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[alsa-devel] [PATCH v2] ASoC: davinci-mcasp: Allow complete shutdown of McASP when not in use
by Peter Ujfalusi 05 Mar '15

05 Mar '15

Rearrange the pm_runtime_get/put_sync calls so the IP will be turned off when it is not in use. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Changs since v1: - use om_runtime_put instead of the _sync variant whenever it is feasible - Use pm runtime API to get information about the power state of McASP at supend and use the saved information on resume to do the right thing. Regards, Peter sound/soc/davinci/davinci-mcasp.c | 27 +++++++++++++++++---------- 1 file changed, 17 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 0c882995a357..33cea0728cd2 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -62,6 +62,7 @@ struct davinci_mcasp_context { u32 config_regs[ARRAY_SIZE(context_regs)]; u32 afifo_regs[2]; /* for read/write fifo control registers */ u32 *xrsr_regs; /* for serializer configuration */ + bool pm_state; }; struct davinci_mcasp { @@ -519,7 +520,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); } out: - pm_runtime_put_sync(mcasp->dev); + pm_runtime_put(mcasp->dev); return ret; } @@ -528,6 +529,7 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); + pm_runtime_get_sync(mcasp->dev); switch (div_id) { case 0: /* MCLK divider */ mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, @@ -553,6 +555,7 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, return -EINVAL; } + pm_runtime_put(mcasp->dev); return 0; } @@ -567,6 +570,7 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); + pm_runtime_get_sync(mcasp->dev); if (dir == SND_SOC_CLOCK_OUT) { mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); @@ -579,6 +583,7 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, mcasp->sysclk_freq = freq; + pm_runtime_put(mcasp->dev); return 0; } @@ -1053,6 +1058,10 @@ static int davinci_mcasp_suspend(struct snd_soc_dai *dai) u32 reg; int i; + context->pm_state = pm_runtime_enabled(mcasp->dev) + if (!context->pm_state) + pm_runtime_get_sync(mcasp->dev); + for (i = 0; i < ARRAY_SIZE(context_regs); i++) context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]); @@ -1069,6 +1078,8 @@ static int davinci_mcasp_suspend(struct snd_soc_dai *dai) context->xrsr_regs[i] = mcasp_get_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i)); + pm_runtime_put_sync(mcasp->dev); + return 0; } @@ -1079,6 +1090,8 @@ static int davinci_mcasp_resume(struct snd_soc_dai *dai) u32 reg; int i; + pm_runtime_get_sync(mcasp->dev); + for (i = 0; i < ARRAY_SIZE(context_regs); i++) mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]); @@ -1095,6 +1108,9 @@ static int davinci_mcasp_resume(struct snd_soc_dai *dai) mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), context->xrsr_regs[i]); + if (!context->pm_state) + pm_runtime_put_sync(mcasp->dev); + return 0; } #else @@ -1398,13 +1414,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); - ret = pm_runtime_get_sync(&pdev->dev); - if (IS_ERR_VALUE(ret)) { - dev_err(&pdev->dev, "pm_runtime_get_sync() failed\n"); - pm_runtime_disable(&pdev->dev); - return ret; - } - mcasp->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); if (!mcasp->base) { dev_err(&pdev->dev, "ioremap failed\n"); @@ -1584,14 +1593,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) return 0; err: - pm_runtime_put_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); return ret; } static int davinci_mcasp_remove(struct platform_device *pdev) { - pm_runtime_put_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); return 0; -- 2.3.0

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[alsa-devel] [PATCH] SoC: Fix MAX98357A codec driver dependencies
by Guenter Roeck 05 Mar '15

05 Mar '15

The max98357a driver depends on GPIOLIB. This may cause the following build failure. sound/soc/codecs/max98357a.c: In function 'max98357a_daiops_trigger': sound/soc/codecs/max98357a.c:30:3: error: implicit declaration of function 'gpiod_set_value' sound/soc/codecs/max98357a.c: In function 'max98357a_codec_probe': sound/soc/codecs/max98357a.c:55:2: error: implicit declaration of function 'devm_gpiod_get' sound/soc/codecs/max98357a.c:61:2: error: implicit declaration of function 'gpiod_direction_output' Seen with mips:allmodconfig as well as various randconfig builds. Fixes: af5adf129369 ("ASoC: max98357a: Add MAX98357A codec driver") Cc: Kenneth Westfield <kwestfie(a)codeaurora.org> Signed-off-by: Guenter Roeck <linux(a)roeck-us.net> --- 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 064e6c1..ea9f0e3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -69,7 +69,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C - select SND_SOC_MAX98357A + select SND_SOC_MAX98357A if GPIOLIB select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C -- 2.1.0

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[alsa-devel] [PATCH] ALSA: usb-audio: Check Marantz/Denon USB DACs in a single place
by Takashi Iwai 05 Mar '15

05 Mar '15

There are three places doing the same check. Let's make them together. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 40 +++++++++++++++++----------------------- 1 file changed, 17 insertions(+), 23 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 753a47de8459..353532b8aee4 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1120,17 +1120,24 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) /* Marantz/Denon USB DACs need a vendor cmd to switch * between PCM and native DSD mode */ +static bool is_marantz_denon_dac(unsigned int id) +{ + switch (id) { + case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */ + case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ + case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ + return true; + } + return false; +} + int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, struct audioformat *fmt) { struct usb_device *dev = subs->dev; int err; - switch (subs->stream->chip->usb_id) { - case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */ - case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ - case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ - + if (is_marantz_denon_dac(subs->stream->chip->usb_id)) { /* First switch to alt set 0, otherwise the mode switch cmd * will not be accepted by the DAC */ @@ -1203,17 +1210,10 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, /* Marantz/Denon devices with USB DAC functionality need a delay * after each class compliant request */ - if ((le16_to_cpu(dev->descriptor.idVendor) == 0x154e) && - (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) { - - switch (le16_to_cpu(dev->descriptor.idProduct)) { - case 0x1003: /* Denon DA300-USB */ - case 0x3005: /* Marantz HD-DAC1 */ - case 0x3006: /* Marantz SA-14S1 */ - mdelay(20); - break; - } - } + if (is_marantz_denon_dac(USB_ID(le16_to_cpu(dev->descriptor.idVendor), + le16_to_cpu(dev->descriptor.idProduct))) + && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + mdelay(20); /* Zoom R16/24 needs a tiny delay here, otherwise requests like * get/set frequency return as failed despite actually succeeding. @@ -1268,15 +1268,9 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, } /* Denon/Marantz devices with USB DAC functionality */ - switch (chip->usb_id) { - case USB_ID(0x154e, 0x1003): /* Denon DA300-USB */ - case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ - case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ + if (is_marantz_denon_dac(chip->usb_id)) { if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; - break; - default: - break; } return 0; -- 2.3.1

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[alsa-devel] [PATCH v2 0/2] snd_use_case_get() documentation improvements
by Tanu Kaskinen 05 Mar '15

05 Mar '15

Tanu Kaskinen (2): ucm: reformat snd_use_case_get() doc ucm: improve jack configuration documentation include/use-case.h | 123 +++++++++++++++++++++++++++++++++-------------------- 1 file changed, 77 insertions(+), 46 deletions(-) -- 1.9.3

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[alsa-devel] [PATCH] ASoC: fsl_ssi: Don't try to round-up for PM divisor calculation
by Nicolin Chen 05 Mar '15

05 Mar '15

According to i.MX6 Series Reference Manual, the formula to calculate the sys clock is sysclk rate = bclk rate * (div2 + 1) * (7 * psr + 1) * (pm + 1) * 2 Commit aafa85e71a75 ("ASoC: fsl_ssi: Add DAI master mode support for SSI on i.MX series") added the divisor calculation which relies on the clk_round_rate(). However, at that time, clk_round_rate() didn't provide closest clock rates for some cases because it might not use a correct rounding policy. So using the original formula (pm + 1) for PM divisor was not able to give us a desired clock rate. And then we used (pm + 2) to do the trick. However, the clk-divider driver has been refined a lot since commit b11d282dbea2 ("clk: divider: fix rate calculation for fractional rates") Now using (pm + 2) trick would result an incorrect clock rate. So this patch fixes the problem by removing the useless trick. Reported-by: Stephane Cerveau <scerveau(a)voxtok.com> Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 05575f4..6b0c8f7 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -603,7 +603,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, factor = (div2 + 1) * (7 * psr + 1) * 2; for (i = 0; i < 255; i++) { - tmprate = freq * factor * (i + 2); + tmprate = freq * factor * (i + 1); if (baudclk_is_used) clkrate = clk_get_rate(ssi_private->baudclk); -- 1.9.1

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[alsa-devel] [PATCH] ALSA: line6: Clamp values correctly
by Takashi Iwai 05 Mar '15

05 Mar '15

The usages of clamp() macro in sound/usb/line6/playback.c are just wrong, the low and high values are swapped. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/line6/playback.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/line6/playback.c b/sound/usb/line6/playback.c index 05dee690f487..97ed593f6010 100644 --- a/sound/usb/line6/playback.c +++ b/sound/usb/line6/playback.c @@ -39,7 +39,7 @@ static void change_volume(struct urb *urb_out, int volume[], for (; p < buf_end; ++p) { short pv = le16_to_cpu(*p); int val = (pv * volume[chn & 1]) >> 8; - pv = clamp(val, 0x7fff, -0x8000); + pv = clamp(val, -0x8000, 0x7fff); *p = cpu_to_le16(pv); ++chn; } @@ -54,7 +54,7 @@ static void change_volume(struct urb *urb_out, int volume[], val = p[0] + (p[1] << 8) + ((signed char)p[2] << 16); val = (val * volume[chn & 1]) >> 8; - val = clamp(val, 0x7fffff, -0x800000); + val = clamp(val, -0x800000, 0x7fffff); p[0] = val; p[1] = val >> 8; p[2] = val >> 16; @@ -126,7 +126,7 @@ static void add_monitor_signal(struct urb *urb_out, unsigned char *signal, short pov = le16_to_cpu(*po); short piv = le16_to_cpu(*pi); int val = pov + ((piv * volume) >> 8); - pov = clamp(val, 0x7fff, -0x8000); + pov = clamp(val, -0x8000, 0x7fff); *po = cpu_to_le16(pov); } } -- 2.3.1

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[alsa-devel] [patch] ALSA: msnd: add some missing curly braces
by Dan Carpenter 05 Mar '15

05 Mar '15

There were some curly braces intended here. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c index 17e49a0..b408540 100644 --- a/sound/isa/msnd/msnd_pinnacle_mixer.c +++ b/sound/isa/msnd/msnd_pinnacle_mixer.c @@ -306,11 +306,12 @@ int snd_msndmix_new(struct snd_card *card) spin_lock_init(&chip->mixer_lock); strcpy(card->mixername, "MSND Pinnacle Mixer"); - for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++) + for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++) { err = snd_ctl_add(card, snd_ctl_new1(snd_msnd_controls + idx, chip)); if (err < 0) return err; + } return 0; }

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Re: [alsa-devel] ALSA: line6: Fix volume calculation for big-endian
by Dan Carpenter 05 Mar '15

05 Mar '15

Hello Takashi Iwai, The patch 0416980d0a2b: "ALSA: line6: Fix volume calculation for big-endian" from Jan 28, 2015, leads to the following static checker warning: sound/usb/line6/playback.c:42 change_volume() warn: always clamps to s16min sound/usb/line6/playback.c:57 change_volume() warn: always clamps to (-8388608) sound/usb/line6/playback.c:129 add_monitor_signal() warn: always clamps to s16min sound/usb/line6/playback.c 25 static void change_volume(struct urb *urb_out, int volume[], 26 int bytes_per_frame) 27 { 28 int chn = 0; 29 30 if (volume[0] == 256 && volume[1] == 256) 31 return; /* maximum volume - no change */ 32 33 if (bytes_per_frame == 4) { 34 __le16 *p, *buf_end; 35 36 p = (__le16 *)urb_out->transfer_buffer; 37 buf_end = p + urb_out->transfer_buffer_length / sizeof(*p); 38 39 for (; p < buf_end; ++p) { 40 short pv = le16_to_cpu(*p); 41 int val = (pv * volume[chn & 1]) >> 8; 42 pv = clamp(val, 0x7fff, -0x8000); ^^^^^^^^^^^^^^^^ You didn't really add this, but you might know what was intended here. It is a complete mystery to me. :) 43 *p = cpu_to_le16(pv); 44 ++chn; 45 } 46 } else if (bytes_per_frame == 6) { 47 unsigned char *p, *buf_end; 48 49 p = (unsigned char *)urb_out->transfer_buffer; 50 buf_end = p + urb_out->transfer_buffer_length; 51 52 for (; p < buf_end; p += 3) { 53 int val; 54 55 val = p[0] + (p[1] << 8) + ((signed char)p[2] << 16); 56 val = (val * volume[chn & 1]) >> 8; 57 val = clamp(val, 0x7fffff, -0x800000); ^^^^^^^^^^^^^^^^^^^^ 58 p[0] = val; 59 p[1] = val >> 8; 60 p[2] = val >> 16; 61 ++chn; 62 } 63 } 64 } [snip] 127 short piv = le16_to_cpu(*pi); 128 int val = pov + ((piv * volume) >> 8); 129 pov = clamp(val, 0x7fff, -0x8000); ^^^^^^^^^^^^^^^^ 130 *po = cpu_to_le16(pov); 131 } regards, dan carpenter

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[alsa-devel] [PATCH] ucm: improve jack configuration documentation
by Tanu Kaskinen 05 Mar '15

05 Mar '15

Signed-off-by: Tanu Kaskinen <tanu.kaskinen(a)linux.intel.com> --- I worked with Han Lu on enabling jack detection in PulseAudio with UCM on Broadwell-rt286. Due to less-than-perfect coordination, we both independently wrote documentation for the new UCM values. Han sent his patch first, and it got applied already, but my patch has more detailed documentation, so it might make sense to replace the documentation with my version. include/use-case.h | 26 ++++++++++++++++++++++---- 1 file changed, 22 insertions(+), 4 deletions(-) diff --git a/include/use-case.h b/include/use-case.h index 3728906..6c1fe70 100644 --- a/include/use-case.h +++ b/include/use-case.h @@ -284,10 +284,28 @@ int snd_use_case_get_list(snd_use_case_mgr_t *uc_mgr, * PlaybackMixerID - mixer playback ID * CaptureMixer - name of capture mixer * CaptureMixerID - mixer capture ID - * JackDev - jack device name - * JackControl - jack control name - * JackHWMute - indicate if the HW mutes a device on jack insertion - * or not. + * JackControl, + * JackDev, + * JackHWMute - Jack information for a device. The jack status can be + * reported via a kcontrol and/or via an input device. + * JackControl is the kcontrol name of the jack, and + * JackDev is the input device id of the jack (if the + * full input device path is /dev/input/by-id/foo, the + * JackDev value should be "foo"). UCM configuration + * files should contain both JackControl and JackDev + * when possible, because applications are likely to + * support only one or the other. If JackHWMute is set, + * it indicates that when the jack is plugged in, the + * hardware automatically mutes some other device. The + * JackHWMute value is the name of the muted device. + * Note that JackHWMute should be used only when the + * hardware enforces the automatic muting. If the + * hardware doesn't enforce any muting, it may still be + * tempting to set JackHWMute to trick upper software + * layers to e.g. automatically mute speakers when + * headphones are plugged in, but that's application + * policy configuration that doesn't belong to UCM + * configuration files. */ int snd_use_case_get(snd_use_case_mgr_t *uc_mgr, const char *identifier, -- 1.9.3

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[alsa-devel] [PATCH] ALSA: hda - Fix wrong detection of "Headphone+LO" or "Speaker+LO"
by Ingo Brückl 05 Mar '15

05 Mar '15

Add the constraint mentioned in the comment. It does not apply to a scenario with three DACs and multi-IO where we would normally get a "Master Playback Volume" and a "Front Playback Volume" (prevented without checking for num_dacs). Signed-off-by: Ingo Brückl <ib(a)wupperonline.de> diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index ecee349..4c8910c 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1097,7 +1097,7 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, case AUTO_PIN_LINE_OUT: /* This deals with the case where we have two DACs and * one LO, one HP and one Speaker */ - if (!ch && cfg->speaker_outs && cfg->hp_outs) { + if (!ch && spec->multiout.num_dacs == 2 && cfg->speaker_outs && cfg->hp_outs) { bool hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); bool spk_lo_shared = !path_has_mixer(codec, spec->speaker_paths[0], ctl_type); if (hp_lo_shared && spk_lo_shared) -- 1.7.10

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Allow complete shutdown of McASP when not in use
by Peter Ujfalusi 05 Mar '15

05 Mar '15

Rearrange the pm_runtime_get/put_sync calls so the IP will be turned off when it is not in use. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 23 ++++++++++++++--------- 1 file changed, 14 insertions(+), 9 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 031c1fb44ae7..a51197ac0b92 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -528,6 +528,7 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); + pm_runtime_get_sync(mcasp->dev); switch (div_id) { case 0: /* MCLK divider */ mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, @@ -553,6 +554,7 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, return -EINVAL; } + pm_runtime_put_sync(mcasp->dev); return 0; } @@ -567,6 +569,7 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); + pm_runtime_get_sync(mcasp->dev); if (dir == SND_SOC_CLOCK_OUT) { mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); @@ -579,6 +582,7 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, mcasp->sysclk_freq = freq; + pm_runtime_put_sync(mcasp->dev); return 0; } @@ -1078,6 +1082,9 @@ static int davinci_mcasp_suspend(struct snd_soc_dai *dai) u32 reg; int i; + if (!dai->active) + pm_runtime_get_sync(mcasp->dev); + for (i = 0; i < ARRAY_SIZE(context_regs); i++) context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]); @@ -1094,6 +1101,8 @@ static int davinci_mcasp_suspend(struct snd_soc_dai *dai) context->xrsr_regs[i] = mcasp_get_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i)); + pm_runtime_put_sync(mcasp->dev); + return 0; } @@ -1104,6 +1113,8 @@ static int davinci_mcasp_resume(struct snd_soc_dai *dai) u32 reg; int i; + pm_runtime_get_sync(mcasp->dev); + for (i = 0; i < ARRAY_SIZE(context_regs); i++) mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]); @@ -1120,6 +1131,9 @@ static int davinci_mcasp_resume(struct snd_soc_dai *dai) mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), context->xrsr_regs[i]); + if (!dai->active) + pm_runtime_put_sync(mcasp->dev); + return 0; } #else @@ -1427,13 +1441,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); - ret = pm_runtime_get_sync(&pdev->dev); - if (IS_ERR_VALUE(ret)) { - dev_err(&pdev->dev, "pm_runtime_get_sync() failed\n"); - pm_runtime_disable(&pdev->dev); - return ret; - } - mcasp->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); if (!mcasp->base) { dev_err(&pdev->dev, "ioremap failed\n"); @@ -1633,14 +1640,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) return 0; err: - pm_runtime_put_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); return ret; } static int davinci_mcasp_remove(struct platform_device *pdev) { - pm_runtime_put_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); return 0; -- 2.3.0

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[alsa-devel] Using simple-audio-card with imx6sl-evk
by Fabio Estevam 05 Mar '15

05 Mar '15

Hi, I am trying to get imx6sl-evk to work with simple-audio-card with the following changes: arch/arm/boot/dts/imx6sl-evk.dts | 52 ++++++++++++++++++++++++++++++---------- 1 file changed, 40 insertions(+), 12 deletions(-) diff --git a/arch/arm/boot/dts/imx6sl-evk.dts b/arch/arm/boot/dts/imx6sl-evk.dts index fda4932..24b9849 100644 --- a/arch/arm/boot/dts/imx6sl-evk.dts +++ b/arch/arm/boot/dts/imx6sl-evk.dts @@ -10,6 +10,7 @@ #include <dt-bindings/gpio/gpio.h> #include <dt-bindings/input/input.h> +#include <dt-bindings/sound/fsl-imx-audmux.h> #include "imx6sl.dtsi" / { @@ -92,19 +93,25 @@ }; sound { - compatible = "fsl,imx6sl-evk-wm8962", "fsl,imx-audio-wm8962"; - model = "wm8962-audio"; - ssi-controller = <&ssi2>; - audio-codec = <&codec>; - audio-routing = + compatible = "simple-audio-card"; + simple-audio-card,name = "wm8962-audio"; + simple-audio-card,format = "i2s"; + simple-audio-card,bitclock-master = <&dailink_master>; + simple-audio-card,frame-master = <&dailink_master>; + simple-audio-card,widgets = + "Headphone", "Headphone Jack"; + simple-audio-card,routing = "Headphone Jack", "HPOUTL", - "Headphone Jack", "HPOUTR", - "Ext Spk", "SPKOUTL", - "Ext Spk", "SPKOUTR", - "AMIC", "MICBIAS", - "IN3R", "AMIC"; - mux-int-port = <2>; - mux-ext-port = <3>; + "Headphone Jack", "HPOUTR"; + + simple-audio-card,cpu { + sound-dai = <&ssi2>; + }; + + dailink_master: simple-audio-card,codec { + sound-dai = <&codec>; + }; }; }; @@ -112,6 +119,26 @@ pinctrl-names = "default"; pinctrl-0 = <&pinctrl_audmux3>; status = "okay"; + + ssi2 { + fsl,audmux-port = <1>; + fsl,port-config = < + (IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TFSEL(2) | + IMX_AUDMUX_V2_PTCR_TCLKDIR | + IMX_AUDMUX_V2_PTCR_TCSEL(2)) + IMX_AUDMUX_V2_PDCR_RXDSEL(2) + >; + }; + + audmuxpins3 { + fsl,audmux-port = <2>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN + IMX_AUDMUX_V2_PDCR_RXDSEL(1) + >; + }; }; &ecspi1 { @@ -255,6 +282,7 @@ compatible = "wlf,wm8962"; reg = <0x1a>; clocks = <&clks IMX6SL_CLK_EXTERN_AUDIO>; + #sound-dai-cells = <0>; DCVDD-supply = <&vgen3_reg>; DBVDD-supply = <&reg_aud3v>; AVDD-supply = <&vgen3_reg>; Then I get the following probe error: [ 2.030249] input: WM8962 Beep Generator as /devices/soc0/soc/2100000.aips-bus/21a4000.i2c/i2c-1/1-001a/input/input1 [ 2.042388] wm8962 1-001a: simple-card: set_sysclk error [ 2.047805] asoc-simple-card sound: ASoC: failed to init 202c000.ssi-wm8962: -22 [ 2.055242] asoc-simple-card sound: ASoC: failed to instantiate card -22 [ 2.065117] asoc-simple-card: probe of sound failed with error -22 I compared how imx-wm8962 calls snd_soc_dai_set_sysclk and did the same on simple card (quick hack): --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -134,7 +134,7 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, } if (set->sysclk) { - ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0); + ret = snd_soc_dai_set_sysclk(dai, 1, set->sysclk, 0); if (ret && ret != -ENOTSUPP) { dev_err(dai->dev, "simple-card: set_sysclk error\n"); goto err; ,and this allows the probe to succeed. When I try to play I get: root@freescale /home$ aplay clarinet.wav [ 16.424340] wm8962 1-001a: Unsupported sysclk ratio 680 Playing WAVE 'clarinet.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono and the track is played with the wrong frequency and wrong sysclk ratio. Any suggestions as to how to use simple-card on this board? Thanks, Fabio Estevam

2 1

[alsa-devel] [PATCH 0/5] ASoC: davinci: Use edma-pcm and remove davinci-pcm
by Peter Ujfalusi 05 Mar '15

05 Mar '15

Hi, This series will deprecate and removes the davinci-pcm platform driver and converts the daVinci DAI drivers to use the edma-pcm. The main feature of davinci-pcm has been the so called ping-pong mode, which can help in situations when the McASP/McBSP/ASP/VCIF experiences underrun or overrun in playback or capture. This was due to the fact the davinci-pcm w/o ping-pong needed to reprogram the paRAM slot after each period, this could cause delay in DMA operation, which leads to starvation. The edm-pcm does not need reprogramming and it seams to be working as good as the davinci-pcm with ping-pong. I have tested this series on OMAP-L138 EVM with McASP and McBSP. VCIF has been only compile tested since I do not have access to the HW. The edma-pcm is already used by AM335x and AM437x and it has been tested on those platforms extensively. After this series we no longer have legacy eDMA users which clears the path for the eDMA driver stack cleanup. Regards, Peter --- Peter Ujfalusi (5): ASoC: davinci: Select SND_EDMA_SOC when SND_DAVINCI_SOC is enabled ASoC: davinci-i2s: Convert to use edma-pcm ASoC: davinci-vcif: Convert to use edma-pcm ASoC: davinci-mcasp: Deprecate the use of davinci-pcm in favor of edma-pcm ASoC: davinci: Remove unused davinci-pcm platform driver sound/soc/davinci/Kconfig | 18 +- sound/soc/davinci/Makefile | 2 - sound/soc/davinci/davinci-i2s.c | 67 ++- sound/soc/davinci/davinci-mcasp.c | 87 +--- sound/soc/davinci/davinci-pcm.c | 861 -------------------------------------- sound/soc/davinci/davinci-pcm.h | 41 -- sound/soc/davinci/davinci-vcif.c | 55 ++- 7 files changed, 79 insertions(+), 1052 deletions(-) delete mode 100644 sound/soc/davinci/davinci-pcm.c delete mode 100644 sound/soc/davinci/davinci-pcm.h -- 2.3.0

3 13

[alsa-devel] [ALSA-LIB][PATCH] Add support for compress offload API
by Qais Yousef 05 Mar '15

05 Mar '15

Merge tinycompress library into alsa-lib using LGPL license. Only function names were modified to match the coding style in alsa-lib, prepending snd_compr to function names and structs. Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Mark Brown <broonie(a)kernel.org> --- configure.ac | 9 + include/Makefile.am | 4 + include/compress.h | 245 ++++++++++++++++ include/sound/compress_offload.h | 191 +++++++++++++ include/sound/compress_params.h | 404 ++++++++++++++++++++++++++ src/Makefile.am | 7 + src/compress/Makefile.am | 8 + src/compress/compress.c | 599 +++++++++++++++++++++++++++++++++++++++ 8 files changed, 1467 insertions(+) create mode 100644 include/compress.h create mode 100644 include/sound/compress_offload.h create mode 100644 include/sound/compress_params.h create mode 100644 src/compress/Makefile.am create mode 100644 src/compress/compress.c diff --git a/configure.ac b/configure.ac index f0995e3ae787..a768730781e0 100644 --- a/configure.ac +++ b/configure.ac @@ -368,6 +368,9 @@ AC_ARG_ENABLE(mixer, AC_ARG_ENABLE(pcm, AS_HELP_STRING([--disable-pcm], [disable the PCM component]), [build_pcm="$enableval"], [build_pcm="yes"]) +AC_ARG_ENABLE(compress, + AS_HELP_STRING([--disable-compress], [disable the compress component]), + [build_compress="$enableval"], [build_compress="yes"]) AC_ARG_ENABLE(rawmidi, AS_HELP_STRING([--disable-rawmidi], [disable the raw MIDI component]), [build_rawmidi="$enableval"], [build_rawmidi="yes"]) @@ -418,6 +421,7 @@ AC_SUBST(PYTHON_INCLUDES) AM_CONDITIONAL([BUILD_MIXER], [test x$build_mixer = xyes]) AM_CONDITIONAL([BUILD_PCM], [test x$build_pcm = xyes]) +AM_CONDITIONAL([BUILD_COMPRESS], [test x$build_compress = xyes]) AM_CONDITIONAL([BUILD_RAWMIDI], [test x$build_rawmidi = xyes]) AM_CONDITIONAL([BUILD_HWDEP], [test x$build_hwdep = xyes]) AM_CONDITIONAL([BUILD_SEQ], [test x$build_seq = xyes]) @@ -431,6 +435,9 @@ fi if test "$build_pcm" = "yes"; then AC_DEFINE([BUILD_PCM], "1", [Build PCM component]) fi +if test "$build_compress" = "yes"; then + AC_DEFINE([BUILD_COMPRESS], "1", [Build compress component]) +fi if test "$build_rawmidi" = "yes"; then AC_DEFINE([BUILD_RAWMIDI], "1", [Build raw MIDI component]) fi @@ -641,6 +648,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ include/Makefile include/sound/Makefile src/Versions src/Makefile \ src/control/Makefile src/mixer/Makefile \ src/pcm/Makefile src/pcm/scopes/Makefile \ + src/compress/Makefile \ src/rawmidi/Makefile src/timer/Makefile \ src/hwdep/Makefile src/seq/Makefile src/ucm/Makefile \ src/alisp/Makefile \ @@ -693,6 +701,7 @@ cat >> include/asoundlib.h <<EOF #include <alsa/conf.h> EOF test "$build_pcm" = "yes" && echo "#include <alsa/pcm.h>" >> include/asoundlib.h +test "$build_compress" = "yes" && echo "#include <alsa/compress.h>" >> include/asoundlib.h test "$build_rawmidi" = "yes" && echo "#include <alsa/rawmidi.h>" >> include/asoundlib.h test "$build_pcm" = "yes" && echo "#include <alsa/timer.h>" >> include/asoundlib.h test "$build_hwdep" = "yes" && echo "#include <alsa/hwdep.h>" >> include/asoundlib.h diff --git a/include/Makefile.am b/include/Makefile.am index 4baa03af69e1..395a2ed60d70 100644 --- a/include/Makefile.am +++ b/include/Makefile.am @@ -30,6 +30,10 @@ alsainclude_HEADERS += pcm_ioplug.h endif endif +if BUILD_COMPRESS +alsainclude_HEADERS += compress.h +endif + if BUILD_RAWMIDI alsainclude_HEADERS += rawmidi.h endif diff --git a/include/compress.h b/include/compress.h new file mode 100644 index 000000000000..250ce0c3f7c4 --- /dev/null +++ b/include/compress.h @@ -0,0 +1,245 @@ +/* + * tinycompress library for compress audio offload in alsa + * Copyright (c) 2011-2012, Intel Corporation. + * + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU Lesser General Public License, + * version 2.1, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public + * License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to + * the Free Software Foundation, Inc., + * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + */ + + +#ifndef __ALSA_COMPRESS_H +#define __ALSA_COMPRESS_H + +#include <stdbool.h> +#include <linux/types.h> + +#if defined(__cplusplus) +extern "C" { +#endif +/* + * struct compr_config: config structure, needs to be filled by app + * If fragment_size or fragments are zero, this means "don't care" + * and tinycompress will choose values that the driver supports + * + * @fragment_size: size of fragment requested, in bytes + * @fragments: number of fragments + * @codec: codec type and parameters requested + */ +struct snd_compr_config { + __u32 fragment_size; + __u32 fragments; + struct snd_codec *codec; +}; + +struct snd_compr_gapless_mdata { + __u32 encoder_delay; + __u32 encoder_padding; +}; + +#define COMPRESS_OUT 0x20000000 +#define COMPRESS_IN 0x10000000 + +struct snd_compr; +struct snd_compr_tstamp; + +/* + * snd_compr_open: open a new compress stream + * returns the valid struct snd_compr on success, NULL on failure + * If config does not specify a requested fragment size, on return + * it will be updated with the size and number of fragments that + * were configured + * + * @card: sound card number + * @device: device number + * @flags: device flags can be COMPRESS_OUT or COMPRESS_IN + * @config: stream config requested. Returns actual fragment config + */ +struct snd_compr *snd_compr_open(unsigned int card, unsigned int device, + unsigned int flags, struct snd_compr_config *config); + +/* + * snd_compr_close: close the compress stream + * + * @compr: compress stream to be closed + */ +void snd_compr_close(struct snd_compr *compr); + +/* + * snd_compr_get_hpointer: get the hw timestamp + * return 0 on success, negative on error + * + * @compr: compress stream on which query is made + * @avail: buffer availble for write/read, in bytes + * @tstamp: hw time + */ +int snd_compr_get_hpointer(struct snd_compr *compr, + unsigned int *avail, struct timespec *tstamp); + + +/* + * snd_compr_get_tstamp: get the raw hw timestamp + * return 0 on success, negative on error + * + * @compr: compress stream on which query is made + * @samples: number of decoded samples played + * @sampling_rate: sampling rate of decoded samples + */ +int snd_compr_get_tstamp(struct snd_compr *compr, + unsigned int *samples, unsigned int *sampling_rate); + +/* + * snd_compr_write: write data to the compress stream + * return bytes written on success, negative on error + * By default this is a blocking call and will not return + * until all bytes have been written or there was a + * write error. + * If non-blocking mode has been enabled with snd_compr_nonblock(), + * this function will write all bytes that can be written without + * blocking and will then return the number of bytes successfully + * written. If the return value is not an error and is < size + * the caller can use snd_compr_wait() to block until the driver + * is ready for more data. + * + * @compr: compress stream to be written to + * @buf: pointer to data + * @size: number of bytes to be written + */ +int snd_compr_write(struct snd_compr *compr, const void *buf, unsigned int size); + +/* + * snd_compr_read: read data from the compress stream + * return bytes read on success, negative on error + * By default this is a blocking call and will block until + * size bytes have been written or there was a read error. + * If non-blocking mode was enabled using snd_compr_nonblock() + * the behaviour will change to read only as many bytes as + * are currently available (if no bytes are available it + * will return immediately). The caller can then use + * snd_compr_wait() to block until more bytes are available. + * + * @compr: compress stream from where data is to be read + * @buf: pointer to data buffer + * @size: size of given buffer + */ +int snd_compr_read(struct snd_compr *compr, void *buf, unsigned int size); + +/* + * snd_compr_start: start the compress stream + * return 0 on success, negative on error + * + * @compr: compress stream to be started + */ +int snd_compr_start(struct snd_compr *compr); + +/* + * snd_compr_stop: stop the compress stream + * return 0 on success, negative on error + * + * @compr: compress stream to be stopped + */ +int snd_compr_stop(struct snd_compr *compr); + +/* + * snd_compr_pause: pause the compress stream + * return 0 on success, negative on error + * + * @compr: compress stream to be paused + */ +int snd_compr_pause(struct snd_compr *compr); + +/* + * snd_compr_resume: resume the compress stream + * return 0 on success, negative on error + * + * @compr: compress stream to be resumed + */ +int snd_compr_resume(struct snd_compr *compr); + +/* + * snd_compr_drain: drain the compress stream + * return 0 on success, negative on error + * + * @compr: compress stream to be drain + */ +int snd_compr_drain(struct snd_compr *compr); + +/* + * snd_compr_next_track: set the next track for stream + * + * return 0 on success, negative on error + * + * @compr: compress stream to be transistioned to next track + */ +int snd_compr_next_track(struct snd_compr *compr); + +/* + * snd_compr_partial_drain: drain will return after the last frame is decoded + * by DSP and will play the , All the data written into compressed + * ring buffer is decoded + * + * return 0 on success, negative on error + * + * @compr: compress stream to be drain + */ +int snd_compr_partial_drain(struct snd_compr *compr); + +/* + * snd_compr_set_gapless_metadata: set gapless metadata of a compress strem + * + * return 0 on success, negative on error + * + * @compr: compress stream for which metadata has to set + * @mdata: metadata encoder delay and padding + */ + +int snd_compr_set_gapless_metadata(struct snd_compr *compr, + struct snd_compr_gapless_mdata *mdata); + +/* + * snd_compr_is_codec_supported: check if the given codec is supported + * returns true when supported, false if not + * + * @card: sound card number + * @device: device number + * @flags: stream flags + * @codec: codec type and parameters to be checked + */ +bool snd_compr_is_codec_supported(unsigned int card, unsigned int device, + unsigned int flags, struct snd_codec *codec); + +/* + * snd_compr_set_max_poll_wait: set the maximum time tinycompress + * will wait for driver to signal a poll(). Interval is in + * milliseconds. + * Pass interval of -1 to disable timeout and make poll() wait + * until driver signals. + * If this function is not used the timeout defaults to 20 seconds. + */ +void snd_compr_set_max_poll_wait(struct snd_compr *compr, int milliseconds); + +/* Enable or disable non-blocking mode for write and read */ +void snd_compr_nonblock(struct snd_compr *compr, int nonblock); + +/* Wait for ring buffer to ready for next read or write */ +int snd_compr_wait(struct snd_compr *compr, int timeout_ms); + +int snd_compr_is_running(struct snd_compr *compr); + +int snd_compr_is_ready(struct snd_compr *compr); + +/* Returns a human readable reason for the last error */ +const char *snd_compr_get_error(struct snd_compr *compr); + +#endif diff --git a/include/sound/compress_offload.h b/include/sound/compress_offload.h new file mode 100644 index 000000000000..22ed8cb7800b --- /dev/null +++ b/include/sound/compress_offload.h @@ -0,0 +1,191 @@ +/* + * compress_offload.h - compress offload header definations + * + * Copyright (C) 2011 Intel Corporation + * Authors: Vinod Koul <vinod.koul(a)linux.intel.com> + * Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.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 + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ +#ifndef __COMPRESS_OFFLOAD_H +#define __COMPRESS_OFFLOAD_H + +#include <linux/types.h> +#include <sound/asound.h> +#include <sound/compress_params.h> + + +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 2) +/** + * struct snd_compressed_buffer - compressed buffer + * @fragment_size: size of buffer fragment in bytes + * @fragments: number of such fragments + */ +struct snd_compressed_buffer { + __u32 fragment_size; + __u32 fragments; +} __attribute__((packed, aligned(4))); + +/** + * struct snd_compr_params - compressed stream params + * @buffer: buffer description + * @codec: codec parameters + * @no_wake_mode: dont wake on fragment elapsed + */ +struct snd_compr_params { + struct snd_compressed_buffer buffer; + struct snd_codec codec; + __u8 no_wake_mode; +} __attribute__((packed, aligned(4))); + +/** + * struct snd_compr_tstamp - timestamp descriptor + * @byte_offset: Byte offset in ring buffer to DSP + * @copied_total: Total number of bytes copied from/to ring buffer to/by DSP + * @pcm_frames: Frames decoded or encoded by DSP. This field will evolve by + * large steps and should only be used to monitor encoding/decoding + * progress. It shall not be used for timing estimates. + * @pcm_io_frames: Frames rendered or received by DSP into a mixer or an audio + * output/input. This field should be used for A/V sync or time estimates. + * @sampling_rate: sampling rate of audio + */ +struct snd_compr_tstamp { + __u32 byte_offset; + __u32 copied_total; + __u32 pcm_frames; + __u32 pcm_io_frames; + __u32 sampling_rate; +} __attribute__((packed, aligned(4))); + +/** + * struct snd_compr_avail - avail descriptor + * @avail: Number of bytes available in ring buffer for writing/reading + * @tstamp: timestamp infomation + */ +struct snd_compr_avail { + __u64 avail; + struct snd_compr_tstamp tstamp; +} __attribute__((packed, aligned(4))); + +enum snd_compr_direction { + SND_COMPRESS_PLAYBACK = 0, + SND_COMPRESS_CAPTURE +}; + +/** + * struct snd_compr_caps - caps descriptor + * @codecs: pointer to array of codecs + * @direction: direction supported. Of type snd_compr_direction + * @min_fragment_size: minimum fragment supported by DSP + * @max_fragment_size: maximum fragment supported by DSP + * @min_fragments: min fragments supported by DSP + * @max_fragments: max fragments supported by DSP + * @num_codecs: number of codecs supported + * @reserved: reserved field + */ +struct snd_compr_caps { + __u32 num_codecs; + __u32 direction; + __u32 min_fragment_size; + __u32 max_fragment_size; + __u32 min_fragments; + __u32 max_fragments; + __u32 codecs[MAX_NUM_CODECS]; + __u32 reserved[11]; +} __attribute__((packed, aligned(4))); + +/** + * struct snd_compr_codec_caps - query capability of codec + * @codec: codec for which capability is queried + * @num_descriptors: number of codec descriptors + * @descriptor: array of codec capability descriptor + */ +struct snd_compr_codec_caps { + __u32 codec; + __u32 num_descriptors; + struct snd_codec_desc descriptor[MAX_NUM_CODEC_DESCRIPTORS]; +} __attribute__((packed, aligned(4))); + +/** + * enum sndrv_compress_encoder + * @SNDRV_COMPRESS_ENCODER_PADDING: no of samples appended by the encoder at the + * end of the track + * @SNDRV_COMPRESS_ENCODER_DELAY: no of samples inserted by the encoder at the + * beginning of the track + */ +enum sndrv_compress_encoder { + SNDRV_COMPRESS_ENCODER_PADDING = 1, + SNDRV_COMPRESS_ENCODER_DELAY = 2, +}; + +/** + * struct snd_compr_metadata - compressed stream metadata + * @key: key id + * @value: key value + */ +struct snd_compr_metadata { + __u32 key; + __u32 value[8]; +} __attribute__((packed, aligned(4))); + +/** + * compress path ioctl definitions + * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP + * SNDRV_COMPRESS_GET_CODEC_CAPS: Query capability of a codec + * SNDRV_COMPRESS_SET_PARAMS: Set codec and stream parameters + * Note: only codec params can be changed runtime and stream params cant be + * SNDRV_COMPRESS_GET_PARAMS: Query codec params + * SNDRV_COMPRESS_TSTAMP: get the current timestamp value + * SNDRV_COMPRESS_AVAIL: get the current buffer avail value. + * This also queries the tstamp properties + * SNDRV_COMPRESS_PAUSE: Pause the running stream + * SNDRV_COMPRESS_RESUME: resume a paused stream + * SNDRV_COMPRESS_START: Start a stream + * SNDRV_COMPRESS_STOP: stop a running stream, discarding ring buffer content + * and the buffers currently with DSP + * SNDRV_COMPRESS_DRAIN: Play till end of buffers and stop after that + * SNDRV_COMPRESS_IOCTL_VERSION: Query the API version + */ +#define SNDRV_COMPRESS_IOCTL_VERSION _IOR('C', 0x00, int) +#define SNDRV_COMPRESS_GET_CAPS _IOWR('C', 0x10, struct snd_compr_caps) +#define SNDRV_COMPRESS_GET_CODEC_CAPS _IOWR('C', 0x11,\ + struct snd_compr_codec_caps) +#define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params) +#define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec) +#define SNDRV_COMPRESS_SET_METADATA _IOW('C', 0x14,\ + struct snd_compr_metadata) +#define SNDRV_COMPRESS_GET_METADATA _IOWR('C', 0x15,\ + struct snd_compr_metadata) +#define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) +#define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) +#define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) +#define SNDRV_COMPRESS_RESUME _IO('C', 0x31) +#define SNDRV_COMPRESS_START _IO('C', 0x32) +#define SNDRV_COMPRESS_STOP _IO('C', 0x33) +#define SNDRV_COMPRESS_DRAIN _IO('C', 0x34) +#define SNDRV_COMPRESS_NEXT_TRACK _IO('C', 0x35) +#define SNDRV_COMPRESS_PARTIAL_DRAIN _IO('C', 0x36) +/* + * TODO + * 1. add mmap support + * + */ +#define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */ +#define SND_COMPR_TRIGGER_NEXT_TRACK 8 +#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 9 +#endif diff --git a/include/sound/compress_params.h b/include/sound/compress_params.h new file mode 100644 index 000000000000..d9bd9ca0d5b0 --- /dev/null +++ b/include/sound/compress_params.h @@ -0,0 +1,404 @@ +/* + * compress_params.h - codec types and parameters for compressed data + * streaming interface + * + * Copyright (C) 2011 Intel Corporation + * Authors: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> + * Vinod Koul <vinod.koul(a)linux.intel.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 + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * The definitions in this file are derived from the OpenMAX AL version 1.1 + * and OpenMAX IL v 1.1.2 header files which contain the copyright notice below. + * + * Copyright (c) 2007-2010 The Khronos Group Inc. + * + * Permission is hereby granted, free of charge, to any person obtaining + * a copy of this software and/or associated documentation files (the + * "Materials "), to deal in the Materials without restriction, including + * without limitation the rights to use, copy, modify, merge, publish, + * distribute, sublicense, and/or sell copies of the Materials, and to + * permit persons to whom the Materials are furnished to do so, subject to + * the following conditions: + * + * The above copyright notice and this permission notice shall be included + * in all copies or substantial portions of the Materials. + * + * THE MATERIALS ARE PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS + * OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF + * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. + * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY + * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, + * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE + * MATERIALS OR THE USE OR OTHER DEALINGS IN THE MATERIALS. + * + */ +#ifndef __SND_COMPRESS_PARAMS_H +#define __SND_COMPRESS_PARAMS_H + +#include <linux/types.h> + +/* AUDIO CODECS SUPPORTED */ +#define MAX_NUM_CODECS 32 +#define MAX_NUM_CODEC_DESCRIPTORS 32 +#define MAX_NUM_BITRATES 32 +#define MAX_NUM_SAMPLE_RATES 32 + +/* Codecs are listed linearly to allow for extensibility */ +#define SND_AUDIOCODEC_PCM ((__u32) 0x00000001) +#define SND_AUDIOCODEC_MP3 ((__u32) 0x00000002) +#define SND_AUDIOCODEC_AMR ((__u32) 0x00000003) +#define SND_AUDIOCODEC_AMRWB ((__u32) 0x00000004) +#define SND_AUDIOCODEC_AMRWBPLUS ((__u32) 0x00000005) +#define SND_AUDIOCODEC_AAC ((__u32) 0x00000006) +#define SND_AUDIOCODEC_WMA ((__u32) 0x00000007) +#define SND_AUDIOCODEC_REAL ((__u32) 0x00000008) +#define SND_AUDIOCODEC_VORBIS ((__u32) 0x00000009) +#define SND_AUDIOCODEC_FLAC ((__u32) 0x0000000A) +#define SND_AUDIOCODEC_IEC61937 ((__u32) 0x0000000B) +#define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C) +#define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D) +#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_G729 + +/* + * Profile and modes are listed with bit masks. This allows for a + * more compact representation of fields that will not evolve + * (in contrast to the list of codecs) + */ + +#define SND_AUDIOPROFILE_PCM ((__u32) 0x00000001) + +/* MP3 modes are only useful for encoders */ +#define SND_AUDIOCHANMODE_MP3_MONO ((__u32) 0x00000001) +#define SND_AUDIOCHANMODE_MP3_STEREO ((__u32) 0x00000002) +#define SND_AUDIOCHANMODE_MP3_JOINTSTEREO ((__u32) 0x00000004) +#define SND_AUDIOCHANMODE_MP3_DUAL ((__u32) 0x00000008) + +#define SND_AUDIOPROFILE_AMR ((__u32) 0x00000001) + +/* AMR modes are only useful for encoders */ +#define SND_AUDIOMODE_AMR_DTX_OFF ((__u32) 0x00000001) +#define SND_AUDIOMODE_AMR_VAD1 ((__u32) 0x00000002) +#define SND_AUDIOMODE_AMR_VAD2 ((__u32) 0x00000004) + +#define SND_AUDIOSTREAMFORMAT_UNDEFINED ((__u32) 0x00000000) +#define SND_AUDIOSTREAMFORMAT_CONFORMANCE ((__u32) 0x00000001) +#define SND_AUDIOSTREAMFORMAT_IF1 ((__u32) 0x00000002) +#define SND_AUDIOSTREAMFORMAT_IF2 ((__u32) 0x00000004) +#define SND_AUDIOSTREAMFORMAT_FSF ((__u32) 0x00000008) +#define SND_AUDIOSTREAMFORMAT_RTPPAYLOAD ((__u32) 0x00000010) +#define SND_AUDIOSTREAMFORMAT_ITU ((__u32) 0x00000020) + +#define SND_AUDIOPROFILE_AMRWB ((__u32) 0x00000001) + +/* AMRWB modes are only useful for encoders */ +#define SND_AUDIOMODE_AMRWB_DTX_OFF ((__u32) 0x00000001) +#define SND_AUDIOMODE_AMRWB_VAD1 ((__u32) 0x00000002) +#define SND_AUDIOMODE_AMRWB_VAD2 ((__u32) 0x00000004) + +#define SND_AUDIOPROFILE_AMRWBPLUS ((__u32) 0x00000001) + +#define SND_AUDIOPROFILE_AAC ((__u32) 0x00000001) + +/* AAC modes are required for encoders and decoders */ +#define SND_AUDIOMODE_AAC_MAIN ((__u32) 0x00000001) +#define SND_AUDIOMODE_AAC_LC ((__u32) 0x00000002) +#define SND_AUDIOMODE_AAC_SSR ((__u32) 0x00000004) +#define SND_AUDIOMODE_AAC_LTP ((__u32) 0x00000008) +#define SND_AUDIOMODE_AAC_HE ((__u32) 0x00000010) +#define SND_AUDIOMODE_AAC_SCALABLE ((__u32) 0x00000020) +#define SND_AUDIOMODE_AAC_ERLC ((__u32) 0x00000040) +#define SND_AUDIOMODE_AAC_LD ((__u32) 0x00000080) +#define SND_AUDIOMODE_AAC_HE_PS ((__u32) 0x00000100) +#define SND_AUDIOMODE_AAC_HE_MPS ((__u32) 0x00000200) + +/* AAC formats are required for encoders and decoders */ +#define SND_AUDIOSTREAMFORMAT_MP2ADTS ((__u32) 0x00000001) +#define SND_AUDIOSTREAMFORMAT_MP4ADTS ((__u32) 0x00000002) +#define SND_AUDIOSTREAMFORMAT_MP4LOAS ((__u32) 0x00000004) +#define SND_AUDIOSTREAMFORMAT_MP4LATM ((__u32) 0x00000008) +#define SND_AUDIOSTREAMFORMAT_ADIF ((__u32) 0x00000010) +#define SND_AUDIOSTREAMFORMAT_MP4FF ((__u32) 0x00000020) +#define SND_AUDIOSTREAMFORMAT_RAW ((__u32) 0x00000040) + +#define SND_AUDIOPROFILE_WMA7 ((__u32) 0x00000001) +#define SND_AUDIOPROFILE_WMA8 ((__u32) 0x00000002) +#define SND_AUDIOPROFILE_WMA9 ((__u32) 0x00000004) +#define SND_AUDIOPROFILE_WMA10 ((__u32) 0x00000008) + +#define SND_AUDIOMODE_WMA_LEVEL1 ((__u32) 0x00000001) +#define SND_AUDIOMODE_WMA_LEVEL2 ((__u32) 0x00000002) +#define SND_AUDIOMODE_WMA_LEVEL3 ((__u32) 0x00000004) +#define SND_AUDIOMODE_WMA_LEVEL4 ((__u32) 0x00000008) +#define SND_AUDIOMODE_WMAPRO_LEVELM0 ((__u32) 0x00000010) +#define SND_AUDIOMODE_WMAPRO_LEVELM1 ((__u32) 0x00000020) +#define SND_AUDIOMODE_WMAPRO_LEVELM2 ((__u32) 0x00000040) +#define SND_AUDIOMODE_WMAPRO_LEVELM3 ((__u32) 0x00000080) + +#define SND_AUDIOSTREAMFORMAT_WMA_ASF ((__u32) 0x00000001) +/* + * Some implementations strip the ASF header and only send ASF packets + * to the DSP + */ +#define SND_AUDIOSTREAMFORMAT_WMA_NOASF_HDR ((__u32) 0x00000002) + +#define SND_AUDIOPROFILE_REALAUDIO ((__u32) 0x00000001) + +#define SND_AUDIOMODE_REALAUDIO_G2 ((__u32) 0x00000001) +#define SND_AUDIOMODE_REALAUDIO_8 ((__u32) 0x00000002) +#define SND_AUDIOMODE_REALAUDIO_10 ((__u32) 0x00000004) +#define SND_AUDIOMODE_REALAUDIO_SURROUND ((__u32) 0x00000008) + +#define SND_AUDIOPROFILE_VORBIS ((__u32) 0x00000001) + +#define SND_AUDIOMODE_VORBIS ((__u32) 0x00000001) + +#define SND_AUDIOPROFILE_FLAC ((__u32) 0x00000001) + +/* + * Define quality levels for FLAC encoders, from LEVEL0 (fast) + * to LEVEL8 (best) + */ +#define SND_AUDIOMODE_FLAC_LEVEL0 ((__u32) 0x00000001) +#define SND_AUDIOMODE_FLAC_LEVEL1 ((__u32) 0x00000002) +#define SND_AUDIOMODE_FLAC_LEVEL2 ((__u32) 0x00000004) +#define SND_AUDIOMODE_FLAC_LEVEL3 ((__u32) 0x00000008) +#define SND_AUDIOMODE_FLAC_LEVEL4 ((__u32) 0x00000010) +#define SND_AUDIOMODE_FLAC_LEVEL5 ((__u32) 0x00000020) +#define SND_AUDIOMODE_FLAC_LEVEL6 ((__u32) 0x00000040) +#define SND_AUDIOMODE_FLAC_LEVEL7 ((__u32) 0x00000080) +#define SND_AUDIOMODE_FLAC_LEVEL8 ((__u32) 0x00000100) + +#define SND_AUDIOSTREAMFORMAT_FLAC ((__u32) 0x00000001) +#define SND_AUDIOSTREAMFORMAT_FLAC_OGG ((__u32) 0x00000002) + +/* IEC61937 payloads without CUVP and preambles */ +#define SND_AUDIOPROFILE_IEC61937 ((__u32) 0x00000001) +/* IEC61937 with S/PDIF preambles+CUVP bits in 32-bit containers */ +#define SND_AUDIOPROFILE_IEC61937_SPDIF ((__u32) 0x00000002) + +/* + * IEC modes are mandatory for decoders. Format autodetection + * will only happen on the DSP side with mode 0. The PCM mode should + * not be used, the PCM codec should be used instead. + */ +#define SND_AUDIOMODE_IEC_REF_STREAM_HEADER ((__u32) 0x00000000) +#define SND_AUDIOMODE_IEC_LPCM ((__u32) 0x00000001) +#define SND_AUDIOMODE_IEC_AC3 ((__u32) 0x00000002) +#define SND_AUDIOMODE_IEC_MPEG1 ((__u32) 0x00000004) +#define SND_AUDIOMODE_IEC_MP3 ((__u32) 0x00000008) +#define SND_AUDIOMODE_IEC_MPEG2 ((__u32) 0x00000010) +#define SND_AUDIOMODE_IEC_AACLC ((__u32) 0x00000020) +#define SND_AUDIOMODE_IEC_DTS ((__u32) 0x00000040) +#define SND_AUDIOMODE_IEC_ATRAC ((__u32) 0x00000080) +#define SND_AUDIOMODE_IEC_SACD ((__u32) 0x00000100) +#define SND_AUDIOMODE_IEC_EAC3 ((__u32) 0x00000200) +#define SND_AUDIOMODE_IEC_DTS_HD ((__u32) 0x00000400) +#define SND_AUDIOMODE_IEC_MLP ((__u32) 0x00000800) +#define SND_AUDIOMODE_IEC_DST ((__u32) 0x00001000) +#define SND_AUDIOMODE_IEC_WMAPRO ((__u32) 0x00002000) +#define SND_AUDIOMODE_IEC_REF_CXT ((__u32) 0x00004000) +#define SND_AUDIOMODE_IEC_HE_AAC ((__u32) 0x00008000) +#define SND_AUDIOMODE_IEC_HE_AAC2 ((__u32) 0x00010000) +#define SND_AUDIOMODE_IEC_MPEG_SURROUND ((__u32) 0x00020000) + +#define SND_AUDIOPROFILE_G723_1 ((__u32) 0x00000001) + +#define SND_AUDIOMODE_G723_1_ANNEX_A ((__u32) 0x00000001) +#define SND_AUDIOMODE_G723_1_ANNEX_B ((__u32) 0x00000002) +#define SND_AUDIOMODE_G723_1_ANNEX_C ((__u32) 0x00000004) + +#define SND_AUDIOPROFILE_G729 ((__u32) 0x00000001) + +#define SND_AUDIOMODE_G729_ANNEX_A ((__u32) 0x00000001) +#define SND_AUDIOMODE_G729_ANNEX_B ((__u32) 0x00000002) + +/* <FIXME: multichannel encoders aren't supported for now. Would need + an additional definition of channel arrangement> */ + +/* VBR/CBR definitions */ +#define SND_RATECONTROLMODE_CONSTANTBITRATE ((__u32) 0x00000001) +#define SND_RATECONTROLMODE_VARIABLEBITRATE ((__u32) 0x00000002) + +/* Encoder options */ + +struct snd_enc_wma { + __u32 super_block_align; /* WMA Type-specific data */ +}; + + +/** + * struct snd_enc_vorbis + * @quality: Sets encoding quality to n, between -1 (low) and 10 (high). + * In the default mode of operation, the quality level is 3. + * Normal quality range is 0 - 10. + * @managed: Boolean. Set bitrate management mode. This turns off the + * normal VBR encoding, but allows hard or soft bitrate constraints to be + * enforced by the encoder. This mode can be slower, and may also be + * lower quality. It is primarily useful for streaming. + * @max_bit_rate: Enabled only if managed is TRUE + * @min_bit_rate: Enabled only if managed is TRUE + * @downmix: Boolean. Downmix input from stereo to mono (has no effect on + * non-stereo streams). Useful for lower-bitrate encoding. + * + * These options were extracted from the OpenMAX IL spec and Gstreamer vorbisenc + * properties + * + * For best quality users should specify VBR mode and set quality levels. + */ + +struct snd_enc_vorbis { + __s32 quality; + __u32 managed; + __u32 max_bit_rate; + __u32 min_bit_rate; + __u32 downmix; +} __attribute__((packed, aligned(4))); + + +/** + * struct snd_enc_real + * @quant_bits: number of coupling quantization bits in the stream + * @start_region: coupling start region in the stream + * @num_regions: number of regions value + * + * These options were extracted from the OpenMAX IL spec + */ + +struct snd_enc_real { + __u32 quant_bits; + __u32 start_region; + __u32 num_regions; +} __attribute__((packed, aligned(4))); + +/** + * struct snd_enc_flac + * @num: serial number, valid only for OGG formats + * needs to be set by application + * @gain: Add replay gain tags + * + * These options were extracted from the FLAC online documentation + * at http://flac.sourceforge.net/documentation_tools_flac.html + * + * To make the API simpler, it is assumed that the user will select quality + * profiles. Additional options that affect encoding quality and speed can + * be added at a later stage if needed. + * + * By default the Subset format is used by encoders. + * + * TAGS such as pictures, etc, cannot be handled by an offloaded encoder and are + * not supported in this API. + */ + +struct snd_enc_flac { + __u32 num; + __u32 gain; +} __attribute__((packed, aligned(4))); + +struct snd_enc_generic { + __u32 bw; /* encoder bandwidth */ + __s32 reserved[15]; +} __attribute__((packed, aligned(4))); + +union snd_codec_options { + struct snd_enc_wma wma; + struct snd_enc_vorbis vorbis; + struct snd_enc_real real; + struct snd_enc_flac flac; + struct snd_enc_generic generic; +} __attribute__((packed, aligned(4))); + +/** struct snd_codec_desc - description of codec capabilities + * @max_ch: Maximum number of audio channels + * @sample_rates: Sampling rates in Hz, use values like 48000 for this + * @num_sample_rates: Number of valid values in sample_rates array + * @bit_rate: Indexed array containing supported bit rates + * @num_bitrates: Number of valid values in bit_rate array + * @rate_control: value is specified by SND_RATECONTROLMODE defines. + * @profiles: Supported profiles. See SND_AUDIOPROFILE defines. + * @modes: Supported modes. See SND_AUDIOMODE defines + * @formats: Supported formats. See SND_AUDIOSTREAMFORMAT defines + * @min_buffer: Minimum buffer size handled by codec implementation + * @reserved: reserved for future use + * + * This structure provides a scalar value for profiles, modes and stream + * format fields. + * If an implementation supports multiple combinations, they will be listed as + * codecs with different descriptors, for example there would be 2 descriptors + * for AAC-RAW and AAC-ADTS. + * This entails some redundancy but makes it easier to avoid invalid + * configurations. + * + */ + +struct snd_codec_desc { + __u32 max_ch; + __u32 sample_rates[MAX_NUM_SAMPLE_RATES]; + __u32 num_sample_rates; + __u32 bit_rate[MAX_NUM_BITRATES]; + __u32 num_bitrates; + __u32 rate_control; + __u32 profiles; + __u32 modes; + __u32 formats; + __u32 min_buffer; + __u32 reserved[15]; +} __attribute__((packed, aligned(4))); + +/** struct snd_codec + * @id: Identifies the supported audio encoder/decoder. + * See SND_AUDIOCODEC macros. + * @ch_in: Number of input audio channels + * @ch_out: Number of output channels. In case of contradiction between + * this field and the channelMode field, the channelMode field + * overrides. + * @sample_rate: Audio sample rate of input data in Hz, use values like 48000 + * for this. + * @bit_rate: Bitrate of encoded data. May be ignored by decoders + * @rate_control: Encoding rate control. See SND_RATECONTROLMODE defines. + * Encoders may rely on profiles for quality levels. + * May be ignored by decoders. + * @profile: Mandatory for encoders, can be mandatory for specific + * decoders as well. See SND_AUDIOPROFILE defines. + * @level: Supported level (Only used by WMA at the moment) + * @ch_mode: Channel mode for encoder. See SND_AUDIOCHANMODE defines + * @format: Format of encoded bistream. Mandatory when defined. + * See SND_AUDIOSTREAMFORMAT defines. + * @align: Block alignment in bytes of an audio sample. + * Only required for PCM or IEC formats. + * @options: encoder-specific settings + * @reserved: reserved for future use + */ + +struct snd_codec { + __u32 id; + __u32 ch_in; + __u32 ch_out; + __u32 sample_rate; + __u32 bit_rate; + __u32 rate_control; + __u32 profile; + __u32 level; + __u32 ch_mode; + __u32 format; + __u32 align; + union snd_codec_options options; + __u32 reserved[3]; +} __attribute__((packed, aligned(4))); + +#endif diff --git a/src/Makefile.am b/src/Makefile.am index fa255ff43ee0..3930986946cf 100644 --- a/src/Makefile.am +++ b/src/Makefile.am @@ -26,6 +26,10 @@ if BUILD_PCM SUBDIRS += pcm timer libasound_la_LIBADD += pcm/libpcm.la timer/libtimer.la endif +if BUILD_COMPRESS +SUBDIRS += compress +libasound_la_LIBADD += compress/libcompress.la +endif if BUILD_RAWMIDI SUBDIRS += rawmidi libasound_la_LIBADD += rawmidi/librawmidi.la @@ -66,6 +70,9 @@ pcm/libpcm.la: ordinary_pcm/libordinarypcm.la: $(MAKE) -C ordinary_pcm libordinarypcm.la +pcm/libcompress.la: + $(MAKE) -C compress libcompress.la + rawmidi/librawmidi.la: $(MAKE) -C rawmidi librawmidi.la diff --git a/src/compress/Makefile.am b/src/compress/Makefile.am new file mode 100644 index 000000000000..893871ab00a5 --- /dev/null +++ b/src/compress/Makefile.am @@ -0,0 +1,8 @@ +EXTRA_LTLIBRARIES=libcompress.la + +libcompress_la_SOURCES = compress.c + +all: libcompress.la + + +AM_CPPFLAGS=-I$(top_srcdir)/include diff --git a/src/compress/compress.c b/src/compress/compress.c new file mode 100644 index 000000000000..e3fe828f2b1b --- /dev/null +++ b/src/compress/compress.c @@ -0,0 +1,599 @@ +/* + * tinycompress library for compress audio offload in alsa + * Copyright (c) 2011-2012, Intel Corporation. + * + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU Lesser General Public License, + * version 2.1, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public + * License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to + * the Free Software Foundation, Inc., + * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include <stdio.h> +#include <stdlib.h> +#include <fcntl.h> +#include <stdarg.h> +#include <string.h> +#include <errno.h> +#include <unistd.h> +#include <poll.h> +#include <stdbool.h> +#include <sys/ioctl.h> +#include <sys/mman.h> +#include <sys/time.h> +#include <limits.h> + +#include <linux/types.h> +#include <linux/ioctl.h> +#define __force +#define __bitwise +#define __user +#include <sound/asound.h> +#include "sound/compress_params.h" +#include "sound/compress_offload.h" + +#include "compress.h" + +#define COMPR_ERR_MAX 128 + +/* Default maximum time we will wait in a poll() - 20 seconds */ +#define DEFAULT_MAX_POLL_WAIT_MS 20000 + +struct snd_compr { + int fd; + unsigned int flags; + char error[COMPR_ERR_MAX]; + struct snd_compr_config *config; + int running; + int max_poll_wait_ms; + int nonblocking; + unsigned int gapless_metadata; + unsigned int next_track; +}; + +static int oops(struct snd_compr *compr, int e, const char *fmt, ...) +{ + va_list ap; + int sz; + + va_start(ap, fmt); + vsnprintf(compr->error, COMPR_ERR_MAX, fmt, ap); + va_end(ap); + sz = strlen(compr->error); + + snprintf(compr->error + sz, COMPR_ERR_MAX - sz, + ": %s", strerror(e)); + errno = e; + + return -1; +} + +const char *snd_compr_get_error(struct snd_compr *compr) +{ + return compr->error; +} +static struct snd_compr bad_compress = { + .fd = -1, +}; + +int snd_compr_is_running(struct snd_compr *compr) +{ + return ((compr->fd > 0) && compr->running) ? 1 : 0; +} + +int snd_compr_is_ready(struct snd_compr *compr) +{ + return (compr->fd > 0) ? 1 : 0; +} + +static int snd_compr_get_version(struct snd_compr *compr) +{ + int version = 0; + + if (ioctl(compr->fd, SNDRV_COMPRESS_IOCTL_VERSION, &version)) { + oops(compr, errno, "cant read version"); + return -1; + } + return version; +} + +static bool _snd_compr_snd_compr_is_codec_supported(struct snd_compr *compr, + struct snd_compr_config *config, const struct snd_compr_caps *caps) +{ + bool codec = false; + unsigned int i; + + for (i = 0; i < caps->num_codecs; i++) { + if (caps->codecs[i] == config->codec->id) { + /* found the codec */ + codec = true; + break; + } + } + if (codec == false) { + oops(compr, ENXIO, "this codec is not supported"); + return false; + } + + if (config->fragment_size < caps->min_fragment_size) { + oops(compr, EINVAL, "requested fragment size %d is below min supported %d", + config->fragment_size, caps->min_fragment_size); + return false; + } + if (config->fragment_size > caps->max_fragment_size) { + oops(compr, EINVAL, "requested fragment size %d is above max supported %d", + config->fragment_size, caps->max_fragment_size); + return false; + } + if (config->fragments < caps->min_fragments) { + oops(compr, EINVAL, "requested fragments %d are below min supported %d", + config->fragments, caps->min_fragments); + return false; + } + if (config->fragments > caps->max_fragments) { + oops(compr, EINVAL, "requested fragments %d are above max supported %d", + config->fragments, caps->max_fragments); + return false; + } + + /* TODO: match the codec properties */ + return true; +} + +static bool _snd_compr_is_codec_type_supported(int fd, struct snd_codec *codec) +{ + struct snd_compr_caps caps; + bool found = false; + unsigned int i; + + if (ioctl(fd, SNDRV_COMPRESS_GET_CAPS, &caps)) { + oops(&bad_compress, errno, "cannot get device caps"); + return false; + } + + for (i = 0; i < caps.num_codecs; i++) { + if (caps.codecs[i] == codec->id) { + /* found the codec */ + found = true; + break; + } + } + /* TODO: match the codec properties */ + return found; +} + +static inline void +snd_compr_fill_params(struct snd_compr_config *config, struct snd_compr_params *params) +{ + params->buffer.fragment_size = config->fragment_size; + params->buffer.fragments = config->fragments; + memcpy(&params->codec, config->codec, sizeof(params->codec)); +} + +struct snd_compr *snd_compr_open(unsigned int card, unsigned int device, + unsigned int flags, struct snd_compr_config *config) +{ + struct snd_compr *compr; + struct snd_compr_params params; + struct snd_compr_caps caps; + char fn[256]; + + if (!config) { + oops(&bad_compress, EINVAL, "passed bad config"); + return &bad_compress; + } + + compr = calloc(1, sizeof(struct snd_compr)); + if (!compr) { + oops(&bad_compress, errno, "cannot allocate compress object"); + return &bad_compress; + } + + compr->next_track = 0; + compr->gapless_metadata = 0; + compr->config = calloc(1, sizeof(*config)); + if (!compr->config) + goto input_fail; + + snprintf(fn, sizeof(fn), "/dev/snd/comprC%uD%u", card, device); + + compr->max_poll_wait_ms = DEFAULT_MAX_POLL_WAIT_MS; + + compr->flags = flags; + if (!((flags & COMPRESS_OUT) || (flags & COMPRESS_IN))) { + oops(&bad_compress, EINVAL, "can't deduce device direction from given flags"); + goto config_fail; + } + + if (flags & COMPRESS_OUT) { + compr->fd = open(fn, O_RDONLY); + } else { + compr->fd = open(fn, O_WRONLY); + } + if (compr->fd < 0) { + oops(&bad_compress, errno, "cannot open device '%s'", fn); + goto config_fail; + } + + if (ioctl(compr->fd, SNDRV_COMPRESS_GET_CAPS, &caps)) { + oops(compr, errno, "cannot get device caps"); + goto codec_fail; + } + + /* If caller passed "don't care" fill in default values */ + if ((config->fragment_size == 0) || (config->fragments == 0)) { + config->fragment_size = caps.min_fragment_size; + config->fragments = caps.max_fragments; + } + +#if 0 + /* FIXME need to turn this On when DSP supports + * and treat in no support case + */ + if (_snd_compr_snd_compr_is_codec_supported(compr, config, &caps) == false) { + oops(compr, errno, "codec not supported\n"); + goto codec_fail; + } +#endif + + memcpy(compr->config, config, sizeof(*compr->config)); + snd_compr_fill_params(config, &params); + + if (ioctl(compr->fd, SNDRV_COMPRESS_SET_PARAMS, &params)) { + oops(&bad_compress, errno, "cannot set device"); + goto codec_fail; + } + + return compr; + +codec_fail: + close(compr->fd); + compr->fd = -1; +config_fail: + free(compr->config); +input_fail: + free(compr); + return &bad_compress; +} + +void snd_compr_close(struct snd_compr *compr) +{ + if (compr == &bad_compress) + return; + + if (compr->fd >= 0) + close(compr->fd); + compr->running = 0; + compr->fd = -1; + free(compr->config); + free(compr); +} + +int snd_compr_get_hpointer(struct snd_compr *compr, + unsigned int *avail, struct timespec *tstamp) +{ + struct snd_compr_avail kavail; + __u64 time; + + if (!snd_compr_is_ready(compr)) + return oops(compr, ENODEV, "device not ready"); + + if (ioctl(compr->fd, SNDRV_COMPRESS_AVAIL, &kavail)) + return oops(compr, errno, "cannot get avail"); + if (0 == kavail.tstamp.sampling_rate) + return oops(compr, ENODATA, "sample rate unknown"); + *avail = (unsigned int)kavail.avail; + time = kavail.tstamp.pcm_io_frames / kavail.tstamp.sampling_rate; + tstamp->tv_sec = time; + time = kavail.tstamp.pcm_io_frames % kavail.tstamp.sampling_rate; + tstamp->tv_nsec = time * 1000000000 / kavail.tstamp.sampling_rate; + return 0; +} + +int snd_compr_get_tstamp(struct snd_compr *compr, + unsigned int *samples, unsigned int *sampling_rate) +{ + struct snd_compr_tstamp ktstamp; + + if (!snd_compr_is_ready(compr)) + return oops(compr, ENODEV, "device not ready"); + + if (ioctl(compr->fd, SNDRV_COMPRESS_TSTAMP, &ktstamp)) + return oops(compr, errno, "cannot get tstamp"); + + *samples = ktstamp.pcm_io_frames; + *sampling_rate = ktstamp.sampling_rate; + return 0; +} + +int snd_compr_write(struct snd_compr *compr, const void *buf, unsigned int size) +{ + struct snd_compr_avail avail; + struct pollfd fds; + int to_write = 0; /* zero indicates we haven't written yet */ + int written, total = 0, ret; + const char* cbuf = buf; + const unsigned int frag_size = compr->config->fragment_size; + + if (!(compr->flags & COMPRESS_IN)) + return oops(compr, EINVAL, "Invalid flag set"); + if (!snd_compr_is_ready(compr)) + return oops(compr, ENODEV, "device not ready"); + fds.fd = compr->fd; + fds.events = POLLOUT; + + /*TODO: treat auto start here first */ + while (size) { + if (ioctl(compr->fd, SNDRV_COMPRESS_AVAIL, &avail)) + return oops(compr, errno, "cannot get avail"); + + /* We can write if we have at least one fragment available + * or there is enough space for all remaining data + */ + if ((avail.avail < frag_size) && (avail.avail < size)) { + + if (compr->nonblocking) + return total; + + ret = poll(&fds, 1, compr->max_poll_wait_ms); + if (fds.revents & POLLERR) { + return oops(compr, EIO, "poll returned error!"); + } + /* A pause will cause -EBADFD or zero. + * This is not an error, just stop writing */ + if ((ret == 0) || (ret == -EBADFD)) + break; + if (ret < 0) + return oops(compr, errno, "poll error"); + if (fds.revents & POLLOUT) { + continue; + } + } + /* write avail bytes */ + if (size > avail.avail) + to_write = avail.avail; + else + to_write = size; + written = write(compr->fd, cbuf, to_write); + /* If play was paused the write returns -EBADFD */ + if (written == -EBADFD) + break; + if (written < 0) + return oops(compr, errno, "write failed!"); + + size -= written; + cbuf += written; + total += written; + } + return total; +} + +int snd_compr_read(struct snd_compr *compr, void *buf, unsigned int size) +{ + struct snd_compr_avail avail; + struct pollfd fds; + int to_read = 0; + int num_read, total = 0, ret; + char* cbuf = buf; + const unsigned int frag_size = compr->config->fragment_size; + + if (!(compr->flags & COMPRESS_OUT)) + return oops(compr, EINVAL, "Invalid flag set"); + if (!snd_compr_is_ready(compr)) + return oops(compr, ENODEV, "device not ready"); + fds.fd = compr->fd; + fds.events = POLLIN; + + while (size) { + if (ioctl(compr->fd, SNDRV_COMPRESS_AVAIL, &avail)) + return oops(compr, errno, "cannot get avail"); + + if ( (avail.avail < frag_size) && (avail.avail < size) ) { + /* Less than one fragment available and not at the + * end of the read, so poll + */ + if (compr->nonblocking) + return total; + + ret = poll(&fds, 1, compr->max_poll_wait_ms); + if (fds.revents & POLLERR) { + return oops(compr, EIO, "poll returned error!"); + } + /* A pause will cause -EBADFD or zero. + * This is not an error, just stop reading */ + if ((ret == 0) || (ret == -EBADFD)) + break; + if (ret < 0) + return oops(compr, errno, "poll error"); + if (fds.revents & POLLIN) { + continue; + } + } + /* read avail bytes */ + if (size > avail.avail) + to_read = avail.avail; + else + to_read = size; + num_read = read(compr->fd, cbuf, to_read); + /* If play was paused the read returns -EBADFD */ + if (num_read == -EBADFD) + break; + if (num_read < 0) + return oops(compr, errno, "read failed!"); + + size -= num_read; + cbuf += num_read; + total += num_read; + } + + return total; +} + +int snd_compr_start(struct snd_compr *compr) +{ + if (!snd_compr_is_ready(compr)) + return oops(compr, ENODEV, "device not ready"); + if (ioctl(compr->fd, SNDRV_COMPRESS_START)) + return oops(compr, errno, "cannot start the stream"); + compr->running = 1; + return 0; + +} + +int snd_compr_stop(struct snd_compr *compr) +{ + if (!snd_compr_is_running(compr)) + return oops(compr, ENODEV, "device not ready"); + if (ioctl(compr->fd, SNDRV_COMPRESS_STOP)) + return oops(compr, errno, "cannot stop the stream"); + return 0; +} + +int snd_compr_pause(struct snd_compr *compr) +{ + if (!snd_compr_is_running(compr)) + return oops(compr, ENODEV, "device not ready"); + if (ioctl(compr->fd, SNDRV_COMPRESS_PAUSE)) + return oops(compr, errno, "cannot pause the stream"); + return 0; +} + +int snd_compr_resume(struct snd_compr *compr) +{ + if (ioctl(compr->fd, SNDRV_COMPRESS_RESUME)) + return oops(compr, errno, "cannot resume the stream"); + return 0; +} + +int snd_compr_drain(struct snd_compr *compr) +{ + if (!snd_compr_is_running(compr)) + return oops(compr, ENODEV, "device not ready"); + if (ioctl(compr->fd, SNDRV_COMPRESS_DRAIN)) + return oops(compr, errno, "cannot drain the stream"); + return 0; +} + +int snd_compr_partial_drain(struct snd_compr *compr) +{ + if (!snd_compr_is_running(compr)) + return oops(compr, ENODEV, "device not ready"); + + if (!compr->next_track) + return oops(compr, EPERM, "next track not signalled"); + if (ioctl(compr->fd, SNDRV_COMPRESS_PARTIAL_DRAIN)) + return oops(compr, errno, "cannot drain the stream\n"); + compr->next_track = 0; + return 0; +} + +int snd_compr_next_track(struct snd_compr *compr) +{ + if (!snd_compr_is_running(compr)) + return oops(compr, ENODEV, "device not ready"); + + if (!compr->gapless_metadata) + return oops(compr, EPERM, "metadata not set"); + if (ioctl(compr->fd, SNDRV_COMPRESS_NEXT_TRACK)) + return oops(compr, errno, "cannot set next track\n"); + compr->next_track = 1; + compr->gapless_metadata = 0; + return 0; +} + +int snd_compr_set_gapless_metadata(struct snd_compr *compr, + struct snd_compr_gapless_mdata *mdata) +{ + struct snd_compr_metadata metadata; + int version; + + if (!snd_compr_is_ready(compr)) + return oops(compr, ENODEV, "device not ready"); + + version = snd_compr_get_version(compr); + if (version <= 0) + return -1; + + if (version < SNDRV_PROTOCOL_VERSION(0, 1, 1)) + return oops(compr, ENXIO, "gapless apis not supported in kernel"); + + metadata.key = SNDRV_COMPRESS_ENCODER_PADDING; + metadata.value[0] = mdata->encoder_padding; + if (ioctl(compr->fd, SNDRV_COMPRESS_SET_METADATA, &metadata)) + return oops(compr, errno, "can't set metadata for stream\n"); + + metadata.key = SNDRV_COMPRESS_ENCODER_DELAY; + metadata.value[0] = mdata->encoder_delay; + if (ioctl(compr->fd, SNDRV_COMPRESS_SET_METADATA, &metadata)) + return oops(compr, errno, "can't set metadata for stream\n"); + compr->gapless_metadata = 1; + return 0; +} + +bool snd_compr_is_codec_supported(unsigned int card, unsigned int device, + unsigned int flags, struct snd_codec *codec) +{ + unsigned int dev_flag; + bool ret; + int fd; + char fn[256]; + + snprintf(fn, sizeof(fn), "/dev/snd/comprC%uD%u", card, device); + + if (flags & COMPRESS_OUT) + dev_flag = O_RDONLY; + else + dev_flag = O_WRONLY; + + fd = open(fn, dev_flag); + if (fd < 0) + return oops(&bad_compress, errno, "cannot open device '%s'", fn); + + ret = _snd_compr_is_codec_type_supported(fd, codec); + + close(fd); + return ret; +} + +void snd_compr_set_max_poll_wait(struct snd_compr *compr, int milliseconds) +{ + compr->max_poll_wait_ms = milliseconds; +} + +void snd_compr_nonblock(struct snd_compr *compr, int nonblock) +{ + compr->nonblocking = !!nonblock; +} + +int snd_compr_wait(struct snd_compr *compr, int timeout_ms) +{ + struct pollfd fds; + int ret; + + fds.fd = compr->fd; + fds.events = POLLOUT | POLLIN; + + ret = poll(&fds, 1, timeout_ms); + if (ret > 0) { + if (fds.revents & POLLERR) + return oops(compr, EIO, "poll returned error!"); + if (fds.revents & (POLLOUT | POLLIN)) + return 0; + } + if (ret == 0) + return oops(compr, ETIME, "poll timed out"); + if (ret < 0) + return oops(compr, errno, "poll error"); + + return oops(compr, EIO, "poll signalled unhandled event"); +} + -- 2.1.0

