe16ba..dca64ae 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -917,7 +917,7 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); - int idx = rl6231_calc_dmic_clk(rt5677->sysclk); + int idx = rl6231_calc_dmic_clk(rt5677->lrck[RT5677_AIF1] << 8); if (idx < 0) dev_err(codec->dev, "Failed to set DMIC clock\n"); -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH v3] ASoC: rt5645: Adds push button support for rt5650
by Bard Liao 29 Apr '15

29 Apr '15

rt5650 support headset button detection. Currently, the button detection is only implemented for rt5650 codec. The button detection configuration register's default value is different from rt5645. And we didn't touch the register in the driver, so we will get the wrong value when we dump the registers. We will fix it in another patch. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * redo the using of rt5645_set_jack_detect base on the latest code base. --- sound/soc/codecs/rt5645.c | 266 +++++++++++++++++++++++++++----- sound/soc/codecs/rt5645.h | 8 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- 3 files changed, 237 insertions(+), 39 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 2cab2eb..a81a24f 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2371,6 +2371,8 @@ static int rt5645_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, static int rt5645_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_PREPARE: if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { @@ -2401,8 +2403,9 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: snd_soc_write(codec, RT5645_DEPOP_M2, 0x1100); - snd_soc_update_bits(codec, RT5645_GEN_CTRL1, - RT5645_DIG_GATE_CTRL, 0); + if (!rt5645->en_button_func) + snd_soc_update_bits(codec, RT5645_GEN_CTRL1, + RT5645_DIG_GATE_CTRL, 0); snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_VREF1 | RT5645_PWR_MB | RT5645_PWR_BG | RT5645_PWR_VREF2 | @@ -2417,28 +2420,71 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int rt5645_jack_detect(struct snd_soc_codec *codec) +static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, + bool enable) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - int gpio_state, jack_type = 0; - unsigned int val; - if (!gpio_is_valid(rt5645->pdata.hp_det_gpio)) { - dev_err(codec->dev, "invalid gpio\n"); - return -EINVAL; + if (enable) { + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "ADC L power"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "ADC R power"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "LDO2"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync_unlocked(&codec->dapm); + snd_soc_update_bits(codec, + RT5645_INT_IRQ_ST, 0x8, 0x8); + snd_soc_update_bits(codec, + RT5650_4BTN_IL_CMD2, 0x8000, 0x8000); + snd_soc_read(codec, RT5650_4BTN_IL_CMD1); + pr_debug("%s read %x = %x\n", __func__, RT5650_4BTN_IL_CMD1, + snd_soc_read(codec, RT5650_4BTN_IL_CMD1)); + } else { + snd_soc_update_bits(codec, RT5650_4BTN_IL_CMD2, 0x8000, 0x0); + snd_soc_update_bits(codec, RT5645_INT_IRQ_ST, 0x8, 0x0); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "ADC L power"); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "ADC R power"); + if (rt5645->pdata.jd_mode == 0) + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "LDO2"); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync_unlocked(&codec->dapm); } - gpio_state = gpio_get_value(rt5645->pdata.hp_det_gpio); +} - dev_dbg(codec->dev, "gpio = %d(%d)\n", rt5645->pdata.hp_det_gpio, - gpio_state); +static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + unsigned int val; - if ((rt5645->pdata.gpio_hp_det_active_high && gpio_state) || - (!rt5645->pdata.gpio_hp_det_active_high && !gpio_state)) { - snd_soc_dapm_force_enable_pin(&codec->dapm, "micbias1"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "micbias2"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "Mic Det Power"); - snd_soc_dapm_sync(&codec->dapm); + if (jack_insert) { + if (codec->component.card->instantiated) { + snd_soc_dapm_force_enable_pin(&codec->dapm, + "micbias1"); + snd_soc_dapm_force_enable_pin(&codec->dapm, + "micbias2"); + snd_soc_dapm_force_enable_pin(&codec->dapm, + "LDO2"); + snd_soc_dapm_force_enable_pin(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } else { + /* Power up necessary bits for JD if dapm is + not ready yet */ + snd_soc_update_bits(codec, RT5645_PWR_ANLG2, + RT5645_PWR_MB1 | RT5645_PWR_MB2, + RT5645_PWR_MB1 | RT5645_PWR_MB2); + snd_soc_update_bits(codec, RT5645_PWR_MIXER, + RT5645_PWR_LDO2, RT5645_PWR_LDO2); + snd_soc_update_bits(codec, RT5645_PWR_VOL, + RT5645_PWR_MIC_DET, RT5645_PWR_MIC_DET); + } snd_soc_write(codec, RT5645_IN1_CTRL1, 0x0006); snd_soc_write(codec, RT5645_JD_CTRL3, 0x00b0); @@ -2452,32 +2498,62 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec) val = snd_soc_read(codec, RT5645_IN1_CTRL3) & 0x7; dev_dbg(codec->dev, "val = %d\n", val); - if (val == 1 || val == 2) - jack_type = SND_JACK_HEADSET; - else - jack_type = SND_JACK_HEADPHONE; + if (codec->component.card->instantiated) { + snd_soc_dapm_disable_pin(&codec->dapm, "micbias1"); + snd_soc_dapm_disable_pin(&codec->dapm, "micbias2"); + if (rt5645->pdata.jd_mode == 0) + snd_soc_dapm_disable_pin(&codec->dapm, "LDO2"); + snd_soc_dapm_disable_pin(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } else { + snd_soc_update_bits(codec, RT5645_PWR_ANLG2, + RT5645_PWR_MB1 | RT5645_PWR_MB2, 0); + if (rt5645->pdata.jd_mode == 0) + snd_soc_update_bits(codec, RT5645_PWR_MIXER, + RT5645_PWR_LDO2, 0); + snd_soc_update_bits(codec, RT5645_PWR_VOL, + RT5645_PWR_MIC_DET, 0); + } - snd_soc_dapm_disable_pin(&codec->dapm, "micbias1"); - snd_soc_dapm_disable_pin(&codec->dapm, "micbias2"); - if (rt5645->pdata.jd_mode == 0) - snd_soc_dapm_disable_pin(&codec->dapm, "LDO2"); - snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); - snd_soc_dapm_sync(&codec->dapm); + if (val == 1 || val == 2) { + rt5645->jack_type = SND_JACK_HEADSET; + if (rt5645->en_button_func) { + msleep(100); + rt5645_enable_push_button_irq(codec, true); + } + } else { + rt5645->jack_type = SND_JACK_HEADPHONE; + } + + } else { /* jack out */ + rt5645->jack_type = 0; + if (rt5645->en_button_func) + rt5645_enable_push_button_irq(codec, false); } - snd_soc_jack_report(rt5645->hp_jack, jack_type, SND_JACK_HEADPHONE); - snd_soc_jack_report(rt5645->mic_jack, jack_type, SND_JACK_MICROPHONE); - return 0; + return rt5645->jack_type; } int rt5645_set_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack) + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack, + struct snd_soc_jack *btn_jack) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); rt5645->hp_jack = hp_jack; rt5645->mic_jack = mic_jack; - rt5645_jack_detect(codec); + rt5645->btn_jack = btn_jack; + if (rt5645->btn_jack && rt5645->codec_type == CODEC_TYPE_RT5650) { + rt5645->en_button_func = true; + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ); + regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1, + RT5645_HP_CB_MASK, RT5645_HP_CB_PU); + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1, + RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); + } + rt5645_irq_detection(rt5645); return 0; } @@ -2488,7 +2564,7 @@ static void rt5645_jack_detect_work(struct work_struct *work) struct rt5645_priv *rt5645 = container_of(work, struct rt5645_priv, jack_detect_work.work); - rt5645_jack_detect(rt5645->codec); + rt5645_irq_detection(rt5645); } static irqreturn_t rt5645_irq(int irq, void *data) @@ -2501,6 +2577,125 @@ static irqreturn_t rt5645_irq(int irq, void *data) return IRQ_HANDLED; } +static int rt5645_button_detect(struct snd_soc_codec *codec) +{ + int btn_type, val; + + val = snd_soc_read(codec, RT5650_4BTN_IL_CMD1); + pr_debug("val=0x%x\n", val); + btn_type = val & 0xfff0; + snd_soc_write(codec, RT5650_4BTN_IL_CMD1, val); + + return btn_type; +} + +static int rt5645_irq_detection(struct rt5645_priv *rt5645) +{ + int val, btn_type, gpio_state = 0, report = 0; + + switch (rt5645->pdata.jd_mode) { + case 0: /* Not using rt5645 JD */ + if (gpio_is_valid(rt5645->pdata.hp_det_gpio)) { + gpio_state = gpio_get_value(rt5645->pdata.hp_det_gpio); + dev_dbg(rt5645->codec->dev, "gpio = %d(%d)\n", + rt5645->pdata.hp_det_gpio, gpio_state); + } + if ((rt5645->pdata.gpio_hp_det_active_high && gpio_state) || + (!rt5645->pdata.gpio_hp_det_active_high && + !gpio_state)) { + report = rt5645_jack_detect(rt5645->codec, 1); + } else { + report = rt5645_jack_detect(rt5645->codec, 0); + } + snd_soc_jack_report(rt5645->hp_jack, + report, SND_JACK_HEADPHONE); + snd_soc_jack_report(rt5645->mic_jack, + report, SND_JACK_MICROPHONE); + return report; + case 1: /* 2 port */ + val = snd_soc_read(rt5645->codec, RT5645_A_JD_CTRL1) & 0x0070; + break; + default: /* 1 port */ + val = snd_soc_read(rt5645->codec, RT5645_A_JD_CTRL1) & 0x0020; + break; + + } + + switch (val) { + /* jack in */ + case 0x30: /* 2 port */ + case 0x0: /* 1 port or 2 port */ + if (rt5645->jack_type == 0) { + report = rt5645_jack_detect(rt5645->codec, 1); + /* for push button and jack out */ + break; + } + btn_type = 0; + if (snd_soc_read(rt5645->codec, RT5645_INT_IRQ_ST) & 0x4) { + /* button pressed */ + report = SND_JACK_HEADSET; + btn_type = rt5645_button_detect(rt5645->codec); + /* rt5650 can report three kinds of button behavior, + one click, double click and hold. However, + currently we will report button pressed/released + event. So all the three button behaviors are + treated as button pressed. */ + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + report |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + report |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + report |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + report |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + dev_err(rt5645->codec->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + } + if (btn_type == 0)/* button release */ + report = rt5645->jack_type; + + break; + /* jack out */ + case 0x70: /* 2 port */ + case 0x10: /* 2 port */ + case 0x20: /* 1 port */ + report = 0; + snd_soc_update_bits(rt5645->codec, + RT5645_INT_IRQ_ST, 0x1, 0x0); + rt5645_jack_detect(rt5645->codec, 0); + break; + default: + break; + } + + snd_soc_jack_report(rt5645->hp_jack, report, SND_JACK_HEADPHONE); + snd_soc_jack_report(rt5645->mic_jack, report, SND_JACK_MICROPHONE); + if (rt5645->en_button_func) + snd_soc_jack_report(rt5645->btn_jack, + report, SND_JACK_MICROPHONE); + + return report; +} + static int rt5645_probe(struct snd_soc_codec *codec) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); @@ -2840,8 +3035,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (rt5645->pdata.en_jd_func) { regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3, - RT5645_IRQ_CLK_GATE_CTRL | RT5645_MICINDET_MANU, - RT5645_IRQ_CLK_GATE_CTRL | RT5645_MICINDET_MANU); + RT5645_IRQ_CLK_GATE_CTRL, RT5645_IRQ_CLK_GATE_CTRL); regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, RT5645_CBJ_BST1_EN, RT5645_CBJ_BST1_EN); regmap_update_bits(rt5645->regmap, RT5645_JD_CTRL3, diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index db78e94..4473636 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2184,6 +2184,7 @@ struct rt5645_priv { struct i2c_client *i2c; struct snd_soc_jack *hp_jack; struct snd_soc_jack *mic_jack; + struct snd_soc_jack *btn_jack; struct delayed_work jack_detect_work; int codec_type; @@ -2196,9 +2197,12 @@ struct rt5645_priv { int pll_src; int pll_in; int pll_out; + + int jack_type; + bool en_button_func; }; int rt5645_set_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack); - + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack, + struct snd_soc_jack *btn_jack); #endif /* __RT5645_H__ */ diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 20a28b2..26e01f3 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -185,7 +185,7 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) return ret; } - rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack); + rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack, NULL); return ret; } -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [RFC 0/9] Add SKL ipc handling code
by Vinod Koul 29 Apr '15

29 Apr '15

This series adds support for managing DSP and posting IPC and processing IPC replies. Subhransu S. Prusty (9): ASoC: hda: Add helper to read/write to dsp mmio space ASoC: hda: Add IPC library for SKL platform ASoC: hda: Add config option for SKL ipc library ASoC: hda: Add DSP init and boot up functionality ASoC: hda: Add dsp loader ops ASoC: hda: Add Code Loader DMA support ASoC: hda: Add DSP library functions for SKL platform ASoC: hda: Add for CL DMA interrupt handling ASoC: hda: Export API to change DSP power state include/sound/soc-hda-sst-cldma.h | 234 ++++++++ include/sound/soc-hda-sst-dsp.h | 266 +++++++++ include/sound/soc-hda-sst-ipc.h | 134 +++++ sound/soc/hda/Kconfig | 7 + sound/soc/hda/Makefile | 3 + sound/soc/hda/intel/Makefile | 6 + sound/soc/hda/intel/soc-hda-sst-cldma.c | 297 +++++++++ sound/soc/hda/intel/soc-hda-sst-dsp.c | 510 ++++++++++++++++ sound/soc/hda/intel/soc-hda-sst-ipc.c | 963 ++++++++++++++++++++++++++++++ sound/soc/hda/intel/soc-hda-sst-skl-fw.h | 163 +++++ sound/soc/hda/intel/soc-hda-sst-skl.c | 292 +++++++++ 11 files changed, 2875 insertions(+) create mode 100644 include/sound/soc-hda-sst-cldma.h create mode 100644 include/sound/soc-hda-sst-dsp.h create mode 100644 include/sound/soc-hda-sst-ipc.h create mode 100644 sound/soc/hda/intel/Makefile create mode 100644 sound/soc/hda/intel/soc-hda-sst-cldma.c create mode 100644 sound/soc/hda/intel/soc-hda-sst-dsp.c create mode 100644 sound/soc/hda/intel/soc-hda-sst-ipc.c create mode 100644 sound/soc/hda/intel/soc-hda-sst-skl-fw.h create mode 100644 sound/soc/hda/intel/soc-hda-sst-skl.c -- 1.7.9.5

3 33

[alsa-devel] [PATCH 1/2] ASoC: Intel: Support rt5650 codec for Cherrytrail & Braswell
by yang.a.fang＠intel.com 29 Apr '15

29 Apr '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> rt5650 and rt5645 are similar codec so reuse the cht_bsw_rt5645 driver Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/intel/Kconfig | 7 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 108 +++++++++++++++++++++++++------ 2 files changed, 94 insertions(+), 21 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index ee03dbd..63d7659 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -112,12 +112,13 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH If unsure select "N". config SND_SOC_INTEL_CHT_BSW_RT5645_MACH - tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645 codec" - depends on X86_INTEL_LPSS + tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645/5650 codec" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + select SND_SOC_INTEL_BAYTRAIL select SND_SOC_RT5645 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI help This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell - platforms with RT5645 audio codec. + platforms with RT5645/5650 audio codec. If unsure select "N". diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 20a28b2..8115444 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -21,6 +21,7 @@ */ #include <linux/module.h> +#include <linux/acpi.h> #include <linux/platform_device.h> #include <linux/slab.h> #include <sound/pcm.h> @@ -33,9 +34,16 @@ #define CHT_PLAT_CLK_3_HZ 19200000 #define CHT_CODEC_DAI "rt5645-aif1" +struct cht_acpi_card { + char *codec_id; + int codec_type; + struct snd_soc_card *soc_card; +}; + struct cht_mc_private { struct snd_soc_jack hp_jack; struct snd_soc_jack mic_jack; + struct cht_acpi_card *acpi_card; }; static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) @@ -94,7 +102,7 @@ static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { platform_clock_control, SND_SOC_DAPM_POST_PMD), }; -static const struct snd_soc_dapm_route cht_audio_map[] = { +static const struct snd_soc_dapm_route cht_5645_audio_map[] = { {"IN1P", NULL, "Headset Mic"}, {"IN1N", NULL, "Headset Mic"}, {"DMIC L1", NULL, "Int Mic"}, @@ -115,6 +123,27 @@ static const struct snd_soc_dapm_route cht_audio_map[] = { {"Ext Spk", NULL, "Platform Clock"}, }; +static const struct snd_soc_dapm_route cht_5650_audio_map[] = { + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + {"DMIC L2", NULL, "Int Mic"}, + {"DMIC R2", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx" }, + {"codec_in1", NULL, "ssp2 Rx" }, + {"ssp2 Rx", NULL, "AIF1 Capture"}, + {"Headphone", NULL, "Platform Clock"}, + {"Headset Mic", NULL, "Platform Clock"}, + {"Int Mic", NULL, "Platform Clock"}, + {"Ext Spk", NULL, "Platform Clock"}, +}; + static const struct snd_kcontrol_new cht_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -239,7 +268,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", .platform_name = "sst-mfld-platform", - .ignore_suspend = 1, + .nonatomic = true, .dynamic = 1, .dpcm_playback = 1, .dpcm_capture = 1, @@ -267,7 +296,7 @@ static struct snd_soc_dai_link cht_dailink[] = { | SND_SOC_DAIFMT_CBS_CFS, .init = cht_codec_init, .be_hw_params_fixup = cht_codec_fixup, - .ignore_suspend = 1, + .nonatomic = true, .dpcm_playback = 1, .dpcm_capture = 1, .ops = &cht_be_ssp2_ops, @@ -275,43 +304,86 @@ static struct snd_soc_dai_link cht_dailink[] = { }; /* SoC card */ -static struct snd_soc_card snd_soc_card_cht = { - .name = "chtrt5645", - .dai_link = cht_dailink, - .num_links = ARRAY_SIZE(cht_dailink), - .dapm_widgets = cht_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), - .dapm_routes = cht_audio_map, - .num_dapm_routes = ARRAY_SIZE(cht_audio_map), - .controls = cht_mc_controls, - .num_controls = ARRAY_SIZE(cht_mc_controls), +static struct snd_soc_card snd_soc_card_cht[] = { + { + .name = "chtrt5645", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_5645_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_5645_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), + }, + { + .name = "chtrt5650", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_5650_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_5650_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), + } }; +static struct cht_acpi_card snd_soc_cards[] = { + {"10EC5645", CODEC_TYPE_RT5645, &snd_soc_card_cht[0]}, + {"10EC5650", CODEC_TYPE_RT5650, &snd_soc_card_cht[1]}, +}; + +static acpi_status snd_acpi_codec_match(acpi_handle handle, u32 level, + void *context, void **ret) +{ + *(bool *)context = true; + return AE_OK; +} + static int snd_cht_mc_probe(struct platform_device *pdev) { int ret_val = 0; + int i; struct cht_mc_private *drv; + struct snd_soc_card *card = snd_soc_cards[0].soc_card; + bool found = false; + char codec_name[32]; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); if (!drv) return -ENOMEM; - snd_soc_card_cht.dev = &pdev->dev; - snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); - ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht); + for (i = 0; i < ARRAY_SIZE(snd_soc_cards); i++) { + if (ACPI_SUCCESS(acpi_get_devices( + snd_soc_cards[i].codec_id, + snd_acpi_codec_match, + &found, NULL)) && found) { + dev_dbg(&pdev->dev, + "found codec %s\n", snd_soc_cards[i].codec_id); + card = snd_soc_cards[i].soc_card; + drv->acpi_card = &snd_soc_cards[i]; + break; + } + } + card->dev = &pdev->dev; + sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id); + /* set correct codec name */ + strcpy((char *)card->dai_link[2].codec_name, codec_name); + snd_soc_card_set_drvdata(card, drv); + ret_val = devm_snd_soc_register_card(&pdev->dev, card); if (ret_val) { dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret_val); return ret_val; } - platform_set_drvdata(pdev, &snd_soc_card_cht); + platform_set_drvdata(pdev, card); return ret_val; } static struct platform_driver snd_cht_mc_driver = { .driver = { .name = "cht-bsw-rt5645", - .pm = &snd_soc_pm_ops, }, .probe = snd_cht_mc_probe, }; -- 1.7.9.5

5 10

[alsa-devel] [PATCH 05/12] trivial: ASoC: adau1977: fix typo s/Substraction/Subtraction/
by Antonio Ospite 29 Apr '15

29 Apr '15

Signed-off-by: Antonio Ospite <ao2(a)ao2.it> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/adau1977.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c index 7ad8e15..dc8ad08 100644 --- a/sound/soc/codecs/adau1977.c +++ b/sound/soc/codecs/adau1977.c @@ -202,7 +202,7 @@ static const struct snd_soc_dapm_route adau1977_dapm_routes[] = { ADAU1977_REG_DC_HPF_CAL, (x) - 1, 1, 0) #define ADAU1977_DC_SUB_SWITCH(x) \ - SOC_SINGLE("ADC" #x " DC Substraction Capture Switch", \ + SOC_SINGLE("ADC" #x " DC Subtraction Capture Switch", \ ADAU1977_REG_DC_HPF_CAL, (x) + 3, 1, 0) static const struct snd_kcontrol_new adau1977_snd_controls[] = { -- 2.1.4

3 4

[alsa-devel] Emu10k2 - support for 32 bit DMA mode
by Peter Zubaj 29 Apr '15

29 Apr '15

Hi, Looks like audigy emu10k2 (probably emu10k1 - sb live too) support two modes for DMA. Second mode is useful for 64 bit os with more then 2 GB of ram (fixes problems with big soundfont loading) 1) 32MB from 2 GB address space using 8192 pages (used now as default) 2) 16MB from 4 GB address space using 4096 pages Mode is set using HCFG_EXPANDED_MEM flag in HCFG register. Also format of emu10k2 page table is then different. If someone wants to play with it, attached is patch to enable this mode for audigy. It is tested only on audigy rx (8 GB ram, 64 bit os). Regards, Peter

2 3

Re: [alsa-devel] M-Audio Midisport 2x2 Anniversary edition, latency problem
by Clemens Ladisch 28 Apr '15

28 Apr '15

Ilya Dikariev wrote: > Вторник, 28 апреля 2015, 15:30 +02:00 от Clemens Ladisch <clemens(a)ladisch.de>: >> Ilya Dikariev wrote: >>> I bought a MIDI-interface von M-Audio (see subject), but there are >>> strange things with it on my desktop (Arch, latest kernel). The >>> latency is about 5 seconds. >> >> How are you measuring the latency? > > I measured it only with my feeling of time. What are the specific events between which there is an interval of 5 seconds? Regards, Clemens

1 0

[alsa-devel] [PATCH] ASoC: rt5645: fixed section mismatch
by Sudip Mukherjee 28 Apr '15

28 Apr '15

while building as a module we are getting warning about section mismatch. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/codecs/rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 14570b3..0ed79ca 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2684,7 +2684,7 @@ static int strago_quirk_cb(const struct dmi_system_id *id) return 1; } -static struct dmi_system_id dmi_platform_intel_braswell[] __initdata = { +static struct dmi_system_id dmi_platform_intel_braswell[] = { { .ident = "Intel Strago", .callback = strago_quirk_cb, -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ALSA: emux: Fix mutex deadlock in OSS emulation
by Takashi Iwai 28 Apr '15

28 Apr '15

The OSS emulation in synth-emux helper has a potential AB/BA deadlock at the simultaneous closing and opening: close -> snd_seq_release() -> sne_seq_free_client() -> snd_seq_delete_all_ports(): takes client->ports_mutex -> port_delete() -> snd_emux_unuse(): takes emux->register_mutex open -> snd_seq_oss_open() -> snd_emux_open_seq_oss(): takes emux->register_mutex -> snd_seq_event_port_attach() -> snd_seq_create_port(): takes client->ports_mutex This patch addresses the deadlock by reducing the rance taking emux->register_mutex in snd_emux_open_seq_oss(). The lock is needed for the refcount handling, so move it locally. The calls in emux_seq.c are already with the mutex, thus they are replaced with the version without mutex lock/unlock. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/synth/emux/emux_oss.c | 11 +---------- sound/synth/emux/emux_seq.c | 27 +++++++++++++++++++++------ 2 files changed, 22 insertions(+), 16 deletions(-) diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c index ab37add269ae..82e350e9501c 100644 --- a/sound/synth/emux/emux_oss.c +++ b/sound/synth/emux/emux_oss.c @@ -118,12 +118,8 @@ snd_emux_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure) if (snd_BUG_ON(!arg || !emu)) return -ENXIO; - mutex_lock(&emu->register_mutex); - - if (!snd_emux_inc_count(emu)) { - mutex_unlock(&emu->register_mutex); + if (!snd_emux_inc_count(emu)) return -EFAULT; - } memset(&callback, 0, sizeof(callback)); callback.owner = THIS_MODULE; @@ -135,7 +131,6 @@ snd_emux_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure) if (p == NULL) { snd_printk(KERN_ERR "can't create port\n"); snd_emux_dec_count(emu); - mutex_unlock(&emu->register_mutex); return -ENOMEM; } @@ -148,8 +143,6 @@ snd_emux_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure) reset_port_mode(p, arg->seq_mode); snd_emux_reset_port(p); - - mutex_unlock(&emu->register_mutex); return 0; } @@ -195,13 +188,11 @@ snd_emux_close_seq_oss(struct snd_seq_oss_arg *arg) if (snd_BUG_ON(!emu)) return -ENXIO; - mutex_lock(&emu->register_mutex); snd_emux_sounds_off_all(p); snd_soundfont_close_check(emu->sflist, SF_CLIENT_NO(p->chset.port)); snd_seq_event_port_detach(p->chset.client, p->chset.port); snd_emux_dec_count(emu); - mutex_unlock(&emu->register_mutex); return 0; } diff --git a/sound/synth/emux/emux_seq.c b/sound/synth/emux/emux_seq.c index 188fda0effb0..a0209204ae48 100644 --- a/sound/synth/emux/emux_seq.c +++ b/sound/synth/emux/emux_seq.c @@ -267,8 +267,8 @@ snd_emux_event_input(struct snd_seq_event *ev, int direct, void *private_data, /* * increment usage count */ -int -snd_emux_inc_count(struct snd_emux *emu) +static int +__snd_emux_inc_count(struct snd_emux *emu) { emu->used++; if (!try_module_get(emu->ops.owner)) @@ -282,12 +282,21 @@ snd_emux_inc_count(struct snd_emux *emu) return 1; } +int snd_emux_inc_count(struct snd_emux *emu) +{ + int ret; + + mutex_lock(&emu->register_mutex); + ret = __snd_emux_inc_count(emu); + mutex_unlock(&emu->register_mutex); + return ret; +} /* * decrease usage count */ -void -snd_emux_dec_count(struct snd_emux *emu) +static void +__snd_emux_dec_count(struct snd_emux *emu) { module_put(emu->card->module); emu->used--; @@ -296,6 +305,12 @@ snd_emux_dec_count(struct snd_emux *emu) module_put(emu->ops.owner); } +void snd_emux_dec_count(struct snd_emux *emu) +{ + mutex_lock(&emu->register_mutex); + __snd_emux_dec_count(emu); + mutex_unlock(&emu->register_mutex); +} /* * Routine that is called upon a first use of a particular port @@ -315,7 +330,7 @@ snd_emux_use(void *private_data, struct snd_seq_port_subscribe *info) mutex_lock(&emu->register_mutex); snd_emux_init_port(p); - snd_emux_inc_count(emu); + __snd_emux_inc_count(emu); mutex_unlock(&emu->register_mutex); return 0; } @@ -338,7 +353,7 @@ snd_emux_unuse(void *private_data, struct snd_seq_port_subscribe *info) mutex_lock(&emu->register_mutex); snd_emux_sounds_off_all(p); - snd_emux_dec_count(emu); + __snd_emux_dec_count(emu); mutex_unlock(&emu->register_mutex); return 0; } -- 2.3.5

