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

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.

1 0

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.

1 0

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.

1 0

[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

3 7

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.

2 1

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

2 6

[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

2 6

[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

2 1

[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, };

2 1