3 4

[alsa-devel] Facing issues in a2dp sink profile support on linux
by jags gediya 05 Mar '15

05 Mar '15

Below is the detail of my software SDK: Kernel version : 3.10.17 Build tool : Yocto Processor : i.MX6 Sololite Pulseaudio version : 5.0 Bluez version : 4.101 I want to play music from my phone to the linux box running the above described setup. When i play song from the mobile, I am seeing the ACL data through hcidump. but it is not playing on the headphone. In the pulseaudio logs, i am seeing the errors. Below are my changes, for pulseaudio and bluetooth. /etc/pulse/daemon.conf resample-method = trivial enable-remixing = no enable-lfe-remixing = no default-sample-format = s16le default-sample-rate = 48000 alternate-sample-rate = 24000 default-sample-channels = 2 /etc/pulse/system.pa .ifexists module-bluetooth-policy.so load-module module-bluetooth-policy .endif .ifexists module-bluetooth-discover.so load-module module-bluetooth-discover .endif Below are my steps to connect to mobile. pulseaudio --system --realtime -vvvvv --log-time=1 --log-target=file:pulse.log & ADAPTER=`dbus-send --print-reply --system --dest=org.bluez / org.bluez.Manager.DefaultAdapter | cut -d'"' -f 2 | grep org` DEVICE_PATH=`dbus-send --system --dest=org.bluez --print-reply $ADAPTER org.bluez.Adapter.ListDevices | grep path | cut -d'"' -f 2` dbus-send --print-reply --system --dest=org.bluez $DEVICE_PATH org.bluez.AudioSource.Connect in, "pactl modules list short" I am seeing, module-bluetooth-device address="00:73:8D:35:2A:B8" path="/org/bluez/351/hci0/dev_00_73_8D_35_2A_B8" module-loopback source="bluez_source.00_73_8D_35_2A_B8" source_dont_move="true" sink_input_properties="media.role=music" pactl list sources short bluez_source.00_73_8D_35_2A_B8 module-bluetooth-device.c s16le 2ch 44100Hz SUSPENDED this is the logs, i am getting. D ( 1.163| 0.002) [pulseaudio] alsa-sink.c: Read hardware volume: 0: 75% 1: 75% D ( 1.163| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Thread starting up D ( 1.164| 0.000) [alsa-sink-HiFi sgtl5000-0] core-util.c: SCHED_RR|SCHED_RESET_ON_FORK worked. I ( 1.164| 0.000) [alsa-sink-HiFi sgtl5000-0] core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5. I ( 1.165| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Starting playback. D ( 1.165| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 1.165| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 1.166| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 1.166| 0.000) [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... I ( 1.167| 0.000) [pulseaudio] conf.c: Unknown parameters 0 I ( 1.167| 0.000) [pulseaudio] pcm.c: Unknown PCM front:0 I ( 1.167| 0.000) [pulseaudio] alsa-util.c: Error opening PCM device front:0: Invalid argument D ( 1.167| 0.000) [pulseaudio] alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... D ( 1.169| 0.001) [pulseaudio] alsa-util.c: Managed to open hw:0 I ( 1.169| 0.000) [pulseaudio] alsa-util.c: cannot disable ALSA period wakeups D ( 1.169| 0.000) [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms D ( 1.174| 0.005) [pulseaudio] alsa-util.c: Set buffer size first (to 88200 samples), period size second (to 88200 samples). I ( 1.175| 0.000) [pulseaudio] alsa-util.c: ALSA period wakeups were not disabled I ( 1.175| 0.000) [pulseaudio] alsa-source.c: Successfully opened device hw:0. I ( 1.175| 0.000) [pulseaudio] alsa-source.c: Selected mapping 'Analog Stereo' (analog-stereo). I ( 1.175| 0.000) [pulseaudio] alsa-source.c: Successfully enabled mmap() mode. I ( 1.175| 0.000) [pulseaudio] alsa-source.c: Successfully enabled timer-based scheduling mode. I ( 1.177| 0.001) [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' D ( 1.179| 0.002) [pulseaudio] alsa-mixer.c: Added 1 ports I ( 1.180| 0.000) [pulseaudio] source.c: Created source 1 "alsa_input.platform-sound.25.analog-stereo" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right . . . . I ( 1.194| 0.000) [pulseaudio] module.c: Loaded "module-alsa-card" (index: #0; argument: "device_id="0" name="platform-sound.25" card_name="alsa_card.platform-sound.25" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes card_properties="module-udev-detect.discovered=1""). I ( 1.194| 0.000) [pulseaudio] module-udev-detect.c: Card /devices/soc0/sound.25/sound/card0 (alsa_card.platform-sound.25) module loaded. I ( 1.194| 0.000) [pulseaudio] module-udev-detect.c: Found 1 cards. I ( 1.194| 0.000) [pulseaudio] module.c: Loaded "module-udev-detect" (index: #1; argument: ""). D ( 1.195| 0.000) [pulseaudio] cli-command.c: Checking for existence of '/usr/lib/pulse-4.0/modules/module-esound-protocol-unix.so': failure W ( 1.212| 0.016) [pulseaudio] authkey.c: Failed to open cookie file '/var/run/pulse/.config/pulse/cookie': No such file or directory W ( 1.212| 0.000) [pulseaudio] authkey.c: Failed to load authorization key '/var/run/pulse/.config/pulse/cookie': No such file or directory W ( 1.212| 0.000) [pulseaudio] authkey.c: Failed to open cookie file '/var/run/pulse/.pulse-cookie': No such file or directory W ( 1.212| 0.000) [pulseaudio] authkey.c: Failed to load authorization key '/var/run/pulse/.pulse-cookie': No such file or directory D ( 1.213| 0.000) [pulseaudio] authkey.c: Got 0 bytes from cookie file '/var/run/pulse/.config/pulse/cookie', expected 256 I ( 1.214| 0.001) [pulseaudio] module.c: Loaded "module-native-protocol-unix" (index: #2; argument: "auth-anonymous=1"). I ( 1.228| 0.013) [pulseaudio] module-stream-restore.c: Successfully opened database file '/var/lib/pulse/79e7ab36632556972e086ba654e49e31-stream-volumes'. . . . . D ( 107.002| 34.839) [pulseaudio] bluetooth-util.c: dbus: interface=org.bluez.AudioSource, path=/org/bluez/351/hci0/dev_00_73_8D_35_2A_B8, member=PropertyChanged D ( 107.003| 0.000) [pulseaudio] bluetooth-util.c: Device /org/bluez/351/hci0/dev_00_73_8D_35_2A_B8 interface org.bluez.AudioSource property 'State' changed to value 'connecting' D ( 107.363| 0.360) [pulseaudio] bluetooth-util.c: dbus: interface=org.bluez.Device, path=/org/bluez/351/hci0/dev_00_73_8D_35_2A_B8, member=PropertyChanged D ( 107.496| 0.132) [pulseaudio] bluetooth-util.c: dbus: interface=org.bluez.MediaEndpoint, path=/MediaEndpoint/A2DPSink, member=SelectConfiguration D ( 107.496| 0.000) [pulseaudio] bluetooth-util.c: dbus: interface=org.bluez.MediaEndpoint, path=/MediaEndpoint/A2DPSink, member=SelectConfiguration D ( 107.510| 0.013) [pulseaudio] bluetooth-util.c: dbus: interface=org.bluez.MediaEndpoint, path=/MediaEndpoint/A2DPSink, member=SetConfiguration D ( 107.510| 0.000) [pulseaudio] bluetooth-util.c: dbus: interface=org.bluez.MediaEndpoint, path=/MediaEndpoint/A2DPSink, member=SetConfiguration D ( 107.510| 0.000) [pulseaudio] bluetooth-util.c: Transport /org/bluez/351/hci0/dev_00_73_8D_35_2A_B8/fd0 profile 1 available D ( 107.567| 0.056) [pulseaudio] bluetooth-util.c: dbus: interface=org.bluez.AudioSource, path=/org/bluez/351/hci0/dev_00_73_8D_35_2A_B8, member=PropertyChanged D ( 107.567| 0.000) [pulseaudio] bluetooth-util.c: Device /org/bluez/351/hci0/dev_00_73_8D_35_2A_B8 interface org.bluez.AudioSource property 'State' changed to value 'connected' D ( 107.567| 0.000) [pulseaudio] bluetooth-util.c: Transport /org/bluez/351/hci0/dev_00_73_8D_35_2A_B8/fd0 (profile a2dp_source) changed state from disconnected to idle. D ( 107.567| 0.000) [pulseaudio] module-bluetooth-discover.c: Loading module-bluetooth-device address="00:73:8D:35:2A:B8" path="/org/bluez/351/hci0/dev_00_73_8D_35_2A_B8" I ( 107.594| 0.027) [pulseaudio] card.c: Created 1 "bluez_card.00_73_8D_35_2A_B8" W ( 107.595| 0.000) [pulseaudio] module-bluetooth-device.c: Default profile not connected, selecting off profile I ( 107.595| 0.000) [pulseaudio] module.c: Loaded "module-bluetooth-device" (index: #13; argument: "address="00:73:8D:35:2A:B8" path="/org/bluez/351/hci0/dev_00_73_8D_35_2A_B8""). . . . . D ( 195.725| 0.003) [pulseaudio] module-bluetooth-device.c: Acquiring transport /org/bluez/351/hci0/dev_00_73_8D_35_2A_B8/fd0 I ( 195.725| 0.000) [pulseaudio] bluetooth-util.c: Failed optional acquire of transport /org/bluez/351/hci0/dev_00_73_8D_35_2A_B8/fd0 I ( 195.725| 0.000) [pulseaudio] module-bluetooth-device.c: Failed optional acquire of transport /org/bluez/351/hci0/dev_00_73_8D_35_2A_B8/fd0 I ( 195.725| 0.000) [pulseaudio] module-bluetooth-device.c: SBC parameters: I ( 195.725| 0.000) [pulseaudio] module-bluetooth-device.c: allocation=0 I ( 195.725| 0.000) [pulseaudio] module-bluetooth-device.c: subbands=1 I ( 195.725| 0.000) [pulseaudio] module-bluetooth-device.c: blocks=3 I ( 195.725| 0.000) [pulseaudio] module-bluetooth-device.c: bitpool=19 D ( 195.726| 0.000) [pulseaudio] module-device-restore.c: Database contains invalid data for key: source:bluez_source.00_73_8D_35_2A_B8:null I ( 195.726| 0.000) [pulseaudio] source.c: Created source 2 "bluez_source.00_73_8D_35_2A_B8" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I ( 195.726| 0.000) [pulseaudio] source.c: bluetooth.protocol = "a2dp_source" I ( 195.726| 0.000) [pulseaudio] source.c: device.description = "jags gediya" I ( 195.726| 0.000) [pulseaudio] source.c: device.string = "00:73:8D:35:2A:B8" I ( 195.726| 0.000) [pulseaudio] source.c: device.api = "bluez" I ( 195.726| 0.000) [pulseaudio] source.c: device.class = "sound" I ( 195.726| 0.000) [pulseaudio] source.c: device.bus = "bluetooth" I ( 195.726| 0.000) [pulseaudio] source.c: device.form_factor = "phone" I ( 195.726| 0.000) [pulseaudio] source.c: bluez.path = "/org/bluez/351/hci0/dev_00_73_8D_35_2A_B8" I ( 195.726| 0.000) [pulseaudio] source.c: bluez.class = "0x5a020c" I ( 195.726| 0.000) [pulseaudio] source.c: bluez.name = "jags gediya" I ( 195.726| 0.000) [pulseaudio] source.c: device.icon_name = "audio-card-bluetooth" D ( 195.728| 0.001) [bluetooth] module-bluetooth-device.c: IO Thread starting up D ( 195.728| 0.000) [bluetooth] core-util.c: SCHED_RR|SCHED_RESET_ON_FORK worked. I ( 195.728| 0.000) [bluetooth] core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5. D ( 195.729| 0.000) [pulseaudio] module-suspend-on-idle.c: Source bluez_source.00_73_8D_35_2A_B8 becomes idle, timeout in 5 seconds. D ( 195.747| 0.018) [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.platform-sound.25.analog-stereo becomes busy, resuming. D ( 195.747| 0.000) [pulseaudio] sink.c: Suspend cause of sink alsa_output.platform-sound.25.analog-stereo is 0x0000, resuming I ( 195.747| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Trying resume... I ( 195.749| 0.001) [alsa-sink-HiFi sgtl5000-0] alsa-util.c: cannot disable ALSA period wakeups D ( 195.749| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-util.c: Maximum hw buffer size is 371 ms D ( 195.757| 0.007) [alsa-sink-HiFi sgtl5000-0] alsa-util.c: Set buffer size first (to 16384 samples), period size second (to 8192 samples). I ( 195.758| 0.001) [alsa-sink-HiFi sgtl5000-0] alsa-util.c: ALSA period wakeups were not disabled D ( 195.758| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: hwbuf_unused=0 D ( 195.758| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: setting avail_min=15502 I ( 195.759| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Time scheduling watermark is 20.00ms I ( 195.759| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Resumed successfully.... . . . I ( 195.773| 0.000) [pulseaudio] protocol-native.c: Connection died. D ( 195.945| 0.172) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! I ( 195.945| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Underrun! I ( 195.945| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Increasing wakeup watermark to 30.00 ms D ( 196.131| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! I ( 196.131| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Underrun! I ( 196.131| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Increasing wakeup watermark to 40.00 ms D ( 196.316| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! I ( 196.317| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Increasing wakeup watermark to 50.00 ms I ( 196.502| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Increasing wakeup watermark to 56.64 ms I ( 196.688| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Increasing minimal latency to 1.00 ms D ( 196.688| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Latency set to 66.67ms D ( 196.688| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: hwbuf_unused=53780 D ( 196.688| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: setting avail_min=13888 D ( 196.689| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Latency set to 66.67ms D ( 196.689| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: hwbuf_unused=53780 D ( 196.689| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: setting avail_min=13888 I ( 196.874| 0.184) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Increasing minimal latency to 2.00 ms D ( 196.874| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Latency set to 66.67ms D ( 196.874| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: hwbuf_unused=53780 D ( 196.874| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: setting avail_min=13888 . . D ( 304.751| 0.000) [alsa-sink-HiFi sgtl5000-0] ratelimit.c: 47 events suppressed D ( 304.751| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.751| 0.000) [pulseaudio] module-suspend-on-idle.c: Source bluez_source.00_73_8D_35_2A_B8 becomes busy, resuming. D ( 304.751| 0.000) [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D ( 304.752| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.752| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.753| 0.001) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.753| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.754| 0.000) [pulseaudio] module-loopback.c: Loopback overall latency is 63.82 ms + 0.00 ms + 74.89 ms = 138.70 ms D ( 304.754| 0.000) [pulseaudio] module-loopback.c: Should buffer 23512 bytes, buffered at minimum 0 bytes I ( 304.754| 0.000) [pulseaudio] module-loopback.c: New rate of 43513 Hz not within 2‰ of 44100 Hz, forcing smaller adjustment D ( 304.754| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.759| 0.005) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.760| 0.000) [pulseaudio] module-loopback.c: [alsa_output.platform-sound.25.analog-stereo] Updated sampling rate to 44011 Hz. D ( 304.760| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.760| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.760| 0.000) [pulseaudio] module-loopback.c: Loopback overall latency is 56.99 ms + 0.00 ms + 81.77 ms = 138.76 ms D ( 304.761| 0.000) [pulseaudio] module-loopback.c: Should buffer 23472 bytes, buffered at minimum 0 bytes I ( 304.761| 0.000) [pulseaudio] module-loopback.c: New rate of 43514 Hz not within 2‰ of 44011 Hz, forcing smaller adjustment D ( 304.761| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.764| 0.002) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.764| 0.000) [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D ( 304.764| 0.000) [pulseaudio] module-loopback.c: [alsa_output.platform-sound.25.analog-stereo] Updated sampling rate to 43922 Hz. D ( 304.915| 0.150) [alsa-sink-HiFi sgtl5000-0] module-loopback.c: Requesting rewind due to end of underrun. D ( 304.922| 0.007) [alsa-sink-HiFi sgtl5000-0] module-loopback.c: Requesting rewind due to end of underrun. D ( 304.925| 0.002) [alsa-sink-HiFi sgtl5000-0] module-loopback.c: Requesting rewind due to end of underrun. I ( 304.936| 0.011) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Increasing wakeup watermark to 30.00 ms I ( 304.955| 0.018) [alsa-sink-HiFi sgtl5000-0] module-loopback.c: Could not peek into queue D ( 304.955| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 304.966| 0.011) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 304.966| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 304.971| 0.004) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 304.973| 0.002) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 304.976| 0.002) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 304.976| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 304.979| 0.003) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 304.980| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 304.998| 0.018) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.017| 0.018) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.021| 0.004) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.021| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.026| 0.004) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.026| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.031| 0.004) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.031| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.036| 0.004) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.036| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.055| 0.018) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.074| 0.018) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.080| 0.006) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.081| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.086| 0.005) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.086| 0.000) [alsa-sink-HiFi sgtl5000-0] module-loopback.c: Requesting rewind due to end of underrun. D ( 305.086| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.092| 0.005) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.092| 0.000) [alsa-sink-HiFi sgtl5000-0] module-loopback.c: Requesting rewind due to end of underrun. D ( 305.092| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.095| 0.003) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.095| 0.000) [alsa-sink-HiFi sgtl5000-0] module-loopback.c: Requesting rewind due to end of underrun. D ( 305.095| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.114| 0.018) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) I ( 305.122| 0.007) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Increasing wakeup watermark to 40.00 ms I ( 305.129| 0.007) [alsa-sink-HiFi sgtl5000-0] module-loopback.c: Could not peek into queue D ( 305.129| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.140| 0.010) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.140| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.144| 0.003) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.144| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.148| 0.003) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.148| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.152| 0.003) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.152| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.179| 0.026) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.196| 0.017) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.197| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.202| 0.004) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.202| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.207| 0.005) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.207| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.212| 0.004) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.212| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.239| 0.026) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.254| 0.015) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) . . I ( 312.115| 0.000) [pulseaudio] protocol-native.c: Connection died. D ( 314.782| 2.666) [alsa-sink-HiFi sgtl5000-0] ratelimit.c: 27 events suppressed D ( 314.782| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 314.967| 0.184) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 315.153| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 315.339| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 315.525| 0.186) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 315.711| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 315.896| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 316.082| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 316.268| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 316.454| 0.186) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 316.641| 0.187) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 319.797| 3.155) [alsa-sink-HiFi sgtl5000-0] ratelimit.c: 16 events suppressed D ( 319.797| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 319.983| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 320.172| 0.188) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 320.354| 0.182) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 320.540| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 320.726| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 320.912| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 321.097| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 321.284| 0.187) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 321.469| 0.184) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 321.655| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! . . . D ( 381.721| 0.066) [pulseaudio] module-loopback.c: Loopback overall latency is 0.00 ms + 1412.45 ms + 62.23 ms = 1474.68 ms D ( 381.721| 0.000) [pulseaudio] module-loopback.c: Should buffer 64888 bytes, buffered at minimum 210984 bytes I ( 381.721| 0.000) [pulseaudio] module-loopback.c: New rate of 47752 Hz not within 2‰ of 43608 Hz, forcing smaller adjustment D ( 381.722| 0.000) [pulseaudio] module-loopback.c: [alsa_output.platform-sound.25.analog-stereo] Updated sampling rate to 43695 Hz. >From the logs, what can be the issue?

1 0

[alsa-devel] [PATCH] amixer: Don't set only the first item in sset_enum()
by Peter Meerwald 05 Mar '15

05 Mar '15

From: Peter Meerwald <p.meerwald(a)bct-electronic.com> Signed-off-by: Peter Meerwald <pmeerw(a)pmeerw.net> --- amixer/amixer.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/amixer/amixer.c b/amixer/amixer.c index ed60e7c..36c92eb 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -1278,7 +1278,7 @@ static int get_enum_item_index(snd_mixer_elem_t *elem, char **ptrp) static int sset_enum(snd_mixer_elem_t *elem, unsigned int argc, char **argv) { - unsigned int idx, chn = 0; + unsigned int idx, item = 0; int check_flag = ignore_error ? 0 : -1; for (idx = 1; idx < argc; idx++) { @@ -1287,7 +1287,7 @@ static int sset_enum(snd_mixer_elem_t *elem, unsigned int argc, char **argv) int ival = get_enum_item_index(elem, &ptr); if (ival < 0) return check_flag; - if (snd_mixer_selem_set_enum_item(elem, chn, ival) >= 0) + if (snd_mixer_selem_set_enum_item(elem, item++, ival) >= 0) check_flag = 1; /* skip separators */ while (*ptr == ',' || isspace(*ptr)) -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 1/2] ucm: document some standard values
by han.lu＠intel.com 05 Mar '15

05 Mar '15

From: "Lu, Han" <han.lu(a)intel.com> add standard value JackDev, JackControl and JackHWMute for speaker mute control on jack insertion. These values will be applied on pulseaudio. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/include/use-case.h b/include/use-case.h index 697377a..3728906 100644 --- a/include/use-case.h +++ b/include/use-case.h @@ -284,6 +284,10 @@ int snd_use_case_get_list(snd_use_case_mgr_t *uc_mgr, * PlaybackMixerID - mixer playback ID * CaptureMixer - name of capture mixer * CaptureMixerID - mixer capture ID + * JackDev - jack device name + * JackControl - jack control name + * JackHWMute - indicate if the HW mutes a device on jack insertion + * or not. */ int snd_use_case_get(snd_use_case_mgr_t *uc_mgr, const char *identifier, -- 2.1.0

2 3

[alsa-devel] [PATCH 00/30] ASoC: Move jack registration to the card level
by Lars-Peter Clausen 05 Mar '15

05 Mar '15

Audio jacks are typically card level elements and when initially introduced the API to register a jack took a card as its parameter. Back at that time there was only one CODEC per card and DAPM support was only available at the CODEC level and all DAPM elements, including the card level elements, were registered with the CODEC's DAPM context. So naturally for all its DAPM operations the jack API used the CODEC's DAPM context. With the multi-component patchset support for multiple CODECs per card was introduced and since it was no longer clear which DAPM context to use for the jack operations the jack API was updated to take a CODEC instead of a card to make it unambiguous again. Shortly after DAPM support was decoupled from the CODECs and became available at the card level as well. Since then card drivers have been updated to register their card level DAPM elements with the card's DAPM context, but the jack API is still CODEC centric and uses the CODEC's DAPM context. This leaves us with some issues. The DAPM pins that are managed by the jack detection logic are part of the card DAPM context but are accessed through a CODEC DAPM context. Currently this works fine and is more of an inelegance, but might break in the future if we take a more hierarchical approach to DAPM contexts. Furthermore with componentization progressing systems that do not register a snd_soc_codec might appear, while these system may still want to able to register a jack. This patchset introduces a new function called snd_soc_card_jack_register() which allows to register the jacks at the card level again and updates all the existing card drivers to use the new API. - Lars Lars-Peter Clausen (30): ASoC: sn95031: Pass CODEC to sn95031_jack_detection() ASoC: Allow to register jacks at the card level ASoC: simple-card: Register jacks at the card level ASoC: imx-es8328: Register jacks at the card level ASoC: wm1133-ev: Register jacks at the card level ASoC: broadwell: Register jacks at the card level ASoC: byt-max98090: Register jacks at the card level ASoC: cht_bsw_rt5645: Register jacks at the card level ASoC: mfld_machine: Register jacks at the card level ASoC: ams-deltea: Register jacks at the card level ASoC: omap-abe-twl6040: Register jacks at the card level ASoC: omap-twl4030: Register jacks at the card level ASoC: rx51: Register jacks at the card level ASoC: hx4700: Register jacks at the card level ASoC: palm27x: Register jacks at the card level ASoC: ttc-dkb: Register jacks at the card level ASoC: z2: Register jacks at the card level ASoC: h1980_uda1380: Register jacks at the card level ASoC: littlemill: Register jacks at the card level ASoC: lowland: Register jacks at the card level ASoC: rx1950_uda1380: Register jacks at the card level ASoC: smartq: Register jacks at the card level ASoC: speyside: Register jacks at the card level ASoC: tobermory: Register jacks at the card level ASoC: tegra_alc5632: Register jacks at the card level ASoC: tegra_max98090: Register jacks at the card level ASoC: tegra_rt5640: Register jacks at the card level ASoC: tegra_rt5677: Register jacks at the card level ASoC: tegra_wm8903: Register jacks at the card level ASoC: Remove snd_soc_jack_new() include/sound/soc.h | 8 +++++--- sound/soc/codecs/sn95031.c | 14 +++++++------ sound/soc/codecs/sn95031.h | 3 ++- sound/soc/fsl/imx-es8328.c | 6 +++--- sound/soc/fsl/wm1133-ev1.c | 12 +++++------ sound/soc/generic/simple-card.c | 20 +++++++++--------- sound/soc/intel/broadwell.c | 12 +++-------- sound/soc/intel/byt-max98090.c | 11 +++------- sound/soc/intel/cht_bsw_rt5645.c | 12 +++++------ sound/soc/intel/mfld_machine.c | 24 +++++++++------------- sound/soc/omap/ams-delta.c | 4 ++-- sound/soc/omap/omap-abe-twl6040.c | 10 ++++----- sound/soc/omap/omap-twl4030.c | 12 ++++------- sound/soc/omap/rx51.c | 6 +++--- sound/soc/pxa/hx4700.c | 11 +++------- sound/soc/pxa/palm27x.c | 11 +++------- sound/soc/pxa/ttc-dkb.c | 15 ++++++-------- sound/soc/pxa/z2.c | 10 +++------ sound/soc/samsung/h1940_uda1380.c | 9 ++------ sound/soc/samsung/littlemill.c | 12 +++++------ sound/soc/samsung/lowland.c | 14 ++++--------- sound/soc/samsung/rx1950_uda1380.c | 9 ++------ sound/soc/samsung/smartq_wm8987.c | 11 ++++------ sound/soc/samsung/speyside.c | 14 ++++--------- sound/soc/samsung/tobermory.c | 13 ++++-------- sound/soc/soc-jack.c | 42 +++++++++++++++++++++++--------------- sound/soc/tegra/tegra_alc5632.c | 9 ++++---- sound/soc/tegra/tegra_max98090.c | 22 +++++++++----------- sound/soc/tegra/tegra_rt5640.c | 10 +++------ sound/soc/tegra/tegra_rt5677.c | 14 ++++++------- sound/soc/tegra/tegra_wm8903.c | 18 ++++++++-------- 31 files changed, 167 insertions(+), 231 deletions(-) -- 1.8.0

5 38

[alsa-devel] [PATCH][ASoC]fsl_ssi: fix of_property_read_u32_array return value check
by Maciej S. Szmigiero 05 Mar '15

05 Mar '15

of_property_read_u32_array returns 0 on success, so the return value shouldn't be inverted twice, first on assignment then in condition expression. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1227,7 +1227,7 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, ssi_private->dma_params_tx.addr = ssi_private->ssi_phys + CCSR_SSI_STX0; ssi_private->dma_params_rx.addr = ssi_private->ssi_phys + CCSR_SSI_SRX0; - ret = !of_property_read_u32_array(np, "dmas", dmas, 4); + ret = of_property_read_u32_array(np, "dmas", dmas, 4); if (ssi_private->use_dma && !ret && dmas[2] == IMX_DMATYPE_SSI_DUAL) { ssi_private->use_dual_fifo = true; /* When using dual fifo mode, we need to keep watermark

2 1

[alsa-devel] [PATCH 1/3] ASoC: ak4671: Fix control-less DAPM routes
by Lars-Peter Clausen 05 Mar '15

05 Mar '15

Routes without a control must use NULL for the control name. The ak4671 driver uses "NULL" instead in a few places. Previous to commit 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") the DAPM core silently ignored non-NULL controls on non-mixer and non-mux routes. But starting with that commit it will complain and not add the route breaking the ak4671 driver in the process. This patch replaces the incorrect "NULL" control name with NULL to fix the issue. Fixes: 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ak4671.c | 44 ++++++++++++++++++++++---------------------- 1 file changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 632e89f..2a58b1d 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -343,25 +343,25 @@ static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = { }; static const struct snd_soc_dapm_route ak4671_intercon[] = { - {"DAC Left", "NULL", "PMPLL"}, - {"DAC Right", "NULL", "PMPLL"}, - {"ADC Left", "NULL", "PMPLL"}, - {"ADC Right", "NULL", "PMPLL"}, + {"DAC Left", NULL, "PMPLL"}, + {"DAC Right", NULL, "PMPLL"}, + {"ADC Left", NULL, "PMPLL"}, + {"ADC Right", NULL, "PMPLL"}, /* Outputs */ - {"LOUT1", "NULL", "LOUT1 Mixer"}, - {"ROUT1", "NULL", "ROUT1 Mixer"}, - {"LOUT2", "NULL", "LOUT2 Mix Amp"}, - {"ROUT2", "NULL", "ROUT2 Mix Amp"}, - {"LOUT3", "NULL", "LOUT3 Mixer"}, - {"ROUT3", "NULL", "ROUT3 Mixer"}, + {"LOUT1", NULL, "LOUT1 Mixer"}, + {"ROUT1", NULL, "ROUT1 Mixer"}, + {"LOUT2", NULL, "LOUT2 Mix Amp"}, + {"ROUT2", NULL, "ROUT2 Mix Amp"}, + {"LOUT3", NULL, "LOUT3 Mixer"}, + {"ROUT3", NULL, "ROUT3 Mixer"}, {"LOUT1 Mixer", "DACL", "DAC Left"}, {"ROUT1 Mixer", "DACR", "DAC Right"}, {"LOUT2 Mixer", "DACHL", "DAC Left"}, {"ROUT2 Mixer", "DACHR", "DAC Right"}, - {"LOUT2 Mix Amp", "NULL", "LOUT2 Mixer"}, - {"ROUT2 Mix Amp", "NULL", "ROUT2 Mixer"}, + {"LOUT2 Mix Amp", NULL, "LOUT2 Mixer"}, + {"ROUT2 Mix Amp", NULL, "ROUT2 Mixer"}, {"LOUT3 Mixer", "DACSL", "DAC Left"}, {"ROUT3 Mixer", "DACSR", "DAC Right"}, @@ -381,18 +381,18 @@ static const struct snd_soc_dapm_route ak4671_intercon[] = { {"LIN2", NULL, "Mic Bias"}, {"RIN2", NULL, "Mic Bias"}, - {"ADC Left", "NULL", "LIN MUX"}, - {"ADC Right", "NULL", "RIN MUX"}, + {"ADC Left", NULL, "LIN MUX"}, + {"ADC Right", NULL, "RIN MUX"}, /* Analog Loops */ - {"LIN1 Mixing Circuit", "NULL", "LIN1"}, - {"RIN1 Mixing Circuit", "NULL", "RIN1"}, - {"LIN2 Mixing Circuit", "NULL", "LIN2"}, - {"RIN2 Mixing Circuit", "NULL", "RIN2"}, - {"LIN3 Mixing Circuit", "NULL", "LIN3"}, - {"RIN3 Mixing Circuit", "NULL", "RIN3"}, - {"LIN4 Mixing Circuit", "NULL", "LIN4"}, - {"RIN4 Mixing Circuit", "NULL", "RIN4"}, + {"LIN1 Mixing Circuit", NULL, "LIN1"}, + {"RIN1 Mixing Circuit", NULL, "RIN1"}, + {"LIN2 Mixing Circuit", NULL, "LIN2"}, + {"RIN2 Mixing Circuit", NULL, "RIN2"}, + {"LIN3 Mixing Circuit", NULL, "LIN3"}, + {"RIN3 Mixing Circuit", NULL, "RIN3"}, + {"LIN4 Mixing Circuit", NULL, "LIN4"}, + {"RIN4 Mixing Circuit", NULL, "RIN4"}, {"LOUT1 Mixer", "LINL1", "LIN1 Mixing Circuit"}, {"ROUT1 Mixer", "RINR1", "RIN1 Mixing Circuit"}, -- 1.8.0

4 5

[alsa-devel] [PATCH RESEND] ASoC: rt5670: Fix the speaker mono output issue
by Bard Liao 04 Mar '15

04 Mar '15

We need to set left/right control for the speaker amp to get stereo output on speaker. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- We need to set different value according to codec version. And the codec version is stored in in RT5670_VENDOR_ID register. --- sound/soc/codecs/rt5670.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 04ca247..ba8cc14 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2700,6 +2700,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, regmap_write(rt5670->regmap, RT5670_RESET, 0); + regmap_read(rt5670->regmap, RT5670_VENDOR_ID, &val); + if (val >= 4) + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0980); + else + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0d00); + ret = regmap_register_patch(rt5670->regmap, init_list, ARRAY_SIZE(init_list)); if (ret != 0) -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: omap-pcm: Allow only formats with 1, 2, and 4 byte physical size
by Jyri Sarha 04 Mar '15

04 Mar '15

sDMA support only transfer elements with 1, 2, and 4 byte physical size. Initialize the pcm driver accordingly. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/omap/omap-pcm.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index f4b05bc..e49ee23 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -39,7 +39,7 @@ #define pcm_omap1510() 0 #endif -static const struct snd_pcm_hardware omap_pcm_hardware = { +static struct snd_pcm_hardware omap_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | @@ -53,6 +53,24 @@ static const struct snd_pcm_hardware omap_pcm_hardware = { .buffer_bytes_max = 128 * 1024, }; +/* sDMA supports only 1, 2, and 4 byte transfer elements. */ +static void omap_pcm_limit_supported_formats(void) +{ + int i; + + for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { + switch (snd_pcm_format_physical_width(i)) { + case 8: + case 16: + case 32: + omap_pcm_hardware.formats |= (1LL << i); + break; + default: + break; + } + } +} + /* this may get called several times by oss emulation */ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -235,6 +253,7 @@ static struct snd_soc_platform_driver omap_soc_platform = { int omap_pcm_platform_register(struct device *dev) { + omap_pcm_limit_supported_formats(); return devm_snd_soc_register_platform(dev, &omap_soc_platform); } EXPORT_SYMBOL_GPL(omap_pcm_platform_register); -- 1.9.1

4 5

Re: [alsa-devel] [ASoC]fsl_ssi: fix of_property_read_u32_array return value check
by Mark Brown 04 Mar '15

04 Mar '15

On Sun, Feb 15, 2015 at 11:11:43PM +0100, Maciej S. Szmigiero wrote: > of_property_read_u32_array returns 0 on success, so the return value shouldn't be inverted twice, > first on assignment then in condition expression. This doesn't apply to current code, please check and resend. Please also keep your changelog under 80 columns as covered in SubmittingPatches.

2 2

[alsa-devel] [PATCH] ASoC: omap-pcm: Correct dma mask
by Peter Ujfalusi 04 Mar '15

04 Mar '15

DMA_BIT_MASK of 64 is not valid dma address mask for OMAPs, it should be set to 32. The 64 was introduced by commit (in 2009): a152ff24b978 ASoC: OMAP: Make DMA 64 aligned But the dma_mask and coherent_dma_mask can not be used to specify alignment. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 54d59e4f8f5f..d52a6d1ae55d 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -227,7 +227,7 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) struct snd_pcm *pcm = rtd->pcm; int ret; - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(64)); + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); if (ret) return ret; -- 2.3.0

2 3

[alsa-devel] [TINYCOMPRESS][PATCH] crec: support streaming output to stdout
by Richard Fitzgerald 04 Mar '15

04 Mar '15

From: Ammar Zahid Ali Syed <aali(a)opensource.wolfsonmicro.com> It's useful for testing to be able to stream the captured audio to stdout instead of having to go via a file. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Ammar Ali <aali(a)opensource.wolfsonmicro.com> --- crec.c | 73 ++++++++++++++++++++++++++++++++++++++++----------------------- 1 files changed, 46 insertions(+), 27 deletions(-) diff --git a/crec.c b/crec.c index e3f189f..258fade 100644 --- a/crec.c +++ b/crec.c @@ -77,6 +77,8 @@ static int verbose; static int file; +static FILE *finfo; +static bool streamed; static const unsigned int DEFAULT_CHANNELS = 1; static const unsigned int DEFAULT_RATE = 44100; @@ -151,7 +153,7 @@ static void size_wave_header(struct wave_header *header, uint32_t size) static void usage(void) { - fprintf(stderr, "usage: crec [OPTIONS] filename\n" + fprintf(stderr, "usage: crec [OPTIONS] [filename]\n" "-c\tcard number\n" "-d\tdevice node\n" "-b\tbuffer size\n" @@ -162,6 +164,8 @@ static void usage(void) "-C\tSpecify the number of channels (default %u)\n" "-R\tSpecify the sample rate (default %u)\n" "-F\tSpecify the format: S16_LE, S32_LE (default S16_LE)\n\n" + "If filename is not given the output is\n" + "written to stdout\n\n" "Example:\n" "\tcrec -c 1 -d 2 test.wav\n" "\tcrec -f 5 test.wav\n", @@ -180,7 +184,7 @@ static int print_time(struct compress *compress) fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); return -1; } else { - printf("DSP recorded %jd.%jd\n", + fprintf(finfo, "DSP recorded %jd.%jd\n", (intmax_t)tstamp.tv_sec, (intmax_t)tstamp.tv_nsec*1000); } return 0; @@ -195,6 +199,10 @@ static int finish_record() if (!file) return -ENOENT; + /* can't rewind if streaming to stdout */ + if (streamed) + return 0; + /* Get amount of data written to file */ ret = lseek(file, 0, SEEK_END); if (ret < 0) @@ -256,22 +264,27 @@ void capture_samples(char *name, unsigned int card, unsigned int device, length = length * rate * (samplebits / 8) * channels; if (verbose) - printf("%s: entry, reading %u bytes\n", __func__, length); - - file = open(name, O_RDWR | O_CREAT, S_IRUSR | S_IWUSR | S_IRGRP | S_IWGRP); - if (file == -1) { - fprintf(stderr, "Unable to open file '%s'\n", name); - exit(EXIT_FAILURE); - } + fprintf(finfo, "%s: entry, reading %u bytes\n", __func__, length); + if (!name) { + file = STDOUT_FILENO; + } else { + file = open(name, O_RDWR | O_CREAT, S_IRUSR | S_IWUSR | S_IRGRP | S_IWGRP); + if (file == -1) { + fprintf(stderr, "Unable to open file '%s'\n", name); + exit(EXIT_FAILURE); + } + } /* Write a header, will update with size once record is complete */ - init_wave_header(&header, channels, rate, samplebits); - written = write(file, &header, sizeof(header)); - if (written != sizeof(header)) { + if (!streamed) { + init_wave_header(&header, channels, rate, samplebits); + written = write(file, &header, sizeof(header)); + if (written != sizeof(header)) { fprintf(stderr, "Error writing output file header: %s\n", strerror(errno)); goto file_exit; - } + } + } memset(&codec, 0, sizeof(codec)); memset(&config, 0, sizeof(config)); @@ -299,7 +312,7 @@ void capture_samples(char *name, unsigned int card, unsigned int device, }; if (verbose) - printf("%s: Opened compress device\n", __func__); + fprintf(finfo, "%s: Opened compress device\n", __func__); size = config.fragment_size; buffer = malloc(size * config.fragments); @@ -308,15 +321,15 @@ void capture_samples(char *name, unsigned int card, unsigned int device, goto comp_exit; } - printf("Recording file %s On Card %u device %u, with buffer of %lu bytes\n", + fprintf(finfo, "Recording file %s On Card %u device %u, with buffer of %lu bytes\n", name, card, device, buffer_size); - printf("Format %u Channels %u, %u Hz\n", - codec.id, codec.ch_out, rate); + fprintf(finfo, "Codec %u Format %u Channels %u, %u Hz\n", + codec.id, codec.format, codec.ch_out, rate); compress_start(compress); if (verbose) - printf("%s: Capturing audio NOW!!!\n", __func__); + fprintf(finfo, "%s: Capturing audio NOW!!!\n", __func__); do { if (length && size > length - total_read) @@ -344,7 +357,7 @@ void capture_samples(char *name, unsigned int card, unsigned int device, } if (verbose) { print_time(compress); - printf("%s: read %d\n", __func__, read); + fprintf(finfo, "%s: read %d\n", __func__, read); } } } while (!length || total_read < length); @@ -362,7 +375,7 @@ void capture_samples(char *name, unsigned int card, unsigned int device, } if (verbose) - printf("%s: exit success\n", __func__); + fprintf(finfo, "%s: exit success\n", __func__); free(buffer); close(file); @@ -379,7 +392,7 @@ file_exit: close(file); if (verbose) - printf("%s: exit failure\n", __func__); + fprintf(finfo, "%s: exit failure\n", __func__); exit(EXIT_FAILURE); } @@ -408,7 +421,7 @@ int main(int argc, char **argv) exit(EXIT_FAILURE); } - if (argc < 2) + if (argc < 1) usage(); verbose = 0; @@ -456,15 +469,21 @@ int main(int argc, char **argv) exit(EXIT_FAILURE); } } - if (optind >= argc) - usage(); - - file = argv[optind]; + if (optind >= argc) { + file = NULL; + finfo = fopen("/dev/null", "w"); + streamed = true; + } else { + file = argv[optind]; + finfo = stdout; + streamed = false; + } capture_samples(file, card, device, buffer_size, frag, length, rate, channels, format); - printf("Finish capturing... Close Normally\n"); + fprintf(finfo, "Finish capturing... Close Normally\n"); + exit(EXIT_SUCCESS); } -- 1.7.2.5

2 4

[alsa-devel] [PATCH 0/4] Armada 38x and 370 audio
by Marcin Wojtas 04 Mar '15

04 Mar '15

Hi, I send a series of patches adding I2S - S/PDIF support for Marvell Armada 38x SoC. It is same controller as in Armada 370, however two quirks had to be taken into consideration: * new PLL frequency setting * exceptive pinout of I2S and S/PDIF formats Kirkwood-i2s driver was enhanced with obtaining mem resources by name. Sound interface was enabled for Armada 385 DB board, using simple-card DT binding for both I2S and S/PDIF. In addition to that minor fix was applied for Armada 370 audio DT description (stable-CC'ed). Any comments and remarks are welcome. Best regards, Marcin Marcin Wojtas (4): ASoC: kirkwood: enable Kirkwood driver for Armada 38x platforms ARM: mvebu: add audio I2S controller to Armada 38x Device Tree ARM: mvebu: add audio support to Armada 385 DB ARM: mvebu: fix routing for analog audio input of Armada 370 DB platform .../devicetree/bindings/sound/mvebu-audio.txt | 14 ++- arch/arm/boot/dts/armada-370-db.dts | 2 +- arch/arm/boot/dts/armada-388-db.dts | 69 +++++++++++ arch/arm/boot/dts/armada-38x.dtsi | 17 +++ sound/soc/kirkwood/kirkwood-i2s.c | 132 ++++++++++++++++++++- sound/soc/kirkwood/kirkwood.h | 2 + 6 files changed, 232 insertions(+), 4 deletions(-) -- 1.8.3.1

3 13

[alsa-devel] [PATCH 1/2] ucm: document some standard values
by han.lu＠intel.com 04 Mar '15

04 Mar '15

From: "Lu, Han" <han.lu(a)intel.com> add standard value JackDev, JackControl and JackHWMute for speaker mute control on jack insertion. These values will be applied on pulseaudio. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/include/use-case.h b/include/use-case.h index 697377a..3728906 100644 --- a/include/use-case.h +++ b/include/use-case.h @@ -284,6 +284,10 @@ int snd_use_case_get_list(snd_use_case_mgr_t *uc_mgr, * PlaybackMixerID - mixer playback ID * CaptureMixer - name of capture mixer * CaptureMixerID - mixer capture ID + * JackDev - jack device name + * JackControl - jack control name + * JackHWMute - indicate if the HW mutes a device on jack insertion + * or not. */ int snd_use_case_get(snd_use_case_mgr_t *uc_mgr, const char *identifier, -- 2.1.0

2 2

[alsa-devel] [PATCH] ALSA: Fix spelling typo in Documentation/DocBook/alsa-driver-api.xml
by Masanari Iida 04 Mar '15

04 Mar '15

This patch fix spelling typo found in alsa-driver-api.xml. It is because this file is generated from comments in source files, I have to fix source files. --- include/sound/compress_driver.h | 4 ++-- include/sound/control.h | 2 +- include/sound/soc.h | 2 +- include/uapi/sound/compress_offload.h | 2 +- sound/core/pcm_dmaengine.c | 4 ++-- 5 files changed, 7 insertions(+), 7 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f48089d..ffafda5b 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -70,7 +70,7 @@ struct snd_compr_runtime { * @device: device pointer * @direction: stream direction, playback/recording * @metadata_set: metadata set flag, true when set - * @next_track: has userspace signall next track transistion, true when set + * @next_track: has userspace signal next track transition, true when set * @private_data: pointer to DSP private data */ struct snd_compr_stream { @@ -95,7 +95,7 @@ struct snd_compr_stream { * and the stream properties * @get_params: retrieve the codec parameters, mandatory * @set_metadata: Set the metadata values for a stream - * @get_metadata: retreives the requested metadata values from stream + * @get_metadata: retrieves the requested metadata values from stream * @trigger: Trigger operations like start, pause, resume, drain, stop. * This callback is mandatory * @pointer: Retrieve current h/w pointer information. Mandatory diff --git a/include/sound/control.h b/include/sound/control.h index 75f3054..95aad6d 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -227,7 +227,7 @@ snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) * Add a virtual slave control to the given master. * Unlike snd_ctl_add_slave(), the element added via this function * is supposed to have volatile values, and get callback is called - * at each time quried from the master. + * at each time queried from the master. * * When the control peeks the hardware values directly and the value * can be changed by other means than the put callback of the element, diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade1..cf0bb15 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1469,7 +1469,7 @@ static inline struct snd_soc_codec *snd_soc_kcontrol_codec( } /** - * snd_soc_kcontrol_platform() - Returns the platform that registerd the control + * snd_soc_kcontrol_platform() - Returns the platform that registered the control * @kcontrol: The control for which to get the platform * * Note: This function will only work correctly if the control has been diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 22ed8cb..e00d8cb 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -75,7 +75,7 @@ struct snd_compr_tstamp { /** * struct snd_compr_avail - avail descriptor * @avail: Number of bytes available in ring buffer for writing/reading - * @tstamp: timestamp infomation + * @tstamp: timestamp information */ struct snd_compr_avail { __u64 avail; diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 6542c40..fba365a 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -289,7 +289,7 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_request_channel); * * The function should usually be called from the pcm open callback. Note that * this function will use private_data field of the substream's runtime. So it - * is not availabe to your pcm driver implementation. + * is not available to your pcm driver implementation. */ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, struct dma_chan *chan) @@ -328,7 +328,7 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open); * This function will request a DMA channel using the passed filter function and * data. The function should usually be called from the pcm open callback. Note * that this function will use private_data field of the substream's runtime. So - * it is not availabe to your pcm driver implementation. + * it is not available to your pcm driver implementation. */ int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream, dma_filter_fn filter_fn, void *filter_data) -- 2.3.1.167.g7f4ba4b

3 3

[alsa-devel] [PATCH 00/10] initial clkdev cleanups
by Russell King - ARM Linux 04 Mar '15

04 Mar '15

Here's some initial clkdev cleanups. These are targetted for the next merge window, and while the initial patches can be merged independently, I'd prefer to keep the series together as further work on solving the problems which unique struct clk's has introduced is needed. The initial cleanups are more about using the correct clkdev function than anything else: there's no point interating over a array of clk_lookup structs, adding each one in turn when we have had a function which does this since forever. I'm also killing a chunk of seemingly unused code in the omap3isp driver. Lastly, I'm introducing a clkdev_create() helper, which combines the clkdev_alloc() + clkdev_add() pattern which keeps cropping up. Individual patches copied to appropriate people, but they will all appear on the mailing lists. arch/arm/mach-lpc32xx/clock.c | 5 +-- arch/arm/mach-omap2/clkt2xxx_virt_prcm_set.c | 12 ++----- arch/arm/mach-omap2/omap_device.c | 24 +++++-------- arch/arm/plat-orion/common.c | 6 +--- arch/sh/kernel/cpu/sh4a/clock-sh7734.c | 3 +- arch/sh/kernel/cpu/sh4a/clock-sh7757.c | 4 +-- arch/sh/kernel/cpu/sh4a/clock-sh7785.c | 4 +-- arch/sh/kernel/cpu/sh4a/clock-sh7786.c | 4 +-- arch/sh/kernel/cpu/sh4a/clock-shx3.c | 4 +-- drivers/clk/clk-s2mps11.c | 4 +-- drivers/clk/clkdev.c | 52 +++++++++++++++++++++------- drivers/clk/versatile/clk-impd1.c | 4 +-- drivers/media/platform/omap3isp/isp.c | 18 ---------- drivers/media/platform/omap3isp/isp.h | 1 - include/linux/clkdev.h | 3 ++ include/media/omap3isp.h | 6 ---- sound/soc/sh/migor.c | 3 +- 17 files changed, 68 insertions(+), 89 deletions(-) -- FTTC broadband for 0.8mile line: currently at 10.5Mbps down 400kbps up according to speedtest.net.

10 26

[alsa-devel] [patch] ALSA: seq_midi_emul: small array underflow
by Dan Carpenter 03 Mar '15

03 Mar '15

In snd_opl3_calc_pitch() then the limit is: if (pitchbend > 0x1FFF) pitchbend = 0x1FFF; But it can underflow meaning that segment can be as low as SHORT_MIN / 0x1000 and we can read 6 elements before the start of the opl3_note_table[] array. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/include/sound/seq_midi_emul.h b/include/sound/seq_midi_emul.h index 8139d8c..c02b840 100644 --- a/include/sound/seq_midi_emul.h +++ b/include/sound/seq_midi_emul.h @@ -44,7 +44,7 @@ struct snd_midi_channel { unsigned char midi_aftertouch; /* Aftertouch (key pressure) */ unsigned char midi_pressure; /* Channel pressure */ unsigned char midi_program; /* Instrument number */ - short midi_pitchbend; /* Pitch bend amount */ + unsigned short midi_pitchbend; /* Pitch bend amount */ unsigned char control[128]; /* Current value of all controls */ unsigned char note[128]; /* Current status for all notes */

2 5

[alsa-devel] [PATCH] ASoC: simple-card: Add a NULL pointer check in asoc_simple_card_dai_link_of
by Vishal Thanki 03 Mar '15

03 Mar '15

Make sure devm_kzalloc() succeeds. Signed-off-by: Vishal Thanki <vishalthanki(a)gmail.com> diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index f7c6734..fb550b5 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -372,6 +372,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, strlen(dai_link->cpu_dai_name) + strlen(dai_link->codec_dai_name) + 2, GFP_KERNEL); + if (!name) { + ret = -ENOMEM; + goto dai_link_of_err; + } + sprintf(name, "%s-%s", dai_link->cpu_dai_name, dai_link->codec_dai_name); dai_link->name = dai_link->stream_name = name; --

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt5670: Add disabled item in dmic pin enum
by Bard Liao 03 Mar '15

03 Mar '15

Currently, we will configure dmic related pin definition if pdata.dmic_en is true. However, there is no disable option in the enum. So, any dmic is used, all 3 dmic related pins will be configured. It may cause unexpected pin definition. This patch adds a disable item for each dmic enum and take it as default. So the driver will not set the pin configuration if we don't set dmicx_data_pin in platform data. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.h | 3 +++ 1 file changed, 3 insertions(+) mode change 100755 => 100644 sound/soc/codecs/rt5670.h diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h old mode 100755 new mode 100644 index 21f8e18..0a67adb --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1950,17 +1950,20 @@ enum { }; enum { + RT5670_DMIC1_DISABLED, RT5670_DMIC_DATA_GPIO6, RT5670_DMIC_DATA_IN2P, RT5670_DMIC_DATA_GPIO7, }; enum { + RT5670_DMIC2_DISABLED, RT5670_DMIC_DATA_GPIO8, RT5670_DMIC_DATA_IN3N, }; enum { + RT5670_DMIC3_DISABLED, RT5670_DMIC_DATA_GPIO9, RT5670_DMIC_DATA_GPIO10, RT5670_DMIC_DATA_GPIO5, -- 1.8.1.1.439.g50a6b54

2 4

[alsa-devel] [PATCH 1/2] ASoC: wm8804: Fix small issues in probe error path
by Charles Keepax 02 Mar '15

02 Mar '15

The regulator notifiers were not being cleared up on the error path in wm8804_probe and the nothing was being cleared up if snd_soc_register_codec failed. This patch fixes these issues. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8804.c | 18 +++++++++++++++--- 1 files changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 1bd4ace..6131c2a 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -607,6 +607,7 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) dev_err(dev, "Failed to register regulator notifier: %d\n", ret); + return ret; } } @@ -614,7 +615,7 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) wm8804->supplies); if (ret) { dev_err(dev, "Failed to enable supplies: %d\n", ret); - goto err_reg_enable; + goto err_reg_notify; } ret = regmap_read(regmap, WM8804_RST_DEVID1, &id1); @@ -651,11 +652,22 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) goto err_reg_enable; } - return snd_soc_register_codec(dev, &soc_codec_dev_wm8804, - &wm8804_dai, 1); + ret = snd_soc_register_codec(dev, &soc_codec_dev_wm8804, + &wm8804_dai, 1); + if (ret < 0) { + dev_err(dev, "Failed to register CODEC: %d\n", ret); + goto err_reg_enable; + } + + return 0; err_reg_enable: regulator_bulk_disable(ARRAY_SIZE(wm8804->supplies), wm8804->supplies); +err_reg_notify: + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); ++i) + regulator_unregister_notifier(wm8804->supplies[i].consumer, + &wm8804->disable_nb[i]); + return ret; } EXPORT_SYMBOL_GPL(wm8804_probe); -- 1.7.2.5

2 5

[alsa-devel] [PATCH] ASoC: sam9g20_wm8731: drop machine_is_xxx
by Alexandre Belloni 02 Mar '15

02 Mar '15

Atmel based boards can now only be used with device tree. Drop non DT initialization. Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- sound/soc/atmel/sam9g20_wm8731.c | 68 ++++++++++++++++++---------------------- 1 file changed, 31 insertions(+), 37 deletions(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 531728975bbb..639a85d0999c 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -46,8 +46,6 @@ #include <sound/pcm_params.h> #include <sound/soc.h> -#include <asm/mach-types.h> - #include "../codecs/wm8731.h" #include "atmel-pcm.h" #include "atmel_ssc_dai.h" @@ -197,9 +195,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) int ret; if (!np) { - if (!(machine_is_at91sam9g20ek() || - machine_is_at91sam9g20ek_2mmc())) - return -ENODEV; + return -ENODEV; } ret = atmel_ssc_set_audio(0); @@ -236,39 +232,37 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) card->dev = &pdev->dev; /* Parse device node info */ - if (np) { - ret = snd_soc_of_parse_card_name(card, "atmel,model"); - if (ret) - goto err; - - ret = snd_soc_of_parse_audio_routing(card, - "atmel,audio-routing"); - if (ret) - goto err; - - /* Parse codec info */ - at91sam9g20ek_dai.codec_name = NULL; - codec_np = of_parse_phandle(np, "atmel,audio-codec", 0); - if (!codec_np) { - dev_err(&pdev->dev, "codec info missing\n"); - return -EINVAL; - } - at91sam9g20ek_dai.codec_of_node = codec_np; - - /* Parse dai and platform info */ - at91sam9g20ek_dai.cpu_dai_name = NULL; - at91sam9g20ek_dai.platform_name = NULL; - cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); - if (!cpu_np) { - dev_err(&pdev->dev, "dai and pcm info missing\n"); - return -EINVAL; - } - at91sam9g20ek_dai.cpu_of_node = cpu_np; - at91sam9g20ek_dai.platform_of_node = cpu_np; - - of_node_put(codec_np); - of_node_put(cpu_np); + ret = snd_soc_of_parse_card_name(card, "atmel,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, + "atmel,audio-routing"); + if (ret) + goto err; + + /* Parse codec info */ + at91sam9g20ek_dai.codec_name = NULL; + codec_np = of_parse_phandle(np, "atmel,audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "codec info missing\n"); + return -EINVAL; + } + at91sam9g20ek_dai.codec_of_node = codec_np; + + /* Parse dai and platform info */ + at91sam9g20ek_dai.cpu_dai_name = NULL; + at91sam9g20ek_dai.platform_name = NULL; + cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "dai and pcm info missing\n"); + return -EINVAL; } + at91sam9g20ek_dai.cpu_of_node = cpu_np; + at91sam9g20ek_dai.platform_of_node = cpu_np; + + of_node_put(codec_np); + of_node_put(cpu_np); ret = snd_soc_register_card(card); if (ret) { -- 2.1.0

2 8

[alsa-devel] Issue with snd-aloop and alsaloop in my embedded linux custom board
by Srinivasan S 02 Mar '15

02 Mar '15

Dear Lars, I hope you were providing solutions for alsa issues, I am running out of the below issue, hope could you please help me in resolving this issue or could you please redirect to the respective links where my queries can be posted & get the solutions for the problem The below is the virtual cards created after configuring snd-aloop in the linux kernel snd-aloop provided 2 pass-through devices as shown in the below card 0, device 0 card 0, device 1 But am unable to perform the loop back ie., what ever am playing am unable to record in the virtual device As per the logs below, am using the below commands to perform loopback, could you please let me know why am unable to perform the loopback with the below commands or please let me know am I missing any configurations aplay -D hw:0,0,0 play.wav arecord -D hw:0,1,0 record.wav root@am335x-evm:/# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 1: UDA1345TS [TI UDA1345TS], device 0: UDA134x uda134x-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 root@am335x-evm:/# Kindly do the needful as early as possible, Many Thanks in advance

1 1

[alsa-devel] [PATCH] ASoC: wm8731: let codec to manage clock by itself
by Bo Shen 02 Mar '15

02 Mar '15

Let the wm8731 codec to manage clock by itself. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8731.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index b115ed8..ecd8424 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -13,6 +13,7 @@ * published by the Free Software Foundation. */ +#include <linux/clk.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> @@ -45,6 +46,7 @@ static const char *wm8731_supply_names[WM8731_NUM_SUPPLIES] = { /* codec private data */ struct wm8731_priv { struct regmap *regmap; + struct clk *mclk; struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES]; const struct snd_pcm_hw_constraint_list *constraints; unsigned int sysclk; @@ -389,6 +391,9 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, switch (clk_id) { case WM8731_SYSCLK_XTAL: case WM8731_SYSCLK_MCLK: + if (IS_ENABLED(CONFIG_COMMON_CLK)) + if (clk_set_rate(wm8731->mclk, freq)) + return -EINVAL; wm8731->sysclk_type = clk_id; break; default: @@ -490,6 +495,9 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + if (IS_ENABLED(CONFIG_COMMON_CLK)) + if (clk_prepare_enable(wm8731->mclk)) + return -EINVAL; break; case SND_SOC_BIAS_PREPARE: break; @@ -508,6 +516,8 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: + if (IS_ENABLED(CONFIG_COMMON_CLK)) + clk_disable_unprepare(wm8731->mclk); snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); @@ -666,6 +676,15 @@ static int wm8731_spi_probe(struct spi_device *spi) if (wm8731 == NULL) return -ENOMEM; + if (IS_ENABLED(CONFIG_COMMON_CLK)) { + wm8731->mclk = devm_clk_get(&spi->dev, "mclk"); + if (IS_ERR(wm8731->mclk)) { + ret = PTR_ERR(wm8731->mclk); + dev_err(&spi->dev, "Failed to get MCLK\n"); + return ret; + } + } + mutex_init(&wm8731->lock); wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap); @@ -717,6 +736,15 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, if (wm8731 == NULL) return -ENOMEM; + if (IS_ENABLED(CONFIG_COMMON_CLK)) { + wm8731->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(wm8731->mclk)) { + ret = PTR_ERR(wm8731->mclk); + dev_err(&i2c->dev, "Failed to get MCLK\n"); + return ret; + } + } + mutex_init(&wm8731->lock); wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); -- 2.3.0.rc0

3 4

[alsa-devel] [PATCH v2 0/3] Switch to ioremap_wc in the SRAM allocator code
by Abhilash Kesavan 01 Mar '15

01 Mar '15

Fix alignment faults seen during play-back of files with specific sampling rates such as 44.1K. This is based on the discussion here: http://www.spinics.net/lists/arm-kernel/msg384647.html Patch 1 is needed as m32r arch does not define ioremap_wc. Patch 2 adds a resource managed helper function for ioremap_wc which is used in patch 3. Changes since v1: - Fix the compilation error with m32r due to missing ioremap_wc definition. - Fixed minor indentation issues. Note: Other architectures either have ioremap_wc defined or are including generic "io.h"/"iomap.h". c6x arch is the only one I am not sure of. Abhilash Kesavan (3): m32r: add definition of ioremap_wc to io.h lib: devres: add a helper function for ioremap_wc misc: sram: switch to ioremap_wc from ioremap Documentation/driver-model/devres.txt | 1 + arch/m32r/include/asm/io.h | 1 + drivers/misc/sram.c | 17 ++++++++++++++--- include/linux/io.h | 2 ++ lib/devres.c | 28 ++++++++++++++++++++++++++++ 5 files changed, 46 insertions(+), 3 deletions(-) -- 1.7.9.5

5 7

[alsa-devel] [patch] emu1010: avoid pop noises
by Mihail Zenkov 28 Feb '15

28 Feb '15

Fix pop noises on startup and on poweroff. Signed-off-by: Mihail Zenkov <mihail.zenkov(a)gmail.com> --- linux-3.19/include/sound/emu10k1.h +++ linux-3.19-1/include/sound/emu10k1.h @@ -1809,6 +1809,8 @@ uint subsystem, struct snd_emu10k1 ** remu); +void snd_emu10k1_shutdown(struct pci_dev *pci); + int snd_emu10k1_pcm(struct snd_emu10k1 * emu, int device, struct snd_pcm ** rpcm); int snd_emu10k1_pcm_mic(struct snd_emu10k1 * emu, int device, struct snd_pcm ** rpcm); int snd_emu10k1_pcm_efx(struct snd_emu10k1 * emu, int device, struct snd_pcm ** rpcm); --- linux-3.19/sound/pci/emu10k1/emu10k1.c +++ linux-3.19-1/sound/pci/emu10k1/emu10k1.c @@ -282,6 +282,7 @@ .driver = { .pm = SND_EMU10K1_PM_OPS, }, + .shutdown = snd_emu10k1_shutdown, }; module_pci_driver(emu10k1_driver); --- linux-3.19/sound/pci/emu10k1/emu10k1_main.c +++ linux-3.19-1/sound/pci/emu10k1/emu10k1_main.c @@ -361,6 +361,9 @@ /* Enable analog/digital outs on audigy */ if (emu->card_capabilities->emu_model) { ; /* Disable all access to A_IOCFG for the emu1010 */ + /* Wait before unmute to avoid pop noise */ + msleep_interruptible(500); + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); /* Unmute all */ } else if (emu->card_capabilities->i2c_adc) { ; /* Disable A_IOCFG for Audigy 2 ZS Notebook */ } else if (emu->audigy) { @@ -1097,7 +1100,6 @@ EMU_DST_ALICE_I2S2_LEFT, EMU_SRC_DOCK_ADC3_LEFT1); snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_ALICE_I2S2_RIGHT, EMU_SRC_DOCK_ADC3_RIGHT1); - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x01); /* Unmute all */ snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp); @@ -1253,17 +1255,12 @@ /* snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); */ /* Output spdif */ /* TEMP: Select 48kHz SPDIF out */ - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x0); /* Mute all */ snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 0x0); /* Default fallback clock 48kHz */ /* Word Clock source, Internal 48kHz x1 */ snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K); /* snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X); */ emu->emu1010.internal_clock = 1; /* 48000 */ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12); /* Set LEDs on Audio Dock */ - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x1); /* Unmute all */ - /* snd_emu1010_fpga_write(emu, 0x7, 0x0); */ /* Mute all */ - /* snd_emu1010_fpga_write(emu, 0x7, 0x1); */ /* Unmute all */ - /* snd_emu1010_fpga_write(emu, 0xe, 0x12); */ /* Set LEDs on Audio Dock */ return 0; } @@ -2142,3 +2139,16 @@ } } #endif + +void snd_emu10k1_shutdown(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct snd_emu10k1 *emu = card->private_data; + + if (emu->card_capabilities->emu_model) { + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE); /* Mute all */ + /* Clean FPGA to avoid pop noise on shutdown */ + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0x02); + } +} +EXPORT_SYMBOL(snd_emu10k1_shutdown);

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm8804: Split out bus drivers
by Charles Keepax 28 Feb '15