1 0

[alsa-devel] [PATCH V2] Update email-id of author
by Rajeev Kumar 28 Apr '15

28 Apr '15

rajeev-dlh.kumar(a)st.com email-id doesn't exist anymore as I have left the company. Replace ST's id with Rajeev Kumar <rajeevkumar.linux(a)gmail.com> Signed-off-by: Rajeev Kumar <rajeevkumar.linux(a)gmail.com> --- include/sound/designware_i2s.h | 2 +- include/sound/spear_dma.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h index 26f406e..3a8fca9 100644 --- a/include/sound/designware_i2s.h +++ b/include/sound/designware_i2s.h @@ -1,5 +1,5 @@ /* - * Copyright (ST) 2012 Rajeev Kumar (rajeev-dlh.kumar(a)st.com) + * Copyright (ST) 2012 Rajeev Kumar (rajeevkumar.linux(a)gmail.com) * * 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 diff --git a/include/sound/spear_dma.h b/include/sound/spear_dma.h index 65aca51..e290de4 100644 --- a/include/sound/spear_dma.h +++ b/include/sound/spear_dma.h @@ -1,7 +1,7 @@ /* * linux/spear_dma.h * -* Copyright (ST) 2012 Rajeev Kumar (rajeev-dlh.kumar(a)st.com) +* Copyright (ST) 2012 Rajeev Kumar (rajeevkumar.linux(a)gmail.com) * * 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 -- 1.7.9.5

2 1

[alsa-devel] [RFC PATCH 1/2] ALSA: hda - divide controller and codec dependency on i915 gfx power well
by mengdong.lin＠intel.com 28 Apr '15

28 Apr '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch can improve power saving for Intel platforms on which only the display audio codec is in the shared i915 power well: - divide the controller and codec dependency on i915 power well by adding flag "need_i915_power" for the chip (controller) and codec respectively. - If the controller does not need the power well, the driver will release the power after probe. And if only the codec needs the power, its runtime PM ops will request/release the power. Background: - For Haswell/Broadwell, which has a separate HD-A controller for display audio, both the controller and the display codec are in the i915 power well. - For Baytrail/Braswell, the display and analog audio share the same HDA controller and link, and only the display codec is in the i915 power well. - For Skylake, the display and analog audio share the same HDA controller but use separate links. Only the display codec is in the i915 power well. And in legacy mode we take the two links as one. So it can follow Baytrail/Braswell. Open questions: - Move the hda_i915 to sound/hda at first? So both legacy and new drivers can use it as Takashi suggested? The PCI ID check in haswell_set_bclk will no longer be needed, since it's only called when chip's "need_i915_power" flag is set and so only called for Haswell & Broadwell. - move flag "need_i915_power" from struct hda_codec to hdac_device, for the same reason as above? - If the audio component moves from struct hda_intel to hdac_bus, how can the controller access the component? Need to add "bus" to struct azx? - implement codec resume/suspend ops to get/put i915 power if only the codec needs the power? Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 2a8aa9d..baa01a0 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -169,6 +169,8 @@ struct hdac_bus_ops { /* get a response from the last command */ int (*get_response)(struct hdac_bus *bus, unsigned int addr, unsigned int *res); + /* request/release display power */ + int (*display_power)(struct hdac_bus *bus, bool enable); }; #define HDA_UNSOL_QUEUE_SIZE 64 @@ -222,6 +224,11 @@ static inline void snd_hdac_codec_link_down(struct hdac_device *codec) clear_bit(codec->addr, &codec->bus->codec_powered); } +static inline int snd_hdac_display_power(struct hdac_device *codec, bool enable) +{ + return codec->bus->ops->display_power(codec->bus, enable); +} + /* * generic array helpers */ diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c old mode 100755 new mode 100644 index 18a021a..034488e --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -524,9 +524,21 @@ static int _hda_bus_get_response(struct hdac_bus *_bus, unsigned int addr, return bus->rirb_error ? -EIO : 0; } +static int _hda_bus_display_power(struct hdac_bus *_bus, bool enable) +{ + struct hda_bus *bus = container_of(_bus, struct hda_bus, core); + + if (bus->ops.display_power) + return bus->ops.display_power(bus, enable); + else + return -EINVAL; + +} + static const struct hdac_bus_ops bus_ops = { .command = _hda_bus_command, .get_response = _hda_bus_get_response, + .display_power = _hda_bus_display_power, }; /** @@ -949,6 +961,10 @@ void snd_hda_codec_register(struct hda_codec *codec) return; if (device_is_registered(hda_codec_dev(codec))) { snd_hda_register_beep_device(codec); + + if (codec->need_i915_power) + snd_hdac_display_power(&codec->core, true); + pm_runtime_enable(hda_codec_dev(codec)); /* it was powered up in snd_hda_codec_new(), now all done */ snd_hda_power_down(codec); @@ -3196,6 +3212,9 @@ static int hda_codec_runtime_suspend(struct device *dev) if (codec_has_clkstop(codec) && codec_has_epss(codec) && (state & AC_PWRST_CLK_STOP_OK)) snd_hdac_codec_link_down(&codec->core); + + if (codec->need_i915_power) + snd_hdac_display_power(&codec->core, false); return 0; } @@ -3205,6 +3224,9 @@ static int hda_codec_runtime_resume(struct device *dev) printk("amanda: hda_codec_runtime_resume, codec addr %d \n", codec->addr); + if (codec->need_i915_power) + snd_hdac_display_power(&codec->core, true); + snd_hdac_codec_link_up(&codec->core); hda_call_codec_resume(codec); pm_runtime_mark_last_busy(dev); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 9075ac2..32d7fa7 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -64,6 +64,7 @@ struct hda_bus_ops { void (*load_dsp_cleanup)(struct hda_bus *bus, struct snd_dma_buffer *dmab); #endif + int (*display_power)(struct hda_bus *bus, bool enable); }; /* @@ -281,6 +282,7 @@ struct hda_codec { 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_save_node:1; /* advanced PM for each widget */ + unsigned int need_i915_power:1; /* only codec needs i915 power */ #ifdef CONFIG_PM unsigned long power_on_acct; unsigned long power_off_acct; diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 26ce990..b7eaf2d 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1354,6 +1354,17 @@ static unsigned int azx_get_response(struct hda_bus *bus, return azx_rirb_get_response(bus, addr); } + +static unsigned int azx_display_power(struct hda_bus *bus, bool enable) +{ + struct azx *chip = bus->private_data; + + if (chip->ops->display_power) + return chip->ops->display_power(chip, enable); + else + return -EINVAL; +} + #ifdef CONFIG_SND_HDA_DSP_LOADER /* * DSP loading code (e.g. for CA0132) @@ -1819,6 +1830,7 @@ static struct hda_bus_ops bus_ops = { .load_dsp_trigger = azx_load_dsp_trigger, .load_dsp_cleanup = azx_load_dsp_cleanup, #endif + .display_power = azx_display_power, }; /* HD-audio bus initialization */ diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index be1b7de..2d6c6c2 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -277,6 +277,7 @@ struct hda_controller_ops { struct vm_area_struct *area); /* Check if current position is acceptable */ int (*position_check)(struct azx *chip, struct azx_dev *azx_dev); + int (*display_power)(struct azx *chip, bool enable); }; struct azx_pcm { @@ -358,6 +359,7 @@ struct azx { unsigned int align_buffer_size:1; unsigned int region_requested:1; unsigned int disabled:1; /* disabled by VGA-switcher */ + unsigned int need_i915_power:1; /* both controller & display codec needs i915 power */ /* for debugging */ unsigned int last_cmd[AZX_MAX_CODECS]; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 34040d2..f6c3e37 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -543,6 +543,13 @@ static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev) return 0; } +static int azx_intel_display_power(struct azx *chip, bool enable) +{ + struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + + return hda_display_power(hda, enable); +} + /* * Check whether the current DMA position is acceptable for updating * periods. Returns non-zero if it's OK. @@ -796,7 +803,8 @@ static int azx_suspend(struct device *dev) if (chip->msi) pci_disable_msi(chip->pci); - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL + && chip->need_i915_power) hda_display_power(hda, false); return 0; } @@ -816,7 +824,8 @@ static int azx_resume(struct device *dev) if (chip->disabled || hda->init_failed) return 0; - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL + && chip->need_i915_power) { hda_display_power(hda, true); haswell_set_bclk(hda); } @@ -859,7 +868,8 @@ static int azx_runtime_suspend(struct device *dev) azx_stop_chip(chip); azx_enter_link_reset(chip); azx_clear_irq_pending(chip); - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL + && chip->need_i915_power) hda_display_power(hda, false); return 0; @@ -885,7 +895,8 @@ static int azx_runtime_resume(struct device *dev) if (!azx_has_pm_runtime(chip)) return 0; - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL + && chip->need_i915_power) { hda_display_power(hda, true); haswell_set_bclk(hda); } @@ -1105,7 +1116,8 @@ static int azx_free(struct azx *chip) release_firmware(chip->fw); #endif if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { - hda_display_power(hda, false); + if (chip->need_i915_power) + hda_display_power(hda, false); hda_i915_exit(hda); } kfree(hda); @@ -1757,6 +1769,7 @@ static const struct hda_controller_ops pci_hda_ops = { .substream_free_pages = substream_free_pages, .pcm_mmap_prepare = pcm_mmap_prepare, .position_check = azx_position_check, + .display_power = azx_intel_display_power, }; static int azx_probe(struct pci_dev *pci, @@ -1855,12 +1868,12 @@ static int azx_probe_continue(struct azx *chip) #ifdef CONFIG_SND_HDA_I915 err = hda_i915_init(hda); if (err < 0) - goto out_free; + goto i915_power_fail; err = hda_display_power(hda, true); if (err < 0) { dev_err(chip->card->dev, "Cannot turn on display power on i915\n"); - goto out_free; + goto i915_power_fail; } #endif } @@ -1911,6 +1924,10 @@ static int azx_probe_continue(struct azx *chip) pm_runtime_put_noidle(&pci->dev); out_free: + if (!chip->need_i915_power) + hda_display_power(hda, false); + +i915_power_fail: if (err < 0) hda->init_failed = 1; complete_all(&hda->probe_wait); -- 1.9.1

3 10

[alsa-devel] Audio crackles with 4.1-rc1
by Jonathan McDowell 28 Apr '15

28 Apr '15

Having upgraded to 4.1-rc1 from 4.0 I'm now hearing audio crackles at regular intervals. I'm fairly sure this is due to the HDA power save as once audio is playing things are fine, it's just when starting to play audio that I hear the crackle. System is a Dell Latitude E7240. I haven't tried a bisect yet but will attempt to find some time to do so in the next few days. It looks like there have been some changes in sound/hda/ between 4.0 + 4.1-rc1 so I'll concentrate on those first. Hopefully relevant dmesg output for the device: [ 23.551075] snd_hda_intel 0000:00:03.0: enabling device (0000 -> 0002) [ 23.551279] snd_hda_intel 0000:00:03.0: bound 0000:00:02.0 (ops 0xffffffff81891a00) [ 23.575852] snd_hda_codec_realtek hdaudioC1D0: autoconfig for ALC3226: line_outs=1 (0x16/0x0/0x0/0x0/0x0) type:line [ 23.575855] snd_hda_codec_realtek hdaudioC1D0: speaker_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 23.575857] snd_hda_codec_realtek hdaudioC1D0: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 23.575858] snd_hda_codec_realtek hdaudioC1D0: mono: mono_out=0x0 [ 23.575859] snd_hda_codec_realtek hdaudioC1D0: inputs: [ 23.575861] snd_hda_codec_realtek hdaudioC1D0: Dock Mic=0x19 [ 23.575863] snd_hda_codec_realtek hdaudioC1D0: Headset Mic=0x1a [ 23.575870] snd_hda_codec_realtek hdaudioC1D0: Internal Mic=0x12 [ 23.582324] input: HDA Intel HDMI HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:03.0/sound/card0/input12 [ 23.582397] input: HDA Intel HDMI HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:03.0/sound/card0/input13 [ 23.582747] input: HDA Intel HDMI HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:03.0/sound/card0/input14 [ 23.593267] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/sound/card1/input9 [ 23.594102] input: HDA Intel PCH Dock Mic as /devices/pci0000:00/0000:00:1b.0/sound/card1/input15 [ 23.594268] input: HDA Intel PCH Dock Line Out as /devices/pci0000:00/0000:00:1b.0/sound/card1/input16 [ 23.594346] input: HDA Intel PCH Front Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card1/input17 I have: CONFIG_SND_HDA_POWER_SAVE_DEFAULT=600 set in my kernel config. There's no dmesg output around the times when the crackling is occurring. J. -- Friends are God's apology for relations.

2 4

[alsa-devel] M-Audio Midisport 2x2 Anniversary edition, latency problem
by Ilya Dikariev 28 Apr '15

28 Apr '15

Hi all, I bought a MIDI-interface von M-Audio (see subject), but there are strange things with it on my desktop (Arch, latest kernel). The latency is about 5 seconds. If I reconnect cabels, the latency can change from 5 sec to 0.5 sec, or to ~0sec, but in 90% of all the tries, the latency stay at 3-5 sec. I tried it on my older laptop (kernel 2.6.32 from Ubuntu 10.04) and it works! I think it's a regression. How can I help to fix this bug? best regards, Ilya.

2 1

[alsa-devel] [PATCH] conf/ucm: broadwell-rt286: change to use the correct jack kcontrol name
by Jie Yang 28 Apr '15

28 Apr '15

The headset jack has two kctls: "Headphone Jack" and "Mic Jack", we need switch speaker output according to the former JackControl. Here correct it. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- src/conf/ucm/broadwell-rt286/HiFi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/conf/ucm/broadwell-rt286/HiFi b/src/conf/ucm/broadwell-rt286/HiFi index 08f5c50..58983ea 100644 --- a/src/conf/ucm/broadwell-rt286/HiFi +++ b/src/conf/ucm/broadwell-rt286/HiFi @@ -43,7 +43,7 @@ SectionDevice."Headphones" { Value { PlaybackChannels "2" JackDev "rt286-jack" - JackControl "Headset Jack" + JackControl "Headphone Jack" JackHWMute "Speaker" } } -- 1.9.1

2 1

[alsa-devel] [PATCH 03/12] trivial: ASoC: pxa: pxa-ssp.c: indent with TABs, not spaces
by Antonio Ospite 28 Apr '15

28 Apr '15

Signed-off-by: Antonio Ospite <ao2(a)ao2.it> Cc: Daniel Mack <daniel(a)zonque.org> Cc: Mark Brown <broonie(a)kernel.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/pxa/pxa-ssp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index fbe2e93..eb181ab 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -586,7 +586,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - sspsp = pxa_ssp_read_reg(ssp, SSPSP); + sspsp = pxa_ssp_read_reg(ssp, SSPSP); if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) { /* This is a special case where the bitclk is 64fs -- 2.1.4

1 0

[alsa-devel] [PATCH v10 0/9] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 28 Apr '15

28 Apr '15

Based on broonie/ASoC v10: - add the generic dt-card - define the audio ports from a DT graph of ports (Russell King) - reuse HDMI constants (Andrew Jackson - Jyri Sarha) - alloc rate_constraints in codec (Jyri Sarha) - fix bad number of channels (Jyri Sarha) - correct codec generation from config (Russell King - Jyri Sarha) - no module init/exit (Russell King) v9: - back to a TDA998x specific CODEC - more comments - change magic values to constants v8: - change some comments about the patches v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (9): ASoC: kirkwood: dynamically build the DAI array ASoC: kirkwood: check the DAI type from the DAI name ASoC: kirkwood: accept the DAI definitions from a graph of ports drm/i2c: tda998x: Add support of a DT graph of ports drm/i2c: tda998x: Change drvdata for audio extension ASoC: tda998x: add a codec to the HDMI transmitter drm/i2c: tda998x: set cts_n according to the sample width ASoC: core: export snd_soc_get_dai_name ASoC: add generic dt-card support .../devicetree/bindings/drm/i2c/tda998x.txt | 51 ++++ .../devicetree/bindings/sound/mvebu-audio.txt | 30 +++ drivers/gpu/drm/i2c/tda998x_drv.c | 237 +++++++++++++++++- include/sound/soc.h | 2 + include/sound/tda998x.h | 32 +++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 175 +++++++++++++ sound/soc/generic/Kconfig | 2 + sound/soc/generic/Makefile | 2 + sound/soc/generic/dt-card.c | 275 +++++++++++++++++++++ sound/soc/kirkwood/kirkwood-i2s.c | 141 ++++++----- sound/soc/kirkwood/kirkwood.h | 4 +- sound/soc/soc-core.c | 5 +- 14 files changed, 877 insertions(+), 86 deletions(-) create mode 100644 include/sound/tda998x.h create mode 100644 sound/soc/codecs/tda998x.c create mode 100644 sound/soc/generic/dt-card.c -- 2.1.4

3 13

[alsa-devel] [PATCH v10 0/7] ALSA: jack: Refactoring for jack kctls
by Jie Yang 28 Apr '15

28 Apr '15

This patch series aligns jack handling between the HDA and ASoC subsystems. The HDA subsystem currently creates a kcontrol for each HDA jack registered. ASoC subsystem creates input device jacks for each jack registered. This series aligns both jack APIs to create kcontrol jacks. This implementation keeps compatibility with existing HDA jacks so no userspace changes are required. Changes in v10: 1. rebase it on top of tiwai/sound.git for-linus branch; 2. Liam helps fix some typos and syntax errors in comments and descriptions. Changes in v9: 1. move struct snd_jack_kctl definition to jack.c, to hides the internal implementation details; 2. some small code fixing; 3. update documentation file. Changes in v8: 1. remove snd_jack_kctl from HDA part, change snd_jack_new param to use bool for creating kcontrol at jack creating stage; 2. create jack for phantom jack, with no input device; 3. remove HDA jack get_unique_index() and index related code. Changes in v7: 1. move name generating and index getting part into ctljack.c; 2. remove exposing private_data and destructor to HDA, then jack core can handle kctls by itself totally. 3. some small code fixing. Changes in v6: 1. refine kctl name generating function; 2. integrate HDA jack kctl more deeply, support phantom jack in snd_jack_new(); 3. add documentation for Jack kcontrols to explain how to use it. Changes in v5: 1. remove SND_KCTL_JACK config item, we need jack kctl once CONFIG_SND_JACK is selected. Changes in v4: 1. use snd_ctl_find_id() to get avaliable index; 2. add initial kctl for snd_jack_new(compatible for HDA); 3. add struct snd_jack_kctl * field to struct hda_jack_tbl; 4. new kctls for soc jack during jack pins creating. 5. add a patch to remove exporting snd_kctl_jack_new(). 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 (7): ALSA: jack: implement kctl creating for jack devices ALSA: Jack: handle jack embedded kcontrol creating within ctljack ALSA: jack: extend snd_jack_new to support phantom jack ALSA: hda - Update to use the new jack kctls method ASoC: jack: create kctls according to jack pins info ALSA: jack: remove exporting ctljack functions ALSA: Docs: Add documentation for Jack kcontrols Documentation/sound/alsa/Jack-Controls.txt | 43 ++++++++++ include/sound/control.h | 2 +- include/sound/jack.h | 13 ++- sound/core/Kconfig | 3 - sound/core/Makefile | 3 +- sound/core/ctljack.c | 41 +++++++-- sound/core/jack.c | 133 ++++++++++++++++++++++++++--- sound/pci/hda/Kconfig | 2 +- sound/pci/hda/hda_jack.c | 90 ++++++------------- sound/pci/hda/hda_jack.h | 5 +- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/soc-jack.c | 3 +- 13 files changed, 242 insertions(+), 100 deletions(-) create mode 100644 Documentation/sound/alsa/Jack-Controls.txt -- 1.9.1

2 8

[alsa-devel] [PATCH] ASoC: rt5645: fixed kbuild err
by yang.a.fang＠intel.com 28 Apr '15

28 Apr '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> kbuild robot reports following error/warnings sound/soc/codecs/rt5645.c: In function 'rt5645_i2c_probe': >> sound/soc/codecs/rt5645.c:2720:4: error: implicit declaration of >> function 'devm_gpiod_get_index' >> [-Werror=implicit-function-declaration] gpiod = devm_gpiod_get_index(&i2c->dev, "rt5645", 0); ^ >> sound/soc/codecs/rt5645.c:2720:10: warning: assignment makes pointer >> from integer without a cast gpiod = devm_gpiod_get_index(&i2c->dev, "rt5645", 0); ^ >> sound/soc/codecs/rt5645.c:2722:4: error: implicit declaration of >> function 'gpiod_direction_input' >> [-Werror=implicit-function-declaration] if (IS_ERR(gpiod) || gpiod_direction_input(gpiod)) { ^ >> sound/soc/codecs/rt5645.c:2726:5: error: implicit declaration of >> function 'desc_to_gpio' [-Werror=implicit-function-declaration] rt5645->pdata.hp_det_gpio = desc_to_gpio(gpiod); ^ >> sound/soc/codecs/rt5645.c:2728:7: error: implicit declaration of >> function 'gpiod_is_active_low' >> [-Werror=implicit-function-declaration] = !gpiod_is_active_low(gpiod); ^ cc1: some warnings being treated as errors Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/codecs/rt5645.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index fb561b4..2cab2eb 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -18,6 +18,7 @@ #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/acpi.h> #include <linux/dmi.h> #include <sound/core.h> -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt5645: Add ACPI match ID
by yang.a.fang＠intel.com 28 Apr '15

28 Apr '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> This patch adds the ACPI match ID for rt5645/5650 codec Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/codecs/rt5645.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index c9a4c5b..e16724a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -18,6 +18,7 @@ #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/gpio.h> +#include <linux/acpi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -2618,6 +2619,15 @@ static const struct i2c_device_id rt5645_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5645_i2c_id); +#ifdef CONFIG_ACPI +static struct acpi_device_id rt5645_acpi_match[] = { + { "10EC5645", 0 }, + { "10EC5650", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5645_acpi_match); +#endif + static int rt5645_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2834,6 +2844,7 @@ static struct i2c_driver rt5645_i2c_driver = { .driver = { .name = "rt5645", .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5645_acpi_match), }, .probe = rt5645_i2c_probe, .remove = rt5645_i2c_remove, -- 1.7.9.5

2 5

[alsa-devel] [PATCH v2] ASoC: rt5645: Adds push button support for rt5650
by bardliao＠realtek.com 27 Apr '15