28 Feb '15

From: Charles Keepax <ckeepax(a)gmail.com> Simplify dependencies by using new style split out bus interfaces. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 18 +++++- sound/soc/codecs/Makefile | 4 + sound/soc/codecs/wm8804-i2c.c | 64 +++++++++++++++++++ sound/soc/codecs/wm8804-spi.c | 56 ++++++++++++++++ sound/soc/codecs/wm8804.c | 139 +++++------------------------------------ sound/soc/codecs/wm8804.h | 7 ++ 6 files changed, 162 insertions(+), 126 deletions(-) create mode 100644 sound/soc/codecs/wm8804-i2c.c create mode 100644 sound/soc/codecs/wm8804-spi.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ea9f0e3..0bddd92 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -141,7 +141,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8770 if SPI_MASTER select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8782 - select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8804_I2C if I2C + select SND_SOC_WM8804_SPI if SPI_MASTER select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C select SND_SOC_WM8904 if I2C @@ -744,8 +745,19 @@ config SND_SOC_WM8782 tristate config SND_SOC_WM8804 - tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver" - depends on SND_SOC_I2C_AND_SPI + tristate + +config SND_SOC_WM8804_I2C + tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver I2C" + depends on I2C + select SND_SOC_WM8804 + select REGMAP_I2C + +config SND_SOC_WM8804_SPI + tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver SPI" + depends on SPI_MASTER + select SND_SOC_WM8804 + select REGMAP_SPI config SND_SOC_WM8900 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 69b8666..7acb6c1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -145,6 +145,8 @@ snd-soc-wm8770-objs := wm8770.o snd-soc-wm8776-objs := wm8776.o snd-soc-wm8782-objs := wm8782.o snd-soc-wm8804-objs := wm8804.o +snd-soc-wm8804-i2c-objs := wm8804-i2c.o +snd-soc-wm8804-spi-objs := wm8804-spi.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8904-objs := wm8904.o @@ -323,6 +325,8 @@ obj-$(CONFIG_SND_SOC_WM8770) += snd-soc-wm8770.o obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o obj-$(CONFIG_SND_SOC_WM8782) += snd-soc-wm8782.o obj-$(CONFIG_SND_SOC_WM8804) += snd-soc-wm8804.o +obj-$(CONFIG_SND_SOC_WM8804_I2C) += snd-soc-wm8804-i2c.o +obj-$(CONFIG_SND_SOC_WM8804_SPI) += snd-soc-wm8804-spi.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8904) += snd-soc-wm8904.o diff --git a/sound/soc/codecs/wm8804-i2c.c b/sound/soc/codecs/wm8804-i2c.c new file mode 100644 index 0000000..5bd4af2 --- /dev/null +++ b/sound/soc/codecs/wm8804-i2c.c @@ -0,0 +1,64 @@ +/* + * wm8804-i2c.c -- WM8804 S/PDIF transceiver driver - I2C + * + * Copyright 2015 Cirrus Logic Inc + * + * Author: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/i2c.h> + +#include "wm8804.h" + +static int wm8804_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_i2c(i2c, &wm8804_regmap_config); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return wm8804_probe(&i2c->dev, regmap); +} + +static int wm8804_i2c_remove(struct i2c_client *i2c) +{ + wm8804_remove(&i2c->dev); + return 0; +} + +static const struct i2c_device_id wm8804_i2c_id[] = { + { "wm8804", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8804_i2c_id); + +static const struct of_device_id wm8804_of_match[] = { + { .compatible = "wlf,wm8804", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8804_of_match); + +static struct i2c_driver wm8804_i2c_driver = { + .driver = { + .name = "wm8804", + .owner = THIS_MODULE, + .of_match_table = wm8804_of_match, + }, + .probe = wm8804_i2c_probe, + .remove = wm8804_i2c_remove, + .id_table = wm8804_i2c_id +}; + +module_i2c_driver(wm8804_i2c_driver); + +MODULE_DESCRIPTION("ASoC WM8804 driver - I2C"); +MODULE_AUTHOR("Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8804-spi.c b/sound/soc/codecs/wm8804-spi.c new file mode 100644 index 0000000..287e11e --- /dev/null +++ b/sound/soc/codecs/wm8804-spi.c @@ -0,0 +1,56 @@ +/* + * wm8804-spi.c -- WM8804 S/PDIF transceiver driver - SPI + * + * Copyright 2015 Cirrus Logic Inc + * + * Author: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/spi/spi.h> + +#include "wm8804.h" + +static int wm8804_spi_probe(struct spi_device *spi) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_spi(spi, &wm8804_regmap_config); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return wm8804_probe(&spi->dev, regmap); +} + +static int wm8804_spi_remove(struct spi_device *spi) +{ + wm8804_remove(&spi->dev); + return 0; +} + +static const struct of_device_id wm8804_of_match[] = { + { .compatible = "wlf,wm8804", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8804_of_match); + +static struct spi_driver wm8804_spi_driver = { + .driver = { + .name = "wm8804", + .owner = THIS_MODULE, + .of_match_table = wm8804_of_match, + }, + .probe = wm8804_spi_probe, + .remove = wm8804_spi_remove +}; + +module_spi_driver(wm8804_spi_driver); + +MODULE_DESCRIPTION("ASoC WM8804 driver - SPI"); +MODULE_AUTHOR("Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index b2b0e68..b5a04fc 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -15,10 +15,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> -#include <linux/i2c.h> #include <linux/of_device.h> -#include <linux/spi/spi.h> -#include <linux/regmap.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> #include <sound/core.h> @@ -518,7 +515,7 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm8804_remove(struct snd_soc_codec *codec) +static int wm8804_codec_remove(struct snd_soc_codec *codec) { struct wm8804_priv *wm8804; int i; @@ -531,7 +528,7 @@ static int wm8804_remove(struct snd_soc_codec *codec) return 0; } -static int wm8804_probe(struct snd_soc_codec *codec) +static int wm8804_codec_probe(struct snd_soc_codec *codec) { struct wm8804_priv *wm8804; int i, id1, id2, ret; @@ -649,8 +646,8 @@ static struct snd_soc_dai_driver wm8804_dai = { }; static const struct snd_soc_codec_driver soc_codec_dev_wm8804 = { - .probe = wm8804_probe, - .remove = wm8804_remove, + .probe = wm8804_codec_probe, + .remove = wm8804_codec_remove, .set_bias_level = wm8804_set_bias_level, .idle_bias_off = true, @@ -658,13 +655,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .num_controls = ARRAY_SIZE(wm8804_snd_controls), }; -static const struct of_device_id wm8804_of_match[] = { - { .compatible = "wlf,wm8804", }, - { } -}; -MODULE_DEVICE_TABLE(of, wm8804_of_match); - -static const struct regmap_config wm8804_regmap_config = { +const struct regmap_config wm8804_regmap_config = { .reg_bits = 8, .val_bits = 8, @@ -675,128 +666,30 @@ static const struct regmap_config wm8804_regmap_config = { .reg_defaults = wm8804_reg_defaults, .num_reg_defaults = ARRAY_SIZE(wm8804_reg_defaults), }; +EXPORT_SYMBOL_GPL(wm8804_regmap_config); -#if defined(CONFIG_SPI_MASTER) -static int wm8804_spi_probe(struct spi_device *spi) +int wm8804_probe(struct device *dev, struct regmap *regmap) { struct wm8804_priv *wm8804; - int ret; - wm8804 = devm_kzalloc(&spi->dev, sizeof *wm8804, GFP_KERNEL); + wm8804 = devm_kzalloc(dev, sizeof(*wm8804), GFP_KERNEL); if (!wm8804) return -ENOMEM; - wm8804->regmap = devm_regmap_init_spi(spi, &wm8804_regmap_config); - if (IS_ERR(wm8804->regmap)) { - ret = PTR_ERR(wm8804->regmap); - return ret; - } - - spi_set_drvdata(spi, wm8804); + dev_set_drvdata(dev, wm8804); - ret = snd_soc_register_codec(&spi->dev, - &soc_codec_dev_wm8804, &wm8804_dai, 1); - - return ret; -} - -static int wm8804_spi_remove(struct spi_device *spi) -{ - snd_soc_unregister_codec(&spi->dev); - return 0; -} - -static struct spi_driver wm8804_spi_driver = { - .driver = { - .name = "wm8804", - .owner = THIS_MODULE, - .of_match_table = wm8804_of_match, - }, - .probe = wm8804_spi_probe, - .remove = wm8804_spi_remove -}; -#endif - -#if IS_ENABLED(CONFIG_I2C) -static int wm8804_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) -{ - struct wm8804_priv *wm8804; - int ret; - - wm8804 = devm_kzalloc(&i2c->dev, sizeof *wm8804, GFP_KERNEL); - if (!wm8804) - return -ENOMEM; + wm8804->regmap = regmap; - wm8804->regmap = devm_regmap_init_i2c(i2c, &wm8804_regmap_config); - if (IS_ERR(wm8804->regmap)) { - ret = PTR_ERR(wm8804->regmap); - return ret; - } - - i2c_set_clientdata(i2c, wm8804); - - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_wm8804, &wm8804_dai, 1); - return ret; -} - -static int wm8804_i2c_remove(struct i2c_client *i2c) -{ - snd_soc_unregister_codec(&i2c->dev); - return 0; -} - -static const struct i2c_device_id wm8804_i2c_id[] = { - { "wm8804", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, wm8804_i2c_id); - -static struct i2c_driver wm8804_i2c_driver = { - .driver = { - .name = "wm8804", - .owner = THIS_MODULE, - .of_match_table = wm8804_of_match, - }, - .probe = wm8804_i2c_probe, - .remove = wm8804_i2c_remove, - .id_table = wm8804_i2c_id -}; -#endif - -static int __init wm8804_modinit(void) -{ - int ret = 0; - -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&wm8804_i2c_driver); - if (ret) { - printk(KERN_ERR "Failed to register wm8804 I2C driver: %d\n", - ret); - } -#endif -#if defined(CONFIG_SPI_MASTER) - ret = spi_register_driver(&wm8804_spi_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register wm8804 SPI driver: %d\n", - ret); - } -#endif - return ret; + return snd_soc_register_codec(dev, &soc_codec_dev_wm8804, + &wm8804_dai, 1); } -module_init(wm8804_modinit); +EXPORT_SYMBOL_GPL(wm8804_probe); -static void __exit wm8804_exit(void) +void wm8804_remove(struct device *dev) { -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&wm8804_i2c_driver); -#endif -#if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&wm8804_spi_driver); -#endif + snd_soc_unregister_codec(dev); } -module_exit(wm8804_exit); +EXPORT_SYMBOL_GPL(wm8804_remove); MODULE_DESCRIPTION("ASoC WM8804 driver"); MODULE_AUTHOR("Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com>"); diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index e72d4f4..a39a256 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -13,6 +13,8 @@ #ifndef _WM8804_H #define _WM8804_H +#include <linux/regmap.h> + /* * Register values. */ @@ -62,4 +64,9 @@ #define WM8804_MCLKDIV_256FS 0 #define WM8804_MCLKDIV_128FS 1 +extern const struct regmap_config wm8804_regmap_config; + +int wm8804_probe(struct device *dev, struct regmap *regmap); +void wm8804_remove(struct device *dev); + #endif /* _WM8804_H */ -- 1.7.2.5

1 2

[alsa-devel] [PATCH 0/7] HD-audio regmap support
by Takashi Iwai 28 Feb '15

28 Feb '15

Hi, this patch adds the regmap support to HD-audio and convert the house-made amp/verb caches to regmap. Because of the asymmetrical nature of HD-audio verbs, the adaption of regmap isn't straightforward at all, and has many limitations. But this still reduces the complexity in struct hda_codec a bit. The patches are applied on top of the previous patch series (found in topic/hda-bus branch of sound git tree). This patchset is also found in sound git tree topic/hda-regmap branch. Takashi === Takashi Iwai (7): ALSA: hda - Add regmap support ALSA: hda - Use regmap for codec parameter reads ALSA: hda - Use regmap for amp accesses ALSA: hda - Use regmap for parameter caches, too ALSA: hda - Allow driver to add vendor-specific verbs for regmap ALSA: hda - Use regmap for command verb caches, too ALSA: hda - Reduce verbs during generic parser initialization sound/pci/hda/Kconfig | 1 + sound/pci/hda/Makefile | 3 +- sound/pci/hda/hda_beep.c | 3 +- sound/pci/hda/hda_bind.c | 6 + sound/pci/hda/hda_codec.c | 645 +++++++---------------------------------- sound/pci/hda/hda_codec.h | 45 +-- sound/pci/hda/hda_generic.c | 32 +- sound/pci/hda/hda_jack.c | 6 +- sound/pci/hda/hda_local.h | 31 +- sound/pci/hda/hda_regmap.c | 409 ++++++++++++++++++++++++++ sound/pci/hda/hda_regmap.h | 38 +++ sound/pci/hda/patch_analog.c | 30 +- sound/pci/hda/patch_conexant.c | 12 +- sound/pci/hda/patch_hdmi.c | 24 +- sound/pci/hda/patch_realtek.c | 11 +- sound/pci/hda/patch_si3054.c | 5 +- sound/pci/hda/patch_sigmatel.c | 38 ++- 17 files changed, 660 insertions(+), 679 deletions(-) create mode 100644 sound/pci/hda/hda_regmap.c create mode 100644 sound/pci/hda/hda_regmap.h -- 2.3.0

2 11

[alsa-devel] [PATCH 00/10] Modernization of HD-audio bus and PM stuff
by Takashi Iwai 28 Feb '15

28 Feb '15

Hi, here is yet another patchset for HD-audio modernization. The device binding is replaced with the standard bus, and the power-save stuff is rewritten with the standard runtime PM. In addition, a few cleanup patches follow. There won't (and shouldn't) be big changes from the functionality POV. This is rather a large restructuring inside. It should work mostly without any visible changes. You can see a bit finer control for runtime PM in sysfs now, for example, but the original interface via power_save option is still kept. Takashi === Takashi Iwai (10): ALSA: hda - Decouple PCM and hwdep devices from codec object ALSA: hda - Bind codecs via standard bus ALSA: hda - Move codec suspend/resume to codec driver ALSA: hda - Use standard runtime PM for codec power-save control ALSA: hda - Drop power_save value indirection in hda_bus ALSA: hda - Replace bus pm_notify with the standard runtime PM framework ALSA: hda - Power down codec automatically at registration ALSA: hda - Set parent of input beep devices ALSA: hda - Remove channel mode helper functions ALSA: hda - Clear pcm pointer assigned to hda_pcm at device removal sound/pci/hda/Makefile | 2 +- sound/pci/hda/hda_beep.c | 2 +- sound/pci/hda/hda_bind.c | 321 ++++++++++++++++++ sound/pci/hda/hda_codec.c | 731 ++++++++--------------------------------- sound/pci/hda/hda_codec.h | 108 ++---- sound/pci/hda/hda_controller.c | 62 ++-- sound/pci/hda/hda_controller.h | 4 +- sound/pci/hda/hda_generic.c | 18 +- sound/pci/hda/hda_hwdep.c | 3 - sound/pci/hda/hda_intel.c | 33 +- sound/pci/hda/hda_local.h | 41 +-- sound/pci/hda/hda_tegra.c | 23 +- sound/pci/hda/hda_trace.h | 24 -- sound/pci/hda/patch_analog.c | 16 +- sound/pci/hda/patch_ca0110.c | 16 +- sound/pci/hda/patch_ca0132.c | 16 +- sound/pci/hda/patch_cirrus.c | 16 +- sound/pci/hda/patch_cmedia.c | 16 +- sound/pci/hda/patch_conexant.c | 16 +- sound/pci/hda/patch_hdmi.c | 27 +- sound/pci/hda/patch_realtek.c | 16 +- sound/pci/hda/patch_si3054.c | 16 +- sound/pci/hda/patch_sigmatel.c | 20 +- sound/pci/hda/patch_via.c | 16 +- 24 files changed, 572 insertions(+), 991 deletions(-) create mode 100644 sound/pci/hda/hda_bind.c -- 2.3.0

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[alsa-devel] [PATCH] ALSA: hda - Make slave strings more user-friendly
by Ingo Brückl 28 Feb '15

28 Feb '15

"Line Out" has been used throughout, so don't abbreviate. Signed-off-by: Ingo Brückl <ib(a)wupperonline.de> diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b680b4e..834730f 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1103,9 +1103,9 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, if (hp_lo_shared && spk_lo_shared) return spec->vmaster_mute.hook ? "PCM" : "Master"; if (hp_lo_shared) - return "Headphone+LO"; + return "Headphone+Line Out"; if (spk_lo_shared) - return "Speaker+LO"; + return "Speaker+Line Out"; } } @@ -4619,7 +4619,7 @@ static const char * const slave_pfxs[] = { "CLFE", "Bass Speaker", "PCM", "Speaker Front", "Speaker Surround", "Speaker CLFE", "Speaker Side", "Headphone Front", "Headphone Surround", "Headphone CLFE", - "Headphone Side", "Headphone+LO", "Speaker+LO", + "Headphone Side", "Headphone+Line Out", "Speaker+Line Out", NULL, }; -- 1.7.10

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[alsa-devel] [PATCH 0/9] Allow hot-unplugging hd-audio codec driver
by Takashi Iwai 28 Feb '15

28 Feb '15

Hi, here is yet another patchset for HD-audio. This time it's about robustifying the dynamic unbind of a codec driver. With these patches, the codec driver can be freely bound / unbound in run time. We had a similar feature via hwdep reconfig sysfs entry, but this is a more generic form. The patchset is applied on top of the previous one, topic/hda-regmap branch of sound git tree. This patchset itself is found in topic/hda-unbind branch. Takashi === Takashi Iwai (9): ALSA: hda - Split snd_hda_build_pcms() ALSA: hda - Add card field to hda_codec struct ALSA: hda - Remove superfluous memory allocation error messages ALSA: hda - Allocate hda_pcm objects dynamically ALSA: hda - Don't assume non-NULL PCM ops ALSA: core: Re-add snd_device_disconnect() ALSA: hda - Implement unbind more safely ALSA: hda - Build PCMs and controls at codec driver probe ALSA: hda - Drop codec driver module refcounting include/sound/core.h | 3 +- sound/core/device.c | 43 +++-- sound/core/init.c | 5 +- sound/pci/hda/hda_beep.c | 4 +- sound/pci/hda/hda_bind.c | 33 ++-- sound/pci/hda/hda_codec.c | 363 ++++++++++++++++++++--------------------- sound/pci/hda/hda_codec.h | 27 ++- sound/pci/hda/hda_controller.c | 53 +++--- sound/pci/hda/hda_generic.c | 28 ++-- sound/pci/hda/hda_generic.h | 2 +- sound/pci/hda/hda_hwdep.c | 2 +- sound/pci/hda/hda_intel.c | 15 +- sound/pci/hda/hda_jack.c | 8 +- sound/pci/hda/hda_local.h | 1 + sound/pci/hda/hda_proc.c | 8 +- sound/pci/hda/hda_sysfs.c | 2 +- sound/pci/hda/hda_tegra.c | 10 -- sound/pci/hda/hda_trace.h | 4 +- sound/pci/hda/patch_ca0132.c | 31 ++-- sound/pci/hda/patch_hdmi.c | 36 ++-- sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_si3054.c | 11 +- sound/pci/hda/patch_via.c | 14 +- 23 files changed, 349 insertions(+), 356 deletions(-) -- 2.3.0

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[alsa-devel] [patch] ALSA: oxfw: fix a condition in start_stream()
by Dan Carpenter 27 Feb '15

27 Feb '15

Static checkers complain that this function doesn't return negative error codes. The comments for amdtp_stream_wait_callback() say that we should stop the stream if it returns false. Fixes: f3699e2c7745 ('ALSA: oxfw: Change the way to start stream') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 29ccb36..ee67581 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -172,7 +172,7 @@ static int start_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream, /* Wait first packet */ err = amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT); - if (err < 0) + if (err == 0) stop_stream(oxfw, stream); end: return err;

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[alsa-devel] [PATCH 1/1] ARM: TLV320AIC23 SoC Audio Codec: Fix errors reported related to input routing signals.
by Gabriel Dobato 27 Feb '15

27 Feb '15

From: dobatog <dobatog(a)gmail.com> The following patch, based on stable v3.19, corrects the errors that are reported in the boot trace in reference to the Input Signals of the audio codec TLV320AIC23: tlv320aic23-codec 1-001a: Control not supported for path LLINEIN -> [NULL] -> Line Input tlv320aic23-codec 1-001a: ASoC: no dapm match for LLINEIN --> NULL --> Line Input tlv320aic23-codec 1-001a: ASoC: Failed to add route LLINEIN -> NULL -> Line Input tlv320aic23-codec 1-001a: Control not supported for path RLINEIN -> [NULL] -> Line Input tlv320aic23-codec 1-001a: ASoC: no dapm match for RLINEIN --> NULL --> Line Input tlv320aic23-codec 1-001a: ASoC: Failed to add route RLINEIN -> NULL -> Line Input tlv320aic23-codec 1-001a: Control not supported for path MICIN -> [NULL] -> Mic Input tlv320aic23-codec 1-001a: ASoC: no dapm match for MICIN --> NULL --> Mic Input tlv320aic23-codec 1-001a: ASoC: Failed to add route MICIN -> NULL -> Mic Input I am trying to set the sound system in a CM-510 (Compulab - SoM) based board with DT, using Simple-Card-Audio and the TLV320AIC23 audio codec included in the SoM . I faced this problem and thanks to the help of Sebastian Hesselbarth who noticed the possible error located in the structure snd_soc_dapm_route tlv320aic23_intercon[] , now errors are not reported. Tested on CM-510 (Compulab SoM Board). Signed-off-by: dobatog <dobatog(a)gmail.com> --- sound/soc/codecs/tlv320aic23.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index cc17e7e..891f3aa 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -174,10 +174,10 @@ static const struct snd_soc_dapm_route tlv320aic23_intercon[] = { {"ROUT", NULL, "Output Mixer"}, /* Inputs */ - {"Line Input", "NULL", "LLINEIN"}, - {"Line Input", "NULL", "RLINEIN"}, + {"Line Input", NULL, "LLINEIN"}, + {"Line Input", NULL, "RLINEIN"}, - {"Mic Input", "NULL", "MICIN"}, + {"Mic Input", NULL, "MICIN"}, /* input mux */ {"Capture Source", "Line", "Line Input"}, -- 1.9.1

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Re: [alsa-devel] ASoC: rt286: add jack detection disable with NULL jack passed
by Dan Carpenter 27 Feb '15

27 Feb '15

Hello Jie Yang, The patch e2cef68d5903: "ASoC: rt286: add jack detection disable with NULL jack passed" from Feb 10, 2015, leads to the following static checker warning: sound/soc/codecs/rt286.c:400 rt286_mic_detect() warn: suspicious bitop condition sound/soc/codecs/rt286.c 392 int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) 393 { 394 struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); 395 396 rt286->jack = jack; 397 398 if (jack) { 399 /* enable IRQ */ 400 if (rt286->jack->status | SND_JACK_HEADPHONE) ^^^^^^^^^^^^^^^^^^^^ You probably intended bitwise AND here? Bitwise OR is always true. 401 snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO1"); 402 regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x2); 403 /* Send an initial empty report */ 404 snd_soc_jack_report(rt286->jack, rt286->jack->status, 405 SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); 406 } else { 407 /* disable IRQ */ 408 regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x0); 409 snd_soc_dapm_disable_pin(&codec->dapm, "LDO1"); 410 } 411 snd_soc_dapm_sync(&codec->dapm); 412 413 return 0; 414 } regards, dan carpenter

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Re: [alsa-devel] ALSA 1.0.29 release
by humbert.olivier.1＠free.fr 26 Feb '15

26 Feb '15

Congratulations to everyone who was involved in this. Merci, Olivier

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[alsa-devel] New alsa-lib release plans?
by Jurgen Kramer 26 Feb '15

26 Feb '15

It has been a while since the last ALSA lib release. What are the plans for 1.0.28.1 or 1.0.29? There are fixes and enhancements since 1.0.28 which would be nice to find in a production release. Jurgen

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Re: [alsa-devel] [ALSA] ca0106: power down SPI DAC channels when not in use
by Dan Carpenter 26 Feb '15

26 Feb '15

Hello Trent Piepho, The patch 485100706b4b: "[ALSA] ca0106: power down SPI DAC channels when not in use" from Jul 25, 2007, leads to the following static checker warning: sound/pci/ca0106/ca0106_main.c:601 snd_ca0106_pcm_open_playback_channel() warn: bool is not less than zero. sound/pci/ca0106/ca0106_main.c 597 598 /* Front channel dac should already be on */ 599 if (channel_id != PCM_FRONT_CHANNEL) { 600 err = snd_ca0106_pcm_power_dac(chip, channel_id, 1); Because of snd_ca0106_spi_write() then this function appears to return zero on success and 1 on failure. That's very strange. None of the callers expect a 1 return. If they care about the return code then they expect negative error codes. 601 if (err < 0) 602 return err; 603 } 604 regards, dan carpenter

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Re: [alsa-devel] Intel HDA / ALC662 analog surround problem
by Raymond Yau 26 Feb '15

26 Feb '15

> > > You have to post the output of alsa-info.sh > > Please see the attached file. 0.655493] sound hdaudioC1D0: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:line [ 0.655494] sound hdaudioC1D0: speaker_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 0.655495] sound hdaudioC1D0: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) [ 0.655496] sound hdaudioC1D0: mono: mono_out=0x0 [ 0.655496] sound hdaudioC1D0: dig-out=0x1e/0x0 [ 0.655497] sound hdaudioC1D0: inputs: [ 0.655498] sound hdaudioC1D0: Front Mic=0x19 [ 0.655499] sound hdaudioC1D0: Rear Mic=0x18 [ 0.655499] sound hdaudioC1D0: Line=0x1a This is a known bug of hda_generic.c for those desktop with internal speaker and three audio jacks at rear panel (e.g. those lenovo thinkcenter with alc66x codec) The driver prefer to assign volume control/dac to headphone, line out and internal speaker instead of line out and the other two multi io jacks The workaround is to change the internal speaker as not connected node 0x15 or You need to increase badness of multi io so that driver select multi io as the best config > > > The channel mode should be automatically added for those desktop with three > > audio jacks (green line out, blue line in and pink mic) at rear panel > > Although I do have these jacks, there unfortunately isn't a channel mode. > > > either your computer BIOS does not set the correct pin default or it can be > > disabled by hint > > Do you really mean "disabled" or do you mean to enable them by hint? > > As you can see I don't have a hwC1D0/hints (and no reconfig), so I cannot try > setting multi_io to make the driver re-configuring itself. Would an early > patching make sense (or maybe I missing a kernel config option for the > hints)? > > If I only knew more about these "pins" (there seem to be 11, from 0x12 to > 0x1e) I could try to add a model for the ALC66x since the existing ones > (mario, asus-mode?, etc) don't make any differences. >

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Re: [alsa-devel] No sound and systemd journal filling when inserting headphones when power-saving is enabled
by Takashi Iwai 26 Feb '15

26 Feb '15

At Tue, 24 Feb 2015 22:26:32 +0000, Dang Sananikone wrote: > > On 24 February 2015 at 20:57, Takashi Iwai <tiwai(a)suse.de> wrote: > > > At Tue, 24 Feb 2015 20:11:58 +0000, > > Dang Sananikone wrote: > > > > > > Thanks, in that case I'll simply attach the files to this post. There are > > > two files: > > > > > > * alsa-info-notworking.txt: This is the alsa info generated when > > > power-saving is enabled and after "speaker-test" has been invoked. > > > * alsa-info-working.txt: This is the alsa info generated when > > power-saving > > > is disabled and after "speaker-test" has been invoked. > > > > > > > > > To reproduce the problem: > > > > > > Prerequisite: > > > Set "options snd_hda_intel power_save=1" > > > > > > Instructions: > > > 1. Boot up laptop. > > > 2. Insert headphone socket into jack. > > > 3. In terminal, type "speaker-test -c 2". > > > > > > Expected Result: > > > The speaker-test program hangs, and the systemd journal starts filling up > > > with "sound hdaudioC0D0: hda-codec: out of range cmd 0:1:716:ffffffff" > > > messages. > > > > OK, then try to pass power_save_controller=0 option to snd-hda-intel > > module. It might be that the controller gets screwed up by some > > reason by power-saving. > > > > If I've understood you correctly I've set the following options: "options > snd_hda_intel power_save=1 power_save_controller=0". > > Then I've plugged in my headphones before invoking speaker-test. With no > audio playing I hear a blip sound every second. Every second? Is this equal with the length you specified to power_save option? i.e. if you pass power_save=10, the frequency of blip noise changes, too? > Then I invoke "speaker-test -c 2" and I hear the speaker test audio I > expect to hear. The program does not hang. After killing the program and > with no audio playing I hear the blip sound every second. So, the problem was basically the HD-audio controller screwed up by the power-saving. BTW, didn't you see the same problem with the explicit suspend/resume (S3 and S4)? For testing it, try a bigger value in power_save so that you go to S3/S4 before the power save is triggered. > > If this still doesn't help, we need more logs. Build the kernel with > > tracing support, and get the HD-audio verb traces as > > Documentation/sound/alsa/HD-Audio.txt, section Tracepoints. > > > > > > Takashi > > > > Going back to this setting I set out to get the HD-audio verbs to provide > more information: "options snd_hda_intel power_save=1". > > I've attached the trace file which was generated after following your > instructions (I think the kernel already had tracing support enabled > judging by the /proc/config.gz config parameters). The trace already shows the bad read from the controller, so it's likely an issue in the HDA controller and/or communication. In anyway, below is a patch to disable runtime PM of the controller again. It's equivalent as passing power_save_controller=1. Could you give it a test? thanks, Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Disable runtime PM for Panther Point again This is essentially a partial revert of the commit [b1920c21102a: 'ALSA: hda - Enable runtime PM on Panther Point']. There was a bug report showing the HD-audio bus hang during runtime PM on HP Spectre XT. Reported-by: Dang Sananikone <dang.sananikone(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 36d2f20db7a4..4ca3d5d02436 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1966,7 +1966,7 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Panther Point */ { PCI_DEVICE(0x8086, 0x1e20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Lynx Point */ { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, -- 2.3.0

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[alsa-devel] About iMX6SL AUDMUX (port 7) and SSI3
by Xuebing Wang 26 Feb '15

26 Feb '15

Hi community: I am working on ASoC based sound driver for our iMX6SL based board. There are 3 I2S peripherals on the board: - iMX6SL pads AUD3 (as in AUD3_TXC), connect to peripheral Codec3 - iMX6SL pads AUD5 connect to Codec5 - iMX6SL pads Aud6, connect to Codec6 Q1) Is internal SSI3 controller (ssi@02030000) actually *hard-wired* to AUDMUX port 7? Q2) Can I configure audmux as below? - External AUD3 -> internal SSI3 (which is actually audmux port 7?) - External AUD5 -> internal SSI2 (ssi@0202c000) - External AUD6 -> internal SSI1 (ssi@02028000) Thanks. -- Xuebing

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[alsa-devel] [PATCH v2 RESEND] ASoC: OMAP: mcbsp: ensure that CLKX and CLKR are not used as ouput pins when they are used as input clock for the SRG.
by Thomas Niederprüm 26 Feb '15

26 Feb '15

This patch fixes faulty behaviour in a setup where the input clock for the SRG is fed through the CLKR/CLKX pin but the McBSP is configured to be master (SND_SOC_DAIFMT_CBS_CFS). In that case of course CLKR/CLKX must not be configured as output pin. Otherwise the input clock is messed up horribly. This patch makes it possible to use the CLKR/CLKX pin rather than CLKS to inject a reference clock in setups where McBSP is master and not both rx and tx are used. However for this to work it has to be ensured that set_dai_sysclk() is called after set_dai_fmt(). This was tested on a beagleboard-xm using McBSP1 to drive a i2s DAC through the tx lines (CLKX,FSX,DX). Using this patch the CLKR pin is used to inject an external reference clock. Signed-off-by: Thomas Niederprüm <niederp(a)physik.uni-kl.de> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- changes since v1: - added comments explaining the bit masking to disable output on CLKR/CLKX sound/soc/omap/omap-mcbsp.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index bd3ef2a..662269c 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -530,8 +530,19 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, case OMAP_MCBSP_SYSCLK_CLKX_EXT: regs->srgr2 |= CLKSM; + regs->pcr0 |= SCLKME; + /* + * If McBSP is master but yet the CLKX/CLKR pin drives the SRG, + * disable output on those pins. This enables to inject the + * reference clock through CLKX/CLKR. For this to work + * set_dai_sysclk() _needs_ to be called after set_dai_fmt(). + */ + regs->pcr0 &= ~CLKXM; + break; case OMAP_MCBSP_SYSCLK_CLKR_EXT: regs->pcr0 |= SCLKME; + /* Disable ouput on CLKR pin in master mode */ + regs->pcr0 &= ~CLKRM; break; default: err = -ENODEV; -- 2.1.1

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[alsa-devel] [Patch] ASoC: max98357a: Use standard DAI names
by Kenneth Westfield 26 Feb '15

26 Feb '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> Use the standard naming convention for the codec DAI. Signed-off-by: Kenneth Westfield <kwestfie(a)codeaurora.org> --- based on next/topic/max98357a branch sound/soc/codecs/max98357a.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 4ee23fbc4e1237268ae104319a9cfe4fc369934e..bf3e933ee895752352e90f610195f39fc6cc058f 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -85,9 +85,9 @@ static struct snd_soc_dai_ops max98357a_dai_ops = { }; static struct snd_soc_dai_driver max98357a_dai_driver = { - .name = "max98357a", + .name = "HiFi", .playback = { - .stream_name = "max98357a-playback", + .stream_name = "HiFi Playback", .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32, -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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Re: [alsa-devel] Adjust the the codec and dai_link close sequence
by Mark Brown 26 Feb '15

26 Feb '15

On Mon, Jan 19, 2015 at 06:22:00PM -0800, Zhao Ye wrote: Please fix your mailer to word wrap within paragraphs at something confortably less than 80 columns, it makes your messages much easier to read. > We meet a issue, when we close the audio path, we found the dai link > will closed after the codec. And the codec dai register will write > fail. This is because our audio codec is a I2C device, and the codec > dai’s dai_link register is in the I2C device. If we close the codec > chip fisrt, The whole I2C chip’s power will be off, then we write the > dai_link’s register will be fail. I'm a little unclear on the exact scenario here but this sounds like the CODEC driver is probably buggy - it should either be putting the device regmap into cache only mode when powered off (if there is no need for the writes to take immediate effect) or not powering off until after whatever cleanup is being done in the DAI is done. Without knowing more about what the CODEC is trying to do at each stage it's hard to say exactly but it doesn't sound like an ordering issue. > So I think we should adjust the > rtd->dai_link->ops->shutdown(substream); before to > codec_dai->driver->ops->shutdown(substream, codec_dai); in the > soc_pcm_close() function. Doing that then means that we run into trouble on systems where the CODEC somehow depends on clocks from the DAI (which does happen)... really there's no perfect ordering here and attempting to shuffle this stuff around is likely to create bugs in other places as their dependencies on the current ordering get exposed. If there is some core work needed it would be better to do something that explicitly shows the dependencies so we don't get this sort of fragility.

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[alsa-devel] [PATCH 1/2] ASoC: rt5677: Correct the routing paths of that after IF1/2 DACx Mux
by Oder Chiou 26 Feb '15

26 Feb '15

The patch corrects the routing paths of that after IF1/2 DACx Mux Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index ab62777..f87c929 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3297,8 +3297,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IB45 Bypass Mux", "Bypass", "IB45 Mux" }, { "IB45 Bypass Mux", "Pass SRC", "IB45 Mux" }, - { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6" }, - { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6" }, + { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6 Mux" }, + { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6 Mux" }, { "IB6 Mux", "SLB DAC 6", "SLB DAC6" }, { "IB6 Mux", "STO4 ADC MIX L", "Stereo4 ADC MIXL" }, { "IB6 Mux", "IF4 DAC L", "IF4 DAC L" }, @@ -3306,8 +3306,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IB6 Mux", "STO2 ADC MIX L", "Stereo2 ADC MIXL" }, { "IB6 Mux", "STO3 ADC MIX L", "Stereo3 ADC MIXL" }, - { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7" }, - { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7" }, + { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7 Mux" }, + { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7 Mux" }, { "IB7 Mux", "SLB DAC 7", "SLB DAC7" }, { "IB7 Mux", "STO4 ADC MIX R", "Stereo4 ADC MIXR" }, { "IB7 Mux", "IF4 DAC R", "IF4 DAC R" }, @@ -3648,15 +3648,15 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "DAC1 FS", NULL, "DAC1 MIXL" }, { "DAC1 FS", NULL, "DAC1 MIXR" }, - { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2" }, - { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2" }, + { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2 Mux" }, + { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2 Mux" }, { "DAC2 L Mux", "IF3 DAC L", "IF3 DAC L" }, { "DAC2 L Mux", "IF4 DAC L", "IF4 DAC L" }, { "DAC2 L Mux", "SLB DAC 2", "SLB DAC2" }, { "DAC2 L Mux", "OB 2", "OutBound2" }, - { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3" }, - { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3" }, + { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3 Mux" }, + { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3 Mux" }, { "DAC2 R Mux", "IF3 DAC R", "IF3 DAC R" }, { "DAC2 R Mux", "IF4 DAC R", "IF4 DAC R" }, { "DAC2 R Mux", "SLB DAC 3", "SLB DAC3" }, @@ -3664,29 +3664,29 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "DAC2 R Mux", "Haptic Generator", "Haptic Generator" }, { "DAC2 R Mux", "VAD ADC", "VAD ADC Mux" }, - { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4" }, - { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4" }, + { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4 Mux" }, + { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4 Mux" }, { "DAC3 L Mux", "IF3 DAC L", "IF3 DAC L" }, { "DAC3 L Mux", "IF4 DAC L", "IF4 DAC L" }, { "DAC3 L Mux", "SLB DAC 4", "SLB DAC4" }, { "DAC3 L Mux", "OB 4", "OutBound4" }, - { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC4" }, - { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC4" }, + { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC5 Mux" }, + { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC5 Mux" }, { "DAC3 R Mux", "IF3 DAC R", "IF3 DAC R" }, { "DAC3 R Mux", "IF4 DAC R", "IF4 DAC R" }, { "DAC3 R Mux", "SLB DAC 5", "SLB DAC5" }, { "DAC3 R Mux", "OB 5", "OutBound5" }, - { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6" }, - { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6" }, + { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6 Mux" }, + { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6 Mux" }, { "DAC4 L Mux", "IF3 DAC L", "IF3 DAC L" }, { "DAC4 L Mux", "IF4 DAC L", "IF4 DAC L" }, { "DAC4 L Mux", "SLB DAC 6", "SLB DAC6" }, { "DAC4 L Mux", "OB 6", "OutBound6" }, - { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7" }, - { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7" }, + { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7 Mux" }, + { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7 Mux" }, { "DAC4 R Mux", "IF3 DAC R", "IF3 DAC R" }, { "DAC4 R Mux", "IF4 DAC R", "IF4 DAC R" }, { "DAC4 R Mux", "SLB DAC 7", "SLB DAC7" }, -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH] ASoC: Intel: wrap runtime_pm usage count under CONFIG_PM
by Vinod Koul 26 Feb '15

26 Feb '15

The struct dev_pm_ops defines usage_count only when CONFIG_PM is defined. So we should use this variable only in cases where this falg is true. So we define a local variable and read the value under this flag. In non PM cases, we set this to 1. Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/sst/sst_drv_interface.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c index 549af7d7f6d0..f0e4b99b3aeb 100644 --- a/sound/soc/intel/sst/sst_drv_interface.c +++ b/sound/soc/intel/sst/sst_drv_interface.c @@ -139,17 +139,23 @@ static int sst_power_control(struct device *dev, bool state) { struct intel_sst_drv *ctx = dev_get_drvdata(dev); int ret = 0; + int usage_count = 0; + +#ifdef CONFIG_PM + usage_count = atomic_read(&dev->power.usage_count); +#else + usage_count = 1; +#endif if (state == true) { ret = pm_runtime_get_sync(dev); - dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", - atomic_read(&dev->power.usage_count)); + + dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count); if (ret < 0) { dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret); return ret; } - if ((ctx->sst_state == SST_RESET) && - (atomic_read(&dev->power.usage_count) == 1)) { + if ((ctx->sst_state == SST_RESET) && (usage_count == 1)) { ret = sst_load_fw(ctx); if (ret) { dev_err(dev, "FW download fail %d\n", ret); @@ -158,8 +164,7 @@ static int sst_power_control(struct device *dev, bool state) } } } else { - dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", - atomic_read(&dev->power.usage_count)); + dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", usage_count); return sst_pm_runtime_put(ctx); } return ret; -- 1.7.9.5

2 1

[alsa-devel] audio mcbsp1 broken for beagleboard-xm -OMAP
by noman pouigt 25 Feb '15

25 Feb '15

Hello, I am trying to integrate max98090 codec with $subject board. I am running the mcBSP in slave mode as i have not changed the hw_parms for omap-twl4030.c file. With below changes i see wait_for_avail is getting stuck i.e. mcbsp is not getting any interrupt and nothing is playing. Kernel version: VERSION = 3 PATCHLEVEL = 19 SUBLEVEL = 0 EXTRAVERSION = NAME = Diseased Newt Setup information: Using beagle board -xm and connected max98090 codec to mcbsp1 pins. Connected only 4 pins. Changed the omap-twl4030.c machine file as below to disable the twl4030: static struct snd_soc_dai_link omap_twl4030_dai_links[] = { { + .name = "max98090", + .stream_name = "max98090", .cpu_dai_name = "omap-mcbsp.1", + .codec_dai_name = "HiFi", .platform_name = "omap-mcbsp.1", + .codec_name = "max98090.1-0010", .ops = &omap_twl4030_ops, }, DTSI file change:arch/arm/boot/dts/omap3-beagle-xm.dts &i2c2 { clock-frequency = <400000>; + + max98090: max98090@10 { + compatible = "maxim,max98090"; + reg = <0x10>; + }; }; omap/omap-twl4030.c added in hw_params below function to set internal oscillator for 98090 for generating PLL for master clock. snd_soc_dai_set_sysclk(codec_dai, 26000000, 0, SND_SOC_CLOCK_IN); I am able to see that the cpu_dai is mapped with codec_dai and configuration is ok. kernel logs at https://ideone.com/oezNSj I think i don't need to configure the mcbsp1 pins right as it uses default. Please guide me.I have tried IRC channel also but not getting the right folks. https://github.com/CircuitCo/BeagleBoard-xM-RevC/blob/master/BeagleBoard-xM… Thanks,

2 6

[alsa-devel] [patch] ALSA: cmipci: remove a stray space character
by Dan Carpenter 25 Feb '15

25 Feb '15

Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 1d0f2ca..6cf464d 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2062,7 +2062,7 @@ static int snd_cmipci_get_volume(struct snd_kcontrol *kcontrol, val = (snd_cmipci_mixer_read(cm, reg.right_reg) >> reg.right_shift) & reg.mask; if (reg.invert) val = reg.mask - val; - ucontrol->value.integer.value[1] = val; + ucontrol->value.integer.value[1] = val; } spin_unlock_irq(&cm->reg_lock); return 0;

2 1

[alsa-devel] [patch] ALSA: azt3328: some indenting cleanups
by Dan Carpenter 25 Feb '15

25 Feb '15

A few minor tweaks to make things line up correctly. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index a40a2b4..33b2a0a 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1385,8 +1385,8 @@ snd_azf3328_ctrl_codec_activity(struct snd_azf3328 *chip, .running) && (!chip->codecs[peer_codecs[codec_type].other2] .running)); - } - if (call_function) + } + if (call_function) snd_azf3328_ctrl_enable_codecs(chip, enable); /* ...and adjust clock, too @@ -2126,7 +2126,8 @@ static struct snd_pcm_ops snd_azf3328_i2s_out_ops = { static int snd_azf3328_pcm(struct snd_azf3328 *chip) { -enum { AZF_PCMDEV_STD, AZF_PCMDEV_I2S_OUT, NUM_AZF_PCMDEVS }; /* pcm devices */ + /* pcm devices */ + enum { AZF_PCMDEV_STD, AZF_PCMDEV_I2S_OUT, NUM_AZF_PCMDEVS }; struct snd_pcm *pcm; int err;

2 1

[alsa-devel] [patch] sound/sb_midi: a couple indenting fixes
by Dan Carpenter 25 Feb '15

25 Feb '15

Let's make things line up a little bit better. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/oss/sb_midi.c b/sound/oss/sb_midi.c index f139028..551ee75 100644 --- a/sound/oss/sb_midi.c +++ b/sound/oss/sb_midi.c @@ -179,14 +179,14 @@ void sb_dsp_midi_init(sb_devc * devc, struct module *owner) { printk(KERN_WARNING "Sound Blaster: failed to allocate MIDI memory.\n"); sound_unload_mididev(dev); - return; + return; } memcpy((char *) midi_devs[dev], (char *) &sb_midi_operations, sizeof(struct midi_operations)); if (owner) - midi_devs[dev]->owner = owner; - + midi_devs[dev]->owner = owner; + midi_devs[dev]->devc = devc;

2 1

[alsa-devel] [patch] sound/sb_ess: white space cleanups
by Dan Carpenter 25 Feb '15

25 Feb '15

These weren't aligned on the same lines as the surrounding code and the printk was quite messy. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index b47a690..57f7d25 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -604,7 +604,7 @@ static void ess_audio_output_block_audio2 ess_chgmixer (devc, 0x78, 0x03, 0x03); /* Go */ devc->irq_mode_16 = IMODE_OUTPUT; - devc->intr_active_16 = 1; + devc->intr_active_16 = 1; } static void ess_audio_output_block @@ -1183,17 +1183,12 @@ FKS_test (devc); chip = "ES1688"; } - printk ( KERN_INFO "ESS chip %s %s%s\n" - , chip - , ( devc->sbmo.esstype == ESSTYPE_DETECT || devc->sbmo.esstype == ESSTYPE_LIKE20 - ? "detected" - : "specified" - ) - , ( devc->sbmo.esstype == ESSTYPE_LIKE20 - ? " (kernel 2.0 compatible)" - : "" - ) - ); + printk(KERN_INFO "ESS chip %s %s%s\n", chip, + (devc->sbmo.esstype == ESSTYPE_DETECT || + devc->sbmo.esstype == ESSTYPE_LIKE20) ? + "detected" : "specified", + devc->sbmo.esstype == ESSTYPE_LIKE20 ? + " (kernel 2.0 compatible)" : ""); sprintf(name,"ESS %s AudioDrive (rev %d)", chip, ess_minor & 0x0f); } else {

2 1

[alsa-devel] [patch] sound/oss/opl3: remove some stray whitespace
by Dan Carpenter 25 Feb '15

25 Feb '15

Removed an extra tab and a extra space character. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c index 607cee4..b6d19ad 100644 --- a/sound/oss/opl3.c +++ b/sound/oss/opl3.c @@ -666,7 +666,7 @@ static int opl3_start_note (int dev, int voice, int note, int volume) opl3_command(map->ioaddr, FNUM_LOW + map->voice_num, data); data = 0x20 | ((block & 0x7) << 2) | ((fnum >> 8) & 0x3); - devc->voc[voice].keyon_byte = data; + devc->voc[voice].keyon_byte = data; opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num, data); if (voice_mode == 4) opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num + 3, data); @@ -717,7 +717,7 @@ static void freq_to_fnum (int freq, int *block, int *fnum) static void opl3_command (int io_addr, unsigned int addr, unsigned int val) { - int i; + int i; /* * The original 2-OP synth requires a quite long delay after writing to a

2 1

[alsa-devel] [patch] sound: sys_timer: indent poll_def_tmr() correctly
by Dan Carpenter 25 Feb '15

25 Feb '15

The indenting here was really whacky and not consistent from one line to the next. I also reverse the "if (opened)" and "if (tmr_running)" tests so that I could remove two indent levels. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/oss/sys_timer.c b/sound/oss/sys_timer.c index 9f03983..2226dda 100644 --- a/sound/oss/sys_timer.c +++ b/sound/oss/sys_timer.c @@ -50,29 +50,24 @@ tmr2ticks(int tmr_value) static void poll_def_tmr(unsigned long dummy) { + if (!opened) + return; + def_tmr.expires = (1) + jiffies; + add_timer(&def_tmr); - if (opened) - { + if (!tmr_running) + return; - { - def_tmr.expires = (1) + jiffies; - add_timer(&def_tmr); - } + spin_lock(&lock); + tmr_ctr++; + curr_ticks = ticks_offs + tmr2ticks(tmr_ctr); - if (tmr_running) - { - spin_lock(&lock); - tmr_ctr++; - curr_ticks = ticks_offs + tmr2ticks(tmr_ctr); - - if (curr_ticks >= next_event_time) - { - next_event_time = (unsigned long) -1; - sequencer_timer(0); - } - spin_unlock(&lock); - } - } + if (curr_ticks >= next_event_time) { + next_event_time = (unsigned long) -1; + sequencer_timer(0); + } + + spin_unlock(&lock); } static void

2 1

[alsa-devel] No sound from speakers with Realtek ALC3226
by John Frankish 25 Feb '15

25 Feb '15

I posted this on the alsa-user list, but did not get a reply. Although it occasionally works, I am usually unable to get any sound out of the speakers on a dell latitude e7240 using alsa-1.0.28 and linux-3.16.6 - see details at the end of the post. 1. HDMI is the default sound device - why, the analogue device would make more sense? 2. On the analogue sound device, the headphones are the default instead of the speakers. On the few occasions I get sound out of the speakers, the headphones need to be muted first - even though no headphones are connected. 3. Shutting windows 7 64bit down with "shutdown.exe -s" seems to help with getting sound from the speakers with linux on the same machine. Most of the time, I see the error: snd_hda_intel 0000:00:1b.0: IRQ timing workaround is activated for card #1. Suggest a bigger bdl_pos_adj. I've tried various module options: options snd_hda_intel probe_only=0,1 options snd-hda-intel enable_msi=1 bdl_pos_adj=1,32 [and 48 and 64] options snd-hda-intel id=PCH index=1 ..all to no avail. Any additional troubleshooting suggestions (or a patch) would be much appreciated. John ---------- $ lspci | grep -i audio 00:03.0 Audio device: Intel Corporation Haswell-ULT HD Audio Controller (rev0b) 00:1b.0 Audio device: Intel Corporation 8 Series HD Audio Controller (rev04) $ cat /proc/asound/cards 0 [HDMI ]: HDA-Intel - HDA Intel HDMI HDA Intel HDMI at 0xf7e34000 irq 65 1 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xf7e30000 irq 64 $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 0: ALC3226 Analog [ALC3226 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 $ cat /proc/asound/card*/codec* | grep Codec Codec: Intel Haswell HDMI Codec: Realtek ALC3226

2 7

Re: [alsa-devel] Issues w/ Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG
by Takashi Iwai 25 Feb '15

25 Feb '15

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

2 3

[alsa-devel] Issues w/ Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG
by Alnie 25 Feb '15

25 Feb '15

I am having a problem getting this card working in Mint 17.1 (64bit). I have found reports from others around the net claiming to have this card working (around 2010/2011 mostly). I also tried xubuntu with the same results. On a fresh distro when I install the card, it is recognized but isn't associated with any driver, if I install the 3.16 kernel or the alsa-daily repo it becomes associated with a driver but I get the following errors from dmesg... [ 3.713183] snd_hda_intel 0000:06:00.0: Probing card using HDA DKMS, version 0.201502151246~ubuntu14.04.1 [ 3.720010] snd_hda_intel 0000:06:00.0: CORB reset timeout#1, CORBRP = 0 [ 4.744035] snd_hda_intel 0000:06:00.0: Codec #1 probe error; disabling it... [ 4.748009] snd_hda_intel 0000:06:00.0: CORB reset timeout#1, CORBRP = 0 [ 9.772075] snd_hda_intel 0000:06:00.0: no AFG or MFG node found [ 9.772100] snd_hda_intel 0000:06:00.0: no codecs initialized and here is my lspci output... 05:00.0 PCI bridge: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG PCI to PCIe Bridge (prog-if 00 [Normal decode]) Physical Slot: 3 Flags: bus master, fast devsel, latency 0 Bus: primary=05, secondary=06, subordinate=0c, sec-latency=36 Memory behind bridge: f4300000-f43fffff Capabilities: [50] Power Management version 3 Capabilities: [60] MSI: Enable- Count=1/16 Maskable- 64bit+ Capabilities: [80] Subsystem: Creative Labs Device 0040 Capabilities: [90] Express PCI-Express to PCI/PCI-X Bridge, MSI 00 Capabilities: [100] Advanced Error Reporting 06:00.0 Audio device: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG Subsystem: Creative Labs SB1040 Flags: bus master, fast Back2Back, medium devsel, latency 32, IRQ 19 Memory at f4300000 (32-bit, non-prefetchable) [size=16K] Capabilities: [dc] Power Management version 3 Kernel driver in use: snd_hda_intel thank you for any advice & feedback on this.

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[alsa-devel] [Patch V6 00/10] ASoC: QCOM: Add support for ipq806x SOC
by Kenneth Westfield 25 Feb '15

25 Feb '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> This patch series adds support for I2S audio playback on the Storm board, which contains a Qualcomm Technologies ipq806x SOC and a Maxim max98357a DAC/amp. The ipq806x SOC has audio-related hardware blocks in its low-power audio subsystem (or LPASS). One of the relevant blocks in the LPASS is its low-power audio interface (or LPAIF). This contains an MI2S port, which is what these drivers are configured to use. The LPAIF also contains a DMA engine that is dedicated to moving audio samples into the transmit FIFO of the MI2S port. One bus from the MI2S port of the SOC is connected to the DAC/amp for stereo playback. This bus is configured so that the SOC is bus master and consists of DATA, LRCLK, and BCLK. The DAC/amp does not need MCLK to operate. In addition, a single GPIO pin from the SOC is connected to the same DAC/amp, which gives enable/disable control over the DAC/amp. The specific drivers added are: * a CPU DAI driver for controlling the MI2S port * a platform driver for controlling the LPAIF DMA engine * a machine driver that instantiates a dai-link for playback The LPASS also contains clocks that need to be controlled. Those drivers have been submitted as a separate patch series: [PATCH v3 0/8] qcom audio clock control drivers http://lkml.org/lkml/2015/1/19/656 Even though the ipq806x LPASS does not contain an audio DSP, other SOCs do have one. For those SOCs, the audio DSP typically controls the hardware blocks in the LPASS. Hence, different CPU DAI driver(s) would need to be used in order to facilitate audio with the DSP. As such, the LPASS DT contains an adsp subnode, which is disabled for this SOC. The same subnode should be enabled and populated for other SOCs that do contain an audio DSP. Not using the audio DSP would require different CPU DAI driver(s), in addition to possible bootloader and/or firmware changes. Corresponding additions to the device tree for the ipq806x SOC and its documentation has also been added. Also, as this is a new directory, the MAINTAINERS file has been updated as well. = Changes since V5 [Patch V5 00/12] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2015-February/087832.h… * Correctly use Storm as the build target label and DT binding label. * Added audio DSP sub-node to the LPASS device tree, disabled for this SOC. * Added logic to CPU DAI driver to fail the probe() if a DSP is present. * Use the standard naming convention for the DAI link properties. * General code cleanup. = Changes since V4 [Patch V4 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2015-February/087499.h… * Replaced simple-card with a machine driver to resolve the system clock configuration, rather than having the CPU DAI driver do it. * Added header files to avoid indirect header dependencies and implicit forward declarations. * Tweaked the ISR to match the conventions of the surrounding code. * Removed the usage of the low-power memory as it is not needed. * Removed the use of the DRV_NAME constant. * Added explicit dependency on gpiolib for the codec driver. * Moved the MODULE_DEVICE_TABLE macro inside the CONFIG_OF conditional. * Modified the documentation to reflect the changes. * General code cleanup. = Changes since V3 [Patch V3 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085694.h… * Placed the content of the inline functions into the callbacks. * Replaced use of readl/writel register access functions with regmap access functions. Notable exception is the ISR, which uses ioread32/iowrite32. * Rearranged the sequencing of the hardware block enables to fit within the ASoC framework callbacks, while remaining functional. REQ 1: The hardware requires the enable sequence to be: LPAIF-DMA[enable],then LPAIF-MI2S[enable], then DAC-GPIO[enable] REQ 2: The hardware requires the disable sequence to be: DAC-GPIO[disable], then LPAIF-MI2S[disable] * Corrected the implementation of the pointer callback. * Utilize the LPM to buffer audio samples, rather than memory external to LPASS. * Corrected the interrupt clearing in the ISR. * Implemented a default system clock (defined by the simple-card DT node), and optional LPASS DT node modifiers that can alter the system clock in order to expand the range of available bit clock frequencies. * Addressed all of the remaining issues raised by Mark Brown. * General code cleanup. = Changes since V2 [Patch v2 00/11] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085186.h… * Removed the PCM platform driver from the DTS platform and tied it to the CPU DAI driver. * Changed I2S pinctrl to use generic naming convention and moved control to CPU DAI driver. It should be controlled now by soc-core's pinctrl_pm_* functionality. * Added stub DAPM support in codec driver. As the DAC GPIO needs to be enabled last when starting playback, and disabled first when stopping playback, it seems as though the trigger function may be the place for this. Suggestions are welcome for a better place to put this. * Removed machine driver and tied DAI drivers to simple-audio-card. * Packaged the build files and Maxim codec files together in one change. * Removed QCOM as vendor from Maxim code and documentation. * Separated the SOC and board definitions into the correct DTS files. * Update device tree documentation to reflect changes. * General code cleanup. = Changes since V1 [PATCH 0/9] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084322.h… * Remove the native LPAIF driver, and move its functionality to the CPU DAI driver. * Add a codec driver to manage the pins going to the external DAC (previously managed by the machine driver). * Use devm_* and dev_* where possible. * ISR only handles relevant DMA channel now. * Update device tree documentation to reflect changes. * General code cleanup. Kenneth Westfield (10): MAINTAINERS: Add QCOM audio ASoC maintainer ASoC: qcom: Document LPASS CPU bindings ASoC: qcom: Document Storm bindings ASoC: qcom: add LPASS header files ASoC: qcom: Add LPASS CPU DAI driver ASoC: qcom: Add LPASS platform driver ASoC: qcom: Add Storm machine driver ASoC: qcom: Add ability to build QCOM drivers ASoC: Allow for building QCOM drivers ARM: dts: Model IPQ LPASS audio hardware .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 49 ++ Documentation/devicetree/bindings/sound/storm.txt | 23 + MAINTAINERS | 7 + arch/arm/boot/dts/qcom-ipq8064.dtsi | 26 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/qcom/Kconfig | 23 + sound/soc/qcom/Makefile | 11 + sound/soc/qcom/lpass-cpu.c | 508 ++++++++++++++++++++ sound/soc/qcom/lpass-lpaif-ipq806x.h | 172 +++++++ sound/soc/qcom/lpass-platform.c | 526 +++++++++++++++++++++ sound/soc/qcom/lpass.h | 51 ++ sound/soc/qcom/storm.c | 162 +++++++ 13 files changed, 1560 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt create mode 100644 Documentation/devicetree/bindings/sound/storm.txt create mode 100644 sound/soc/qcom/Kconfig create mode 100644 sound/soc/qcom/Makefile create mode 100644 sound/soc/qcom/lpass-cpu.c create mode 100644 sound/soc/qcom/lpass-lpaif-ipq806x.h create mode 100644 sound/soc/qcom/lpass-platform.c create mode 100644 sound/soc/qcom/lpass.h create mode 100644 sound/soc/qcom/storm.c -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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Re: [alsa-devel] Ask for help. ARM:mvebu (CM-A510), add sound support using TLV320AIC23 audio-codec
by Sebastian Hesselbarth 24 Feb '15

24 Feb '15

[adding alsa-devel and Jean-Francois] Gabriel, you should try to add people and lists you expect help from. On 23.02.2015 21:27, Gabriel Dobato wrote: > I am trying to set the sound system using "simple-audio-card" and > TLV320AIC23B audio codec in Compulab CM-510 SoM. > > This is my structure in DT, according to > ./KERNEL/Documentation/devicetree/bindings/sound/simple-card.txt, > /KERNEL/Documentation/devicetree/bindings/sound/widgets.txt and the > source code of the audio-codec ./KERNEL/sound/soc/codecs/tlv320aic23.c : > > sound { > compatible = "simple-audio-card"; > simple-audio-card,name = "OnboardTLV320AIC23B"; > simple-audio-card,format = "i2s"; > simple-audio-card,bitclock-master = <&dailink_master>; > simple-audio-card,frame-master = <&dailink_master>; > simple-audio-card,widgets = > "Microphone", "Mic Input", > "Line", "Line Input", > "Line", "Line Out", > "Speaker", "Speaker", > "Headphone", "Headphone Jack"; > > simple-audio-card,routing = > "Line Out", "LOUT", > "Line Out", "ROUT", > "Line Out", "RHPOUT", > "Line Out", "LHPOUT", > "LLINEIN","Line Input", > "RLINEIN","Line Input", > "MICIN","Mic Input"; > simple-audio-card,cpu { > sound-dai = <&audio0 0>; > }; > dailink_master: simple-audio-card,codec { > sound-dai = <&opt_audio>; > }; > }; > }; > > /* TLV320AIC23 */ > &opt_audio { > status = "okay"; > #sound-dai-cells= <0>; > }; The full node looks like: opt_audio: audio@1a { compatible = "ti,tlv320aic23"; reg = <0x1a>; #sound-dai-cells= <0>; status = "okay"; }; > And this is the kernel trace: > > ... > tlv320aic23-codec 0-001a: Control not supported for path LLINEIN -> > [NULL] -> Line Input > tlv320aic23-codec 0-001a: ASoC: no dapm match for LLINEIN --> NULL --> > Line Input > tlv320aic23-codec 0-001a: ASoC: Failed to add route LLINEIN -> NULL -> > Line Input > tlv320aic23-codec 0-001a: Control not supported for path RLINEIN -> > [NULL] -> Line Input > tlv320aic23-codec 0-001a: ASoC: no dapm match for RLINEIN --> NULL --> > Line Input > tlv320aic23-codec 0-001a: ASoC: Failed to add route RLINEIN -> NULL -> > Line Input > tlv320aic23-codec 0-001a: Control not supported for path MICIN -> [NULL] > -> Mic Input > tlv320aic23-codec 0-001a: ASoC: no dapm match for MICIN --> NULL --> Mic > Input > tlv320aic23-codec 0-001a: ASoC: Failed to add route MICIN -> NULL -> Mic > Input > usb 2-1.2: new high-speed USB device number 3 using orion-ehci > asoc-simple-card sound: tlv320aic23-hifi <-> i2s mapping ok > ... > > ALSA device list: > #0: OnboardTLV320AIC23B > ... > > [....] Setting up ALSA...amixer: Invalid command! It doesn't help much without knowing what amixer is trying to do. > I think I have written the node according to the documentation, but > probably I miss something. Looking at sound/soc/codecs/tlv320aic23.c: static const struct snd_soc_dapm_route tlv320aic23_intercon[] = { /* Output Mixer */ {"Output Mixer", "Line Bypass Switch", "Line Input"}, {"Output Mixer", "Playback Switch", "DAC"}, {"Output Mixer", "Mic Sidetone Switch", "Mic Input"}, /* Outputs */ {"RHPOUT", NULL, "Output Mixer"}, {"LHPOUT", NULL, "Output Mixer"}, {"LOUT", NULL, "Output Mixer"}, {"ROUT", NULL, "Output Mixer"}, /* Inputs */ {"Line Input", "NULL", "LLINEIN"}, {"Line Input", "NULL", "RLINEIN"}, {"Mic Input", "NULL", "MICIN"}, /* input mux */ {"Capture Source", "Line", "Line Input"}, {"Capture Source", "Mic", "Mic Input"}, {"ADC", NULL, "Capture Source"}, }; The lines with the failing routes really look suspicious, i.e. there is strings "NULL" where I'd expect plain NULL. What happens if you amend the three lines and replace the "NULL" with NULL? Sebastian

2 2

Re: [alsa-devel] No sound and systemd journal filling when inserting headphones when power-saving is enabled
by Takashi Iwai 24 Feb '15

24 Feb '15

At Tue, 24 Feb 2015 20:11:58 +0000, Dang Sananikone wrote: > > Thanks, in that case I'll simply attach the files to this post. There are > two files: > > * alsa-info-notworking.txt: This is the alsa info generated when > power-saving is enabled and after "speaker-test" has been invoked. > * alsa-info-working.txt: This is the alsa info generated when power-saving > is disabled and after "speaker-test" has been invoked. > > > To reproduce the problem: > > Prerequisite: > Set "options snd_hda_intel power_save=1" > > Instructions: > 1. Boot up laptop. > 2. Insert headphone socket into jack. > 3. In terminal, type "speaker-test -c 2". > > Expected Result: > The speaker-test program hangs, and the systemd journal starts filling up > with "sound hdaudioC0D0: hda-codec: out of range cmd 0:1:716:ffffffff" > messages. OK, then try to pass power_save_controller=0 option to snd-hda-intel module. It might be that the controller gets screwed up by some reason by power-saving. If this still doesn't help, we need more logs. Build the kernel with tracing support, and get the HD-audio verb traces as Documentation/sound/alsa/HD-Audio.txt, section Tracepoints. Takashi