27 Apr '15

From: Bard Liao <bardliao(a)realtek.com> rt5650 support headset button detection. Currently, the button detection is only implemented for rt5650 codec. The button detection configuration register's default value is different from rt5645. And we didn't touch the register in the driver, so we will get the wrong value when we dump the registers. We will fix it in another patch. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * remove the change of removing the value of 0xdd in rt5645_reg[]. * add comment of handle three button detection code in one case. Mark, Can we have a separate codec driver for rt5650? rt5650 will be more popular than rt5645. Many customers are searching rt5650.c but can't find one. We have to told them rt5650 and rt5645 share the same codec driver. Also, some registers or functions are different between rt5645 and rt5650. Thanks. --- sound/soc/codecs/rt5645.c | 266 +++++++++++++++++++++++++++++++++------ sound/soc/codecs/rt5645.h | 8 +- sound/soc/intel/cht_bsw_rt5645.c | 2 +- 3 files changed, 237 insertions(+), 39 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 69528ae..2d80568 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2368,6 +2368,8 @@ static int rt5645_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, static int rt5645_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_PREPARE: if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { @@ -2398,8 +2400,9 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: snd_soc_write(codec, RT5645_DEPOP_M2, 0x1100); - snd_soc_update_bits(codec, RT5645_GEN_CTRL1, - RT5645_DIG_GATE_CTRL, 0); + if (!rt5645->en_button_func) + snd_soc_update_bits(codec, RT5645_GEN_CTRL1, + RT5645_DIG_GATE_CTRL, 0); snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_VREF1 | RT5645_PWR_MB | RT5645_PWR_BG | RT5645_PWR_VREF2 | @@ -2414,28 +2417,71 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int rt5645_jack_detect(struct snd_soc_codec *codec) +static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, + bool enable) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - int gpio_state, jack_type = 0; - unsigned int val; - if (!gpio_is_valid(rt5645->pdata.hp_det_gpio)) { - dev_err(codec->dev, "invalid gpio\n"); - return -EINVAL; + if (enable) { + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "ADC L power"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "ADC R power"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "LDO2"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync_unlocked(&codec->dapm); + snd_soc_update_bits(codec, + RT5645_INT_IRQ_ST, 0x8, 0x8); + snd_soc_update_bits(codec, + RT5650_4BTN_IL_CMD2, 0x8000, 0x8000); + snd_soc_read(codec, RT5650_4BTN_IL_CMD1); + pr_debug("%s read %x = %x\n", __func__, RT5650_4BTN_IL_CMD1, + snd_soc_read(codec, RT5650_4BTN_IL_CMD1)); + } else { + snd_soc_update_bits(codec, RT5650_4BTN_IL_CMD2, 0x8000, 0x0); + snd_soc_update_bits(codec, RT5645_INT_IRQ_ST, 0x8, 0x0); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "ADC L power"); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "ADC R power"); + if (rt5645->pdata.jd_mode == 0) + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "LDO2"); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync_unlocked(&codec->dapm); } - gpio_state = gpio_get_value(rt5645->pdata.hp_det_gpio); +} - dev_dbg(codec->dev, "gpio = %d(%d)\n", rt5645->pdata.hp_det_gpio, - gpio_state); +static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + unsigned int val; - if ((rt5645->pdata.gpio_hp_det_active_high && gpio_state) || - (!rt5645->pdata.gpio_hp_det_active_high && !gpio_state)) { - snd_soc_dapm_force_enable_pin(&codec->dapm, "micbias1"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "micbias2"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "Mic Det Power"); - snd_soc_dapm_sync(&codec->dapm); + if (jack_insert) { + if (codec->component.card->instantiated) { + snd_soc_dapm_force_enable_pin(&codec->dapm, + "micbias1"); + snd_soc_dapm_force_enable_pin(&codec->dapm, + "micbias2"); + snd_soc_dapm_force_enable_pin(&codec->dapm, + "LDO2"); + snd_soc_dapm_force_enable_pin(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } else { + /* Power up necessary bits for JD if dapm is + not ready yet */ + snd_soc_update_bits(codec, RT5645_PWR_ANLG2, + RT5645_PWR_MB1 | RT5645_PWR_MB2, + RT5645_PWR_MB1 | RT5645_PWR_MB2); + snd_soc_update_bits(codec, RT5645_PWR_MIXER, + RT5645_PWR_LDO2, RT5645_PWR_LDO2); + snd_soc_update_bits(codec, RT5645_PWR_VOL, + RT5645_PWR_MIC_DET, RT5645_PWR_MIC_DET); + } snd_soc_write(codec, RT5645_IN1_CTRL1, 0x0006); snd_soc_write(codec, RT5645_JD_CTRL3, 0x00b0); @@ -2449,32 +2495,62 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec) val = snd_soc_read(codec, RT5645_IN1_CTRL3) & 0x7; dev_dbg(codec->dev, "val = %d\n", val); - if (val == 1 || val == 2) - jack_type = SND_JACK_HEADSET; - else - jack_type = SND_JACK_HEADPHONE; + if (codec->component.card->instantiated) { + snd_soc_dapm_disable_pin(&codec->dapm, "micbias1"); + snd_soc_dapm_disable_pin(&codec->dapm, "micbias2"); + if (rt5645->pdata.jd_mode == 0) + snd_soc_dapm_disable_pin(&codec->dapm, "LDO2"); + snd_soc_dapm_disable_pin(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } else { + snd_soc_update_bits(codec, RT5645_PWR_ANLG2, + RT5645_PWR_MB1 | RT5645_PWR_MB2, 0); + if (rt5645->pdata.jd_mode == 0) + snd_soc_update_bits(codec, RT5645_PWR_MIXER, + RT5645_PWR_LDO2, 0); + snd_soc_update_bits(codec, RT5645_PWR_VOL, + RT5645_PWR_MIC_DET, 0); + } - snd_soc_dapm_disable_pin(&codec->dapm, "micbias1"); - snd_soc_dapm_disable_pin(&codec->dapm, "micbias2"); - if (rt5645->pdata.jd_mode == 0) - snd_soc_dapm_disable_pin(&codec->dapm, "LDO2"); - snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); - snd_soc_dapm_sync(&codec->dapm); + if (val == 1 || val == 2) { + rt5645->jack_type = SND_JACK_HEADSET; + if (rt5645->en_button_func) { + msleep(100); + rt5645_enable_push_button_irq(codec, true); + } + } else { + rt5645->jack_type = SND_JACK_HEADPHONE; + } + + } else { /* jack out */ + rt5645->jack_type = 0; + if (rt5645->en_button_func) + rt5645_enable_push_button_irq(codec, false); } - snd_soc_jack_report(rt5645->hp_jack, jack_type, SND_JACK_HEADPHONE); - snd_soc_jack_report(rt5645->mic_jack, jack_type, SND_JACK_MICROPHONE); - return 0; + return rt5645->jack_type; } int rt5645_set_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack) + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack, + struct snd_soc_jack *btn_jack) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); rt5645->hp_jack = hp_jack; rt5645->mic_jack = mic_jack; - rt5645_jack_detect(codec); + rt5645->btn_jack = btn_jack; + if (rt5645->btn_jack && rt5645->codec_type == CODEC_TYPE_RT5650) { + rt5645->en_button_func = true; + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ); + regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1, + RT5645_HP_CB_MASK, RT5645_HP_CB_PU); + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1, + RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); + } + rt5645_irq_detection(rt5645); return 0; } @@ -2485,7 +2561,7 @@ static void rt5645_jack_detect_work(struct work_struct *work) struct rt5645_priv *rt5645 = container_of(work, struct rt5645_priv, jack_detect_work.work); - rt5645_jack_detect(rt5645->codec); + rt5645_irq_detection(rt5645); } static irqreturn_t rt5645_irq(int irq, void *data) @@ -2498,6 +2574,125 @@ static irqreturn_t rt5645_irq(int irq, void *data) return IRQ_HANDLED; } +static int rt5645_button_detect(struct snd_soc_codec *codec) +{ + int btn_type, val; + + val = snd_soc_read(codec, RT5650_4BTN_IL_CMD1); + pr_debug("val=0x%x\n", val); + btn_type = val & 0xfff0; + snd_soc_write(codec, RT5650_4BTN_IL_CMD1, val); + + return btn_type; +} + +static int rt5645_irq_detection(struct rt5645_priv *rt5645) +{ + int val, btn_type, gpio_state = 0, report = 0; + + switch (rt5645->pdata.jd_mode) { + case 0: /* Not using rt5645 JD */ + if (gpio_is_valid(rt5645->pdata.hp_det_gpio)) { + gpio_state = gpio_get_value(rt5645->pdata.hp_det_gpio); + dev_dbg(rt5645->codec->dev, "gpio = %d(%d)\n", + rt5645->pdata.hp_det_gpio, gpio_state); + } + if ((rt5645->pdata.gpio_hp_det_active_high && gpio_state) || + (!rt5645->pdata.gpio_hp_det_active_high && + !gpio_state)) { + report = rt5645_jack_detect(rt5645->codec, 1); + } else { + report = rt5645_jack_detect(rt5645->codec, 0); + } + snd_soc_jack_report(rt5645->hp_jack, + report, SND_JACK_HEADPHONE); + snd_soc_jack_report(rt5645->mic_jack, + report, SND_JACK_MICROPHONE); + return report; + case 1: /* 2 port */ + val = snd_soc_read(rt5645->codec, RT5645_A_JD_CTRL1) & 0x0070; + break; + default: /* 1 port */ + val = snd_soc_read(rt5645->codec, RT5645_A_JD_CTRL1) & 0x0020; + break; + + } + + switch (val) { + /* jack in */ + case 0x30: /* 2 port */ + case 0x0: /* 1 port or 2 port */ + if (rt5645->jack_type == 0) { + report = rt5645_jack_detect(rt5645->codec, 1); + /* for push button and jack out */ + break; + } + btn_type = 0; + if (snd_soc_read(rt5645->codec, RT5645_INT_IRQ_ST) & 0x4) { + /* button pressed */ + report = SND_JACK_HEADSET; + btn_type = rt5645_button_detect(rt5645->codec); + /* rt5650 can report three kinds of button behavior, + one click, double click and hold. However, + currently we will report button pressed/released + event. So all the three button behaviors are + treated as button pressed. */ + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + report |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + report |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + report |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + report |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + dev_err(rt5645->codec->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + } + if (btn_type == 0)/* button release */ + report = rt5645->jack_type; + + break; + /* jack out */ + case 0x70: /* 2 port */ + case 0x10: /* 2 port */ + case 0x20: /* 1 port */ + report = 0; + snd_soc_update_bits(rt5645->codec, + RT5645_INT_IRQ_ST, 0x1, 0x0); + rt5645_jack_detect(rt5645->codec, 0); + break; + default: + break; + } + + snd_soc_jack_report(rt5645->hp_jack, report, SND_JACK_HEADPHONE); + snd_soc_jack_report(rt5645->mic_jack, report, SND_JACK_MICROPHONE); + if (rt5645->en_button_func) + snd_soc_jack_report(rt5645->btn_jack, + report, SND_JACK_MICROPHONE); + + return report; +} + static int rt5645_probe(struct snd_soc_codec *codec) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); @@ -2784,8 +2979,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (rt5645->pdata.en_jd_func) { regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3, - RT5645_IRQ_CLK_GATE_CTRL | RT5645_MICINDET_MANU, - RT5645_IRQ_CLK_GATE_CTRL | RT5645_MICINDET_MANU); + RT5645_IRQ_CLK_GATE_CTRL, RT5645_IRQ_CLK_GATE_CTRL); regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, RT5645_CBJ_BST1_EN, RT5645_CBJ_BST1_EN); regmap_update_bits(rt5645->regmap, RT5645_JD_CTRL3, diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index db78e94..4473636 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2184,6 +2184,7 @@ struct rt5645_priv { struct i2c_client *i2c; struct snd_soc_jack *hp_jack; struct snd_soc_jack *mic_jack; + struct snd_soc_jack *btn_jack; struct delayed_work jack_detect_work; int codec_type; @@ -2196,9 +2197,12 @@ struct rt5645_priv { int pll_src; int pll_in; int pll_out; + + int jack_type; + bool en_button_func; }; int rt5645_set_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack); - + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack, + struct snd_soc_jack *btn_jack); #endif /* __RT5645_H__ */ diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/cht_bsw_rt5645.c index bd29617..3d0b549 100644 --- a/sound/soc/intel/cht_bsw_rt5645.c +++ b/sound/soc/intel/cht_bsw_rt5645.c @@ -185,7 +185,7 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) return ret; } - rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack); + rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack, NULL); return ret; } -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 1/4] ASoC: Add helper function getting CODEC's DAPM context
by Lars-Peter Clausen 27 Apr '15

27 Apr '15

The DAPM context in the snd_soc_codec struct is redundant and scheduled to be replaced by the DAPM context in the snd_soc_component struct. This patch introduces a new helper function snd_soc_codec_get_dapm() which should be used for getting the DAPM context for a CODEC rather then directly accessing the dapm field. Once there are no more direct users of the dapm field left it is possible to transparently switch all drivers to the component DAPM context by updating snd_soc_codec_get_dapm() function. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f6226914..d57dc7c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -819,7 +819,7 @@ struct snd_soc_codec { /* component */ struct snd_soc_component component; - /* dapm */ + /* Don't access this directly, use snd_soc_codec_get_dapm() */ struct snd_soc_dapm_context dapm; #ifdef CONFIG_DEBUG_FS @@ -1282,6 +1282,18 @@ static inline struct snd_soc_dapm_context *snd_soc_component_get_dapm( } /** + * snd_soc_codec_get_dapm() - Returns the DAPM context for the CODEC + * @codec: The CODEC for which to get the DAPM context + * + * Note: Use this function instead of directly accessing the CODEC's dapm field + */ +static inline struct snd_soc_dapm_context *snd_soc_codec_get_dapm( + struct snd_soc_codec *codec) +{ + return &codec->dapm; +} + +/** * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol * @kcontrol: The kcontrol * -- 1.8.0

2 4

[alsa-devel] [PATCH RFT] ASoC: rt5645: Fix mask for setting RT5645_DMIC_2_DP_GPIO12 bit
by Axel Lin 27 Apr '15

27 Apr '15

Current code uses wrong mask when setting RT5645_DMIC_2_DP_GPIO12 bit, fix it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Bard, I don't have the h/w to test, so please review and test this patch. Thanks, Axel sound/soc/codecs/rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 69528ae..a249f65 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2770,7 +2770,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, case RT5645_DMIC_DATA_GPIO12: regmap_update_bits(rt5645->regmap, RT5645_DMIC_CTRL1, - RT5645_DMIC_1_DP_MASK, RT5645_DMIC_2_DP_GPIO12); + RT5645_DMIC_2_DP_MASK, RT5645_DMIC_2_DP_GPIO12); regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, RT5645_GP12_PIN_MASK, RT5645_GP12_PIN_DMIC2_SDA); -- 2.1.0

2 1

[alsa-devel] [PATCH: RESEND] Update email-id of author
by Rajeev Kumar 27 Apr '15

27 Apr '15

rajeev-dlh.kumar(a)st.com email-id doesn't exist anymore as I have left the company. Replace ST's id with Rajeev Kumar <rajeevkumar.linux(a)gmail.com> It also updates .mailmap file to fix address for 'git shortlog' Signed-off-by: Rajeev Kumar <rajeevkumar.linux(a)gmail.com> --- .mailmap | 1 + arch/arm/mach-spear/generic.h | 2 +- arch/arm/mach-spear/include/mach/irqs.h | 2 +- arch/arm/mach-spear/include/mach/spear.h | 2 +- arch/arm/mach-spear/spear6xx.c | 2 +- drivers/input/keyboard/spear-keyboard.c | 2 +- drivers/rtc/rtc-spear.c | 4 ++-- include/linux/platform_data/keyboard-spear.h | 2 +- include/sound/designware_i2s.h | 2 +- include/sound/spear_dma.h | 2 +- 10 files changed, 11 insertions(+), 10 deletions(-) diff --git a/.mailmap b/.mailmap index 0d971cf..c570755 100644 --- a/.mailmap +++ b/.mailmap @@ -96,6 +96,7 @@ Peter A Jonsson <pj(a)ludd.ltu.se> Peter Oruba <peter(a)oruba.de> Peter Oruba <peter.oruba(a)amd.com> Praveen BP <praveenbp(a)ti.com> +Rajeev Kumar <rajeevkumar.linux(a)gmail.com> Rajesh Shah <rajesh.shah(a)intel.com> Ralf Baechle <ralf(a)linux-mips.org> Ralf Wildenhues <Ralf.Wildenhues(a)gmx.de> diff --git a/arch/arm/mach-spear/generic.h b/arch/arm/mach-spear/generic.h index a99d90a..0bfd2ea 100644 --- a/arch/arm/mach-spear/generic.h +++ b/arch/arm/mach-spear/generic.h @@ -2,7 +2,7 @@ * spear machine family generic header file * * Copyright (C) 2009-2012 ST Microelectronics - * Rajeev Kumar <rajeev-dlh.kumar(a)st.com> + * Rajeev Kumar <rajeevkumar.linux(a)gmail.com> * Viresh Kumar <viresh.linux(a)gmail.com> * * This file is licensed under the terms of the GNU General Public diff --git a/arch/arm/mach-spear/include/mach/irqs.h b/arch/arm/mach-spear/include/mach/irqs.h index 92da0a8..022cf98 100644 --- a/arch/arm/mach-spear/include/mach/irqs.h +++ b/arch/arm/mach-spear/include/mach/irqs.h @@ -2,7 +2,7 @@ * IRQ helper macros for spear machine family * * Copyright (C) 2009-2012 ST Microelectronics - * Rajeev Kumar <rajeev-dlh.kumar(a)st.com> + * Rajeev Kumar <rajeevkumar.linux(a)gmail.com> * Viresh Kumar <viresh.linux(a)gmail.com> * * This file is licensed under the terms of the GNU General Public diff --git a/arch/arm/mach-spear/include/mach/spear.h b/arch/arm/mach-spear/include/mach/spear.h index f2d6a01..0d2f14d 100644 --- a/arch/arm/mach-spear/include/mach/spear.h +++ b/arch/arm/mach-spear/include/mach/spear.h @@ -2,7 +2,7 @@ * SPEAr3xx/6xx Machine family specific definition * * Copyright (C) 2009,2012 ST Microelectronics - * Rajeev Kumar<rajeev-dlh.kumar(a)st.com> + * Rajeev Kumar <rajeevkumar.linux(a)gmail.com> * Viresh Kumar <viresh.linux(a)gmail.com> * * This file is licensed under the terms of the GNU General Public diff --git a/arch/arm/mach-spear/spear6xx.c b/arch/arm/mach-spear/spear6xx.c index da26fa5b..c6dcb0d 100644 --- a/arch/arm/mach-spear/spear6xx.c +++ b/arch/arm/mach-spear/spear6xx.c @@ -4,7 +4,7 @@ * SPEAr6XX machines common source file * * Copyright (C) 2009 ST Microelectronics - * Rajeev Kumar<rajeev-dlh.kumar(a)st.com> + * Rajeev Kumar <rajeevkumar.linux(a)gmail.com> * * Copyright 2012 Stefan Roese <sr(a)denx.de> * diff --git a/drivers/input/keyboard/spear-keyboard.c b/drivers/input/keyboard/spear-keyboard.c index f42a543..623d451 100644 --- a/drivers/input/keyboard/spear-keyboard.c +++ b/drivers/input/keyboard/spear-keyboard.c @@ -3,7 +3,7 @@ * Based on omap-keypad driver * * Copyright (C) 2010 ST Microelectronics - * Rajeev Kumar<rajeev-dlh.kumar(a)st.com> + * Rajeev Kumar <rajeevkumar.linux(a)gmail.com> * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any diff --git a/drivers/rtc/rtc-spear.c b/drivers/rtc/rtc-spear.c index d2cdb98..d14be33 100644 --- a/drivers/rtc/rtc-spear.c +++ b/drivers/rtc/rtc-spear.c @@ -2,7 +2,7 @@ * drivers/rtc/rtc-spear.c * * Copyright (C) 2010 ST Microelectronics - * Rajeev Kumar<rajeev-dlh.kumar(a)st.com> + * Rajeev Kumar <rajeevkumar.linux(a)gmail.com> * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any @@ -504,6 +504,6 @@ static struct platform_driver spear_rtc_driver = { module_platform_driver(spear_rtc_driver); MODULE_ALIAS("platform:rtc-spear"); -MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar(a)st.com>"); +MODULE_AUTHOR("Rajeev Kumar <rajeevkumar.linux(a)gmail.com>"); MODULE_DESCRIPTION("ST SPEAr Realtime Clock Driver (RTC)"); MODULE_LICENSE("GPL"); diff --git a/include/linux/platform_data/keyboard-spear.h b/include/linux/platform_data/keyboard-spear.h index 9248e3a..5e3ff65 100644 --- a/include/linux/platform_data/keyboard-spear.h +++ b/include/linux/platform_data/keyboard-spear.h @@ -1,6 +1,6 @@ /* * Copyright (C) 2010 ST Microelectronics - * Rajeev Kumar<rajeev-dlh.kumar(a)st.com> + * Rajeev Kumar <rajeevkumar.linux(a)gmail.com> * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h index 26f406e..3a8fca9 100644 --- a/include/sound/designware_i2s.h +++ b/include/sound/designware_i2s.h @@ -1,5 +1,5 @@ /* - * Copyright (ST) 2012 Rajeev Kumar (rajeev-dlh.kumar(a)st.com) + * Copyright (ST) 2012 Rajeev Kumar (rajeevkumar.linux(a)gmail.com) * * 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 diff --git a/include/sound/spear_dma.h b/include/sound/spear_dma.h index 65aca51..e290de4 100644 --- a/include/sound/spear_dma.h +++ b/include/sound/spear_dma.h @@ -1,7 +1,7 @@ /* * linux/spear_dma.h * -* Copyright (ST) 2012 Rajeev Kumar (rajeev-dlh.kumar(a)st.com) +* Copyright (ST) 2012 Rajeev Kumar (rajeevkumar.linux(a)gmail.com) * * 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 -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: dmaengine_pcm: Make FLAG_NO_RESIDUE internal
by Lars-Peter Clausen 27 Apr '15

27 Apr '15

Whether residue can be reported or not is not a property of the audio controller but of the DMA controller. The FLAG_NO_RESIDUE was initially added when the DMAengine framework had no support for describing the residue reporting capabilities of the controller. Support for this was added quite a while ago and recently the DMAengine framework started to complain if a driver does not describe its capabilities and a lot of patches have been merged that add support for this where it was missing. So it should be safe to assume that driver on actively used platforms properly implement the DMA capabilities API. This patch makes the FLAG_NO_RESIDUE internal and no longer allows audio controller drivers to manually set the flag. If a DMA driver against expectations does not support reporting its capabilities for now the generic DMAengine PCM driver will now emit a warning and simply assume that residue reporting is not supported. In the future this might be changed to aborting with an error. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/dmaengine_pcm.h | 5 ----- sound/soc/atmel/atmel-pcm-dma.c | 3 +-- sound/soc/cirrus/ep93xx-pcm.c | 1 - sound/soc/fsl/fsl_sai.c | 3 +-- sound/soc/soc-generic-dmaengine-pcm.c | 25 ++++++++++++++----------- sound/soc/ux500/ux500_pcm.c | 1 - 6 files changed, 16 insertions(+), 22 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index eb73a3a..f86ef5e 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -91,11 +91,6 @@ void snd_dmaengine_pcm_set_config_from_dai_data( */ #define SND_DMAENGINE_PCM_FLAG_NO_DT BIT(1) /* - * The platforms dmaengine driver does not support reporting the amount of - * bytes that are still left to transfer. - */ -#define SND_DMAENGINE_PCM_FLAG_NO_RESIDUE BIT(2) -/* * The PCM is half duplex and the DMA channel is shared between capture and * playback. */ diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index b6625c8..dd57a9e 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -124,8 +124,7 @@ static const struct snd_dmaengine_pcm_config atmel_dmaengine_pcm_config = { int atmel_pcm_dma_platform_register(struct device *dev) { - return snd_dmaengine_pcm_register(dev, &atmel_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); + return snd_dmaengine_pcm_register(dev, &atmel_dmaengine_pcm_config, 0); } EXPORT_SYMBOL(atmel_pcm_dma_platform_register); diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c index 5f66447..67a7333 100644 --- a/sound/soc/cirrus/ep93xx-pcm.c +++ b/sound/soc/cirrus/ep93xx-pcm.c @@ -60,7 +60,6 @@ int devm_ep93xx_pcm_platform_register(struct device *dev) { return devm_snd_dmaengine_pcm_register(dev, &ep93xx_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ec79c3d..ee2671b 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -664,8 +664,7 @@ static int fsl_sai_probe(struct platform_device *pdev) if (sai->sai_on_imx) return imx_pcm_dma_init(pdev); else - return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); + return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); } static const struct of_device_id fsl_sai_ids[] = { diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index c9917ca..6fd1906 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -24,6 +24,12 @@ #include <sound/dmaengine_pcm.h> +/* + * The platforms dmaengine driver does not support reporting the amount of + * bytes that are still left to transfer. + */ +#define SND_DMAENGINE_PCM_FLAG_NO_RESIDUE BIT(31) + struct dmaengine_pcm { struct dma_chan *chan[SNDRV_PCM_STREAM_LAST + 1]; const struct snd_dmaengine_pcm_config *config; @@ -222,14 +228,18 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( return snd_dmaengine_pcm_request_channel(fn, dma_data->filter_data); } -static bool dmaengine_pcm_can_report_residue(struct dma_chan *chan) +static bool dmaengine_pcm_can_report_residue(struct device *dev, + struct dma_chan *chan) { struct dma_slave_caps dma_caps; int ret; ret = dma_get_slave_caps(chan, &dma_caps); - if (ret != 0) - return true; + if (ret != 0) { + dev_warn(dev, "Failed to get DMA channel capabilities, falling back to period counting: %d\n", + ret); + return false; + } if (dma_caps.residue_granularity == DMA_RESIDUE_GRANULARITY_DESCRIPTOR) return false; @@ -289,14 +299,7 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - /* - * This will only return false if we know for sure that at least - * one channel does not support residue reporting. If the DMA - * driver does not implement the slave_caps API we rely having - * the NO_RESIDUE flag set manually in case residue reporting is - * not supported. - */ - if (!dmaengine_pcm_can_report_residue(pcm->chan[i])) + if (!dmaengine_pcm_can_report_residue(dev, pcm->chan[i])) pcm->flags |= SND_DMAENGINE_PCM_FLAG_NO_RESIDUE; } diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 51a66a8..f12c01d 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -147,7 +147,6 @@ int ux500_pcm_register_platform(struct platform_device *pdev) pcm_config = &ux500_dmaengine_pcm_config; ret = snd_dmaengine_pcm_register(&pdev->dev, pcm_config, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | SND_DMAENGINE_PCM_FLAG_COMPAT); if (ret < 0) { dev_err(&pdev->dev, -- 1.8.0

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[alsa-devel] [PATCH v9 0/7] ALSA: jack: Refactoring for jack kctls
by Jie Yang 27 Apr '15

27 Apr '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 new implementation, both HDA and soc jack kctls works fine, they are compatible for old HDA jack kctls, too. For soc, a. snd_jack_new() with NULL jack_kctl; b. for each pin, call snd_jack_add_kctls() to add one kctl(it will call snd_ctl_add() to add kctl to card); For HDA, With snd_jack_new(), we can use phantom_jack = true for phantom jack creating, and the jack kctl pointer will be filled with the new created kcontrol. Changes in v9: 1. move struct snd_jack_kctl definition to jack.c, to hides the internal implementation details; 2. some small code fixing; 3. update documentation file. Changes in v8: 1. remove snd_jack_kctl from HDA part, change snd_jack_new param to use bool for creating kcontrol at jack creating stage; 2. create jack for phantom jack, with no input device; 3. remove HDA jack get_unique_index() and index related code. Changes in v7: 1. move name generating and index getting part into ctljack.c; 2. remove exposing private_data and destructor to hda, then jack core can handle kctls by itself totally. 3. some small code fixing. Changes in v6: 1. refine kctl name generating function; 2. integrate HDA jack kctl more deeply, support phantom jack in snd_jack_new(); 3. add documentation for Jack kcontrols to explain how to use it. Changes in v5: 1. remove SND_KCTL_JACK config item, we need jack kctl once CONFIG_SND_JACK is selected. Changes in v4: 1. use snd_ctl_find_id() to get avaliable index; 2. add initial kctl for snd_jack_new(compatible for HDA); 3. add struct snd_jack_kctl * field to struct hda_jack_tbl; 4. new kctls for soc jack during jack pins creating. 5. add a patch to remove exporting snd_kctl_jack_new(). 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 (7): ALSA: jack: implement kctl creating for jack device ALSA: Jack: handle jack embedded kcontrol creating within ctljack ALSA: jack: extend snd_jack_new to support phantom jack ALSA: hda - Update to use the new jack kctls method ASoC: jack: create kctls according to jack pins info ALSA: jack: remove exporting ctljack functions ALSA: Docs: Add documentation for Jack kcontrols Documentation/sound/alsa/Jack-Controls.txt | 49 ++++++++++ include/sound/control.h | 2 +- include/sound/jack.h | 13 ++- sound/core/Kconfig | 3 - sound/core/Makefile | 3 +- sound/core/ctljack.c | 41 ++++++-- sound/core/jack.c | 152 ++++++++++++++++++++++++++--- sound/pci/hda/Kconfig | 2 +- sound/pci/hda/hda_jack.c | 90 +++++------------ sound/pci/hda/hda_jack.h | 5 +- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/soc-jack.c | 3 +- 13 files changed, 266 insertions(+), 101 deletions(-) create mode 100644 Documentation/sound/alsa/Jack-Controls.txt -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda - Fix missing va_end() call in snd_hda_codec_pcm_new()
by Takashi Iwai 27 Apr '15

27 Apr '15

Reported by coverity CID 1296024. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 27333e0d8ebe..b49feff0a319 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -873,14 +873,15 @@ struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec, struct hda_pcm *pcm; va_list args; - va_start(args, fmt); pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); if (!pcm) return NULL; pcm->codec = codec; kref_init(&pcm->kref); + va_start(args, fmt); pcm->name = kvasprintf(GFP_KERNEL, fmt, args); + va_end(args); if (!pcm->name) { kfree(pcm); return NULL; -- 2.3.5

1 0

[alsa-devel] [PATCH] ALSA: emux: Fix mutex deadlock at unloading
by Takashi Iwai 27 Apr '15

27 Apr '15

The emux-synth driver has a possible AB/BA mutex deadlock at unloading the emu10k1 driver: snd_emux_free() -> snd_emux_detach_seq(): mutex_lock(&emu->register_mutex) -> snd_seq_delete_kernel_client() -> snd_seq_free_client(): mutex_lock(&register_mutex) snd_seq_release() -> snd_seq_free_client(): mutex_lock(&register_mutex) -> snd_seq_delete_all_ports() -> snd_emux_unuse(): mutex_lock(&emu->register_mutex) Basically snd_emux_detach_seq() doesn't need a protection of emu->register_mutex as it's already being unregistered. So, we can get rid of this for avoiding the deadlock. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/synth/emux/emux_seq.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/synth/emux/emux_seq.c b/sound/synth/emux/emux_seq.c index 7778b8e19782..188fda0effb0 100644 --- a/sound/synth/emux/emux_seq.c +++ b/sound/synth/emux/emux_seq.c @@ -124,12 +124,10 @@ snd_emux_detach_seq(struct snd_emux *emu) if (emu->voices) snd_emux_terminate_all(emu); - mutex_lock(&emu->register_mutex); if (emu->client >= 0) { snd_seq_delete_kernel_client(emu->client); emu->client = -1; } - mutex_unlock(&emu->register_mutex); } -- 2.3.5

1 0

Re: [alsa-devel] [PATCH 3/8] mfd: arizona: Add support for WM8998 and WM1814
by Mark Brown 27 Apr '15

27 Apr '15

On Tue, Apr 21, 2015 at 01:33:51PM +0100, Richard Fitzgerald wrote: > + switch (arizona->type) { > + case WM8998: > + case WM1814: > + /* Some bits are shifted on WM8998, > + * rearrange to match the standard bit layout > + */ > + val[0] = ((val[0] & 0x60e0) >> 1) | > + ((val[0] & 0x1e00) >> 2) | > + (val[0] & 0x000f); > + break; Are you sure this approach is going to scale (and avoid confusion)?