1 0

[alsa-devel] [PATCH] ASoC: Add max98925 codec driver
by Anish Kumar 24 Feb '15

24 Feb '15

Signed-off-by: Anish Kumar <yesanishhere(a)gmail.com> --- sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98925.c | 1229 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max98925.h | 845 +++++++++++++++++++++++++++++ 4 files changed, 2086 insertions(+) create mode 100644 sound/soc/codecs/max98925.c create mode 100644 sound/soc/codecs/max98925.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 45b7256..a4ccae9 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -47,6 +47,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C + select SND_SOC_MAX98925 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C @@ -285,6 +286,15 @@ config SND_SOC_MAX9850 config SND_SOC_OMAP_HDMI_CODEC tristate +config SND_SOC_MAX98925 + bool "Support MAX98925 stereo system" + depends on I2C + help + MAX98925 is a high-efficiency mono + Class DG audio amplifier. Use this + option to enable the system consisting + of left and right power amplifier. + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6a3b3c3..313a7c4 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -37,6 +37,7 @@ snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o +snd-soc-max98925-objs := max98925.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o @@ -162,6 +163,7 @@ obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o +obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c new file mode 100644 index 0000000..e3a1984 --- /dev/null +++ b/sound/soc/codecs/max98925.c @@ -0,0 +1,1229 @@ +/* + * max98925.c -- ALSA SoC Stereo MAX98925 driver + * Copyright 2013-15 Maxim Integrated Products + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/cdev.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max98925.h" + +#define VCC_I2C_MIN_UV 1800000 +#define VCC_I2C_MAX_UV 1800000 +#define I2C_LOAD_UA 300000 + +enum max98925_type { + MAX98925L, + MAX98925R, +}; + +static int max98925_hpf_l; +static int max98925_hpf_r; +static int max98925_dai_sel_l; +static int max98925_dai_sel_r; +static struct max98925_priv *max98925; + +static const char *const dai_text[] = { + "L_Data", "R_Data", "LR_Data", "LR_Data_Div2", +}; + +static const char *const hpf_text[] = { + "hpf_disable", "hpf_dc_block", "hpf_enable_100", + "hpf_enable_200", "hpf_enable_400", "hpf_enable_800", +}; + +static const struct soc_enum max98925_dai_l_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(dai_text), dai_text), +}; + +static const struct soc_enum max98925_dai_r_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(dai_text), dai_text), +}; + +static const struct soc_enum max98925_hpfout_r_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(hpf_text), hpf_text), +}; + +static const struct soc_enum max98925_hpfout_l_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(hpf_text), hpf_text), +}; + +static struct reg_default max98925_reg[] = { + { 0x00, 0x00 }, /* Battery Voltage Data */ + { 0x01, 0x00 }, /* Boost Voltage Data */ + { 0x02, 0x00 }, /* Live Status0 */ + { 0x03, 0x00 }, /* Live Status1 */ + { 0x04, 0x00 }, /* Live Status2 */ + { 0x05, 0x00 }, /* State0 */ + { 0x06, 0x00 }, /* State1 */ + { 0x07, 0x00 }, /* State2 */ + { 0x08, 0x00 }, /* Flag0 */ + { 0x09, 0x00 }, /* Flag1 */ + { 0x0A, 0x00 }, /* Flag2 */ + { 0x0B, 0x00 }, /* IRQ Enable0 */ + { 0x0C, 0x00 }, /* IRQ Enable1 */ + { 0x0D, 0x00 }, /* IRQ Enable2 */ + { 0x0E, 0x00 }, /* IRQ Clear0 */ + { 0x0F, 0x00 }, /* IRQ Clear1 */ + { 0x10, 0x00 }, /* IRQ Clear2 */ + { 0x11, 0xC0 }, /* Map0 */ + { 0x12, 0x00 }, /* Map1 */ + { 0x13, 0x00 }, /* Map2 */ + { 0x14, 0xF0 }, /* Map3 */ + { 0x15, 0x00 }, /* Map4 */ + { 0x16, 0xAB }, /* Map5 */ + { 0x17, 0x89 }, /* Map6 */ + { 0x18, 0x00 }, /* Map7 */ + { 0x19, 0x00 }, /* Map8 */ + { 0x1A, 0x06 }, /* DAI Clock Mode 1 */ + { 0x1B, 0xC0 }, /* DAI Clock Mode 2 */ + { 0x1C, 0x00 }, /* DAI Clock Divider Denominator MSBs */ + { 0x1D, 0x00 }, /* DAI Clock Divider Denominator LSBs */ + { 0x1E, 0xF0 }, /* DAI Clock Divider Numerator MSBs */ + { 0x1F, 0x00 }, /* DAI Clock Divider Numerator LSBs */ + { 0x20, 0x50 }, /* Format */ + { 0x21, 0x00 }, /* TDM Slot Select */ + { 0x22, 0x00 }, /* DOUT Configuration VMON */ + { 0x23, 0x00 }, /* DOUT Configuration IMON */ + { 0x24, 0x00 }, /* DOUT Configuration VBAT */ + { 0x25, 0x00 }, /* DOUT Configuration VBST */ + { 0x26, 0x00 }, /* DOUT Configuration FLAG */ + { 0x27, 0xFF }, /* DOUT HiZ Configuration 1 */ + { 0x28, 0xFF }, /* DOUT HiZ Configuration 2 */ + { 0x29, 0xFF }, /* DOUT HiZ Configuration 3 */ + { 0x2A, 0xFF }, /* DOUT HiZ Configuration 4 */ + { 0x2B, 0x02 }, /* DOUT Drive Strength */ + { 0x2C, 0x90 }, /* Filters */ + { 0x2D, 0x00 }, /* Gain */ + { 0x2E, 0x02 }, /* Gain Ramping */ + { 0x2F, 0x00 }, /* Speaker Amplifier */ + { 0x30, 0x0A }, /* Threshold */ + { 0x31, 0x00 }, /* ALC Attack */ + { 0x32, 0x80 }, /* ALC Atten and Release */ + { 0x33, 0x00 }, /* ALC Infinite Hold Release */ + { 0x34, 0x92 }, /* ALC Configuration */ + { 0x35, 0x01 }, /* Boost Converter */ + { 0x36, 0x00 }, /* Block Enable */ + { 0x37, 0x00 }, /* Configuration */ + { 0x38, 0x00 }, /* Global Enable */ + { 0x3A, 0x00 }, /* Boost Limiter */ + { 0xFF, 0x50 }, /* Revision ID */ +}; + +void max98925_regmap_update_bits_stereo(struct max98925_priv *max98925, + unsigned int reg, unsigned int mask, unsigned int val) +{ + regmap_update_bits(max98925->regmap_l, reg, mask, val); + regmap_update_bits(max98925->regmap_r, reg, mask, val); +} + +static int max98925_get_hpf_l(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = max98925_hpf_l; + return 0; +} + +static int max98925_put_hpf_l(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + struct snd_soc_dapm_update update; + unsigned int reg = MAX98925_R02C_FILTERS; + unsigned int mask = M98925_DAC_HPF_MASK; + + switch (ucontrol->value.integer.value[0]) { + case 0: + regmap_update_bits(max98925->regmap_l, reg, mask, + M98925_DAC_HPF_DISABLE); + break; + case 1: + regmap_update_bits(max98925->regmap_l, reg, mask, + M98925_DAC_HPF_DC_BLOCK); + break; + case 2: + regmap_update_bits(max98925->regmap_l, reg, mask, + M98925_DAC_HPF_EN_100); + break; + case 3: + regmap_update_bits(max98925->regmap_l, reg, mask, + M98925_DAC_HPF_EN_200); + break; + case 4: + regmap_update_bits(max98925->regmap_l, reg, mask, + M98925_DAC_HPF_EN_400); + break; + case 5: + regmap_update_bits(max98925->regmap_l, reg, mask, + M98925_DAC_HPF_EN_800); + break; + default: + return -EINVAL; + } + update.kcontrol = kcontrol; + update.reg = SND_SOC_NOPM; + max98925_hpf_l = ucontrol->value.integer.value[0]; + snd_soc_dapm_mux_update_power(&codec->dapm, + kcontrol, max98925_hpf_l, e, &update); + return 0; +} + +static int max98925_get_hpf_r(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = max98925_hpf_r; + return 0; +} + +static int max98925_put_hpf_r(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + struct snd_soc_dapm_update update; + unsigned int reg = MAX98925_R02C_FILTERS; + unsigned int mask = M98925_DAC_HPF_MASK; + + switch (ucontrol->value.integer.value[0]) { + case 0: + regmap_update_bits(max98925->regmap_r, reg, mask, + M98925_DAC_HPF_DISABLE); + break; + case 1: + regmap_update_bits(max98925->regmap_r, reg, mask, + M98925_DAC_HPF_DC_BLOCK); + break; + case 2: + regmap_update_bits(max98925->regmap_r, reg, mask, + M98925_DAC_HPF_EN_100); + break; + case 3: + regmap_update_bits(max98925->regmap_r, reg, mask, + M98925_DAC_HPF_EN_200); + break; + case 4: + regmap_update_bits(max98925->regmap_r, reg, mask, + M98925_DAC_HPF_EN_400); + break; + case 5: + regmap_update_bits(max98925->regmap_r, reg, mask, + M98925_DAC_HPF_EN_800); + break; + default: + return -EINVAL; + } + max98925_hpf_r = ucontrol->value.integer.value[0]; + update.kcontrol = kcontrol; + update.reg = SND_SOC_NOPM; + snd_soc_dapm_mux_update_power(&codec->dapm, + kcontrol, max98925_hpf_r, e, &update); + return 0; +} + +static int max98925_get_dai_sel_l(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = max98925_dai_sel_l; + return 0; +} + +static int max98925_put_dai_sel_l(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + struct snd_soc_dapm_update update; + unsigned int reg = MAX98925_R02D_GAIN; + unsigned int mask = M98925_DAC_IN_SEL_MASK; + + switch (ucontrol->value.integer.value[0]) { + case 0: + regmap_update_bits(max98925->regmap_l, + reg, mask, M98925_DAC_IN_SEL_LEFT_DAI); + break; + case 1: + regmap_update_bits(max98925->regmap_l, + reg, mask, M98925_DAC_IN_SEL_RIGHT_DAI); + break; + case 2: + regmap_update_bits(max98925->regmap_l, + reg, mask, M98925_DAC_IN_SEL_SUMMED_DAI); + break; + case 3: + regmap_update_bits(max98925->regmap_l, + reg, mask, M98925_DAC_IN_SEL_DIV2_SUMMED_DAI); + break; + default: + return -EINVAL; + } + max98925_dai_sel_l = ucontrol->value.integer.value[0]; + update.kcontrol = kcontrol; + update.reg = SND_SOC_NOPM; + snd_soc_dapm_mux_update_power(&codec->dapm, + kcontrol, max98925_dai_sel_l, e, &update); + return 0; +} + +static int max98925_get_dai_sel_r(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = max98925_dai_sel_r; + return 0; +} + +static int max98925_put_dai_sel_r(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + struct snd_soc_dapm_update update; + unsigned int reg = MAX98925_R02D_GAIN; + unsigned int mask = M98925_DAC_IN_SEL_MASK; + + switch (ucontrol->value.integer.value[0]) { + case 0: + regmap_update_bits(max98925->regmap_r, + reg, mask, M98925_DAC_IN_SEL_LEFT_DAI); + break; + case 1: + regmap_update_bits(max98925->regmap_r, + reg, mask, M98925_DAC_IN_SEL_RIGHT_DAI); + break; + case 2: + regmap_update_bits(max98925->regmap_r, + reg, mask, M98925_DAC_IN_SEL_SUMMED_DAI); + break; + case 3: + regmap_update_bits(max98925->regmap_r, + reg, mask, M98925_DAC_IN_SEL_DIV2_SUMMED_DAI); + break; + default: + return -EINVAL; + } + max98925_dai_sel_r = ucontrol->value.integer.value[0]; + update.kcontrol = kcontrol; + update.reg = SND_SOC_NOPM; + snd_soc_dapm_mux_update_power(&codec->dapm, + kcontrol, max98925_dai_sel_r, e, &update); + return 0; +} + +static int max98925_reg_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, unsigned int reg, + unsigned int mask, unsigned int shift) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + unsigned int sel = ucontrol->value.integer.value[0]; + + max98925_regmap_update_bits_stereo(max98925, reg, mask, sel << shift); + dev_dbg(codec->dev, "%s: register 0x%02X, value 0x%02X\n", + __func__, reg, sel); + return 0; +} + +static int max98925_dac_ev_l(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(max98925->regmap_l, MAX98925_R036_BLOCK_ENABLE, + M98925_SPK_EN_MASK, M98925_SPK_EN_MASK); + return 0; +} + +static int max98925_dac_ev_r(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(max98925->regmap_r, MAX98925_R036_BLOCK_ENABLE, + M98925_SPK_EN_MASK, M98925_SPK_EN_MASK); + return 0; +} + +static const struct snd_kcontrol_new dai_sel_mux[2] = { + SOC_DAPM_ENUM_EXT("DAI_IN_MUX_L Mux", max98925_dai_l_enum, + max98925_get_dai_sel_l, max98925_put_dai_sel_l), + SOC_DAPM_ENUM_EXT("DAI_IN_MUX_R Mux", max98925_dai_r_enum, + max98925_get_dai_sel_r, max98925_put_dai_sel_r), +}; + +static const struct snd_kcontrol_new hpf_mux[2] = { + SOC_ENUM_EXT("R02C_FILTER_R Mux", max98925_hpfout_r_enum, + max98925_get_hpf_r, max98925_put_hpf_r), + SOC_ENUM_EXT("R02C_FILTER_L Mux", max98925_hpfout_l_enum, + max98925_get_hpf_l, max98925_put_hpf_l), +}; + +static const struct snd_soc_dapm_widget max98925_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("DAI_IN_MUX_L MUX", SND_SOC_NOPM, 0, 0, + &dai_sel_mux[0]), + SND_SOC_DAPM_MUX("DAI_IN_MUX_R MUX", SND_SOC_NOPM, 0, 0, + &dai_sel_mux[1]), + SND_SOC_DAPM_MUX("R02C_FILTER_L MUX", SND_SOC_NOPM, 0, 0, + &hpf_mux[0]), + SND_SOC_DAPM_MUX("R02C_FILTER_R MUX", SND_SOC_NOPM, 0, 0, + &hpf_mux[1]), + SND_SOC_DAPM_DAC_E("amp_enable_l", NULL, + SND_SOC_NOPM, 0, 0, + max98925_dac_ev_l, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_DAC_E("amp_enable_r", NULL, + SND_SOC_NOPM, 0, 0, + max98925_dac_ev_r, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_OUTPUT("BE_OUT"), +}; + +static const struct snd_soc_dapm_route max98925_audio_map[] = { + {"DAI_IN_MUX_L MUX", "L_Data", "DAI_OUT"}, + {"DAI_IN_MUX_L MUX", "R_Data", "DAI_OUT"}, + {"DAI_IN_MUX_L MUX", "LR_Data", "DAI_OUT"}, + {"DAI_IN_MUX_L MUX", "LR_Data_Div2", "DAI_OUT"}, + {"R02C_FILTER_L MUX", "hpf_disable", "DAI_IN_MUX_L MUX"}, + {"R02C_FILTER_L MUX", "hpf_dc_block", "DAI_IN_MUX_L MUX"}, + {"R02C_FILTER_L MUX", "hpf_enable_100", "DAI_IN_MUX_L MUX"}, + {"R02C_FILTER_L MUX", "hpf_enable_200", "DAI_IN_MUX_L MUX"}, + {"R02C_FILTER_L MUX", "hpf_enable_400", "DAI_IN_MUX_L MUX"}, + {"R02C_FILTER_L MUX", "hpf_enable_800", "DAI_IN_MUX_L MUX"}, + {"amp_enable_l", NULL, "R02C_FILTER_L MUX"}, + {"BE_OUT", NULL, "amp_enable_l"}, + + {"DAI_IN_MUX_R MUX", "L_Data", "DAI_OUT"}, + {"DAI_IN_MUX_R MUX", "R_Data", "DAI_OUT"}, + {"DAI_IN_MUX_R MUX", "LR_Data", "DAI_OUT"}, + {"DAI_IN_MUX_R MUX", "LR_Data_Div2", "DAI_OUT"}, + {"R02C_FILTER_R MUX", "hpf_disable", "DAI_IN_MUX_R MUX"}, + {"R02C_FILTER_R MUX", "hpf_dc_block", "DAI_IN_MUX_R MUX"}, + {"R02C_FILTER_R MUX", "hpf_enable_100", "DAI_IN_MUX_R MUX"}, + {"R02C_FILTER_R MUX", "hpf_enable_200", "DAI_IN_MUX_R MUX"}, + {"R02C_FILTER_R MUX", "hpf_enable_400", "DAI_IN_MUX_R MUX"}, + {"R02C_FILTER_R MUX", "hpf_enable_800", "DAI_IN_MUX_R MUX"}, + {"amp_enable_r", NULL, "R02C_FILTER_R MUX"}, + {"BE_OUT", NULL, "amp_enable_r"}, +}; + +void max98925_regmap_write_stereo(struct max98925_priv *max98925, + unsigned int reg, unsigned int val) +{ + regmap_write(max98925->regmap_l, reg, val); + regmap_write(max98925->regmap_r, reg, val); +} + +static bool max98925_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98925_R000_VBAT_DATA: + case MAX98925_R001_VBST_DATA: + case MAX98925_R002_LIVE_STATUS0: + case MAX98925_R003_LIVE_STATUS1: + case MAX98925_R004_LIVE_STATUS2: + case MAX98925_R005_STATE0: + case MAX98925_R006_STATE1: + case MAX98925_R007_STATE2: + case MAX98925_R008_FLAG0: + case MAX98925_R009_FLAG1: + case MAX98925_R00A_FLAG2: + case MAX98925_R0FF_VERSION: + return true; + default: + return false; + } +} + +static bool max98925_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98925_R00E_IRQ_CLEAR0: + case MAX98925_R00F_IRQ_CLEAR1: + case MAX98925_R010_IRQ_CLEAR2: + case MAX98925_R033_ALC_HOLD_RLS: + return false; + default: + return true; + } +}; + +DECLARE_TLV_DB_SCALE(max98925_spk_tlv, -600, 100, 0); +static int reg_set_optimum_mode_check(struct regulator *reg, int load_ua) +{ + return (regulator_count_voltages(reg) > 0) ? + regulator_set_optimum_mode(reg, load_ua) : 0; +} + +static int max98925_regulator_config(struct i2c_client *i2c, + bool pullup, bool on) +{ + struct regulator *max98925_vcc_i2c; + int rc; + + if (pullup) { + pr_info("%s: i2c pull up\n", __func__); + max98925_vcc_i2c = regulator_get(&i2c->dev, "vcc_i2c"); + if (IS_ERR(max98925_vcc_i2c)) { + rc = PTR_ERR(max98925_vcc_i2c); + pr_err("%s: regulator get failed rc=%d\n", + __func__, rc); + goto error_get_vtg_i2c; + } + if (regulator_count_voltages(max98925_vcc_i2c) > 0) { + rc = regulator_set_voltage(max98925_vcc_i2c, + VCC_I2C_MIN_UV, VCC_I2C_MAX_UV); + if (rc) { + pr_err("%s: regulator set_vtg failed rc=%d\n", + __func__, rc); + goto error_set_vtg_i2c; + } + } + + rc = reg_set_optimum_mode_check(max98925_vcc_i2c, I2C_LOAD_UA); + if (rc < 0) { + pr_err("%s: regulator vcc_i2c set_opt failed rc=%d\n", + __func__, rc); + goto error_reg_opt_i2c; + } + + rc = regulator_enable(max98925_vcc_i2c); + if (rc) { + pr_err("%s: regulator vcc_i2c enable failed rc=%d\n", + __func__, rc); + goto error_reg_en_vcc_i2c; + } + + } + return 0; +error_set_vtg_i2c: + regulator_put(max98925_vcc_i2c); +error_get_vtg_i2c: + if (regulator_count_voltages(max98925_vcc_i2c) > 0) + regulator_set_voltage(max98925_vcc_i2c, 0, + VCC_I2C_MAX_UV); +error_reg_en_vcc_i2c: + if (pullup) + reg_set_optimum_mode_check(max98925_vcc_i2c, 0); +error_reg_opt_i2c: + regulator_disable(max98925_vcc_i2c); + return rc; +} + +static int max98925_spk_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = max98925->spk_gain; + dev_dbg(codec->dev, "%s: spk_gain setting returned %d\n", __func__, + (int) ucontrol->value.integer.value[0]); + return 0; +} + +static int max98925_spk_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + unsigned int sel = ucontrol->value.integer.value[0]; + + if (sel < ((1 << M98925_SPK_GAIN_WIDTH) - 1)) { + max98925_regmap_update_bits_stereo(max98925, MAX98925_R02D_GAIN, + M98925_SPK_GAIN_MASK, sel << M98925_SPK_GAIN_SHIFT); + max98925->spk_gain = sel; + + dev_dbg(codec->dev, "%s: spk_gain set to %d\n", __func__, sel); + } else { + dev_dbg(codec->dev, "%s: valid speaker gain settings: %d to %d\n", + __func__, 0, + ((1 << M98925_SPK_GAIN_WIDTH) - 1)); + } + return 0; +} + +static int max98925_reg_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, unsigned int reg, + unsigned int mask, unsigned int shift) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + int data; + + regmap_read(max98925->regmap_l, reg, &data); + ucontrol->value.integer.value[0] = + (data & mask) >> shift; + return 0; +} + +static int max98925_spk_ramp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_get(kcontrol, ucontrol, MAX98925_R02E_GAIN_RAMPING, + M98925_SPK_RMP_EN_MASK, M98925_SPK_RMP_EN_SHIFT); +} + +static int max98925_spk_ramp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_put(kcontrol, ucontrol, MAX98925_R02E_GAIN_RAMPING, + M98925_SPK_RMP_EN_MASK, M98925_SPK_RMP_EN_SHIFT); +} + +static int max98925_spk_zcd_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_get(kcontrol, ucontrol, MAX98925_R02E_GAIN_RAMPING, + M98925_SPK_ZCD_EN_MASK, M98925_SPK_ZCD_EN_SHIFT); +} + +static int max98925_spk_zcd_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_put(kcontrol, ucontrol, MAX98925_R02E_GAIN_RAMPING, + M98925_SPK_ZCD_EN_MASK, M98925_SPK_ZCD_EN_SHIFT); +} + +static int max98925_alc_en_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_get(kcontrol, ucontrol, MAX98925_R030_THRESHOLD, + M98925_ALC_EN_MASK, M98925_ALC_EN_SHIFT); +} + +static int max98925_alc_en_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_put(kcontrol, ucontrol, MAX98925_R030_THRESHOLD, + M98925_ALC_EN_MASK, M98925_ALC_EN_SHIFT); +} + +static int max98925_alc_threshold_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_get(kcontrol, ucontrol, MAX98925_R030_THRESHOLD, + M98925_ALC_TH_MASK, M98925_ALC_TH_SHIFT); +} + +static int max98925_alc_threshold_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_put(kcontrol, ucontrol, MAX98925_R030_THRESHOLD, + M98925_ALC_TH_MASK, M98925_ALC_TH_SHIFT); +} + +static const char * const max98925_boost_voltage_text[] = {"8.5V", "8.25V", + "8.0V", "7.75V", "7.5V", "7.25V", "7.0V", "6.75V", + "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", + "6.5V", "6.5V"}; + +static const struct soc_enum max98925_boost_voltage_enum = + SOC_ENUM_SINGLE(MAX98925_R037_CONFIGURATION, M98925_BST_VOUT_SHIFT, 15, + max98925_boost_voltage_text); + +static int max98925_boost_voltage_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_get(kcontrol, ucontrol, MAX98925_R037_CONFIGURATION, + M98925_BST_VOUT_MASK, M98925_BST_VOUT_SHIFT); +} + +static int max98925_boost_voltage_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_put(kcontrol, ucontrol, MAX98925_R037_CONFIGURATION, + M98925_BST_VOUT_MASK, M98925_BST_VOUT_SHIFT); +} + +static const struct snd_kcontrol_new max98925_snd_controls[] = { + + SOC_SINGLE_EXT_TLV("Speaker Gain", MAX98925_R02D_GAIN, + M98925_SPK_GAIN_SHIFT, (1<<M98925_SPK_GAIN_WIDTH)-1, 0, + max98925_spk_gain_get, max98925_spk_gain_put, max98925_spk_tlv), + + SOC_SINGLE_EXT("Speaker Ramp", 0, 0, 1, 0, + max98925_spk_ramp_get, max98925_spk_ramp_put), + + SOC_SINGLE_EXT("Speaker ZCD", 0, 0, 1, 0, + max98925_spk_zcd_get, max98925_spk_zcd_put), + + SOC_SINGLE_EXT("ALC Enable", 0, 0, 1, 0, + max98925_alc_en_get, max98925_alc_en_put), + + SOC_SINGLE_EXT("ALC Threshold", 0, 0, (1<<M98925_ALC_TH_WIDTH)-1, 0, + max98925_alc_threshold_get, max98925_alc_threshold_put), + + SOC_ENUM_EXT("Boost Output Voltage", max98925_boost_voltage_enum, + max98925_boost_voltage_get, max98925_boost_voltage_put), +}; + +/* codec sample rate and n/m dividers parameter table */ +static const struct { + int rate; + int sr; + int divisors[3][2]; +} rate_table[] = { + { + .rate = 8000, + .sr = 0, + .divisors = { {1, 375}, {5, 1764}, {1, 384} } + }, + { + .rate = 11025, + .sr = 1, + .divisors = { {147, 40000}, {1, 256}, {147, 40960} } + }, + { + .rate = 12000, + .sr = 2, + .divisors = { {1, 250}, {5, 1176}, {1, 256} } + }, + { + .rate = 16000, + .sr = 3, + .divisors = { {2, 375}, {5, 882}, {1, 192} } + }, + { + .rate = 22050, + .sr = 4, + .divisors = { {147, 20000}, {1, 128}, {147, 20480} } + }, + { + .rate = 24000, + .sr = 5, + .divisors = { {1, 125}, {5, 588}, {1, 128} } + }, + { + .rate = 32000, + .sr = 6, + .divisors = { {4, 375}, {5, 441}, {1, 96} } + }, + { + .rate = 44100, + .sr = 7, + .divisors = { {147, 10000}, {1, 64}, {147, 10240} } + }, + { + .rate = 48000, + .sr = 8, + .divisors = { {2, 125}, {5, 294}, {1, 64} } + }, +}; + +static inline int max98925_rate_value(int rate, + int clock, int *value, int *n, int *m) +{ + int ret = -EINVAL; + int i; + + for (i = 0; i < ARRAY_SIZE(rate_table); i++) { + if (rate_table[i].rate >= rate) { + *value = rate_table[i].sr; + *n = rate_table[i].divisors[clock][0]; + *m = rate_table[i].divisors[clock][1]; + ret = 0; + break; + } + } + pr_debug("%s: sample rate is %d, returning %d\n", + __func__, rate_table[i].rate, *value); + return ret; +} + +static void max98925_set_sense_data(struct max98925_priv *max98925) +{ + /* + * 1. set VMON slots + */ + regmap_update_bits(max98925->regmap_l, + MAX98925_R022_DOUT_CFG_VMON, + M98925_DAI_VMON_EN_MASK, M98925_DAI_VMON_EN_MASK); + regmap_update_bits(max98925->regmap_l, + MAX98925_R022_DOUT_CFG_VMON, + M98925_DAI_VMON_SLOT_MASK, M98925_DAI_VMON_SLOT_00_01); + /* + * 2. set IMON slots + */ + regmap_update_bits(max98925->regmap_l, + MAX98925_R023_DOUT_CFG_IMON, + M98925_DAI_IMON_EN_MASK, M98925_DAI_IMON_EN_MASK); + regmap_update_bits(max98925->regmap_l, + MAX98925_R023_DOUT_CFG_IMON, + M98925_DAI_IMON_SLOT_MASK, M98925_DAI_IMON_SLOT_02_03); +} + +static void max98925_set_slave(struct max98925_priv *max98925) +{ + /* + * 1. use BCLK instead of MCLK + */ + max98925_regmap_update_bits_stereo(max98925, + MAX98925_R01A_DAI_CLK_MODE1, + M98925_DAI_CLK_SOURCE_MASK, M98925_DAI_CLK_SOURCE_MASK); + /* + * 2. set DAI to slave mode + */ + max98925_regmap_update_bits_stereo(max98925, + MAX98925_R01B_DAI_CLK_MODE2, + M98925_DAI_MAS_MASK, 0); + /* + * 3. set BLCKs to LRCLKs to 64 + */ + max98925_regmap_update_bits_stereo(max98925, + MAX98925_R01B_DAI_CLK_MODE2, + M98925_DAI_BSEL_MASK, M98925_DAI_BSEL_64); + + max98925_set_sense_data(max98925); +} + +static void max98925_set_master(struct max98925_priv *max98925) +{ + /* + * 1. use MCLK for Left channel, right channel always BCLK + */ + regmap_update_bits(max98925->regmap_l, + MAX98925_R01A_DAI_CLK_MODE1, + M98925_DAI_CLK_SOURCE_MASK, 0); + regmap_update_bits(max98925->regmap_r, + MAX98925_R01A_DAI_CLK_MODE1, + M98925_DAI_CLK_SOURCE_MASK, + M98925_DAI_CLK_SOURCE_MASK); + /* + * 2. set left channel DAI to master mode, right channel always slave + */ + regmap_update_bits(max98925->regmap_l, + MAX98925_R01B_DAI_CLK_MODE2, + M98925_DAI_MAS_MASK, M98925_DAI_MAS_MASK); + regmap_update_bits(max98925->regmap_r, + MAX98925_R01B_DAI_CLK_MODE2, + M98925_DAI_MAS_MASK, 0); +} + +static int max98925_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + struct max98925_cdata *cdata; + unsigned int invert = 0; + + dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); + cdata = &max98925->dai[0]; + cdata->fmt = fmt; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + max98925_set_slave(max98925); + break; + case SND_SOC_DAIFMT_CBM_CFM: + max98925_set_master(max98925); + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + break; + case SND_SOC_DAIFMT_DSP_A: + default: + dev_err(codec->dev, "DAI format unsupported, fmt:0x%x", fmt); + return -EINVAL; + } + dev_dbg(codec->dev, "DAI format supported, fmt:0x%x", fmt); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + invert = M98925_DAI_WCI_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + invert = M98925_DAI_BCI_MASK; + break; + case SND_SOC_DAIFMT_IB_IF: + invert = M98925_DAI_BCI_MASK | M98925_DAI_WCI_MASK; + break; + default: + dev_err(codec->dev, "DAI invert mode unsupported"); + return -EINVAL; + } + + max98925_regmap_update_bits_stereo(max98925, MAX98925_R020_FORMAT, + M98925_DAI_BCI_MASK | M98925_DAI_BCI_MASK, invert); + return 0; +} + +static int max98925_set_clock(struct max98925_priv *max98925, + unsigned int rate) +{ + unsigned int dai_sr = 0; + unsigned int clock; + unsigned int mdll; + unsigned int n; + unsigned int m; + + switch (max98925->sysclk) { + case 6000000: + clock = 0; + mdll = M98925_MDLL_MULT_MCLKx16; + break; + case 11289600: + clock = 1; + mdll = M98925_MDLL_MULT_MCLKx8; + break; + case 12000000: + clock = 0; + mdll = M98925_MDLL_MULT_MCLKx8; + break; + case 12288000: + clock = 2; + mdll = M98925_MDLL_MULT_MCLKx8; + break; + default: + dev_info(max98925->codec->dev, "unsupported sysclk %d\n", + max98925->sysclk); + return -EINVAL; + } + + if (max98925_rate_value(rate, clock, &dai_sr, &n, &m)) + return -EINVAL; + + /* + * 1. set DAI_SR to correct LRCLK frequency + */ + max98925_regmap_update_bits_stereo(max98925, + MAX98925_R01B_DAI_CLK_MODE2, + M98925_DAI_SR_MASK, dai_sr << M98925_DAI_SR_SHIFT); + /* + * 2. set DAI m divider + */ + regmap_write(max98925->regmap_l, + MAX98925_R01C_DAI_CLK_DIV_M_MSBS, m >> 8); + regmap_write(max98925->regmap_l, + MAX98925_R01D_DAI_CLK_DIV_M_LSBS, m & 0xFF); + /* + * 3. set DAI n divider + */ + regmap_write(max98925->regmap_l, + MAX98925_R01E_DAI_CLK_DIV_N_MSBS, n >> 8); + regmap_write(max98925->regmap_l, + MAX98925_R01F_DAI_CLK_DIV_N_LSBS, n & 0xFF); + /* + * 4. set MDLL + */ + max98925_regmap_update_bits_stereo(max98925, + MAX98925_R01A_DAI_CLK_MODE1, + M98925_MDLL_MULT_MASK, mdll << M98925_MDLL_MULT_SHIFT); + return 0; +} + +static int max98925_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + struct max98925_cdata *cdata; + unsigned int rate; + + cdata = &max98925->dai[0]; + rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + max98925_regmap_update_bits_stereo(max98925, + MAX98925_R020_FORMAT, + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_16); + break; + case SNDRV_PCM_FORMAT_S24_LE: + max98925_regmap_update_bits_stereo(max98925, + MAX98925_R020_FORMAT, + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_32); + dev_dbg(codec->dev, "%s: (really set to 32 bits)\n", __func__); + break; + case SNDRV_PCM_FORMAT_S32_LE: + max98925_regmap_update_bits_stereo(max98925, + MAX98925_R020_FORMAT, + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_32); + break; + default: + pr_err("%s: format unsupported %d", + __func__, params_format(params)); + return -EINVAL; + } + dev_dbg(codec->dev, "%s: format supported %d", + __func__, params_format(params)); + return max98925_set_clock(max98925, rate); +} + +static int max98925_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s: clk_id %d, freq %d, dir %d\n", + __func__, clk_id, freq, dir); + max98925->sysclk = freq; + return 0; +} + +static int max98925_dai_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct max98925_priv *max98925 = + snd_soc_codec_get_drvdata(codec_dai->codec); + struct snd_soc_codec *codec = codec_dai->codec; + + dev_dbg(codec->dev, "%s: mute %d\n", __func__, mute); + if (mute) { + max98925_regmap_update_bits_stereo(max98925, MAX98925_R02D_GAIN, + M98925_SPK_GAIN_MASK, 0x00); + + max98925_regmap_update_bits_stereo(max98925, + MAX98925_R038_GLOBAL_ENABLE, + M98925_EN_MASK, 0x0); + } else { + max98925_regmap_update_bits_stereo(max98925, + MAX98925_R036_BLOCK_ENABLE, + M98925_BST_EN_MASK | + M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK, + M98925_BST_EN_MASK | + M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK); + + max98925_regmap_write_stereo(max98925, + MAX98925_R038_GLOBAL_ENABLE, M98925_EN_MASK); + } + return 0; +} + +#define MAX98925_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops max98925_dai_ops = { + .set_sysclk = max98925_dai_set_sysclk, + .set_fmt = max98925_dai_set_fmt, + .hw_params = max98925_dai_hw_params, + .digital_mute = max98925_dai_digital_mute, +}; + +static struct snd_soc_dai_driver max98925_dai[] = { + { + .name = "max98925-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98925_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98925_FORMATS, + }, + .ops = &max98925_dai_ops, + } +}; + +static int max98925_probe(struct snd_soc_codec *codec) +{ + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + struct max98925_cdata *cdata; + int ret = 0; + int reg = 0; + + dev_info(codec->dev, "build number %s\n", MAX98925_REVISION); + + max98925->codec = codec; + codec->control_data = max98925->regmap_l; + /* can be configured to any other value supported by this chip */ + max98925->sysclk = 12288000; + max98925->spk_gain = 0x14; + cdata = &max98925->dai[0]; + cdata->rate = (unsigned)-1; + cdata->fmt = (unsigned)-1; + + reg = 0; + ret = regmap_read(max98925->regmap_l, + MAX98925_R0FF_VERSION, &reg); + if ((ret < 0) || + ((reg != MAX98925_VERSION) && + (reg != MAX98925_VERSION1))) { + dev_err(codec->dev, + "L device initialization error (%d 0x%02X)\n", + ret, reg); + goto err_access; + } + dev_info(codec->dev, "L device version 0x%02X\n", reg); + + reg = 0; + ret = regmap_read(max98925->regmap_r, + MAX98925_R0FF_VERSION, &reg); + if ((ret < 0) || ((reg != MAX98925_VERSION) + && (reg != MAX98925_VERSION1))) { + dev_err(codec->dev, + "R device initialization error (%d 0x%02X)\n", + ret, reg); + goto err_access; + } + dev_info(codec->dev, "R device version 0x%02X\n", reg); + max98925_regmap_write_stereo(max98925, + MAX98925_R038_GLOBAL_ENABLE, 0x00); + + /* It's not the default but we need to set DAI_DLY */ + max98925_regmap_write_stereo(max98925, + MAX98925_R020_FORMAT, M98925_DAI_DLY_MASK); + max98925_regmap_write_stereo(max98925, + MAX98925_R021_TDM_SLOT_SELECT, 0xC8); + max98925_regmap_write_stereo(max98925, + MAX98925_R027_DOUT_HIZ_CFG1, 0xFF); + max98925_regmap_write_stereo(max98925, + MAX98925_R028_DOUT_HIZ_CFG2, 0xFF); + max98925_regmap_write_stereo(max98925, + MAX98925_R029_DOUT_HIZ_CFG3, 0xFF); + regmap_write(max98925->regmap_l, + MAX98925_R02A_DOUT_HIZ_CFG4, 0xF0); + regmap_write(max98925->regmap_r, + MAX98925_R02A_DOUT_HIZ_CFG4, 0x0F); + + max98925_regmap_write_stereo(max98925, + MAX98925_R02C_FILTERS, 0xD8); + max98925_regmap_write_stereo(max98925, + MAX98925_R034_ALC_CONFIGURATION, 0xF8); + max98925_regmap_write_stereo(max98925, + MAX98925_R037_CONFIGURATION, 0xF0); + + /* Disable ALC muting */ + max98925_regmap_write_stereo(max98925, + MAX98925_R03A_BOOST_LIMITER, 0xF8); + + regmap_update_bits(max98925->regmap_l, MAX98925_R02D_GAIN, + M98925_DAC_IN_SEL_MASK, M98925_DAC_IN_SEL_LEFT_DAI); + regmap_update_bits(max98925->regmap_r, MAX98925_R02D_GAIN, + M98925_DAC_IN_SEL_MASK, M98925_DAC_IN_SEL_RIGHT_DAI); + +err_access: + return ret; +} + +static struct snd_soc_codec_driver soc_codec_dev_max98925 = { + .probe = max98925_probe, + .controls = max98925_snd_controls, + .num_controls = ARRAY_SIZE(max98925_snd_controls), + .dapm_routes = max98925_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98925_audio_map), + .dapm_widgets = max98925_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98925_dapm_widgets), +}; + +static struct regmap_config max98925_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98925_R0FF_VERSION, + .reg_defaults = max98925_reg, + .num_reg_defaults = ARRAY_SIZE(max98925_reg), + .volatile_reg = max98925_volatile_register, + .readable_reg = max98925_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98925_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + int ret = 0; + + dev_info(&i2c->dev, "%s: enter, device '%s'\n", __func__, id->name); + if (!max98925) { + max98925 = devm_kzalloc(&i2c->dev, + sizeof(*max98925), GFP_KERNEL); + if (!max98925) + return -ENOMEM; + } + i2c_set_clientdata(i2c, max98925); + if (id->driver_data == MAX98925L) { + max98925->regmap_l = + devm_regmap_init_i2c(i2c, &max98925_regmap); + if (IS_ERR(max98925->regmap_l)) { + ret = PTR_ERR(max98925->regmap_l); + dev_err(&i2c->dev, + "Failed to allocate regmap_l: %d\n", ret); + goto err_out; + } + } + + /* Check for second MAX98925 */ + if (id->driver_data == MAX98925R) { + max98925->regmap_r = + devm_regmap_init_i2c(i2c, &max98925_regmap); + if (IS_ERR(max98925->regmap_r)) { + ret = PTR_ERR(max98925->regmap_r); + dev_err(&i2c->dev, + "Failed to allocate regmap_r: %d\n", ret); + goto err_out; + } + } + if (max98925->regmap_r && max98925->regmap_l) { + max98925_regulator_config(i2c, + of_property_read_bool(i2c->dev.of_node, + "max98925,i2c-pull-up"), 1); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98925, + max98925_dai, ARRAY_SIZE(max98925_dai)); + if (ret < 0) + dev_err(&i2c->dev, + "Failed to register codec: %d\n", ret); + } +err_out: + return ret; +} + +static int max98925_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id max98925_i2c_id[] = { + { "max98925L", MAX98925L }, + { "max98925R", MAX98925R }, + { }, +}; +MODULE_DEVICE_TABLE(i2c, max98925_i2c_id); + +static struct i2c_driver max98925_i2c_driver = { + .driver = { + .name = "max98925", + .owner = THIS_MODULE, + .pm = NULL, + }, + .probe = max98925_i2c_probe, + .remove = max98925_i2c_remove, + .id_table = max98925_i2c_id, +}; + +module_i2c_driver(max98925_i2c_driver); + +MODULE_DESCRIPTION("ALSA SoC MAX98925 Stereo driver"); +MODULE_AUTHOR("Ralph Birt <rdbirt(a)gmail.com>, Anish kumar <anish.kumar(a)maximintegrated.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98925.h b/sound/soc/codecs/max98925.h new file mode 100644 index 0000000..36b42d4 --- /dev/null +++ b/sound/soc/codecs/max98925.h @@ -0,0 +1,845 @@ +/* + * max98925.h -- MAX98925 ALSA SoC Audio driver + * + * Copyright 2013-2015 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX98925_H +#define _MAX98925_H + +/* Maximum number of MAX98925 devices in the system this driver can support */ +#define MAX_NUM_MAX98925 2 +#define MAX98925_VERSION 0x51 +#define MAX98925_VERSION1 0x80 + +#define MAX98925_REVISION "0.00.0333" + +/* + * MAX98925 Register Definitions + */ +#define MAX98925_R000_VBAT_DATA 0x00 +#define MAX98925_R001_VBST_DATA 0x01 +#define MAX98925_R002_LIVE_STATUS0 0x02 +#define MAX98925_R003_LIVE_STATUS1 0x03 +#define MAX98925_R004_LIVE_STATUS2 0x04 +#define MAX98925_R005_STATE0 0x05 +#define MAX98925_R006_STATE1 0x06 +#define MAX98925_R007_STATE2 0x07 +#define MAX98925_R008_FLAG0 0x08 +#define MAX98925_R009_FLAG1 0x09 +#define MAX98925_R00A_FLAG2 0x0A +#define MAX98925_R00B_IRQ_ENABLE0 0x0B +#define MAX98925_R00C_IRQ_ENABLE1 0x0C +#define MAX98925_R00D_IRQ_ENABLE2 0x0D +#define MAX98925_R00E_IRQ_CLEAR0 0x0E +#define MAX98925_R00F_IRQ_CLEAR1 0x0F +#define MAX98925_R010_IRQ_CLEAR2 0x10 +#define MAX98925_R011_MAP0 0x11 +#define MAX98925_R012_MAP1 0x12 +#define MAX98925_R013_MAP2 0x13 +#define MAX98925_R014_MAP3 0x14 +#define MAX98925_R015_MAP4 0x15 +#define MAX98925_R016_MAP5 0x16 +#define MAX98925_R017_MAP6 0x17 +#define MAX98925_R018_MAP7 0x18 +#define MAX98925_R019_MAP8 0x19 +#define MAX98925_R01A_DAI_CLK_MODE1 0x1A +#define MAX98925_R01B_DAI_CLK_MODE2 0x1B +#define MAX98925_R01C_DAI_CLK_DIV_M_MSBS 0x1C +#define MAX98925_R01D_DAI_CLK_DIV_M_LSBS 0x1D +#define MAX98925_R01E_DAI_CLK_DIV_N_MSBS 0x1E +#define MAX98925_R01F_DAI_CLK_DIV_N_LSBS 0x1F +#define MAX98925_R020_FORMAT 0x20 +#define MAX98925_R021_TDM_SLOT_SELECT 0x21 +#define MAX98925_R022_DOUT_CFG_VMON 0x22 +#define MAX98925_R023_DOUT_CFG_IMON 0x23 +#define MAX98925_R024_DOUT_CFG_VBAT 0x24 +#define MAX98925_R025_DOUT_CFG_VBST 0x25 +#define MAX98925_R026_DOUT_CFG_FLAG 0x26 +#define MAX98925_R027_DOUT_HIZ_CFG1 0x27 +#define MAX98925_R028_DOUT_HIZ_CFG2 0x28 +#define MAX98925_R029_DOUT_HIZ_CFG3 0x29 +#define MAX98925_R02A_DOUT_HIZ_CFG4 0x2A +#define MAX98925_R02B_DOUT_DRV_STRENGTH 0x2B +#define MAX98925_R02C_FILTERS 0x2C +#define MAX98925_R02D_GAIN 0x2D +#define MAX98925_R02E_GAIN_RAMPING 0x2E +#define MAX98925_R02F_SPK_AMP 0x2F +#define MAX98925_R030_THRESHOLD 0x30 +#define MAX98925_R031_ALC_ATTACK 0x31 +#define MAX98925_R032_ALC_ATTEN_RLS 0x32 +#define MAX98925_R033_ALC_HOLD_RLS 0x33 +#define MAX98925_R034_ALC_CONFIGURATION 0x34 +#define MAX98925_R035_BOOST_CONVERTER 0x35 +#define MAX98925_R036_BLOCK_ENABLE 0x36 +#define MAX98925_R037_CONFIGURATION 0x37 +#define MAX98925_R038_GLOBAL_ENABLE 0x38 +#define MAX98925_R03A_BOOST_LIMITER 0x3A +#define MAX98925_R0FF_VERSION 0xFF + +#define MAX98925_REG_CNT (MAX98925_R03A_BOOST_LIMITER+1) + +/* MAX98925 Register Bit Fields */ + +/* MAX98925_R002_LIVE_STATUS0 */ +#define M98925_THERMWARN_STATUS_MASK (1<<3) +#define M98925_THERMWARN_STATUS_SHIFT 3 +#define M98925_THERMWARN_STATUS_WIDTH 1 +#define M98925_THERMSHDN_STATUS_MASK (1<<1) +#define M98925_THERMSHDN_STATUS_SHIFT 1 +#define M98925_THERMSHDN_STATUS_WIDTH 1 + +/* MAX98925_R003_LIVE_STATUS1 */ +#define M98925_SPKCURNT_STATUS_MASK (1<<5) +#define M98925_SPKCURNT_STATUS_SHIFT 5 +#define M98925_SPKCURNT_STATUS_WIDTH 1 +#define M98925_WATCHFAIL_STATUS_MASK (1<<4) +#define M98925_WATCHFAIL_STATUS_SHIFT 4 +#define M98925_WATCHFAIL_STATUS_WIDTH 1 +#define M98925_ALCINFH_STATUS_MASK (1<<3) +#define M98925_ALCINFH_STATUS_SHIFT 3 +#define M98925_ALCINFH_STATUS_WIDTH 1 +#define M98925_ALCACT_STATUS_MASK (1<<2) +#define M98925_ALCACT_STATUS_SHIFT 2 +#define M98925_ALCACT_STATUS_WIDTH 1 +#define M98925_ALCMUT_STATUS_MASK (1<<1) +#define M98925_ALCMUT_STATUS_SHIFT 1 +#define M98925_ALCMUT_STATUS_WIDTH 1 +#define M98925_ACLP_STATUS_MASK (1<<0) +#define M98925_ACLP_STATUS_SHIFT 0 +#define M98925_ACLP_STATUS_WIDTH 1 + +/* MAX98925_R004_LIVE_STATUS2 */ +#define M98925_SLOTOVRN_STATUS_MASK (1<<6) +#define M98925_SLOTOVRN_STATUS_SHIFT 6 +#define M98925_SLOTOVRN_STATUS_WIDTH 1 +#define M98925_INVALSLOT_STATUS_MASK (1<<5) +#define M98925_INVALSLOT_STATUS_SHIFT 5 +#define M98925_INVALSLOT_STATUS_WIDTH 1 +#define M98925_SLOTCNFLT_STATUS_MASK (1<<4) +#define M98925_SLOTCNFLT_STATUS_SHIFT 4 +#define M98925_SLOTCNFLT_STATUS_WIDTH 1 +#define M98925_VBSTOVFL_STATUS_MASK (1<<3) +#define M98925_VBSTOVFL_STATUS_SHIFT 3 +#define M98925_VBSTOVFL_STATUS_WIDTH 1 +#define M98925_VBATOVFL_STATUS_MASK (1<<2) +#define M98925_VBATOVFL_STATUS_SHIFT 2 +#define M98925_VBATOVFL_STATUS_WIDTH 1 +#define M98925_IMONOVFL_STATUS_MASK (1<<1) +#define M98925_IMONOVFL_STATUS_SHIFT 1 +#define M98925_IMONOVFL_STATUS_WIDTH 1 +#define M98925_VMONOVFL_STATUS_MASK (1<<0) +#define M98925_VMONOVFL_STATUS_SHIFT 0 +#define M98925_VMONOVFL_STATUS_WIDTH 1 + +/* MAX98925_R005_STATE0 */ +#define M98925_THERMWARN_END_STATE_MASK (1<<3) +#define M98925_THERMWARN_END_STATE_SHIFT 3 +#define M98925_THERMWARN_END_STATE_WIDTH 1 +#define M98925_THERMWARN_BGN_STATE_MASK (1<<2) +#define M98925_THERMWARN_BGN_STATE_SHIFT 1 +#define M98925_THERMWARN_BGN_STATE_WIDTH 1 +#define M98925_THERMSHDN_END_STATE_MASK (1<<1) +#define M98925_THERMSHDN_END_STATE_SHIFT 1 +#define M98925_THERMSHDN_END_STATE_WIDTH 1 +#define M98925_THERMSHDN_BGN_STATE_MASK (1<<0) +#define M98925_THERMSHDN_BGN_STATE_SHIFT 0 +#define M98925_THERMSHDN_BGN_STATE_WIDTH 1 + +/* MAX98925_R006_STATE1 */ +#define M98925_SPRCURNT_STATE_MASK (1<<5) +#define M98925_SPRCURNT_STATE_SHIFT 5 +#define M98925_SPRCURNT_STATE_WIDTH 1 +#define M98925_WATCHFAIL_STATE_MASK (1<<4) +#define M98925_WATCHFAIL_STATE_SHIFT 4 +#define M98925_WATCHFAIL_STATE_WIDTH 1 +#define M98925_ALCINFH_STATE_MASK (1<<3) +#define M98925_ALCINFH_STATE_SHIFT 3 +#define M98925_ALCINFH_STATE_WIDTH 1 +#define M98925_ALCACT_STATE_MASK (1<<2) +#define M98925_ALCACT_STATE_SHIFT 2 +#define M98925_ALCACT_STATE_WIDTH 1 +#define M98925_ALCMUT_STATE_MASK (1<<1) +#define M98925_ALCMUT_STATE_SHIFT 1 +#define M98925_ALCMUT_STATE_WIDTH 1 +#define M98925_ALCP_STATE_MASK (1<<0) +#define M98925_ALCP_STATE_SHIFT 0 +#define M98925_ALCP_STATE_WIDTH 1 + +/* MAX98925_R007_STATE2 */ +#define M98925_SLOTOVRN_STATE_MASK (1<<6) +#define M98925_SLOTOVRN_STATE_SHIFT 6 +#define M98925_SLOTOVRN_STATE_WIDTH 1 +#define M98925_INVALSLOT_STATE_MASK (1<<5) +#define M98925_INVALSLOT_STATE_SHIFT 5 +#define M98925_INVALSLOT_STATE_WIDTH 1 +#define M98925_SLOTCNFLT_STATE_MASK (1<<4) +#define M98925_SLOTCNFLT_STATE_SHIFT 4 +#define M98925_SLOTCNFLT_STATE_WIDTH 1 +#define M98925_VBSTOVFL_STATE_MASK (1<<3) +#define M98925_VBSTOVFL_STATE_SHIFT 3 +#define M98925_VBSTOVFL_STATE_WIDTH 1 +#define M98925_VBATOVFL_STATE_MASK (1<<2) +#define M98925_VBATOVFL_STATE_SHIFT 2 +#define M98925_VBATOVFL_STATE_WIDTH 1 +#define M98925_IMONOVFL_STATE_MASK (1<<1) +#define M98925_IMONOVFL_STATE_SHIFT 1 +#define M98925_IMONOVFL_STATE_WIDTH 1 +#define M98925_VMONOVFL_STATE_MASK (1<<0) +#define M98925_VMONOVFL_STATE_SHIFT 0 +#define M98925_VMONOVFL_STATE_WIDTH 1 + +/* MAX98925_R008_FLAG0 */ +#define M98925_THERMWARN_END_FLAG_MASK (1<<3) +#define M98925_THERMWARN_END_FLAG_SHIFT 3 +#define M98925_THERMWARN_END_FLAG_WIDTH 1 +#define M98925_THERMWARN_BGN_FLAG_MASK (1<<2) +#define M98925_THERMWARN_BGN_FLAG_SHIFT 2 +#define M98925_THERMWARN_BGN_FLAG_WIDTH 1 +#define M98925_THERMSHDN_END_FLAG_MASK (1<<1) +#define M98925_THERMSHDN_END_FLAG_SHIFT 1 +#define M98925_THERMSHDN_END_FLAG_WIDTH 1 +#define M98925_THERMSHDN_BGN_FLAG_MASK (1<<0) +#define M98925_THERMSHDN_BGN_FLAG_SHIFT 0 +#define M98925_THERMSHDN_BGN_FLAG_WIDTH 1 + +/* MAX98925_R009_FLAG1 */ +#define M98925_SPKCURNT_FLAG_MASK (1<<5) +#define M98925_SPKCURNT_FLAG_SHIFT 5 +#define M98925_SPKCURNT_FLAG_WIDTH 1 +#define M98925_WATCHFAIL_FLAG_MASK (1<<4) +#define M98925_WATCHFAIL_FLAG_SHIFT 4 +#define M98925_WATCHFAIL_FLAG_WIDTH 1 +#define M98925_ALCINFH_FLAG_MASK (1<<3) +#define M98925_ALCINFH_FLAG_SHIFT 3 +#define M98925_ALCINFH_FLAG_WIDTH 1 +#define M98925_ALCACT_FLAG_MASK (1<<2) +#define M98925_ALCACT_FLAG_SHIFT 2 +#define M98925_ALCACT_FLAG_WIDTH 1 +#define M98925_ALCMUT_FLAG_MASK (1<<1) +#define M98925_ALCMUT_FLAG_SHIFT 1 +#define M98925_ALCMUT_FLAG_WIDTH 1 +#define M98925_ALCP_FLAG_MASK (1<<0) +#define M98925_ALCP_FLAG_SHIFT 0 +#define M98925_ALCP_FLAG_WIDTH 1 + +/* MAX98925_R00A_FLAG2 */ +#define M98925_SLOTOVRN_FLAG_MASK (1<<6) +#define M98925_SLOTOVRN_FLAG_SHIFT 6 +#define M98925_SLOTOVRN_FLAG_WIDTH 1 +#define M98925_INVALSLOT_FLAG_MASK (1<<5) +#define M98925_INVALSLOT_FLAG_SHIFT 5 +#define M98925_INVALSLOT_FLAG_WIDTH 1 +#define M98925_SLOTCNFLT_FLAG_MASK (1<<4) +#define M98925_SLOTCNFLT_FLAG_SHIFT 4 +#define M98925_SLOTCNFLT_FLAG_WIDTH 1 +#define M98925_VBSTOVFL_FLAG_MASK (1<<3) +#define M98925_VBSTOVFL_FLAG_SHIFT 3 +#define M98925_VBSTOVFL_FLAG_WIDTH 1 +#define M98925_VBATOVFL_FLAG_MASK (1<<2) +#define M98925_VBATOVFL_FLAG_SHIFT 2 +#define M98925_VBATOVFL_FLAG_WIDTH 1 +#define M98925_IMONOVFL_FLAG_MASK (1<<1) +#define M98925_IMONOVFL_FLAG_SHIFT 1 +#define M98925_IMONOVFL_FLAG_WIDTH 1 +#define M98925_VMONOVFL_FLAG_MASK (1<<0) +#define M98925_VMONOVFL_FLAG_SHIFT 0 +#define M98925_VMONOVFL_FLAG_WIDTH 1 + +/* MAX98925_R00B_IRQ_ENABLE0 */ +#define M98925_THERMWARN_END_EN_MASK (1<<3) +#define M98925_THERMWARN_END_EN_SHIFT 3 +#define M98925_THERMWARN_END_EN_WIDTH 1 +#define M98925_THERMWARN_BGN_EN_MASK (1<<2) +#define M98925_THERMWARN_BGN_EN_SHIFT 2 +#define M98925_THERMWARN_BGN_EN_WIDTH 1 +#define M98925_THERMSHDN_END_EN_MASK (1<<1) +#define M98925_THERMSHDN_END_EN_SHIFT 1 +#define M98925_THERMSHDN_END_EN_WIDTH 1 +#define M98925_THERMSHDN_BGN_EN_MASK (1<<0) +#define M98925_THERMSHDN_BGN_EN_SHIFT 0 +#define M98925_THERMSHDN_BGN_EN_WIDTH 1 + +/* MAX98925_R00C_IRQ_ENABLE1 */ +#define M98925_SPKCURNT_EN_MASK (1<<5) +#define M98925_SPKCURNT_EN_SHIFT 5 +#define M98925_SPKCURNT_EN_WIDTH 1 +#define M98925_WATCHFAIL_EN_MASK (1<<4) +#define M98925_WATCHFAIL_EN_SHIFT 4 +#define M98925_WATCHFAIL_EN_WIDTH 1 +#define M98925_ALCINFH_EN_MASK (1<<3) +#define M98925_ALCINFH_EN_SHIFT 3 +#define M98925_ALCINFH_EN_WIDTH 1 +#define M98925_ALCACT_EN_MASK (1<<2) +#define M98925_ALCACT_EN_SHIFT 2 +#define M98925_ALCACT_EN_WIDTH 1 +#define M98925_ALCMUT_EN_MASK (1<<1) +#define M98925_ALCMUT_EN_SHIFT 1 +#define M98925_ALCMUT_EN_WIDTH 1 +#define M98925_ALCP_EN_MASK (1<<0) +#define M98925_ALCP_EN_SHIFT 0 +#define M98925_ALCP_EN_WIDTH 1 + +/* MAX98925_R00D_IRQ_ENABLE2 */ +#define M98925_SLOTOVRN_EN_MASK (1<<6) +#define M98925_SLOTOVRN_EN_SHIFT 6 +#define M98925_SLOTOVRN_EN_WIDTH 1 +#define M98925_INVALSLOT_EN_MASK (1<<5) +#define M98925_INVALSLOT_EN_SHIFT 5 +#define M98925_INVALSLOT_EN_WIDTH 1 +#define M98925_SLOTCNFLT_EN_MASK (1<<4) +#define M98925_SLOTCNFLT_EN_SHIFT 4 +#define M98925_SLOTCNFLT_EN_WIDTH 1 +#define M98925_VBSTOVFL_EN_MASK (1<<3) +#define M98925_VBSTOVFL_EN_SHIFT 3 +#define M98925_VBSTOVFL_EN_WIDTH 1 +#define M98925_VBATOVFL_EN_MASK (1<<2) +#define M98925_VBATOVFL_EN_SHIFT 2 +#define M98925_VBATOVFL_EN_WIDTH 1 +#define M98925_IMONOVFL_EN_MASK (1<<1) +#define M98925_IMONOVFL_EN_SHIFT 1 +#define M98925_IMONOVFL_EN_WIDTH 1 +#define M98925_VMONOVFL_EN_MASK (1<<0) +#define M98925_VMONOVFL_EN_SHIFT 0 +#define M98925_VMONOVFL_EN_WIDTH 1 + +/* MAX98925_R00E_IRQ_CLEAR0 */ +#define M98925_THERMWARN_END_CLR_MASK (1<<3) +#define M98925_THERMWARN_END_CLR_SHIFT 3 +#define M98925_THERMWARN_END_CLR_WIDTH 1 +#define M98925_THERMWARN_BGN_CLR_MASK (1<<2) +#define M98925_THERMWARN_BGN_CLR_SHIFT 2 +#define M98925_THERMWARN_BGN_CLR_WIDTH 1 +#define M98925_THERMSHDN_END_CLR_MASK (1<<1) +#define M98925_THERMSHDN_END_CLR_SHIFT 1 +#define M98925_THERMSHDN_END_CLR_WIDTH 1 +#define M98925_THERMSHDN_BGN_CLR_MASK (1<<0) +#define M98925_THERMSHDN_BGN_CLR_SHIFT 0 +#define M98925_THERMSHDN_BGN_CLR_WIDTH 1 + +/* MAX98925_R00F_IRQ_CLEAR1 */ +#define M98925_SPKCURNT_CLR_MASK (1<<5) +#define M98925_SPKCURNT_CLR_SHIFT 5 +#define M98925_SPKCURNT_CLR_WIDTH 1 +#define M98925_WATCHFAIL_CLR_MASK (1<<4) +#define M98925_WATCHFAIL_CLR_SHIFT 4 +#define M98925_WATCHFAIL_CLR_WIDTH 1 +#define M98925_ALCINFH_CLR_MASK (1<<3) +#define M98925_ALCINFH_CLR_SHIFT 3 +#define M98925_ALCINFH_CLR_WIDTH 1 +#define M98925_ALCACT_CLR_MASK (1<<2) +#define M98925_ALCACT_CLR_SHIFT 2 +#define M98925_ALCACT_CLR_WIDTH 1 +#define M98925_ALCMUT_CLR_MASK (1<<1) +#define M98925_ALCMUT_CLR_SHIFT 1 +#define M98925_ALCMUT_CLR_WIDTH 1 +#define M98925_ALCP_CLR_MASK (1<<0) +#define M98925_ALCP_CLR_SHIFT 0 +#define M98925_ALCP_CLR_WIDTH 1 + +/* MAX98925_R010_IRQ_CLEAR2 */ +#define M98925_SLOTOVRN_CLR_MASK (1<<6) +#define M98925_SLOTOVRN_CLR_SHIFT 6 +#define M98925_SLOTOVRN_CLR_WIDTH 1 +#define M98925_INVALSLOT_CLR_MASK (1<<5) +#define M98925_INVALSLOT_CLR_SHIFT 5 +#define M98925_INVALSLOT_CLR_WIDTH 1 +#define M98925_SLOTCNFLT_CLR_MASK (1<<4) +#define M98925_SLOTCNFLT_CLR_SHIFT 4 +#define M98925_SLOTCNFLT_CLR_WIDTH 1 +#define M98925_VBSTOVFL_CLR_MASK (1<<3) +#define M98925_VBSTOVFL_CLR_SHIFT 3 +#define M98925_VBSTOVFL_CLR_WIDTH 1 +#define M98925_VBATOVFL_CLR_MASK (1<<2) +#define M98925_VBATOVFL_CLR_SHIFT 2 +#define M98925_VBATOVFL_CLR_WIDTH 1 +#define M98925_IMONOVFL_CLR_MASK (1<<1) +#define M98925_IMONOVFL_CLR_SHIFT 1 +#define M98925_IMONOVFL_CLR_WIDTH 1 +#define M98925_VMONOVFL_CLR_MASK (1<<0) +#define M98925_VMONOVFL_CLR_SHIFT 0 +#define M98925_VMONOVFL_CLR_WIDTH 1 + +/* MAX98925_R011_MAP0 */ +#define M98925_ER_THERMWARN_EN_MASK (1<<7) +#define M98925_ER_THERMWARN_EN_SHIFT 7 +#define M98925_ER_THERMWARN_EN_WIDTH 1 +#define M98925_ER_THERMWARN_MAP_MASK (0x07<<4) +#define M98925_ER_THERMWARN_MAP_SHIFT 4 +#define M98925_ER_THERMWARN_MAP_WIDTH 3 + +/* MAX98925_R012_MAP1 */ +#define M98925_ER_ALCMUT_EN_MASK (1<<7) +#define M98925_ER_ALCMUT_EN_SHIFT 7 +#define M98925_ER_ALCMUT_EN_WIDTH 1 +#define M98925_ER_ALCMUT_MAP_MASK (0x07<<4) +#define M98925_ER_ALCMUT_MAP_SHIFT 4 +#define M98925_ER_ALCMUT_MAP_WIDTH 3 +#define M98925_ER_ALCP_EN_MASK (1<<3) +#define M98925_ER_ALCP_EN_SHIFT 3 +#define M98925_ER_ALCP_EN_WIDTH 1 +#define M98925_ER_ALCP_MAP_MASK (0x07<<0) +#define M98925_ER_ALCP_MAP_SHIFT 0 +#define M98925_ER_ALCP_MAP_WIDTH 3 + +/* MAX98925_R013_MAP2 */ +#define M98925_ER_ALCINFH_EN_MASK (1<<7) +#define M98925_ER_ALCINFH_EN_SHIFT 7 +#define M98925_ER_ALCINFH_EN_WIDTH 1 +#define M98925_ER_ALCINFH_MAP_MASK (0x07<<4) +#define M98925_ER_ALCINFH_MAP_SHIFT 4 +#define M98925_ER_ALCINFH_MAP_WIDTH 3 +#define M98925_ER_ALCACT_EN_MASK (1<<3) +#define M98925_ER_ALCACT_EN_SHIFT 3 +#define M98925_ER_ALCACT_EN_WIDTH 1 +#define M98925_ER_ALCACT_MAP_MASK (0x07<<0) +#define M98925_ER_ALCACT_MAP_SHIFT 0 +#define M98925_ER_ALCACT_MAP_WIDTH 3 + +/* MAX98925_R014_MAP3 */ +#define M98925_ER_SPKCURNT_EN_MASK (1<<7) +#define M98925_ER_SPKCURNT_EN_SHIFT 7 +#define M98925_ER_SPKCURNT_EN_WIDTH 1 +#define M98925_ER_SPKCURNT_MAP_MASK (0x07<<4) +#define M98925_ER_SPKCURNT_MAP_SHIFT 4 +#define M98925_ER_SPKCURNT_MAP_WIDTH 3 + +/* MAX98925_R015_MAP4 */ +/* RESERVED */ + +/* MAX98925_R016_MAP5 */ +#define M98925_ER_IMONOVFL_EN_MASK (1<<7) +#define M98925_ER_IMONOVFL_EN_SHIFT 7 +#define M98925_ER_IMONOVFL_EN_WIDTH 1 +#define M98925_ER_IMONOVFL_MAP_MASK (0x07<<4) +#define M98925_ER_IMONOVFL_MAP_SHIFT 4 +#define M98925_ER_IMONOVFL_MAP_WIDTH 3 +#define M98925_ER_VMONOVFL_EN_MASK (1<<3) +#define M98925_ER_VMONOVFL_EN_SHIFT 3 +#define M98925_ER_VMONOVFL_EN_WIDTH 1 +#define M98925_ER_VMONOVFL_MAP_MASK (0x07<<0) +#define M98925_ER_VMONOVFL_MAP_SHIFT 0 +#define M98925_ER_VMONOVFL_MAP_WIDTH 3 + +/* MAX98925_R017_MAP6 */ +#define M98925_ER_VBSTOVFL_EN_MASK (1<<7) +#define M98925_ER_VBSTOVFL_EN_SHIFT 7 +#define M98925_ER_VBSTOVFL_EN_WIDTH 1 +#define M98925_ER_VBSTOVFL_MAP_MASK (0x07<<4) +#define M98925_ER_VBSTOVFL_MAP_SHIFT 4 +#define M98925_ER_VBSTOVFL_MAP_WIDTH 3 +#define M98925_ER_VBATOVFL_EN_MASK (1<<3) +#define M98925_ER_VBATOVFL_EN_SHIFT 3 +#define M98925_ER_VBATOVFL_EN_WIDTH 1 +#define M98925_ER_VBATOVFL_MAP_MASK (0x07<<0) +#define M98925_ER_VBATOVFL_MAP_SHIFT 0 +#define M98925_ER_VBATOVFL_MAP_WIDTH 3 + +/* MAX98925_R018_MAP7 */ +#define M98925_ER_INVALSLOT_EN_MASK (1<<7) +#define M98925_ER_INVALSLOT_EN_SHIFT 7 +#define M98925_ER_INVALSLOT_EN_WIDTH 1 +#define M98925_ER_INVALSLOT_MAP_MASK (0x07<<4) +#define M98925_ER_INVALSLOT_MAP_SHIFT 4 +#define M98925_ER_INVALSLOT_MAP_WIDTH 3 +#define M98925_ER_SLOTCNFLT_EN_MASK (1<<3) +#define M98925_ER_SLOTCNFLT_EN_SHIFT 3 +#define M98925_ER_SLOTCNFLT_EN_WIDTH 1 +#define M98925_ER_SLOTCNFLT_MAP_MASK (0x07<<0) +#define M98925_ER_SLOTCNFLT_MAP_SHIFT 0 +#define M98925_ER_SLOTCNFLT_MAP_WIDTH 3 + +/* MAX98925_R019_MAP8 */ +#define M98925_ER_SLOTOVRN_EN_MASK (1<<3) +#define M98925_ER_SLOTOVRN_EN_SHIFT 3 +#define M98925_ER_SLOTOVRN_EN_WIDTH 1 +#define M98925_ER_SLOTOVRN_MAP_MASK (0x07<<0) +#define M98925_ER_SLOTOVRN_MAP_SHIFT 0 +#define M98925_ER_SLOTOVRN_MAP_WIDTH 3 + +/* MAX98925_R01A_DAI_CLK_MODE1 */ +#define M98925_DAI_CLK_SOURCE_MASK (1<<6) +#define M98925_DAI_CLK_SOURCE_SHIFT 6 +#define M98925_DAI_CLK_SOURCE_WIDTH 1 +#define M98925_MDLL_MULT_MASK (0x0F<<0) +#define M98925_MDLL_MULT_SHIFT 0 +#define M98925_MDLL_MULT_WIDTH 4 + +#define M98925_MDLL_MULT_MCLKx8 6 +#define M98925_MDLL_MULT_MCLKx16 8 + +/* MAX98925_R01B_DAI_CLK_MODE2 */ +#define M98925_DAI_SR_MASK (0x0F<<4) +#define M98925_DAI_SR_SHIFT 4 +#define M98925_DAI_SR_WIDTH 4 +#define M98925_DAI_MAS_MASK (1<<3) +#define M98925_DAI_MAS_SHIFT 3 +#define M98925_DAI_MAS_WIDTH 1 +#define M98925_DAI_BSEL_MASK (0x07<<0) +#define M98925_DAI_BSEL_SHIFT 0 +#define M98925_DAI_BSEL_WIDTH 3 + +#define M98925_DAI_BSEL_32 (0 << M98925_DAI_BSEL_SHIFT) +#define M98925_DAI_BSEL_48 (1 << M98925_DAI_BSEL_SHIFT) +#define M98925_DAI_BSEL_64 (2 << M98925_DAI_BSEL_SHIFT) +#define M98925_DAI_BSEL_256 (6 << M98925_DAI_BSEL_SHIFT) + +/* MAX98925_R01C_DAI_CLK_DIV_M_MSBS */ +#define M98925_DAI_M_MSBS_MASK (0xFF<<0) +#define M98925_DAI_M_MSBS_SHIFT 0 +#define M98925_DAI_M_MSBS_WIDTH 8 + +/* MAX98925_R01D_DAI_CLK_DIV_M_LSBS */ +#define M98925_DAI_M_LSBS_MASK (0xFF<<0) +#define M98925_DAI_M_LSBS_SHIFT 0 +#define M98925_DAI_M_LSBS_WIDTH 8 + +/* MAX98925_R01E_DAI_CLK_DIV_N_MSBS */ +#define M98925_DAI_N_MSBS_MASK (0x7F<<0) +#define M98925_DAI_N_MSBS_SHIFT 0 +#define M98925_DAI_N_MSBS_WIDTH 7 + +/* MAX98925_R01F_DAI_CLK_DIV_N_LSBS */ +#define M98925_DAI_N_LSBS_MASK (0xFF<<0) +#define M98925_DAI_N_LSBS_SHIFT 0 +#define M98925_DAI_N_LSBS_WIDTH 8 + +/* MAX98925_R020_FORMAT */ +#define M98925_DAI_CHANSZ_MASK (0x03<<6) +#define M98925_DAI_CHANSZ_SHIFT 6 +#define M98925_DAI_CHANSZ_WIDTH 2 +#define M98925_DAI_EXTBCLK_HIZ_MASK (1<<4) +#define M98925_DAI_EXTBCLK_HIZ_SHIFT 4 +#define M98925_DAI_EXTBCLK_HIZ_WIDTH 1 +#define M98925_DAI_WCI_MASK (1<<3) +#define M98925_DAI_WCI_SHIFT 3 +#define M98925_DAI_WCI_WIDTH 1 +#define M98925_DAI_BCI_MASK (1<<2) +#define M98925_DAI_BCI_SHIFT 2 +#define M98925_DAI_BCI_WIDTH 1 +#define M98925_DAI_DLY_MASK (1<<1) +#define M98925_DAI_DLY_SHIFT 1 +#define M98925_DAI_DLY_WIDTH 1 +#define M98925_DAI_TDM_MASK (1<<0) +#define M98925_DAI_TDM_SHIFT 0 +#define M98925_DAI_TDM_WIDTH 1 + +#define M98925_DAI_CHANSZ_16 (1 << M98925_DAI_CHANSZ_SHIFT) +#define M98925_DAI_CHANSZ_24 (2 << M98925_DAI_CHANSZ_SHIFT) +#define M98925_DAI_CHANSZ_32 (3 << M98925_DAI_CHANSZ_SHIFT) + +/* MAX98925_R021_TDM_SLOT_SELECT */ +#define M98925_DAI_DO_EN_MASK (1<<7) +#define M98925_DAI_DO_EN_SHIFT 7 +#define M98925_DAI_DO_EN_WIDTH 1 +#define M98925_DAI_DIN_EN_MASK (1<<6) +#define M98925_DAI_DIN_EN_SHIFT 6 +#define M98925_DAI_DIN_EN_WIDTH 1 +#define M98925_DAI_INR_SOURCE_MASK (0x07<<3) +#define M98925_DAI_INR_SOURCE_SHIFT 3 +#define M98925_DAI_INR_SOURCE_WIDTH 3 +#define M98925_DAI_INL_SOURCE_MASK (0x07<<0) +#define M98925_DAI_INL_SOURCE_SHIFT 0 +#define M98925_DAI_INL_SOURCE_WIDTH 3 + +/* MAX98925_R022_DOUT_CFG_VMON */ +#define M98925_DAI_VMON_EN_MASK (1<<5) +#define M98925_DAI_VMON_EN_SHIFT 5 +#define M98925_DAI_VMON_EN_WIDTH 1 +#define M98925_DAI_VMON_SLOT_MASK (0x1F<<0) +#define M98925_DAI_VMON_SLOT_SHIFT 0 +#define M98925_DAI_VMON_SLOT_WIDTH 5 + +#define M98925_DAI_VMON_SLOT_00_01 (0 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_01_02 (1 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_02_03 (2 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_03_04 (3 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_04_05 (4 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_05_06 (5 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_06_07 (6 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_07_08 (7 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_08_09 (8 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_09_0A (9 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0A_0B (10 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0B_0C (11 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0C_0D (12 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0D_0E (13 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0E_0F (14 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0F_10 (15 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_10_11 (16 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_11_12 (17 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_12_13 (18 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_13_14 (19 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_14_15 (20 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_15_16 (21 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_16_17 (22 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_17_18 (23 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_18_19 (24 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_19_1A (25 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1A_1B (26 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1B_1C (27 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1C_1D (28 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1D_1E (29 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1E_1F (30 << M98925_DAI_VMON_SLOT_SHIFT) + +/* MAX98925_R023_DOUT_CFG_IMON */ +#define M98925_DAI_IMON_EN_MASK (1<<5) +#define M98925_DAI_IMON_EN_SHIFT 5 +#define M98925_DAI_IMON_EN_WIDTH 1 +#define M98925_DAI_IMON_SLOT_MASK (0x1F<<0) +#define M98925_DAI_IMON_SLOT_SHIFT 0 +#define M98925_DAI_IMON_SLOT_WIDTH 5 + +#define M98925_DAI_IMON_SLOT_00_01 (0 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_01_02 (1 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_02_03 (2 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_03_04 (3 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_04_05 (4 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_05_06 (5 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_06_07 (6 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_07_08 (7 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_08_09 (8 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_09_0A (9 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0A_0B (10 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0B_0C (11 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0C_0D (12 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0D_0E (13 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0E_0F (14 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0F_10 (15 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_10_11 (16 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_11_12 (17 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_12_13 (18 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_13_14 (19 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_14_15 (20 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_15_16 (21 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_16_17 (22 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_17_18 (23 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_18_19 (24 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_19_1A (25 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1A_1B (26 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1B_1C (27 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1C_1D (28 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1D_1E (29 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1E_1F (30 << M98925_DAI_IMON_SLOT_SHIFT) + +/* MAX98925_R024_DOUT_CFG_VBAT */ +#define M98925_DAI_VBAT_EN_MASK (1<<5) +#define M98925_DAI_VBAT_EN_SHIFT 5 +#define M98925_DAI_VBAT_EN_WIDTH 1 +#define M98925_DAI_VBAT_SLOT_MASK (0x1F<<0) +#define M98925_DAI_VBAT_SLOT_SHIFT 0 +#define M98925_DAI_VBAT_SLOT_WIDTH 5 + +/* MAX98925_R025_DOUT_CFG_VBST */ +#define M98925_DAI_VBST_EN_MASK (1<<5) +#define M98925_DAI_VBST_EN_SHIFT 5 +#define M98925_DAI_VBST_EN_WIDTH 1 +#define M98925_DAI_VBST_SLOT_MASK (0x1F<<0) +#define M98925_DAI_VBST_SLOT_SHIFT 0 +#define M98925_DAI_VBST_SLOT_WIDTH 5 + +/* MAX98925_R026_DOUT_CFG_FLAG */ +#define M98925_DAI_FLAG_EN_MASK (1<<5) +#define M98925_DAI_FLAG_EN_SHIFT 5 +#define M98925_DAI_FLAG_EN_WIDTH 1 +#define M98925_DAI_FLAG_SLOT_MASK (0x1F<<0) +#define M98925_DAI_FLAG_SLOT_SHIFT 0 +#define M98925_DAI_FLAG_SLOT_WIDTH 5 + +/* MAX98925_R027_DOUT_HIZ_CFG1 */ +#define M98925_DAI_SLOT_HIZ_CFG1_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG1_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG1_WIDTH 8 + +/* MAX98925_R028_DOUT_HIZ_CFG2 */ +#define M98925_DAI_SLOT_HIZ_CFG2_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG2_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG2_WIDTH 8 + +/* MAX98925_R029_DOUT_HIZ_CFG3 */ +#define M98925_DAI_SLOT_HIZ_CFG3_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG3_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG3_WIDTH 8 + +/* MAX98925_R02A_DOUT_HIZ_CFG4 */ +#define M98925_DAI_SLOT_HIZ_CFG4_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG4_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG4_WIDTH 8 + +/* MAX98925_R02B_DOUT_DRV_STRENGTH */ +#define M98925_DAI_OUT_DRIVE_MASK (0x03<<0) +#define M98925_DAI_OUT_DRIVE_SHIFT 0 +#define M98925_DAI_OUT_DRIVE_WIDTH 2 + +/* MAX98925_R02C_FILTERS */ +#define M98925_ADC_DITHER_EN_MASK (1<<7) +#define M98925_ADC_DITHER_EN_SHIFT 7 +#define M98925_ADC_DITHER_EN_WIDTH 1 +#define M98925_IV_DCB_EN_MASK (1<<6) +#define M98925_IV_DCB_EN_SHIFT 6 +#define M98925_IV_DCB_EN_WIDTH 1 +#define M98925_DAC_DITHER_EN_MASK (1<<4) +#define M98925_DAC_DITHER_EN_SHIFT 4 +#define M98925_DAC_DITHER_EN_WIDTH 1 +#define M98925_DAC_FILTER_MODE_MASK (1<<3) +#define M98925_DAC_FILTER_MODE_SHIFT 3 +#define M98925_DAC_FILTER_MODE_WIDTH 1 +#define M98925_DAC_HPF_MASK (0x07<<0) +#define M98925_DAC_HPF_SHIFT 0 +#define M98925_DAC_HPF_WIDTH 3 +#define M98925_DAC_HPF_DISABLE (0 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_DC_BLOCK (1 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_EN_100 (2 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_EN_200 (3 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_EN_400 (4 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_EN_800 (5 << M98925_DAC_HPF_SHIFT) + +/* MAX98925_R02D_GAIN */ +#define M98925_DAC_IN_SEL_MASK (0x03<<5) +#define M98925_DAC_IN_SEL_SHIFT 5 +#define M98925_DAC_IN_SEL_WIDTH 2 +#define M98925_SPK_GAIN_MASK (0x1F<<0) +#define M98925_SPK_GAIN_SHIFT 0 +#define M98925_SPK_GAIN_WIDTH 5 + +#define M98925_DAC_IN_SEL_LEFT_DAI (0 << M98925_DAC_IN_SEL_SHIFT) +#define M98925_DAC_IN_SEL_RIGHT_DAI (1 << M98925_DAC_IN_SEL_SHIFT) +#define M98925_DAC_IN_SEL_SUMMED_DAI (2 << M98925_DAC_IN_SEL_SHIFT) +#define M98925_DAC_IN_SEL_DIV2_SUMMED_DAI (3 << M98925_DAC_IN_SEL_SHIFT) + +/* MAX98925_R02E_GAIN_RAMPING */ +#define M98925_SPK_RMP_EN_MASK (1<<1) +#define M98925_SPK_RMP_EN_SHIFT 1 +#define M98925_SPK_RMP_EN_WIDTH 1 +#define M98925_SPK_ZCD_EN_MASK (1<<0) +#define M98925_SPK_ZCD_EN_SHIFT 0 +#define M98925_SPK_ZCD_EN_WIDTH 1 + +/* MAX98925_R02F_SPK_AMP */ +#define M98925_SPK_MODE_MASK (1<<0) +#define M98925_SPK_MODE_SHIFT 0 +#define M98925_SPK_MODE_WIDTH 1 + +/* MAX98925_R030_THRESHOLD */ +#define M98925_ALC_EN_MASK (1<<5) +#define M98925_ALC_EN_SHIFT 5 +#define M98925_ALC_EN_WIDTH 1 +#define M98925_ALC_TH_MASK (0x1F<<0) +#define M98925_ALC_TH_SHIFT 0 +#define M98925_ALC_TH_WIDTH 5 + +/* MAX98925_R031_ALC_ATTACK */ +#define M98925_ALC_ATK_STEP_MASK (0x0F<<4) +#define M98925_ALC_ATK_STEP_SHIFT 4 +#define M98925_ALC_ATK_STEP_WIDTH 4 +#define M98925_ALC_ATK_RATE_MASK (0x7<<0) +#define M98925_ALC_ATK_RATE_SHIFT 0 +#define M98925_ALC_ATK_RATE_WIDTH 3 + +/* MAX98925_R032_ALC_ATTEN_RLS */ +#define M98925_ALC_MAX_ATTEN_MASK (0x0F<<4) +#define M98925_ALC_MAX_ATTEN_SHIFT 4 +#define M98925_ALC_MAX_ATTEN_WIDTH 4 +#define M98925_ALC_RLS_RATE_MASK (0x7<<0) +#define M98925_ALC_RLS_RATE_SHIFT 0 +#define M98925_ALC_RLS_RATE_WIDTH 3 + +/* MAX98925_R033_ALC_HOLD_RLS */ +#define M98925_ALC_RLS_TGR_MASK (1<<0) +#define M98925_ALC_RLS_TGR_SHIFT 0 +#define M98925_ALC_RLS_TGR_WIDTH 1 + +/* MAX98925_R034_ALC_CONFIGURATION */ +#define M98925_ALC_MUTE_EN_MASK (1<<7) +#define M98925_ALC_MUTE_EN_SHIFT 7 +#define M98925_ALC_MUTE_EN_WIDTH 1 +#define M98925_ALC_MUTE_DLY_MASK (0x07<<4) +#define M98925_ALC_MUTE_DLY_SHIFT 4 +#define M98925_ALC_MUTE_DLY_WIDTH 3 +#define M98925_ALC_RLS_DBT_MASK (0x07<<0) +#define M98925_ALC_RLS_DBT_SHIFT 0 +#define M98925_ALC_RLS_DBT_WIDTH 3 + +/* MAX98925_R035_BOOST_CONVERTER */ +#define M98925_BST_SYNC_MASK (1<<7) +#define M98925_BST_SYNC_SHIFT 7 +#define M98925_BST_SYNC_WIDTH 1 +#define M98925_BST_PHASE_MASK (0x03<<4) +#define M98925_BST_PHASE_SHIFT 4 +#define M98925_BST_PHASE_WIDTH 2 +#define M98925_BST_SKIP_MODE_MASK (0x03<<0) +#define M98925_BST_SKIP_MODE_SHIFT 0 +#define M98925_BST_SKIP_MODE_WIDTH 2 + +/* MAX98925_R036_BLOCK_ENABLE */ +#define M98925_BST_EN_MASK (1<<7) +#define M98925_BST_EN_SHIFT 7 +#define M98925_BST_EN_WIDTH 1 +#define M98925_WATCH_EN_MASK (1<<6) +#define M98925_WATCH_EN_SHIFT 6 +#define M98925_WATCH_EN_WIDTH 1 +#define M98925_CLKMON_EN_MASK (1<<5) +#define M98925_CLKMON_EN_SHIFT 5 +#define M98925_CLKMON_EN_WIDTH 1 +#define M98925_SPK_EN_MASK (1<<4) +#define M98925_SPK_EN_SHIFT 4 +#define M98925_SPK_EN_WIDTH 1 +#define M98925_ADC_VBST_EN_MASK (1<<3) +#define M98925_ADC_VBST_EN_SHIFT 3 +#define M98925_ADC_VBST_EN_WIDTH 1 +#define M98925_ADC_VBAT_EN_MASK (1<<2) +#define M98925_ADC_VBAT_EN_SHIFT 2 +#define M98925_ADC_VBAT_EN_WIDTH 1 +#define M98925_ADC_IMON_EN_MASK (1<<1) +#define M98925_ADC_IMON_EN_SHIFT 1 +#define M98925_ADC_IMON_EN_WIDTH 1 +#define M98925_ADC_VMON_EN_MASK (1<<0) +#define M98925_ADC_VMON_EN_SHIFT 0 +#define M98925_ADC_VMON_EN_WIDTH 1 + +/* MAX98925_R037_CONFIGURATION */ +#define M98925_BST_VOUT_MASK (0x0F<<4) +#define M98925_BST_VOUT_SHIFT 4 +#define M98925_BST_VOUT_WIDTH 4 +#define M98925_THERMWARN_LEVEL_MASK (0x03<<2) +#define M98925_THERMWARN_LEVEL_SHIFT 2 +#define M98925_THERMWARN_LEVEL_WIDTH 2 +#define M98925_WATCH_TIME_MASK (0x03<<0) +#define M98925_WATCH_TIME_SHIFT 0 +#define M98925_WATCH_TIME_WIDTH 2 + +/* MAX98925_R038_GLOBAL_ENABLE */ +#define M98925_EN_MASK (1<<7) +#define M98925_EN_SHIFT 7 +#define M98925_EN_WIDTH 1 + +/* MAX98925_R03A_BOOST_LIMITER */ +#define M98925_BST_ILIM_MASK (0x1F<<3) +#define M98925_BST_ILIM_SHIFT 3 +#define M98925_BST_ILIM_WIDTH 5 + +/* MAX98925_R0FF_VERSION */ +#define M98925_REV_ID_MASK (0xFF<<0) +#define M98925_REV_ID_SHIFT 0 +#define M98925_REV_ID_WIDTH 8 + +struct max98925_cdata { + unsigned int rate; + unsigned int fmt; +}; + +struct max98925_priv { + struct regmap *regmap_l; + struct regmap *regmap_r; + struct snd_soc_codec *codec; + struct max98925_pdata *pdata; + unsigned int spk_gain; + unsigned int sysclk; + unsigned int iv_status; + struct max98925_cdata dai[1]; +}; +#endif -- 1.9.3

3 4

[alsa-devel] [PATCH] ASoC: Add max98925 codec driver
by Anish Kumar 24 Feb '15

24 Feb '15

Signed-off-by: Anish Kumar <yesanishhere(a)gmail.com> --- sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98925.c | 843 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max98925.h | 845 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 1699 insertions(+) create mode 100644 sound/soc/codecs/max98925.c create mode 100644 sound/soc/codecs/max98925.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 45b7256..7d9002c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -47,6 +47,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C + select SND_SOC_MAX98925 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C @@ -285,6 +286,14 @@ config SND_SOC_MAX9850 config SND_SOC_OMAP_HDMI_CODEC tristate +config SND_SOC_MAX98925 + tristate "Support MAX98925 audio codec" + depends on I2C + help + MAX98925 is a high-efficiency mono + Class DG audio amplifier. Use this + option to enable the codec. + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6a3b3c3..d439d47 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -38,6 +38,7 @@ snd-soc-max98088-objs := max98088.o snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o +snd-soc-max98925-objs := max98925.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-omap-hdmi-codec-objs := omap-hdmi.o @@ -233,6 +234,7 @@ obj-$(CONFIG_SND_SOC_WM8994) += snd-soc-wm8994.o obj-$(CONFIG_SND_SOC_WM8995) += snd-soc-wm8995.o obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o obj-$(CONFIG_SND_SOC_WM9090) += snd-soc-wm9090.o +obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c new file mode 100644 index 0000000..35af23b --- /dev/null +++ b/sound/soc/codecs/max98925.c @@ -0,0 +1,843 @@ +/* + * max98925.c -- ALSA SoC Stereo MAX98925 driver + * Copyright 2013-15 Maxim Integrated Products + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/cdev.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max98925.h" + +static const char *const dai_text[] = { + "left", "right", "leftright", "leftrightdiv2", +}; + +static const char *const hpf_text[] = { + "disable", "dc_block", "100Hz", "200Hz", "400Hz", "800Hz", +}; + +static struct reg_default max98925_reg[] = { + { 0x00, 0x00 }, /* Battery Voltage Data */ + { 0x01, 0x00 }, /* Boost Voltage Data */ + { 0x02, 0x00 }, /* Live Status0 */ + { 0x03, 0x00 }, /* Live Status1 */ + { 0x04, 0x00 }, /* Live Status2 */ + { 0x05, 0x00 }, /* State0 */ + { 0x06, 0x00 }, /* State1 */ + { 0x07, 0x00 }, /* State2 */ + { 0x08, 0x00 }, /* Flag0 */ + { 0x09, 0x00 }, /* Flag1 */ + { 0x0A, 0x00 }, /* Flag2 */ + { 0x0B, 0x00 }, /* IRQ Enable0 */ + { 0x0C, 0x00 }, /* IRQ Enable1 */ + { 0x0D, 0x00 }, /* IRQ Enable2 */ + { 0x0E, 0x00 }, /* IRQ Clear0 */ + { 0x0F, 0x00 }, /* IRQ Clear1 */ + { 0x10, 0x00 }, /* IRQ Clear2 */ + { 0x11, 0xC0 }, /* Map0 */ + { 0x12, 0x00 }, /* Map1 */ + { 0x13, 0x00 }, /* Map2 */ + { 0x14, 0xF0 }, /* Map3 */ + { 0x15, 0x00 }, /* Map4 */ + { 0x16, 0xAB }, /* Map5 */ + { 0x17, 0x89 }, /* Map6 */ + { 0x18, 0x00 }, /* Map7 */ + { 0x19, 0x00 }, /* Map8 */ + { 0x1A, 0x06 }, /* DAI Clock Mode 1 */ + { 0x1B, 0xC0 }, /* DAI Clock Mode 2 */ + { 0x1C, 0x00 }, /* DAI Clock Divider Denominator MSBs */ + { 0x1D, 0x00 }, /* DAI Clock Divider Denominator LSBs */ + { 0x1E, 0xF0 }, /* DAI Clock Divider Numerator MSBs */ + { 0x1F, 0x00 }, /* DAI Clock Divider Numerator LSBs */ + { 0x20, 0x50 }, /* Format */ + { 0x21, 0x00 }, /* TDM Slot Select */ + { 0x22, 0x00 }, /* DOUT Configuration VMON */ + { 0x23, 0x00 }, /* DOUT Configuration IMON */ + { 0x24, 0x00 }, /* DOUT Configuration VBAT */ + { 0x25, 0x00 }, /* DOUT Configuration VBST */ + { 0x26, 0x00 }, /* DOUT Configuration FLAG */ + { 0x27, 0xFF }, /* DOUT HiZ Configuration 1 */ + { 0x28, 0xFF }, /* DOUT HiZ Configuration 2 */ + { 0x29, 0xFF }, /* DOUT HiZ Configuration 3 */ + { 0x2A, 0xFF }, /* DOUT HiZ Configuration 4 */ + { 0x2B, 0x02 }, /* DOUT Drive Strength */ + { 0x2C, 0x90 }, /* Filters */ + { 0x2D, 0x00 }, /* Gain */ + { 0x2E, 0x02 }, /* Gain Ramping */ + { 0x2F, 0x00 }, /* Speaker Amplifier */ + { 0x30, 0x0A }, /* Threshold */ + { 0x31, 0x00 }, /* ALC Attack */ + { 0x32, 0x80 }, /* ALC Atten and Release */ + { 0x33, 0x00 }, /* ALC Infinite Hold Release */ + { 0x34, 0x92 }, /* ALC Configuration */ + { 0x35, 0x01 }, /* Boost Converter */ + { 0x36, 0x00 }, /* Block Enable */ + { 0x37, 0x00 }, /* Configuration */ + { 0x38, 0x00 }, /* Global Enable */ + { 0x3A, 0x00 }, /* Boost Limiter */ + { 0xFF, 0x50 }, /* Revision ID */ +}; + +static int max98925_reg_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, unsigned int reg, + unsigned int mask, unsigned int shift) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + unsigned int sel = ucontrol->value.integer.value[0]; + + regmap_update_bits(max98925->regmap, reg, mask, sel<<shift); + dev_dbg(codec->dev, "%s: register 0x%02X, value 0x%02X\n", + __func__, reg, sel); + return 0; +} + +static const struct soc_enum max98925_enum[] = { + SOC_ENUM_SINGLE(MAX98925_R02D_GAIN, 5, 4, dai_text), + SOC_ENUM_SINGLE(MAX98925_R02C_FILTERS, 0, 6, hpf_text), +}; + +static const struct snd_kcontrol_new max98925_dai_sel = + SOC_DAPM_ENUM("Route", max98925_enum[0]); + +static const struct snd_kcontrol_new max98925_hpf_sel = + SOC_DAPM_ENUM("Route", max98925_enum[1]); + +static int max98925_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(max98925->regmap, + MAX98925_R036_BLOCK_ENABLE, + M98925_BST_EN_MASK | + M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK, + M98925_BST_EN_MASK | + M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK); + regmap_write(max98925->regmap, + MAX98925_R038_GLOBAL_ENABLE, M98925_EN_MASK); + return 0; +} + +static const struct snd_soc_dapm_widget max98925_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("DAI_IN_MUX MUX", SND_SOC_NOPM, 0, 0, + &max98925_dai_sel), + SND_SOC_DAPM_MUX("R02C_FILTER MUX", SND_SOC_NOPM, 0, 0, + &max98925_hpf_sel), + SND_SOC_DAPM_DAC_E("amp_enable", NULL, + MAX98925_R036_BLOCK_ENABLE, + M98925_SPK_EN_SHIFT, 0, + max98925_dac_event, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_OUTPUT("BE_OUT"), +}; + +static const struct snd_soc_dapm_route max98925_audio_map[] = { + {"DAI_IN_MUX MUX", "left", "DAI_OUT"}, + {"DAI_IN_MUX MUX", "right", "DAI_OUT"}, + {"DAI_IN_MUX MUX", "leftright", "DAI_OUT"}, + {"DAI_IN_MUX MUX", "leftrightdiv2", "DAI_OUT"}, + {"R02C_FILTER MUX", "disable", "DAI_IN_MUX MUX"}, + {"R02C_FILTER MUX", "dc_block", "DAI_IN_MUX MUX"}, + {"R02C_FILTER MUX", "100Hz", "DAI_IN_MUX MUX"}, + {"R02C_FILTER MUX", "200Hz", "DAI_IN_MUX MUX"}, + {"R02C_FILTER MUX", "400Hz", "DAI_IN_MUX MUX"}, + {"R02C_FILTER MUX", "800Hz", "DAI_IN_MUX MUX"}, + {"amp_enable", NULL, "R02C_FILTER MUX"}, + {"BE_OUT", NULL, "amp_enable"}, +}; + +static bool max98925_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98925_R000_VBAT_DATA: + case MAX98925_R001_VBST_DATA: + case MAX98925_R002_LIVE_STATUS0: + case MAX98925_R003_LIVE_STATUS1: + case MAX98925_R004_LIVE_STATUS2: + case MAX98925_R005_STATE0: + case MAX98925_R006_STATE1: + case MAX98925_R007_STATE2: + case MAX98925_R008_FLAG0: + case MAX98925_R009_FLAG1: + case MAX98925_R00A_FLAG2: + case MAX98925_R0FF_VERSION: + return true; + default: + return false; + } +} + +static bool max98925_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98925_R00E_IRQ_CLEAR0: + case MAX98925_R00F_IRQ_CLEAR1: + case MAX98925_R010_IRQ_CLEAR2: + case MAX98925_R033_ALC_HOLD_RLS: + return false; + default: + return true; + } +} + +DECLARE_TLV_DB_SCALE(max98925_spk_tlv, -600, 100, 0); + +static int max98925_spk_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = max98925->spk_gain; + dev_dbg(codec->dev, "%s: spk_gain setting returned %d\n", __func__, + (int) ucontrol->value.integer.value[0]); + return 0; +} + +static int max98925_spk_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + unsigned int sel = ucontrol->value.integer.value[0]; + + if (sel < ((1 << M98925_SPK_GAIN_WIDTH) - 1)) { + regmap_update_bits(max98925->regmap, MAX98925_R02D_GAIN, + M98925_SPK_GAIN_MASK, sel << M98925_SPK_GAIN_SHIFT); + max98925->spk_gain = sel; + + dev_dbg(codec->dev, "%s: spk_gain set to %d\n", __func__, sel); + } else { + dev_dbg(codec->dev, "%s: valid speaker gain settings: %d to %d\n", + __func__, 0, + ((1 << M98925_SPK_GAIN_WIDTH) - 1)); + } + return 0; +} + +static int max98925_reg_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, unsigned int reg, + unsigned int mask, unsigned int shift) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + int data; + + regmap_read(max98925->regmap, reg, &data); + ucontrol->value.integer.value[0] = + (data & mask) >> shift; + return 0; +} + +static int max98925_spk_ramp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_get(kcontrol, ucontrol, MAX98925_R02E_GAIN_RAMPING, + M98925_SPK_RMP_EN_MASK, M98925_SPK_RMP_EN_SHIFT); +} + +static int max98925_spk_ramp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_put(kcontrol, ucontrol, MAX98925_R02E_GAIN_RAMPING, + M98925_SPK_RMP_EN_MASK, M98925_SPK_RMP_EN_SHIFT); +} + +static int max98925_spk_zcd_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_get(kcontrol, ucontrol, MAX98925_R02E_GAIN_RAMPING, + M98925_SPK_ZCD_EN_MASK, M98925_SPK_ZCD_EN_SHIFT); +} + +static int max98925_spk_zcd_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_put(kcontrol, ucontrol, MAX98925_R02E_GAIN_RAMPING, + M98925_SPK_ZCD_EN_MASK, M98925_SPK_ZCD_EN_SHIFT); +} + +static int max98925_alc_en_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_get(kcontrol, ucontrol, MAX98925_R030_THRESHOLD, + M98925_ALC_EN_MASK, M98925_ALC_EN_SHIFT); +} + +static int max98925_alc_en_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_put(kcontrol, ucontrol, MAX98925_R030_THRESHOLD, + M98925_ALC_EN_MASK, M98925_ALC_EN_SHIFT); +} + +static int max98925_alc_threshold_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_get(kcontrol, ucontrol, MAX98925_R030_THRESHOLD, + M98925_ALC_TH_MASK, M98925_ALC_TH_SHIFT); +} + +static int max98925_alc_threshold_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_put(kcontrol, ucontrol, MAX98925_R030_THRESHOLD, + M98925_ALC_TH_MASK, M98925_ALC_TH_SHIFT); +} + +static const char * const max98925_boost_voltage_text[] = { + "8.5V", "8.25V", "8.0V", "7.75V", "7.5V", "7.25V", "7.0V", "6.75V", + "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V" +}; + +static const struct soc_enum max98925_boost_voltage_enum = + SOC_ENUM_SINGLE(MAX98925_R037_CONFIGURATION, M98925_BST_VOUT_SHIFT, 15, + max98925_boost_voltage_text); + +static int max98925_boost_voltage_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_get(kcontrol, ucontrol, MAX98925_R037_CONFIGURATION, + M98925_BST_VOUT_MASK, M98925_BST_VOUT_SHIFT); +} + +static int max98925_boost_voltage_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_put(kcontrol, ucontrol, MAX98925_R037_CONFIGURATION, + M98925_BST_VOUT_MASK, M98925_BST_VOUT_SHIFT); +} + +static const struct snd_kcontrol_new max98925_snd_controls[] = { + + SOC_SINGLE_EXT_TLV("Speaker Gain", MAX98925_R02D_GAIN, + M98925_SPK_GAIN_SHIFT, (1<<M98925_SPK_GAIN_WIDTH)-1, 0, + max98925_spk_gain_get, max98925_spk_gain_put, max98925_spk_tlv), + + SOC_SINGLE_EXT("Speaker Ramp", 0, 0, 1, 0, + max98925_spk_ramp_get, max98925_spk_ramp_put), + + SOC_SINGLE_EXT("Speaker ZCD", 0, 0, 1, 0, + max98925_spk_zcd_get, max98925_spk_zcd_put), + + SOC_SINGLE_EXT("ALC Enable", 0, 0, 1, 0, + max98925_alc_en_get, max98925_alc_en_put), + + SOC_SINGLE_EXT("ALC Threshold", 0, 0, (1<<M98925_ALC_TH_WIDTH)-1, 0, + max98925_alc_threshold_get, max98925_alc_threshold_put), + + SOC_ENUM_EXT("Boost Output Voltage", max98925_boost_voltage_enum, + max98925_boost_voltage_get, max98925_boost_voltage_put), +}; + +/* codec sample rate and n/m dividers parameter table */ +static const struct { + int rate; + int sr; + int divisors[3][2]; +} rate_table[] = { + { + .rate = 8000, + .sr = 0, + .divisors = { {1, 375}, {5, 1764}, {1, 384} } + }, + { + .rate = 11025, + .sr = 1, + .divisors = { {147, 40000}, {1, 256}, {147, 40960} } + }, + { + .rate = 12000, + .sr = 2, + .divisors = { {1, 250}, {5, 1176}, {1, 256} } + }, + { + .rate = 16000, + .sr = 3, + .divisors = { {2, 375}, {5, 882}, {1, 192} } + }, + { + .rate = 22050, + .sr = 4, + .divisors = { {147, 20000}, {1, 128}, {147, 20480} } + }, + { + .rate = 24000, + .sr = 5, + .divisors = { {1, 125}, {5, 588}, {1, 128} } + }, + { + .rate = 32000, + .sr = 6, + .divisors = { {4, 375}, {5, 441}, {1, 96} } + }, + { + .rate = 44100, + .sr = 7, + .divisors = { {147, 10000}, {1, 64}, {147, 10240} } + }, + { + .rate = 48000, + .sr = 8, + .divisors = { {2, 125}, {5, 294}, {1, 64} } + }, +}; + +static inline int max98925_rate_value(int rate, + int clock, int *value, int *n, int *m) +{ + int ret = -EINVAL; + int i; + + for (i = 0; i < ARRAY_SIZE(rate_table); i++) { + if (rate_table[i].rate >= rate) { + *value = rate_table[i].sr; + *n = rate_table[i].divisors[clock][0]; + *m = rate_table[i].divisors[clock][1]; + ret = 0; + break; + } + } + pr_debug("%s: sample rate is %d, returning %d\n", + __func__, rate_table[i].rate, *value); + return ret; +} + +static void max98925_set_sense_data(struct max98925_priv *max98925) +{ + /* + * 1. set VMON slots + */ + regmap_update_bits(max98925->regmap, + MAX98925_R022_DOUT_CFG_VMON, + M98925_DAI_VMON_EN_MASK, M98925_DAI_VMON_EN_MASK); + regmap_update_bits(max98925->regmap, + MAX98925_R022_DOUT_CFG_VMON, + M98925_DAI_VMON_SLOT_MASK, max98925->v_slot); + /* + * 2. set IMON slots + */ + regmap_update_bits(max98925->regmap, + MAX98925_R023_DOUT_CFG_IMON, + M98925_DAI_IMON_EN_MASK, M98925_DAI_IMON_EN_MASK); + regmap_update_bits(max98925->regmap, + MAX98925_R023_DOUT_CFG_IMON, + M98925_DAI_IMON_SLOT_MASK, max98925->i_slot); +} + +static void max98925_set_slave(struct max98925_priv *max98925) +{ + /* + * 1. set DAI to slave mode + */ + regmap_update_bits(max98925->regmap, + MAX98925_R01B_DAI_CLK_MODE2, + M98925_DAI_MAS_MASK, 0); + max98925_set_sense_data(max98925); +} + +static void max98925_set_master(struct max98925_priv *max98925) +{ + /* + * set left channel DAI to master mode, right channel always slave + */ + regmap_update_bits(max98925->regmap, + MAX98925_R01B_DAI_CLK_MODE2, + M98925_DAI_MAS_MASK, M98925_DAI_MAS_MASK); +} + +static int max98925_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + unsigned int invert = 0; + + dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + max98925_set_slave(max98925); + break; + case SND_SOC_DAIFMT_CBM_CFM: + max98925_set_master(max98925); + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + invert = M98925_DAI_WCI_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + invert = M98925_DAI_BCI_MASK; + break; + case SND_SOC_DAIFMT_IB_IF: + invert = M98925_DAI_BCI_MASK | M98925_DAI_WCI_MASK; + break; + default: + dev_err(codec->dev, "DAI invert mode unsupported"); + return -EINVAL; + } + + regmap_update_bits(max98925->regmap, MAX98925_R020_FORMAT, + M98925_DAI_BCI_MASK | M98925_DAI_BCI_MASK, invert); + return 0; +} + +static int max98925_set_clock(struct max98925_priv *max98925, + struct snd_pcm_hw_params *params) +{ + unsigned int dai_sr = 0; + unsigned int clock; + unsigned int mdll; + unsigned int n; + unsigned int m; + int rate = params_rate(params); + + /* BCLK/LRCLK ratio calculation */ + int blr_clk_ratio = params_channels(params) * max98925->ch_size; + switch (blr_clk_ratio) { + case 32: + regmap_update_bits(max98925->regmap, + MAX98925_R01B_DAI_CLK_MODE2, + M98925_DAI_BSEL_MASK, M98925_DAI_BSEL_32); + break; + case 48: + regmap_update_bits(max98925->regmap, + MAX98925_R01B_DAI_CLK_MODE2, + M98925_DAI_BSEL_MASK, M98925_DAI_BSEL_48); + break; + case 64: + regmap_update_bits(max98925->regmap, + MAX98925_R01B_DAI_CLK_MODE2, + M98925_DAI_BSEL_MASK, M98925_DAI_BSEL_64); + break; + default: + return -EINVAL; + } + + switch (max98925->sysclk) { + case 6000000: + clock = 0; + mdll = M98925_MDLL_MULT_MCLKx16; + break; + case 11289600: + clock = 1; + mdll = M98925_MDLL_MULT_MCLKx8; + break; + case 12000000: + clock = 0; + mdll = M98925_MDLL_MULT_MCLKx8; + break; + case 12288000: + clock = 2; + mdll = M98925_MDLL_MULT_MCLKx8; + break; + default: + dev_info(max98925->codec->dev, "unsupported sysclk %d\n", + max98925->sysclk); + return -EINVAL; + } + + if (max98925_rate_value(rate, clock, &dai_sr, &n, &m)) + return -EINVAL; + + /* + * 1. set DAI_SR to correct LRCLK frequency + */ + regmap_update_bits(max98925->regmap, + MAX98925_R01B_DAI_CLK_MODE2, + M98925_DAI_SR_MASK, dai_sr << M98925_DAI_SR_SHIFT); + /* + * 2. set DAI m divider + */ + regmap_write(max98925->regmap, + MAX98925_R01C_DAI_CLK_DIV_M_MSBS, m >> 8); + regmap_write(max98925->regmap, + MAX98925_R01D_DAI_CLK_DIV_M_LSBS, m & 0xFF); + /* + * 3. set DAI n divider + */ + regmap_write(max98925->regmap, + MAX98925_R01E_DAI_CLK_DIV_N_MSBS, n >> 8); + regmap_write(max98925->regmap, + MAX98925_R01F_DAI_CLK_DIV_N_LSBS, n & 0xFF); + /* + * 4. set MDLL + */ + regmap_update_bits(max98925->regmap, + MAX98925_R01A_DAI_CLK_MODE1, + M98925_MDLL_MULT_MASK, mdll << M98925_MDLL_MULT_SHIFT); + return 0; +} + +static int max98925_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(max98925->regmap, + MAX98925_R020_FORMAT, + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_16); + max98925->ch_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(max98925->regmap, + MAX98925_R020_FORMAT, + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_32); + max98925->ch_size = 32; + break; + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(max98925->regmap, + MAX98925_R020_FORMAT, + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_32); + max98925->ch_size = 32; + break; + default: + pr_err("%s: format unsupported %d", + __func__, params_format(params)); + return -EINVAL; + } + dev_dbg(codec->dev, "%s: format supported %d", + __func__, params_format(params)); + return max98925_set_clock(max98925, params); +} + +static int max98925_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case 0: + /* + * 1. use MCLK for Left channel, right channel always BCLK + */ + regmap_update_bits(max98925->regmap, + MAX98925_R01A_DAI_CLK_MODE1, + M98925_DAI_CLK_SOURCE_MASK, 0); + break; + case 1: + /* + * configure dai clock source to BCLK instead of MCLK(default) + */ + regmap_update_bits(max98925->regmap, + MAX98925_R01A_DAI_CLK_MODE1, + M98925_DAI_CLK_SOURCE_MASK, + M98925_DAI_CLK_SOURCE_MASK); + break; + default: + return -EINVAL; + } + max98925->sysclk = freq; + return 0; +} + +static int max98925_dai_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct max98925_priv *max98925 = + snd_soc_codec_get_drvdata(codec_dai->codec); + struct snd_soc_codec *codec = codec_dai->codec; + + dev_dbg(codec->dev, "%s: mute %d\n", __func__, mute); + if (mute) { + regmap_update_bits(max98925->regmap, MAX98925_R02D_GAIN, + M98925_SPK_GAIN_MASK, 0x00); + + regmap_update_bits(max98925->regmap, + MAX98925_R038_GLOBAL_ENABLE, + M98925_EN_MASK, 0x0); + } + return 0; +} + +#define MAX98925_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops max98925_dai_ops = { + .set_sysclk = max98925_dai_set_sysclk, + .set_fmt = max98925_dai_set_fmt, + .hw_params = max98925_dai_hw_params, + .digital_mute = max98925_dai_digital_mute, +}; + +static struct snd_soc_dai_driver max98925_dai[] = { + { + .name = "max98925-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98925_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98925_FORMATS, + }, + .ops = &max98925_dai_ops, + } +}; + +static int max98925_probe(struct snd_soc_codec *codec) +{ + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + int reg = 0; + + dev_info(codec->dev, "build number %s\n", MAX98925_REVISION); + + max98925->codec = codec; + codec->control_data = max98925->regmap; + ret = regmap_read(max98925->regmap, + MAX98925_R0FF_VERSION, &reg); + if ((ret < 0) || + ((reg != MAX98925_VERSION) && + (reg != MAX98925_VERSION1))) { + dev_err(codec->dev, + "device initialization error (%d 0x%02X)\n", + ret, reg); + goto err_access; + } + regmap_write(max98925->regmap, MAX98925_R038_GLOBAL_ENABLE, 0x00); + /* It's not the default but we need to set DAI_DLY */ + regmap_write(max98925->regmap, + MAX98925_R020_FORMAT, M98925_DAI_DLY_MASK); + regmap_write(max98925->regmap, MAX98925_R021_TDM_SLOT_SELECT, 0xC8); + regmap_write(max98925->regmap, MAX98925_R027_DOUT_HIZ_CFG1, 0xFF); + regmap_write(max98925->regmap, MAX98925_R028_DOUT_HIZ_CFG2, 0xFF); + regmap_write(max98925->regmap, MAX98925_R029_DOUT_HIZ_CFG3, 0xFF); + regmap_write(max98925->regmap, MAX98925_R02A_DOUT_HIZ_CFG4, 0xF0); + regmap_write(max98925->regmap, MAX98925_R02C_FILTERS, 0xD8); + regmap_write(max98925->regmap, MAX98925_R034_ALC_CONFIGURATION, 0xF8); + regmap_write(max98925->regmap, MAX98925_R037_CONFIGURATION, 0xF0); + + /* Disable ALC muting */ + regmap_write(max98925->regmap, MAX98925_R03A_BOOST_LIMITER, 0xF8); + dev_info(codec->dev, "device version 0x%02X\n", reg); +err_access: + return ret; +} + +static struct snd_soc_codec_driver soc_codec_dev_max98925 = { + .probe = max98925_probe, + .controls = max98925_snd_controls, + .num_controls = ARRAY_SIZE(max98925_snd_controls), + .dapm_routes = max98925_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98925_audio_map), + .dapm_widgets = max98925_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98925_dapm_widgets), +}; + +static struct regmap_config max98925_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98925_R0FF_VERSION, + .reg_defaults = max98925_reg, + .num_reg_defaults = ARRAY_SIZE(max98925_reg), + .volatile_reg = max98925_volatile_register, + .readable_reg = max98925_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98925_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + int ret = 0; + u32 value; + struct max98925_priv *max98925; + + dev_info(&i2c->dev, "%s: enter, device '%s'\n", __func__, id->name); + max98925 = devm_kzalloc(&i2c->dev, + sizeof(*max98925), GFP_KERNEL); + if (!max98925) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98925); + max98925->regmap = devm_regmap_init_i2c(i2c, &max98925_regmap); + if (IS_ERR(max98925->regmap)) { + ret = PTR_ERR(max98925->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + goto err_out; + } + + if (!of_property_read_u32(i2c->dev.of_node, "vmon-slot-no", &value)) { + if (value > M98925_DAI_VMON_SLOT_1E_1F) { + dev_err(&i2c->dev, "vmon slot number is wrong:\n"); + return -EINVAL; + } + max98925->v_slot = value; + } + if (!of_property_read_u32(i2c->dev.of_node, "imon-slot-no", &value)) { + if (value > M98925_DAI_IMON_SLOT_1E_1F) { + dev_err(&i2c->dev, "imon slot number is wrong:\n"); + return -EINVAL; + } + max98925->i_slot = value; + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98925, + max98925_dai, ARRAY_SIZE(max98925_dai)); + if (ret < 0) + dev_err(&i2c->dev, + "Failed to register codec: %d\n", ret); +err_out: + return ret; +} + +static int max98925_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id max98925_i2c_id[] = { + { "max98925", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max98925_i2c_id); + +static struct i2c_driver max98925_i2c_driver = { + .driver = { + .name = "max98925", + .owner = THIS_MODULE, + .pm = NULL, + }, + .probe = max98925_i2c_probe, + .remove = max98925_i2c_remove, + .id_table = max98925_i2c_id, +}; + +module_i2c_driver(max98925_i2c_driver); + +MODULE_DESCRIPTION("ALSA SoC MAX98925 driver"); +MODULE_AUTHOR("Ralph Birt <rdbirt(a)gmail.com>, Anish kumar <anish.kumar(a)maximintegrated.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98925.h b/sound/soc/codecs/max98925.h new file mode 100644 index 0000000..1842c15 --- /dev/null +++ b/sound/soc/codecs/max98925.h @@ -0,0 +1,845 @@ +/* + * max98925.h -- MAX98925 ALSA SoC Audio driver + * + * Copyright 2013-2015 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX98925_H +#define _MAX98925_H + +/* Maximum number of MAX98925 devices in the system this driver can support */ +#define MAX_NUM_MAX98925 2 +#define MAX98925_VERSION 0x51 +#define MAX98925_VERSION1 0x80 + +#define MAX98925_REVISION "0.00.0333" + +/* + * MAX98925 Register Definitions + */ +#define MAX98925_R000_VBAT_DATA 0x00 +#define MAX98925_R001_VBST_DATA 0x01 +#define MAX98925_R002_LIVE_STATUS0 0x02 +#define MAX98925_R003_LIVE_STATUS1 0x03 +#define MAX98925_R004_LIVE_STATUS2 0x04 +#define MAX98925_R005_STATE0 0x05 +#define MAX98925_R006_STATE1 0x06 +#define MAX98925_R007_STATE2 0x07 +#define MAX98925_R008_FLAG0 0x08 +#define MAX98925_R009_FLAG1 0x09 +#define MAX98925_R00A_FLAG2 0x0A +#define MAX98925_R00B_IRQ_ENABLE0 0x0B +#define MAX98925_R00C_IRQ_ENABLE1 0x0C +#define MAX98925_R00D_IRQ_ENABLE2 0x0D +#define MAX98925_R00E_IRQ_CLEAR0 0x0E +#define MAX98925_R00F_IRQ_CLEAR1 0x0F +#define MAX98925_R010_IRQ_CLEAR2 0x10 +#define MAX98925_R011_MAP0 0x11 +#define MAX98925_R012_MAP1 0x12 +#define MAX98925_R013_MAP2 0x13 +#define MAX98925_R014_MAP3 0x14 +#define MAX98925_R015_MAP4 0x15 +#define MAX98925_R016_MAP5 0x16 +#define MAX98925_R017_MAP6 0x17 +#define MAX98925_R018_MAP7 0x18 +#define MAX98925_R019_MAP8 0x19 +#define MAX98925_R01A_DAI_CLK_MODE1 0x1A +#define MAX98925_R01B_DAI_CLK_MODE2 0x1B +#define MAX98925_R01C_DAI_CLK_DIV_M_MSBS 0x1C +#define MAX98925_R01D_DAI_CLK_DIV_M_LSBS 0x1D +#define MAX98925_R01E_DAI_CLK_DIV_N_MSBS 0x1E +#define MAX98925_R01F_DAI_CLK_DIV_N_LSBS 0x1F +#define MAX98925_R020_FORMAT 0x20 +#define MAX98925_R021_TDM_SLOT_SELECT 0x21 +#define MAX98925_R022_DOUT_CFG_VMON 0x22 +#define MAX98925_R023_DOUT_CFG_IMON 0x23 +#define MAX98925_R024_DOUT_CFG_VBAT 0x24 +#define MAX98925_R025_DOUT_CFG_VBST 0x25 +#define MAX98925_R026_DOUT_CFG_FLAG 0x26 +#define MAX98925_R027_DOUT_HIZ_CFG1 0x27 +#define MAX98925_R028_DOUT_HIZ_CFG2 0x28 +#define MAX98925_R029_DOUT_HIZ_CFG3 0x29 +#define MAX98925_R02A_DOUT_HIZ_CFG4 0x2A +#define MAX98925_R02B_DOUT_DRV_STRENGTH 0x2B +#define MAX98925_R02C_FILTERS 0x2C +#define MAX98925_R02D_GAIN 0x2D +#define MAX98925_R02E_GAIN_RAMPING 0x2E +#define MAX98925_R02F_SPK_AMP 0x2F +#define MAX98925_R030_THRESHOLD 0x30 +#define MAX98925_R031_ALC_ATTACK 0x31 +#define MAX98925_R032_ALC_ATTEN_RLS 0x32 +#define MAX98925_R033_ALC_HOLD_RLS 0x33 +#define MAX98925_R034_ALC_CONFIGURATION 0x34 +#define MAX98925_R035_BOOST_CONVERTER 0x35 +#define MAX98925_R036_BLOCK_ENABLE 0x36 +#define MAX98925_R037_CONFIGURATION 0x37 +#define MAX98925_R038_GLOBAL_ENABLE 0x38 +#define MAX98925_R03A_BOOST_LIMITER 0x3A +#define MAX98925_R0FF_VERSION 0xFF + +#define MAX98925_REG_CNT (MAX98925_R03A_BOOST_LIMITER+1) + +/* MAX98925 Register Bit Fields */ + +/* MAX98925_R002_LIVE_STATUS0 */ +#define M98925_THERMWARN_STATUS_MASK (1<<3) +#define M98925_THERMWARN_STATUS_SHIFT 3 +#define M98925_THERMWARN_STATUS_WIDTH 1 +#define M98925_THERMSHDN_STATUS_MASK (1<<1) +#define M98925_THERMSHDN_STATUS_SHIFT 1 +#define M98925_THERMSHDN_STATUS_WIDTH 1 + +/* MAX98925_R003_LIVE_STATUS1 */ +#define M98925_SPKCURNT_STATUS_MASK (1<<5) +#define M98925_SPKCURNT_STATUS_SHIFT 5 +#define M98925_SPKCURNT_STATUS_WIDTH 1 +#define M98925_WATCHFAIL_STATUS_MASK (1<<4) +#define M98925_WATCHFAIL_STATUS_SHIFT 4 +#define M98925_WATCHFAIL_STATUS_WIDTH 1 +#define M98925_ALCINFH_STATUS_MASK (1<<3) +#define M98925_ALCINFH_STATUS_SHIFT 3 +#define M98925_ALCINFH_STATUS_WIDTH 1 +#define M98925_ALCACT_STATUS_MASK (1<<2) +#define M98925_ALCACT_STATUS_SHIFT 2 +#define M98925_ALCACT_STATUS_WIDTH 1 +#define M98925_ALCMUT_STATUS_MASK (1<<1) +#define M98925_ALCMUT_STATUS_SHIFT 1 +#define M98925_ALCMUT_STATUS_WIDTH 1 +#define M98925_ACLP_STATUS_MASK (1<<0) +#define M98925_ACLP_STATUS_SHIFT 0 +#define M98925_ACLP_STATUS_WIDTH 1 + +/* MAX98925_R004_LIVE_STATUS2 */ +#define M98925_SLOTOVRN_STATUS_MASK (1<<6) +#define M98925_SLOTOVRN_STATUS_SHIFT 6 +#define M98925_SLOTOVRN_STATUS_WIDTH 1 +#define M98925_INVALSLOT_STATUS_MASK (1<<5) +#define M98925_INVALSLOT_STATUS_SHIFT 5 +#define M98925_INVALSLOT_STATUS_WIDTH 1 +#define M98925_SLOTCNFLT_STATUS_MASK (1<<4) +#define M98925_SLOTCNFLT_STATUS_SHIFT 4 +#define M98925_SLOTCNFLT_STATUS_WIDTH 1 +#define M98925_VBSTOVFL_STATUS_MASK (1<<3) +#define M98925_VBSTOVFL_STATUS_SHIFT 3 +#define M98925_VBSTOVFL_STATUS_WIDTH 1 +#define M98925_VBATOVFL_STATUS_MASK (1<<2) +#define M98925_VBATOVFL_STATUS_SHIFT 2 +#define M98925_VBATOVFL_STATUS_WIDTH 1 +#define M98925_IMONOVFL_STATUS_MASK (1<<1) +#define M98925_IMONOVFL_STATUS_SHIFT 1 +#define M98925_IMONOVFL_STATUS_WIDTH 1 +#define M98925_VMONOVFL_STATUS_MASK (1<<0) +#define M98925_VMONOVFL_STATUS_SHIFT 0 +#define M98925_VMONOVFL_STATUS_WIDTH 1 + +/* MAX98925_R005_STATE0 */ +#define M98925_THERMWARN_END_STATE_MASK (1<<3) +#define M98925_THERMWARN_END_STATE_SHIFT 3 +#define M98925_THERMWARN_END_STATE_WIDTH 1 +#define M98925_THERMWARN_BGN_STATE_MASK (1<<2) +#define M98925_THERMWARN_BGN_STATE_SHIFT 1 +#define M98925_THERMWARN_BGN_STATE_WIDTH 1 +#define M98925_THERMSHDN_END_STATE_MASK (1<<1) +#define M98925_THERMSHDN_END_STATE_SHIFT 1 +#define M98925_THERMSHDN_END_STATE_WIDTH 1 +#define M98925_THERMSHDN_BGN_STATE_MASK (1<<0) +#define M98925_THERMSHDN_BGN_STATE_SHIFT 0 +#define M98925_THERMSHDN_BGN_STATE_WIDTH 1 + +/* MAX98925_R006_STATE1 */ +#define M98925_SPRCURNT_STATE_MASK (1<<5) +#define M98925_SPRCURNT_STATE_SHIFT 5 +#define M98925_SPRCURNT_STATE_WIDTH 1 +#define M98925_WATCHFAIL_STATE_MASK (1<<4) +#define M98925_WATCHFAIL_STATE_SHIFT 4 +#define M98925_WATCHFAIL_STATE_WIDTH 1 +#define M98925_ALCINFH_STATE_MASK (1<<3) +#define M98925_ALCINFH_STATE_SHIFT 3 +#define M98925_ALCINFH_STATE_WIDTH 1 +#define M98925_ALCACT_STATE_MASK (1<<2) +#define M98925_ALCACT_STATE_SHIFT 2 +#define M98925_ALCACT_STATE_WIDTH 1 +#define M98925_ALCMUT_STATE_MASK (1<<1) +#define M98925_ALCMUT_STATE_SHIFT 1 +#define M98925_ALCMUT_STATE_WIDTH 1 +#define M98925_ALCP_STATE_MASK (1<<0) +#define M98925_ALCP_STATE_SHIFT 0 +#define M98925_ALCP_STATE_WIDTH 1 + +/* MAX98925_R007_STATE2 */ +#define M98925_SLOTOVRN_STATE_MASK (1<<6) +#define M98925_SLOTOVRN_STATE_SHIFT 6 +#define M98925_SLOTOVRN_STATE_WIDTH 1 +#define M98925_INVALSLOT_STATE_MASK (1<<5) +#define M98925_INVALSLOT_STATE_SHIFT 5 +#define M98925_INVALSLOT_STATE_WIDTH 1 +#define M98925_SLOTCNFLT_STATE_MASK (1<<4) +#define M98925_SLOTCNFLT_STATE_SHIFT 4 +#define M98925_SLOTCNFLT_STATE_WIDTH 1 +#define M98925_VBSTOVFL_STATE_MASK (1<<3) +#define M98925_VBSTOVFL_STATE_SHIFT 3 +#define M98925_VBSTOVFL_STATE_WIDTH 1 +#define M98925_VBATOVFL_STATE_MASK (1<<2) +#define M98925_VBATOVFL_STATE_SHIFT 2 +#define M98925_VBATOVFL_STATE_WIDTH 1 +#define M98925_IMONOVFL_STATE_MASK (1<<1) +#define M98925_IMONOVFL_STATE_SHIFT 1 +#define M98925_IMONOVFL_STATE_WIDTH 1 +#define M98925_VMONOVFL_STATE_MASK (1<<0) +#define M98925_VMONOVFL_STATE_SHIFT 0 +#define M98925_VMONOVFL_STATE_WIDTH 1 + +/* MAX98925_R008_FLAG0 */ +#define M98925_THERMWARN_END_FLAG_MASK (1<<3) +#define M98925_THERMWARN_END_FLAG_SHIFT 3 +#define M98925_THERMWARN_END_FLAG_WIDTH 1 +#define M98925_THERMWARN_BGN_FLAG_MASK (1<<2) +#define M98925_THERMWARN_BGN_FLAG_SHIFT 2 +#define M98925_THERMWARN_BGN_FLAG_WIDTH 1 +#define M98925_THERMSHDN_END_FLAG_MASK (1<<1) +#define M98925_THERMSHDN_END_FLAG_SHIFT 1 +#define M98925_THERMSHDN_END_FLAG_WIDTH 1 +#define M98925_THERMSHDN_BGN_FLAG_MASK (1<<0) +#define M98925_THERMSHDN_BGN_FLAG_SHIFT 0 +#define M98925_THERMSHDN_BGN_FLAG_WIDTH 1 + +/* MAX98925_R009_FLAG1 */ +#define M98925_SPKCURNT_FLAG_MASK (1<<5) +#define M98925_SPKCURNT_FLAG_SHIFT 5 +#define M98925_SPKCURNT_FLAG_WIDTH 1 +#define M98925_WATCHFAIL_FLAG_MASK (1<<4) +#define M98925_WATCHFAIL_FLAG_SHIFT 4 +#define M98925_WATCHFAIL_FLAG_WIDTH 1 +#define M98925_ALCINFH_FLAG_MASK (1<<3) +#define M98925_ALCINFH_FLAG_SHIFT 3 +#define M98925_ALCINFH_FLAG_WIDTH 1 +#define M98925_ALCACT_FLAG_MASK (1<<2) +#define M98925_ALCACT_FLAG_SHIFT 2 +#define M98925_ALCACT_FLAG_WIDTH 1 +#define M98925_ALCMUT_FLAG_MASK (1<<1) +#define M98925_ALCMUT_FLAG_SHIFT 1 +#define M98925_ALCMUT_FLAG_WIDTH 1 +#define M98925_ALCP_FLAG_MASK (1<<0) +#define M98925_ALCP_FLAG_SHIFT 0 +#define M98925_ALCP_FLAG_WIDTH 1 + +/* MAX98925_R00A_FLAG2 */ +#define M98925_SLOTOVRN_FLAG_MASK (1<<6) +#define M98925_SLOTOVRN_FLAG_SHIFT 6 +#define M98925_SLOTOVRN_FLAG_WIDTH 1 +#define M98925_INVALSLOT_FLAG_MASK (1<<5) +#define M98925_INVALSLOT_FLAG_SHIFT 5 +#define M98925_INVALSLOT_FLAG_WIDTH 1 +#define M98925_SLOTCNFLT_FLAG_MASK (1<<4) +#define M98925_SLOTCNFLT_FLAG_SHIFT 4 +#define M98925_SLOTCNFLT_FLAG_WIDTH 1 +#define M98925_VBSTOVFL_FLAG_MASK (1<<3) +#define M98925_VBSTOVFL_FLAG_SHIFT 3 +#define M98925_VBSTOVFL_FLAG_WIDTH 1 +#define M98925_VBATOVFL_FLAG_MASK (1<<2) +#define M98925_VBATOVFL_FLAG_SHIFT 2 +#define M98925_VBATOVFL_FLAG_WIDTH 1 +#define M98925_IMONOVFL_FLAG_MASK (1<<1) +#define M98925_IMONOVFL_FLAG_SHIFT 1 +#define M98925_IMONOVFL_FLAG_WIDTH 1 +#define M98925_VMONOVFL_FLAG_MASK (1<<0) +#define M98925_VMONOVFL_FLAG_SHIFT 0 +#define M98925_VMONOVFL_FLAG_WIDTH 1 + +/* MAX98925_R00B_IRQ_ENABLE0 */ +#define M98925_THERMWARN_END_EN_MASK (1<<3) +#define M98925_THERMWARN_END_EN_SHIFT 3 +#define M98925_THERMWARN_END_EN_WIDTH 1 +#define M98925_THERMWARN_BGN_EN_MASK (1<<2) +#define M98925_THERMWARN_BGN_EN_SHIFT 2 +#define M98925_THERMWARN_BGN_EN_WIDTH 1 +#define M98925_THERMSHDN_END_EN_MASK (1<<1) +#define M98925_THERMSHDN_END_EN_SHIFT 1 +#define M98925_THERMSHDN_END_EN_WIDTH 1 +#define M98925_THERMSHDN_BGN_EN_MASK (1<<0) +#define M98925_THERMSHDN_BGN_EN_SHIFT 0 +#define M98925_THERMSHDN_BGN_EN_WIDTH 1 + +/* MAX98925_R00C_IRQ_ENABLE1 */ +#define M98925_SPKCURNT_EN_MASK (1<<5) +#define M98925_SPKCURNT_EN_SHIFT 5 +#define M98925_SPKCURNT_EN_WIDTH 1 +#define M98925_WATCHFAIL_EN_MASK (1<<4) +#define M98925_WATCHFAIL_EN_SHIFT 4 +#define M98925_WATCHFAIL_EN_WIDTH 1 +#define M98925_ALCINFH_EN_MASK (1<<3) +#define M98925_ALCINFH_EN_SHIFT 3 +#define M98925_ALCINFH_EN_WIDTH 1 +#define M98925_ALCACT_EN_MASK (1<<2) +#define M98925_ALCACT_EN_SHIFT 2 +#define M98925_ALCACT_EN_WIDTH 1 +#define M98925_ALCMUT_EN_MASK (1<<1) +#define M98925_ALCMUT_EN_SHIFT 1 +#define M98925_ALCMUT_EN_WIDTH 1 +#define M98925_ALCP_EN_MASK (1<<0) +#define M98925_ALCP_EN_SHIFT 0 +#define M98925_ALCP_EN_WIDTH 1 + +/* MAX98925_R00D_IRQ_ENABLE2 */ +#define M98925_SLOTOVRN_EN_MASK (1<<6) +#define M98925_SLOTOVRN_EN_SHIFT 6 +#define M98925_SLOTOVRN_EN_WIDTH 1 +#define M98925_INVALSLOT_EN_MASK (1<<5) +#define M98925_INVALSLOT_EN_SHIFT 5 +#define M98925_INVALSLOT_EN_WIDTH 1 +#define M98925_SLOTCNFLT_EN_MASK (1<<4) +#define M98925_SLOTCNFLT_EN_SHIFT 4 +#define M98925_SLOTCNFLT_EN_WIDTH 1 +#define M98925_VBSTOVFL_EN_MASK (1<<3) +#define M98925_VBSTOVFL_EN_SHIFT 3 +#define M98925_VBSTOVFL_EN_WIDTH 1 +#define M98925_VBATOVFL_EN_MASK (1<<2) +#define M98925_VBATOVFL_EN_SHIFT 2 +#define M98925_VBATOVFL_EN_WIDTH 1 +#define M98925_IMONOVFL_EN_MASK (1<<1) +#define M98925_IMONOVFL_EN_SHIFT 1 +#define M98925_IMONOVFL_EN_WIDTH 1 +#define M98925_VMONOVFL_EN_MASK (1<<0) +#define M98925_VMONOVFL_EN_SHIFT 0 +#define M98925_VMONOVFL_EN_WIDTH 1 + +/* MAX98925_R00E_IRQ_CLEAR0 */ +#define M98925_THERMWARN_END_CLR_MASK (1<<3) +#define M98925_THERMWARN_END_CLR_SHIFT 3 +#define M98925_THERMWARN_END_CLR_WIDTH 1 +#define M98925_THERMWARN_BGN_CLR_MASK (1<<2) +#define M98925_THERMWARN_BGN_CLR_SHIFT 2 +#define M98925_THERMWARN_BGN_CLR_WIDTH 1 +#define M98925_THERMSHDN_END_CLR_MASK (1<<1) +#define M98925_THERMSHDN_END_CLR_SHIFT 1 +#define M98925_THERMSHDN_END_CLR_WIDTH 1 +#define M98925_THERMSHDN_BGN_CLR_MASK (1<<0) +#define M98925_THERMSHDN_BGN_CLR_SHIFT 0 +#define M98925_THERMSHDN_BGN_CLR_WIDTH 1 + +/* MAX98925_R00F_IRQ_CLEAR1 */ +#define M98925_SPKCURNT_CLR_MASK (1<<5) +#define M98925_SPKCURNT_CLR_SHIFT 5 +#define M98925_SPKCURNT_CLR_WIDTH 1 +#define M98925_WATCHFAIL_CLR_MASK (1<<4) +#define M98925_WATCHFAIL_CLR_SHIFT 4 +#define M98925_WATCHFAIL_CLR_WIDTH 1 +#define M98925_ALCINFH_CLR_MASK (1<<3) +#define M98925_ALCINFH_CLR_SHIFT 3 +#define M98925_ALCINFH_CLR_WIDTH 1 +#define M98925_ALCACT_CLR_MASK (1<<2) +#define M98925_ALCACT_CLR_SHIFT 2 +#define M98925_ALCACT_CLR_WIDTH 1 +#define M98925_ALCMUT_CLR_MASK (1<<1) +#define M98925_ALCMUT_CLR_SHIFT 1 +#define M98925_ALCMUT_CLR_WIDTH 1 +#define M98925_ALCP_CLR_MASK (1<<0) +#define M98925_ALCP_CLR_SHIFT 0 +#define M98925_ALCP_CLR_WIDTH 1 + +/* MAX98925_R010_IRQ_CLEAR2 */ +#define M98925_SLOTOVRN_CLR_MASK (1<<6) +#define M98925_SLOTOVRN_CLR_SHIFT 6 +#define M98925_SLOTOVRN_CLR_WIDTH 1 +#define M98925_INVALSLOT_CLR_MASK (1<<5) +#define M98925_INVALSLOT_CLR_SHIFT 5 +#define M98925_INVALSLOT_CLR_WIDTH 1 +#define M98925_SLOTCNFLT_CLR_MASK (1<<4) +#define M98925_SLOTCNFLT_CLR_SHIFT 4 +#define M98925_SLOTCNFLT_CLR_WIDTH 1 +#define M98925_VBSTOVFL_CLR_MASK (1<<3) +#define M98925_VBSTOVFL_CLR_SHIFT 3 +#define M98925_VBSTOVFL_CLR_WIDTH 1 +#define M98925_VBATOVFL_CLR_MASK (1<<2) +#define M98925_VBATOVFL_CLR_SHIFT 2 +#define M98925_VBATOVFL_CLR_WIDTH 1 +#define M98925_IMONOVFL_CLR_MASK (1<<1) +#define M98925_IMONOVFL_CLR_SHIFT 1 +#define M98925_IMONOVFL_CLR_WIDTH 1 +#define M98925_VMONOVFL_CLR_MASK (1<<0) +#define M98925_VMONOVFL_CLR_SHIFT 0 +#define M98925_VMONOVFL_CLR_WIDTH 1 + +/* MAX98925_R011_MAP0 */ +#define M98925_ER_THERMWARN_EN_MASK (1<<7) +#define M98925_ER_THERMWARN_EN_SHIFT 7 +#define M98925_ER_THERMWARN_EN_WIDTH 1 +#define M98925_ER_THERMWARN_MAP_MASK (0x07<<4) +#define M98925_ER_THERMWARN_MAP_SHIFT 4 +#define M98925_ER_THERMWARN_MAP_WIDTH 3 + +/* MAX98925_R012_MAP1 */ +#define M98925_ER_ALCMUT_EN_MASK (1<<7) +#define M98925_ER_ALCMUT_EN_SHIFT 7 +#define M98925_ER_ALCMUT_EN_WIDTH 1 +#define M98925_ER_ALCMUT_MAP_MASK (0x07<<4) +#define M98925_ER_ALCMUT_MAP_SHIFT 4 +#define M98925_ER_ALCMUT_MAP_WIDTH 3 +#define M98925_ER_ALCP_EN_MASK (1<<3) +#define M98925_ER_ALCP_EN_SHIFT 3 +#define M98925_ER_ALCP_EN_WIDTH 1 +#define M98925_ER_ALCP_MAP_MASK (0x07<<0) +#define M98925_ER_ALCP_MAP_SHIFT 0 +#define M98925_ER_ALCP_MAP_WIDTH 3 + +/* MAX98925_R013_MAP2 */ +#define M98925_ER_ALCINFH_EN_MASK (1<<7) +#define M98925_ER_ALCINFH_EN_SHIFT 7 +#define M98925_ER_ALCINFH_EN_WIDTH 1 +#define M98925_ER_ALCINFH_MAP_MASK (0x07<<4) +#define M98925_ER_ALCINFH_MAP_SHIFT 4 +#define M98925_ER_ALCINFH_MAP_WIDTH 3 +#define M98925_ER_ALCACT_EN_MASK (1<<3) +#define M98925_ER_ALCACT_EN_SHIFT 3 +#define M98925_ER_ALCACT_EN_WIDTH 1 +#define M98925_ER_ALCACT_MAP_MASK (0x07<<0) +#define M98925_ER_ALCACT_MAP_SHIFT 0 +#define M98925_ER_ALCACT_MAP_WIDTH 3 + +/* MAX98925_R014_MAP3 */ +#define M98925_ER_SPKCURNT_EN_MASK (1<<7) +#define M98925_ER_SPKCURNT_EN_SHIFT 7 +#define M98925_ER_SPKCURNT_EN_WIDTH 1 +#define M98925_ER_SPKCURNT_MAP_MASK (0x07<<4) +#define M98925_ER_SPKCURNT_MAP_SHIFT 4 +#define M98925_ER_SPKCURNT_MAP_WIDTH 3 + +/* MAX98925_R015_MAP4 */ +/* RESERVED */ + +/* MAX98925_R016_MAP5 */ +#define M98925_ER_IMONOVFL_EN_MASK (1<<7) +#define M98925_ER_IMONOVFL_EN_SHIFT 7 +#define M98925_ER_IMONOVFL_EN_WIDTH 1 +#define M98925_ER_IMONOVFL_MAP_MASK (0x07<<4) +#define M98925_ER_IMONOVFL_MAP_SHIFT 4 +#define M98925_ER_IMONOVFL_MAP_WIDTH 3 +#define M98925_ER_VMONOVFL_EN_MASK (1<<3) +#define M98925_ER_VMONOVFL_EN_SHIFT 3 +#define M98925_ER_VMONOVFL_EN_WIDTH 1 +#define M98925_ER_VMONOVFL_MAP_MASK (0x07<<0) +#define M98925_ER_VMONOVFL_MAP_SHIFT 0 +#define M98925_ER_VMONOVFL_MAP_WIDTH 3 + +/* MAX98925_R017_MAP6 */ +#define M98925_ER_VBSTOVFL_EN_MASK (1<<7) +#define M98925_ER_VBSTOVFL_EN_SHIFT 7 +#define M98925_ER_VBSTOVFL_EN_WIDTH 1 +#define M98925_ER_VBSTOVFL_MAP_MASK (0x07<<4) +#define M98925_ER_VBSTOVFL_MAP_SHIFT 4 +#define M98925_ER_VBSTOVFL_MAP_WIDTH 3 +#define M98925_ER_VBATOVFL_EN_MASK (1<<3) +#define M98925_ER_VBATOVFL_EN_SHIFT 3 +#define M98925_ER_VBATOVFL_EN_WIDTH 1 +#define M98925_ER_VBATOVFL_MAP_MASK (0x07<<0) +#define M98925_ER_VBATOVFL_MAP_SHIFT 0 +#define M98925_ER_VBATOVFL_MAP_WIDTH 3 + +/* MAX98925_R018_MAP7 */ +#define M98925_ER_INVALSLOT_EN_MASK (1<<7) +#define M98925_ER_INVALSLOT_EN_SHIFT 7 +#define M98925_ER_INVALSLOT_EN_WIDTH 1 +#define M98925_ER_INVALSLOT_MAP_MASK (0x07<<4) +#define M98925_ER_INVALSLOT_MAP_SHIFT 4 +#define M98925_ER_INVALSLOT_MAP_WIDTH 3 +#define M98925_ER_SLOTCNFLT_EN_MASK (1<<3) +#define M98925_ER_SLOTCNFLT_EN_SHIFT 3 +#define M98925_ER_SLOTCNFLT_EN_WIDTH 1 +#define M98925_ER_SLOTCNFLT_MAP_MASK (0x07<<0) +#define M98925_ER_SLOTCNFLT_MAP_SHIFT 0 +#define M98925_ER_SLOTCNFLT_MAP_WIDTH 3 + +/* MAX98925_R019_MAP8 */ +#define M98925_ER_SLOTOVRN_EN_MASK (1<<3) +#define M98925_ER_SLOTOVRN_EN_SHIFT 3 +#define M98925_ER_SLOTOVRN_EN_WIDTH 1 +#define M98925_ER_SLOTOVRN_MAP_MASK (0x07<<0) +#define M98925_ER_SLOTOVRN_MAP_SHIFT 0 +#define M98925_ER_SLOTOVRN_MAP_WIDTH 3 + +/* MAX98925_R01A_DAI_CLK_MODE1 */ +#define M98925_DAI_CLK_SOURCE_MASK (1<<6) +#define M98925_DAI_CLK_SOURCE_SHIFT 6 +#define M98925_DAI_CLK_SOURCE_WIDTH 1 +#define M98925_MDLL_MULT_MASK (0x0F<<0) +#define M98925_MDLL_MULT_SHIFT 0 +#define M98925_MDLL_MULT_WIDTH 4 + +#define M98925_MDLL_MULT_MCLKx8 6 +#define M98925_MDLL_MULT_MCLKx16 8 + +/* MAX98925_R01B_DAI_CLK_MODE2 */ +#define M98925_DAI_SR_MASK (0x0F<<4) +#define M98925_DAI_SR_SHIFT 4 +#define M98925_DAI_SR_WIDTH 4 +#define M98925_DAI_MAS_MASK (1<<3) +#define M98925_DAI_MAS_SHIFT 3 +#define M98925_DAI_MAS_WIDTH 1 +#define M98925_DAI_BSEL_MASK (0x07<<0) +#define M98925_DAI_BSEL_SHIFT 0 +#define M98925_DAI_BSEL_WIDTH 3 + +#define M98925_DAI_BSEL_32 (0 << M98925_DAI_BSEL_SHIFT) +#define M98925_DAI_BSEL_48 (1 << M98925_DAI_BSEL_SHIFT) +#define M98925_DAI_BSEL_64 (2 << M98925_DAI_BSEL_SHIFT) +#define M98925_DAI_BSEL_256 (6 << M98925_DAI_BSEL_SHIFT) + +/* MAX98925_R01C_DAI_CLK_DIV_M_MSBS */ +#define M98925_DAI_M_MSBS_MASK (0xFF<<0) +#define M98925_DAI_M_MSBS_SHIFT 0 +#define M98925_DAI_M_MSBS_WIDTH 8 + +/* MAX98925_R01D_DAI_CLK_DIV_M_LSBS */ +#define M98925_DAI_M_LSBS_MASK (0xFF<<0) +#define M98925_DAI_M_LSBS_SHIFT 0 +#define M98925_DAI_M_LSBS_WIDTH 8 + +/* MAX98925_R01E_DAI_CLK_DIV_N_MSBS */ +#define M98925_DAI_N_MSBS_MASK (0x7F<<0) +#define M98925_DAI_N_MSBS_SHIFT 0 +#define M98925_DAI_N_MSBS_WIDTH 7 + +/* MAX98925_R01F_DAI_CLK_DIV_N_LSBS */ +#define M98925_DAI_N_LSBS_MASK (0xFF<<0) +#define M98925_DAI_N_LSBS_SHIFT 0 +#define M98925_DAI_N_LSBS_WIDTH 8 + +/* MAX98925_R020_FORMAT */ +#define M98925_DAI_CHANSZ_MASK (0x03<<6) +#define M98925_DAI_CHANSZ_SHIFT 6 +#define M98925_DAI_CHANSZ_WIDTH 2 +#define M98925_DAI_EXTBCLK_HIZ_MASK (1<<4) +#define M98925_DAI_EXTBCLK_HIZ_SHIFT 4 +#define M98925_DAI_EXTBCLK_HIZ_WIDTH 1 +#define M98925_DAI_WCI_MASK (1<<3) +#define M98925_DAI_WCI_SHIFT 3 +#define M98925_DAI_WCI_WIDTH 1 +#define M98925_DAI_BCI_MASK (1<<2) +#define M98925_DAI_BCI_SHIFT 2 +#define M98925_DAI_BCI_WIDTH 1 +#define M98925_DAI_DLY_MASK (1<<1) +#define M98925_DAI_DLY_SHIFT 1 +#define M98925_DAI_DLY_WIDTH 1 +#define M98925_DAI_TDM_MASK (1<<0) +#define M98925_DAI_TDM_SHIFT 0 +#define M98925_DAI_TDM_WIDTH 1 + +#define M98925_DAI_CHANSZ_16 (1 << M98925_DAI_CHANSZ_SHIFT) +#define M98925_DAI_CHANSZ_24 (2 << M98925_DAI_CHANSZ_SHIFT) +#define M98925_DAI_CHANSZ_32 (3 << M98925_DAI_CHANSZ_SHIFT) + +/* MAX98925_R021_TDM_SLOT_SELECT */ +#define M98925_DAI_DO_EN_MASK (1<<7) +#define M98925_DAI_DO_EN_SHIFT 7 +#define M98925_DAI_DO_EN_WIDTH 1 +#define M98925_DAI_DIN_EN_MASK (1<<6) +#define M98925_DAI_DIN_EN_SHIFT 6 +#define M98925_DAI_DIN_EN_WIDTH 1 +#define M98925_DAI_INR_SOURCE_MASK (0x07<<3) +#define M98925_DAI_INR_SOURCE_SHIFT 3 +#define M98925_DAI_INR_SOURCE_WIDTH 3 +#define M98925_DAI_INL_SOURCE_MASK (0x07<<0) +#define M98925_DAI_INL_SOURCE_SHIFT 0 +#define M98925_DAI_INL_SOURCE_WIDTH 3 + +/* MAX98925_R022_DOUT_CFG_VMON */ +#define M98925_DAI_VMON_EN_MASK (1<<5) +#define M98925_DAI_VMON_EN_SHIFT 5 +#define M98925_DAI_VMON_EN_WIDTH 1 +#define M98925_DAI_VMON_SLOT_MASK (0x1F<<0) +#define M98925_DAI_VMON_SLOT_SHIFT 0 +#define M98925_DAI_VMON_SLOT_WIDTH 5 + +#define M98925_DAI_VMON_SLOT_00_01 (0 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_01_02 (1 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_02_03 (2 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_03_04 (3 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_04_05 (4 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_05_06 (5 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_06_07 (6 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_07_08 (7 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_08_09 (8 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_09_0A (9 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0A_0B (10 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0B_0C (11 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0C_0D (12 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0D_0E (13 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0E_0F (14 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0F_10 (15 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_10_11 (16 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_11_12 (17 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_12_13 (18 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_13_14 (19 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_14_15 (20 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_15_16 (21 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_16_17 (22 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_17_18 (23 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_18_19 (24 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_19_1A (25 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1A_1B (26 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1B_1C (27 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1C_1D (28 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1D_1E (29 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1E_1F (30 << M98925_DAI_VMON_SLOT_SHIFT) + +/* MAX98925_R023_DOUT_CFG_IMON */ +#define M98925_DAI_IMON_EN_MASK (1<<5) +#define M98925_DAI_IMON_EN_SHIFT 5 +#define M98925_DAI_IMON_EN_WIDTH 1 +#define M98925_DAI_IMON_SLOT_MASK (0x1F<<0) +#define M98925_DAI_IMON_SLOT_SHIFT 0 +#define M98925_DAI_IMON_SLOT_WIDTH 5 + +#define M98925_DAI_IMON_SLOT_00_01 (0 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_01_02 (1 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_02_03 (2 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_03_04 (3 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_04_05 (4 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_05_06 (5 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_06_07 (6 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_07_08 (7 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_08_09 (8 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_09_0A (9 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0A_0B (10 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0B_0C (11 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0C_0D (12 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0D_0E (13 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0E_0F (14 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0F_10 (15 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_10_11 (16 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_11_12 (17 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_12_13 (18 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_13_14 (19 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_14_15 (20 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_15_16 (21 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_16_17 (22 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_17_18 (23 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_18_19 (24 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_19_1A (25 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1A_1B (26 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1B_1C (27 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1C_1D (28 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1D_1E (29 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1E_1F (30 << M98925_DAI_IMON_SLOT_SHIFT) + +/* MAX98925_R024_DOUT_CFG_VBAT */ +#define M98925_DAI_VBAT_EN_MASK (1<<5) +#define M98925_DAI_VBAT_EN_SHIFT 5 +#define M98925_DAI_VBAT_EN_WIDTH 1 +#define M98925_DAI_VBAT_SLOT_MASK (0x1F<<0) +#define M98925_DAI_VBAT_SLOT_SHIFT 0 +#define M98925_DAI_VBAT_SLOT_WIDTH 5 + +/* MAX98925_R025_DOUT_CFG_VBST */ +#define M98925_DAI_VBST_EN_MASK (1<<5) +#define M98925_DAI_VBST_EN_SHIFT 5 +#define M98925_DAI_VBST_EN_WIDTH 1 +#define M98925_DAI_VBST_SLOT_MASK (0x1F<<0) +#define M98925_DAI_VBST_SLOT_SHIFT 0 +#define M98925_DAI_VBST_SLOT_WIDTH 5 + +/* MAX98925_R026_DOUT_CFG_FLAG */ +#define M98925_DAI_FLAG_EN_MASK (1<<5) +#define M98925_DAI_FLAG_EN_SHIFT 5 +#define M98925_DAI_FLAG_EN_WIDTH 1 +#define M98925_DAI_FLAG_SLOT_MASK (0x1F<<0) +#define M98925_DAI_FLAG_SLOT_SHIFT 0 +#define M98925_DAI_FLAG_SLOT_WIDTH 5 + +/* MAX98925_R027_DOUT_HIZ_CFG1 */ +#define M98925_DAI_SLOT_HIZ_CFG1_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG1_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG1_WIDTH 8 + +/* MAX98925_R028_DOUT_HIZ_CFG2 */ +#define M98925_DAI_SLOT_HIZ_CFG2_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG2_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG2_WIDTH 8 + +/* MAX98925_R029_DOUT_HIZ_CFG3 */ +#define M98925_DAI_SLOT_HIZ_CFG3_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG3_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG3_WIDTH 8 + +/* MAX98925_R02A_DOUT_HIZ_CFG4 */ +#define M98925_DAI_SLOT_HIZ_CFG4_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG4_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG4_WIDTH 8 + +/* MAX98925_R02B_DOUT_DRV_STRENGTH */ +#define M98925_DAI_OUT_DRIVE_MASK (0x03<<0) +#define M98925_DAI_OUT_DRIVE_SHIFT 0 +#define M98925_DAI_OUT_DRIVE_WIDTH 2 + +/* MAX98925_R02C_FILTERS */ +#define M98925_ADC_DITHER_EN_MASK (1<<7) +#define M98925_ADC_DITHER_EN_SHIFT 7 +#define M98925_ADC_DITHER_EN_WIDTH 1 +#define M98925_IV_DCB_EN_MASK (1<<6) +#define M98925_IV_DCB_EN_SHIFT 6 +#define M98925_IV_DCB_EN_WIDTH 1 +#define M98925_DAC_DITHER_EN_MASK (1<<4) +#define M98925_DAC_DITHER_EN_SHIFT 4 +#define M98925_DAC_DITHER_EN_WIDTH 1 +#define M98925_DAC_FILTER_MODE_MASK (1<<3) +#define M98925_DAC_FILTER_MODE_SHIFT 3 +#define M98925_DAC_FILTER_MODE_WIDTH 1 +#define M98925_DAC_HPF_MASK (0x07<<0) +#define M98925_DAC_HPF_SHIFT 0 +#define M98925_DAC_HPF_WIDTH 3 +#define M98925_DAC_HPF_DISABLE (0 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_DC_BLOCK (1 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_EN_100 (2 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_EN_200 (3 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_EN_400 (4 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_EN_800 (5 << M98925_DAC_HPF_SHIFT) + +/* MAX98925_R02D_GAIN */ +#define M98925_DAC_IN_SEL_MASK (0x03<<5) +#define M98925_DAC_IN_SEL_SHIFT 5 +#define M98925_DAC_IN_SEL_WIDTH 2 +#define M98925_SPK_GAIN_MASK (0x1F<<0) +#define M98925_SPK_GAIN_SHIFT 0 +#define M98925_SPK_GAIN_WIDTH 5 + +#define M98925_DAC_IN_SEL_LEFT_DAI (0 << M98925_DAC_IN_SEL_SHIFT) +#define M98925_DAC_IN_SEL_RIGHT_DAI (1 << M98925_DAC_IN_SEL_SHIFT) +#define M98925_DAC_IN_SEL_SUMMED_DAI (2 << M98925_DAC_IN_SEL_SHIFT) +#define M98925_DAC_IN_SEL_DIV2_SUMMED_DAI (3 << M98925_DAC_IN_SEL_SHIFT) + +/* MAX98925_R02E_GAIN_RAMPING */ +#define M98925_SPK_RMP_EN_MASK (1<<1) +#define M98925_SPK_RMP_EN_SHIFT 1 +#define M98925_SPK_RMP_EN_WIDTH 1 +#define M98925_SPK_ZCD_EN_MASK (1<<0) +#define M98925_SPK_ZCD_EN_SHIFT 0 +#define M98925_SPK_ZCD_EN_WIDTH 1 + +/* MAX98925_R02F_SPK_AMP */ +#define M98925_SPK_MODE_MASK (1<<0) +#define M98925_SPK_MODE_SHIFT 0 +#define M98925_SPK_MODE_WIDTH 1 + +/* MAX98925_R030_THRESHOLD */ +#define M98925_ALC_EN_MASK (1<<5) +#define M98925_ALC_EN_SHIFT 5 +#define M98925_ALC_EN_WIDTH 1 +#define M98925_ALC_TH_MASK (0x1F<<0) +#define M98925_ALC_TH_SHIFT 0 +#define M98925_ALC_TH_WIDTH 5 + +/* MAX98925_R031_ALC_ATTACK */ +#define M98925_ALC_ATK_STEP_MASK (0x0F<<4) +#define M98925_ALC_ATK_STEP_SHIFT 4 +#define M98925_ALC_ATK_STEP_WIDTH 4 +#define M98925_ALC_ATK_RATE_MASK (0x7<<0) +#define M98925_ALC_ATK_RATE_SHIFT 0 +#define M98925_ALC_ATK_RATE_WIDTH 3 + +/* MAX98925_R032_ALC_ATTEN_RLS */ +#define M98925_ALC_MAX_ATTEN_MASK (0x0F<<4) +#define M98925_ALC_MAX_ATTEN_SHIFT 4 +#define M98925_ALC_MAX_ATTEN_WIDTH 4 +#define M98925_ALC_RLS_RATE_MASK (0x7<<0) +#define M98925_ALC_RLS_RATE_SHIFT 0 +#define M98925_ALC_RLS_RATE_WIDTH 3 + +/* MAX98925_R033_ALC_HOLD_RLS */ +#define M98925_ALC_RLS_TGR_MASK (1<<0) +#define M98925_ALC_RLS_TGR_SHIFT 0 +#define M98925_ALC_RLS_TGR_WIDTH 1 + +/* MAX98925_R034_ALC_CONFIGURATION */ +#define M98925_ALC_MUTE_EN_MASK (1<<7) +#define M98925_ALC_MUTE_EN_SHIFT 7 +#define M98925_ALC_MUTE_EN_WIDTH 1 +#define M98925_ALC_MUTE_DLY_MASK (0x07<<4) +#define M98925_ALC_MUTE_DLY_SHIFT 4 +#define M98925_ALC_MUTE_DLY_WIDTH 3 +#define M98925_ALC_RLS_DBT_MASK (0x07<<0) +#define M98925_ALC_RLS_DBT_SHIFT 0 +#define M98925_ALC_RLS_DBT_WIDTH 3 + +/* MAX98925_R035_BOOST_CONVERTER */ +#define M98925_BST_SYNC_MASK (1<<7) +#define M98925_BST_SYNC_SHIFT 7 +#define M98925_BST_SYNC_WIDTH 1 +#define M98925_BST_PHASE_MASK (0x03<<4) +#define M98925_BST_PHASE_SHIFT 4 +#define M98925_BST_PHASE_WIDTH 2 +#define M98925_BST_SKIP_MODE_MASK (0x03<<0) +#define M98925_BST_SKIP_MODE_SHIFT 0 +#define M98925_BST_SKIP_MODE_WIDTH 2 + +/* MAX98925_R036_BLOCK_ENABLE */ +#define M98925_BST_EN_MASK (1<<7) +#define M98925_BST_EN_SHIFT 7 +#define M98925_BST_EN_WIDTH 1 +#define M98925_WATCH_EN_MASK (1<<6) +#define M98925_WATCH_EN_SHIFT 6 +#define M98925_WATCH_EN_WIDTH 1 +#define M98925_CLKMON_EN_MASK (1<<5) +#define M98925_CLKMON_EN_SHIFT 5 +#define M98925_CLKMON_EN_WIDTH 1 +#define M98925_SPK_EN_MASK (1<<4) +#define M98925_SPK_EN_SHIFT 4 +#define M98925_SPK_EN_WIDTH 1 +#define M98925_ADC_VBST_EN_MASK (1<<3) +#define M98925_ADC_VBST_EN_SHIFT 3 +#define M98925_ADC_VBST_EN_WIDTH 1 +#define M98925_ADC_VBAT_EN_MASK (1<<2) +#define M98925_ADC_VBAT_EN_SHIFT 2 +#define M98925_ADC_VBAT_EN_WIDTH 1 +#define M98925_ADC_IMON_EN_MASK (1<<1) +#define M98925_ADC_IMON_EN_SHIFT 1 +#define M98925_ADC_IMON_EN_WIDTH 1 +#define M98925_ADC_VMON_EN_MASK (1<<0) +#define M98925_ADC_VMON_EN_SHIFT 0 +#define M98925_ADC_VMON_EN_WIDTH 1 + +/* MAX98925_R037_CONFIGURATION */ +#define M98925_BST_VOUT_MASK (0x0F<<4) +#define M98925_BST_VOUT_SHIFT 4 +#define M98925_BST_VOUT_WIDTH 4 +#define M98925_THERMWARN_LEVEL_MASK (0x03<<2) +#define M98925_THERMWARN_LEVEL_SHIFT 2 +#define M98925_THERMWARN_LEVEL_WIDTH 2 +#define M98925_WATCH_TIME_MASK (0x03<<0) +#define M98925_WATCH_TIME_SHIFT 0 +#define M98925_WATCH_TIME_WIDTH 2 + +/* MAX98925_R038_GLOBAL_ENABLE */ +#define M98925_EN_MASK (1<<7) +#define M98925_EN_SHIFT 7 +#define M98925_EN_WIDTH 1 + +/* MAX98925_R03A_BOOST_LIMITER */ +#define M98925_BST_ILIM_MASK (0x1F<<3) +#define M98925_BST_ILIM_SHIFT 3 +#define M98925_BST_ILIM_WIDTH 5 + +/* MAX98925_R0FF_VERSION */ +#define M98925_REV_ID_MASK (0xFF<<0) +#define M98925_REV_ID_SHIFT 0 +#define M98925_REV_ID_WIDTH 8 + +struct max98925_cdata { + unsigned int rate; + unsigned int fmt; +}; + +struct max98925_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct max98925_pdata *pdata; + unsigned int sysclk; + unsigned int v_slot; + unsigned int i_slot; + unsigned int spk_gain; + unsigned int ch_size; +}; +#endif -- 1.9.3