2 1

[alsa-devel] [PATCH] ALSA: emu10k1: Fix card shortname string buffer overflow
by Takashi Iwai 27 Apr '15

27 Apr '15

Some models provide too long string for the shortname that has 32bytes including the terminator, and it results in a non-terminated string exposed to the user-space. This isn't too critical, though, as the string is stopped at the succeeding longname string. This patch fixes such entries by dropping "SB" prefix (it's enough to fit within 32 bytes, so far). Meanwhile, it also changes strcpy() with strlcpy() to make sure that this kind of problem won't happen in future, too. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/emu10k1/emu10k1.c | 6 ++++-- sound/pci/emu10k1/emu10k1_main.c | 4 ++-- 2 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 37d0220a094c..db7a2e5e4a14 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -183,8 +183,10 @@ static int snd_card_emu10k1_probe(struct pci_dev *pci, } #endif - strcpy(card->driver, emu->card_capabilities->driver); - strcpy(card->shortname, emu->card_capabilities->name); + strlcpy(card->driver, emu->card_capabilities->driver, + sizeof(card->driver)); + strlcpy(card->shortname, emu->card_capabilities->name, + sizeof(card->shortname)); snprintf(card->longname, sizeof(card->longname), "%s (rev.%d, serial:0x%x) at 0x%lx, irq %i", card->shortname, emu->revision, emu->serial, emu->port, emu->irq); diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 54079f5d5673..4f8cf5e7e45f 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1446,7 +1446,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { * */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x20011102, - .driver = "Audigy2", .name = "SB Audigy 2 ZS Notebook [SB0530]", + .driver = "Audigy2", .name = "Audigy 2 ZS Notebook [SB0530]", .id = "Audigy2", .emu10k2_chip = 1, .ca0108_chip = 1, @@ -1596,7 +1596,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .adc_1361t = 1, /* 24 bit capture instead of 16bit */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10051102, - .driver = "Audigy2", .name = "SB Audigy 2 Platinum EX [SB0280]", + .driver = "Audigy2", .name = "Audigy 2 Platinum EX [SB0280]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, -- 2.3.5

1 0

[alsa-devel] [PATCH 1/2] sound: Close holes in struct snd_pcm_hw_rule
by Lars-Peter Clausen 27 Apr '15

27 Apr '15

On a 64-bit system there are two 32-bit holes due to the alignment of 64-bit fields. Reordering things slightly gets rid of those holes, reducing the size of the struct by 17% percent of its original size. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/pcm.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 0cb7f3f..d632809 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -224,9 +224,10 @@ typedef int (*snd_pcm_hw_rule_func_t)(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule { unsigned int cond; - snd_pcm_hw_rule_func_t func; int var; int deps[4]; + + snd_pcm_hw_rule_func_t func; void *private; }; -- 1.8.0

2 3

[alsa-devel] [PATCH 1/2] ALSA: hda - Fix mute-LED fixed mode
by Takashi Iwai 27 Apr '15

27 Apr '15

The mute-LED mode control has the fixed on/off states that are supposed to remain on/off regardless of the master switch. However, this doesn't work actually because the vmaster hook is called in the vmaster code itself. This patch fixes it by calling the hook indirectly after checking the mute LED mode. Reported-and-tested-by: Pali Rohár <pali.rohar(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 21 ++++++++++++--------- 1 file changed, 12 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 873ed1bce12b..27333e0d8ebe 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2082,6 +2082,16 @@ static struct snd_kcontrol_new vmaster_mute_mode = { .put = vmaster_mute_mode_put, }; +/* meta hook to call each driver's vmaster hook */ +static void vmaster_hook(void *private_data, int enabled) +{ + struct hda_vmaster_mute_hook *hook = private_data; + + if (hook->mute_mode != HDA_VMUTE_FOLLOW_MASTER) + enabled = hook->mute_mode; + hook->hook(hook->codec, enabled); +} + /** * snd_hda_add_vmaster_hook - Add a vmaster hook for mute-LED * @codec: the HDA codec @@ -2100,9 +2110,9 @@ int snd_hda_add_vmaster_hook(struct hda_codec *codec, if (!hook->hook || !hook->sw_kctl) return 0; - snd_ctl_add_vmaster_hook(hook->sw_kctl, hook->hook, codec); hook->codec = codec; hook->mute_mode = HDA_VMUTE_FOLLOW_MASTER; + snd_ctl_add_vmaster_hook(hook->sw_kctl, vmaster_hook, hook); if (!expose_enum_ctl) return 0; kctl = snd_ctl_new1(&vmaster_mute_mode, hook); @@ -2128,14 +2138,7 @@ void snd_hda_sync_vmaster_hook(struct hda_vmaster_mute_hook *hook) */ if (hook->codec->bus->shutdown) return; - switch (hook->mute_mode) { - case HDA_VMUTE_FOLLOW_MASTER: - snd_ctl_sync_vmaster_hook(hook->sw_kctl); - break; - default: - hook->hook(hook->codec, hook->mute_mode); - break; - } + snd_ctl_sync_vmaster_hook(hook->sw_kctl); } EXPORT_SYMBOL_GPL(snd_hda_sync_vmaster_hook); -- 2.3.5

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[alsa-devel] Pop noise on startup when headphones are plugged in (Dell XPS13 9333)
by Gabriele Mazzotta 27 Apr '15

27 Apr '15

Hi, I've recently found that commit a551d91473 ("ALSA: hda - Use regmap for command verb caches, too") is somehow causing a pop noise on startup when headphones are plugged in, but I couldn't figure out the exact cause. Was this observed on other systems (mine is a Dell XPS13 9333, Realtek ALC3661)? Does anyone have any idea of what the cause might be? Regards, Gabriele

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[alsa-devel] [PATCH] ASoC: Intel: fix Coccinelle kfree warning
by Firo Yang 27 Apr '15

27 Apr '15

It's not necessary to use kfree() to free memory allocated with devm_kzalloc in a probe function and using kfree leads to a double free. Signed-off-by: Firo Yang <firogm(a)gmail.com> --- sound/soc/intel/haswell/sst-haswell-ipc.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index 344a1e9..324eceb 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -2201,7 +2201,6 @@ dma_err: dsp_new_err: sst_ipc_fini(ipc); ipc_init_err: - kfree(hsw); return ret; } EXPORT_SYMBOL_GPL(sst_hsw_dsp_init); -- 2.1.0

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[alsa-devel] [PATCH] rajeev-dlh.kumar@st.com email-id doesn't exist anymore as I have left the company. Replace ST's id with Rajeev Kumar <rajeevkumar.linux@gmail.com>
by Rajeev Kumar 27 Apr '15

27 Apr '15

It also updates .mailmap file to fix address for 'git shortlog' Signed-off-by: Rajeev Kumar <rajeevkumar.linux(a)gmail.com> --- .mailmap | 1 + arch/arm/mach-spear/generic.h | 2 +- arch/arm/mach-spear/include/mach/irqs.h | 2 +- arch/arm/mach-spear/include/mach/spear.h | 2 +- arch/arm/mach-spear/spear6xx.c | 2 +- drivers/input/keyboard/spear-keyboard.c | 2 +- drivers/rtc/rtc-spear.c | 4 ++-- include/linux/platform_data/keyboard-spear.h | 2 +- include/sound/designware_i2s.h | 2 +- include/sound/spear_dma.h | 2 +- 10 files changed, 11 insertions(+), 10 deletions(-) diff --git a/.mailmap b/.mailmap index 0d971cf..c570755 100644 --- a/.mailmap +++ b/.mailmap @@ -96,6 +96,7 @@ Peter A Jonsson <pj(a)ludd.ltu.se> Peter Oruba <peter(a)oruba.de> Peter Oruba <peter.oruba(a)amd.com> Praveen BP <praveenbp(a)ti.com> +Rajeev Kumar <rajeevkumar.linux(a)gmail.com> Rajesh Shah <rajesh.shah(a)intel.com> Ralf Baechle <ralf(a)linux-mips.org> Ralf Wildenhues <Ralf.Wildenhues(a)gmx.de> diff --git a/arch/arm/mach-spear/generic.h b/arch/arm/mach-spear/generic.h index a99d90a..0bfd2ea 100644 --- a/arch/arm/mach-spear/generic.h +++ b/arch/arm/mach-spear/generic.h @@ -2,7 +2,7 @@ * spear machine family generic header file * * Copyright (C) 2009-2012 ST Microelectronics - * Rajeev Kumar <rajeev-dlh.kumar(a)st.com> + * Rajeev Kumar <rajeevkumar.linux(a)gmail.com> * Viresh Kumar <viresh.linux(a)gmail.com> * * This file is licensed under the terms of the GNU General Public diff --git a/arch/arm/mach-spear/include/mach/irqs.h b/arch/arm/mach-spear/include/mach/irqs.h index 92da0a8..022cf98 100644 --- a/arch/arm/mach-spear/include/mach/irqs.h +++ b/arch/arm/mach-spear/include/mach/irqs.h @@ -2,7 +2,7 @@ * IRQ helper macros for spear machine family * * Copyright (C) 2009-2012 ST Microelectronics - * Rajeev Kumar <rajeev-dlh.kumar(a)st.com> + * Rajeev Kumar <rajeevkumar.linux(a)gmail.com> * Viresh Kumar <viresh.linux(a)gmail.com> * * This file is licensed under the terms of the GNU General Public diff --git a/arch/arm/mach-spear/include/mach/spear.h b/arch/arm/mach-spear/include/mach/spear.h index f2d6a01..0d2f14d 100644 --- a/arch/arm/mach-spear/include/mach/spear.h +++ b/arch/arm/mach-spear/include/mach/spear.h @@ -2,7 +2,7 @@ * SPEAr3xx/6xx Machine family specific definition * * Copyright (C) 2009,2012 ST Microelectronics - * Rajeev Kumar<rajeev-dlh.kumar(a)st.com> + * Rajeev Kumar <rajeevkumar.linux(a)gmail.com> * Viresh Kumar <viresh.linux(a)gmail.com> * * This file is licensed under the terms of the GNU General Public diff --git a/arch/arm/mach-spear/spear6xx.c b/arch/arm/mach-spear/spear6xx.c index da26fa5b..c6dcb0d 100644 --- a/arch/arm/mach-spear/spear6xx.c +++ b/arch/arm/mach-spear/spear6xx.c @@ -4,7 +4,7 @@ * SPEAr6XX machines common source file * * Copyright (C) 2009 ST Microelectronics - * Rajeev Kumar<rajeev-dlh.kumar(a)st.com> + * Rajeev Kumar <rajeevkumar.linux(a)gmail.com> * * Copyright 2012 Stefan Roese <sr(a)denx.de> * diff --git a/drivers/input/keyboard/spear-keyboard.c b/drivers/input/keyboard/spear-keyboard.c index f42a543..623d451 100644 --- a/drivers/input/keyboard/spear-keyboard.c +++ b/drivers/input/keyboard/spear-keyboard.c @@ -3,7 +3,7 @@ * Based on omap-keypad driver * * Copyright (C) 2010 ST Microelectronics - * Rajeev Kumar<rajeev-dlh.kumar(a)st.com> + * Rajeev Kumar <rajeevkumar.linux(a)gmail.com> * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any diff --git a/drivers/rtc/rtc-spear.c b/drivers/rtc/rtc-spear.c index d2cdb98..d14be33 100644 --- a/drivers/rtc/rtc-spear.c +++ b/drivers/rtc/rtc-spear.c @@ -2,7 +2,7 @@ * drivers/rtc/rtc-spear.c * * Copyright (C) 2010 ST Microelectronics - * Rajeev Kumar<rajeev-dlh.kumar(a)st.com> + * Rajeev Kumar <rajeevkumar.linux(a)gmail.com> * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any @@ -504,6 +504,6 @@ static struct platform_driver spear_rtc_driver = { module_platform_driver(spear_rtc_driver); MODULE_ALIAS("platform:rtc-spear"); -MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar(a)st.com>"); +MODULE_AUTHOR("Rajeev Kumar <rajeevkumar.linux(a)gmail.com>"); MODULE_DESCRIPTION("ST SPEAr Realtime Clock Driver (RTC)"); MODULE_LICENSE("GPL"); diff --git a/include/linux/platform_data/keyboard-spear.h b/include/linux/platform_data/keyboard-spear.h index 9248e3a..5e3ff65 100644 --- a/include/linux/platform_data/keyboard-spear.h +++ b/include/linux/platform_data/keyboard-spear.h @@ -1,6 +1,6 @@ /* * Copyright (C) 2010 ST Microelectronics - * Rajeev Kumar<rajeev-dlh.kumar(a)st.com> + * Rajeev Kumar <rajeevkumar.linux(a)gmail.com> * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h index 26f406e..3a8fca9 100644 --- a/include/sound/designware_i2s.h +++ b/include/sound/designware_i2s.h @@ -1,5 +1,5 @@ /* - * Copyright (ST) 2012 Rajeev Kumar (rajeev-dlh.kumar(a)st.com) + * Copyright (ST) 2012 Rajeev Kumar (rajeevkumar.linux(a)gmail.com) * * 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 diff --git a/include/sound/spear_dma.h b/include/sound/spear_dma.h index 65aca51..e290de4 100644 --- a/include/sound/spear_dma.h +++ b/include/sound/spear_dma.h @@ -1,7 +1,7 @@ /* * linux/spear_dma.h * -* Copyright (ST) 2012 Rajeev Kumar (rajeev-dlh.kumar(a)st.com) +* Copyright (ST) 2012 Rajeev Kumar (rajeevkumar.linux(a)gmail.com) * * 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 -- 1.7.9.5

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[alsa-devel] [PATCH v3 0/2] Add support for select accessory detect mode to HPDETL or HPDETR
by Inha Song 27 Apr '15

27 Apr '15

This set of patches adds support for select accessory detect mode to HPDETL or HPDETR. Changes in v3: - Set the hpdet_channel default value to HPDETL If the value is unknown or invalid. Changes in v2: - Use the value in pdata instead of hpdet_channel in extcon_info. - Wrap arizona_extcon_of_get in IS_ENABLED(CONFIG_OF). - Change hpdet_channel type to unsigned from signed in pdata. - Move ARIZONA_ACCDET_MODE_* define to dt-binding header and directly set it to pdata. Inha Song (2): extcon: arizona: Add support for select accessory detect mode when headphone detection mfd: arizona: Update DT binding to support hpdet channel Documentation/devicetree/bindings/mfd/arizona.txt | 6 ++++ drivers/extcon/extcon-arizona.c | 38 ++++++++++++++++++----- include/dt-bindings/mfd/arizona.h | 8 +++++ include/linux/mfd/arizona/pdata.h | 3 ++ 4 files changed, 47 insertions(+), 8 deletions(-) create mode 100644 include/dt-bindings/mfd/arizona.h -- 2.0.0.390.gcb682f8

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Re: [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 25 Apr '15

25 Apr '15

Takashi Sakamoto wrote: > I do not understand techinical details, but the sound was > > perfect even with 192kHz/24bit audio using libffado. > > Ditto. libffado works with such broken stream by ignoring its packet > index. I guess that you've continued to use your device with > uns-ynchronized state for several years because no one reports it. > Well, one reason I believed I was doing fine with my Orpheus is one of the experiments I did in November 2010. ------------------------------------------------------------------------- I copied a 44.1kHz/16bit sound file into a USB memory, and played it on PS3. The optical output (SPDIF) of PS3 is fed into the optical input of Orpheus, and the data is captured using ffado and jackd. Two files, the original and the captured, are compared and showed perfect match. https://spinorlab.wordpress.com/2015/04/25/orpheus-and-bit-perfect/ (see https://spinorlab.files.wordpress.com/2015/04/orpheusbitperfect.jpg) ------------------------------------------------------------------------- I know I should have compared by not only capturing by Orpheus but playing by Orpheus, but I saved my effort assuming the reverse way should also be OK. (Actually, I did not expect any negative results. Was I too optimistic?) *** Therefore, I still can not believe I am experiencing packet losses (without noticing). I mean not with juju+libffado+jackd. I really don't know what is happening with snd-bebob. *** > So if it's only a > > #define TIMER_SOMETHING value_anything > > type of things... > > I cannot understand what you mean. > Never mind. I am not a very technical person anyway. I just thought if it is due to a VERY STRICT timeout checking policy, relaxing the value of timeout might help. > Unless fully understanding specifications about your device, you cannot > make firm statement about it, I think. The drivers installed in Windows > or OS X can set the synchronization when plugging-in. > You are correct! (See the item 8 below.) I did some experiment today, although I am not sure if it has any technical meaning. --------------- My PC runnging on Windows 7 --------------- 1. Run Control Panel, and set Sync Source to PC DAW. 2. Play music with Audacity -> O.K. -> Stop music. 3. Kill Control panel. 4. Set Orpheus to Stand-by Mode 5. Remove Firewire Cable 6. Recover from Stand-by Mode 7. Connect Firewire Cable 8. Run Control Panel, and Sync Source now reads Local, /*** this is not what I expected, but it is so. ***/ so set Sync Source to PC DAW again. 9. Play music with Audacity -> O.K. -> Keep playing 10. Kill Control Panel. -> Music still playing O.K. -> Stop Music 11. Set Orpheus to Stand-by Mode 12. Remove Firewire Cable --------------- Reboot My PC to start Linux --------------- 13. Recover from Stand-by Mode 14. Connect Firewire Cable 15. Enter the following CLIs. % dmesg [ 21.838971] r8169 0000:02:00.0 eth0: link down [ 24.281647] r8169 0000:02:00.0 eth0: link up [ 281.110728] irq_handler: 36 callbacks suppressed [ 281.110737] firewire_ohci 0000:08:00.0: isochronous cycle inconsistent [ 281.612586] firewire_core 0000:08:00.0: created device fw1: GUID 00119800010001f9, S400 [ 281.612747] firewire_core 0000:08:00.0: phy config: new root=ffc1, gap_count=5 % ls -l /proc/asound lrwxrwxrwx 1 root root 5 Apr 25 21:55 Orpheus -> card3 % cat /proc/asound/card3/firewire/clock Sampling rate: 44100 Clock Source: External (MSU-dest: 7) % aplay -D plughw:3,0 test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:1297: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (46439 46440) PERIOD_SIZE: 2048 PERIOD_BYTES: 8192 PERIODS: 4 BUFFER_TIME: (185759 185760) BUFFER_SIZE: 8192 BUFFER_BYTES: 32768 TICK_TIME: 0 % dmesg [ 21.838971] r8169 0000:02:00.0 eth0: link down [ 24.281647] r8169 0000:02:00.0 eth0: link up [ 281.110728] irq_handler: 36 callbacks suppressed [ 281.110737] firewire_ohci 0000:08:00.0: isochronous cycle inconsistent [ 281.612586] firewire_core 0000:08:00.0: created device fw1: GUID 00119800010001f9, S400 [ 281.612747] firewire_core 0000:08:00.0: phy config: new root=ffc1, gap_count=5 *** Regards, Mike Ood

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[alsa-devel] [RFC] dice: simplifying address registration
by Takashi Sakamoto 25 Apr '15

25 Apr '15

Clemens, Would I request your comments about an attached patch? Currently Dice driver keeps address ranges for each unit instance. But there's a way to reuse the same address range. I think this idea can simplify driver probe processing and save resources of host controller. Regards Takashi Sakamoto

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Re: [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 25 Apr '15

25 Apr '15

Takashi Sakamoto wrote: > > % cat /proc/asound/card2/firewire/clock > > Sampling rate: 44100 > > Clock Source: External (MSU-dest: 7) > > According to this output, the internal clock of your device is set to > synchronize to external source such as S/PDIF or word clock. Oops. Let me boot my Windows7 again, and see.. https://spinorlab.wordpress.com/2015/04/24/orpheus-and-its-sync-source/ (see orpheussyncsource.jpg) The control program by PrismSound tells that the Sync Source is Local, which means it is a Master and System is sync'ed to Orpheus's local clock. http://www.prismsound.com/music_recording/products_subs/orpheus/online_manu… (Contents -> Orpheus hardware -> Synchronization) (Contents -> Orpheus software -> Orpheus Control Panel) Funny? Now reboot again, and going back to Linux. ------------------ % dmesg <- Orpheus was powered on already. ------------------ [ 20.927594] r8169 0000:02:00.0 eth0: link down [ 23.299665] r8169 0000:02:00.0 eth0: link up [ 27.676032] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 29.974533] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 32.282923] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 34.961027] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 37.259624] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 47.732497] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 50.030918] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 52.339515] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 55.037594] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 57.316079] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 ------------------ % ls -l /proc/asound lrwxrwxrwx 1 root root 5 Apr 24 20:24 Orpheus -> card0 ------------------ % cat /proc/asound/card0/firewire/clock Sampling rate: 44100 Clock Source: External (MSU-dest: 7) <- as Takashi pointed out! ------------------ % aplay -D plughw:0,0 test.wav <- same happens as is expected ------------------ Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:1297: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (46439 46440) PERIOD_SIZE: 2048 PERIOD_BYTES: 8192 PERIODS: 4 BUFFER_TIME: (185759 185760) BUFFER_SIZE: 8192 BUFFER_BYTES: 32768 TICK_TIME: 0 ------------------ % dmesg <- the last line is added. ------------------ [ 20.927594] r8169 0000:02:00.0 eth0: link down [ 23.299665] r8169 0000:02:00.0 eth0: link up [ 27.676032] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 29.974533] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 32.282923] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 34.961027] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 37.259624] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 47.732497] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 50.030918] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 52.339515] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 55.037594] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 57.316079] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 1483.932836] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 *** > If your device is implemented with no customization of the > setting and you didn't give any external sources to your > device, the device is not synchronized. > (If you give it correctly, please inform it to us.) I am not sure, but I think I gave it correctly to Local Sync Source. > libffado also supports this device and I guess you can > start streaming with jackd and firewire backend Yes, as I have mentioned, I have been always successful with alsa (vanilla without snd-*) + libffado + jackd. > because libffado implements loose > checker of packet discontinuity. I do not understand techinical details, but the sound was perfect even with 192kHz/24bit audio using libffado. So if it's only a #define TIMER_SOMETHING value_anything type of things... > Currently, libffado has a bug to fail to switch clock > source for usual BeBoB driver. > So we have no way to set the source of clock correctly. > I've tried to fix it but get stuck deeply into > ffado's fault and given up. As for me, clock source settings, etc., which is written into its EEPROM when powered off, can be occasionally set via the control program by PrismSound either on Windows/OS X, only when it becomes necessary. So personally I really do not matter, if I understand the situation correctly. > For our information: > http://subversion.ffado.org/wiki/AvcModels/PrismSoundOrpheus I hope some day this level of information becomes usefull for me. Regards, Mike Ood

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Re: [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 25 Apr '15

25 Apr '15

Mike Ood wrote: > As for me, clock source settings, etc., > which is written into its EEPROM when powered off, > can be occasionally set via the control program > by PrismSound either on Windows/OS X, > only when it becomes necessary. > I think I was at least partially wrong. The manual says: http://www.prismsound.com/music_recording/products_subs/orpheus/online_manu… (Contents -> Orpheus hardware -> Stand-alone operation): --------------------------------- Stand-alone operation --------------------------------- It is possible to operate Orpheus without a FireWire connection to a host computer. This is done by setting up the unit as required using the Orpheus Control Panel applet whilst it is connected to a host computer via its FireWire interface, then placing the unit in standby by pressing the standby button, before disconnecting the unit from the host and power source. When the unit is re-powered, and detects that no FireWire connection is active, <========== I missed this part! it reloads the settings which were previously stored. As Takashi Sakamoto mentioned: > Currently, libffado has a bug to fail to switch clock > source for usual BeBoB driver. > So we have no way to set the source of clock correctly. > I've tried to fix it but get stuck deeply into > ffado's fault and given up. this could be a problem for me, too. Anyway, the functionality I expect snd-bebob to have is no more than what is offered by juju+libffado+jack. Mike Ood

1 0

[alsa-devel] [PATCH 1/3] ASoC: tas571x: Add DT binding document
by Kevin Cernekee 24 Apr '15

24 Apr '15

Document the bindings for the soon-to-be-added tas571x driver. Signed-off-by: Kevin Cernekee <cernekee(a)chromium.org> --- .../devicetree/bindings/sound/tas571x.txt | 34 ++++++++++++++++++++++ 1 file changed, 34 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas571x.txt diff --git a/Documentation/devicetree/bindings/sound/tas571x.txt b/Documentation/devicetree/bindings/sound/tas571x.txt new file mode 100644 index 000000000000..ac704c1f143d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas571x.txt @@ -0,0 +1,34 @@ +Texas Instruments TAS5711/TAS5717/TAS5719 stereo power amplifiers + +The codec is controlled through an I2C interface. It also has two other +signals that can be wired up to GPIOs: reset (strongly recommended), and +powerdown (optional). + +Required properties: + +- compatible: "ti,tas5711", "ti,tas5717", or "ti,tas5719" +- reg: The I2C address of the device +- #sound-dai-cells: must be equal to 0 + +Optional properties: + +- reset-gpios: GPIO specifier for the TAS571x's active low reset line +- pdn-gpios: GPIO specifier for the TAS571x's active low powerdown line +- clocks: clock phandle for the MCLK input +- clock-names: should be "mclk" +- VDD-supply: regulator phandle for the AVDD/DVDD/HP_VDD supply +- PVDD-supply: regulator phandle for the PVDD supply + +Example: + + tas5717: audio-codec@2a { + compatible = "ti,tas5717"; + reg = <0x2a>; + #sound-dai-cells = <0>; + + reset-gpios = <&gpio5 1 GPIO_ACTIVE_HIGH>; + pdn-gpios = <&gpio5 2 GPIO_ACTIVE_HIGH>; + + clocks = <&clk_core CLK_I2S>; + clock-names = "mclk"; + }; -- 2.2.0.rc0.207.ga3a616c

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[alsa-devel] [PATCH 00/21] HD-audio controller driver split
by Takashi Iwai 24 Apr '15