4 5

[alsa-devel] [PATCH v4] ASoC: rt286: add jack detection disable with NULL jack passed
by Jie Yang 24 Feb '15

24 Feb '15

Some platforms, e.g. WSB, don't need jack detection when system is in Suspend, for power save reason. Here add headphone/mic jack detection disable feature with NULL jack passed in, when disabled, it will disable interrupt, and disable LDO1, which is used for jack detection when headphone is plugged in. Signed-off-by: Jie Yang <yang.jie(a)intel.com> Reviewed-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index d5cc1bc..c778028 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -394,9 +394,20 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) rt286->jack = jack; - /* Send an initial empty report */ - snd_soc_jack_report(rt286->jack, 0, - SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + if (jack) { + /* enable IRQ */ + if (rt286->jack->status | SND_JACK_HEADPHONE) + snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO1"); + regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x2); + /* Send an initial empty report */ + snd_soc_jack_report(rt286->jack, rt286->jack->status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + } else { + /* disable IRQ */ + regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x0); + snd_soc_dapm_disable_pin(&codec->dapm, "LDO1"); + } + snd_soc_dapm_sync(&codec->dapm); return 0; } -- 1.9.1

2 1

[alsa-devel] No sound and systemd journal filling when inserting headphones when power-saving is enabled
by Dang Sananikone 24 Feb '15

24 Feb '15

Hi, I previously posted in alsa user mailing list about a problem I was facing with having no sound and the systemd journal filling when using headphones: http://article.gmane.org/gmane.linux.alsa.user/38948 I subsequently realised that I had enabled power-saving a few weeks ago and it was this power-saving that was causing the problem. My response to myself is here: http://article.gmane.org/gmane.linux.alsa.user/38953 I am now satisfied that disabling power-saving has resolved my issue, but there has been a suggestion that I report this issue, so here it is. Apologies if this is the wrong place (this is my first time on the alsa mailing list). Let me know if you want me to create a bug ticket and where it should be created (the bug tracker URL link in the alsa project page is not working).

2 1

[alsa-devel] [PATCH 0/20] ASoC: rsnd: add Synchronous SRC and U/O error salvage support
by Kuninori Morimoto 24 Feb '15

24 Feb '15

Hi Mark. These patches adds [1] Synchronous SRC, [2] under/over run error salvage support to Renesas sound driver. 1) - 16) : prepare cleanup/tidyup for 17) - 19) 17) : Synchronous SRC 18)- 19) : salvage support for under/over run error 20) : cleanup for 18) - 19) Current Renesas sound can use Asynchronous/Synchronous sampling rate convert. Asynchronous sampling rate convert is already supported on upstream. ("feature" is supported, "DT" bindings is not yet supported) 17) adds Synchronous sampling rate convert. Renesas sound has HW bug (?) which switches L/R channel if under/over run error had occurred. 18) - 19) adds salvage supported for it. 20) is cleanup patch which is based on 18)-19) patch Kuninori Morimoto (20): 1. ASoC: rsnd: add .fallback callback 2. ASoC: rsnd: add callback status check method 3. ASoC: rsnd: rsnd_src_ssiu_stop() stops SSIU compulsorily 4. ASoC: rsnd: tidyup PIO/DMA mode settings method 5. ASoC: rsnd: tidyup SSI interrupt enable/disable method 6. ASoC: rsnd: care SSIWSR register in rsnd_ssi_hw_start() 7. ASoC: rsnd: clear status register when HW start 8. ASoC: rsnd: synchronize SSI start/stop sequence between PIO/DMA mode 9. ASoC: rsnd: show master clock rate when ADG probe 10. ASoC: rsnd: move snd_kcontrol_new fucntions to core.c 11. ASoC: rsnd: tidyup rsnd_io_to_runtime() macro 12. ASoC: rsnd: use rsnd_src_convert_rate() once on rsnd_src_set_convert_rate_gen2() 13. ASoC: rsnd: initialize SRC on rsnd_src_init() 14. ASoC: rsnd: set SRC_ROUTE_MODE0 on each rsnd_src_set_convert_rate() 15. ASoC: rsnd: share SSI starting method between PIO/DMA mode 16. ASoC: rsnd: remove un-necessary parameter from rsnd_src_start/stop() 17. ASoC: rsnd: add Synchronous SRC mode 18. ASoC: rsnd: add salvage support for under/over flow error on SSI 19. ASoC: rsnd: add salvage support for under/over flow error on SRC 20. ASoC: rsnd: rename SSI function name of PIO include/sound/rcar_snd.h | 9 +- sound/soc/sh/rcar/adg.c | 2 +- sound/soc/sh/rcar/core.c | 166 +++++++++++++++++++- sound/soc/sh/rcar/dvc.c | 177 ++-------------------- sound/soc/sh/rcar/gen.c | 15 ++ sound/soc/sh/rcar/rsnd.h | 94 +++++++++++- sound/soc/sh/rcar/src.c | 378 +++++++++++++++++++++++++++++++++++++++------- sound/soc/sh/rcar/ssi.c | 205 +++++++++++++++---------- 8 files changed, 736 insertions(+), 310 deletions(-)

3 40

[alsa-devel] [PATCH 1/2] ASoC: core: Add extra dapm properties for Device Tree
by Nicolin Chen 24 Feb '15

24 Feb '15

The current helper functions, snd_soc_of_parse_audio_simple_widgets() and snd_soc_of_parse_audio_routing(), set dapm_widgets and dapm_routes without caring if they are already set by using build-in widgets and routes in the card driver. So there could be one of them, build-in one or Device Tree one, overrided by the other depending on which one was assigned later. This patch adds an extra pair of dapm_widgets and dapm_routes for DT use only so as to prevent unexpected overriding. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 16 ++++++++++++---- 2 files changed, 17 insertions(+), 4 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index ac8b333..7443062 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1068,11 +1068,16 @@ struct snd_soc_card { /* * Card-specific routes and widgets. + * Note: of_dapm_xxx for Device Tree; Otherwise for driver build-in. */ const struct snd_soc_dapm_widget *dapm_widgets; int num_dapm_widgets; const struct snd_soc_dapm_route *dapm_routes; int num_dapm_routes; + const struct snd_soc_dapm_widget *of_dapm_widgets; + int num_of_dapm_widgets; + const struct snd_soc_dapm_route *of_dapm_routes; + int num_of_dapm_routes; bool fully_routed; struct work_struct deferred_resume_work; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c024962..2695a12 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1486,6 +1486,10 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets, card->num_dapm_widgets); + if (card->of_dapm_widgets) + snd_soc_dapm_new_controls(&card->dapm, card->of_dapm_widgets, + card->num_of_dapm_widgets); + /* initialise the sound card only once */ if (card->probe) { ret = card->probe(card); @@ -1541,6 +1545,10 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); + if (card->of_dapm_routes) + snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes, + card->num_of_dapm_routes); + for (i = 0; i < card->num_links; i++) { struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; dai_link = &card->dai_link[i]; @@ -3187,8 +3195,8 @@ int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, widgets[i].name = wname; } - card->dapm_widgets = widgets; - card->num_dapm_widgets = num_widgets; + card->of_dapm_widgets = widgets; + card->num_of_dapm_widgets = num_widgets; return 0; } @@ -3272,8 +3280,8 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } } - card->num_dapm_routes = num_routes; - card->dapm_routes = routes; + card->num_of_dapm_routes = num_routes; + card->of_dapm_routes = routes; return 0; } -- 1.9.1

2 2

[alsa-devel] [PATCH 0/2] ASoC: pcm512x: Add knobs to allow and control overclocking
by Peter Rosin 24 Feb '15

24 Feb '15

From: Peter Rosin <peda(a)axentia.se> Hi! I wasn't sure if I should add Documentation/* for these sysfs knobs or not? A lot of knobs do not seem to have docs (no specific example, just a gut feeling). And I'm not sure how I should name the doc-file since the pcm512x driver handles devices connected with both i2c and spi. And what KernelVersion should I enter? That depends... So, this isn't perfect, suggestions welcome. The first patch is a preparatory patch that makes 2/2 slightly simpler. I can fold them together if that's desired. Or I could split 2/2 up in three logical patches, one for each of the independent PLL/DSP/DAC knobs. Another "feature" that one might want, is that attempts to change the overclocking should fail with EBUSY when the device is already in use. But I haven't invested enough time to find out how I should determine when to fail, so for this version the driver accepts the change but doesn't enforce the new limit until the clock rates are recalculated. Cheers, Peter Peter Rosin (2): ASoC: pcm512x: Rearrange to not repeat dacsrc_rate / dac_div ASoC: pcm512x: Allow independently overclocking PLL, DAC and DSP Documentation/ABI/testing/sysfs-i2c-pcm512x | 35 +++++ sound/soc/codecs/pcm512x.c | 189 ++++++++++++++++++++++++--- 2 files changed, 205 insertions(+), 19 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-i2c-pcm512x -- 1.7.10.4

3 12

[alsa-devel] [PATCH] ALSA: hda: controller code - do not export static functions
by Jaroslav Kysela 24 Feb '15

24 Feb '15

It is a bad idea to export static functions. GCC for some platforms shows errors like: error: __ksymtab_azx_get_response causes a section type conflict Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/hda_controller.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index ebb7a64..f63d2cb5 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -978,7 +978,6 @@ static int azx_alloc_cmd_io(struct azx *chip) dev_err(chip->card->dev, "cannot allocate CORB/RIRB\n"); return err; } -EXPORT_SYMBOL_GPL(azx_alloc_cmd_io); static void azx_init_cmd_io(struct azx *chip) { @@ -1043,7 +1042,6 @@ static void azx_init_cmd_io(struct azx *chip) azx_writeb(chip, RIRBCTL, AZX_RBCTL_DMA_EN | AZX_RBCTL_IRQ_EN); spin_unlock_irq(&chip->reg_lock); } -EXPORT_SYMBOL_GPL(azx_init_cmd_io); static void azx_free_cmd_io(struct azx *chip) { @@ -1053,7 +1051,6 @@ static void azx_free_cmd_io(struct azx *chip) azx_writeb(chip, CORBCTL, 0); spin_unlock_irq(&chip->reg_lock); } -EXPORT_SYMBOL_GPL(azx_free_cmd_io); static unsigned int azx_command_addr(u32 cmd) { @@ -1333,7 +1330,6 @@ static int azx_send_cmd(struct hda_bus *bus, unsigned int val) else return azx_corb_send_cmd(bus, val); } -EXPORT_SYMBOL_GPL(azx_send_cmd); /* get a response */ static unsigned int azx_get_response(struct hda_bus *bus, @@ -1347,7 +1343,6 @@ static unsigned int azx_get_response(struct hda_bus *bus, else return azx_rirb_get_response(bus, addr); } -EXPORT_SYMBOL_GPL(azx_get_response); #ifdef CONFIG_SND_HDA_DSP_LOADER /* -- 1.9.3

2 1

[alsa-devel] [PATCH] ASoC: fsl_ssi: Don't try to round-up for PM divisor calculation
by Nicolin Chen 24 Feb '15

24 Feb '15

According to i.MX6 Series Reference Manual, the formula to calculate the sys clock is sysclk rate = bclk rate * (div2 + 1) * (7 * psr + 1) * (pm + 1) * 2 Commit aafa85e71a75 ("ASoC: fsl_ssi: Add DAI master mode support for SSI on i.MX series") added the divisor calculation which relies on the clk_round_rate(). However, at that time, clk_round_rate() didn't provide closest clock rates for some cases because it might not use a correct rounding policy. So using the original formula (pm + 1) for PM divisor was not able to give us a desired clock rate. And then we used (pm + 2) to do the trick. However, the clk-divider driver has been refined a lot since commit b11d282dbea2 ("clk: divider: fix rate calculation for fractional rates") Now using (pm + 2) trick would result an incorrect clock rate. So this patch fixes the problem by removing the useless trick. Reported-by: Stephane Cerveau <scerveau(a)voxtok.com> Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- [ I also checked that only 3.18 stable tree has the refinements for clk-divider driver, so I guess it might not be able to apply to previous long-term trees if being treated as bug fix. -- Nicolin ] sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index d7365c5..3e7c644 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -603,7 +603,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, factor = (div2 + 1) * (7 * psr + 1) * 2; for (i = 0; i < 255; i++) { - tmprate = freq * factor * (i + 2); + tmprate = freq * factor * (i + 1); if (baudclk_is_used) clkrate = clk_get_rate(ssi_private->baudclk); -- 1.9.1

2 2

[alsa-devel] [PATCH v2] ASoC: sta32x: fix register range in regmap.
by Thomas Niederprüm 24 Feb '15

24 Feb '15

The STA32X_AUTO3 is a writable register that currently does not appear in the regmap ranges(neither read nor write). By adding this register to the register ranges there is no gap anymore and the existing register ranges can be joined. This fixes a regression introduced in commit a1be4cead9b9504aa6fc93b624975601cec8c188 where the driver was moved to direct regmap usage and the STA32X_AUTO3 register was missed. That made it impossible to choose the preset EQ mode set through the STA32X_AUTO3 register. Signed-off-by: Thomas Niederprüm <niederp(a)physik.uni-kl.de> --- changes since v1: - updated to apply to linux-next. - added commit id that introduced the regression in the commit message. sound/soc/codecs/sta32x.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index e28c185..fc686ca 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -106,13 +106,11 @@ static const struct reg_default sta32x_regs[] = { }; static const struct regmap_range sta32x_write_regs_range[] = { - regmap_reg_range(STA32X_CONFA, STA32X_AUTO2), - regmap_reg_range(STA32X_C1CFG, STA32X_FDRC2), + regmap_reg_range(STA32X_CONFA, STA32X_FDRC2), }; static const struct regmap_range sta32x_read_regs_range[] = { - regmap_reg_range(STA32X_CONFA, STA32X_AUTO2), - regmap_reg_range(STA32X_C1CFG, STA32X_FDRC2), + regmap_reg_range(STA32X_CONFA, STA32X_FDRC2), }; static const struct regmap_range sta32x_volatile_regs_range[] = { -- 2.1.1

2 1

[alsa-devel] [PATCH] ASoC: improve usage of gpiod API
by Uwe Kleine-König 24 Feb '15

24 Feb '15

Since 39b2bbe3d715 (gpio: add flags argument to gpiod_get*() functions) which appeared in v3.17-rc1, the gpiod_get* functions take an additional parameter that allows to specify direction and initial value for output. Simplify drivers accordingly. Also there is an *_optional variant that serves well here. The sematics is slightly changed here by using it as error checking is more strict now: If GPIOLIB is not enabled an error is returned instead of just ignoring the gpio. On one hand this is bad for devices that don't "have" the respective gpio as the driver is failing now. On the other hand there is no means to assert that this gpio is really not needed or if only the driver to control it is not available. The latter is a real reason to fail and so it's defensive to fail here, too. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/adau1977.c | 17 +++++------------ sound/soc/codecs/cs35l32.c | 19 ++++++------------- sound/soc/codecs/cs4265.c | 19 ++++++------------- sound/soc/codecs/sta350.c | 30 +++++++++--------------------- sound/soc/codecs/tas2552.c | 13 +++---------- 5 files changed, 29 insertions(+), 69 deletions(-) diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c index 70ab35744aba..7ad8e156e2df 100644 --- a/sound/soc/codecs/adau1977.c +++ b/sound/soc/codecs/adau1977.c @@ -938,22 +938,15 @@ int adau1977_probe(struct device *dev, struct regmap *regmap, adau1977->dvdd_reg = NULL; } - adau1977->reset_gpio = devm_gpiod_get(dev, "reset"); - if (IS_ERR(adau1977->reset_gpio)) { - ret = PTR_ERR(adau1977->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return PTR_ERR(adau1977->reset_gpio); - adau1977->reset_gpio = NULL; - } + adau1977->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(adau1977->reset_gpio)) + return PTR_ERR(adau1977->reset_gpio); dev_set_drvdata(dev, adau1977); - if (adau1977->reset_gpio) { - ret = gpiod_direction_output(adau1977->reset_gpio, 0); - if (ret) - return ret; + if (adau1977->reset_gpio) ndelay(100); - } ret = adau1977_power_enable(adau1977); if (ret) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index f2b8aad21274..60598b230341 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -437,20 +437,13 @@ static int cs35l32_i2c_probe(struct i2c_client *i2c_client, } /* Reset the Device */ - cs35l32->reset_gpio = devm_gpiod_get(&i2c_client->dev, - "reset-gpios"); - if (IS_ERR(cs35l32->reset_gpio)) { - ret = PTR_ERR(cs35l32->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - cs35l32->reset_gpio = NULL; - } else { - ret = gpiod_direction_output(cs35l32->reset_gpio, 0); - if (ret) - return ret; + cs35l32->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs35l32->reset_gpio)) + return PTR_ERR(cs35l32->reset_gpio); + + if (cs35l32->reset_gpio) gpiod_set_value_cansleep(cs35l32->reset_gpio, 1); - } /* initialize codec */ ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_AB, &reg); diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index ce6086835ebd..cac48ddf3ba6 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -605,21 +605,14 @@ static int cs4265_i2c_probe(struct i2c_client *i2c_client, return ret; } - cs4265->reset_gpio = devm_gpiod_get(&i2c_client->dev, - "reset-gpios"); - if (IS_ERR(cs4265->reset_gpio)) { - ret = PTR_ERR(cs4265->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - cs4265->reset_gpio = NULL; - } else { - ret = gpiod_direction_output(cs4265->reset_gpio, 0); - if (ret) - return ret; + cs4265->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs4265->reset_gpio)) + return PTR_ERR(cs4265->reset_gpio); + + if (cs4265->reset_gpio) { mdelay(1); gpiod_set_value_cansleep(cs4265->reset_gpio, 1); - } i2c_set_clientdata(i2c_client, cs4265); diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c index bda2ee18769e..669e3228241e 100644 --- a/sound/soc/codecs/sta350.c +++ b/sound/soc/codecs/sta350.c @@ -1213,27 +1213,15 @@ static int sta350_i2c_probe(struct i2c_client *i2c, #endif /* GPIOs */ - sta350->gpiod_nreset = devm_gpiod_get(dev, "reset"); - if (IS_ERR(sta350->gpiod_nreset)) { - ret = PTR_ERR(sta350->gpiod_nreset); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta350->gpiod_nreset = NULL; - } else { - gpiod_direction_output(sta350->gpiod_nreset, 0); - } - - sta350->gpiod_power_down = devm_gpiod_get(dev, "power-down"); - if (IS_ERR(sta350->gpiod_power_down)) { - ret = PTR_ERR(sta350->gpiod_power_down); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta350->gpiod_power_down = NULL; - } else { - gpiod_direction_output(sta350->gpiod_power_down, 0); - } + sta350->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(sta350->gpiod_nreset)) + return PTR_ERR(sta350->gpiod_nreset); + + sta350->gpiod_power_down = devm_gpiod_get(dev, "power-down", + GPIOD_OUT_LOW); + if (IS_ERR(sta350->gpiod_power_down)) + return PTR_ERR(sta350->gpiod_power_down); /* regulators */ for (i = 0; i < ARRAY_SIZE(sta350->supplies); i++) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index ae23acdd2708..dfb4ff5cc9ea 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -485,16 +485,9 @@ static int tas2552_probe(struct i2c_client *client, if (data == NULL) return -ENOMEM; - data->enable_gpio = devm_gpiod_get(dev, "enable"); - if (IS_ERR(data->enable_gpio)) { - ret = PTR_ERR(data->enable_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - data->enable_gpio = NULL; - } else { - gpiod_direction_output(data->enable_gpio, 0); - } + data->enable_gpio = devm_gpiod_get(dev, "enable", GPIOD_OUT_LOW); + if (IS_ERR(data->enable_gpio)) + return PTR_ERR(data->enable_gpio); data->tas2552_client = client; data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: rt5670: Set RT5670_IRQ_CTRL1 non volatile
by Bard Liao 24 Feb '15

24 Feb '15

RT5670_IRQ_CTRL1(0xbd) is a non volatile register. And we need to restore its value after suspend/resume. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 6aa4ab2..04ca247 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -225,7 +225,6 @@ static bool rt5670_volatile_register(struct device *dev, unsigned int reg) case RT5670_ADC_EQ_CTRL1: case RT5670_EQ_CTRL1: case RT5670_ALC_CTRL_1: - case RT5670_IRQ_CTRL1: case RT5670_IRQ_CTRL2: case RT5670_INT_IRQ_ST: case RT5670_IL_CMD: -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v2 0/4] ASoC: Intel: BSW driver updates
by Vinod Koul 24 Feb '15

24 Feb '15

This series adds rest of the remaining 4 patches posted in v1. This is generated on top of Mark's topic/intel Subhransu S. Prusty (2): ASoC: Intel: Remove ignore suspend support ASoC: Intel: Remove soc pm handling to allow platform driver handle it Vinod Koul (2): ASoC: Intel: add pm support in sst ipc driver ASoC: Intel: Move the fw download to power_control sound/soc/intel/cht_bsw_rt5672.c | 3 - sound/soc/intel/sst/sst.c | 128 +++++++++++++++++++++++++++++-- sound/soc/intel/sst/sst.h | 9 +++ sound/soc/intel/sst/sst_drv_interface.c | 27 ++++++- 4 files changed, 152 insertions(+), 15 deletions(-) -- 1.7.9.5

2 5

[alsa-devel] [PATCH 00/13] ASoC: Intel: BSW driver updates
by Vinod Koul 24 Feb '15

24 Feb '15

This patch series first adds nonatomic pcm ops support in ASoC (it is already availble in ALSA). Then adds driver support. Later it adds puase/resume and suspend/resume support to the intel driver Subhransu S. Prusty (3): ASoC: Intel: Add memcpy32_fromio as well ASoC: Intel: Remove ignore suspend support ASoC: Intel: Remove soc pm handling to allow platform driver handle it Vinod Koul (10): ASoC: core: allow pcms to be registered as nonatomic ASoC: Intel: mark cht machine driver with nonatomic trigger ASoC: Intel: update MMX ID to 3 ASoC: Intel: add support for pause and resume in sst ASoC: Intel: add support for pcm stream suspend/resume ASoC: Intel: add support for platform suspend ASoC: Intel: save and restore the CSR register ASoC: Intel: reset the DSP while suspending ASoC: Intel: add pm support in sst ipc driver ASoC: Intel: Move the fw download to power_control include/sound/soc.h | 3 + sound/soc/intel/cht_bsw_rt5672.c | 5 +- sound/soc/intel/sst-atom-controls.h | 2 +- sound/soc/intel/sst-mfld-platform-pcm.c | 60 ++++++++++++++ sound/soc/intel/sst-mfld-platform.h | 1 + sound/soc/intel/sst/sst.c | 134 ++++++++++++++++++++++++++++--- sound/soc/intel/sst/sst.h | 12 +++ sound/soc/intel/sst/sst_drv_interface.c | 58 ++++++++++++- sound/soc/intel/sst/sst_loader.c | 10 ++- sound/soc/soc-pcm.c | 1 + 10 files changed, 268 insertions(+), 18 deletions(-) -- 1.7.9.5

2 16

Re: [alsa-devel] Intel HDA / ALC662 analog surround problem
by Raymond Yau 24 Feb '15

24 Feb '15

> I'm using alsa 1.0.28 (lib and utils). Everything works fine so far, i.e. stereo output through speakers and headphone as well as S/PDIF and all possible capturing. > According to my mainboard manual, the codec supports 5.1 (analog) surround by using the rear "line out" jack (for rear surround) and "mic" jack (for center and sub) - which doesn't work. (I don't have even a clue what to do to activate.) > My questions: > 1) Does 5.1 analog surround work for the ALC662? If yes, how? https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/plain/Document… Channel Mode This is an enum control to change the surround-channel setup, appears only when the surround channels are available. It gives the number of channels to be used, "2ch", "4ch", "6ch", and "8ch". According to the configuration, this also controls the jack-retasking of multi-I/O jacks. >2) If not, is it a limitation of the kernel oder alsa? >HDA-Intel.conf says: > HDA-Intel.pcm.surround51.0 cards.HDA-Intel.pcm.front.0 > Does this mean that 5.1 surround is restricted to stereo in any case? This mean softvol plugin is also used by 5.1 and stereo Do "pcm playback volume" affect the volume after you change channel mode from 2ch to 6 ch so that mic and line in jacks are retasked as output ?

2 2

[alsa-devel] [PATCH 0/6] ASoC: samsung: Spring sound support
by Andreas Färber 23 Feb '15

23 Feb '15

Hello, This series prepares sound support for the HP Chromebook 11. Regards, Andreas Andreas Färber (6): ASoC: max98088: Document DT bindings ASoC: max98088: Add DT bindings ARM: dts: Add max98089 to exynos5250-spring ASoC: samsung: Document binding for max98089 based Snow driver ASoC: samsung: Extend Snow driver to support max98089 ARM: dts: Add sound support to exynos5250-spring .../devicetree/bindings/sound/max98088.txt | 16 ++++++++++++ Documentation/devicetree/bindings/sound/snow.txt | 1 + arch/arm/boot/dts/exynos5250-spring.dts | 30 ++++++++++++++++++++++ sound/soc/codecs/max98088.c | 8 ++++++ sound/soc/samsung/Kconfig | 1 + sound/soc/samsung/snow.c | 1 + 6 files changed, 57 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98088.txt -- 2.1.4

7 29

[alsa-devel] [PATCH] ASoC: Intel: fix machine driver warnings
by yang.a.fang＠intel.com 23 Feb '15

23 Feb '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> this patch will fix below sparse warnings warning: incorrect type in argument 2 (different base types) expected unsigned int [unsigned] val got restricted snd_pcm_format_t [usertype] <noident> sound/soc/intel/haswell.c:61:37 sound/soc/intel/broadwell.c:115:37: sound/soc/intel/bytcr_dpcm_rt5640.c:118:37: sound/soc/intel/cht_bsw_rt5672.c:183:37: sound/soc/intel/cht_bsw_rt5645.c:208:37: Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/intel/broadwell.c | 2 +- sound/soc/intel/bytcr_dpcm_rt5640.c | 2 +- sound/soc/intel/cht_bsw_rt5645.c | 2 +- sound/soc/intel/cht_bsw_rt5672.c | 2 +- sound/soc/intel/haswell.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 9cf7d01..0ca4015 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -112,7 +112,7 @@ static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP0 to 16 bit */ snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S16_LE); + (__force int)SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/bytcr_dpcm_rt5640.c b/sound/soc/intel/bytcr_dpcm_rt5640.c index 5930862..b171443 100644 --- a/sound/soc/intel/bytcr_dpcm_rt5640.c +++ b/sound/soc/intel/bytcr_dpcm_rt5640.c @@ -115,7 +115,7 @@ static int byt_codec_fixup(struct snd_soc_pcm_runtime *rtd, /* 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); + (__force int)SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/cht_bsw_rt5645.c index b6f8377..7781fbe 100644 --- a/sound/soc/intel/cht_bsw_rt5645.c +++ b/sound/soc/intel/cht_bsw_rt5645.c @@ -205,7 +205,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, /* 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); + (__force int)SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index ff01662..b1842fa 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -180,7 +180,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, /* 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); + (__force int)SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/haswell.c b/sound/soc/intel/haswell.c index 35edf51..b19f1ce 100644 --- a/sound/soc/intel/haswell.c +++ b/sound/soc/intel/haswell.c @@ -58,7 +58,7 @@ static int haswell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP0 to 16 bit */ snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S16_LE); + (__force int)SNDRV_PCM_FORMAT_S16_LE); return 0; } -- 1.7.9.5

4 14

Re: [alsa-devel] Build regressions/improvements in v4.0-rc1
by Geert Uytterhoeven 23 Feb '15

23 Feb '15

On Mon, Feb 23, 2015 at 9:33 AM, Geert Uytterhoeven <geert(a)linux-m68k.org> wrote: > Below is the list of build error/warning regressions/improvements in > v4.0-rc1[1] compared to v3.19[2]. > > Summarized: > - build errors: +11/-10 > [1] http://kisskb.ellerman.id.au/kisskb/head/8494/ (256 out of 257 configs) > [2] http://kisskb.ellerman.id.au/kisskb/head/8427/ (255 out of 257 configs) > > > *** ERRORS *** > > 11 regressions: > + /home/kisskb/slave/src/arch/arm/include/asm/atomic.h: error: #error SMP not supported on pre-ARMv6 CPUs: => 137:2 > + /home/kisskb/slave/src/arch/arm/include/asm/cmpxchg.h: error: #error "SMP is not supported on this platform": => 114:2 arm-randconfig SMP does depend on CPU_V6K || CPU_V7 and on HAVE_SMP... strange... > + /home/kisskb/slave/src/fs/dax.c: error: implicit declaration of function 'copy_user_page' [-Werror=implicit-function-declaration]: => 266:2 sh-randconfig > + /home/kisskb/slave/src/sound/soc/codecs/max98357a.c: error: implicit declaration of function 'devm_gpiod_get' [-Werror=implicit-function-declaration]: => 55:2 > + /home/kisskb/slave/src/sound/soc/codecs/max98357a.c: error: implicit declaration of function 'gpiod_direction_output' [-Werror=implicit-function-declaration]: => 61:2 > + /home/kisskb/slave/src/sound/soc/codecs/max98357a.c: error: implicit declaration of function 'gpiod_set_value' [-Werror=implicit-function-declaration]: => 30:3 s390-allyesconfig s390-allmodconfig parisc-allmodconfig sh-allmodconfig sh-allyesconfig cris-allyesconfig cris-allmodconfig mips-allmodconfig > + error: asm9260_timer.c: undefined reference to `of_io_request_and_map': => .init.text+0x4014), .init.text+0x5c84) sparc64-allmodconfig sparc-allmodconfig > + error: book3s_64_vio_hv.c: undefined reference to `power7_wakeup_loss': => .text+0x408) pseries_defconfig > + error: smp.c: undefined reference to `.__flush_disable_L1': => .text+0xb24) powerpc-randconfig Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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[alsa-devel] [PATCH 0/4] ALSA: fireworks/bebob/dice/oxfw: fix kernel oops at unplugging during streaming
by Takashi Sakamoto 23 Feb '15

23 Feb '15

Each FireWire sound driver causes kernel oops at unplugging during streaming. This bug affects: - snd-bebob (in 3.16 or later) - snd-fireworks (in 3.16 or later) - snd-dice (in 3.19 or later) - snd-oxfw (in 3.19 or later) When streaming, any ALSA character devices are opened and sound card object is referred. Then 'struct snd_card.private_free()' is not executed in a context of 'struct fw_driver.remove()'. It's executed in a context of an application process in which the last ALSA character device is closed. In this case, any operation to instances of FireWire unit is invalid because it's already released, and causes kernel oops. I confirm GPF or Null-pointer-dereference. To fix this problem, this patchset do: 1.delegate reference counting of an instance of FireWire unit from AMDTP helper functions to each driver, then do it in 'struct snd_card.private_free()'. 2.move stream or transaction destructor from 'struct fw_driver.remove()' to 'struct snd_card.private_free()'. With this patchset, each driver keeps a reference to FireWire unit till all of ALSA applications close its character devices. This may be against IEEE 1394 bus driver developers' expectation to high-level drivers. Takashi Sakamoto (4): ALSA: fireworks/bebob/dice/oxfw: add reference-counting for FireWire unit ALSA: firewire-lib: remove reference counting ALSA: fireworks/bebob/dice/oxfw: allow stream destructor after releasing runtime ALSA: fireworks/bebob/dice/oxfw: make it possible to shutdown safely sound/firewire/amdtp.c | 3 +-- sound/firewire/bebob/bebob.c | 20 ++++++++++++++++---- sound/firewire/bebob/bebob_stream.c | 16 ++++------------ sound/firewire/dice/dice-stream.c | 18 ++++++++++++------ sound/firewire/dice/dice.c | 16 +++++++++++----- sound/firewire/fireworks/fireworks.c | 20 ++++++++++++++------ sound/firewire/fireworks/fireworks_stream.c | 19 ++++++++++--------- sound/firewire/oxfw/oxfw-stream.c | 6 ++++-- sound/firewire/oxfw/oxfw.c | 21 ++++++++++++++------- 9 files changed, 86 insertions(+), 53 deletions(-) -- 2.1.0

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[alsa-devel] [PATCH 0/3] Expose more controls from tegra-max98090
by Tomeu Vizoso 21 Feb '15

21 Feb '15

Hello, we need some more controls in userspace so policy can be applied at events such as microphone and headphone jacks being plugged in, to be used by Tegra-based Chromebooks. Regards, Tomeu Tomeu Vizoso (3): ASoC: tegra: Expose Headphones pin to userspace ASoC: tegra: Add sink for the internal mic to tegra_max98090 ASoC: tegra: Add control for the Mic Jack pin .../devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt | 1 + sound/soc/tegra/tegra_max98090.c | 4 ++++ 2 files changed, 5 insertions(+) -- 1.9.3

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[alsa-devel] [PATCH] ASoC: Add support for NAU8824 codec to ASoC
by Wan ZongShun 21 Feb '15

21 Feb '15

The NAU88L24 is an ultra-low power high performance audio codec designed for tablet PC, ultra portable laptops by Nuvoton, now add linux driver support for it. Signed-off-by: Wan ZongShun <linux(a)mcuos.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8824.c | 770 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8824.h | 379 ++++++++++++++++++++++ 4 files changed, 1156 insertions(+) create mode 100644 sound/soc/codecs/nau8824.c create mode 100644 sound/soc/codecs/nau8824.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 064e6c1..862b7bd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8824 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER @@ -463,6 +464,10 @@ config SND_SOC_MAX98357A config SND_SOC_MAX9850 tristate +config SND_SOC_NAU8824 + tristate "Nuvoton NAU8824 CODEC" + depends on I2C + config SND_SOC_PCM1681 tristate "Texas Instruments PCM1681 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 69b8666..acb7bda 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -68,6 +68,7 @@ snd-soc-max98357a-objs := max98357a.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8824-objs := nau8824.o snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o @@ -250,6 +251,7 @@ obj-$(CONFIG_SND_SOC_MAX98357A) += snd-soc-max98357a.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8824) += snd-soc-nau8824.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c new file mode 100644 index 0000000..80c2283 --- /dev/null +++ b/sound/soc/codecs/nau8824.c @@ -0,0 +1,770 @@ +/* + * linux/sound/soc/codecs/nau8824.c + * + * Copyright 2015 Nuvoton Technology Corp. + * Author: Meng-Huang Kuo <mhkuo(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/regmap.h> +#include <linux/acpi.h> +#include <asm/div64.h> +#include <sound/jack.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/delay.h> + +#include "nau8824.h" + +static int nau8824_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); +static int nau8824_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt); +static int nau8824_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level); +static int nau8824_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir); +static const DECLARE_TLV_DB_SCALE(out_spk_vol_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(out_hp_vol_tlv, -3000, 100, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -12800, 50, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -12800, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_left_vol_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(adc_right_vol_tlv, -12750, 50, 1); + +static const struct snd_kcontrol_new nau8824_snd_controls[] = { + /* SP Class-D mute control */ + SOC_DOUBLE("SP Playback Switch", NAU8824_CLASS_D_GAIN1, + NAU8824_CLASS_D_SFT, NAU8824_CLASS_D_SFT, 1, 0), + /* SP Class-D driver output stage gain control */ + SOC_SINGLE_TLV("SP Left Volume", NAU8824_CLASS_D_GAIN2, + NAU8824_CLASS_D_GAIN_L_SFT, + NAU8824_SPK_VOL_RSCL_RANGE, 0, out_spk_vol_tlv), + SOC_SINGLE_TLV("SP Right Volume", NAU8824_CLASS_D_GAIN1, + NAU8824_CLASS_D_GAIN_R_SFT, + NAU8824_SPK_VOL_RSCL_RANGE, 0, + out_spk_vol_tlv), + /* SP Class-D mute control */ + SOC_DOUBLE("HP Playback Switch", NAU8824_HP_MUTE, NAU8824_L_MUTE_SFT, + NAU8824_R_MUTE_SFT, 1, 1), + SOC_SINGLE_TLV("HP Left Volume", NAU8824_HP_VOL, NAU8824_L_VOL_SFT, + NAU8824_VOL_RSCL_RANGE, 1, out_hp_vol_tlv), + SOC_SINGLE_TLV("HP Right Volume", NAU8824_HP_VOL, NAU8824_R_VOL_SFT, + NAU8824_VOL_RSCL_RANGE, 1, out_hp_vol_tlv), + /* DMIC control */ + SOC_DOUBLE("DMIC L R Switch", NAU8824_ENA_CTRL, NAU8824_DMIC_CH0_SFT, + NAU8824_DMIC_CH1_SFT, 1, 0), + SOC_SINGLE("DMIC Enable", NAU8824_BIAS_ADJ, NAU8824_DMIC1_SFT, 1, 0), + SOC_SINGLE("VMID Switch", NAU8824_BIAS_ADJ, NAU8824_VMID_SFT, 1, 0), + SOC_SINGLE("BIAS Switch", NAU8824_BOOST, NAU8824_G_BIAS_SFT, 1, 0), + SOC_DOUBLE_R_TLV("MIC L R Gain", NAU8824_ADC_CH0_DGAIN_CTRL, + NAU8824_ADC_CH1_DGAIN_CTRL, 0, + NAU8824_ADC_VOL_RSCL_RANGE, 0, adc_vol_tlv), +}; + +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, NAU8824_CLK_DIVIDER, + NAU8824_CLK_DMIC_SRC_MASK, + NAU8824_CLK_DMIC_DIV_4); + break; + default: + break; + } + return 0; +} + +static const struct snd_kcontrol_new nau8824_rec_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", SND_SOC_NOPM, + 0, 0, 0), +}; + +static const struct snd_kcontrol_new nau8824_rec_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", SND_SOC_NOPM, + 0, 0, 0), +}; + +static const struct snd_kcontrol_new nau8824_spo_mix[] = { + SOC_DAPM_SINGLE("SP L Switch", SND_SOC_NOPM, + 0, 0, 0), + SOC_DAPM_SINGLE("SP R Switch", SND_SOC_NOPM, + 0, 0, 0), +}; + +static const struct snd_kcontrol_new nau8824_hpo_mix[] = { + SOC_DAPM_SINGLE("HP L Switch", + NAU8824_HPO_MIXER, NAU8824_M_HPVOL_L_HM_SFT, + 1, 0), + SOC_DAPM_SINGLE("HP R Switch", + NAU8824_HPO_MIXER, NAU8824_M_HPVOL_R_HM_SFT, + 1, 0), +}; + +static const char * const nau8824_stereo_adc1_src[] = { "ADC", "DMIC" }; + +static const SOC_ENUM_SINGLE_DECL(nau8824_stereo_adc_l1_enum, NAU8824_ENA_CTRL, + NAU8824_DMIC_CH0_SFT, nau8824_stereo_adc1_src); + +static const SOC_ENUM_SINGLE_DECL(nau8824_stereo_adc_r1_enum, NAU8824_ENA_CTRL, + NAU8824_DMIC_CH1_SFT, nau8824_stereo_adc1_src); + +static const struct snd_kcontrol_new nau8824_sto_adc_l1_mux = + SOC_DAPM_ENUM("Stereo ADC L1 source", + nau8824_stereo_adc_l1_enum); + +static const struct snd_kcontrol_new nau8824_sto_adc_r1_mux = + SOC_DAPM_ENUM("Stereo ADC R1 source", + nau8824_stereo_adc_r1_enum); + +static int nau8824_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, NAU8824_ANALOG_CTRL2, + NAU8824_CLASS_D_CLAMP_MSK, + NAU8824_CLASS_D_CLAMP_BETTER); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, NAU8824_CLASS_D_GAIN1, + NAU8824_CLASS_D_MASK, NAU8824_CLASS_D_DIS); + default: + break; + } + return 0; +} + +static int nau8824_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + return 0; +} + +static int nau8824_dac1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + return 0; +} + +static int micbias_power_on_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + if (SND_SOC_DAPM_EVENT_ON(event)) + snd_soc_update_bits(codec, NAU8824_BOOST, NAU8824_G_BIAS_MASK, + NAU8824_G_BIAS_EN); + else if (SND_SOC_DAPM_EVENT_OFF(event)) + snd_soc_update_bits(codec, NAU8824_BOOST, NAU8824_G_BIAS_MASK, + NAU8824_G_BIAS_DIS); + return 0; +} + +static int vmid_power_on_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + if (SND_SOC_DAPM_EVENT_ON(event)) + snd_soc_update_bits(codec, NAU8824_BIAS_ADJ, + NAU8824_VMID_MASK, NAU8824_VMID_EN); + else if (SND_SOC_DAPM_EVENT_OFF(event)) + snd_soc_update_bits(codec, NAU8824_BIAS_ADJ, + NAU8824_VMID_MASK, NAU8824_VMID_DIS); + return 0; +} + +static const struct snd_soc_dapm_widget nau8824_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("DMIC1"), + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + + SND_SOC_DAPM_SUPPLY("micbias", SND_SOC_NOPM, 0, 0, + micbias_power_on_event, SND_SOC_DAPM_PRE_PMU + | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_SUPPLY("vmid", SND_SOC_NOPM, 0, 0, vmid_power_on_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA("DMIC L1", NAU8824_ENA_CTRL, NAU8824_DMIC_CH0_SFT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC R1", NAU8824_ENA_CTRL, NAU8824_DMIC_CH1_SFT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, set_dmic_clk, + SND_SOC_DAPM_PRE_PMU), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", SND_SOC_NOPM, 0, 0, nau8824_rec_l_mix, + ARRAY_SIZE(nau8824_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", SND_SOC_NOPM, 0, 0, nau8824_rec_r_mix, + ARRAY_SIZE(nau8824_rec_r_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &nau8824_sto_adc_l1_mux), + SND_SOC_DAPM_MUX("Stereo ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &nau8824_sto_adc_r1_mux), + /* ADC IF1 */ + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC L1", NULL, NAU8824_DAC_CTRL, + NAU8824_DAC_L_SFT, 0, nau8824_dac1_event, + SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("DAC R1", NULL, NAU8824_DAC_CTRL, + NAU8824_DAC_R_SFT, 0, nau8824_dac1_event, + SND_SOC_DAPM_PRE_PMD), + + /* SPO/HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("SPO MIX", SND_SOC_NOPM, 0, 0, nau8824_spo_mix, + ARRAY_SIZE(nau8824_spo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX", SND_SOC_NOPM, 0, 0, nau8824_hpo_mix, + ARRAY_SIZE(nau8824_hpo_mix)), + + SND_SOC_DAPM_PGA_S("HP amp", 1, NAU8824_CLASS_G_CTRL, + NAU8824_CLASS_G_SHIFT, 0, nau8824_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("SPK amp", 1, NAU8824_CLASS_D_GAIN1, + NAU8824_CLASS_D_SFT, 0, nau8824_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8824_dapm_routes[] = { + + {"DMIC L1", NULL, "DMIC1"}, + {"DMIC R1", NULL, "DMIC1"}, + + {"BST1", NULL, "IN1P"}, + {"BST1", NULL, "IN1N"}, + {"BST2", NULL, "IN2P"}, + {"BST2", NULL, "IN2N"}, + {"IN2P", NULL, "micbias"}, + {"IN2P", NULL, "vmid"}, + + {"RECMIXL", "BST1 Switch", "BST1"}, + {"RECMIXR", "BST2 Switch", "BST2"}, + + {"ADC L", NULL, "RECMIXL"}, + {"ADC R", NULL, "RECMIXR"}, + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC CLK"}, + + {"Stereo ADC L1 Mux", "ADC", "ADC L"}, + {"Stereo ADC L1 Mux", "DMIC", "DMIC L1"}, + {"Stereo ADC R1 Mux", "ADC", "ADC R"}, + {"Stereo ADC R1 Mux", "DMIC", "DMIC R1"}, + {"AIF1TX", NULL, "IF1 ADC"}, + + {"DAC L1", NULL, "AIF1RX"}, + {"DAC R1", NULL, "AIF1RX"}, + {"SPO MIX", "SP L Switch", "DAC L1"}, + {"SPO MIX", "SP R Switch", "DAC R1"}, + {"SPK amp", NULL, "SPO MIX"}, + {"SPK", NULL, "SPK amp"}, + + {"HPO MIX", "HP L Switch", "DAC L1"}, + {"HPO MIX", "HP R Switch", "DAC R1"}, + {"HP amp", NULL, "HPO MIX"}, + {"HPOL", NULL, "HP amp"}, + {"HPOR", NULL, "HP amp"}, +}; + +static int nau8824_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *tmp) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + unsigned int val_len = 0; + + dev_dbg(codec->dev, "%s\n", __func__); + dev_dbg(codec->dev, "##- Data length: %d\n", params_format(params)); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len |= NAU8824_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= NAU8824_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S8: + case SNDRV_PCM_FORMAT_S32_LE: + val_len |= NAU8824_I2S_DL_32; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, NAU8824_I2S_PCM_CTRL_1, NAU8824_I2S_DL_MASK, + val_len); + return 0; +} + +static int nau8824_dac_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + if (mute) + snd_soc_update_bits(codec, NAU8824_DAC_MUTE_CTRL, + NAU8824_SOFT_MUTE_MASK, NAU8824_SOFT_MUTE_EN); + else + snd_soc_update_bits(codec, NAU8824_DAC_MUTE_CTRL, + NAU8824_SOFT_MUTE_MASK, NAU8824_SOFT_MUTE_DIS); + return 0; +} + +static int nau8824_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int reg_val = 0, reg_val_2 = 0; + + dev_dbg(codec->dev, "%s: Entered\n", __func__); + dev_dbg(codec->dev, "###nau8824_set_dai_fmt %x\n", fmt); + dev_dbg(codec->dev, "##+ nau8824_set_dai_fmt (%x)\n", fmt); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + reg_val_2 |= NAU8824_I2S_MS_M; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBS_CFM: + break; + default: + dev_alert(codec->dev, "Invalid DAI master/slave interface\n"); + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= NAU8824_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + reg_val |= NAU8824_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= NAU8824_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + reg_val |= NAU8824_I2S_DF_RIGHT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= NAU8824_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= NAU8824_I2S_DF_PCM_B; + reg_val |= NAU8824_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, NAU8824_I2S_PCM_CTRL_1, + NAU8824_I2S_DL_MASK | NAU8824_I2S_DF_MASK + | NAU8824_I2S_BP_MASK | NAU8824_I2S_PCMB_MASK, + reg_val); + snd_soc_update_bits(codec, NAU8824_I2S_PCM_CTRL_2, NAU8824_I2S_MS_MASK, + reg_val_2); + + dev_dbg(codec->dev, "##-nau8824_set_dai_fmt Master\n"); + dev_dbg(codec->dev, "%s: Exiting\n", __func__); + return 0; +} + +static int nau8824_FLL_freerun_mode(struct snd_soc_codec *codec, int IsEnable) +{ + if (IsEnable == true) { + snd_soc_write(codec, 0x03, 0x0853); + snd_soc_write(codec, 0x09, 0xE000); + snd_soc_write(codec, 0x03, 0x8853); + } else { + snd_soc_write(codec, 0x03, 0x0853); + snd_soc_write(codec, 0x09, 0x6000); + snd_soc_write(codec, 0x03, 0x8853); + } + return 0; +} + +void set_sys_clk(struct snd_soc_codec *codec, int sys_clk) +{ + pr_debug("%s sys_clk=%x\n", __func__, sys_clk); + switch (sys_clk) { + case NAU8824_INTERNALCLOCK: + nau8824_FLL_freerun_mode(codec, true); + break; + + case NAU8824_MCLK: + default: + nau8824_FLL_freerun_mode(codec, false); + break; + } +} + +static int nau8824_set_dai_pll(struct snd_soc_dai *dai, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "In nau8824: dai_set_pll\n"); + return 0; +} + +static int nau8824_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + int divider = 1; + + if (clk_id == NAU8824_MCLK) { + set_sys_clk(codec, NAU8824_MCLK); + dev_dbg(codec->dev, "%s: input freq = %d divider = %d", + __func__, freq, divider); + + } else if (clk_id == NAU8824_INTERNALCLOCK) { + set_sys_clk(codec, NAU8824_INTERNALCLOCK); + } else { + dev_err(codec->dev, "Wrong clock src\n"); + return -EINVAL; + } + return 0; +} + +static int nau8824_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + dev_dbg(codec->dev, "## nau8824_set_bias_level %d\n", level); + if (level == codec->dapm.bias_level) { + dev_dbg(codec->dev, "##set_bias_level: level returning...\r\n"); + return -EINVAL; + } + + switch (level) { + case SND_SOC_BIAS_ON: + /* All power is driven by DAPM system*/ + dev_dbg(codec->dev, "###nau8824_set_bias_level BIAS_ON\n"); + snd_soc_update_bits(codec, NAU8824_BIAS_ADJ, + NAU8824_VMID_MASK, NAU8824_VMID_EN); + snd_soc_update_bits(codec, NAU8824_BOOST, + NAU8824_G_BIAS_MASK, NAU8824_G_BIAS_EN); + break; + + case SND_SOC_BIAS_PREPARE: + dev_dbg(codec->dev, "###nau8824_set_bias_level BIAS_PREPARE\n"); + break; + + case SND_SOC_BIAS_STANDBY: + dev_dbg(codec->dev, "###nau8824_set_bias_level STANDBY\n"); + break; + + case SND_SOC_BIAS_OFF: + dev_dbg(codec->dev, "###nau8824_set_bias_level OFF\n"); + set_sys_clk(codec, NAU8824_INTERNALCLOCK); + snd_soc_update_bits(codec, NAU8824_BIAS_ADJ, NAU8824_VMID_MASK, + NAU8824_VMID_DIS); + snd_soc_update_bits(codec, NAU8824_BOOST, + NAU8824_G_BIAS_MASK, NAU8824_G_BIAS_DIS); + break; + default: + break; + } + codec->dapm.bias_level = level; + dev_dbg(codec->dev, "## nau8824_set_bias_level %d\n", level); + + return 0; +} + +static int nau8824_suspend(struct snd_soc_codec *codec) +{ + dev_dbg(codec->dev, "%s: Entered\n", __func__); + nau8824_set_bias_level(codec, SND_SOC_BIAS_OFF); + dev_dbg(codec->dev, "%s: Exiting\n", __func__); + return 0; +} + +static int nau8824_resume(struct snd_soc_codec *codec) +{ + dev_dbg(codec->dev, "%s: Entered\n", __func__); + nau8824_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + dev_dbg(codec->dev, "%s: Exiting\n", __func__); + return 0; +} + +struct nau8824_init_reg { + u8 reg; + u16 val; +}; + +static struct nau8824_init_reg init_list[] = { + {0x0000, 0x0001}, + {0x0000, 0x0000}, + {0x0066, 0x0060}, + {0x0076, 0x3000}, + {0x0004, 0x0002}, + {0x0005, 0xDD44}, + {0x0006, 0x0007}, + {0x0007, 0x0810}, + {0x0008, 0xC000}, + {0x0074, 0x1106}, + {0x0073, 0x330C}, + {0x0080, 0x0000}, + {0x006D, 0x0100}, + {0x006E, 0x0001}, + {0x006A, 0x0008}, + {0x006D, 0x0180}, + {0x006D, 0x1480}, + {0x006E, 0x0014}, + {0x0050, 0x2007}, + {0x0080, 0x3020}, + {0x006B, 0xC000}, + {0x007B, 0x1E1E}, + {0x006B, 0xC005}, + {0x0068, 0x8000}, + {0x0076, 0x300F}, + {0x007F, 0x000F}, + {0x0080, 0x0020}, + {0x006B, 0x0005}, + {0x007B, 0x0000}, + {0x0002, 0x0800}, + {0x0025, 0x1000}, + {0x0024, 0x0002}, + {0x0002, 0x0AA5}, + {0x0020, 0x0000}, + {0x0025, 0x1082}, + {0x0031, 0x0800}, + {0x0032, 0x0100}, + {0x0033, 0x0300}, + {0x0034, 0x0000}, + {0x001C, 0x000A}, + {0x001F, 0x2000}, + {0x0001, 0x0030}, + {0x0002, 0x3AA5}, + {0x0067, 0x4040}, + {0x007F, 0x300F}, + {0x0077, 0x0010}, + {0x0078, 0x3C00}, + {0x0076, 0x380F}, + {0x0076, 0x300F}, + {0x007A, 0x0202}, + {0x0067, 0x0000}, + {0x0072, 0x40A0}, + {0x0079, 0x0300}, + {0x0023, 0x1010}, + {0x0024, 0x0002}, + {0x0038, 0x1486}, + {0x003C, 0x1486}, + {0x0066, 0x0064}, + {0x0001, 0x40F0}, + {0x0001, 0x48F0}, + {0x0025, 0x3082}, + {0x0020, 0x0003}, + {0x0024, 0x8102}, + {0x002D, 0x0108}, + {0x002E, 0x0308}, + {0x002F, 0x0500}, + {0x0030, 0x0700}, + {0x001C, 0x000A}, + {0x001F, 0x2000}, + {0x0002, 0x3BA5}, + {0x0001, 0x48F3}, + {0x0002, 0xFBA5}, + {0x0014, 0x2210}, + {0x007c, 0x1e1e}, +}; + +#define NAU8824_INIT_REG_LEN ARRAY_SIZE(init_list) +static int nau8824_reg_init(struct snd_soc_codec *codec) +{ + int i; + + mdelay(1); + for (i = 0; i < NAU8824_INIT_REG_LEN; i++) { + if (init_list[i].reg == 0xFFFF) + mdelay(1); + else + snd_soc_write(codec, init_list[i].reg, + init_list[i].val); + } + return 0; +} + +static int nau8824_codec_probe(struct snd_soc_codec *codec) +{ + int ret = 0; + struct nau8824_priv *nau8824; + + nau8824 = snd_soc_codec_get_drvdata(codec); + codec->control_data = nau8824->regmap; + mutex_init(&nau8824->mutex); + nau8824_reg_init(codec); + + /* Dynamic Headset detection enabled */ + snd_soc_update_bits(codec, 0x01, 0x400, 0x400); + snd_soc_update_bits(codec, 0x02, 0x0008, 0x0008); + snd_soc_update_bits(codec, 0x0f, 0x0300, 0x0100); + snd_soc_write(codec, 0x09, 0xE000); + snd_soc_write(codec, NAU8824_IRQ_SETTING, 0x1006); + snd_soc_write(codec, 0x13, 0x1615); + snd_soc_write(codec, 0x15, 0x0414); + snd_soc_update_bits(codec, 0x16, 0xFF00, 0x5900); + snd_soc_update_bits(codec, 0x66, 0x0070, 0x0060); + + /* Program codec to use internal clock */ + nau8824_FLL_freerun_mode(codec, true); + return ret; +} + +static int nau8824_codec_remove(struct snd_soc_codec *codec) +{ + nau8824_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_driver_nau8824 = { + .probe = nau8824_codec_probe, + .remove = nau8824_codec_remove, + .suspend = nau8824_suspend, + .resume = nau8824_resume, + .set_bias_level = nau8824_set_bias_level, + .controls = nau8824_snd_controls, + .num_controls = ARRAY_SIZE(nau8824_snd_controls), + .dapm_widgets = nau8824_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8824_dapm_widgets), + .dapm_routes = nau8824_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8824_dapm_routes), + .reg_word_size = sizeof(u16), + .reg_cache_default = init_list, + .set_sysclk = nau8824_set_sysclk, +}; + +static struct snd_soc_dai_ops nau8824_dai_ops = { + .hw_params = nau8824_hw_params, + .set_pll = nau8824_set_dai_pll, + .set_fmt = nau8824_set_dai_fmt, + .digital_mute = nau8824_dac_mute, +}; + +static struct snd_soc_dai_driver nau8824_dai_driver[] = { +{ + .name = "nau8824-aif1", + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8824_RATES, + .formats = NAU8824_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8824_RATES, + .formats = NAU8824_FORMATS, + }, + .ops = &nau8824_dai_ops, +} +}; + +static int nau8824_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct nau8824_priv *nau8824; + int ret; + + pr_debug("%s enter", __func__); + nau8824 = kzalloc(sizeof(*nau8824), GFP_KERNEL); + if (nau8824 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, nau8824); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_nau8824, + nau8824_dai_driver, + ARRAY_SIZE(nau8824_dai_driver)); + if (ret < 0) + kfree(nau8824); + + return ret; +} + +static int nau8824_i2c_remove(struct i2c_client *i2c) +{ + struct nau8824_priv *nau8824 = dev_get_drvdata(&i2c->dev); + + snd_soc_unregister_codec(nau8824->dev); + + return 0; +} + +static const struct i2c_device_id nau8824_i2c_id[] = { + { "nau8824", 0}, + {"10508824:00", 0}, + {"10508824", 0}, + { } +}; + +MODULE_DEVICE_TABLE(i2c, nau8824_i2c_id); +static struct i2c_driver nau8824_i2c_driver = { + .driver = { + .name = "nau8824", + .owner = THIS_MODULE, + }, + .probe = nau8824_i2c_probe, + .remove = (nau8824_i2c_remove), + .id_table = nau8824_i2c_id, +}; +module_i2c_driver(nau8824_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8824 codec driver"); +MODULE_AUTHOR("Nuvoton"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/codecs/nau8824.h b/sound/soc/codecs/nau8824.h new file mode 100644 index 0000000..1745dac --- /dev/null +++ b/sound/soc/codecs/nau8824.h @@ -0,0 +1,379 @@ +/* + * linux/sound/soc/codecs/nau8824.h + * + * Copyright 2015 Nuvoton Technology Corp. + * Author: Meng-Huang Kuo <mhkuo(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _NAU8824_H +#define _NAU8824_H + +#define NAU8824_RESET 0x00 +#define NAU8824_ENA_CTRL 0x01 +#define NAU8824_CLK_EN 0x02 +#define NAU8824_CLK_DIVIDER 0x03 +#define NAU8824_FL_1 0x04 +#define NAU8824_FL_2 0x05 +#define NAU8824_FL_3 0x06 +#define NAU8824_FL_4 0x07 +#define NAU8824_FL_5 0x08 +#define NAU8824_FL_6 0x09 +#define NAU8824_IRQ_STATUS 0x10 +#define NAU8824_IRQ_CLEAR 0x11 +#define NAU8824_IRQ_SETTING 0x12 +#define NAU8824_SAR_ADC 0x13 +#define NAU8824_VDET_COEFFICIENT 0x14 +#define NAU8824_I2S_PCM_CTRL_1 0x1C +#define NAU8824_I2S_PCM_CTRL_2 0x1D +#define NAU8824_ADC_CH0_DGAIN_CTRL 0x2D +#define NAU8824_ADC_CH1_DGAIN_CTRL 0x2E +#define NAU8824_DAC_MUTE_CTRL 0x31 +#define NAU8824_DAC0_DIG_VOL 0x32 +#define NAU8824_DAC1_DIG_VOL 0x33 +#define NAU8824_CLASS_G_CTRL 0x50 +#define NAU8824_SAR_ADC_OUTPUT 0x59 +#define NAU8824_BIAS_ADJ 0x66 +#define NAU8824_ANALOG_CTRL2 0x6A +#define NAU8824_HPO_MIXER 0x6B +#define NAU8824_CLASS_D_GAIN1 0x6D +#define NAU8824_CLASS_D_GAIN2 0x6E +#define NAU8824_DAC_CTRL 0x73 +#define NAU8824_MIC_BIAS 0x74 +#define NAU8824_HP_MUTE 0x75 +#define NAU8824_BOOST 0x76 +#define NAU8824_HP_VOL 0x7B + +/* reg. NAU8824_ENA_CTRL (0x01) */ +#define NAU8824_DMIC_EDGE_CH23_MASK (0x01 << 15) +#define NAU8824_DMIC_EDGE_CH23_SFT 15 +#define NAU8824_DMIC_EDGE_CH01_MASK (0x01 << 14) +#define NAU8824_DMIC_EDGE_CH01_SFT 14 +#define NAU8824_ADC_OP_EN_MASK (0x01 << 11) +#define NAU8824_ADC_OP_EN_SFT 11 +#define NAU8824_ADC_OP_EN (0x01 << 11) +#define NAU8824_ADC_OP_DIS (0x00 << 11) +#define NAU8824_INT_SLEEP_MASK (0x01 << 10) +#define NAU8824_INT_SLEEP_SFT 10 +#define NAU8824_INT_SLEEP_ON (0x01 << 10) +#define NAU8824_INT_SLEEP_OFF (0x00 << 10) +#define NAU8824_DMIC_CH3_MASK (0x01 << 9) +#define NAU8824_DMIC_CH3_SFT 9 +#define NAU8824_DMIC_CH3_ON (0x01 << 9) +#define NAU8824_DMIC_CH3_OFF (0x00 << 9) +#define NAU8824_DMIC_CH2_MASK (0x01 << 8) +#define NAU8824_DMIC_CH2_SFT 8 +#define NAU8824_DMIC_CH2_ON (0x01 << 8) +#define NAU8824_DMIC_CH2_OFF (0x00 << 8) +#define NAU8824_DMIC_CH1_MASK (0x01 << 7) +#define NAU8824_DMIC_CH1_SFT 7 +#define NAU8824_DMIC_CH1_ON (0x01 << 7) +#define NAU8824_DMIC_CH1_OFF (0x00 << 7) +#define NAU8824_DMIC_CH0_MASK (0x01 << 6) +#define NAU8824_DMIC_CH0_SFT 6 +#define NAU8824_DMIC_CH0_ON (0x01 << 6) +#define NAU8824_DMIC_CH0_OFF (0x00 << 6) +#define NAU8824_DAC_CH1_EN_MASK (0x01 << 5) +#define NAU8824_DAC_CH1_EN_SFT 5 +#define NAU8824_DAC_CH1_EN (0x01 << 5) +#define NAU8824_DAC_CH1_DIS (0x00 << 5) +#define NAU8824_DAC_CH0_EN_MASK (0x01 << 4) +#define NAU8824_DAC_CH0_EN_SFT 4 +#define NAU8824_DAC_CH0_EN (0x01 << 4) +#define NAU8824_DAC_CH0_DIS (0x00 << 4) +#define NAU8824_ADC_CH3_EN_MASK (0x01 << 3) +#define NAU8824_ADC_CH3_EN_SFT 3 +#define NAU8824_ADC_CH3_EN (0x01 << 3) +#define NAU8824_ADC_CH3_DIS (0x00 << 3) +#define NAU8824_ADC_CH2_EN_MASK (0x01 << 2) +#define NAU8824_ADC_CH2_EN_SFT 2 +#define NAU8824_ADC_CH2_EN (0x01 << 2) +#define NAU8824_ADC_CH2_DIS (0x00 << 2) +#define NAU8824_ADC_CH1_EN_MASK (0x01 << 1) +#define NAU8824_ADC_CH1_EN_SFT 1 +#define NAU8824_ADC_CH1_EN (0x01 << 1) +#define NAU8824_ADC_CH1_DIS (0x00 << 1) +#define NAU8824_ADC_CH0_EN_MASK (0x01 << 0) +#define NAU8824_ADC_CH0_EN_SFT 0 +#define NAU8824_ADC_CH0_EN (0x01 << 0) +#define NAU8824_ADC_CH0_DIS (0x00 << 0) + +/* reg. NAU8824_CLK_EN (0x02) */ +#define NAU8824_CLK_SAR_EN_MASK (0x1 << 3) +#define NAU8824_CLK_SAR_EN_SFT 3 +#define NAU8824_CLK_SAR_EN (0x1 << 3) +#define NAU8824_CLK_SAR_DIS (0x0 << 3) + +/* reg. NAU8824_CLK_DIVIDER (0x03) */ +#define NAU8824_SYSCLK_SRC_MASK (0x1 << 15) +#define NAU8824_SYSCLK_SRC_SFT 15 +#define NAU8824_SYSCLK_SRC_VCO (0x1 << 15) +#define NAU8824_SYSCLK_SRC_MCLK (0x0 << 15) +#define NAU8824_CLK_CODEC_SRC_MASK (0x1 << 13) +#define NAU8824_CLK_CODEC_SRC_SFT 13 +#define NAU8824_CODEC_SRC_SYSCLK (0x1 << 13) +#define NAU8824_CODEC_SRC_MCLK (0x0 << 13) +#define NAU8824_CLK_DMIC_SRC_MASK (0x7 << 10) +#define NAU8824_CLK_DMIC_SRC_SFT 10 +#define NAU8824_CLK_DMIC_DIV_4 (0x2 << 10) + +/* reg. NAU8824_IRQ_STATUS (0x10) */ +#define NAU8824_IRQ_KEY_RELEASE (0x1 << 5) +#define NAU8824_IRQ_LONGKEY (0x1 << 4) +#define NAU8824_IRQ_SHORTKEY (0x1 << 3) +#define NAU8824_IRQ_MIC_INSERTED (0x1 << 2) +#define NAU8824_IRQ_JACK_EJECTED (0x1 << 1) +#define NAU8824_IRQ_JACK_INSERTED (0x1 << 0) + +/* reg. NAU8824_IRQ_CLEAR (0x11) */ +#define NAU8824_IRQ_KEY_RELEASE_CLR (0x1 << 5) +#define NAU8824_IRQ_LONGKEY_CLR (0x1 << 4) +#define NAU8824_IRQ_SHORTKEY_CLR (0x1 << 3) +#define NAU8824_IRQ_MIC_INSERTED_CLR (0x1 << 2) +#define NAU8824_IRQ_JACK_EJECTED_CLR (0x1 << 1) +#define NAU8824_IRQ_JACK_INSERTED_CLR (0x1 << 0) + +/* reg NAU8824_SAR_ADC (0x13) */ +#define NAU8824_SAR_EN_MASK (0x1 << 12) +#define NAU8824_SAR_EN_SFT 12 +#define NAU8824_SAR_EN (0x1 << 12) +#define NAU8824_SAR_DIS (0x0 << 12) + +/* reg. NAU8824_I2S_PCM_CTRL_1 (0x1C) */ +#define NAU8824_I2S_BP_MASK (0x1 << 7) +#define NAU8824_I2S_BP_SFT 7 +#define NAU8824_I2S_BP_NOR (0x0 << 7) +#define NAU8824_I2S_BP_INV (0x1 << 7) +#define NAU8824_I2S_PCMB_MASK (0x1 << 6) +#define NAU8824_I2S_PCMB_SFT 6 +#define NAU8824_I2S_PCMB_EN (0x1 << 6) +#define NAU8824_I2S_PCMB_DIS (0x0 << 6) +#define NAU8824_I2S_DL_MASK (0x3 << 2) +#define NAU8824_I2S_DL_SFT 2 +#define NAU8824_I2S_DF_MASK (0x3) +#define NAU8824_I2S_DF_SFT 0 +#define NAU8824_I2S_DF_RIGHT (0x0) +#define NAU8824_I2S_DF_LEFT (0x1) +#define NAU8824_I2S_DF_I2S (0x2) +#define NAU8824_I2S_DF_PCM_A (0x3) +#define NAU8824_I2S_DF_PCM_B (0x3) + +/* reg. NAU8824_I2S_PCM_CTRL_2 (0x1D) */ +#define NAU8824_I2S_BCLKDIV_MASK 7 +#define NAU8824_I2S_BCLKDIV_SFT 0 + +#define NAU8824_I2S_MS_MASK (0x1 << 3) +#define NAU8824_I2S_MS_SFT 3 +#define NAU8824_I2S_MS_M (0x1 << 3) +#define NAU8824_I2S_MS_S (0x0 << 3) + +/* reg. NAU8824_DAC_MUTE_CTRL (0x31) */ +#define NAU8824_SOFT_MUTE_MASK (0x1 << 13) +#define NAU8824_SOFT_MUTE_SFT 13 +#define NAU8824_SOFT_MUTE_EN (0x1 << 13) +#define NAU8824_SOFT_MUTE_DIS (0x0 << 13) + +/* reg NAU8824_DAC0_DIG_VOL (0x32) */ +#define NAU8824_DAC0_GAIN_MASK (0x1FF << 0) + +/* reg NAU8824_DAC1_DIG_VOL (0x33) */ +#define NAU8824_DAC1_GAIN_MASK (0x1FF << 0) + +/* reg. NAU8824_CLASS_G_CTRL (0x50) */ +#define NAU8824_CLASS_G_CLK_SRC_MSK (0x3 << 14) +#define NAU8824_CLASS_G_CLK_DIS (0x3 << 14) +#define NAU8824_CLASS_G_CLK_SRC_MCLK (0x2 << 14) +#define NAU8824_CLASS_G_CLK_SRC_660K (0x1 << 14) +#define NAU8824_CLASS_G_CLK_SRC_2M (0x0 << 14) +#define NAU8824_CLASS_G_TIMER_MSK (0x3F << 8) +#define NAU8824_CLASS_G_TIMER_SFT 8 +#define NAU8824_CLASS_G_THRSLD_MSK (0x3 << 4) +#define NAU8824_CLASS_G_THRSLD_SFT 4 +#define NAU8824_CLASS_G_RIGHT_EN_MSK (0x1 << 2) +#define NAU8824_CLASS_G_RIGHT_EN (0x1 << 2) +#define NAU8824_CLASS_G_RIGHT_DIS (0x0 << 2) +#define NAU8824_CLASS_G_LEFT_EN_MSK (0x1 << 1) +#define NAU8824_CLASS_G_LEFT_EN (0x1 << 1) +#define NAU8824_CLASS_G_LEFT_DIS (0x0 << 1) +#define NAU8824_CLASS_G_EN_MSK (0x1 << 0) +#define NAU8824_CLASS_G_SHIFT 0 +#define NAU8824_CLASS_G_EN (0x1 << 0) +#define NAU8824_CLASS_G_DIS (0x0 << 0) + +/* reg NAU8824_SAR_ADC_OUTPUT (0x59) */ +#define NAU8824_ADC_OUTPUT_MASK (0xFF << 0) +#define NAU8824_ADC_OUTPUT_SFT 0 + +/* reg. NAU8824_BIAS_ADJ_2 (0x66) */ +#define NAU8824_VMID_MASK (0x1 << 6) +#define NAU8824_VMID_SFT 6 +#define NAU8824_VMID_EN (0x1 << 6) +#define NAU8824_VMID_DIS (0x0 << 6) +#define NAU8824_VMID_OPTION_MASK (0x1 << 4) +#define NAU8824_VMID_OPTION_SFT 4 +#define NAU8824_DMIC2_MASK (0x1 << 3) +#define NAU8824_DMIC2_SFT 3 +#define NAU8824_DMIC2_EN (0x1 << 3) +#define NAU8824_DMIC2_DIS (0x0 << 3) +#define NAU8824_DMIC1_MASK (0x1 << 2) +#define NAU8824_DMIC1_SFT 2 +#define NAU8824_DMIC1_EN (0x1 << 2) +#define NAU8824_DMIC1_DIS (0x0 << 2) + +/* reg. NAU8824_ANALOG_CTRL2 (0x6A) */ +#define NAU8824_CLASS_D_CLAMP_MSK (0x1 << 3) +#define NAU8824_CLASS_D_CLAMP_SFT 3 +#define NAU8824_CLASS_D_CLAMP_BETTER (0x1 << 3) +#define NAU8824_CLASS_D_CLAMP_DEFAULT (0x0 << 3) + +/* reg. NAU8824_HPO_MIXER (0x6B) */ +#define NAU8824_M_HPVOL_R_HM (0x1 << 2) +#define NAU8824_M_HPVOL_R_HM_SFT 2 +#define NAU8824_M_HPVOL_L_HM (0x1 << 0) +#define NAU8824_M_HPVOL_L_HM_SFT 0 + +/* reg. NAU8824_CLASS_D_GAIN1 (0x6D) */ +#define NAU8824_CLASS_D_GAIN_R_MSK (0x1F << 8) +#define NAU8824_CLASS_D_GAIN_R_SFT 8 +#define NAU8824_CLASS_D_MASK (0x1 << 7) +#define NAU8824_CLASS_D_SFT 7 +#define NAU8824_CLASS_D_EN_MSK (0x1 << 7) +#define NAU8824_CLASS_D_EN (0x1 << 7) +#define NAU8824_CLASS_D_DIS (0x0 << 7) + +/* reg. NAU8824_CLASS_D_GAIN2 (0x6E) */ +#define NAU8824_CLASS_D_GAIN_L_MSK (0x1F << 0) +#define NAU8824_CLASS_D_GAIN_L_SFT 0 + +/* reg. NAU8824_DAC_CTRL (0x73) */ +#define NAU8824_DAC_R_MASK (0x1 << 13) +#define NAU8824_DAC_R_SFT 13 +#define NAU8824_DAC_R_EN (0x1 << 13) +#define NAU8824_DAC_R_DIS (0x0 << 13) +#define NAU8824_DAC_L_MASK (0x1 << 12) +#define NAU8824_DAC_L_SFT 12 +#define NAU8824_DAC_L_EN (0x1 << 12) +#define NAU8824_DAC_L_DIS (0x0 << 12) + +/* reg. NAU8824_MIC_BIAS (0x74) */ +#define NAU8824_BIAS_POWER_MASK (0x1 << 8) +#define NAU8824_BIAS_POWER_SFT 8 +#define NAU8824_BIAS_POWER_ON (0x1 << 8) +#define NAU8824_BIAS_POWER_DOWN (0x0 << 8) +#define NAU8824_BIAS_LEVEL_MASK (0x7 << 0) +#define NAU8824_BIAS_LEVEL_SFT 0 +#define NAU8824_BIAS_LEVEL_VDDA (0x0 << 0) + +/* reg. NAU8824_HP_MUTE (0x75) */ +#define NAU8824_R_MUTE (0x1 << 14) +#define NAU8824_R_MUTE_SFT 14 +#define NAU8824_L_MUTE (0x1 << 6) +#define NAU8824_L_MUTE_SFT 6 + +/* reg. NAU8824_BOOST (0x76) */ +#define NAU8824_VMID_PRECHARGE_MASK (0x1 << 13) +#define NAU8824_VMID_PRECHARGE_SFT 13 +#define NAU8824_VMID_PRECHARGE_DIS (0x1 << 13) +#define NAU8824_VMID_PRECHARGE_EN (0x0 << 13) +#define NAU8824_G_BIAS_MASK (0x1 << 12) +#define NAU8824_G_BIAS_SFT 12 +#define NAU8824_G_BIAS_EN (0x1 << 12) +#define NAU8824_G_BIAS_DIS (0x0 << 12) + +/* reg. NAU8824_HP_VOL (0x7B) */ +#define NAU8824_R_VOL_MASK (0x1f << 8) +#define NAU8824_R_VOL_SFT 8 +#define NAU8824_SPK_L_VOL_MASK (0x1f) +#define NAU8824_SPK_R_VOL_MASK (0x1f << 8) +#define NAU8824_L_VOL_MASK (0x1f) +#define NAU8824_L_VOL_SFT 0 + +/* Volume Rescale */ +#define NAU8824_VOL_RSCL_MAX 0x1E +#define NAU8824_VOL_RSCL_RANGE 0x1E +#define NAU8824_SPK_VOL_RSCL_MAX 0x19 +#define NAU8824_SPK_VOL_RSCL_RANGE 0x19 +#define NAU8824_DAC_VOL_RSCL_RANGE 0x164 +#define NAU8824_ADC_VOL_RSCL_RANGE 0x164 + +/* Data Format */ +#define NAU8824_I2S_DL_32 (0x3 << 2) +#define NAU8824_I2S_DL_24 (0x2 << 2) +#define NAU8824_I2S_DL_20 (0x1 << 2) +#define NAU8824_I2S_DL_16 (0x0 << 2) + +#define NAU8824_FREQ_25000000 25000000 +#define NAU8824_PLL_CLKIN_MCLK 0 +#define NAU8824_BUTTONPRESS_MASK 0x20 +#define NAU8824_HSPLUG_MASK 0x10 + +enum { + NAU8824_INTERNALCLOCK = 0, + NAU8824_MCLK, +}; + +enum { + NAU8824_J_IN_EVENT, + NAU8824_J_OUT_EVENT, + NAU8824_BP_EVENT, + NAU8824_BR_EVENT, + NAU8824_UN_EVENT, +}; + +enum { + S_NAU8824_J_OUT, + S_NAU8824_J_IN_TEMP, + S_NAU8824_J_IN, + S_NAU8824_BUTTON_DECT, + S_NAU8824_NO_BUTTON_DECT, +}; + +#define NAU8824_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8824_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct nau8824_jack_data { + struct snd_soc_jack *jack; + int report; +}; + +struct nau8824_gpio_setup { + unsigned int reg; + u8 value; +}; + +struct nau8824_pdata { + unsigned int audio_mclk1; + unsigned int gpio_irq; + int naudint_irq; + int headset_detect; + int button_press_detect; +}; + +struct nau8824_priv { + struct nau8824_jack_data hs_jack; + struct workqueue_struct *workqueue; + struct delayed_work delayed_work; + struct snd_soc_codec *codec; + u8 i2c_regs_status; + struct device *dev; + struct regmap *regmap; + struct mutex mutex; + struct nau8824_pdata pdata; + unsigned int irq; + bool jd_status; + bool bp_status; + int jack_type; +}; + +int nau8824_device_init(struct nau8824_priv *nau8824); +void nau8824_device_exit(struct nau8824_priv *nau8824); +void nau8824_enable_mic_bias(struct snd_soc_codec *codec, int enable); +int nau8824_query_jack_status(struct snd_soc_codec *codec); +int nau8824_query_btn_press(struct snd_soc_codec *codec); +void nau8824_btn_press_intr_enable(struct snd_soc_codec *codec, int enable); + +#endif /* _NAU8824_H */ -- 1.9.1 --- 此电子邮件没有病毒和恶意软件，因为 avast! 防病毒保护处于活动状态。 http://www.avast.com