24 Apr '15

Hi, here is a patch series to split HD-audio controller driver code to snd-hda-core helper module. It's not targeted for 4.1 but rather for 4.2 together with ASoC HD-audio. There is no real changes from the functionality POV. It's mostly code splits and cleanups. The latest patches are found in topic/hda branch of sound git tree. Takashi === Jeeja KP (2): ALSA: hda - moved alloc/free stream pages function to controller library ALSA: hda - add generic functions to set hdac stream params Ramesh Babu (1): ALSA: hda - add ASoC device type for hda core Takashi Iwai (18): ALSA: hda - Handle error from get_response bus ops directly ALSA: hda - Add the controller helper codes to hda-core module ALSA: hda - Add DSP loader to core library code ALSA: hda - Merge codec and controller helpers ALSA: hda - Move send_cmd / get_response to hdac_bus_ops ALSA: hda - Pass bus io_ops directly from the top-level driver ALSA: hda - Migrate hdac_stream into legacy driver ALSA: hda - Migrate more hdac_stream codes ALSA: hda - Embed bus into controller object ALSA: hda - Minor refactoring ALSA: hda - Move PCM format and rate handling code to core library ALSA: hda - Add missing inclusion of <linux/clocksource.h> ALSA: hda - Reenable tracepoints for controller ALSA: hda/tegra - Fix build error and warning ALSA: hda - Drop azx_sd_read*/write*() macros ALSA: hda - Replace hda_bus_ops with static binding ALSA: hda - Move prepared flag into struct hdac_stream ALSA: hda - Replace open codes with snd_hdac_stream_set_params() include/sound/hda_register.h | 152 +++++ include/sound/hdaudio.h | 286 ++++++++- sound/hda/Kconfig | 3 + sound/hda/Makefile | 2 +- sound/hda/hdac_bus.c | 20 +- sound/hda/hdac_controller.c | 507 +++++++++++++++ sound/hda/hdac_device.c | 300 +++++++++ sound/hda/hdac_stream.c | 686 ++++++++++++++++++++ sound/pci/hda/Kconfig | 3 - sound/pci/hda/Makefile | 3 +- sound/pci/hda/hda_codec.c | 421 +------------ sound/pci/hda/hda_codec.h | 80 +-- sound/pci/hda/hda_controller.c | 1330 ++++++++------------------------------- sound/pci/hda/hda_controller.h | 270 +------- sound/pci/hda/hda_intel.c | 234 ++++--- sound/pci/hda/hda_intel.h | 1 + sound/pci/hda/hda_intel_trace.h | 4 +- sound/pci/hda/hda_tegra.c | 76 +-- sound/pci/hda/patch_ca0132.c | 7 +- 19 files changed, 2457 insertions(+), 1928 deletions(-) create mode 100644 include/sound/hda_register.h create mode 100644 sound/hda/hdac_controller.c create mode 100644 sound/hda/hdac_stream.c -- 2.3.5

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Re: [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 24 Apr '15

24 Apr '15

Clemens wrote: > Does this device (with this cable) work in Windows or OS X? Definitely. Let me now reset my PC so that it will boot from its Windows7 partion. With Audacity and a vendor supplied control software, my Orpheus plays 44.1kHz 2-ch sound perfectly. Also, >>>> I have been using my PrismSound Orpheus successfully with juju+ffado+jackd as: for more than five years, not with the same PC but with the same cable, during which time I have never experienced a trouble. And before doing clean install of Ubuntu 15.04 (kernel 3.16) a few days ago, I confirmed that: % jackd -R -dfirewire -r44100 -p4096 -n3 & % aplay -D pcm.jack test.wav or % alsaplayer -i text -o jack -d "system:playback_1,system:playback_2" test.wav works all right on Ubuntu 14.04 with kernel 3.13. So I have an impression that this is not a hardware but a software problem, perhaps specific to snd-bebob.. *** > At the moment, a missing packet results in the driver just stopping the > stream. There would be better ways to handle this, but I haven't yet > found the time to implement this. Any suggestions will be appreciated. Thank you again for taking your time. Regards, Mike Ood

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Re: [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 24 Apr '15

24 Apr '15

Clemens wrote: >> You should replace "hw" with "plughw" in the device name. So you were referring to: % aplay -L plughw:CARD=Orpheus,DEV=0 Orpheus, Orpheus PCM Hardware device with all software conversions > Any error message in the system log? % tail dmesg?.txt ==> dmesg1.txt <== this is just after booting. [ 21.439026] r8169 0000:02:00.0 eth0: link down [ 23.809013] r8169 0000:02:00.0 eth0: link up ==> dmesg2.txt <== this is just after connecting Orpheus. [ 21.439026] r8169 0000:02:00.0 eth0: link down [ 23.809013] r8169 0000:02:00.0 eth0: link up [ 110.079264] firewire_core 0000:08:00.0: phy config: new root=ffc1, gap_count=5 [ 110.343319] firewire_core 0000:08:00.0: phy config: new root=ffc1, gap_count=5 [ 110.580310] firewire_core 0000:08:00.0: created device fw1: GUID 00119800010001f9, S400 [ 113.660273] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 ==> dmesg3.txt <== this is just after % aplay -D plughw:2,0 test.wav [ 21.439026] r8169 0000:02:00.0 eth0: link down [ 23.809013] r8169 0000:02:00.0 eth0: link up [ 110.079264] firewire_core 0000:08:00.0: phy config: new root=ffc1, gap_count=5 [ 110.343319] firewire_core 0000:08:00.0: phy config: new root=ffc1, gap_count=5 [ 110.580310] firewire_core 0000:08:00.0: created device fw1: GUID 00119800010001f9, S400 [ 113.660273] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 128.010593] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 133.416998] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 [ 136.275142] snd-bebob fw1.0: Detect discontinuity of CIP: 50 70 [ 137.654178] snd-bebob fw1.0: Detect discontinuity of CIP: 70 90 % ls -l dmesg?.txt -rw-rw-r-- 1 user1 user1 72728 Apr 24 08:42 dmesg1.txt -rw-rw-r-- 1 user1 user1 73050 Apr 24 08:42 dmesg2.txt -rw-rw-r-- 1 user1 user1 73318 Apr 24 08:43 dmesg3.txt % egrep 'Apr 24 08:42|Apr 24 08:43' /var/log/syslog Apr 24 08:42:24 host gnome-session[1466]: WARN 2015-04-24 08:42:24 unity.dash.gsettingsscopereader GSettingsScopes.cpp:108 Error fetching protocol metadata for scope: social.scope : Valid key file could not be found in search dirs Apr 24 08:42:44 host kernel: [ 110.079264] firewire_core 0000:08:00.0: phy config: new root=ffc1, gap_count=5 Apr 24 08:42:45 host kernel: [ 110.343319] firewire_core 0000:08:00.0: phy config: new root=ffc1, gap_count=5 Apr 24 08:42:45 host kernel: [ 110.580310] firewire_core 0000:08:00.0: created device fw1: GUID 00119800010001f9, S400 Apr 24 08:42:48 host kernel: [ 113.660273] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 Apr 24 08:43:02 host kernel: [ 128.010593] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 Apr 24 08:43:08 host kernel: [ 133.416998] snd-bebob fw1.0: Detect discontinuity of CIP: 00 08 Apr 24 08:43:11 host rtkit-daemon[1163]: Supervising 4 threads of 1 processes of 1 users. Apr 24 08:43:11 host rtkit-daemon[1163]: Successfully made thread 1900 of process 1540 (n/a) owned by '1000' RT at priority 5. Apr 24 08:43:11 host rtkit-daemon[1163]: Supervising 5 threads of 1 processes of 1 users. Apr 24 08:43:11 host kernel: [ 136.275142] snd-bebob fw1.0: Detect discontinuity of CIP: 50 70 Apr 24 08:43:12 host kernel: [ 137.654178] snd-bebob fw1.0: Detect discontinuity of CIP: 70 90 Apr 24 08:43:14 host pulseaudio[1540]: [alsa-source-BeBoB] alsa-util.c: Could not recover from POLLERR|POLLNVAL|POLLHUP and XRUN: Connection timed out Apr 24 08:43:17 host systemd[1]: Stopping User Manager for UID 112... *** Any clues? Mike Ood -- % uname -a Linux host 3.19.0-15-generic #15-Ubuntu SMP Thu Apr 16 23:32:37 UTC 2015 x86_64 x86_64 x86_64 GNU/Linux --

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[alsa-devel] [PATCH] ASoC: rt5677: add register patch for PLL
by Bard Liao 24 Apr '15

24 Apr '15

The PLL output will be unstable in some cases. We can fix it by setting some registers. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5677.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index af18258..9c07a0c 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -62,6 +62,9 @@ static const struct reg_default init_list[] = { {RT5677_PR_BASE + 0x1e, 0x0000}, {RT5677_PR_BASE + 0x12, 0x0eaa}, {RT5677_PR_BASE + 0x14, 0x018a}, + {RT5677_PR_BASE + 0x15, 0x0490}, + {RT5677_PR_BASE + 0x38, 0x0f71}, + {RT5677_PR_BASE + 0x39, 0x0f71}, }; #define RT5677_INIT_REG_LEN ARRAY_SIZE(init_list) -- 1.8.1.1.439.g50a6b54

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Re: [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 24 Apr '15

24 Apr '15

I am now trying with my capture program, "myprog.cpp" instead of "aplay.c". ------------------------------- % alsacap ; alscap -R ------------------------------- Card 2, ID `Orpheus', name `Orpheus' Device 0, ID `BeBoB', name `Orpheus PCM', 1 subdevices (1 available) 12 channels, sampling rate 44100..44100 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' Card 2, ID `Orpheus', name `Orpheus' Device 0, ID `BeBoB', name `Orpheus PCM', 1 subdevices (1 available) 10 channels, sampling rate 44100..44100 Hz Sample formats: S32_LE Subdevice 0, name `subdevice #0' ------------------------------- % myprog hw:2,0 44100 10 ------------------------------- Sound::Sound: sound_device = hw:2,0, rate = 44100, channels = 10 Sound::asound_set_hwparams: begin... Sound::asound_set_hwparams: buffer_time = 92879, dir = 1 Sound::asound_set_hwparams: buffer_size = 4096 Sound::asound_set_hwparams: period_time = 46439 , dir = 1 Sound::asound_set_hwparams: period_size = 2048, dir = 0 Sound::asound_set_hwparams: Unable to set hw params: Connection timed out Sound::Sound: setting of hwparams failed. Connection timed out ------------------------------- Sound::asound_set_hwparams(snd_pcm_t *, snd_pcm_hw_params_t *) ------------------------------- err = snd_pcm_hw_params_any(handle, params); err = snd_pcm_hw_params_set_rate_resample(handle, params, resample); err = snd_pcm_hw_params_set_access(handle, params, err = snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S32_LE); //if Orphesu err = snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16); //else err = snd_pcm_hw_params_set_channels(handle, params, channels); err = snd_pcm_hw_params_set_rate_near(handle, params, &rrate, 0); err = snd_pcm_hw_params_set_buffer_time_near(handle, params, &buffer_time, err = snd_pcm_hw_params_get_buffer_size(params, &size); err = snd_pcm_hw_params_set_period_time_near(handle, params, &period_time, err = snd_pcm_hw_params_get_period_size(params, &size, &dir); err = snd_pcm_hw_params(handle, params); if (err < 0) { cout << "Sound::asound_set_hwparams: Unable to set hw params: " << snd_strerror(err) << endl; return err; } ------------------------------- No messages were found in % demsg and % cat /var/log/syslog concerning the above run of % myprog. So is it (partly) due to "Connection timed out"? Mike Ood

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[alsa-devel] [PATCH] ALSA: hda - fix headset mic detection problem for one more machine
by Hui Wang 24 Apr '15

24 Apr '15

We have two machines with alc256 codec in the pin quirk table, so moving the common pins to ALC256_STANDARD_PINS. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1447909 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 24 +++++++++++++++--------- 1 file changed, 15 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 03975d03..2a72da7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5228,6 +5228,16 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {0x1b, 0x411111f0}, \ {0x1e, 0x411111f0} +#define ALC256_STANDARD_PINS \ + {0x12, 0x90a60140}, \ + {0x14, 0x90170110}, \ + {0x19, 0x411111f0}, \ + {0x1a, 0x411111f0}, \ + {0x1b, 0x411111f0}, \ + {0x1d, 0x40700001}, \ + {0x1e, 0x411111f0}, \ + {0x21, 0x02211020} + #define ALC282_STANDARD_PINS \ {0x14, 0x90170110}, \ {0x18, 0x411111f0}, \ @@ -5328,15 +5338,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1d, 0x40700001}, {0x21, 0x02211050}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0x90a60140}, - {0x13, 0x40000000}, - {0x14, 0x90170110}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, - {0x1b, 0x411111f0}, - {0x1d, 0x40700001}, - {0x1e, 0x411111f0}, - {0x21, 0x02211020}), + ALC256_STANDARD_PINS, + {0x13, 0x40000000}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC256_STANDARD_PINS, + {0x13, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, {0x13, 0x40000000}, -- 1.9.1

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[alsa-devel] [ANNOUNCE] Audio Mini Summit 2014, Oct. 14, Düsseldorf, Germany
by Takashi Iwai 24 Apr '15

24 Apr '15

Hi, like previous years, we're going to hold a meeting to discuss lowlevel audio on Linux. This time it's along with Linux Plumbers conference, but for avoiding the conflicts with other LPC mini conferences, it'll be held on Tuesday October 14, at the same venue as LPC, Congress Center Düsseldorf, Germany. If anyone is interested, feel free to sign up in the attendee list below. It's a public accessible Google doc, so you can edit freely. http://goo.gl/VbXLPW (or https://docs.google.com/spreadsheets/d/1wBAiAyZSnMjVFJbxdXE5JoVdKGIBIWt-4tu…) The topics that have been currently raised are found in the URL below. If you have any more topics to discuss there, just put your favorite one in the URL below as well. http://goo.gl/JSasQ9 (or https://docs.google.com/spreadsheets/d/1cXzON65QYNrTXi31AIQHffUZ9NWwJuNEJAC…) Thanks! Takashi

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[alsa-devel] [RFC v2] ASoC: Topology File format
by Liam Girdwood 24 Apr '15

24 Apr '15

Here's V2 of the file format the userspace topology tool will read to generate the binary output file. The output file uses the topology UAPI structures to represent the objects in the text file. This is still WiP and is based on the UCM file format from Jaroslav. Changes from V1. o Added BE's and Codec <-> codec links o Added enumerated and binary controls o Added private data o Moved object names from name field into object top level (like section names). o Control ops binding changed to string+number # Firmware Topology Configuration for Broadwell # TLV scale used by both global and stream volumes SectionTLV."hsw_vol_tlv" { DBScale [ min "-9000" step "300" mute "1" ] } # Values for Equalizer 1 SectionText."EQU1" { Values [ "100Hz", "200Hz", "1kHz" ] } # Private data for Equalizer 1 SectionData."EQU1" { file "/path/to/file" }; # Values for Input Muxes 1,2,3 SectionText."In1" { Values [ "Line", "Mic", ] } # Controls from default pipeline SectionControl."Default Pipeline" { # All controls in this section belong to this Index group/pipeline Index "1" # Master DSP volume Mixer."Master Playback Volume" [ # Channel register and shift for Front Left/Right Channel."FL" [ reg "0" shift "0" ] Channel."FR" [ reg "0" shift "8" ] # max control value and whether value is inverted max "31" invert "false" # control uses bespoke driver get/put/info ID 0 ops "DRV0" # uses TLV data above tlv_array "hsw_vol_tlv" ] # Offload 0 volume Mixer."Media0 Playback Volume" [ # Channel register and shift for Front Left/Right Channel."FL" [ reg "1" shift "0" ] Channel."FR" [ reg "1" shift "8" ] # max control value and whether value is inverted max "31" invert "false" # control uses bespoke driver get/put/info ID 0 ops "DRV0" # uses TLV data above tlv_array "hsw_vol_tlv" ] # Enum controls Enum."Equ1" [ # Texts for enum texts "EQU1" # Channel register and shift for Front Left/Right Channel."FL" [ reg "2" shift "0" ] Channel."FR" [ reg "1" shift "8" ] # control uses bespoke driver get/put/info ID 1 ops "DRV1" # Private data for this control data "EQU1" ] Enum."Input Select" [ # Texts for enum texts "In1" # Channel register and shift for Front Left/Right Channel."FL" [ reg "2" shift "0" ] Channel."FR" [ reg "1" shift "8" ] # control uses bespoke driver get/put/info ID 1 ops "DRV2" ] # Byte controls Bytes [ base "0" num_regs "16" mask "0xff" ] } # DAPM Widgets for main pipeline SectionWidget."Pipeline1" { # All controls in this section belong to this Index group/pipeline Index "1" # SSP AIF input AIF_IN."SSP0 CODEC IN" [ no_pm "true" ] # SSP AIF output AIF_OUT."SSP0 CODEC OUT" [ no_pm "true" ] # Virtual playback mixer MIXER."Playback VMixer" [ no_pm "true" ] # Test input mux MUX."Capture Select" [ no_pm "true" enum "Input Select" ] } # DAPM Routes SectionGraph."Pipeline 1 Graph" { Index "1" Routes [ "Playback VMixer, , System Playback" "Playback VMixer, , Offload0 Playback" "Playback VMixer, , Offload1 Playback" "SSP0 CODEC OUT, , Playback VMixer" "Analog Capture, , SSP0 CODEC IN" ] } # PCM Configurations supported by FW SectionPCMConfig."PCM 48k Stereo 24bit" { Playback [ format "S24_LE" rate "48000" channels "2" tdm_slot "0xf" ] Capture [ format "S24_LE" rate "48000" channels "2" tdm_slot "0xf" ] } SectionPCMConfig."PCM 48k Stereo 16bit" { Playback [ format "S16_LE" rate "48000" channels "2" tdm_slot "0xf" ] Capture [ format "S16_LE" rate "48000" channels "2" tdm_slot "0xf" ] } SectionPCMConfig."PCM 48k 2P/4C 16bit" { Playback [ format "S16_LE" rate "48000" channels "2" tdm_slot "0xf" ] Capture [ format "S16_LE" rate "48000" channels "4" tdm_slot "0xf" ] } SectionPCMConfig."PCM 8k Mono 16bit" { Playback [ format "S16_LE" rate "8000" channels "1" ] Capture [ format "S16_LE" rate "8000" channels "1" ] } # PCM capabilities supported by FW SectionPCMCapabilities."System Playback" { Capabilities [ formats "S24_LE, S16_LE" rate_min "48000" rate_max "48000" channels_min "2" channels_max "2" ] } SectionPCMCapabilities."Analog Capture" { Capabilities [ formats "S24_LE, S16_LE" rate_min "48000" rate_max "48000" channels_min "2" channels_max "4" ] } SectionPCMCapabilities."Offload Pin Playback" { Capabilities [ formats "S24_LE, S16_LE" rate_min "8000" rate_max "192000" channels_min "2" channels_max "2" ] } SectionPCMCapabilities."Loopback Capture" { Capabilities [ formats "S16_LE" rate_min "48000" rate_max "48000" channels_min "2" channels_max "2" ] } SectionPCMCapabilities."BT" { Capabilities [ formats "S16_LE" rate_min "8000" rate_max "8000" channels_min "1" channels_max "1" ] } # PCM devices exported by Firmware SectionPCM."System Pin" { Index "1" # used for binding to the PCM ID "0" Playback [ Capabilities "System Playback" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" ] Capture [ Capabilities "Analog Capture" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" Config "PCM 48k 2P/4C 16bit" ] } SectionPCM."Offload Pin 0" { Index "1" # used for binding to the PCM ID "1" Playback [ Capabilities "Offload Pin Playback" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" ] } SectionPCM."Offload Pin 1" { Index "1" # used for binding to the PCM ID "2" Playback [ Capabilities "Offload Pin Playback" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" ] } SectionPCM."Loopback Pin" { Index "1" # used for binding to the PCM ID "3" Capture [ Capabilities "Loopback Capture" Config "PCM 48k Stereo 16bit" ] } # Backend DAI links SectionBE."SSP0-Codec" { Index "1" # used for binding to the BE DAI link ID "0" # BE DAI link capabilities and supported configs Playback [ Capabilities "System Playback" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" ] Capture [ Capabilities "Analog Capture" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" Config "PCM 48k 2P/4C 16bit" ] } SectionBE."SSP1-BT" { Index "1" # used for binding to the BE DAI link ID "1" # BE DAI link capabilities and supported configs Playback [ Capabilities "BT" Config "PCM 8k Mono 16bit" ] Capture [ Capabilities "BT" Config "PCM 8k Mono 16bit" ] } # Codec <-> Codec style links SectionCC."FM-Codec" { Index "1" # used for binding to the CC link ID "0" # CC DAI link capabilities and supported configs Playback [ Capabilities "System Playback" Config "PCM 48k Stereo 16bit" ] Capture [ Capabilities "Analog Capture" Config "PCM 48k Stereo 16bit" ] }

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[alsa-devel] [PATCH 0/8] Proc file cleanups
by Takashi Iwai 23 Apr '15

23 Apr '15

Hi, here is a series of patches for cleaning up the ALSA proc file management code. The biggest change is the use of seq_file for text reads, but all are internal and the API is kept intact. The rest are mostly small cleanups and refactoring. The patches are found in topic/proc branch of sound git tree. Takashi === Takashi Iwai (8): ALSA: core: Use seq_file for text proc file reads ALSA: core: Fix possible memory leaks at error path in info.c ALSA: core: Remove child proc file elements recursively ALSA: core: Manage asound root directory with snd_info_entry ALSA: core: Remove superfluous exit calls for proc entries ALSA: core: Don't ignore errors at creating proc files ALSA: core: Build conditionally and remove superfluous ifdefs ALSA: core: Clean up OSS proc file management include/sound/core.h | 4 - include/sound/info.h | 29 +- sound/core/Makefile | 8 +- sound/core/info.c | 779 ++++++++++++++++++---------------------------- sound/core/info_oss.c | 29 +- sound/core/init.c | 33 +- sound/core/seq/Makefile | 3 +- sound/core/seq/seq_info.c | 19 +- sound/core/sound.c | 24 +- sound/core/sound_oss.c | 30 +- 10 files changed, 373 insertions(+), 585 deletions(-) -- 2.3.5

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Re: [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 23 Apr '15

23 Apr '15

Clemens wrote: > You should not write an .asoundrc. > You should replace "hw" with "plughw" in the device name. Somewhat different messages, but still aborts immediately. (There is a relay click sound from Orpheus, as it always does before playing music, so it is getting slightly better..) % ls -l .asound* -rw-rw-r-- 1 user1 user1 65 Apr 23 21:48 .asoundrc-nogo % grep asound /usr/share/alsa/alsa.conf "/etc/asound.conf" "~/.asoundrc" % sudo ls -l /etc/asound.conf ls: cannot access /etc/asound.conf: No such file or directory % aplay -D plughw:2,0 test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:1297: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (46439 46440) PERIOD_SIZE: 2048 PERIOD_BYTES: 8192 PERIODS: 4 BUFFER_TIME: (185759 185760) BUFFER_SIZE: 8192 BUFFER_BYTES: 32768 TICK_TIME: 0 % aplay -D plughw:2 test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:1297: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (46439 46440) PERIOD_SIZE: 2048 PERIOD_BYTES: 8192 PERIODS: 4 BUFFER_TIME: (185759 185760) BUFFER_SIZE: 8192 BUFFER_BYTES: 32768 TICK_TIME: 0 % aplay -D default:2,0 test.wav ALSA lib conf.c:4705:(snd_config_expand) Unknown parameters 2,0 ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM default:2,0 aplay: main:722: audio open error: Invalid argument % aplay -D default:2 test.wav ALSA lib conf.c:4705:(snd_config_expand) Unknown parameters 2 ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM default:2 aplay: main:722: audio open error: Invalid argument Regards, Mike Ood

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[alsa-devel] alsa-driver-build on kernel 3.13
by Raymond Yau 23 Apr '15

23 Apr '15

Do alsa-driver-build still support on kernel 3.13 ? Some of the modules have been built but not installed on Ubuntu 14.04 LTS [ 7.547726] snd_hda_codec: Unknown symbol snd_hdac_get_connections (err 0) [ 7.547733] snd_hda_codec: Unknown symbol snd_hdac_power_up_pm (err 0) [ 7.547749] snd_hda_codec: Unknown symbol snd_hdac_regmap_read_raw (err 0) [ 7.547762] snd_hda_codec: Unknown symbol snd_hdac_power_down (err 0) [ 7.547768] snd_hda_codec: Unknown symbol snd_hdac_make_cmd (err 0) [ 7.547783] snd_hda_codec: Unknown symbol snd_hdac_regmap_init (err 0) [ 7.547797] snd_hda_codec: Unknown symbol snd_hdac_refresh_widgets (err 0) [ 7.547824] snd_hda_codec: Unknown symbol snd_hdac_bus_exit (err 0) [ 7.547832] snd_hda_codec: Unknown symbol _snd_hdac_read_parm (err 0) [ 7.547845] snd_hda_codec: Unknown symbol snd_hdac_bus_exec_verb_unlocked (err 0) [ 7.547849] snd_hda_codec: Unknown symbol snd_hdac_override_parm (err 0) [ 7.547857] snd_hda_codec: Unknown symbol snd_hdac_device_register (err 0) [ 7.547862] snd_hda_codec: Unknown symbol snd_hdac_regmap_update_raw (err 0) [ 7.547878] snd_hda_codec: Unknown symbol snd_hdac_bus_init (err 0) [ 7.547886] snd_hda_codec: Unknown symbol snd_hdac_read_parm_uncached (err 0) [ 7.547892] snd_hda_codec: Unknown symbol snd_hdac_device_unregister (err 0) [ 7.547896] snd_hda_codec: Unknown symbol snd_hdac_device_init (err 0) [ 7.547908] snd_hda_codec: Unknown symbol snd_hdac_power_down_pm (err 0) [ 7.547924] snd_hda_codec: Unknown symbol snd_hdac_regmap_write_raw (err 0) [ 7.547933] snd_hda_codec: Unknown symbol snd_hdac_regmap_exit (err 0) [ 7.547943] snd_hda_codec: Unknown symbol snd_hda_bus_type (err 0) [ 7.547949] snd_hda_codec: Unknown symbol snd_hdac_device_exit (err 0) [ 7.547956] snd_hda_codec: Unknown symbol snd_hdac_get_sub_nodes (err 0) [ 7.547964] snd_hda_codec: Unknown symbol snd_hdac_exec_verb (err 0) [ 7.547973] snd_hda_codec: Unknown symbol snd_array_new (err 0) [ 7.547981] snd_hda_codec: Unknown symbol snd_array_free (err 0) [ 7.547994] snd_hda_codec: Unknown symbol snd_hdac_power_up (err 0) [ 7.548197] snd_hda_codec: Unknown symbol snd_hdac_get_connections (err 0) [ 7.548202] snd_hda_codec: Unknown symbol snd_hdac_power_up_pm (err 0) [ 7.548214] snd_hda_codec: Unknown symbol snd_hdac_regmap_read_raw (err 0) [ 7.548225] snd_hda_codec: Unknown symbol snd_hdac_power_down (err 0) [ 7.548231] snd_hda_codec: Unknown symbol snd_hdac_make_cmd (err 0) [ 7.548244] snd_hda_codec: Unknown symbol snd_hdac_regmap_init (err 0) [ 7.548256] snd_hda_codec: Unknown symbol snd_hdac_refresh_widgets (err 0) [ 7.548281] snd_hda_codec: Unknown symbol snd_hdac_bus_exit (err 0) [ 7.548287] snd_hda_codec: Unknown symbol _snd_hdac_read_parm (err 0) [ 7.548298] snd_hda_codec: Unknown symbol snd_hdac_bus_exec_verb_unlocked (err 0) [ 7.548302] snd_hda_codec: Unknown symbol snd_hdac_override_parm (err 0) [ 7.548310] snd_hda_codec: Unknown symbol snd_hdac_device_register (err 0) [ 7.548315] snd_hda_codec: Unknown symbol snd_hdac_regmap_update_raw (err 0) [ 7.548330] snd_hda_codec: Unknown symbol snd_hdac_bus_init (err 0) [ 7.548338] snd_hda_codec: Unknown symbol snd_hdac_read_parm_uncached (err 0) [ 7.548343] snd_hda_codec: Unknown symbol snd_hdac_device_unregister (err 0) [ 7.548347] snd_hda_codec: Unknown symbol snd_hdac_device_init (err 0) [ 7.548357] snd_hda_codec: Unknown symbol snd_hdac_power_down_pm (err 0) [ 7.548372] snd_hda_codec: Unknown symbol snd_hdac_regmap_write_raw (err 0) [ 7.548380] snd_hda_codec: Unknown symbol snd_hdac_regmap_exit (err 0) [ 7.548389] snd_hda_codec: Unknown symbol snd_hda_bus_type (err 0) [ 7.548395] snd_hda_codec: Unknown symbol snd_hdac_device_exit (err 0) [ 7.548402] snd_hda_codec: Unknown symbol snd_hdac_get_sub_nodes (err 0) [ 7.548410] snd_hda_codec: Unknown symbol snd_hdac_exec_verb (err 0) [ 7.548418] snd_hda_codec: Unknown symbol snd_array_new (err 0) [ 7.548426] snd_hda_codec: Unknown symbol snd_array_free (err 0) [ 7.548438] snd_hda_codec: Unknown symbol snd_hdac_power_up (err 0)