1 0

[alsa-devel] [PATCH v2] ASoC: rt5677: Add the chip type to distinguish the setting of the clock source
by Oder Chiou 21 Feb '15

21 Feb '15

There is only one clock source in the rt5676, so the chip type is added to distinguish the setting of the clock source in the VAD function. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 28 ++++++++++++++++++++++------ sound/soc/codecs/rt5677.h | 6 ++++++ 2 files changed, 28 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5d0bb87..ab62777 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -718,11 +718,24 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) RT5677_LDO1_SEL_MASK, 0x0); regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, RT5677_PWR_LDO1, RT5677_PWR_LDO1); - regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, - RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); - regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, - RT5677_PLL2_PR_SRC_MASK | RT5677_DSP_CLK_SRC_MASK, - RT5677_PLL2_PR_SRC_MCLK2 | RT5677_DSP_CLK_SRC_BYPASS); + switch (rt5677->type) { + case RT5677: + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, + RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_PLL2_PR_SRC_MASK | + RT5677_DSP_CLK_SRC_MASK, + RT5677_PLL2_PR_SRC_MCLK2 | + RT5677_DSP_CLK_SRC_BYPASS); + break; + case RT5676: + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_DSP_CLK_SRC_MASK, + RT5677_DSP_CLK_SRC_BYPASS); + break; + default: + break; + } regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x07ff); regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07fd); rt5677_set_dsp_mode(codec, true); @@ -4733,7 +4746,8 @@ static const struct regmap_config rt5677_regmap = { }; static const struct i2c_device_id rt5677_i2c_id[] = { - { "rt5677", 0 }, + { "rt5677", RT5677 }, + { "rt5676", RT5676 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); @@ -4850,6 +4864,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5677); + rt5677->type = id->driver_data; + if (pdata) rt5677->pdata = *pdata; diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index c0a625f..07df96b 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1665,6 +1665,11 @@ enum { RT5677_IRQ_JD3, }; +enum rt5677_type { + RT5677, + RT5676, +}; + struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; @@ -1681,6 +1686,7 @@ struct rt5677_priv { int pll_in; int pll_out; int pow_ldo2; /* POW_LDO2 pin */ + enum rt5677_type type; #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; #endif -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v2 0/3] Expose more controls from tegra-max98090
by Tomeu Vizoso 21 Feb '15

21 Feb '15

v2: * Added ack tags from Stephen Hello, we need some more controls in userspace so policy can be applied at events such as microphone and headphone jacks being plugged in, to be used by Tegra-based Chromebooks. Regards, Tomeu Tomeu Vizoso (3): ASoC: tegra: Expose Headphones pin to userspace ASoC: tegra: Add sink for the internal mic to tegra_max98090 ASoC: tegra: Add control for the Mic Jack pin .../devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt | 1 + sound/soc/tegra/tegra_max98090.c | 4 ++++ 2 files changed, 5 insertions(+) -- 1.9.3

2 4

[alsa-devel] [PATCH] ASoC: sta32x: fix register range in regmap.
by Thomas Niederprüm 21 Feb '15

21 Feb '15

The STA32X_AUTO3 is a writable register that currently does not appear in the regmap ranges(neither read nor write). By adding this register to the register ranges there is no gap anymore and the existing register ranges can be joined. This fixes a regression that made it impossible to choose the preset EQ mode set through the STA32X_AUTO3 register. Signed-off-by: Thomas Niederprüm <niederp(a)physik.uni-kl.de> --- I realized that I missed one register in the DT enabling patchset, so here I send a patch to clean up my mess :) Thomas sound/soc/codecs/sta32x.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index bc6edb1..d320227 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -107,13 +107,11 @@ static const struct reg_default sta32x_regs[] = { }; static const struct regmap_range sta32x_write_regs_range[] = { - regmap_reg_range(STA32X_CONFA, STA32X_AUTO2), - regmap_reg_range(STA32X_C1CFG, STA32X_FDRC2), + regmap_reg_range(STA32X_CONFA, STA32X_FDRC2), }; static const struct regmap_range sta32x_read_regs_range[] = { - regmap_reg_range(STA32X_CONFA, STA32X_AUTO2), - regmap_reg_range(STA32X_C1CFG, STA32X_STATUS), + regmap_reg_range(STA32X_CONFA, STA32X_STATUS), }; static const struct regmap_range sta32x_volatile_regs_range[] = { -- 2.1.1

2 1

[alsa-devel] [Patch V3 1/2] ASoC: max98357a: Add missing header files
by Kenneth Westfield 21 Feb '15

21 Feb '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> Add missing header files to avoid implicit declarations and indirect inclusions. Signed-off-by: Kenneth Westfield <kwestfie(a)codeaurora.org> --- for fix/max98357a branch changes from v2: - split patch into logical changes - rebased to new branch changes from v1: - matched subject line to subsystem style sound/soc/codecs/max98357a.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index f493fb6fd4ea764d617b3d7dd5f3b1ebe1e0fe20..e9e6efbc21dd527299fb459ec01903fa8c24dae2 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -12,10 +12,19 @@ * max98357a.c -- MAX98357A ALSA SoC Codec driver */ -#include <linux/module.h> +#include <linux/device.h> +#include <linux/err.h> #include <linux/gpio.h> #include <linux/gpio/consumer.h> +#include <linux/kernel.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> #include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> #define DRV_NAME "max98357a" -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

2 2

[alsa-devel] [PATCH] ASoC: cirrus: tlv320aic23 needs I2C
by Arnd Bergmann 21 Feb '15

21 Feb '15

The tlv320aic23 codec is selected by the ep93xx snapper platform, which are missing a dependency on I2C, and that can result in this build error, as found during randconfig builds: .../codecs/tlv320aic23-i2c.c: In function 'tlv320aic23_i2c_probe': .../codecs/tlv320aic23-i2c.c:27:2: error: implicit declaration of function 'i2c_check_functionality' [-Werror=implicit-function-declaration] if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) ^ This adds the missing dependency. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 7b7fbcd49e5e..c7cd60f009e9 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -16,7 +16,7 @@ config SND_EP93XX_SOC_AC97 config SND_EP93XX_SOC_SNAPPERCL15 tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" - depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 + depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C select SND_EP93XX_SOC_I2S select SND_SOC_TLV320AIC23_I2C help

2 1

[alsa-devel] [PATCH v2] ASoC: Samsung: add missing I2C/SPI dependencies
by Arnd Bergmann 21 Feb '15

21 Feb '15

A few sound drivers for the samsung platforms are missing dependencies on I2C or SPI, which can lead to build errors like codecs/rt5631.c:1737:1: warning: data definition has no type or storage class 31_i2c_driver); codecs/rt5631.c:1737:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] codecs/rt5631.c:1737:1: warning: parameter names (without types) in function declaration codecs/rt5631.c:1726:26: warning: 'rt5631_i2c_driver' defined but not used [-Wunused-variable] I have gone through all the ones that did not already have an I2C dependency and added the ones that I found missing, namely arndale, odroid-x2, littlemill, bells and speyside and this patch adds all the dependencies. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 3cebf6ca03df..0632a36852c8 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -174,7 +174,7 @@ 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 + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C && SPI_MASTER select SND_SAMSUNG_I2S select SND_SOC_WM8996 select SND_SOC_WM9081 @@ -189,7 +189,7 @@ config SND_SOC_TOBERMORY config SND_SOC_BELLS tristate "Audio support for Wolfson Bells" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA && I2C && SPI_MASTER select SND_SAMSUNG_I2S select SND_SOC_WM5102 select SND_SOC_WM5110 @@ -206,7 +206,7 @@ config SND_SOC_LOWLAND config SND_SOC_LITTLEMILL tristate "Audio support for Wolfson Littlemill" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C select SND_SAMSUNG_I2S select MFD_WM8994 select SND_SOC_WM8994 @@ -223,7 +223,7 @@ config SND_SOC_SNOW config SND_SOC_ODROIDX2 tristate "Audio support for Odroid-X2 and Odroid-U3" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SOC_MAX98090 select SND_SAMSUNG_I2S help @@ -231,6 +231,6 @@ config SND_SOC_ODROIDX2 config SND_SOC_ARNDALE_RT5631_ALC5631 tristate "Audio support for RT5631(ALC5631) on Arndale Board" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SAMSUNG_I2S select SND_SOC_RT5631

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Ensure DSP controls are always persistent
by Charles Keepax 21 Feb '15

21 Feb '15

From: Nikesh Oswal <Nikesh.Oswal(a)wolfsonmicro.com> Currently DSP controls are persistent (across DSP On/Off) only if they were set whilst the DSP is off. This change makes the controls persistent irrespective of when they are set. Signed-off-by: Nikesh Oswal <Nikesh.Oswal(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 5 ++--- 1 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ff67b33..185ee85 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -420,10 +420,9 @@ static int wm_coeff_put(struct snd_kcontrol *kcontrol, memcpy(ctl->cache, p, ctl->len); - if (!ctl->enabled) { - ctl->set = 1; + ctl->set = 1; + if (!ctl->enabled) return 0; - } return wm_coeff_write_control(kcontrol, p, ctl->len); } -- 1.7.2.5

2 2

[alsa-devel] [PATCH] ALSA: firewire-lib: fix an unexpected byte sequence for micro sign
by Takashi Sakamoto 21 Feb '15

21 Feb '15

The sign for microsecond (U+0085, MICRO SIGN) was encoded to '0x c2 b5' by UTF-8 character encoding scheme. But the byte sequence was converted to '0x c3 82 c2 b5' in a previous commit. As a result, the byte sequence cannot represent microsecond sign in UTF-8 or ASCII. This may confuse developers. This commit replaces the sign to string expression with 'microseconds' to purge superfluous troubles. Fixes: 5c697e5b46ef("ALSA: firewire-lib: remove rx_blocks_for_midi quirk") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 0d58018..7b7b183 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -33,7 +33,7 @@ */ #define MAX_MIDI_RX_BLOCKS 8 -#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 Âµs */ +#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 microseconds */ /* isochronous header parameters */ #define ISO_DATA_LENGTH_SHIFT 16 -- 2.1.0

2 1

[alsa-devel] [PATCH v3 0/5] alsa-lib: audio timestamping evolutions
by Pierre-Louis Bossart 21 Feb '15

21 Feb '15

alsa-lib side of kernel patches already merged on topic/timestamp. Pierre-Louis Bossart (5): sync asound.h with kernel version add support for get/set_audio_htstamp_config pcm: add helper functions to query timestamping capabilities pcm: add support for new STATUS_EXT ioctl test: fix audio_time with new get/set audio_tstamp_config include/pcm.h | 47 ++++- include/sound/asound.h | 37 +++- src/pcm/pcm.c | 73 +++++++- src/pcm/pcm_hw.c | 16 +- test/audio_time.c | 491 +++++++++++++++++++++++++++++++------------------ 5 files changed, 475 insertions(+), 189 deletions(-) -- 1.9.1

1 5

[alsa-devel] [PATCH v6 0/7] audio timestamping evolutions
by Pierre-Louis Bossart 21 Feb '15

21 Feb '15

This series of patches was inspired by recent threads on the alsa mailing list, as well issues detected with existing and upcoming hardware: 1. there was a request from the PulseAudio community to get more information from drivers to make rewinds safer. While the conclusion was that it's nearly impossible for a driver developer to provide this information, there are however ways to assess the granularity of the hw_ptr updates using timestamping capabilities, and indirectly understand how safe rewinds might be. 2. There was also a request to add a start_at capability based either on system hr timers or a wall clock, which requires a means to expose both types of information to applications. Rather than adding new sets of timestamps, it is suggested the new start_at functionality relies on the new definition provides by these patches 3. For new hardware, there is a neeed to let low-level drivers handle timestamping instead of having the ALSA core do it. Similarly there is a need to let the low-level driver update the initial estimate for the trigger timestamp if there are delays to start a stream (eg. with USB) These patches try to provide an answer to these multiple needs by building on the work done two years ago to expose wall clock information to applications. The evolution is to let application select which audio timestamp they are interested in, track the delay and drifts between recurring measurements and get, when possible, an idea of the accuracy of the underlying hardware. A backwards compatible mode is provided in case userspace doesn't provide any timestamp selection (results based on HDAudio wallclock for playback, hw_ptr in all other cases). The first 4 patches are corrections for misses in the way the system and trigger timestamps are handled, and the last 6 provide the additional audio timestamping selection. A second batch is planned to enable hardware capabilities in a low-level drivers. A corresponding set of patches is available for alsa-lib. V2 changes: trigger_tstamp: trigger_tstamp_latched, pending redefined as bool trigger_tstamp_latched reset in snd_pcm_pre_start() audio_ts_config, report: keep separate structure but use different bitfields for in and out. use u32 instead of __u32, add comments that these structures are internal COMPAT backwards compatible mode, uses WALL_CLOCK/LINK for HDAudio playback and DEFAULT (hw_ptr) everywhere else INFO bits: reclaimed 32-bits from hw_params, renamed as info_ext moved all timestamp info to info_ext snd_pcm_status: read only 32-bit audio_tstamp_data, ignore all other fields V3/4: Addressed feedback from Jaroslav: no change to STATUS ioctl, new functionality introduced in STATUS_EXT ioctl with r/w params. bumped PCM protocol for detection on STATUS_EXT in userspace used 32-bit word for accuracy report, no mantissa/exponent packing rolled back info_ext changes, all INFO fields remain in same word Merged comments from Liam (code simplifications) fixed packing Added driver_tstamp field in case there is a delay in passing the tstamp and audio_tstamp over IPC. v5: no functional change, only addressed Takashi comments on patch 2,3,4 fixed used of bool with true/false and ! in comparisons move trigger_tstamp_pending_update flag to snd_usb_substream, will be used by ALSA core when other devices need it v6: rebased on v3.19-rc7 after merge of initial 4 patches address Takashi's comments: simplified parameter handling, const and zero-clear cleanups in pcm.h, fix git-bisect issue Pierre-Louis Bossart (7): ALSA: core: selection of audio_tstamp type and accuracy reports ALSA: core: pass audio tstamp config from userspace ALSA: core: pass audio tstamp config from userspace in compat mode ALSA: core: add .get_time_info ALSA: hda: replace .wallclock by .get_time_info ALSA: core: remove .wall_clock ALSA: bump PCM protocol to 2.0.13 Documentation/sound/alsa/timestamping.txt | 200 ++++++++++++++++++++++++++++++ include/sound/pcm.h | 66 +++++++++- include/uapi/sound/asound.h | 36 +++++- sound/core/pcm_compat.c | 28 ++++- sound/core/pcm_lib.c | 88 ++++++++----- sound/core/pcm_native.c | 41 +++++- sound/pci/hda/hda_controller.c | 40 ++++-- 7 files changed, 444 insertions(+), 55 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt -- 1.9.1

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[alsa-devel] [PATCH 0/4] Internal PCM fixes / cleanups
by Takashi Iwai 20 Feb '15

20 Feb '15

Hi, while reading the internal PCM (BE) handling code again, I noticed a few potential bugs. Here are fix and cleanup patches. Takashi === Takashi Iwai (4): ALSA: pcm: Minor refactoring in snd_pcm_attach_substream() ALSA: pcm: Don't add internal PCMs to PCM device list ALSA: pcm: Don't notify internal PCMs ALSA: pcm: Don't ignore internal PCMs in snd_pcm_dev_disconnect() sound/core/pcm.c | 92 +++++++++++++++++++++++++------------------------------- 1 file changed, 41 insertions(+), 51 deletions(-) -- 2.3.0

2 7

[alsa-devel] [PATCH] ALSA: hda - Avoid unnecessary power-up at mixer amp changes
by Takashi Iwai 20 Feb '15

20 Feb '15

When the mixer amp is touched by control elements, we don't have to power up always; if the codec was suspended at the time, we can just update the amp cache and it's reflected to the hardware upon resume. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 33 ++++++++++++++++----------------- 1 file changed, 16 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0734cb35887d..5e755eb6f19a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2148,11 +2148,10 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_amp_read); static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val, - bool init_only) + bool init_only, bool cache_only) { struct hda_amp_info *info; unsigned int caps; - unsigned int cache_only; if (snd_BUG_ON(mask & ~0xff)) mask &= 0xff; @@ -2170,7 +2169,7 @@ static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, return 0; } info->vol[ch] = val; - cache_only = info->head.dirty = codec->cached_write; + info->head.dirty |= cache_only; caps = info->amp_caps; mutex_unlock(&codec->hash_mutex); if (!cache_only) @@ -2194,7 +2193,8 @@ static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val) { - return codec_amp_update(codec, nid, ch, direction, idx, mask, val, false); + return codec_amp_update(codec, nid, ch, direction, idx, mask, val, + false, codec->cached_write); } EXPORT_SYMBOL_GPL(snd_hda_codec_amp_update); @@ -2241,7 +2241,8 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_amp_stereo); int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch, int dir, int idx, int mask, int val) { - return codec_amp_update(codec, nid, ch, dir, idx, mask, val, true); + return codec_amp_update(codec, nid, ch, dir, idx, mask, val, true, + codec->cached_write); } EXPORT_SYMBOL_GPL(snd_hda_codec_amp_init); @@ -2383,8 +2384,8 @@ update_amp_value(struct hda_codec *codec, hda_nid_t nid, maxval = get_amp_max_value(codec, nid, dir, 0); if (val > maxval) val = maxval; - return snd_hda_codec_amp_update(codec, nid, ch, dir, idx, - HDA_AMP_VOLMASK, val); + return codec_amp_update(codec, nid, ch, dir, idx, HDA_AMP_VOLMASK, val, + false, !hda_codec_is_power_on(codec)); } /** @@ -2434,14 +2435,12 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change = 0; - snd_hda_power_up(codec); if (chs & 1) { change = update_amp_value(codec, nid, 0, dir, idx, ofs, *valp); valp++; } if (chs & 2) change |= update_amp_value(codec, nid, 1, dir, idx, ofs, *valp); - snd_hda_power_down(codec); return change; } EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_put); @@ -3109,19 +3108,19 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change = 0; - snd_hda_power_up(codec); if (chs & 1) { - change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx, - HDA_AMP_MUTE, - *valp ? 0 : HDA_AMP_MUTE); + change = codec_amp_update(codec, nid, 0, dir, idx, + HDA_AMP_MUTE, + *valp ? 0 : HDA_AMP_MUTE, false, + !hda_codec_is_power_on(codec)); valp++; } if (chs & 2) - change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx, - HDA_AMP_MUTE, - *valp ? 0 : HDA_AMP_MUTE); + change |= codec_amp_update(codec, nid, 1, dir, idx, + HDA_AMP_MUTE, + *valp ? 0 : HDA_AMP_MUTE, false, + !hda_codec_is_power_on(codec)); hda_call_check_power_status(codec, nid); - snd_hda_power_down(codec); return change; } EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_put); -- 2.3.0

1 0

[alsa-devel] [PATCH 0/6] Minor HD-audio cleanups
by Takashi Iwai 20 Feb '15

20 Feb '15

Hi, this is a series of minor cleanups for HD-audio codes. No functional changes but just code refactoring for the further works. The patches are found in topic/hda-cleanup branch of sound git tree, too. Takashi === Takashi Iwai (6): ALSA: hda - Drop hda_bus_template for snd_hda_bus_new() ALSA: hda - Embed struct hda_bus_unsolicited into struct hda_bus ALSA: hda - Introduce azx_has_pm_runtime() macro ALSA: hda - Fold hda_priv.h into hda_controller.h ALSA: hda - Drop azx_mixer_create() ALSA: hda - Split azx_codec_create() to two phases sound/pci/hda/hda_codec.c | 54 +----- sound/pci/hda/hda_codec.h | 22 +-- sound/pci/hda/hda_controller.c | 96 +++++----- sound/pci/hda/hda_controller.h | 397 +++++++++++++++++++++++++++++++++++++++- sound/pci/hda/hda_i915.c | 2 +- sound/pci/hda/hda_intel.c | 23 ++- sound/pci/hda/hda_intel.h | 2 +- sound/pci/hda/hda_local.h | 17 -- sound/pci/hda/hda_priv.h | 406 ----------------------------------------- sound/pci/hda/hda_tegra.c | 9 +- 10 files changed, 477 insertions(+), 551 deletions(-) delete mode 100644 sound/pci/hda/hda_priv.h -- 2.3.0

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[alsa-devel] Correct 5.1 settings for Asus mobo AT4NM10-I with IDT 92HD73C1X5
by Misan Thropos 19 Feb '15

19 Feb '15

I got it to work with the help of "debianuser" on #alsa@freenode and hdajackretask which can be found in alsa-tools-gui (if you use Debian). Had to change some pin settings from default configuration to make it work. You can get the working configuration from: http://www.alsa-project.org/db/?f=71b06cd49f390e0ff4061d069a2efcb93b3148d2 It enables the S/PDIF digital output too. Maybe you like to take that information for a model for IDT 92HD73C1X5. cheers, Misan

2 1

[alsa-devel] [PATCH 0/19] ASoC: rsnd: move 2nd DMAC from DMAEngine to rsnd driver
by Kuninori Morimoto 19 Feb '15

19 Feb '15

Hi Mark, Arnd, Vinod Cc: Laurent, SH-ML Renesas R-Car sound (= rsnd) needs 2 DMAC which are called as Audio DMAC (= 1st DMAC) and Audio DMAC peri peri (2nd DMAC). And rsnd had assumed that 1st / 2nd DMACs are implemented as DMAEngine. But, in result of DMA ML long term discussion, 2nd DMAC was concluded that it is not a general purpose DMAC (2nd DMAC is for Device to Device inside sound system). Additionally, current DMAEngine can't support Device to Device, and we don't have correct DT bindings for it at this point. So the easiest solution for it is that move it from DMAEngine to rsnd driver. http://thread.gmane.org/gmane.linux.ports.sh.devel/41063/focus=43829 These patches add Audio DMAC peri peri implementation in rsnd driver. I will remove current DMAEngine side Audio DMAC peri peri implement if these are accepted. Arnd, Vinod I'm happy if you can send Ack for these patches 1) - 5) are cleanup / tidyup rsnd driver 6) - 11) adds Audio DMAC peri peri implementation in rsnd driver 12) change dma-names for 1st DMAC on DT 13) - 14) cleanup. these are no longer needed now 15) - 19) DT bindings text update 12) changes dma-names for 1st DMAC. It was assumed that 1st / 2nd DMAC are used as DMAEngine. dma-names had same naming rule for both DMAC to avoid complex DT mapping issue, But now, 2nd DMAC is no longer DMAEngine, we don't need to use this naming rule anymore. Now, no SoC/platform is using DMA for rsnd driver yet in upstream. I believe 12) is no problem for DT. Kuninori Morimoto (19): 1) ASoC: rsnd: remove SH-DMA-BASE specific implementation 2) ASoC: rsnd: remove un-needed parameter from rsnd_dma_init() 3) ASoC: rsnd: remove unused rsnd_dma_available() 4) ASoC: rsnd: remove un-needed parameter from rsnd_dma_quit() 5) ASoC: rsnd: tidyup rsnd_dma_to_mod() macro declaration position 6) ASoC: rsnd: enable to get resource by name 7) ASoC: rsnd: add rsnd_gen_get_phy_addr() to get physical address 8) ASoC: rsnd: add dma.c for Audio DMAC / Audio DMAC peri peri 9) ASoC: rsnd: move rsnd_gen_dma_addr() from gen.c to dma.c 10) ASoC: rsnd: enable to care 1st / 2nd DMAC on rsnd_dma_xxx() 11) ASoC: rsnd: add Audio DMAC peri peri support rework 12) ASoC: rsnd: dma-names cares 1st DMAC only 13) ASoC: rsnd: remove rsnd_dma::addr 14) ASoC: rsnd: remove rsnd_dma::dir 15) ASoC: rsnd: tidyup rcar_sound,ssi sample code 16) ASoC: rsnd: add sample code of rcar_sound,src irq 17) ASoC: rsnd: add sample code of reg-names 18) ASoC: rsnd: add sample code of missing clocks 19) ASoC: rsnd: add sample code of dma entry .../devicetree/bindings/sound/renesas,rsnd.txt | 131 +++-- sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/core.c | 224 +------- sound/soc/sh/rcar/dma.c | 583 ++++++++++++++++++++ sound/soc/sh/rcar/gen.c | 145 +---- sound/soc/sh/rcar/rsnd.h | 50 +- sound/soc/sh/rcar/src.c | 5 +- sound/soc/sh/rcar/ssi.c | 5 +- 8 files changed, 725 insertions(+), 420 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] baytrail-pcm-audio: audio broken on Lenovo Thinkpad Tablet 10
by Sebastien Bourdeauducq 18 Feb '15

18 Feb '15

Hi, the Thinkpad Tablet 10 has an impressive amount of problems under Linux, among which the audio is completely broken. The kernel log is: [ 7.532413] rt5640 i2c-10EC5640:00: Device with ID register 6271 is not rt5640/39 [ 7.908208] byt-rt5640 byt-rt5640: ASoC: CPU DAI baytrail-pcm-audio not registered [ 7.908231] platform byt-rt5640: Driver byt-rt5640 requests probe deferral [ 7.931549] baytrail-pcm-audio baytrail-pcm-audio: error: invalid DMA engine 0 [ 7.931557] baytrail-pcm-audio baytrail-pcm-audio: sst_dma_new failed -22 [ 7.967040] baytrail-pcm-audio baytrail-pcm-audio: FW version: 04.05.13.a0 [ 7.967047] baytrail-pcm-audio baytrail-pcm-audio: Build type: a0 [ 7.967051] baytrail-pcm-audio baytrail-pcm-audio: Build date: Apr 2 2014 14:14:39 [ 7.967212] byt-rt5640 byt-rt5640: ASoC: CODEC DAI rt5640-aif1 not registered [ 7.967248] platform byt-rt5640: Driver byt-rt5640 requests probe deferral I'm running kernel 3.19 with the latest firmware from the linux-firmware repository. Sebastien -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 1

[alsa-devel] [PATCH 0/3] Fixing snd_use_case_get() bugs
by Tanu Kaskinen 17 Feb '15

17 Feb '15

Tanu Kaskinen (3): ucm: fix incorrect error code sign ucm: fix the logic of choosing the default cdev ucm: fix some variable constness issues include/use-case.h | 5 +++-- src/ucm/main.c | 51 ++++++++++++++++++++++++++++++++------------------- 2 files changed, 35 insertions(+), 21 deletions(-) -- 1.9.3

2 4

[alsa-devel] [PATCH] ALSA: usb: Fix support for Denon DA-300USB DAC (ID 154e:1003)
by Takashi Iwai 17 Feb '15

17 Feb '15

From: Frank C Guenther <bugzilla.frnkcg(a)spamgourmet.com> Fix problem where playback of Denon DA-300USB DAC sometimes does not start and leads to error messages like "clock source 41 is not valid, cannot use". Solution: Treat this device the same as other Denon/Marantz devices in sound/usb/quirks.c. Tested with both PCM and DSD formats. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=93261 Signed-off-by: Frank C Guenther <bugzilla.frnkcg(a)spamgourmet.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5c0c05314df0..753a47de8459 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1127,6 +1127,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, int err; switch (subs->stream->chip->usb_id) { + case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */ case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ @@ -1206,6 +1207,7 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) { switch (le16_to_cpu(dev->descriptor.idProduct)) { + case 0x1003: /* Denon DA300-USB */ case 0x3005: /* Marantz HD-DAC1 */ case 0x3006: /* Marantz SA-14S1 */ mdelay(20); @@ -1267,6 +1269,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* Denon/Marantz devices with USB DAC functionality */ switch (chip->usb_id) { + case USB_ID(0x154e, 0x1003): /* Denon DA300-USB */ case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ if (fp->altsetting == 2) -- 2.3.0

1 0

[alsa-devel] [PATCH] ASoC: Intel: add SNDRV_PCM_INFO_DRAIN_TRIGGER flag
by libin.yang＠intel.com 17 Feb '15

17 Feb '15

From: Libin Yang <libin.yang(a)intel.com> Add SNDRV_PCM_INFO_DRAIN_TRIGGER in snd_pcm_hardware.info to call sst_hsw_stream_set_silence_start() when PCM draining. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index d6fa9d5..7e21e8f 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -91,7 +91,8 @@ static const struct snd_pcm_hardware hsw_pcm_hardware = { SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP | + SNDRV_PCM_INFO_DRAIN_TRIGGER, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = PAGE_SIZE, -- 1.9.1

3 3

[alsa-devel] [PATCH 40/45] include/uapi/sound/emu10k1.h: include sound/asound.h
by Mikko Rapeli 17 Feb '15

17 Feb '15

Fixes userspace compilation errors like: error: field ‘id’ has incomplete type struct snd_ctl_elem_id id; /* full control ID definition */ Signed-off-by: Mikko Rapeli <mikko.rapeli(a)iki.fi> --- include/uapi/sound/emu10k1.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h index d1bbaf7..ec1535b 100644 --- a/include/uapi/sound/emu10k1.h +++ b/include/uapi/sound/emu10k1.h @@ -23,8 +23,7 @@ #define _UAPI__SOUND_EMU10K1_H #include <linux/types.h> - - +#include <sound/asound.h> /* * ---- FX8010 ---- -- 2.1.4

2 1

[alsa-devel] [PATCH 39/45] include/uapi/sound/asequencer.h: include sound/asound.h
by Mikko Rapeli 17 Feb '15

17 Feb '15

Fixes userspace compilation error: error: unknown type name ‘snd_seq_client_type_t’ snd_seq_client_type_t type; /* client type */ Signed-off-by: Mikko Rapeli <mikko.rapeli(a)iki.fi> --- include/uapi/sound/asequencer.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/uapi/sound/asequencer.h b/include/uapi/sound/asequencer.h index 09c8a00..5a5fa49 100644 --- a/include/uapi/sound/asequencer.h +++ b/include/uapi/sound/asequencer.h @@ -22,6 +22,7 @@ #ifndef _UAPI__SOUND_ASEQUENCER_H #define _UAPI__SOUND_ASEQUENCER_H +#include <sound/asound.h> /** version of the sequencer */ #define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1) -- 2.1.4

2 1

[alsa-devel] [PATCH 35/45] include/uapi/sound/asound.h: include stdlib.h in userspace
by Mikko Rapeli 17 Feb '15

17 Feb '15

Fixes compiler errors like: error: field ‘trigger_tstamp’ has incomplete type error: invalid application of ‘sizeof’ to incomplete t ype ‘struct timespec’ Signed-off-by: Mikko Rapeli <mikko.rapeli(a)iki.fi> --- include/uapi/sound/asound.h | 3 +++ 1 file changed, 3 insertions(+) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 941d32f..af156b0 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -25,6 +25,9 @@ #include <linux/types.h> +#ifndef __KERNEL__ +#include <stdlib.h> +#endif /* * protocol version -- 2.1.4

2 1

[alsa-devel] [PATCH 0/3] ALSA: hdspm - Remove various code duplication
by Adrian Knoth 17 Feb '15

17 Feb '15

From: Adrian Knoth <aknoth(a)google.com> Hi! As promised last week, here's a series that removes code duplication in hdspm.c. Adrian Knoth (3): ALSA: hdspm - DRY cleanup in .open callbacks ALSA: hdspm - DRY cleanup in .release callback ALSA: hdspm - DRY cleanup of snd_pcm_ops sound/pci/rme9652/hdspm.c | 141 ++++++++++++---------------------------------- 1 file changed, 35 insertions(+), 106 deletions(-) -- 2.1.0

2 4

[alsa-devel] [PATCH v2] ALSA: usb-audio: Don't attempt to get Lifecam HD-5000 sample rate
by Joe Turner 17 Feb '15

17 Feb '15

Adds a quirk to disable the check that the sample rate has been set correctly, as the Lifecam does not support getting the sample rate. This means that we don't need to wait for the USB timeout when attempting to get the sample rate. Waiting for the timeout causes problems in some applications, which give up on the device acquisition process before it has had time to complete, resulting in no sound. Signed-off-by: Joe Turner <joe(a)oampo.co.uk> --- Changes in v2: - Fixed code style, as suggested by Takashi Iwai <tiwai(a)suse.de> sound/usb/clock.c | 5 +++++ sound/usb/quirks.c | 5 +++++ sound/usb/quirks.h | 2 ++ 3 files changed, 12 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 03fed66..9d6e709 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -303,6 +303,11 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, return err; } + /* Don't check the sample rate for devices which we know don't + * support reading */ + if (snd_usb_get_sample_rate_quirk(chip)) + return 0; + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a739841..50add5e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1111,6 +1111,11 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, } } +int snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) +{ + /* MS Lifecam HD-5000 doesn't support reading the sample rate. */ + return chip->usb_id == USB_ID(0x045E, 0x076D); +} /* Marantz/Denon USB DACs need a vendor cmd to switch * between PCM and native DSD mode diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index 1b86238..428c158 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -21,6 +21,8 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, void snd_usb_set_format_quirk(struct snd_usb_substream *subs, struct audioformat *fmt); +int snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip); + int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat *fp); -- 2.3.0

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[alsa-devel] iMX6 SSI driver: fsl_ssi vs imx-ssi
by Xuebing Wang 17 Feb '15

17 Feb '15

Hi community, I am developing an ASOC codec driver based on iMX6SL platform. I noticed there are fsl_ssi.c and imx-ssi.c in folder sound/soc/fsl. Would you please tell me what are the differences between these 2, and which one should I use? Thanks. -- Xuebing

2 2

[alsa-devel] [PATCH] .gitignore: add test-driver
by Tanu Kaskinen 16 Feb '15

16 Feb '15

Signed-off-by: Tanu Kaskinen <tanu.kaskinen(a)linux.intel.com> --- .gitignore | 1 + 1 file changed, 1 insertion(+) diff --git a/.gitignore b/.gitignore index 05a2459..e1cc163 100644 --- a/.gitignore +++ b/.gitignore @@ -11,6 +11,7 @@ Makefile libtool depcomp ltmain.sh +test-driver version Makefile.in aclocal.m4 -- 1.9.3

2 3

[alsa-devel] [PATCH 0/6] Modernize sequecner device binding
by Takashi Iwai 16 Feb '15

16 Feb '15

Hi, this is a series of patches for 3.21 (or whatever the next-to-next) to sanitze and modernize the device binding code of ALSA sequencer core. Takashi Takashi Iwai (6): ALSA: seq: Move EXPORT_SYMBOL() after each function ALSA: seq: Don't compile snd_seq_device_load_drivers() for built-in ALSA: seq: Rewrite sequencer device binding with standard bus ALSA: seq: Clean up device and driver structs ALSA: seq: Define driver object in each driver ALSA: seq: Drop snd_seq_autoload_lock() and _unlock() include/sound/seq_device.h | 46 +-- include/sound/seq_kernel.h | 6 +- sound/core/seq/oss/seq_oss.c | 22 +- sound/core/seq/oss/seq_oss_synth.c | 6 +- sound/core/seq/oss/seq_oss_synth.h | 4 +- sound/core/seq/seq_device.c | 569 +++++++++---------------------------- sound/core/seq/seq_dummy.c | 6 +- sound/core/seq/seq_midi.c | 36 +-- sound/drivers/opl3/opl3_seq.c | 34 +-- sound/drivers/opl4/opl4_seq.c | 33 +-- sound/isa/sb/emu8000_synth.c | 35 +-- sound/pci/emu10k1/emu10k1_synth.c | 35 +-- 12 files changed, 256 insertions(+), 576 deletions(-) -- 2.3.0

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[alsa-devel] [PATCH] MS Lifecam HD-5000: Don't attempt to get the sample rate
by Joe Turner 14 Feb '15

14 Feb '15

Hi Folks, This is my first time doing anything kernel-related, so apologies in advance for any faux-pas which are to follow. I hope this is at least going to the correct place! Currently after setting the sample rate of USB devices the driver tries to get the sample rate back from the device, to check that the two match up. The Lifecam doesn't support getting the sample rate, and so this operation times-out, and we continue regardless. Because the USB timeout is pretty long this causes issues with some apps, which begin to doubt the ability of the device to produce audio before the timeout is up and misbehave in various ways. See https://www.google.co.uk/search?q=%22cannot+get+freq+at+ep+0x82%22 for a range of reports relating to this issue. This patch adds a quirk so that rather than check the sample rate and timeout, we don't bother checking at all. Hope this all looks okay. Cheers, Joe

2 1

[alsa-devel] [PATCH] ALSA: hda/tegra check correct return value from ioremap_resource
by eliot＠blennerhassett.gen.nz 14 Feb '15

14 Feb '15

From: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> Signed-off-by: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> --- sound/pci/hda/hda_tegra.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 227990b..375e94f 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -329,8 +329,8 @@ static int hda_tegra_init_chip(struct azx *chip, struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); hda->regs = devm_ioremap_resource(dev, res); - if (IS_ERR(chip->remap_addr)) - return PTR_ERR(chip->remap_addr); + if (IS_ERR(hda->regs)) + return PTR_ERR(hda->regs); chip->remap_addr = hda->regs + HDA_BAR0; chip->addr = res->start + HDA_BAR0; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fsl-asoc-card: Add snd_soc_of_parse_audio_routing()
by Nicolin Chen 14 Feb '15

14 Feb '15

This patch adds snd_soc_of_parse_audio_routing() to get dapm routes via Device Tree. Because snd_soc_of_parse_audio_routing() overrides dpam_routes, we use snd_soc_dapm_add_routes() to append the original build-in DPCM routes instead of linking them to dpam_routes directly. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl-asoc-card.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 3f6959c..bdc9671 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -388,6 +388,8 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) struct device *dev = card->dev; int ret; + snd_soc_dapm_add_routes(&card->dapm, audio_map, ARRAY_SIZE(audio_map)); + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret) { @@ -503,15 +505,19 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.dev = &pdev->dev; priv->card.name = priv->name; priv->card.dai_link = priv->dai_link; - priv->card.dapm_routes = audio_map; priv->card.late_probe = fsl_asoc_card_late_probe; - priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets; priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets); memcpy(priv->dai_link, fsl_asoc_card_dai, sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); + ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing"); + if (ret) { + dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret); + goto asrc_fail; + } + /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; priv->dai_link[0].codec_of_node = codec_np; -- 1.9.1

2 4

[alsa-devel] [Patch V4 00/10] ASoC: QCOM: Add support for ipq806x SOC
by Kenneth Westfield 14 Feb '15

14 Feb '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> This patch series adds support for I2S audio playback on the Qualcomm Technologies ipq806x SOC. The ipq806x SOC has audio-related hardware blocks in its low-power audio subsystem (or LPASS). One of the relevant blocks in the LPASS is its low-power audio interface (or LPAIF). This contains an MI2S port, which is what these drivers are configured to use. The LPAIF also contains a DMA engine that is dedicated to moving audio samples into the transmit FIFO of the MI2S port. In addition, there is also low-power memory (LPM) within the audio subsystem, which is used for buffering the audio samples. The development board being used for testing contains the ipq806x SOC and a Maxim max98357a DAC/amp. One bus from the MI2S port of the SOC is connected to the DAC/amp for stereo playback. This bus is configured so that the SOC is bus master and consists of DATA, LRCLK, and BCLK. The DAC/amp does not need MCLK to operate. In addition, a single GPIO pin from the SOC is connected to the same DAC/amp, which gives enable/disable control over the DAC/amp. The specific drivers added are: * a codec DAI driver for controlling the DAC/amp * a CPU DAI driver for controlling the MI2S port * a platform driver for controlling the LPAIF DMA engine These drivers, together, are tied into simple-audio-card to complete the audio implementation. Corresponding additions to the device tree for the ipq806x SOC and its documentation has also been added. Also, as this is a new directory, the MAINTAINERS file has been updated as well. The LPASS also contains clocks that need to be controlled. Those drivers have been submitted as a separate patch series: [PATCH v3 0/8] qcom audio clock control drivers http://lkml.org/lkml/2015/1/19/656 = Changes since V3 [Patch V3 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085694.h… * Placed the content of the inline functions into the callbacks. * Replaced use of readl/writel register access functions with regmap access functions. Notable exception is the ISR, which uses ioread32/iowrite32. * Rearranged the sequencing of the hardware block enables to fit within the ASoC framework callbacks, while remaining functional. REQ 1: The hardware requires the enable sequence to be: LPAIF-DMA[enable],then LPAIF-MI2S[enable], then DAC-GPIO[enable] REQ 2: The hardware requires the disable sequence to be: DAC-GPIO[disable], then LPAIF-MI2S[disable] * Corrected the implementation of the pointer callback. * Utilize the LPM to buffer audio samples, rather than memory external to LPASS. * Corrected the interrupt clearing in the ISR. * Implemented a default system clock (defined by the simple-card DT node), and optional LPASS DT node modifiers that can alter the system clock in order to expand the range of available bit clock frequencies. * Addressed all of the remaining issues raised by Mark Brown. * General code cleanup. = Changes since V2 [Patch v2 00/11] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085186.h… * Removed the PCM platform driver from the DTS platform and tied it to the CPU DAI driver. * Changed I2S pinctrl to use generic naming convention and moved control to CPU DAI driver. It should be controlled now by soc-core's pinctrl_pm_* functionality. * Added stub DAPM support in codec driver. As the DAC GPIO needs to be enabled last when starting playback, and disabled first when stopping playback, it seems as though the trigger function may be the place for this. Suggestions are welcome for a better place to put this. * Removed machine driver and tied DAI drivers to simple-audio-card. * Packaged the build files and Maxim codec files together in one change. * Removed QCOM as vendor from Maxim code and documentation. * Separated the SOC and board definitions into the correct DTS files. * Update device tree documentation to reflect changes. * General code cleanup. = Changes since V1 [PATCH 0/9] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084322.h… * Remove the native LPAIF driver, and move its functionality to the CPU DAI driver. * Add a codec driver to manage the pins going to the external DAC (previously managed by the machine driver). * Use devm_* and dev_* where possible. * ISR only handles relevant DMA channel now. * Update device tree documentation to reflect changes. * General code cleanup. Kenneth Westfield (10): MAINTAINERS: Add QCOM audio ASoC maintainer ASoC: max98357a: Document MAX98357A bindings ASoC: qcom: Document LPASS CPU bindings ASoC: codec: Add MAX98357A codec driver ASoC: ipq806x: add LPASS header files ASoC: ipq806x: Add LPASS CPU DAI driver ASoC: ipq806x: Add LPASS platform driver ASoC: qcom: Add ability to build QCOM drivers ASoC: Allow for building QCOM drivers ARM: dts: Model IPQ LPASS audio hardware .../devicetree/bindings/sound/max98357a.txt | 14 + .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 66 +++ MAINTAINERS | 7 + arch/arm/boot/dts/qcom-ipq8064.dtsi | 23 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98357a.c | 138 +++++ sound/soc/qcom/Kconfig | 27 + sound/soc/qcom/Makefile | 6 + sound/soc/qcom/lpass-cpu.c | 557 +++++++++++++++++++++ sound/soc/qcom/lpass-lpaif-ipq806x.h | 170 +++++++ sound/soc/qcom/lpass-platform.c | 514 +++++++++++++++++++ sound/soc/qcom/lpass.h | 64 +++ 15 files changed, 1594 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98357a.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt create mode 100644 sound/soc/codecs/max98357a.c create mode 100644 sound/soc/qcom/Kconfig create mode 100644 sound/soc/qcom/Makefile create mode 100644 sound/soc/qcom/lpass-cpu.c create mode 100644 sound/soc/qcom/lpass-lpaif-ipq806x.h create mode 100644 sound/soc/qcom/lpass-platform.c create mode 100644 sound/soc/qcom/lpass.h -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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[alsa-devel] [PATCH] ASoC: Intel: set initial runtime PM status to active for ACPI-enumerated ADSP
by mengdong.lin＠intel.com 14 Feb '15

14 Feb '15

From: Mengdong Lin <mengdong.lin(a)intel.com> The ADSP on Braswell/Baytrail is an ACPI device. This patch sets its initial runtime PM status to active. Otherwise, its initial status is suspended and runtime_suspend ops will not be called after probe and thus cannot further trigger ACPI _PS3 (D3) method to put the device into low power D3cold state. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 8a8d56a..d6ea800 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -379,6 +379,10 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx) * initially active. So change the state to active before * enabling the pm */ + + if (!acpi_disabled) + pm_runtime_set_active(ctx->dev); + pm_runtime_enable(ctx->dev); if (acpi_disabled) -- 1.9.1

2 1

[alsa-devel] [Patch] ASoC: codec: Add missing deps/hdrs to MAX98357A
by Kenneth Westfield 14 Feb '15

14 Feb '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> For the max98357a codec driver: - Add missing build dependancy to GPIOLIB in Kconfig. - Add header files to avoid implicit declarations and indirect inclusions. - Remove use of DRV_NAME constant. Signed-off-by: Kenneth Westfield <kwestfie(a)codeaurora.org> --- for topic/max98357a branch sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/max98357a.c | 23 +++++++++++++++-------- 2 files changed, 16 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6ecac1e4428eafc491d6cfc25110f6b3dbf699ed..0500b3ed5fa5ac4d45663ad659da9db3003bc5a3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -69,7 +69,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C - select SND_SOC_MAX98357A + select SND_SOC_MAX98357A if GPIOLIB select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 1806333ea29e5a6b49e2a0adb8c58257d34b7508..4ee23fbc4e1237268ae104319a9cfe4fc369934e 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -12,11 +12,19 @@ * max98357a.c -- MAX98357A ALSA SoC Codec driver */ -#include <linux/module.h> +#include <linux/device.h> +#include <linux/err.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/kernel.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> #include <sound/soc.h> - -#define DRV_NAME "max98357a" +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) @@ -77,9 +85,9 @@ static struct snd_soc_dai_ops max98357a_dai_ops = { }; static struct snd_soc_dai_driver max98357a_dai_driver = { - .name = DRV_NAME, + .name = "max98357a", .playback = { - .stream_name = DRV_NAME "-playback", + .stream_name = "max98357a-playback", .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32, @@ -117,7 +125,7 @@ static int max98357a_platform_remove(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id max98357a_device_id[] = { - { .compatible = "maxim," DRV_NAME, }, + { .compatible = "maxim,max98357a" }, {} }; MODULE_DEVICE_TABLE(of, max98357a_device_id); @@ -125,7 +133,7 @@ MODULE_DEVICE_TABLE(of, max98357a_device_id); static struct platform_driver max98357a_platform_driver = { .driver = { - .name = DRV_NAME, + .name = "max98357a", .of_match_table = of_match_ptr(max98357a_device_id), }, .probe = max98357a_platform_probe, @@ -135,4 +143,3 @@ module_platform_driver(max98357a_platform_driver); MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:" DRV_NAME); -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

2 1

[alsa-devel] [PATCH] ASoC: omap-hdmi-audio: Add missing error check for devm_kzalloc
by Kiran Padwal 14 Feb '15

14 Feb '15

This patch add a missing check on the return value of devm_kzalloc, which would cause a NULL pointer dereference in a OOM situation. Signed-off-by: Kiran Padwal <kiran.padwal(a)smartplayin.com> --- sound/soc/omap/omap-hdmi-audio.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c index ccfb41c..f7eb42a 100644 --- a/sound/soc/omap/omap-hdmi-audio.c +++ b/sound/soc/omap/omap-hdmi-audio.c @@ -352,6 +352,9 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) return ret; card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + card->name = devm_kasprintf(dev, GFP_KERNEL, "HDMI %s", dev_name(ad->dssdev)); card->owner = THIS_MODULE; -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Add support for CBS_CFM mode
by Peter Ujfalusi 14 Feb '15

14 Feb '15

Support for setups where codec is bitclock slave and frame master. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 750124030576..c0e485cba9eb 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -441,6 +441,18 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); mcasp->bclk_master = 1; break; + case SND_SOC_DAIFMT_CBS_CFM: + /* codec is clock slave and frame master */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); + mcasp->bclk_master = 1; + break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); -- 2.3.0

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[alsa-devel] [PATCH 0/4] ASoC: samsung: clean up references to outdated Kconfig symbols
by Paul Bolle 14 Feb '15

14 Feb '15

A few series of ARM cleanups regarding S5PC100 and S5PV210 were included in v3.17. These series removed these five Kconfig symbols: MACH_AQUILA MACH_GONI MACH_SMDKC100 MACH_SMDKC110 MACH_SMDKV210 This series, done on top of next-20150119, removes the remaining references to those five symbols. (The first two patches are actually resends.) Paul Bolle (4): ASoC: samsung: Remove goni or aquila with the WM8994 ASoC: samsung: Remove PCM support for WM8580 on SMDK ASoC: samsung: MACH_SMDKC100 isn't supported anymore ASoC: samsung: remove outdated Kconfig references sound/soc/samsung/Kconfig | 24 +-- sound/soc/samsung/Makefile | 4 - sound/soc/samsung/goni_wm8994.c | 289 ------------------------------------- sound/soc/samsung/smdk_wm8580pcm.c | 176 ---------------------- 4 files changed, 2 insertions(+), 491 deletions(-) delete mode 100644 sound/soc/samsung/goni_wm8994.c delete mode 100644 sound/soc/samsung/smdk_wm8580pcm.c -- 1.9.3

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[alsa-devel] sound/pci/hda/hda_tegra.c
by Eliot Blennerhassett 13 Feb '15

13 Feb '15

Browsing code I came across this... sound/pci/hda/hda_tegra.c static int hda_tegra_init_chip(struct azx *chip, struct platform_device *pdev) ... res = platform_get_resource(pdev, IORESOURCE_MEM, 0); hda->regs = devm_ioremap_resource(dev, res); Checking chip->remap_addr before assignment looks wierd to me, am I missing something? Should it be 'hda->regs' here? if (IS_ERR(chip->remap_addr)) return PTR_ERR(chip->remap_addr); chip->remap_addr = hda->regs + HDA_BAR0; chip->addr = res->start + HDA_BAR0;

2 1

[alsa-devel] [PATCH] ALSA: hda - enable mute led quirk for one more hp machine.
by Hui Wang 13 Feb '15

13 Feb '15

Otherwise, the mute led can't work at all. Tested-by: Taihsiang Ho <taihsiang.ho(a)canonical.com> Cc: <stable(a)vger.kernel.org> BugLink: https://bugs.launchpad.net/bugs/1410704 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- 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 ddb9308..b2b24a8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4937,6 +4937,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x225f, "HP", ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY), /* ALC282 */ + SND_PCI_QUIRK(0x103c, 0x21f9, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2210, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2214, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2236, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), -- 1.9.1

2 1

[alsa-devel] [PATCH 0/5] ALSA: control: improve core codes with intrusion
by Takashi Sakamoto 13 Feb '15

13 Feb '15

This patchset is for improving core control codes. Most of them have any effects for processing, messaging or status codes. This patchset is based on my previous one: [alsa-devel] [PATCH 0/8] ALSA: control: refactoring core codes http://mailman.alsa-project.org/pipermail/alsa-devel/2015-February/087783.h… Takashi Sakamoto (5): ALSA: control: drop a sub-effect for caller's data ALSA: control: use dev_dbg() for warning to add duplicated controls ALSA: control: fix logic error about control count in a device ALSA: control: improve returned value because of the rest ALSA: control: save stack usage at creating new instance sound/core/control.c | 209 ++++++++++++++++++++++++++++----------------------- 1 file changed, 116 insertions(+), 93 deletions(-) -- 2.1.0

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[alsa-devel] [PATCH 0/4 v3] ASoC: rsnd: tidyup .remove method
by Kuninori Morimoto 13 Feb '15

13 Feb '15

Hi Mark These are v3 of unbind/bind issue fixup patches. v2 patch-set was Kuninori Morimoto (6): 1) ASoC: rsnd: set device data before snd_soc_register_platform/component 2) ASoC: soc-core: indicate unregister debug message once 3) ASoC: soc-core: add snd_soc_remove_card() 4) ASoC: soc-core: deactivate pins in snd_soc_instantiate_card() 5) ASoC: soc-core: call soc_cleanup_card_debugfs() from snd_soc_unregister_card() 6) ASoC: soc-core: call snd_soc_remove_card() when component del Now, 1) / 2) are applied already. So, v3 patches is same as 3) - 6)

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[alsa-devel] davinci-mcasp: extension to use AHCLKX pin as external clock source
by Urs Fässler 12 Feb '15

12 Feb '15

The AHCLKX pin seems not to be supported as external source. The first patch is a general fix to allow external clock. The second allows you to select the AHCLKX pin as clock source. Patch built against v3.19-rc7

2 4

[alsa-devel] Status of ALSA FireWire stack as of Linux 3.19
by Takashi Sakamoto 12 Feb '15

12 Feb '15

Dear all, I'm just a developer, not a maintainer of this stack, but often receive personal messages from users. I think it helpful to explain about its status and known issues here, as of Linux 3.19. == Impovements in 3.19 release * ALSA FireWire stack gain a thlottle for MIDI data rate. No devices can disorder due to MIDI buffer overflow. http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084652.h… * ALSA Dice driver becomes to handle more models with 176.4/192.0 kHz support and both of playback/capture for PCM/MIDI substreams. * ALSA firewire-speakers driver was obsoleted by ALSA OXFW driver to handles more models based on OXFW 970/971. * As a result, ALSA supports approximately 140 FireWire models in comparison to Linux 3.15. == Model dependent issues There're some reports about model dependent issues. * PreSonus FP10 (former known as FirePod) * When ALSA BeBoB driver re-establishes connections, this model causes hung-up, then reset automatically. The cause is unknown. * Terratek PHASE 88 Rack FW * https://github.com/takaswie/snd-firewire-improve/issues/8 * According to a reporter, when ALSA BeBoB driver establishes connections according to CMP, this model transfers no packets. The cause is unknown. * Behringer XENIX UFX 1604 * http://sourceforge.net/p/ffado/mailman/message/32846563/ * When ALSA BeBoB driver communicates, this model sometimes misses to transfer responses. The cause is unknown but I guess that this model expects vendor-specific command. * Echo AudioFire 2 * http://mailman.alsa-project.org/pipermail/alsa-devel/2014-August/080379.html * At least, firmware version 5.7.3 for this model has a quirk about AMDTP processing. It's not compliant to this protocol. Even if ALSA Fireworks driver communicates successfully, ALSA FireWire stack detects protocol error. * Weiss engineering DAC 202 * According to a reporter, when ALSA Dice driver probes this model it's trials are ETIMEDOUT at setting sampling rate. The cause is unknown. == Stack and driver issues There're still some issues about ALSA FireWire stack and drivers. Most of them are written in my report: https://github.com/takaswie/alsa-firewire-report * 9.2 A lack of arrangement for severe latency * 9.3 A lack of delay for timestamp processing * 9.4 A lack of PCM configuration for virtual surround nodes * 9.6 A lack of control elements for ALSA control interface * 9.7 A lack of packet sorting * 9.9 Various results at bus reset * 9.10 A lack of synchronizing several devices on the same bus * Sometimes kernel oops at disconnecting devices during streaming. * I guess three contexts have somewhat race condition: * Application process with snd_pcm_prepare() * Isochronous context * .remove callback by OHCI 1394 host controller driver == The other issues * PulseAudio 6,0 (not released yet) gain multichannel profiles. http://www.freedesktop.org/wiki/Software/PulseAudio/Notes/6.0/ ALSA FireWire sound devices can be available, while in most case the display name is the name of OHCI 3194 host controller. https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/1381475 I've posted RFC patch to pulseaudio-discuss but I think it better to touch hardware database which systemd produces. http://lists.freedesktop.org/archives/pulseaudio-discuss/2015-January/02294… == My plan for 3.21 release * Some userspace library is under development for developers of mixer application. * libhinawa http://mailman.alsa-project.org/pipermail/alsa-devel/2015-January/086969.ht… * alsa-gir https://github.com/takaswie/alsa-gir * An offload of control functionality from ALSA OXFW driver to userspace * Currently ALSA OXFW driver includes control functionality in kernel side, while this can be implemented in userspace. This functionaliy utilizes certailn commands which the other drivers can use for the same purpose. * Need to discuss about it, later. * ALSA BeBoB driver improvement * ALSA BeBoB driver's bootup command for M-Audio models depends on little-endiannness machine. I need more time to prepare for big-endianness machine to test my patch for this issue. * Some models expect BridgeCo.-specific command to change sampling rate. Currently ALSA BeBoB driver uses commands in AV/C general specification. Let's start to replace these comands. Regards Takashi Sakamoto

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[alsa-devel] [Patch V2] ASoC: max98357a: Add missing deps and headers
by Kenneth Westfield 12 Feb '15

12 Feb '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> Add missing build dependancy to GPIOLIB in Kconfig. Add header files to avoid implicit declarations and indirect inclusions. Remove use of DRV_NAME constant. Signed-off-by: Kenneth Westfield <kwestfie(a)codeaurora.org> --- for topic/max98357a branch changes from v1: - matched subject line to subsystem style sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/max98357a.c | 23 +++++++++++++++-------- 2 files changed, 16 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6ecac1e4428eafc491d6cfc25110f6b3dbf699ed..0500b3ed5fa5ac4d45663ad659da9db3003bc5a3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -69,7 +69,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C - select SND_SOC_MAX98357A + select SND_SOC_MAX98357A if GPIOLIB select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 1806333ea29e5a6b49e2a0adb8c58257d34b7508..4ee23fbc4e1237268ae104319a9cfe4fc369934e 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -12,11 +12,19 @@ * max98357a.c -- MAX98357A ALSA SoC Codec driver */ -#include <linux/module.h> +#include <linux/device.h> +#include <linux/err.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/kernel.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> #include <sound/soc.h> - -#define DRV_NAME "max98357a" +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) @@ -77,9 +85,9 @@ static struct snd_soc_dai_ops max98357a_dai_ops = { }; static struct snd_soc_dai_driver max98357a_dai_driver = { - .name = DRV_NAME, + .name = "max98357a", .playback = { - .stream_name = DRV_NAME "-playback", + .stream_name = "max98357a-playback", .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32, @@ -117,7 +125,7 @@ static int max98357a_platform_remove(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id max98357a_device_id[] = { - { .compatible = "maxim," DRV_NAME, }, + { .compatible = "maxim,max98357a" }, {} }; MODULE_DEVICE_TABLE(of, max98357a_device_id); @@ -125,7 +133,7 @@ MODULE_DEVICE_TABLE(of, max98357a_device_id); static struct platform_driver max98357a_platform_driver = { .driver = { - .name = DRV_NAME, + .name = "max98357a", .of_match_table = of_match_ptr(max98357a_device_id), }, .probe = max98357a_platform_probe, @@ -135,4 +143,3 @@ module_platform_driver(max98357a_platform_driver); MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:" DRV_NAME); -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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[alsa-devel] [Patch V5 00/12] ASoC: QCOM: Add support for ipq806x SOC
by Kenneth Westfield 12 Feb '15