2 1

Re: [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 23 Apr '15

23 Apr '15

Clemens wrote: > You are trying to play a two-channel file ... > ... on a device that supports only ten channels. This is what I was vaguely afraid of... > An with "hw", you have disabled any automatic format conversion. > Try "plughw:2" or "default:2". Could you be kind enough to suggest how I should write .asoundrc? I only know the obvious one for jack: .asoundrc ---------------- pcm.jackplug { type plug slave { pcm "jack" } } pcm.jack { type jack playback_ports { 0 alsa_pcm:playback_1 1 alsa_pcm:playback_2 } capture_ports { 0 alsa_pcm:capture_1 1 alsa_pcm:capture_2 } } ---------------- Regards, Mike Ood

2 1

[alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 23 Apr '15

23 Apr '15

Hi All, I have been using my PrismSound Orpheus successfully with juju+ffado+jackd as: % jackd -R -dfirewire -r44100 -p4096 -n3 & % aplay -D pcm.jack test.wav or % alsaplayer -i text -o jack -d "system:playback_1,system:playback_2" test.wav on Ubuntu 14.04 with kernel 3.13 or so. *** Today I clean installed ubuntu 14.10 wishing to use snd-bebob, but unfortunately % sudo aplay -D hw:2,0 --channels=2 test.wav does not work with the following messages: Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:1239: Channels count non available aplay.c: 1237 err = snd_pcm_hw_params_set_channels(handle, params, hwparams.channels); 1238 if (err < 0) { 1239 error(_("Channels count non available")); 1240 prg_exit(EXIT_FAILURE); 1241 } My short capture program, which works fine with USB audio devices, does not work either at the same place. myprog.cpp: /* set the count of channels */ err = snd_pcm_hw_params_set_channels(handle, params, channels); if (err < 0) { cout << "Sound::asound_set_hwparams: Channels count " << channels << " is not available for playbacks." << snd_strerror(err) << endl; return err; } Does anybody have any suggestions for me? Mike ODD *** (for your information) *** % uname -a Linux host 3.16.0-34-generic #47-Ubuntu SMP Fri Apr 10 18:02:58 UTC 2015 x86_64 x86_64 x86_64 GNU/Linux % ls -l /dev/fw* crw------- 1 root root 250, 0 Apr 23 13:57 /dev/fw0 crw-rw----+ 1 root video 250, 1 Apr 23 13:57 /dev/fw1 % lsmod | grep snd_bebob snd_bebob 40777 0 snd_firewire_lib 29081 1 snd_bebob snd_hwdep 17698 3 snd_usb_audio,snd_bebob,snd_hda_codec snd_pcm 104102 8 snd_firewire_lib,snd_usb_audio,snd_bebob,snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel,snd_hda_controller snd_rawmidi 30876 4 snd_firewire_lib,snd_bebob,snd_usbmidi_lib,snd_seq_midi snd 87611 31 snd_hda_codec_realtek,snd_usb_audio,snd_bebob,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_pcm,snd_seq,snd_rawmidi,snd_hda_codec_generic,snd_usbmidi_lib,snd_hda_codec,snd_hda_intel,snd_seq_device firewire_core 68671 3 snd_firewire_lib,snd_bebob,firewire_ohci % ls -l /proc/asound total 0 dr-xr-xr-x 6 root root 0 Apr 23 21:02 card0 dr-xr-xr-x 3 root root 0 Apr 23 21:02 card1 dr-xr-xr-x 5 root root 0 Apr 23 21:02 card2 dr-xr-xr-x 4 root root 0 Apr 23 21:02 card3 -r--r--r-- 1 root root 0 Apr 23 21:02 cards lrwxrwxrwx 1 root root 5 Apr 23 21:02 CODEC -> card3 -r--r--r-- 1 root root 0 Apr 23 21:02 devices lrwxrwxrwx 1 root root 5 Apr 23 21:02 HDMI -> card1 -r--r--r-- 1 root root 0 Apr 23 21:02 hwdep lrwxrwxrwx 1 root root 5 Apr 23 21:02 MID -> card0 -r--r--r-- 1 root root 0 Apr 23 21:02 modules lrwxrwxrwx 1 root root 5 Apr 23 21:02 Orpheus -> card2 dr-xr-xr-x 2 root root 0 Apr 23 21:02 oss -r--r--r-- 1 root root 0 Apr 23 21:02 pcm dr-xr-xr-x 2 root root 0 Apr 23 21:02 seq -r--r--r-- 1 root root 0 Apr 23 21:02 timers -r--r--r-- 1 root root 0 Apr 23 21:02 version % ls -l /proc/asound/card2 total 0 dr-xr-xr-x 2 root root 0 Apr 23 21:02 firewire -r--r--r-- 1 root root 0 Apr 23 21:02 id -r--r--r-- 1 root root 0 Apr 23 21:02 midi0 dr-xr-xr-x 3 root root 0 Apr 23 21:02 pcm0c dr-xr-xr-x 3 root root 0 Apr 23 21:02 pcm0p % cat /proc/asound/card2/id Orpheus % ls -l /proc/asound/card2/firewire total 0 -r--r--r-- 1 root root 0 Apr 23 21:03 clock -r--r--r-- 1 root root 0 Apr 23 21:03 firmware -r--r--r-- 1 root root 0 Apr 23 21:03 formation % cat /proc/asound/card2/firewire/clock Sampling rate: 44100 Clock Source: External (MSU-dest: 7) % cat /proc/asound/card2/firewire/firmware Manufacturer: bridgeCo Protocol Ver: 3 Build Ver: 0 GUID: 0x00119800010001F9 Model ID: 0x13 Model Rev: 0 Firmware Date: 20100205 Firmware Time: 105129 Firmware ID: 0x10048 Firmware Ver: 6572 Base Addr: 0x400C0080 Max Size: 1519984 Loader Date: 20070118 Loader Time: 101440 % cat /proc/asound/card2/firewire/formation Output Stream from device: Rate PCM MIDI 32000 0 0 44100 10 1 48000 10 1 88200 10 1 96000 10 1 176400 10 1 192000 10 1 Input Stream to device: Rate PCM MIDI 32000 0 0 44100 12 1 48000 12 1 88200 12 1 96000 12 1 176400 12 1 192000 12 1 % ls -l /proc/asound/card2/pcm0c total 0 -r--r--r-- 1 root root 0 Apr 23 21:05 info dr-xr-xr-x 2 root root 0 Apr 23 21:05 sub0 % ls -l /proc/asound/card2/pcm0p total 0 -r--r--r-- 1 root root 0 Apr 23 21:06 info dr-xr-xr-x 2 root root 0 Apr 23 21:06 sub0 % cat /proc/asound/card2/pcm0c/info card: 2 device: 0 subdevice: 0 stream: CAPTURE id: BeBoB name: Orpheus PCM subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 % cat /proc/asound/card2/pcm0p/info card: 2 device: 0 subdevice: 0 stream: PLAYBACK id: BeBoB name: Orpheus PCM subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 *** (end of information) ***

2 1

[alsa-devel] [PATCH] ASoC: Intel: fix the makefile for atom code
by Vinod Koul 23 Apr '15

23 Apr '15

The tom code should be using SND_SST_MFLD_PLATFORM and not the baytrail one. So fix it now Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index cd9aee9871a3..3853ec2ddbc7 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -4,7 +4,7 @@ obj-$(CONFIG_SND_SOC_INTEL_SST) += common/ # Platform Support obj-$(CONFIG_SND_SOC_INTEL_HASWELL) += haswell/ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += baytrail/ -obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += atom/ +obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += atom/ # Machine support obj-$(CONFIG_SND_SOC_INTEL_SST) += boards/ -- 1.7.9.5

2 1

[alsa-devel] [RFC] File format for topology data
by Liam Girdwood 23 Apr '15

23 Apr '15

I've pasted below the topology data for the broadwell firmware using the text file format being used by UCM. Using the UCM format allows to to share parser code and allows us to also have UCM define topology data. I've not yet completed the byte or enum controls but the rest of the data is there for comments. I've also included the recent channel changes suggested in the UAPI header comments. # Firmware Topology Configuration for Broadwell # TLV scale used by both global and stream volumes SectionTLV."hsw_vol_tlv" { DBScale [ min "-9000" step "300" mute "1" ] } # Controls from default pipeline SectionControl."Default Pipeline" { Index "1" # Master DSP volume Mixer [ name "Master Playback Volume" Channel.0 [ map "Left" reg "0" shift "0" ] Channel.1 [ map "Right" reg "0" shift "8" ] max "31" invert "false" get "0" put "0" tlv_array "hsw_vol_tlv" ] # Offload 0 volume Mixer [ name "Media0 Playback Volume" Channel.0 [ map "Left" reg "1" shift "0" ] Channel.1 [ map "Right" reg "1" shift "8" ] max "31" invert "false" get "1" put "1" tlv_array "hsw_vol_tlv" ] # Enum controls Enum [ ... ] # Byte controls Bytes [ ... ] } # DAPM Widgets for main pipeline SectionWidget."Pipeline1" { Index "1" AIF_IN [ name "SSP0 CODEC IN" stream_name "" slot "" reg "0" shift "0" invert "false" ] AIF_OUT [ name "SSP0 CODEC OUT" stream_name "" slot "" reg "0" shift "0" invert "false" ] ... } # DAPM Routes SectionGraph."Pipeline 1 Graph" { Index 1 Routes [ "Playback VMixer, , System Playback" "Playback VMixer, , Offload0 Playback" "Playback VMixer, , Offload1 Playback" "SSP0 CODEC OUT, , Playback VMixer" "Analog Capture, , SSP0 CODEC IN" ] } # PCM Configurations supported by FW SectionPCMConfig."PCM 48k Stereo 24bit" { Playback [ format "S24_LE" rate "48000" channels "2" tdm_slot "0xf" ] Capture [ format "S24_LE" rate "48000" channels "2" tdm_slot "0xf" ] } SectionPCMConfig."PCM 48k Stereo 16bit" { Playback [ format "S16_LE" rate "48000" channels "2" tdm_slot "0xf" ] Capture [ format "S16_LE" rate "48000" channels "2" tdm_slot "0xf" ] } SectionPCMConfig."PCM 48k 2P/4C 16bit" { Playback [ format "S16_LE" rate "48000" channels "2" tdm_slot "0xf" ] Capture [ format "S16_LE" rate "48000" channels "4" tdm_slot "0xf" ] } # PCM capabilities supported by FW SectionPCMCapabilities."System Playback" { Capabilities [ formats "S24_LE, S16_LE" rate_min "48000" rate_max "48000" channels_min "2" channels_max "2" ] } SectionPCMCapabilities."Analog Capture" { Capabilities [ formats "S24_LE, S16_LE" rate_min "48000" rate_max "48000" channels_min "2" channels_max "4" ] } SectionPCMCapabilities."Offload Pin Playback" { Capabilities [ formats "S24_LE, S16_LE" rate_min "8000" rate_max "192000" channels_min "2" channels_max "2" ] } SectionPCMCapabilities."Loopback Capture" { Capabilities [ formats "S16_LE" rate_min "48000" rate_max "48000" channels_min "2" channels_max "2" ] } # PCM devices exported by Firmware SectionPCM."System Pin" { Index 1 Playback [ Capabilities "System Playback" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" ] Capture [ Capabilities "Analog Capture" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" Config "PCM 48k 2P/4C 16bit" ] } SectionPCM."Offload Pin 0" { Index 1 Playback [ Capabilities "Offload Pin Playback" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" ] } SectionPCM."Offload Pin 1" { Index 1 Playback [ Capabilities "Offload Pin Playback" Config "PCM 48k Stereo 24bit" Config "PCM 48k Stereo 16bit" ] } SectionPCM."Loopback Pin" { Index 1 Capture [ Capabilities "Loopback Capture" Config "PCM 48k Stereo 16bit" ] }

3 4

[alsa-devel] [PATCH] ALSA: hda - add AZX_DCAPS_I915_POWERWELL to Baytrail
by mengdong.lin＠intel.com 23 Apr '15

23 Apr '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch addes AZX_DCAPS_I915_POWERWELL to BYT (Baytrail). Like Braswell and Skylake, the HDMI codec on Bytrail is also in the shared power well with GPU. This power well must be turned on before we reset link to probe the codec, to avoid communication failure with the codec. The side effect is that this power is always ON in S0 because the BYT HDMI codec does not support EPSS or D3ClkStop and so the controller doesn't enter D3 at runtime, and the HDMI codec and analog codec share a single physical HD-A link and so we cannot reset the HD-A link freely when we re-enable the power to use the HDMI codec. Next step is to test if an AGP reset or double AGP reset on BYT HDMI codec is okay to bring the HDMI codec back to a functional state after restoring the power. If okay, we can bind the power on/off with the HDMI codec PM without interrupting the analog audio. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e1c2105..34040d2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -297,6 +297,9 @@ enum { AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ AZX_DCAPS_SNOOP_TYPE(SCH)) +#define AZX_DCAPS_INTEL_BAYTRAIL \ + (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_I915_POWERWELL) + #define AZX_DCAPS_INTEL_BRASWELL \ (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_I915_POWERWELL) @@ -1992,7 +1995,7 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* BayTrail */ { PCI_DEVICE(0x8086, 0x0f04), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BAYTRAIL }, /* Braswell */ { PCI_DEVICE(0x8086, 0x2284), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BRASWELL }, -- 1.9.1

4 5

[alsa-devel] [RFC] bind/unbind SoC devices with deferred list
by Kuninori Morimoto 23 Apr '15

23 Apr '15

Hi Takashi, Mark, Lars I worked for bind/unbind issue which was discussed before http://thread.gmane.org/gmane.linux.alsa.devel/134062/focus=134076 http://thread.gmane.org/gmane.linux.alsa.devel/134283/focus=134287 I would like to know your opinion before sending patch to Greg The problem is ASoC has card/platform/cpu/codec relationship for each other, but, we can unbind one of them, and then, other related devices doesn't know about it. Thus, kernel will error if we use sound after that. Here is example of this issue. It unbind cpu (= rcar_sound), playback sound, and, error. And, we can't re-bind after this. // check current sound card > aplay -l **** List of PLAYBACK Hardware Devices **** card 0: rsnddai0ak4642h [rsnd-dai.0-ak4642-hifi], device 0: rsnd-dai.0-ak4642-hifi ak4642-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 // unbind cpu (= rcar_sound) > echo ec500000.rcar_sound > /sys/bus/platform/drivers/rcar_sound/unbind // recheck sound card // card doesn't know cpu was removed > aplay -l **** List of PLAYBACK Hardware Devices **** card 0: rsnddai0ak4642h [rsnd-dai.0-ak4642-hifi], device 0: rsnd-dai.0-ak4642-hifi ak4642-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 // kernel error happen if playback > aplay xxx.wav Unable to handle kernel NULL pointer dereference at virtual address 00000100 pgd = ee97c000 ... // we can't re-bind cpu > echo ec500000.rcar_sound > /sys/bus/platform/drivers/rcar_sound/bind -- freeze -- I hacked this issue to using deferred list, and created attached patch. it is all-in-one patch. but it solved my issue. I would like to know your opinion about this. // check current sound card > aplay -l **** List of PLAYBACK Hardware Devices **** card 0: rsnddai0ak4642h [rsnd-dai.0-ak4642-hifi], device 0: rsnd-dai.0-ak4642-hifi ak4642-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 // unbind cpu (= rcar_sound) > echo ec500000.rcar_sound > /sys/bus/platform/drivers/rcar_sound/unbind (local debug message) release ec500000.rcar_sound (local debug message) release related device - sound (local debug message) related related device - 6-0012 // recheck sound card // it knows there is no available card > aplay -l aplay: device_list:268: no soundcards found... // re-bind lost cpu (= rcar_sound) // it re-bind sound card > echo ec500000.rcar_sound > /sys/bus/platform/drivers/rcar_sound/bind rcar_sound ec500000.rcar_sound: probed asoc-simple-card sound: ak4642-hifi <-> ec500000.rcar_sound mapping ok // let's unbind codec side (= ak4642) this time > echo 6-0012 > /sys/bus/i2c/drivers/ak4642-codec/unbind (local debug message) release 6-0012 (local debug message) release related device - ec500000.rcar_sound (local debug message) related related device - sound // recheck sound card // it knows there is no available card > aplay -l aplay: device_list:268: no soundcards found... // re-bind lost codec (= ak4642) // it re-bind sound card > echo 6-0012 > /sys/bus/i2c/drivers/ak4642-codec/bind rcar_sound ec500000.rcar_sound: probed asoc-simple-card sound: ak4642-hifi <-> ec500000.rcar_sound mapping ok --- patch --- From 9da5324a659ff596a2733f14b768b6b8c5a04f3b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 22 Apr 2015 16:53:49 +0900 Subject: [PATCH] no Subject at this point Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- drivers/base/base.h | 1 + drivers/base/core.c | 9 +++++++++ drivers/base/dd.c | 21 ++++++++++++++++++++- include/linux/device.h | 1 + sound/soc/soc-core.c | 4 ++++ 5 files changed, 35 insertions(+), 1 deletion(-) diff --git a/drivers/base/base.h b/drivers/base/base.h index 251c5d3..1a71a52 100644 --- a/drivers/base/base.h +++ b/drivers/base/base.h @@ -74,6 +74,7 @@ struct device_private { struct klist_node knode_driver; struct klist_node knode_bus; struct list_head deferred_probe; + struct list_head related_dev; struct device *device; }; #define to_device_private_parent(obj) \ diff --git a/drivers/base/core.c b/drivers/base/core.c index 21d1303..643146f 100644 --- a/drivers/base/core.c +++ b/drivers/base/core.c @@ -948,9 +948,18 @@ int device_private_init(struct device *dev) klist_init(&dev->p->klist_children, klist_children_get, klist_children_put); INIT_LIST_HEAD(&dev->p->deferred_probe); + INIT_LIST_HEAD(&dev->p->related_dev); + return 0; } +void device_related_device(struct device *dev, struct device *node) +{ + if (list_empty(&node->p->related_dev)) + list_add(&node->p->related_dev, &dev->p->related_dev); +} +EXPORT_SYMBOL_GPL(device_related_device); + /** * device_add - add device to device hierarchy. * @dev: device. diff --git a/drivers/base/dd.c b/drivers/base/dd.c index e843fdb..3d218c6 100644 --- a/drivers/base/dd.c +++ b/drivers/base/dd.c @@ -516,7 +516,7 @@ EXPORT_SYMBOL_GPL(driver_attach); * __device_release_driver() must be called with @dev lock held. * When called for a USB interface, @dev->parent lock must be held as well. */ -static void __device_release_driver(struct device *dev) +static void _device_release_driver(struct device *dev) { struct device_driver *drv; @@ -552,6 +552,25 @@ static void __device_release_driver(struct device *dev) } } +static void __device_release_driver(struct device *dev) +{ + struct device_private *dp, *d; + + /* release dev itself */ + _device_release_driver(dev); + + list_for_each_entry_safe(dp, d, &dev->p->related_dev, related_dev) { + /* + * temporarily, release related device. + * these are push to deferred_probe_pending_list. + * It can be re-probed if it didn't call device_del() + */ + _device_release_driver(dp->device); + driver_deferred_probe_add(dp->device); + list_del_init(&dp->related_dev); + } +} + /** * device_release_driver - manually detach device from driver. * @dev: device. diff --git a/include/linux/device.h b/include/linux/device.h index 6558af9..2b81804 100644 --- a/include/linux/device.h +++ b/include/linux/device.h @@ -924,6 +924,7 @@ void driver_init(void); extern int __must_check device_register(struct device *dev); extern void device_unregister(struct device *dev); extern void device_initialize(struct device *dev); +extern void device_related_device(struct device *dev, struct device *node); extern int __must_check device_add(struct device *dev); extern void device_del(struct device *dev); extern int device_for_each_child(struct device *dev, void *data, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2373252..01c0d56 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -22,6 +22,7 @@ * o Support TDM on PCM and I2S */ +#include <linux/device.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> @@ -946,6 +947,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) dai_link->cpu_dai_name); return -EPROBE_DEFER; } + device_related_device(card->dev, rtd->cpu_dai->dev); rtd->num_codecs = dai_link->num_codecs; @@ -962,6 +964,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) /* Single codec links expect codec and codec_dai in runtime data */ rtd->codec_dai = codec_dais[0]; rtd->codec = rtd->codec_dai->codec; + device_related_device(card->dev, rtd->codec->dev); /* if there's no platform we match on the empty platform */ platform_name = dai_link->platform_name; @@ -986,6 +989,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) dai_link->platform_name); return -EPROBE_DEFER; } + device_related_device(card->dev, rtd->platform->dev); card->num_rtd++; -- 1.9.1 Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: dapm: Enable autodisable on SOC_DAPM_SINGLE_TLV_AUTODISABLE
by Charles Keepax 22 Apr '15

22 Apr '15

Correct small copy and paste error where autodisable was not being enabled for the SOC_DAPM_SINGLE_TLV_AUTODISABLE control. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 0bc8364..1065095 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -287,7 +287,7 @@ struct device; .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\ .tlv.p = (tlv_array), \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 1) } #define SOC_DAPM_SINGLE_TLV_VIRT(xname, max, tlv_array) \ SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0, tlv_array) #define SOC_DAPM_ENUM(xname, xenum) \ -- 1.7.2.5

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Re: [alsa-devel] Playback - Overwrite buffer with silence
by Clemens Ladisch 22 Apr '15

22 Apr '15

Nilhcraiv wrote: > To set the Buffer size and Periord size I have use the > snd_pcm_hw_params_set_period_size_near() and > snd_pcm_hw_params_set_buffer_size_near() functions. If I do not use > the ...near() functions, I return errors. The allowed values of the hardware parameters depend on the hardware's capabilities. Regards, Clemens

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[alsa-devel] [PATCH v8 0/7] ALSA: jack: Refactoring for jack kctls
by Jie Yang 22 Apr '15

22 Apr '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 new implementation, both HDA and soc jack kctls works fine, they are compatible for old HDA jack kctls, too. For soc, a. snd_jack_new() with NULL jack_kctl; b. for each pin, call snd_jack_add_kctls() to add one kctl(it will call snd_ctl_add() to add kctl to card); For HDA, With snd_jack_new(), we can use phantom_jack = true for phantom jack creating, and the jack kctl pointer will be filled with the new created kcontrol. Changes in v8: 1. remove snd_jack_kctl from HDA part, change snd_jack_new param to use bool for creating kcontrol at jack creating stage; 2. create jack for phantom jack, with no input device; 3. remove HDA jack get_unique_index() and index related code. Changes in v7: 1. move name generating and index getting part into ctljack.c; 2. remove exposing private_data and destructor to hda, then jack core can handle kctls by itself totally. 3. some small code fixing. Changes in v6: 1. refine kctl name generating function; 2. integrate HDA jack kctl more deeply, support phantom jack in snd_jack_new(); 3. add documentation for Jack kcontrols to explain how to use it. Changes in v5: 1. remove SND_KCTL_JACK config item, we need jack kctl once CONFIG_SND_JACK is selected. Changes in v4: 1. use snd_ctl_find_id() to get avaliable index; 2. add initial kctl for snd_jack_new(compatible for HDA); 3. add struct snd_jack_kctl * field to struct hda_jack_tbl; 4. new kctls for soc jack during jack pins creating. 5. add a patch to remove exporting snd_kctl_jack_new(). 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. *** BLURB HERE *** Jie Yang (7): ALSA: jack: implement kctl creating for jack device ALSA: Jack: refactoring snd_kctl_jack_new to support embedded kctl ALSA: jack: extend snd_jack_new to support phantom jack ALSA: hda - Update to use the new jack kctls method ASoC: jack: create kctls according to jack pins info ALSA: jack: remove exporting ctljack functions ALSA: Docs: Add documentation for Jack kcontrols Documentation/sound/alsa/Jack-Controls.txt | 48 ++++++++++ include/sound/control.h | 2 +- include/sound/jack.h | 19 +++- sound/core/Kconfig | 3 - sound/core/Makefile | 3 +- sound/core/ctljack.c | 41 +++++++-- sound/core/jack.c | 142 ++++++++++++++++++++++++++--- sound/pci/hda/Kconfig | 1 - sound/pci/hda/hda_jack.c | 90 +++++------------- sound/pci/hda/hda_jack.h | 5 +- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/soc-jack.c | 3 +- 13 files changed, 260 insertions(+), 101 deletions(-) create mode 100644 Documentation/sound/alsa/Jack-Controls.txt -- 1.9.1

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[alsa-devel] [RFC 1/4] ASoC: topology: Add topology UAPI header.
by Liam Girdwood 22 Apr '15

22 Apr '15

The ASoC topology UAPI header defines the structures required to define any DSP firmware audio topology and control objects from userspace. The following objects are supported :- o kcontrols including TLV controls. o DAPM widgets and graph elements o Vendor bespoke objects. o Coefficient data o FE PCM capabilities and config. o BE link capabilities and config. o Codec <-> codec link capabilities and config. o Topology object manifest. The file format is simple and divided into blocks for each object type and each block has a header that defines it's size and type. Blocks can be in any order of type and can either all be in a single file or spread across more than one file. Blocks also have a group identifier ID so that they can be loaded and unloaded by ID. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dapm.h | 33 +--- include/uapi/sound/asoc.h | 420 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 421 insertions(+), 32 deletions(-) create mode 100644 include/uapi/sound/asoc.h diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index b1ec05b..3263e36 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -15,6 +15,7 @@ #include <linux/types.h> #include <sound/control.h> +#include <sound/asoc.h> struct device; @@ -447,38 +448,6 @@ 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); -/* dapm widget types */ -enum snd_soc_dapm_type { - snd_soc_dapm_input = 0, /* input pin */ - snd_soc_dapm_output, /* output pin */ - snd_soc_dapm_mux, /* selects 1 analog signal from many inputs */ - snd_soc_dapm_mixer, /* mixes several analog signals together */ - snd_soc_dapm_mixer_named_ctl, /* mixer with named controls */ - snd_soc_dapm_pga, /* programmable gain/attenuation (volume) */ - snd_soc_dapm_out_drv, /* output driver */ - snd_soc_dapm_adc, /* analog to digital converter */ - snd_soc_dapm_dac, /* digital to analog converter */ - snd_soc_dapm_micbias, /* microphone bias (power) */ - snd_soc_dapm_mic, /* microphone */ - snd_soc_dapm_hp, /* headphones */ - snd_soc_dapm_spk, /* speaker */ - snd_soc_dapm_line, /* line input/output */ - snd_soc_dapm_switch, /* analog switch */ - snd_soc_dapm_vmid, /* codec bias/vmid - to minimise pops */ - snd_soc_dapm_pre, /* machine specific pre widget - exec first */ - snd_soc_dapm_post, /* machine specific post widget - exec last */ - snd_soc_dapm_supply, /* power/clock supply */ - snd_soc_dapm_regulator_supply, /* external regulator */ - snd_soc_dapm_clock_supply, /* external clock */ - snd_soc_dapm_aif_in, /* audio interface input */ - snd_soc_dapm_aif_out, /* audio interface output */ - snd_soc_dapm_siggen, /* signal generator */ - snd_soc_dapm_dai_in, /* link to DAI structure */ - snd_soc_dapm_dai_out, - snd_soc_dapm_dai_link, /* link between two DAI structures */ - snd_soc_dapm_kcontrol, /* Auto-disabled kcontrol */ -}; - enum snd_soc_dapm_subclass { SND_SOC_DAPM_CLASS_INIT = 0, SND_SOC_DAPM_CLASS_RUNTIME = 1, diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h new file mode 100644 index 0000000..dde7133 --- /dev/null +++ b/include/uapi/sound/asoc.h @@ -0,0 +1,420 @@ +/* + * uapi/sound/asoc.h -- ALSA SoC Firmware Controls and DAPM + * + * Copyright (C) 2012 Texas Instruments Inc. + * Copyright (C) 2015 Intel Corporation. + * + * 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. + * + * Simple file API to load FW that includes mixers, coefficients, DAPM graphs, + * algorithms, equalisers, DAIs, widgets etc. +*/ + +#ifndef __LINUX_UAPI_SND_ASOC_H +#define __LINUX_UAPI_SND_ASOC_H + +#include <linux/types.h> +#include <sound/asound.h> + +/* + * Numeric IDs for stock mixer types that are used to map and enumerate FW + * based mixers. + */ +#define SOC_CONTROL_ID_PUT(p) ((p & 0xff) << 16) +#define SOC_CONTROL_ID_GET(g) ((g & 0xff) << 8) +#define SOC_CONTROL_ID_INFO(i) ((i & 0xff) << 0) +#define SOC_CONTROL_ID(g, p, i) \ + (SOC_CONTROL_ID_PUT(p) | SOC_CONTROL_ID_GET(g) |\ + SOC_CONTROL_ID_INFO(i)) + +#define SOC_CONTROL_GET_ID_PUT(id) ((id & 0xff0000) >> 16) +#define SOC_CONTROL_GET_ID_GET(id) ((id & 0x00ff00) >> 8) +#define SOC_CONTROL_GET_ID_INFO(id) ((id & 0x0000ff) >> 0) + +/* individual kcontrol info types - can be mixed with other types */ +#define SOC_CONTROL_TYPE_EXT 0 /* driver defined */ +#define SOC_CONTROL_TYPE_VOLSW 1 +#define SOC_CONTROL_TYPE_VOLSW_SX 2 +#define SOC_CONTROL_TYPE_VOLSW_S8 3 +#define SOC_CONTROL_TYPE_VOLSW_XR_SX 4 +#define SOC_CONTROL_TYPE_ENUM 6 +#define SOC_CONTROL_TYPE_ENUM_EXT 7 +#define SOC_CONTROL_TYPE_BYTES 8 +#define SOC_CONTROL_TYPE_BOOL_EXT 9 +#define SOC_CONTROL_TYPE_ENUM_VALUE 10 +#define SOC_CONTROL_TYPE_RANGE 11 +#define SOC_CONTROL_TYPE_STROBE 12 +#define SOC_CONTROL_TYPE_BYTES_EXT 13 +#define SOC_CONTROL_TYPE_VOLSW_EXT 14 + +/* convenience macros for compound control type mappings */ +#define SOC_CONTROL_IO_VOLSW \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_VOLSW, \ + SOC_CONTROL_TYPE_VOLSW, \ + SOC_CONTROL_TYPE_VOLSW) +#define SOC_CONTROL_IO_VOLSW_SX \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_VOLSW_SX, \ + SOC_CONTROL_TYPE_VOLSW_SX, \ + SOC_CONTROL_TYPE_VOLSW) +#define SOC_CONTROL_IO_VOLSW_S8 \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_VOLSW_S8, \ + SOC_CONTROL_TYPE_VOLSW_S8, \ + SOC_CONTROL_TYPE_VOLSW_S8) +#define SOC_CONTROL_IO_VOLSW_XR_SX \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_VOLSW_XR_SX, \ + SOC_CONTROL_TYPE_VOLSW_XR_SX, \ + SOC_CONTROL_TYPE_VOLSW_XR_SX) +#define SOC_CONTROL_IO_EXT \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_VOLSW) +#define SOC_CONTROL_IO_ENUM \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_ENUM, \ + SOC_CONTROL_TYPE_ENUM, \ + SOC_CONTROL_TYPE_ENUM) +#define SOC_CONTROL_IO_ENUM_EXT \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_ENUM_EXT) +#define SOC_CONTROL_IO_BYTES \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_BYTES, \ + SOC_CONTROL_TYPE_BYTES, \ + SOC_CONTROL_TYPE_BYTES) +#define SOC_CONTROL_IO_BOOL_EXT \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_BOOL_EXT) +#define SOC_CONTROL_IO_ENUM_VALUE \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_ENUM_VALUE, \ + SOC_CONTROL_TYPE_ENUM_VALUE, \ + SOC_CONTROL_TYPE_ENUM) +#define SOC_CONTROL_IO_RANGE \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_RANGE, \ + SOC_CONTROL_TYPE_RANGE, \ + SOC_CONTROL_TYPE_RANGE) +#define SOC_CONTROL_IO_STROBE \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_STROBE, \ + SOC_CONTROL_TYPE_STROBE, \ + SOC_CONTROL_TYPE_STROBE) +#define SOC_CONTROL_IO_BYTES_EXT \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_BYTES_EXT) +#define SOC_CONTROL_IO_VOLSW_EXT \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_VOLSW) + +/* individual widget kcontrol info types - can be mixed with other types */ +#define SOC_DAPM_TYPE_VOLSW 64 +#define SOC_DAPM_TYPE_ENUM_DOUBLE 65 +#define SOC_DAPM_TYPE_ENUM_VIRT 66 +#define SOC_DAPM_TYPE_ENUM_VALUE 67 +#define SOC_DAPM_TYPE_PIN 68 +#define SOC_DAPM_TYPE_ENUM_EXT 69 + +/* convenience macros for compound widget kcontrol type mappings */ +#define SOC_DAPM_IO_VOLSW \ + SOC_CONTROL_ID(SOC_DAPM_TYPE_VOLSW, \ + SOC_DAPM_TYPE_VOLSW, \ + SOC_DAPM_TYPE_VOLSW) +#define SOC_DAPM_IO_ENUM_DOUBLE \ + SOC_CONTROL_ID(SOC_DAPM_TYPE_ENUM_DOUBLE, \ + SOC_DAPM_TYPE_ENUM_DOUBLE, \ + SOC_CONTROL_TYPE_ENUM) +#define SOC_DAPM_IO_ENUM_VIRT \ + SOC_CONTROL_ID(SOC_DAPM_TYPE_ENUM_VIRT, \ + SOC_DAPM_TYPE_ENUM_VIRT, \ + SOC_CONTROL_TYPE_ENUM) +#define SOC_DAPM_IO_ENUM_VALUE \ + SOC_CONTROL_ID(SOC_DAPM_TYPE_ENUM_VALUE, \ + SOC_DAPM_TYPE_ENUM_VALUE, \ + SOC_CONTROL_TYPE_ENUM) +#define SOC_DAPM_IO_PIN \ + SOC_CONTROL_ID(SOC_DAPM_TYPE_PIN, \ + SOC_DAPM_TYPE_PIN, \ + SOC_DAPM_TYPE_PIN) +#define SOC_DAPM_IO_ENUM_EXT \ + SOC_CONTROL_ID(SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_EXT, \ + SOC_CONTROL_TYPE_ENUM) + +/* dapm widget types */ +enum snd_soc_dapm_type { + snd_soc_dapm_input = 0, /* input pin */ + snd_soc_dapm_output, /* output pin */ + snd_soc_dapm_mux, /* selects 1 analog signal from many inputs */ + snd_soc_dapm_mixer, /* mixes several analog signals together */ + snd_soc_dapm_mixer_named_ctl, /* mixer with named controls */ + snd_soc_dapm_pga, /* programmable gain/attenuation (volume) */ + snd_soc_dapm_out_drv, /* output driver */ + snd_soc_dapm_adc, /* analog to digital converter */ + snd_soc_dapm_dac, /* digital to analog converter */ + snd_soc_dapm_micbias, /* microphone bias (power) */ + snd_soc_dapm_mic, /* microphone */ + snd_soc_dapm_hp, /* headphones */ + snd_soc_dapm_spk, /* speaker */ + snd_soc_dapm_line, /* line input/output */ + snd_soc_dapm_switch, /* analog switch */ + snd_soc_dapm_vmid, /* codec bias/vmid - to minimise pops */ + snd_soc_dapm_pre, /* machine specific pre widget - exec first */ + snd_soc_dapm_post, /* machine specific post widget - exec last */ + snd_soc_dapm_supply, /* power/clock supply */ + snd_soc_dapm_regulator_supply, /* external regulator */ + snd_soc_dapm_clock_supply, /* external clock */ + snd_soc_dapm_aif_in, /* audio interface input */ + snd_soc_dapm_aif_out, /* audio interface output */ + snd_soc_dapm_siggen, /* signal generator */ + snd_soc_dapm_dai_in, /* link to DAI structure */ + snd_soc_dapm_dai_out, + snd_soc_dapm_dai_link, /* link between two DAI structures */ + snd_soc_dapm_kcontrol, /* Auto-disabled kcontrol */ +}; + +/* Header magic number and string sizes */ +#define SND_SOC_TPLG_MAGIC 0x41536F43 /* ASoC */ + +/* string sizes */ +#define SND_SOC_TPLG_NUM_TEXTS 16 + +/* ABI version */ +#define SND_SOC_TPLG_ABI_VERSION 0x2 + +/* + * File and Block header data types. + * Add new generic and vendor types to end of list. + * Generic types are handled by the core whilst vendors types are passed + * to the component drivers for handling. + */ +#define SND_SOC_TPLG_MIXER 1 +#define SND_SOC_TPLG_DAPM_GRAPH 2 +#define SND_SOC_TPLG_DAPM_WIDGET 3 +#define SND_SOC_TPLG_DAI_LINK 4 +#define SND_SOC_TPLG_PCM 5 +#define SND_SOC_TPLG_MANIFEST 6 +#define SND_SOC_TPLG_CODEC_LINK 7 +#define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_CODEC_LINK + +/* vendor block IDs - please add new vendor types to end */ +#define SND_SOC_TPLG_VENDOR_FW 1000 +#define SND_SOC_TPLG_VENDOR_CONFIG 1001 +#define SND_SOC_TPLG_VENDOR_COEFF 1002 +#define SND_SOC_TPLG_VENDOR_CODEC 1003 + +/* + * File and Block Header. + * This header preceeds all object and object arrays below. + */ +struct snd_soc_tplg_hdr { + __le32 magic; + __le32 abi; /* ABI version */ + __le32 version; /* optional vendor specific version details */ + __le32 type; /* SND_SOC_TPLG_ */ + __le32 vendor_type; /* optional vendor specific type info */ + __le32 size; /* data bytes, excluding this header */ + __le32 id; /* identifier for block */ + char reserved[128]; +} __attribute__((packed)); + +/* + * Manifest. List totals for each payload type. Not used in parsing, but will + * be passed to the component driver before any other objects in order for any + * global componnent resource allocations. + */ +struct snd_soc_tplg_manifest { + __le32 control_elems; /* number of control elements */ + __le32 widget_elems; /* number of widget elements */ + __le32 graph_elems; /* number of graph elements */ + __le32 dai_elems; /* number of DAI elements */ + __le32 dai_link_elems; /* number of DAI link elements */ + char reserved[128]; +} __attribute__((packed)); + +/* + * Element Header. Defines number of elements in an object array. + * This prceeeds al mixers, widgets, graph elems, BEs, FEs, CC links. + */ +struct snd_soc_tplg_elems { + __le32 count; /* number of elements */ + /* elements start here */ +} __attribute__((packed)); + +/* + * Private data. + * All topology objects may have private data that can be used by the driver or + * firmware. Core will ignore this data. + */ +struct snd_soc_tplg_private { + __le32 length; + char data[0]; +} __attribute__((packed)); + +/* + * Kcontrol TLV data. + */ +struct snd_soc_tplg_ctl_tlv { + __le32 numid; /* control element numeric identification */ + __le32 length; /* in bytes aligned to 4 */ + char reserved[16]; + /* tlv data starts here */ +} __attribute__((packed)); + +/* + * kcontrol header + */ +struct snd_soc_tplg_control_hdr { + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 index; + __le32 access; + __le32 tlv_size; + char reserved[16]; +} __attribute__((packed)); + +/* + * Mixer kcontrol. + */ +struct snd_soc_tplg_mixer_control { + struct snd_soc_tplg_control_hdr hdr; + __le32 min; + __le32 max; + __le32 platform_max; + __le32 reg; + __le32 rreg; + __le32 shift; + __le32 rshift; + __le32 invert; + char reserved[64]; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * Enumerated kcontrol + */ +struct snd_soc_tplg_enum_control { + struct snd_soc_tplg_control_hdr hdr; + __le32 reg; + __le32 shift_l; + __le32 shift_r; + __le32 items; + __le32 mask; + __le32 count; + char reserved[64]; + char texts[SND_SOC_TPLG_NUM_TEXTS][SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 values[SND_SOC_TPLG_NUM_TEXTS * SNDRV_CTL_ELEM_ID_NAME_MAXLEN / 4]; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * Bytes kcontrol + */ +struct snd_soc_tplg_bytes_ext { + struct snd_soc_tplg_control_hdr hdr; + __le32 max; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * DAPM Graph Element + */ +struct snd_soc_tplg_dapm_graph_elem { + char sink[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char control[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char source[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; +} __attribute__((packed)); + +/* + * DAPM Widget. + */ +struct snd_soc_tplg_dapm_widget { + __le32 id; /* snd_soc_dapm_type */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char sname[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + __le32 reg; /* negative reg = no direct dapm */ + __le32 shift; /* bits to shift */ + __le32 mask; /* non-shifted mask */ + __u32 invert; /* invert the power bit */ + __u32 ignore_suspend; /* kept enabled over suspend */ + __u16 event_flags; + __u16 event_type; + __u16 num_kcontrols; + char reserved[64]; + struct snd_soc_tplg_private priv; + /* + * kcontrols that relate to this widget + * follow here after widget private data + */ +} __attribute__((packed)); + +/* + * Coefficient File Data. + */ +struct snd_soc_file_coeff_data { + __le32 count; /* in elems */ + __le32 size; /* total data size */ + __le32 id; /* associated mixer ID */ + char reserved[16]; + /* data here */ +} __attribute__((packed)); + +/* + * Stream Capabilities + */ +struct snd_soc_tplg_stream_caps { + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le64 formats; /* supported formats SNDRV_PCM_FMTBIT_* */ + __le32 rates; /* supported rates SNDRV_PCM_RATE_* */ + __le32 rate_min; /* min rate */ + __le32 rate_max; /* max rate */ + __le32 channels_min; /* min channels */ + __le32 channels_max; /* max channels */ + __le32 periods_min; /* min number of periods */ + __le32 periods_max; /* max number of periods */ + __le32 period_size_min; /* min period size bytes */ + __le32 period_size_max; /* max period size bytes */ + __le32 buffer_size_min; /* min buffer size bytes */ + __le32 buffer_size_max; /* max buffer size bytes */ + char reserved[64]; +} __attribute__((packed)); + +/* + * FE or BE Stream configuration supported by SW/FW + */ +struct snd_soc_tplg_stream { + __le64 format; /* SNDRV_PCM_FMTBIT_* */ + __le32 rate; /* SNDRV_PCM_RATE_* */ + __le32 period_bytes; /* size of period in bytes */ + __le32 buffer_bytes; /* size of buffer in bytes */ + __le32 channels; /* channels */ + __le32 tdm_slot; /* optional BE bitmask of supported TDM slots */ + __le32 dai_fmt; /* SND_SOC_DAIFMT_ */ + char reserved[16]; +} __attribute__((packed)); + +/* + * Duplex stream configuration supported by SW/FW. + */ +struct snd_soc_tplg_stream_config { + struct snd_soc_tplg_stream playback; + struct snd_soc_tplg_stream capture; + char reserved[16]; +} __attribute__((packed)); + +/* + * Describes SW/FW specific features of PCM or DAI link. + */ +struct snd_soc_tplg_pcm_dai { + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 id; /* unique BE ID - used to match */ + __le32 playback; /* supports playback in DPCM mode */ + __le32 capture; /* supports capture in DPCM mode */ + __le32 compress; /* 1 = compressed; 0 = PCM */ + __le32 num_configs; /* number of configs */ + char reserved[64]; + struct snd_soc_tplg_stream_caps caps[2]; /* capabilities */ + struct snd_soc_tplg_stream_config config[0]; /* supported SW/FW configs */ +}__attribute__((packed)); + +#endif -- 2.1.0

4 14

[alsa-devel] [PATCH 1/2] ASoC: rt5645: Adds push button support for rt5650
by Bard Liao 22 Apr '15

22 Apr '15

rt5650 support headset button detection. Currently, the button detection is only implemented for rt5650 codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 262 +++++++++++++++++++++++++++++++++++++++------- sound/soc/codecs/rt5645.h | 8 +- 2 files changed, 231 insertions(+), 39 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 69528ae..7c59f9c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -198,7 +198,6 @@ static const struct reg_default rt5645_reg[] = { { 0xda, 0x0000 }, { 0xdb, 0x0003 }, { 0xdc, 0x0049 }, - { 0xdd, 0x001b }, { 0xdf, 0x0008 }, { 0xe0, 0x4000 }, { 0xe6, 0x8000 }, @@ -2368,6 +2367,8 @@ static int rt5645_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, static int rt5645_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_PREPARE: if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { @@ -2398,8 +2399,9 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: snd_soc_write(codec, RT5645_DEPOP_M2, 0x1100); - snd_soc_update_bits(codec, RT5645_GEN_CTRL1, - RT5645_DIG_GATE_CTRL, 0); + if (!rt5645->en_button_func) + snd_soc_update_bits(codec, RT5645_GEN_CTRL1, + RT5645_DIG_GATE_CTRL, 0); snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_VREF1 | RT5645_PWR_MB | RT5645_PWR_BG | RT5645_PWR_VREF2 | @@ -2414,28 +2416,71 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int rt5645_jack_detect(struct snd_soc_codec *codec) +static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, + bool enable) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - int gpio_state, jack_type = 0; - unsigned int val; - if (!gpio_is_valid(rt5645->pdata.hp_det_gpio)) { - dev_err(codec->dev, "invalid gpio\n"); - return -EINVAL; + if (enable) { + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "ADC L power"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "ADC R power"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "LDO2"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync_unlocked(&codec->dapm); + snd_soc_update_bits(codec, + RT5645_INT_IRQ_ST, 0x8, 0x8); + snd_soc_update_bits(codec, + RT5650_4BTN_IL_CMD2, 0x8000, 0x8000); + snd_soc_read(codec, RT5650_4BTN_IL_CMD1); + pr_debug("%s read %x = %x\n", __func__, RT5650_4BTN_IL_CMD1, + snd_soc_read(codec, RT5650_4BTN_IL_CMD1)); + } else { + snd_soc_update_bits(codec, RT5650_4BTN_IL_CMD2, 0x8000, 0x0); + snd_soc_update_bits(codec, RT5645_INT_IRQ_ST, 0x8, 0x0); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "ADC L power"); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "ADC R power"); + if (rt5645->pdata.jd_mode == 0) + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "LDO2"); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync_unlocked(&codec->dapm); } - gpio_state = gpio_get_value(rt5645->pdata.hp_det_gpio); +} - dev_dbg(codec->dev, "gpio = %d(%d)\n", rt5645->pdata.hp_det_gpio, - gpio_state); +static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + unsigned int val; - if ((rt5645->pdata.gpio_hp_det_active_high && gpio_state) || - (!rt5645->pdata.gpio_hp_det_active_high && !gpio_state)) { - snd_soc_dapm_force_enable_pin(&codec->dapm, "micbias1"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "micbias2"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "Mic Det Power"); - snd_soc_dapm_sync(&codec->dapm); + if (jack_insert) { + if (codec->component.card->instantiated) { + snd_soc_dapm_force_enable_pin(&codec->dapm, + "micbias1"); + snd_soc_dapm_force_enable_pin(&codec->dapm, + "micbias2"); + snd_soc_dapm_force_enable_pin(&codec->dapm, + "LDO2"); + snd_soc_dapm_force_enable_pin(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } else { + /* Power up necessary bits for JD if dapm is + not ready yet */ + snd_soc_update_bits(codec, RT5645_PWR_ANLG2, + RT5645_PWR_MB1 | RT5645_PWR_MB2, + RT5645_PWR_MB1 | RT5645_PWR_MB2); + snd_soc_update_bits(codec, RT5645_PWR_MIXER, + RT5645_PWR_LDO2, RT5645_PWR_LDO2); + snd_soc_update_bits(codec, RT5645_PWR_VOL, + RT5645_PWR_MIC_DET, RT5645_PWR_MIC_DET); + } snd_soc_write(codec, RT5645_IN1_CTRL1, 0x0006); snd_soc_write(codec, RT5645_JD_CTRL3, 0x00b0); @@ -2449,32 +2494,62 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec) val = snd_soc_read(codec, RT5645_IN1_CTRL3) & 0x7; dev_dbg(codec->dev, "val = %d\n", val); - if (val == 1 || val == 2) - jack_type = SND_JACK_HEADSET; - else - jack_type = SND_JACK_HEADPHONE; + if (codec->component.card->instantiated) { + snd_soc_dapm_disable_pin(&codec->dapm, "micbias1"); + snd_soc_dapm_disable_pin(&codec->dapm, "micbias2"); + if (rt5645->pdata.jd_mode == 0) + snd_soc_dapm_disable_pin(&codec->dapm, "LDO2"); + snd_soc_dapm_disable_pin(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } else { + snd_soc_update_bits(codec, RT5645_PWR_ANLG2, + RT5645_PWR_MB1 | RT5645_PWR_MB2, 0); + if (rt5645->pdata.jd_mode == 0) + snd_soc_update_bits(codec, RT5645_PWR_MIXER, + RT5645_PWR_LDO2, 0); + snd_soc_update_bits(codec, RT5645_PWR_VOL, + RT5645_PWR_MIC_DET, 0); + } - snd_soc_dapm_disable_pin(&codec->dapm, "micbias1"); - snd_soc_dapm_disable_pin(&codec->dapm, "micbias2"); - if (rt5645->pdata.jd_mode == 0) - snd_soc_dapm_disable_pin(&codec->dapm, "LDO2"); - snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); - snd_soc_dapm_sync(&codec->dapm); + if (val == 1 || val == 2) { + rt5645->jack_type = SND_JACK_HEADSET; + if (rt5645->en_button_func) { + msleep(100); + rt5645_enable_push_button_irq(codec, true); + } + } else { + rt5645->jack_type = SND_JACK_HEADPHONE; + } + + } else { /* jack out */ + rt5645->jack_type = 0; + if (rt5645->en_button_func) + rt5645_enable_push_button_irq(codec, false); } - snd_soc_jack_report(rt5645->hp_jack, jack_type, SND_JACK_HEADPHONE); - snd_soc_jack_report(rt5645->mic_jack, jack_type, SND_JACK_MICROPHONE); - return 0; + return rt5645->jack_type; } int rt5645_set_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack) + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack, + struct snd_soc_jack *btn_jack) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); rt5645->hp_jack = hp_jack; rt5645->mic_jack = mic_jack; - rt5645_jack_detect(codec); + rt5645->btn_jack = btn_jack; + if (rt5645->btn_jack && rt5645->codec_type == CODEC_TYPE_RT5650) { + rt5645->en_button_func = true; + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ); + regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1, + RT5645_HP_CB_MASK, RT5645_HP_CB_PU); + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1, + RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); + } + rt5645_irq_detection(rt5645); return 0; } @@ -2485,7 +2560,7 @@ static void rt5645_jack_detect_work(struct work_struct *work) struct rt5645_priv *rt5645 = container_of(work, struct rt5645_priv, jack_detect_work.work); - rt5645_jack_detect(rt5645->codec); + rt5645_irq_detection(rt5645); } static irqreturn_t rt5645_irq(int irq, void *data) @@ -2498,6 +2573,120 @@ static irqreturn_t rt5645_irq(int irq, void *data) return IRQ_HANDLED; } +static int rt5645_button_detect(struct snd_soc_codec *codec) +{ + int btn_type, val; + + val = snd_soc_read(codec, RT5650_4BTN_IL_CMD1); + pr_debug("val=0x%x\n", val); + btn_type = val & 0xfff0; + snd_soc_write(codec, RT5650_4BTN_IL_CMD1, val); + + return btn_type; +} + +static int rt5645_irq_detection(struct rt5645_priv *rt5645) +{ + int val, btn_type, gpio_state = 0, report = 0; + + switch (rt5645->pdata.jd_mode) { + case 0: /* Not using rt5645 JD */ + if (gpio_is_valid(rt5645->pdata.hp_det_gpio)) { + gpio_state = gpio_get_value(rt5645->pdata.hp_det_gpio); + dev_dbg(rt5645->codec->dev, "gpio = %d(%d)\n", + rt5645->pdata.hp_det_gpio, gpio_state); + } + if ((rt5645->pdata.gpio_hp_det_active_high && gpio_state) || + (!rt5645->pdata.gpio_hp_det_active_high && + !gpio_state)) { + report = rt5645_jack_detect(rt5645->codec, 1); + } else { + report = rt5645_jack_detect(rt5645->codec, 0); + } + snd_soc_jack_report(rt5645->hp_jack, + report, SND_JACK_HEADPHONE); + snd_soc_jack_report(rt5645->mic_jack, + report, SND_JACK_MICROPHONE); + return report; + case 1: /* 2 port */ + val = snd_soc_read(rt5645->codec, RT5645_A_JD_CTRL1) & 0x0070; + break; + default: /* 1 port */ + val = snd_soc_read(rt5645->codec, RT5645_A_JD_CTRL1) & 0x0020; + break; + + } + + switch (val) { + /* jack in */ + case 0x30: /* 2 port */ + case 0x0: /* 1 port or 2 port */ + if (rt5645->jack_type == 0) { + report = rt5645_jack_detect(rt5645->codec, 1); + /* for push button and jack out */ + break; + } + btn_type = 0; + if (snd_soc_read(rt5645->codec, RT5645_INT_IRQ_ST) & 0x4) { + /* button pressed */ + report = SND_JACK_HEADSET; + btn_type = rt5645_button_detect(rt5645->codec); + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + report |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + report |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + report |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + report |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + dev_err(rt5645->codec->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + } + if (btn_type == 0)/* button release */ + report = rt5645->jack_type; + + break; + /* jack out */ + case 0x70: /* 2 port */ + case 0x10: /* 2 port */ + case 0x20: /* 1 port */ + report = 0; + snd_soc_update_bits(rt5645->codec, + RT5645_INT_IRQ_ST, 0x1, 0x0); + rt5645_jack_detect(rt5645->codec, 0); + break; + default: + break; + } + + snd_soc_jack_report(rt5645->hp_jack, report, SND_JACK_HEADPHONE); + snd_soc_jack_report(rt5645->mic_jack, report, SND_JACK_MICROPHONE); + if (rt5645->en_button_func) + snd_soc_jack_report(rt5645->btn_jack, + report, SND_JACK_MICROPHONE); + + return report; +} + static int rt5645_probe(struct snd_soc_codec *codec) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); @@ -2784,8 +2973,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (rt5645->pdata.en_jd_func) { regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3, - RT5645_IRQ_CLK_GATE_CTRL | RT5645_MICINDET_MANU, - RT5645_IRQ_CLK_GATE_CTRL | RT5645_MICINDET_MANU); + RT5645_IRQ_CLK_GATE_CTRL, RT5645_IRQ_CLK_GATE_CTRL); regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, RT5645_CBJ_BST1_EN, RT5645_CBJ_BST1_EN); regmap_update_bits(rt5645->regmap, RT5645_JD_CTRL3, diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index db78e94..4473636 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2184,6 +2184,7 @@ struct rt5645_priv { struct i2c_client *i2c; struct snd_soc_jack *hp_jack; struct snd_soc_jack *mic_jack; + struct snd_soc_jack *btn_jack; struct delayed_work jack_detect_work; int codec_type; @@ -2196,9 +2197,12 @@ struct rt5645_priv { int pll_src; int pll_in; int pll_out; + + int jack_type; + bool en_button_func; }; int rt5645_set_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack); - + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack, + struct snd_soc_jack *btn_jack); #endif /* __RT5645_H__ */ -- 1.8.1.1.439.g50a6b54

2 7

[alsa-devel] [PATCH 0/2] Calculate BCLK using TDM slots and remove channels rule
by Jyri Sarha 22 Apr '15

22 Apr '15

The first patch is a bugfix. I did not have the HW see the problem myself, but reading from the code the problem is evident. This should also fix Dan Carpenter's concern about stack usage in davinci_mcasp_hw_rule_channels(), as the whole function is removed. The second patch is just an optimization of the sample-rate rule. In effect I put in use Takashi Iwai's suggestion for fixing the stack usage issue in the channels rule. Jyri Sarha (2): ASoC: davinci-mcasp: Calculate BCLK using TDM slots and remove channels rule ASoC: davinci-macsp: Optimize implicit BLCK sample-rate rule sound/soc/davinci/davinci-mcasp.c | 104 ++++++++++---------------------------- 1 file changed, 27 insertions(+), 77 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH v7 0/7] ALSA: jack: Refactoring for jack kctls
by Jie Yang 22 Apr '15

22 Apr '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 new implementation, both HDA and soc jack kctls works fine, they are compatible for old HDA jack kctls, too. For soc, a. snd_jack_new() with NULL jack_kctl; b. for each pin, call snd_jack_add_kctls() to add one kctl(it will call snd_ctl_add() to add kctl to card); For HDA, With snd_jack_new(), we can use phantom_jack = true for phantom jack creating, and the jack kctl pointer will be filled with the new created kcontrol. Changes in v7: 1. move name generating and index getting part into ctljack.c; 2. remove exposing private_data and destructor to hda, then jack core can handle kctls by itself totally. 3. some small code fixing. Changes in v6: 1. refine kctl name generating function; 2. integrate HDA jack kctl more deeply, support phantom jack in snd_jack_new(); 3. add documentation for Jack kcontrols to explain how to use it. Changes in v5: 1. remove SND_KCTL_JACK config item, we need jack kctl once CONFIG_SND_JACK is selected. Changes in v4: 1. use snd_ctl_find_id() to get avaliable index; 2. add initial kctl for snd_jack_new(compatible for HDA); 3. add struct snd_jack_kctl * field to struct hda_jack_tbl; 4. new kctls for soc jack during jack pins creating. 5. add a patch to remove exporting snd_kctl_jack_new(). 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 (7): ALSA: jack: implement kctl creating for jack device ALSA: Jack: refactoring snd_kctl_jack_new to support embedded kctl ALSA: jack: extend snd_jack_new to support phantom jack ALSA: hda - Update to use the new jack kctls method ASoC: jack: create kctls according to jack pins info ALSA: jack: remove exporting ctljack functions ALSA: Docs: Add documentation for Jack kcontrols Documentation/sound/alsa/Jack-Controls.txt | 48 +++++++++ include/sound/control.h | 2 +- include/sound/jack.h | 19 +++- sound/core/Kconfig | 3 - sound/core/Makefile | 3 +- sound/core/ctljack.c | 41 ++++++-- sound/core/jack.c | 156 +++++++++++++++++++++++++---- sound/pci/hda/Kconfig | 1 - sound/pci/hda/hda_jack.c | 53 ++++------ sound/pci/hda/hda_jack.h | 4 +- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/soc-jack.c | 3 +- 12 files changed, 259 insertions(+), 76 deletions(-) create mode 100644 Documentation/sound/alsa/Jack-Controls.txt -- 1.9.1

3 16

[alsa-devel] [RFC 0/4] ASoC DSP topology
by Liam Girdwood 22 Apr '15

22 Apr '15

It's been about 2 years since posting V1 of the ASoC dynamic firmware patches and a lot has changed since then ... This RFC series is being posted instead of a V2 patch series atm since we are still completing some testing on the codebase and developing a new userspace tool. Expect V2 once more testing is completed shortly. ASoC topology (formerly called dynamic firmware) allows audio DSP topology to be defined as part of the firmware package and not hard coded into the driver. This allows DSP vendors to ship multiple different firmwares or firmwares with programmable topologies for different devices (i.e. phones, tablets, TVs) using a single DSP driver. The DSP topology can be contained as part of the firmware file or exist as separate files. The topology core and file format is generic and not tied to any single vendor or device. The topology core consists of kernel and userspace components :- Kernel. The kernel core reads in the topology data and builds the driver capabilities and audio topology based on this data. This includes building the DAPM graph, setting PCM and DAI capabilities, registering kcontrols and registering DAPM widgets. The userspace tool allows the creation of binary DSP FW topology files based on a topology text files. Changes since V1:- o Name changed to ASoC Topology o Structures from soc.h and soc-dapm.h no longer exposed as UAPIs o Moved to component model from codec/platform model. o Added reserved fields in each UAPI structure. o UAPI structures packed and endianess defined for each structure. o Added support for PCM, DAI and codec <-> codec link capabilities. o Added support for client drivers manifests. o Added support for private data for each dynamic object type. o Single list of dynamic objects rather than list of each object type. Liam

4 8

[alsa-devel] [PATCH v2 0/7] ASoC: hda - add ASoC Skylake HD audio driver
by Vinod Koul 22 Apr '15

22 Apr '15

This version incorporates the comments from Takashi & Mark on last RFC series. This also clubs the ASoC HDA wrapper and controller driver to single series. This is based on for ASoC Skylake HD Audio driver and based on Takashi's topic/hda Dropped the based on comments ALSA: hda - add id table support for hdac device/driver ALSA: hda - Add generic helper function to register/unregister driver ALSA: hda - add generic function to unregister all devices on bus Dropped the below patch and add new patch ALSA: hda - add id table support for hdac device/driver Jeeja KP (6): ALSA: hda - add generic functions to set hdac stream params ASoC: hda - add soc hda codec driver wrapper ASoC: hda - add Skylake platform driver ASoC: hda - add Skylake HD audio driver ASoC: hda - enable ASoC Skylake HD audio driver ASoC: hda - added Skylake I2S machine driver Ramesh Babu (1): ALSA: hda - add ASoC device type for hda core include/sound/hdaudio.h | 4 + sound/hda/hdac_stream.c | 31 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/soc-hda-codec.c | 79 ++++ sound/soc/codecs/soc-hda-codec.h | 43 +++ sound/soc/hda/Kconfig | 43 +++ sound/soc/hda/Makefile | 6 + sound/soc/hda/boards/Makefile | 3 + sound/soc/hda/boards/skl_rt286.c | 313 ++++++++++++++++ sound/soc/hda/hda_skl.c | 748 ++++++++++++++++++++++++++++++++++++++ sound/soc/hda/hda_skl.h | 44 +++ sound/soc/hda/hda_skl_pcm.c | 498 +++++++++++++++++++++++++ 15 files changed, 1819 insertions(+) create mode 100644 sound/soc/codecs/soc-hda-codec.c create mode 100644 sound/soc/codecs/soc-hda-codec.h create mode 100644 sound/soc/hda/Kconfig create mode 100644 sound/soc/hda/Makefile create mode 100644 sound/soc/hda/boards/Makefile create mode 100644 sound/soc/hda/boards/skl_rt286.c create mode 100644 sound/soc/hda/hda_skl.c create mode 100644 sound/soc/hda/hda_skl.h create mode 100644 sound/soc/hda/hda_skl_pcm.c -- 1.7.9.5

3 22

[alsa-devel] [PATCH 1/2] ALSA: asound.h - use SNDRV_CTL_ELEM_ID_NAME_MAXLEN
by Vinod Koul 21 Apr '15

21 Apr '15

we have defined SNDRV_CTL_ELEM_ID_NAME_MAXLEN as size of name array so use this define instead of numeric value Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/uapi/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 46145a5277fe..a45be6bdcf5b 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -864,7 +864,7 @@ struct snd_ctl_elem_id { snd_ctl_elem_iface_t iface; /* interface identifier */ unsigned int device; /* device/client number */ unsigned int subdevice; /* subdevice (substream) number */ - unsigned char name[44]; /* ASCII name of item */ + unsigned char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* ASCII name of item */ unsigned int index; /* index of item */ }; -- 1.7.9.5

5 8

[alsa-devel] [PATCH 1/1] usb-audio: Fix audio output on Roland SC-D70 sound module
by Takamichi Horikawa 21 Apr '15

21 Apr '15

Roland SC-D70 reports its device class as vendor specific class and the quirk QUIRK_AUDIO_FIXED_ENDPOINT was used for audio output. In the quirks table the sampling rate was hard-coded to 44100 Hz and therefore not worked when the sound module was in 48000 Hz mode. In this change the quirk is changed to QUIRK_AUDIO_STANDARD_INTERFACE but as the sound module reports incorrect bSubframeSize in its descriptors, additional change is made in format.c to detect it and to override it (which uses the existing code for Edirol SD-90). Tested both when the sound module was in 44100 Hz mode and 48000 Hz mode and both audio input and output. MIDI related part of the driver is not touched. Signed-off-by: Takamichi Horikawa <takamichiho(a)gmail.com> --- sound/usb/format.c | 5 ++++- sound/usb/quirks-table.h | 30 ++---------------------------- 2 files changed, 6 insertions(+), 29 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index 8bcc87c..789d19e 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -79,7 +79,10 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, format = 1 << UAC_FORMAT_TYPE_I_PCM; } if (format & (1 << UAC_FORMAT_TYPE_I_PCM)) { - if (chip->usb_id == USB_ID(0x0582, 0x0016) /* Edirol SD-90 */ && + if (((chip->usb_id == USB_ID(0x0582, 0x0016)) || + /* Edirol SD-90 */ + (chip->usb_id == USB_ID(0x0582, 0x000c))) && + /* Roland SC-D70 */ sample_width == 24 && sample_bytes == 2) sample_bytes = 3; else if (sample_width > sample_bytes * 8) { diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 07f984d..2f6d3e9 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -816,37 +816,11 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { - .formats = SNDRV_PCM_FMTBIT_S24_3LE, - .channels = 2, - .iface = 0, - .altsetting = 1, - .altset_idx = 1, - .attributes = 0, - .endpoint = 0x01, - .ep_attr = 0x01, - .rates = SNDRV_PCM_RATE_CONTINUOUS, - .rate_min = 44100, - .rate_max = 44100, - } + .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { - .formats = SNDRV_PCM_FMTBIT_S24_3LE, - .channels = 2, - .iface = 1, - .altsetting = 1, - .altset_idx = 1, - .attributes = 0, - .endpoint = 0x81, - .ep_attr = 0x01, - .rates = SNDRV_PCM_RATE_CONTINUOUS, - .rate_min = 44100, - .rate_max = 44100, - } + .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 2, -- 2.3.5

2 1

[alsa-devel] Support Dell headset mode for ALC288
by Kailang 21 Apr '15

21 Apr '15

Hi Takashi, Dell has new platform with ALC288. Attach patch is support headset mode for ALC288. Thanks. BR, Kailang

3 10

[alsa-devel] [PATCH] ALSA: hda - only sync BCLK to the display clock for Haswell & Broadwell
by mengdong.lin＠intel.com 21 Apr '15

21 Apr '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Only Intel Haswell and Broadwell have a separate HD-A controller (PCI device 3) for display audio, which needs to get 24MHz HD-A link BCLK from the variable display core clock through vendor specific registers EM4 & EM5. Other platforms (Baytrail, Braswell and Skylake) don't have this feature. So a new driver quirk AZX_DCAPS_I915_SYNC_CLK is defined for the HSW/BDW display HD-A controller, to sync clock after the shared display power is restored. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index be1b7de..1729004 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -175,6 +175,7 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ #define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ +#define AZX_DCAPS_I915_SYNC_CLK (1 << 31) /* Need to sync BCLK with display clock */ enum { AZX_SNOOP_TYPE_NONE, diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e1c2105..951263d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -289,13 +289,13 @@ enum { #define AZX_DCAPS_INTEL_HASWELL \ (/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_COUNT_LPIB_DELAY |\ AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ - AZX_DCAPS_SNOOP_TYPE(SCH)) + AZX_DCAPS_I915_SYNC_CLK | AZX_DCAPS_SNOOP_TYPE(SCH)) /* Broadwell HDMI can't use position buffer reliably, force to use LPIB */ #define AZX_DCAPS_INTEL_BROADWELL \ (/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_POSFIX_LPIB |\ AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ - AZX_DCAPS_SNOOP_TYPE(SCH)) + AZX_DCAPS_I915_SYNC_CLK | AZX_DCAPS_SNOOP_TYPE(SCH)) #define AZX_DCAPS_INTEL_BRASWELL \ (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_I915_POWERWELL) @@ -815,7 +815,8 @@ static int azx_resume(struct device *dev) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { hda_display_power(hda, true); - haswell_set_bclk(hda); + if (chip->driver_caps & AZX_DCAPS_I915_SYNC_CLK) + haswell_set_bclk(hda); } if (chip->msi) if (pci_enable_msi(pci) < 0) @@ -884,7 +885,8 @@ static int azx_runtime_resume(struct device *dev) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { hda_display_power(hda, true); - haswell_set_bclk(hda); + if (chip->driver_caps & AZX_DCAPS_I915_SYNC_CLK) + haswell_set_bclk(hda); } /* Read STATESTS before controller reset */ @@ -1578,7 +1580,7 @@ static int azx_first_init(struct azx *chip) /* initialize chip */ azx_init_pci(chip); - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { + if (chip->driver_caps & AZX_DCAPS_I915_SYNC_CLK) { struct hda_intel *hda; hda = container_of(chip, struct hda_intel, chip); -- 1.9.1

3 8

[alsa-devel] [PATCH v6 0/6] ALSA: jack: Refactoring for jack kctls
by Jie Yang 21 Apr '15

21 Apr '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 new implementation, both HDA and soc jack kctls works fine, they are compatible for old HDA jack kctls, too. For soc, a. snd_jack_new() with NULL jack_kctl; b. for each pin, call snd_jack_add_kctls() to add one kctl(it will call snd_ctl_add() to add kctl to card); For HDA, With snd_jack_new(), we can use phantom_jack = true for phantom jack creating, and the jack kctl pointer will be filled with the new created kcontrol. Changes in v6: 1. refine kctl name generating function; 2. integrate HDA jack kctl more deeply, support phantom jack in snd_jack_new(); 3. add documentation for Jack kcontrols to explain how to use it. Changes in v5: 1. remove SND_KCTL_JACK config item, we need jack kctl once CONFIG_SND_JACK is selected. Changes in v4: 1. use snd_ctl_find_id() to get avaliable index; 2. add initial kctl for snd_jack_new(compatible for HDA); 3. add struct snd_jack_kctl * field to struct hda_jack_tbl; 4. new kctls for soc jack during jack pins creating. 5. add a patch to remove exporting snd_kctl_jack_new(). 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 (6): ALSA: jack: create jack kcontrols for every jack input ALSA: jack: extend snd_jack_new to support phantom jack ALSA: hda - Update to use the new jack kctls method ASoC: jack: create kctls according to jack pins info ALSA: jack: remove exporting ctljack functions ALSA: Docs: Add documentation for Jack kcontrols Documentation/sound/alsa/Jack-Controls.txt | 48 +++++++ include/sound/jack.h | 21 +++- sound/core/Kconfig | 3 - sound/core/Makefile | 3 +- sound/core/ctljack.c | 2 - sound/core/jack.c | 193 +++++++++++++++++++++++++---- sound/pci/hda/Kconfig | 1 - sound/pci/hda/hda_jack.c | 57 ++++----- sound/pci/hda/hda_jack.h | 4 +- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/soc-jack.c | 3 +- 11 files changed, 264 insertions(+), 73 deletions(-) create mode 100644 Documentation/sound/alsa/Jack-Controls.txt -- 1.9.1

4 12

[alsa-devel] [RFC 4/4] ASoC: topology: Add build support for topology.
by Liam Girdwood 21 Apr '15

21 Apr '15

Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/Makefile | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 5b3c8f6..a132768 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,6 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o +snd-soc-core-objs += soc-topology.o ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o -- 2.1.0

2 1

[alsa-devel] [PATCH] ALSA: hda - Mute headphone pin on suspend on XPS13 9333
by Gabriele Mazzotta 20 Apr '15

20 Apr '15

Muting the headphone output pin right before the codec suspension prevents pop noises when headphones are plugged in (except for a barely audible click noise). This solution allows to truly save some power when headphones are plugged in unlike the previous solution (033b0a7ca9c: "ALSA: hda - Pop noises fix for XPS13 9333") Signed-off-by: Gabriele Mazzotta <gabriele.mzt(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 17 +++++++---------- 1 file changed, 7 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b18b9c6..231d0e4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4176,17 +4176,15 @@ static void alc_fixup_disable_aamix(struct hda_codec *codec, } } -static unsigned int alc_power_filter_xps13(struct hda_codec *codec, - hda_nid_t nid, - unsigned int power_state) +static void alc_shutup_dell_xps13(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + int hp_pin = spec->gen.autocfg.hp_pins[0]; - /* Avoid pop noises when headphones are plugged in */ - if (spec->gen.hp_jack_present) - if (nid == codec->core.afg || nid == 0x02 || nid == 0x15) - return AC_PWRST_D0; - return snd_hda_gen_path_power_filter(codec, nid, power_state); + /* Prevent pop noises when headphones are plugged in */ + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + msleep(20); } static void alc_fixup_dell_xps13(struct hda_codec *codec, @@ -4197,8 +4195,7 @@ static void alc_fixup_dell_xps13(struct hda_codec *codec, struct hda_input_mux *imux = &spec->gen.input_mux; int i; - spec->shutup = alc_no_shutup; - codec->power_filter = alc_power_filter_xps13; + spec->shutup = alc_shutup_dell_xps13; /* Make the internal mic the default input source. */ for (i = 0; i < imux->num_items; i++) { -- 2.1.4

2 2

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Use __leXX for little endian data
by Charles Keepax 20 Apr '15

20 Apr '15

Using uXX for little endian data, was triggering some warnings through sparse: sound/soc/codecs/wm_adsp.c:716:26: sparse: cast to restricted __le16 sound/soc/codecs/wm_adsp.c:736:23: sparse: cast to restricted __le16 sound/soc/codecs/wm_adsp.c:739:23: sparse: cast to restricted __le32 Correct this by changing the casts to use __leXX instead of uXX. Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 00289bf..53fc7f8 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -713,7 +713,7 @@ static int wm_coeff_parse_string(int bytes, const u8 **pos, const u8 **str) length = **pos; break; case 2: - length = le16_to_cpu(*((u16 *)*pos)); + length = le16_to_cpu(*((__le16 *)*pos)); break; default: return 0; @@ -733,10 +733,10 @@ static int wm_coeff_parse_int(int bytes, const u8 **pos) switch (bytes) { case 2: - val = le16_to_cpu(*((u16 *)*pos)); + val = le16_to_cpu(*((__le16 *)*pos)); break; case 4: - val = le32_to_cpu(*((u32 *)*pos)); + val = le32_to_cpu(*((__le32 *)*pos)); break; default: break; -- 1.7.2.5

2 2

[alsa-devel] [PATCH resend] ASoC: adau1701: add regulator consumer support
by pascal.huerst＠gmail.com 20 Apr '15

20 Apr '15

From: Pascal Huerst <pascal.huerst(a)gmail.com> The adau1701 has two power domains, DVDD and AVDD. Enable them both as long as the codec is in use. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- .../devicetree/bindings/sound/adi,adau1701.txt | 4 + sound/soc/codecs/adau1701.c | 125 ++++++++++++++++++--- 2 files changed, 114 insertions(+), 15 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/adi,adau1701.txt b/Documentation/devicetree/bindings/sound/adi,adau1701.txt index 547a49b..0d1128c 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau1701.txt +++ b/Documentation/devicetree/bindings/sound/adi,adau1701.txt @@ -20,6 +20,8 @@ Optional properties: pin configurations as described in the datasheet, table 53. Note that the value of this property has to be prefixed with '/bits/ 8'. + - avdd-supply: Power supply for AVDD, providing 3.3V + - dvdd-supply: Power supply for DVDD, providing 3.3V Examples: @@ -28,6 +30,8 @@ Examples: compatible = "adi,adau1701"; reg = <0x34>; reset-gpio = <&gpio 23 0>; + avdd-supply = <&vdd_3v3_reg>; + dvdd-supply = <&vdd_3v3_reg>; adi,pll-mode-gpios = <&gpio 24 0 &gpio 25 0>; adi,pin-config = /bits/ 8 <0x4 0x7 0x5 0x5 0x4 0x4 0x4 0x4 0x4 0x4 0x4 0x4>; diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index d4e219b..ca94ae8 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -16,6 +16,7 @@ #include <linux/of.h> #include <linux/of_gpio.h> #include <linux/of_device.h> +#include <linux/regulator/consumer.h> #include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> @@ -101,6 +102,10 @@ #define ADAU1701_FIRMWARE "adau1701.bin" +static const char * const supply_names[] = { + "dvdd", "avdd" +}; + struct adau1701 { int gpio_nreset; int gpio_pll_mode[2]; @@ -112,6 +117,7 @@ struct adau1701 { u8 pin_config[12]; struct sigmadsp *sigmadsp; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; static const struct snd_kcontrol_new adau1701_controls[] = { @@ -669,6 +675,13 @@ static int adau1701_probe(struct snd_soc_codec *codec) if (ret) return ret; + ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), + adau1701->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + /* * Let the pll_clkdiv variable default to something that won't happen * at runtime. That way, we can postpone the firmware download from @@ -680,7 +693,7 @@ static int adau1701_probe(struct snd_soc_codec *codec) /* initalize with pre-configured pll mode settings */ ret = adau1701_reset(codec, adau1701->pll_clkdiv, 0); if (ret < 0) - return ret; + goto exit_regulators_disable; /* set up pin config */ val = 0; @@ -696,10 +709,60 @@ static int adau1701_probe(struct snd_soc_codec *codec) regmap_write(adau1701->regmap, ADAU1701_PINCONF_1, val); return 0; + +exit_regulators_disable: + + regulator_bulk_disable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); + return ret; } +static int adau1701_remove(struct snd_soc_codec *codec) +{ + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(adau1701->gpio_nreset)) + gpio_set_value_cansleep(adau1701->gpio_nreset, 0); + + regulator_bulk_disable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); + + return 0; +} + +#ifdef CONFIG_PM +static int adau1701_suspend(struct snd_soc_codec *codec) +{ + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + + regulator_bulk_disable(ARRAY_SIZE(adau1701->supplies), + adau1701->supplies); + + return 0; +} + +static int adau1701_resume(struct snd_soc_codec *codec) +{ + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), + adau1701->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + + return adau1701_reset(codec, adau1701->pll_clkdiv, 0); +} +#else +#define adau1701_resume NULL +#define adau1701_suspend NULL +#endif /* CONFIG_PM */ + static struct snd_soc_codec_driver adau1701_codec_drv = { .probe = adau1701_probe, + .remove = adau1701_remove, + .resume = adau1701_resume, + .suspend = adau1701_suspend, .set_bias_level = adau1701_set_bias_level, .idle_bias_off = true, @@ -730,32 +793,58 @@ static int adau1701_i2c_probe(struct i2c_client *client, struct device *dev = &client->dev; int gpio_nreset = -EINVAL; int gpio_pll_mode[2] = { -EINVAL, -EINVAL }; - int ret; + int ret, i; adau1701 = devm_kzalloc(dev, sizeof(*adau1701), GFP_KERNEL); if (!adau1701) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + adau1701->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(adau1701->supplies), + adau1701->supplies); + if (ret < 0) { + dev_err(dev, "Failed to get regulators: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), + adau1701->supplies); + if (ret < 0) { + dev_err(dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + adau1701->client = client; adau1701->regmap = devm_regmap_init(dev, NULL, client, &adau1701_regmap); - if (IS_ERR(adau1701->regmap)) - return PTR_ERR(adau1701->regmap); + if (IS_ERR(adau1701->regmap)) { + ret = PTR_ERR(adau1701->regmap); + goto exit_regulators_disable; + } + if (dev->of_node) { gpio_nreset = of_get_named_gpio(dev->of_node, "reset-gpio", 0); - if (gpio_nreset < 0 && gpio_nreset != -ENOENT) - return gpio_nreset; + if (gpio_nreset < 0 && gpio_nreset != -ENOENT) { + ret = gpio_nreset; + goto exit_regulators_disable; + } gpio_pll_mode[0] = of_get_named_gpio(dev->of_node, "adi,pll-mode-gpios", 0); - if (gpio_pll_mode[0] < 0 && gpio_pll_mode[0] != -ENOENT) - return gpio_pll_mode[0]; + if (gpio_pll_mode[0] < 0 && gpio_pll_mode[0] != -ENOENT) { + ret = gpio_pll_mode[0]; + goto exit_regulators_disable; + } gpio_pll_mode[1] = of_get_named_gpio(dev->of_node, "adi,pll-mode-gpios", 1); - if (gpio_pll_mode[1] < 0 && gpio_pll_mode[1] != -ENOENT) - return gpio_pll_mode[1]; + if (gpio_pll_mode[1] < 0 && gpio_pll_mode[1] != -ENOENT) { + ret = gpio_pll_mode[1]; + goto exit_regulators_disable; + } of_property_read_u32(dev->of_node, "adi,pll-clkdiv", &adau1701->pll_clkdiv); @@ -769,7 +858,7 @@ static int adau1701_i2c_probe(struct i2c_client *client, ret = devm_gpio_request_one(dev, gpio_nreset, GPIOF_OUT_INIT_LOW, "ADAU1701 Reset"); if (ret < 0) - return ret; + goto exit_regulators_disable; } if (gpio_is_valid(gpio_pll_mode[0]) && @@ -778,13 +867,13 @@ static int adau1701_i2c_probe(struct i2c_client *client, GPIOF_OUT_INIT_LOW, "ADAU1701 PLL mode 0"); if (ret < 0) - return ret; + goto exit_regulators_disable; ret = devm_gpio_request_one(dev, gpio_pll_mode[1], GPIOF_OUT_INIT_LOW, "ADAU1701 PLL mode 1"); if (ret < 0) - return ret; + goto exit_regulators_disable; } adau1701->gpio_nreset = gpio_nreset; @@ -795,11 +884,17 @@ static int adau1701_i2c_probe(struct i2c_client *client, adau1701->sigmadsp = devm_sigmadsp_init_i2c(client, &adau1701_sigmadsp_ops, ADAU1701_FIRMWARE); - if (IS_ERR(adau1701->sigmadsp)) - return PTR_ERR(adau1701->sigmadsp); + if (IS_ERR(adau1701->sigmadsp)) { + ret = PTR_ERR(adau1701->sigmadsp); + goto exit_regulators_disable; + } ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv, &adau1701_dai, 1); + +exit_regulators_disable: + + regulator_bulk_disable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); return ret; } -- 1.9.3

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