12 Feb '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> This patch series adds support for I2S audio playback on the Qualcomm Technologies ipq806x SOC. The ipq806x SOC has audio-related hardware blocks in its low-power audio subsystem (or LPASS). One of the relevant blocks in the LPASS is its low-power audio interface (or LPAIF). This contains an MI2S port, which is what these drivers are configured to use. The LPAIF also contains a DMA engine that is dedicated to moving audio samples into the transmit FIFO of the MI2S port. The development board being used for testing contains the ipq806x SOC and a Maxim max98357a DAC/amp. One bus from the MI2S port of the SOC is connected to the DAC/amp for stereo playback. This bus is configured so that the SOC is bus master and consists of DATA, LRCLK, and BCLK. The DAC/amp does not need MCLK to operate. In addition, a single GPIO pin from the SOC is connected to the same DAC/amp, which gives enable/disable control over the DAC/amp. The specific drivers added are: * a codec DAI driver for controlling the DAC/amp * a CPU DAI driver for controlling the MI2S port * a platform driver for controlling the LPAIF DMA engine * a machine driver that instantiates a dai-link for playback Corresponding additions to the device tree for the ipq806x SOC and its documentation has also been added. Also, as this is a new directory, the MAINTAINERS file has been updated as well. The LPASS also contains clocks that need to be controlled. Those drivers have been submitted as a separate patch series: [PATCH v3 0/8] qcom audio clock control drivers http://lkml.org/lkml/2015/1/19/656 = Changes since V4 [Patch V4 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2015-February/087499.h… * Replaced simple-card with a machine driver to resolve the system clock configuration, rather than having the CPU DAI driver do it. * Added header files to avoid indirect header dependencies and implicit forward declarations. * Tweaked the ISR to match the conventions of the surrounding code. * Removed the usage of the low-power memory as it is not needed. * Removed the use of the DRV_NAME constant. * Added explicit dependancy on gpiolib for the codec driver. * Moved the MODULE_DEVICE_TABLE macro inside the CONFIG_OF conditional. * Modified the documentation to reflect the changes. * General code cleanup. = Changes since V3 [Patch V3 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085694.h… * Placed the content of the inline functions into the callbacks. * Replaced use of readl/writel register access functions with regmap access functions. Notable exception is the ISR, which uses ioread32/iowrite32. * Rearranged the sequencing of the hardware block enables to fit within the ASoC framework callbacks, while remaining functional. REQ 1: The hardware requires the enable sequence to be: LPAIF-DMA[enable],then LPAIF-MI2S[enable], then DAC-GPIO[enable] REQ 2: The hardware requires the disable sequence to be: DAC-GPIO[disable], then LPAIF-MI2S[disable] * Corrected the implementation of the pointer callback. * Utilize the LPM to buffer audio samples, rather than memory external to LPASS. * Corrected the interrupt clearing in the ISR. * Implemented a default system clock (defined by the simple-card DT node), and optional LPASS DT node modifiers that can alter the system clock in order to expand the range of available bit clock frequencies. * Addressed all of the remaining issues raised by Mark Brown. * General code cleanup. = Changes since V2 [Patch v2 00/11] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085186.h… * Removed the PCM platform driver from the DTS platform and tied it to the CPU DAI driver. * Changed I2S pinctrl to use generic naming convention and moved control to CPU DAI driver. It should be controlled now by soc-core's pinctrl_pm_* functionality. * Added stub DAPM support in codec driver. As the DAC GPIO needs to be enabled last when starting playback, and disabled first when stopping playback, it seems as though the trigger function may be the place for this. Suggestions are welcome for a better place to put this. * Removed machine driver and tied DAI drivers to simple-audio-card. * Packaged the build files and Maxim codec files together in one change. * Removed QCOM as vendor from Maxim code and documentation. * Separated the SOC and board definitions into the correct DTS files. * Update device tree documentation to reflect changes. * General code cleanup. = Changes since V1 [PATCH 0/9] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084322.h… * Remove the native LPAIF driver, and move its functionality to the CPU DAI driver. * Add a codec driver to manage the pins going to the external DAC (previously managed by the machine driver). * Use devm_* and dev_* where possible. * ISR only handles relevant DMA channel now. * Update device tree documentation to reflect changes. * General code cleanup. Kenneth Westfield (12): MAINTAINERS: Add QCOM audio ASoC maintainer ASoC: max98357a: Document MAX98357A bindings ASoC: qcom: Document LPASS CPU bindings ASoC: qcom: Document IPQ806X-MAX89357A bindings ASoC: codec: Add MAX98357A codec driver ASoC: ipq806x: add LPASS header files ASoC: ipq806x: Add LPASS CPU DAI driver ASoC: ipq806x: Add LPASS platform driver ASoC: ipq806x: Add @max98357a machine driver ASoC: qcom: Add ability to build QCOM drivers ASoC: Allow for building QCOM drivers ARM: dts: Model IPQ LPASS audio hardware .../devicetree/bindings/sound/max98357a.txt | 14 + .../bindings/sound/qcom,ipq806x-max98357a.txt | 24 + .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 38 ++ MAINTAINERS | 7 + arch/arm/boot/dts/qcom-ipq8064.dtsi | 23 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98357a.c | 145 ++++++ sound/soc/qcom/Kconfig | 26 + sound/soc/qcom/Makefile | 11 + sound/soc/qcom/ipq806x-max98357a.c | 171 +++++++ sound/soc/qcom/lpass-cpu.c | 481 +++++++++++++++++++ sound/soc/qcom/lpass-lpaif-ipq806x.h | 172 +++++++ sound/soc/qcom/lpass-platform.c | 526 +++++++++++++++++++++ sound/soc/qcom/lpass.h | 51 ++ 17 files changed, 1697 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98357a.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,ipq806x-max98357a.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt create mode 100644 sound/soc/codecs/max98357a.c create mode 100644 sound/soc/qcom/Kconfig create mode 100644 sound/soc/qcom/Makefile create mode 100644 sound/soc/qcom/ipq806x-max98357a.c create mode 100644 sound/soc/qcom/lpass-cpu.c create mode 100644 sound/soc/qcom/lpass-lpaif-ipq806x.h create mode 100644 sound/soc/qcom/lpass-platform.c create mode 100644 sound/soc/qcom/lpass.h -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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[alsa-devel] [PATCH] ASoC: max98357a: Fix missing include
by Vincent Stehlé 12 Feb '15

12 Feb '15

This fixes the following compilation errors: sound/soc/codecs/max98357a.c: In function ‘max98357a_daiops_trigger’: sound/soc/codecs/max98357a.c:30:3: error: implicit declaration of function ‘gpiod_set_value’ [-Werror=implicit-function-declaration] sound/soc/codecs/max98357a.c: In function ‘max98357a_codec_probe’: sound/soc/codecs/max98357a.c:55:2: error: implicit declaration of function ‘devm_gpiod_get’ [-Werror=implicit-function-declaration] sound/soc/codecs/max98357a.c:61:2: error: implicit declaration of function ‘gpiod_direction_output’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors Signed-off-by: Vincent Stehlé <vincent.stehle(a)laposte.net> Cc: Kenneth Westfield <kwestfie(a)codeaurora.org> Cc: Mark Brown <broonie(a)kernel.org> --- Hi, Those compilation errors can be seen for example with hppa allmodconfig. Best regards, V. sound/soc/codecs/max98357a.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 1806333..f493fb6 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -14,6 +14,7 @@ #include <linux/module.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <sound/soc.h> #define DRV_NAME "max98357a" -- 2.1.4

2 1

[alsa-devel] [PATCH 0/3] ucm: miscellaneous fixes
by Tanu Kaskinen 11 Feb '15

11 Feb '15

While working on PulseAudio's UCM code, I noticed strange use of const in snd_use_case_get(). The third patch fixes that. While working on the third patch, I also noticed strange code in execute_sequence(). The first patch fixes one obvious bug, but there was another issue which I didn't know how to fix, so I just added a FIXME comment instead (that's just as an RFC, I hope the discussion will reveal what the proper fix should be). Tanu Kaskinen (3): ucm: fix variable mixup ucm: add a FIXME comment ucm: fix inappropriate use of const include/use-case.h | 2 +- src/ucm/main.c | 34 +++++++++++++++++++++------------- 2 files changed, 22 insertions(+), 14 deletions(-) -- 1.9.3

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[alsa-devel] [patch] ALSA: seq: potential out of bounds in do_control()
by Dan Carpenter 11 Feb '15

11 Feb '15

Smatch complains that "control" is user specifigy and needs to be capped. The call tree to understand this warning is quite long. snd_seq_write() <-- get the event from the user snd_seq_client_enqueue_event() snd_seq_deliver_event() deliver_to_subscribers() snd_seq_deliver_single_event() snd_opl3_oss_event_input() snd_midi_process_event() do_control() Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- I have spent some time reviewing this code, but I may have missed something where we verify that control is in bounds. I'm not very familiar with this code and the call tree is fairly long. diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c index 9b6470c..7ba9373 100644 --- a/sound/core/seq/seq_midi_emul.c +++ b/sound/core/seq/seq_midi_emul.c @@ -269,6 +269,9 @@ do_control(struct snd_midi_op *ops, void *drv, struct snd_midi_channel_set *chse { int i; + if (control >= ARRAY_SIZE(chan->control)) + return; + /* Switches */ if ((control >=64 && control <=69) || (control >= 80 && control <= 83)) { /* These are all switches; either off or on so set to 0 or 127 */

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[alsa-devel] [PATCH v2] ALSA: line6: Improve line6_read/write_data() interfaces
by Chris Rorvick 11 Feb '15

11 Feb '15

The address cannot be negative so make it unsigned. Also, an unsigned int is always sufficient for the length, so no need to overdo it with a size_t. Finally, add in range checks to see if the values passed in actually fit where they are used. Signed-off-by: Chris Rorvick <chris(a)rorvick.com> --- This was dropped from an earlier series ("Cleanup reads/writes to Line 6 device memory".) This version leaves the arguements wide (within reason) and does range checking. Regards, Chris sound/usb/line6/driver.c | 14 ++++++++++---- sound/usb/line6/driver.h | 8 ++++---- 2 files changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 99b63a7..4f82b57 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -302,14 +302,17 @@ static void line6_data_received(struct urb *urb) /* Read data from device. */ -int line6_read_data(struct usb_line6 *line6, int address, void *data, - size_t datalen) +int line6_read_data(struct usb_line6 *line6, unsigned address, void *data, + unsigned datalen) { struct usb_device *usbdev = line6->usbdev; int ret; unsigned char len; unsigned count; + if (address > 0xffff || datalen > 0xff) + return EINVAL; + /* query the serial number: */ ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, @@ -370,14 +373,17 @@ EXPORT_SYMBOL_GPL(line6_read_data); /* Write data to device. */ -int line6_write_data(struct usb_line6 *line6, int address, void *data, - size_t datalen) +int line6_write_data(struct usb_line6 *line6, unsigned address, void *data, + unsigned datalen) { struct usb_device *usbdev = line6->usbdev; int ret; unsigned char status; int count; + if (address > 0xffff || datalen > 0xffff) + return EINVAL; + ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, 0x0022, address, data, datalen, diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index 5d20294..7da643e 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -147,8 +147,8 @@ struct usb_line6 { extern char *line6_alloc_sysex_buffer(struct usb_line6 *line6, int code1, int code2, int size); -extern int line6_read_data(struct usb_line6 *line6, int address, void *data, - size_t datalen); +extern int line6_read_data(struct usb_line6 *line6, unsigned address, + void *data, unsigned datalen); extern int line6_read_serial_number(struct usb_line6 *line6, u32 *serial_number); extern int line6_send_raw_message_async(struct usb_line6 *line6, @@ -161,8 +161,8 @@ extern void line6_start_timer(struct timer_list *timer, unsigned long msecs, void (*function)(unsigned long), unsigned long data); extern int line6_version_request_async(struct usb_line6 *line6); -extern int line6_write_data(struct usb_line6 *line6, int address, void *data, - size_t datalen); +extern int line6_write_data(struct usb_line6 *line6, unsigned address, + void *data, unsigned datalen); int line6_probe(struct usb_interface *interface, const struct usb_device_id *id, -- 2.1.0

2 2

[alsa-devel] Memory and Cycles overhead in <snd_soc_init>
by Bhaskara rao Budiredla 11 Feb '15

11 Feb '15

Hi - May I know the purpose of the call to snd_soc_init( ) from kernel_init thread? snd_soc_init( ) is trying to register a dummy platform device "snd-soc-dummy" containing (dummy_codec/dummy_platform) and a sound card device. Neither of those devices are added to platform bus before they are making a call to platform_driver_register( ). Due to that platform_match( ) will fail and they will not be added to platform bus . Thus, snd_soc_init( ) is not doing any useful task during kernel boot time. Could someone help me if my understanding is correct? If I am correct, shall we remove the call to snd_soc_init( ) in kernel_init thread to save boot time and memory? Otherwise please suggest me where those dummy devices and sound card gets added to platform bus and the purpose of creating them? Thanks, Bhaskara

2 3

[alsa-devel] [PATCH 0/6] Cleanup reads/writes to Line 6 device memory
by Chris Rorvick 11 Feb '15

11 Feb '15

The primary fix here is to throttle the requests for status after a read or write operation. In some cases I was seeing more than 1000 before the device was ready with a response. The remainder are smaller improvements related to the read/write logic. All of this seems pretty straightforward with the possible exception of changing the returned error from EINVAL to EIO. The EINVAL seems wrong to me but maybe not? Regards, Chris Chris Rorvick (6): ALSA: line6: Improve line6_read/write_data() interfaces ALSA: line6: Add delay before reading status ALSA: line6: Return error if device not responding ALSA: line6: Return EIO if read/write not successful ALSA: line6: Use explicit type for serial number ALSA: line6: toneport: Use explicit type for firmware version sound/usb/line6/driver.c | 51 +++++++++++++++++++++++++++++++++------------- sound/usb/line6/driver.h | 10 ++++----- sound/usb/line6/pod.c | 4 ++-- sound/usb/line6/toneport.c | 4 ++-- 4 files changed, 46 insertions(+), 23 deletions(-) -- 2.1.0

2 13

[alsa-devel] [PATCH v3 kernel 1/3] snd_pcm_start_at and friends.
by Tim Cussins 11 Feb '15

11 Feb '15

We introduce the kernel-side of the START_AT ioctl. struct runtime is updated to hold information about the currently active start_at timer, if any. This facilitates cancellation via snd_pcm_start_at_abort(), and querying via snd_pcm_status(). struct snd_start_at holds a startat operation and its arguments. Signed-off-by: Tim Cussins <timcussins(a)eml.cc> diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 0e88e7a..2943e1a 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -421,7 +421,10 @@ struct snd_pcm_status { snd_pcm_state_t suspended_state; /* suspended stream state */ __u32 reserved_alignment; /* must be filled with zero */ struct timespec audio_tstamp; /* from sample counter or wall clock */ - unsigned char reserved[56-sizeof(struct timespec)]; /* must be filled with zero */ + int startat_pending; /* 1 if a start_at timer is pending, 0 otherwise */ + int startat_clock_type; /* start_at clock type, if pending */ + struct timespec startat_start_time; /* start_at start time, if pending */ + unsigned char reserved[48-(2*sizeof(struct timespec))]; /* must be filled with zero */ }; struct snd_pcm_mmap_status { @@ -473,6 +476,34 @@ enum { SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, }; +enum { + SNDRV_PCM_STARTAT_OP_SET = 0, + SNDRV_PCM_STARTAT_OP_CANCEL, + SNDRV_PCM_STARTAT_OP_STATUS, + SNDRV_PCM_STARTAT_OP_LAST = SNDRV_PCM_STARTAT_OP_STATUS, +}; + +enum { + SNDRV_PCM_STARTAT_CLOCK_TYPE_GETTIMEOFDAY = 0, + SNDRV_PCM_STARTAT_CLOCK_TYPE_MONOTONIC, + SNDRV_PCM_STARTAT_CLOCK_TYPE_LINK, + SNDRV_PCM_STARTAT_CLOCK_TYPE_LAST = SNDRV_PCM_STARTAT_CLOCK_TYPE_LINK, +}; + +struct snd_start_at { + int op; /* startat operation to be performed */ + union { /* fields for setting a startat timer */ + struct { + int clock_type; /* clock type e.g. SNDRV_PCM_STARTAT_CLOCK_TYPE_GETTIMEOFDAY */ + struct timespec start_time; /* start time */ + } set; + struct { + int clock_type; + struct timespec current_time; + } status; + } args; +}; + /* channel positions */ enum { SNDRV_CHMAP_UNKNOWN = 0, @@ -551,6 +582,8 @@ enum { #define SNDRV_PCM_IOCTL_READN_FRAMES _IOR('A', 0x53, struct snd_xfern) #define SNDRV_PCM_IOCTL_LINK _IOW('A', 0x60, int) #define SNDRV_PCM_IOCTL_UNLINK _IO('A', 0x61) +#define SNDRV_PCM_IOCTL_START_AT _IOW('A', 0x62, struct snd_start_at) + /***************************************************************************** * * -- 1.9.1

3 10

[alsa-devel] [PATCH] ASoC: Intel: Clean data after SST fw fetch
by libin.yang＠intel.com 11 Feb '15

11 Feb '15

From: Libin Yang <libin.yang(a)intel.com> The BDW audio firmware DSP manages the DMA and the DMA cannot be stopped exactly at the end of the playback stream. This means stale samples may be played at PCM stop unless the driver copies silence to the subsequent periods. Signed-off-by: Libin Yang <libin.yang(a)intel.com> Reviewed-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 28 +++++++++++++++++ sound/soc/intel/sst-haswell-ipc.h | 9 ++++++ sound/soc/intel/sst-haswell-pcm.c | 65 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 102 insertions(+) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 0ab1309..394af56 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -31,6 +31,7 @@ #include <linux/dma-mapping.h> #include <linux/debugfs.h> #include <linux/pm_runtime.h> +#include <sound/asound.h> #include "sst-haswell-ipc.h" #include "sst-dsp.h" @@ -242,6 +243,9 @@ struct sst_hsw_stream { u32 (*notify_position)(struct sst_hsw_stream *stream, void *data); void *pdata; + /* record the fw read position when playback */ + snd_pcm_uframes_t old_position; + bool play_silence; struct list_head node; }; @@ -1347,6 +1351,30 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) return 0; } +snd_pcm_uframes_t sst_hsw_stream_get_old_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream) +{ + return stream->old_position; +} + +void sst_hsw_stream_set_old_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, snd_pcm_uframes_t val) +{ + stream->old_position = val; +} + +bool sst_hsw_stream_get_silence_start(struct sst_hsw *hsw, + struct sst_hsw_stream *stream) +{ + return stream->play_silence; +} + +void sst_hsw_stream_set_silence_start(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, bool val) +{ + stream->play_silence = val; +} + /* Stream Information - these calls could be inline but we want the IPC ABI to be opaque to client PCM drivers to cope with any future ABI changes */ int sst_hsw_mixer_get_info(struct sst_hsw *hsw) diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index c1ad901..8580960 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -20,6 +20,7 @@ #include <linux/types.h> #include <linux/kernel.h> #include <linux/platform_device.h> +#include <sound/asound.h> #define SST_HSW_NO_CHANNELS 4 #define SST_HSW_MAX_DX_REGIONS 14 @@ -425,6 +426,14 @@ int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); +snd_pcm_uframes_t sst_hsw_stream_get_old_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream); +void sst_hsw_stream_set_old_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, snd_pcm_uframes_t val); +bool sst_hsw_stream_get_silence_start(struct sst_hsw *hsw, + struct sst_hsw_stream *stream); +void sst_hsw_stream_set_silence_start(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, bool val); int sst_hsw_mixer_get_info(struct sst_hsw *hsw); /* Stream ALSA trigger operations */ diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 78fa01b..d6fa9d5 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -36,6 +36,11 @@ #define HSW_PCM_COUNT 6 #define HSW_VOLUME_MAX 0x7FFFFFFF /* 0dB */ +#define SST_OLD_POSITION(d, r, o) ((d) + \ + frames_to_bytes(r, o)) +#define SST_SAMPLES(r, x) (bytes_to_samples(r, \ + frames_to_bytes(r, (x)))) + /* simple volume table */ static const u32 volume_map[] = { HSW_VOLUME_MAX >> 30, @@ -577,23 +582,34 @@ static int hsw_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); struct hsw_pcm_data *pcm_data; + struct sst_hsw_stream *sst_stream; struct sst_hsw *hsw = pdata->hsw; + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t pos; int dai; dai = mod_map[rtd->cpu_dai->id].dai_id; pcm_data = &pdata->pcm[dai][substream->stream]; + sst_stream = pcm_data->stream; switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + sst_hsw_stream_set_silence_start(hsw, sst_stream, false); sst_hsw_stream_resume(hsw, pcm_data->stream, 0); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sst_hsw_stream_set_silence_start(hsw, sst_stream, false); sst_hsw_stream_pause(hsw, pcm_data->stream, 0); break; + case SNDRV_PCM_TRIGGER_DRAIN: + pos = runtime->control->appl_ptr % runtime->buffer_size; + sst_hsw_stream_set_old_position(hsw, pcm_data->stream, pos); + sst_hsw_stream_set_silence_start(hsw, sst_stream, true); + break; default: break; } @@ -607,13 +623,62 @@ static u32 hsw_notify_pointer(struct sst_hsw_stream *stream, void *data) struct snd_pcm_substream *substream = pcm_data->substream; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct sst_hsw *hsw = pdata->hsw; u32 pos; + snd_pcm_uframes_t position = bytes_to_frames(runtime, + sst_hsw_get_dsp_position(hsw, pcm_data->stream)); + unsigned char *dma_area = runtime->dma_area; + snd_pcm_uframes_t dma_frames = + bytes_to_frames(runtime, runtime->dma_bytes); + snd_pcm_uframes_t old_position; + ssize_t samples; pos = frames_to_bytes(runtime, (runtime->control->appl_ptr % runtime->buffer_size)); dev_vdbg(rtd->dev, "PCM: App pointer %d bytes\n", pos); + /* SST fw don't know where to stop dma + * So, SST driver need to clean the data which has been consumed + */ + if (dma_area == NULL || dma_frames <= 0 + || (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + || !sst_hsw_stream_get_silence_start(hsw, stream)) { + snd_pcm_period_elapsed(substream); + return pos; + } + + old_position = sst_hsw_stream_get_old_position(hsw, stream); + if (position > old_position) { + if (position < dma_frames) { + samples = SST_SAMPLES(runtime, position - old_position); + snd_pcm_format_set_silence(runtime->format, + SST_OLD_POSITION(dma_area, + runtime, old_position), + samples); + } else + dev_err(rtd->dev, "PCM: position is wrong\n"); + } else { + if (old_position < dma_frames) { + samples = SST_SAMPLES(runtime, + dma_frames - old_position); + snd_pcm_format_set_silence(runtime->format, + SST_OLD_POSITION(dma_area, + runtime, old_position), + samples); + } else + dev_err(rtd->dev, "PCM: dma_bytes is wrong\n"); + if (position < dma_frames) { + samples = SST_SAMPLES(runtime, position); + snd_pcm_format_set_silence(runtime->format, + dma_area, samples); + } else + dev_err(rtd->dev, "PCM: position is wrong\n"); + } + sst_hsw_stream_set_old_position(hsw, stream, position); + /* let alsa know we have play a period */ snd_pcm_period_elapsed(substream); return pos; -- 1.9.1

4 4

[alsa-devel] [PATCH] ASoC: fsl_ssi: Fix the incorrect limitation of the bit clock rate
by Nicolin Chen 11 Feb '15

11 Feb '15

According to i.MX Reference Manual, the bit-clock frequency generated by SSI must be never greater than 1/5 of the peripheral clock frequency. This peripheral clock, however, is not baudclk but the IPG clock (i.e. ssi_private->clk in the fsl_ssi driver). So this patch just simply fixes the incorrect limitation applied to the bit clock (baudclk) rate. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_ssi.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 059496e..d7365c5 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -603,10 +603,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, factor = (div2 + 1) * (7 * psr + 1) * 2; for (i = 0; i < 255; i++) { - /* The bclk rate must be smaller than 1/5 sysclk rate */ - if (factor * (i + 1) < 5) - continue; - tmprate = freq * factor * (i + 2); if (baudclk_is_used) @@ -614,6 +610,13 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, else clkrate = clk_round_rate(ssi_private->baudclk, tmprate); + /* + * Hardware limitation: The bclk rate must be + * never greater than 1/5 IPG clock rate + */ + if (clkrate * 5 > clk_get_rate(ssi_private->clk)) + continue; + clkrate /= factor; afreq = clkrate / (i + 1); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5677: Add the chip type to distinguish the setting of the clock source
by Oder Chiou 11 Feb '15

11 Feb '15

There is only one clock source in the rt5676, so the chip type is added to distinguish the setting of the clock source in the VAD function. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 23 +++++++++++++++++------ sound/soc/codecs/rt5677.h | 6 ++++++ 2 files changed, 23 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5d0bb87..3aae8a4 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -718,11 +718,19 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) RT5677_LDO1_SEL_MASK, 0x0); regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, RT5677_PWR_LDO1, RT5677_PWR_LDO1); - regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, - RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); - regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, - RT5677_PLL2_PR_SRC_MASK | RT5677_DSP_CLK_SRC_MASK, - RT5677_PLL2_PR_SRC_MCLK2 | RT5677_DSP_CLK_SRC_BYPASS); + if (rt5677->type == RT5677) { + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, + RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_PLL2_PR_SRC_MASK | + RT5677_DSP_CLK_SRC_MASK, + RT5677_PLL2_PR_SRC_MCLK2 | + RT5677_DSP_CLK_SRC_BYPASS); + } else { + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_DSP_CLK_SRC_MASK, + RT5677_DSP_CLK_SRC_BYPASS); + } regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x07ff); regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07fd); rt5677_set_dsp_mode(codec, true); @@ -4733,7 +4741,8 @@ static const struct regmap_config rt5677_regmap = { }; static const struct i2c_device_id rt5677_i2c_id[] = { - { "rt5677", 0 }, + { "rt5677", RT5677 }, + { "rt5676", RT5676 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); @@ -4850,6 +4859,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5677); + rt5677->type = id->driver_data; + if (pdata) rt5677->pdata = *pdata; diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index c0a625f..07df96b 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1665,6 +1665,11 @@ enum { RT5677_IRQ_JD3, }; +enum rt5677_type { + RT5677, + RT5676, +}; + struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; @@ -1681,6 +1686,7 @@ struct rt5677_priv { int pll_in; int pll_out; int pow_ldo2; /* POW_LDO2 pin */ + enum rt5677_type type; #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; #endif -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [GIT PULL]: Add sst audio firmware for Braswell platforms
by Vinod Koul 11 Feb '15

11 Feb '15

Hello Linux firmware maintainers, Please PULL to receive the Intel audio firmware for audio DSP in Braswell platforms. This firmware is using the same license as the Baytrail firmware already available in linux-firmware git. The following changes since commit b794c5039dcf0b7ebfeb58929d035f7a1d4c80dd: Merge branch 'marvell-mwifiex' (2015-01-27 10:42:47 -0500) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git BSW for you to fetch changes up to f692aba54c8c80cb27a3eb5388594a55a30504ea: linux-firmware: add sst audio firmware for braswell platforms (2015-02-02 21:12:22 -0800) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: add sst audio firmware for braswell platforms WHENCE | 8 ++++++++ intel/fw_sst_22a8.bin | Bin 0 -> 647390 bytes 2 files changed, 8 insertions(+) create mode 100644 intel/fw_sst_22a8.bin Thanks -- ~Vinod

2 2

[alsa-devel] Crackly playback, Roland AIRA TR-8 USB audio almost works with ALSA . . .
by David W. 10 Feb '15

10 Feb '15

Dear ALSA developers, On 30/12/14 19:39, David W. wrote: > On 15/12/14 23:37, David W. wrote: >> On 07/07/14 07:31, Clemens Ladisch wrote: >>> >>> Apparently, this device needs some vendor-specific magic to enable >>> recording. >>> This AIRA TR-8 USB audio device, which fails with an 'input/output error' after ~10 seconds for 'aplay' and 'arecord', seems to require operation in implicit feedback mode: the following patch suggested by Daniel Mack solves the 'input/output' error for 'aplay', but the playback is noisy and corrupted and the kernel complains of 'rogue URB xfers' (see below). Given this information and the quote below from lsusb etc., I am hoping someone would be able to suggest a further patch to get playback and recording working :-) I can patch, compile and test recent linux kernels (whatever is in: https://www.archlinux.org/packages/?sort=&repo=Testing&q=linux) Here's the patch from Daniel which helped: diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 0d8aba5..e35575b 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -346,6 +346,15 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, alts = &iface->altsetting[1]; goto add_sync_ep; + case USB_ID(0x0582, 0x017c): + ep = 0x85; + iface = usb_ifnum_to_if(dev, 3); + + if (!iface || iface->num_altsetting == 0) + return -EINVAL; + + alts = &iface->altsetting[1]; + goto add_sync_ep; } if (attr == USB_ENDPOINT_SYNC_ASYNC && altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && Here's an example of the kind of complaint that the 3.18.2-2-ARCH kernel, with the above patch applied, reports: Jan 12 23:18:20 dwmobile kernel: WARNING: CPU: 2 PID: 1889 at drivers/usb/core/urb.c:450 usb_submit_urb+0x265/0x5f0 [usbcore]() Jan 12 23:18:20 dwmobile kernel: usb 2-1.1: BOGUS urb xfer, pipe 0 != type 1 Jan 12 23:18:20 dwmobile kernel: Modules linked in: snd_usb_audio snd_usbmidi_lib snd_rawmidi snd_seq_device ctr ccm sha256_ssse3 sha256_generic d Jan 12 23:18:20 dwmobile kernel: snd_pcm snd_timer video mei_me battery ac snd intel_agp button intel_gtt acpi_cpufreq shpchp mei soundcore proce Jan 12 23:18:20 dwmobile kernel: CPU: 2 PID: 1889 Comm: aplay Not tainted 3.18.2-2-custom #1 Jan 12 23:18:20 dwmobile kernel: Hardware name: System76, Inc. Pangolin Performance /W76x/M77xCUH Jan 12 23:18:20 dwmobile kernel: 0000000000000000 000000005911a000 ffff88022988faf8 ffffffff8154f134 Jan 12 23:18:20 dwmobile kernel: 0000000000000000 ffff88022988fb50 ffff88022988fb38 ffffffff81072bc1 Jan 12 23:18:20 dwmobile kernel: ffff88022988fb28 ffff88022f1aba00 0000000000000020 0000000000000003 Jan 12 23:18:20 dwmobile kernel: Call Trace: Jan 12 23:18:20 dwmobile kernel: [<ffffffff8154f134>] dump_stack+0x4e/0x71 Jan 12 23:18:20 dwmobile kernel: [<ffffffff81072bc1>] warn_slowpath_common+0x81/0xa0 Jan 12 23:18:20 dwmobile kernel: [<ffffffff81072c35>] warn_slowpath_fmt+0x55/0x70 Jan 12 23:18:20 dwmobile kernel: [<ffffffffa009b005>] ? usb_enable_endpoint+0x85/0x90 [usbcore] Jan 12 23:18:20 dwmobile kernel: [<ffffffffa0098e75>] usb_submit_urb+0x265/0x5f0 [usbcore] Jan 12 23:18:20 dwmobile kernel: [<ffffffffa08d2c65>] snd_usb_endpoint_start+0x125/0x350 [snd_usb_audio] Jan 12 23:18:20 dwmobile kernel: [<ffffffffa08dac89>] start_endpoints+0xb9/0x1a0 [snd_usb_audio] Jan 12 23:18:20 dwmobile kernel: [<ffffffffa08dcbad>] snd_usb_pcm_prepare+0x17d/0x560 [snd_usb_audio] Jan 12 23:18:20 dwmobile kernel: [<ffffffff810d7af6>] ? current_fs_time+0x16/0x60 Jan 12 23:18:20 dwmobile kernel: [<ffffffffa028627b>] snd_pcm_do_prepare+0x1b/0x30 [snd_pcm] Jan 12 23:18:20 dwmobile kernel: [<ffffffffa0285e4f>] snd_pcm_action_single+0x2f/0x70 [snd_pcm] Jan 12 23:18:20 dwmobile kernel: [<ffffffffa0285f06>] snd_pcm_action_nonatomic+0x76/0x80 [snd_pcm] Jan 12 23:18:20 dwmobile kernel: [<ffffffffa0288c18>] snd_pcm_common_ioctl1+0x688/0xbc0 [snd_pcm] Jan 12 23:18:20 dwmobile kernel: [<ffffffffa0289268>] snd_pcm_playback_ioctl1+0x118/0x280 [snd_pcm] Jan 12 23:18:20 dwmobile kernel: [<ffffffffa0289404>] snd_pcm_playback_ioctl+0x34/0x40 [snd_pcm] Jan 12 23:18:20 dwmobile kernel: [<ffffffff811e23e0>] do_vfs_ioctl+0x2d0/0x4b0 Jan 12 23:18:20 dwmobile kernel: [<ffffffff811cf2ac>] ? vfs_write+0x18c/0x200 Jan 12 23:18:20 dwmobile kernel: [<ffffffff811e2641>] SyS_ioctl+0x81/0xa0 Jan 12 23:18:20 dwmobile kernel: [<ffffffff81554ca9>] system_call_fastpath+0x12/0x17 Jan 12 23:18:20 dwmobile kernel: ---[ end trace e6531ceb4fd4f2e5 ]--- Jan 12 23:18:20 dwmobile kernel: ------------[ cut here ]------------ >> >> amidi -l >> >> Dir Device Name >> IO hw:0,0,0 USB X-Session MIDI 1 >> I hw:0,0,1 USB X-Session MIDI 2 >> IO hw:2,0,0 TR-8 MIDI 1 # this device >> IO hw:2,0,1 TR-8 MIDI 2 # this device >> >> >> ./alsacap -d hw:2,0 >> >> *** Exploring configuration space of device 'hw:1,0' for playback *** >> 4 channels >> Sampling rate 96000 Hz >> Sample formats: S32_LE >> Significant bits: 32 >> >> >> Bus 002 Device 005: ID 0582:017c Roland Corp. >> Device Descriptor: >> bLength 18 >> bDescriptorType 1 >> bcdUSB 2.00 >> bDeviceClass 255 Vendor Specific Class >> bDeviceSubClass 0 >> bDeviceProtocol 255 >> bMaxPacketSize0 64 >> idVendor 0x0582 Roland Corp. >> idProduct 0x017c >> bcdDevice 1.00 >> iManufacturer 1 Roland >> iProduct 2 TR-8 >> iSerial 0 >> bNumConfigurations 1 >> Configuration Descriptor: >> bLength 9 >> bDescriptorType 2 >> wTotalLength 176 >> bNumInterfaces 4 >> bConfigurationValue 1 >> iConfiguration 0 >> bmAttributes 0xc0 >> Self Powered >> MaxPower 0mA >> Interface Descriptor: >> bLength 9 >> bDescriptorType 4 >> bInterfaceNumber 0 >> bAlternateSetting 0 >> bNumEndpoints 0 >> bInterfaceClass 255 Vendor Specific Class >> bInterfaceSubClass 255 Vendor Specific Subclass >> bInterfaceProtocol 0 >> iInterface 0 >> Interface Descriptor: >> bLength 9 >> bDescriptorType 4 >> bInterfaceNumber 1 >> bAlternateSetting 0 >> bNumEndpoints 0 >> bInterfaceClass 255 Vendor Specific Class >> bInterfaceSubClass 2 >> bInterfaceProtocol 2 >> iInterface 0 >> # <06 24 F1 02 <inputs> <outputs>> from patch comment by Clemens Ladisch >> ** UNRECOGNIZED: 06 24 f1 01 00 00 >> # Audio OUT 4 channels? >> Interface Descriptor: >> bLength 9 >> bDescriptorType 4 >> bInterfaceNumber 1 >> bAlternateSetting 1 >> bNumEndpoints 1 >> bInterfaceClass 255 Vendor Specific Class >> bInterfaceSubClass 2 >> bInterfaceProtocol 2 >> iInterface 0 >> # seen in other Roland devices >> # http://ubuntuforums.org/archive/index.php/t-1905531.html >> ** UNRECOGNIZED: 07 24 01 01 00 01 00 >> ** UNRECOGNIZED: 0b 24 02 01 04 04 18 01 00 77 01 >> Endpoint Descriptor: >> bLength 7 >> bDescriptorType 5 >> bEndpointAddress 0x0d EP 13 OUT >> bmAttributes 5 >> Transfer Type Isochronous # audio >> Synch Type Asynchronous >> Usage Type Data >> # 4x 24-bit 96kHz == 56 bytes per 4 channels (or 32-bit?) >> wMaxPacketSize 0x00e0 1x 224 bytes >> bInterval 1 >> Interface Descriptor: >> bLength 9 >> bDescriptorType 4 >> bInterfaceNumber 2 >> bAlternateSetting 0 >> bNumEndpoints 0 >> bInterfaceClass 255 Vendor Specific Class >> bInterfaceSubClass 2 >> bInterfaceProtocol 1 >> iInterface 0 >> # Audio IN: 14 channels? >> Interface Descriptor: >> bLength 9 >> bDescriptorType 4 >> bInterfaceNumber 2 >> bAlternateSetting 1 >> bNumEndpoints 1 >> bInterfaceClass 255 Vendor Specific Class >> bInterfaceSubClass 2 >> bInterfaceProtocol 1 >> iInterface 0 >> ** UNRECOGNIZED: 07 24 01 07 00 01 00 >> ** UNRECOGNIZED: 0b 24 02 01 0e 04 18 01 00 77 01 >> Endpoint Descriptor: >> bLength 7 >> bDescriptorType 5 >> bEndpointAddress 0x8e EP 14 IN >> bmAttributes 37 >> Transfer Type Isochronous >> Synch Type Asynchronous >> Usage Type Implicit feedback Data >> # 14 channels x 56 bytes per transfer 96kHz, 24-bit? >> wMaxPacketSize 0x0310 1x 784 bytes >> bInterval 1 >> # MIDI? >> Interface Descriptor: >> bLength 9 >> bDescriptorType 4 >> bInterfaceNumber 3 >> bAlternateSetting 0 >> bNumEndpoints 2 >> bInterfaceClass 255 Vendor Specific Class >> bInterfaceSubClass 3 >> bInterfaceProtocol 0 >> iInterface 0 >> # <06 24 F1 02 <inputs> <outputs>> 2 in 2 out >> ** UNRECOGNIZED: 06 24 f1 02 02 02 >> Endpoint Descriptor: >> bLength 7 >> bDescriptorType 5 >> bEndpointAddress 0x03 EP 3 OUT >> bmAttributes 2 >> Transfer Type Bulk >> Synch Type None >> Usage Type Data >> wMaxPacketSize 0x0200 1x 512 bytes >> bInterval 1 >> Endpoint Descriptor: >> bLength 7 >> bDescriptorType 5 >> bEndpointAddress 0x84 EP 4 IN >> bmAttributes 2 >> Transfer Type Bulk >> Synch Type None >> Usage Type Data >> wMaxPacketSize 0x0200 1x 512 bytes >> bInterval 0 >> # MIDI: same endpoints needs Bulk AND Interrupt packet types? >> Interface Descriptor: >> bLength 9 >> bDescriptorType 4 >> bInterfaceNumber 3 >> bAlternateSetting 1 >> bNumEndpoints 2 >> bInterfaceClass 255 Vendor Specific Class >> bInterfaceSubClass 3 >> bInterfaceProtocol 0 >> iInterface 0 >> Endpoint Descriptor: >> bLength 7 >> bDescriptorType 5 >> bEndpointAddress 0x03 EP 3 OUT >> bmAttributes 3 >> Transfer Type Interrupt >> Synch Type None >> Usage Type Data >> wMaxPacketSize 0x0200 1x 512 bytes >> bInterval 1 >> Endpoint Descriptor: >> bLength 7 >> bDescriptorType 5 >> bEndpointAddress 0x85 EP 5 IN >> bmAttributes 3 >> Transfer Type Interrupt >> Synch Type None >> Usage Type Data >> wMaxPacketSize 0x0200 1x 512 bytes >> bInterval 1 >> Device Qualifier (for other device speed): >> bLength 10 >> bDescriptorType 6 >> bcdUSB 2.00 >> bDeviceClass 255 Vendor Specific Class >> bDeviceSubClass 0 >> bDeviceProtocol 255 >> bMaxPacketSize0 64 >> bNumConfigurations 1 >> can't get debug descriptor: Resource temporarily unavailable >> Device Status: 0x0001 >> Self Powered >> >> >> >> A summary of some USB packets during idle, start playback, stop >> playback, start recording, stop recording. Playback and record was >> probably only using a pair of stereo channels IIRC (maybe even one mono >> channel). I could do the same with all channels if that would be of any use. >> >> >> - at all times, polling seems to be 'Bulk or Interrupt Transfer' to >> Device Object FFFFFA800BBF1C00h (00000094) with I/O Request Packet >> FFFFFA800B97B830h >> >> >> - start playback: 'Select Interface', 'Ifc 1 / Alt 1' and 'Ifc 2 / Alt >> 1', Device Object: FFFFFA800BBF1C00h, I/O Request Packet: FFFFFA800778E780h >> >> 1x 'Create' using I/O Request Packet: FFFFFA800C7A2480h to Device >> Objects: FFFFFA800C060C20h (00000095) and FFFFFA800BE30600h (00000096) >> >> >> - during playback: lots of 'Isoch Transfer' to FFFFFA800BBF1C00h with >> nine different I/O Request Packets >> >> >> - stop playback: 'Select Interface', 'Ifc 1 / Alt 0' and 'Ifc 1 / Alt >> 0', Device Object: FFFFFA800BBF1C00h; I/O Request Packet: >> FFFFFA8007552810h >> >> 1x 'Close' using I/O Request Packet: FFFFFA800C7A2480h to Device >> Objects: FFFFFA800C060C20h and FFFFFA800BE30600h; some 'Cleanups' that >> return 'Invalid Device Request' >> >> >> - start recording: 'Select Interface', 'Ifc 1 / Alt 1' and 'Ifc 2 / Alt >> 1', Device Object: FFFFFA800BBF1C00h, I/O Request Packet: FFFFFA800778E780h >> >> 5x 'Create' using I/O Request Packet: FFFFFA800C0E9C60h to Device >> Objects: FFFFFA800C060C20h and FFFFFA800BE30600h >> 2x 'Create' using I/O Request Packet: FFFFFA800C60A010h to Device >> Objects: FFFFFA800C060C20h and FFFFFA800BE30600h >> 1x 'Create' using I/O Request Packet: FFFFFA800754F010h to Device >> Objects: FFFFFA800C060C20h and FFFFFA800BE30600h >> 2x 'Close' using I/O Request Packet: FFFFFA800C0E9C60h to Device >> Objects: FFFFFA800C060C20h and FFFFFA800BE30600h >> >> >> - during recording: lots of 'Isoch Transfer' to FFFFFA800BBF1C00h with >> nine different I/O Request Packets >> >> >> - stop recording: 'Select Interface', 'Ifc 1 / Alt 0' and 'Ifc 1 / Alt >> 0', Device Object: FFFFFA800BBF1C00h; I/O Request Packet: >> FFFFFA8007552810h >> >> 4x 'Close' using I/O Request Packet: FFFFFA8007543C60h to Device >> Objects: FFFFFA800C060C20h and FFFFFA800BE30600h >> >>

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[alsa-devel] [RFC PATCH 0/3]
by Yakir Yang 10 Feb '15

10 Feb '15

RK3288 hdmi eye-diagram test would fail when pixel clock is 148.5MHz, and single-ended test would failed when display mode is 74.25MHz. - Fix some code style, leave space for next patches. - For hdmi eye-diagram test, we turn on the Transmitter Trailer-B and improve slopeboost to 25%-30% decrease. - For hdmi single-ended test, we set CKLVL & TXLVL to 17 when pixel clock is 74.25MHz, keep CKLVL & TXLVL to 13 when pixel clock is 148.5MHz. Yakir Yang (3): drm: bridge/dw_hdmi: fixed codec style drm: bridge/dw_hdmi_rockchip: improve hdmi eye-diagram test drm: bridge/dw_hdmi: improve hdmi single-end test drivers/gpu/drm/bridge/dw_hdmi.c | 20 ++++++++++---------- drivers/gpu/drm/imx/dw_hdmi-imx.c | 12 ++++++------ drivers/gpu/drm/rockchip/dw_hdmi-rockchip.c | 14 +++++++------- include/drm/bridge/dw_hdmi.h | 5 +++-- 4 files changed, 26 insertions(+), 25 deletions(-) -- 1.7.9.5

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[alsa-devel] [RESEND PATCH 0/6] Add support for the WM8280 and WM8281
by Charles Keepax 10 Feb '15

10 Feb '15

This set of patches adds support for the WM8280 and WM8281 codecs to the Wolfson Arizona drivers. Just another resend on this series, if it would expedite the process I would be happy if we could just merge the MFD patches then I can push the other patches through their respective trees once that hits mainline, but at the moment all the patches are gated by a build dependency on the MFD patches. Thanks, Charles Richard Fitzgerald (6): mfd: arizona: Add support for WM8280/WM8281 Documentation: devicetree: arizona: Add bindings for WM8280 regulator: arizona-micsupp: Add support for WM8280/WM8281 gpio: arizona: Add support for WM8280/WM8281 extcon: arizona: Add support for WM8280/WM8281 ASoC: arizona: Add support for WM8280/WM8281 Documentation/devicetree/bindings/mfd/arizona.txt | 15 +++++++++++---- drivers/extcon/extcon-arizona.c | 1 + drivers/gpio/gpio-arizona.c | 1 + drivers/mfd/Kconfig | 5 +++-- drivers/mfd/arizona-core.c | 15 +++++++++++++-- drivers/mfd/arizona-i2c.c | 2 ++ drivers/mfd/arizona-irq.c | 1 + drivers/mfd/arizona-spi.c | 2 ++ drivers/regulator/arizona-micsupp.c | 1 + include/linux/mfd/arizona/core.h | 1 + sound/soc/codecs/arizona.c | 2 ++ 11 files changed, 38 insertions(+), 8 deletions(-) -- 1.7.2.5

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[alsa-devel] [PATCH] ALSA: hda - Add docking station support for another HP machine
by David Henningsson 10 Feb '15

10 Feb '15

From: TienFu Chen <tienfu.chen(a)canonical.com> BugLink: https://bugs.launchpad.net/bugs/1412800 Signed-off-by: TienFu Chen <tienfu.chen(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- 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 0403061..ddb9308 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4950,6 +4950,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x226e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2271, "HP", ALC286_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x2272, "HP", ALC280_FIXUP_HP_DOCK_PINS), + SND_PCI_QUIRK(0x103c, 0x2273, "HP", ALC280_FIXUP_HP_DOCK_PINS), SND_PCI_QUIRK(0x103c, 0x229e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22b2, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22b7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), -- 1.9.1

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[alsa-devel] [PATCH v3] ASoC: atmel_ssc_dai: Allow more rates
by Peter Rosin 10 Feb '15

10 Feb '15

From: Peter Rosin <peda(a)axentia.se> When the SSC acts as BCK master, use a ratnum rule to limit the rate instead of only doing the standard rates. When the SSC acts as BCK slave, allow any BCK frequency up to the SSC master clock, divided by either of 2, 3 or 6. Put a cap at 384kHz. Who's /ever/ going to need more than that? The divider of 2, 3 or 6 is selected based on the Serial Clock Ratio Considerations section from the SSC documentation: The Transmitter and the Receiver can be programmed to operate with the clock signals provided on either the TK or RK pins. This allows the SSC to support many slave-mode data transfers. In this case, the maximum clock speed allowed on the RK pin is: - Peripheral clock divided by 2 if Receiver Frame Synchro is input - Peripheral clock divided by 3 if Receiver Frame Synchro is output In addition, the maximum clock speed allowed on the TK pin is: - Peripheral clock divided by 6 if Transmit Frame Synchro is input - Peripheral clock divided by 2 if Transmit Frame Synchro is output Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/atmel/atmel_ssc_dai.c | 111 +++++++++++++++++++++++++++++++++++++-- sound/soc/atmel/atmel_ssc_dai.h | 1 + 2 files changed, 108 insertions(+), 4 deletions(-) Changes since v2: - Killed the 500ppm reduction. Just trust the nominal clock rates of the given clocks. - Don't multiply the given SSC clk rate by two to get to the SSC master clock rate. That factor was the result of some confusion on my part. The SSC clk *is* the SSC master clock. Changes since v1: - I have checked all Atmel datasheets I could get my hands on (and that seemed relevant), and in every instance where they have described the SSC, the description have the exact rate limitations on the RK and TK pin that I have implemented here. - Improved the comments. - Rebased on top of the latest patches from Bo Chen. diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 379ac2a6ab16..6b8e64899205 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -187,6 +187,94 @@ static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } +/* + * When the bit clock is input, limit the maximum rate according to the + * Serial Clock Ratio Considerations section from the SSC documentation: + * + * The Transmitter and the Receiver can be programmed to operate + * with the clock signals provided on either the TK or RK pins. + * This allows the SSC to support many slave-mode data transfers. + * In this case, the maximum clock speed allowed on the RK pin is: + * - Peripheral clock divided by 2 if Receiver Frame Synchro is input + * - Peripheral clock divided by 3 if Receiver Frame Synchro is output + * In addition, the maximum clock speed allowed on the TK pin is: + * - Peripheral clock divided by 6 if Transmit Frame Synchro is input + * - Peripheral clock divided by 2 if Transmit Frame Synchro is output + * + * When the bit clock is output, limit the rate according to the + * SSC divider restrictions. + */ +static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct atmel_ssc_info *ssc_p = rule->private; + struct ssc_device *ssc = ssc_p->ssc; + struct snd_interval *i = hw_param_interval(params, rule->var); + struct snd_interval t; + struct snd_ratnum r = { + .den_min = 1, + .den_max = 4095, + .den_step = 1, + }; + unsigned int num = 0, den = 0; + int frame_size; + int mck_div = 2; + int ret; + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) + return frame_size; + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFS: + if ((ssc_p->dir_mask & SSC_DIR_MASK_CAPTURE) + && ssc->clk_from_rk_pin) + /* Receiver Frame Synchro (i.e. capture) + * is output (format is _CFS) and the RK pin + * is used for input (format is _CBM_). + */ + mck_div = 3; + break; + + case SND_SOC_DAIFMT_CBM_CFM: + if ((ssc_p->dir_mask & SSC_DIR_MASK_PLAYBACK) + && !ssc->clk_from_rk_pin) + /* Transmit Frame Synchro (i.e. playback) + * is input (format is _CFM) and the TK pin + * is used for input (format _CBM_ but not + * using the RK pin). + */ + mck_div = 6; + break; + } + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + r.num = ssc_p->mck_rate / mck_div / frame_size; + + ret = snd_interval_ratnum(i, 1, &r, &num, &den); + if (ret >= 0 && den && rule->var == SNDRV_PCM_HW_PARAM_RATE) { + params->rate_num = num; + params->rate_den = den; + } + break; + + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBM_CFM: + t.min = 8000; + t.max = ssc_p->mck_rate / mck_div / frame_size; + t.openmin = t.openmax = 0; + t.integer = 0; + ret = snd_interval_refine(i, &t); + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} /*-------------------------------------------------------------------------*\ * DAI functions @@ -200,6 +288,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; struct atmel_pcm_dma_params *dma_params; int dir, dir_mask; + int ret; pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", ssc_readl(ssc_p->ssc->regs, SR)); @@ -207,6 +296,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, /* Enable PMC peripheral clock for this SSC */ pr_debug("atmel_ssc_dai: Starting clock\n"); clk_enable(ssc_p->ssc->clk); + ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk); /* Reset the SSC to keep it at a clean status */ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); @@ -219,6 +309,17 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, dir_mask = SSC_DIR_MASK_CAPTURE; } + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + atmel_ssc_hw_rule_rate, + ssc_p, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret < 0) { + dev_err(dai->dev, "Failed to specify rate rule: %d\n", ret); + return ret; + } + dma_params = &ssc_dma_params[dai->id][dir]; dma_params->ssc = ssc_p->ssc; dma_params->substream = substream; @@ -783,8 +884,6 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) # define atmel_ssc_resume NULL #endif /* CONFIG_PM */ -#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) - #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -804,12 +903,16 @@ static struct snd_soc_dai_driver atmel_ssc_dai = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .capture = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .ops = &atmel_ssc_dai_ops, }; diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index b1f08d511495..80b153857a88 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -115,6 +115,7 @@ struct atmel_ssc_info { unsigned short rcmr_period; struct atmel_pcm_dma_params *dma_params[2]; struct atmel_ssc_state ssc_state; + unsigned long mck_rate; }; int atmel_ssc_set_audio(int ssc_id); -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: omap: Kconfig: Support for omap5-uevm analog audio
by Peter Ujfalusi 10 Feb '15

10 Feb '15

The analog audio is supported via omap-abe-twl6040 machine driver on omap5-uevm. Update the Kconfig file to reflect this and select the Palmas clock driver which is providing the 32K clock for audio on omap5-uevm. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/Kconfig | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index a2cd3486ac55..e7c78b0406b5 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -100,17 +100,19 @@ config SND_OMAP_SOC_OMAP_TWL4030 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || COMPILE_TEST) + depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || SOC_OMAP5 || COMPILE_TEST) select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 select SND_SOC_DMIC + select COMMON_CLK_PALMAS if SOC_OMAP5 help Say Y if you want to add support for SoC audio on OMAP boards using ABE and twl6040 codec. This driver currently supports: - SDP4430/Blaze boards - PandaBoard (4430) - PandaBoardES (4460) + - omap5-uevm (5432) config SND_OMAP_SOC_OMAP3_PANDORA tristate "SoC Audio support for OMAP3 Pandora" -- 2.2.2

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[alsa-devel] [PATCH] ALSA: control: confirm to return all identical information in 'activate' event data
by Takashi Sakamoto 10 Feb '15

10 Feb '15

When event originator doesn't set numerical ID in identical information, the event data includes no numerical ID, thus userspace applications cannot identify the control just by unique ID in event data. This commit fix this bug so as the event data includes all of identical information. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/control.c b/sound/core/control.c index 6a72b3e..884fddd 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -587,6 +587,7 @@ int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, } ret = 1; unlock: + *id = kctl->id; up_write(&card->controls_rwsem); if (ret > 0) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, id); -- 2.1.0

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Re: [alsa-devel] Regression for Lenovo X240 sound in 3.18
by Takashi Iwai 10 Feb '15

10 Feb '15

At Mon, 9 Feb 2015 12:38:00 +0100 (CET), Jan Engelhardt wrote: > > > On Monday 2015-02-09 11:29, Takashi Iwai wrote: > >> > >> On 3.19-rc3+ [eb74926920cfa756087a82e0b081df837177cb95] I confirm > >> all the volume bars/switches that went missing for me are back again. > >> > >> As said, all is fine here. > > > >Sorry, I've forgotten this pending issue. > >Could you confirm that the patch below works for you? Then I'll queue > >up for 3.20-rc1 and backport appropriately. > > I have no problems with 3.19-rc6. > Your patch does not appear to break anything, but it does not appear > to fix something, either. It does change the controls again though. > I suggest not to apply it. > > http://picpaste.de/3_19_rc6-NgtzogKW.png > http://picpaste.de/3_19_rc6_YOURPATCH-y6IcoXpE.png OK, if 3.19 works as is, then there is no reason to change there. Takashi

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[alsa-devel] [PATCH lib] control: complement comment for subscribe API
by Takashi Sakamoto 09 Feb '15

09 Feb '15

From: Takashi Sakamoto <mocchi(a)Mocchi64.miraclelinux.com> When minus value is given as a parameter, the API returns whether the caller has already subscribed or not. This commit adds a comment about this so as better documentation. Signed-off-by: Takashi Sakamoto <mocchi(a)Mocchi64.miraclelinux.com> --- src/control/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/control/control.c b/src/control/control.c index dd428a1..4a28cf6 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -213,7 +213,7 @@ int snd_ctl_poll_descriptors_revents(snd_ctl_t *ctl, struct pollfd *pfds, unsign /** * \brief Ask to be informed about events (poll, #snd_async_add_ctl_handler, #snd_ctl_read) * \param ctl CTL handle - * \param subscribe 0 = unsubscribe, 1 = subscribe + * \param subscribe 0 = unsubscribe, 1 = subscribe, -1 = check subscribe or not * \return 0 on success otherwise a negative error code */ int snd_ctl_subscribe_events(snd_ctl_t *ctl, int subscribe) -- 2.1.0

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[alsa-devel] [PATCH] ALSA: control: fix failure to return new numerical ID in 'replace' event data
by Takashi Sakamoto 09 Feb '15

09 Feb '15

In 'replace' event data, numerical ID of control is always invalid. This commit fix this bug so as the event data has renewed numerical ID for control. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/control.c b/sound/core/control.c index 884fddd..ab74b7e 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -440,6 +440,7 @@ add: card->controls_count += kcontrol->count; kcontrol->id.numid = card->last_numid + 1; card->last_numid += kcontrol->count; + id = kcontrol->id; count = kcontrol->count; up_write(&card->controls_rwsem); for (idx = 0; idx < count; idx++, id.index++, id.numid++) -- 2.1.0

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[alsa-devel] [PATCH] ALSA: control: evaluate macro instead of numerical value
by Takashi Sakamoto 09 Feb '15

09 Feb '15

SNDRV_CTL_TLV_OP_XXX is defined but not used in core code. Instead, raw numerical value is evaluated. This commit replaces these values to these macros for better looking. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 8aae6ea..6a72b3e 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1077,7 +1077,7 @@ static int snd_ctl_elem_user_tlv(struct snd_kcontrol *kcontrol, int change = 0; void *new_data; - if (op_flag > 0) { + if (op_flag == SNDRV_CTL_TLV_OP_WRITE) { if (size > 1024 * 128) /* sane value */ return -EINVAL; @@ -1333,9 +1333,12 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file, goto __kctl_end; } vd = &kctl->vd[tlv.numid - kctl->id.numid]; - if ((op_flag == 0 && (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_READ) == 0) || - (op_flag > 0 && (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_WRITE) == 0) || - (op_flag < 0 && (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND) == 0)) { + if ((op_flag == SNDRV_CTL_TLV_OP_READ && + (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_READ) == 0) || + (op_flag == SNDRV_CTL_TLV_OP_WRITE && + (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_WRITE) == 0) || + (op_flag == SNDRV_CTL_TLV_OP_CMD && + (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND) == 0)) { err = -ENXIO; goto __kctl_end; } @@ -1352,7 +1355,7 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file, return 0; } } else { - if (op_flag) { + if (op_flag != SNDRV_CTL_TLV_OP_READ) { err = -ENXIO; goto __kctl_end; } -- 2.1.0

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[alsa-devel] [PATCH v5 00/10] audio timestamping evolutions
by Pierre-Louis Bossart 09 Feb '15

09 Feb '15

This series of patches was inspired by recent threads on the alsa mailing list, as well issues detected with existing and upcoming hardware: 1. there was a request from the PulseAudio community to get more information from drivers to make rewinds safer. While the conclusion was that it's nearly impossible for a driver developer to provide this information, there are however ways to assess the granularity of the hw_ptr updates using timestamping capabilities, and indirectly understand how safe rewinds might be. 2. There was also a request to add a start_at capability based either on system hr timers or a wall clock, which requires a means to expose both types of information to applications. Rather than adding new sets of timestamps, it is suggested the new start_at functionality relies on the new definition provides by these patches 3. For new hardware, there is a neeed to let low-level drivers handle timestamping instead of having the ALSA core do it. Similarly there is a need to let the low-level driver update the initial estimate for the trigger timestamp if there are delays to start a stream (eg. with USB) These patches try to provide an answer to these multiple needs by building on the work done two years ago to expose wall clock information to applications. The evolution is to let application select which audio timestamp they are interested in, track the delay and drifts between recurring measurements and get, when possible, an idea of the accuracy of the underlying hardware. A backwards compatible mode is provided in case userspace doesn't provide any timestamp selection (results based on HDAudio wallclock for playback, hw_ptr in all other cases). The first 4 patches are corrections for misses in the way the system and trigger timestamps are handled, and the last 6 provide the additional audio timestamping selection. A second batch is planned to enable hardware capabilities in a low-level drivers. A corresponding set of patches is available for alsa-lib. V2 changes: trigger_tstamp: trigger_tstamp_latched, pending redefined as bool trigger_tstamp_latched reset in snd_pcm_pre_start() audio_ts_config, report: keep separate structure but use different bitfields for in and out. use u32 instead of __u32, add comments that these structures are internal COMPAT backwards compatible mode, uses WALL_CLOCK/LINK for HDAudio playback and DEFAULT (hw_ptr) everywhere else INFO bits: reclaimed 32-bits from hw_params, renamed as info_ext moved all timestamp info to info_ext snd_pcm_status: read only 32-bit audio_tstamp_data, ignore all other fields V3/4: Addressed feedback from Jaroslav: no change to STATUS ioctl, new functionality introduced in STATUS_EXT ioctl with r/w params. bumped PCM protocol for detection on STATUS_EXT in userspace used 32-bit word for accuracy report, no mantissa/exponent packing rolled back info_ext changes, all INFO fields remain in same word Merged comments from Liam (code simplifications) fixed packing Added driver_tstamp field in case there is a delay in passing the tstamp and audio_tstamp over IPC. v5: no functional change, only addressed Takashi comments on patch 2,3,4 fixed used of bool with true/false and ! in comparisons move trigger_tstamp_pending_update flag to snd_usb_substream, will be used by ALSA core when other devices need it Pierre-Louis Bossart (10): ALSA: core: don't override timestamp unconditionally ALSA: core: allow for trigger_tstamp snapshot in .trigger ALSA: hda: read trigger_timestamp immediately after starting DMA ALSA: usb: update trigger timestamp on first non-zero URB submitted ALSA: core: selection of audio_tstamp type and accuracy reports ALSA: core: pass audio tstamp config from userspace ALSA: core: pass audio tstamp config from userspace in compat mode ALSA: core: replace .wall_clock by .get_time_info ALSA: hda: replace .wallclock by .get_time_info ALSA: bump PCM protocol to 2.0.13 Documentation/sound/alsa/timestamping.txt | 200 ++++++++++++++++++++++++++++++ include/sound/pcm.h | 66 +++++++++- include/uapi/sound/asound.h | 36 +++++- sound/core/pcm_compat.c | 36 +++++- sound/core/pcm_lib.c | 88 ++++++++----- sound/core/pcm_native.c | 58 ++++++++- sound/pci/hda/hda_controller.c | 43 +++++-- sound/usb/card.h | 2 + sound/usb/pcm.c | 9 ++ 9 files changed, 479 insertions(+), 59 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: atmel_ssc_dai: Allow more rates
by Peter Rosin 09 Feb '15

09 Feb '15

From: Peter Rosin <peda(a)axentia.se> When the SSC acts as BCK master, use a ratnum rule to limit the rate instead of only doing the standard rates. When the SSC acts as BCK slave, allow any BCK frequency up to within 500ppm of the SSC master clock, possibly divided by 2, 3 or 6. Put a cap at 384kHz. Who's /ever/ going to need more than that? The divider of 2, 3 or 6 is selected based on the Serial Clock Ratio Considerations section from the SSC documentation: The Transmitter and the Receiver can be programmed to operate with the clock signals provided on either the TK or RK pins. This allows the SSC to support many slave-mode data transfers. In this case, the maximum clock speed allowed on the RK pin is: - Peripheral clock divided by 2 if Receiver Frame Synchro is input - Peripheral clock divided by 3 if Receiver Frame Synchro is output In addition, the maximum clock speed allowed on the TK pin is: - Peripheral clock divided by 6 if Transmit Frame Synchro is input - Peripheral clock divided by 2 if Transmit Frame Synchro is output Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/atmel/atmel_ssc_dai.c | 113 +++++++++++++++++++++++++++++++++++++-- sound/soc/atmel/atmel_ssc_dai.h | 1 + 2 files changed, 110 insertions(+), 4 deletions(-) Changes since v1: - I have checked all Atmel datasheets I could get my hands on (and that seemed relevant), and in every instance where they have described the SSC, the description have the exact rate limitations on the RK and TK pin that I have implemented here. - Improved the comments. - Rebased on top of the latest patches from Bo Chen. One thing remains a bit unclear, and that is the 500ppm deduction. Is that really warranted? The number was just pulled out of my hat... Cheers, Peter diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 379ac2a6ab16..767f65bab25d 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -187,6 +187,96 @@ static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } +/* + * When the bit clock is input, limit the maximum rate according to the + * Serial Clock Ratio Considerations section from the SSC documentation: + * + * The Transmitter and the Receiver can be programmed to operate + * with the clock signals provided on either the TK or RK pins. + * This allows the SSC to support many slave-mode data transfers. + * In this case, the maximum clock speed allowed on the RK pin is: + * - Peripheral clock divided by 2 if Receiver Frame Synchro is input + * - Peripheral clock divided by 3 if Receiver Frame Synchro is output + * In addition, the maximum clock speed allowed on the TK pin is: + * - Peripheral clock divided by 6 if Transmit Frame Synchro is input + * - Peripheral clock divided by 2 if Transmit Frame Synchro is output + * + * When the bit clock is output, limit the rate according to the + * SSC divider restrictions. + */ +static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct atmel_ssc_info *ssc_p = rule->private; + struct ssc_device *ssc = ssc_p->ssc; + struct snd_interval *i = hw_param_interval(params, rule->var); + struct snd_interval t; + struct snd_ratnum r = { + .den_min = 1, + .den_max = 4095, + .den_step = 1, + }; + unsigned int num = 0, den = 0; + int frame_size; + int mck_div = 2; + int ret; + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) + return frame_size; + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFS: + if ((ssc_p->dir_mask & SSC_DIR_MASK_CAPTURE) + && ssc->clk_from_rk_pin) + /* Receiver Frame Synchro (i.e. capture) + * is output (format is _CFS) and the RK pin + * is used for input (format is _CBM_). + */ + mck_div = 3; + break; + + case SND_SOC_DAIFMT_CBM_CFM: + if ((ssc_p->dir_mask & SSC_DIR_MASK_PLAYBACK) + && !ssc->clk_from_rk_pin) + /* Transmit Frame Synchro (i.e. playback) + * is input (format is _CFM) and the TK pin + * is used for input (format _CBM_ but not + * using the RK pin). + */ + mck_div = 6; + break; + } + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + r.num = ssc_p->mck_rate / mck_div / frame_size; + + ret = snd_interval_ratnum(i, 1, &r, &num, &den); + if (ret >= 0 && den && rule->var == SNDRV_PCM_HW_PARAM_RATE) { + params->rate_num = num; + params->rate_den = den; + } + break; + + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBM_CFM: + t.min = 8000; + t.max = ssc_p->mck_rate / mck_div / frame_size; + /* Take away 500ppm, just to be on the safe side. */ + t.max -= t.max / 2000; + t.openmin = t.openmax = 0; + t.integer = 0; + ret = snd_interval_refine(i, &t); + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} /*-------------------------------------------------------------------------*\ * DAI functions @@ -200,6 +290,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; struct atmel_pcm_dma_params *dma_params; int dir, dir_mask; + int ret; pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", ssc_readl(ssc_p->ssc->regs, SR)); @@ -207,6 +298,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, /* Enable PMC peripheral clock for this SSC */ pr_debug("atmel_ssc_dai: Starting clock\n"); clk_enable(ssc_p->ssc->clk); + ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk) * 2; /* Reset the SSC to keep it at a clean status */ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); @@ -219,6 +311,17 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, dir_mask = SSC_DIR_MASK_CAPTURE; } + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + atmel_ssc_hw_rule_rate, + ssc_p, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret < 0) { + dev_err(dai->dev, "Failed to specify rate rule: %d\n", ret); + return ret; + } + dma_params = &ssc_dma_params[dai->id][dir]; dma_params->ssc = ssc_p->ssc; dma_params->substream = substream; @@ -783,8 +886,6 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) # define atmel_ssc_resume NULL #endif /* CONFIG_PM */ -#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) - #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -804,12 +905,16 @@ static struct snd_soc_dai_driver atmel_ssc_dai = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .capture = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .ops = &atmel_ssc_dai_ops, }; diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index b1f08d511495..80b153857a88 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -115,6 +115,7 @@ struct atmel_ssc_info { unsigned short rcmr_period; struct atmel_pcm_dma_params *dma_params[2]; struct atmel_ssc_state ssc_state; + unsigned long mck_rate; }; int atmel_ssc_set_audio(int ssc_id); -- 1.7.10.4

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[alsa-devel] [patch] ALSA: riptide: off by one in snd_riptide_joystick_probe()
by Dan Carpenter 09 Feb '15

09 Feb '15

There is an off by one bug in snd_riptide_joystick_probe() because we do the "dev++" toward the start of the function instead of waiting until the end. The impact of this bug is: 1) A static checker warning. 2) If you connect 32 joysticks to your computer they will fail to load with a different error message than intended. 3) If you pass a module option to specify a non-default joystick port then the parameter array has to be shifted one element. The first two are not serious. For the third one, it's probably too late to change it since the bug was introduced in 2009. Either no one noticed and we can leave it alone, or if they did notice, they probably have implemented work arounds so we can't change it now without confusing them. So instead of fixing it the "theoretically correct way", I have decided to just work around it by making the joystick_port[] one element larger. This silences the static checker warning and doesn't affect user space. Fixes: db1005ec6ff8 ('ALSA: riptide - Fix joystick resource handling') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 29f2827..324e34c 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -125,7 +125,11 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; #ifdef SUPPORT_JOYSTICK -static int joystick_port[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS - 1)] = 0x200 }; +/* + * The joystick_port accounting is off by one but I'm afraid to change it for + * fear of breaking userspace. + */ +static int joystick_port[SNDRV_CARDS + 1] = { [0 ... (SNDRV_CARDS)] = 0x200 }; #endif static int mpu_port[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS - 1)] = 0x330 }; static int opl3_port[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS - 1)] = 0x388 };

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Re: [alsa-devel] Regression for Lenovo X240 sound in 3.18
by Takashi Iwai 09 Feb '15

09 Feb '15

At Sat, 10 Jan 2015 13:10:30 +0100 (CET), Jan Engelhardt wrote: > > > On Wednesday 2014-12-31 09:51, Takashi Iwai wrote: > >> > > >> >Ah, I see you disable the automute. So the headphone plug shouldn't > >> >matter. > >> > >> Changing the "Auto-Mute" control does not do anything; the system always > >> exhibits Automute behavior (if that's a bug, it's one which I did not > >> care about so far). > > > >Oh, that's really interesting. I didn't expect such a > >hardware-controlled automute nowadays. We need to investigate this > >more. > >[...] > >[Volumes] On 3.19, this was renamed differently, so that "PCM" can > >appear again by a softvol item like before. > > On 3.19-rc3+ [eb74926920cfa756087a82e0b081df837177cb95] I confirm > all the volume bars/switches that went missing for me are back again. > > In particular, 3.19 shows a new "Speaker+LO" no-switch volume bar > control, and a "Speaker" no-volume switch control. Works for me, > though I wonder if the change was necessary. (In 3.17, the switch and > volume were combined in one control.) > > As said, all is fine here. Sorry, I've forgotten this pending issue. Could you confirm that the patch below works for you? Then I'll queue up for 3.20-rc1 and backport appropriately. thanks, Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Fix lost sound on Thinkpad X240 The commit [7c21539c45ea: ALSA: hda - Add ultra dock support for Thinkpad X240] caused a regression, the sound got lost. The machine seems requiring the fixed binding between DACs and pins, and the commit above resulted in the change of binding. As a fix, this patch adds the binding list to ensure to connect to the correct DAC. Reported-by: Jan Engelhardt <jengelh(a)inai.de> Fixes: 7c21539c45ea ('ALSA: hda - Add ultra dock support for Thinkpad X240.') Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 25 ++++++++++++++++++++++++- 1 file changed, 24 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 040306194e6d..dba31bafcd1c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4226,6 +4226,22 @@ static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec, } } +/* TP X240 seems to need fixed DAC binding; otherwise sound gets lost */ +static void alc_fixup_tpx240_dacs(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + static hda_nid_t preferred_pairs[] = { + 0x14, 0x03, + 0x15, 0x02, + 0x16, 0x02, + 0 + }; + struct alc_spec *spec = codec->spec; + spec->gen.preferred_dacs = preferred_pairs; + } +} + static void alc283_hp_automute_hook(struct hda_codec *codec, struct hda_jack_callback *jack) { @@ -4411,6 +4427,7 @@ enum { ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_FIXUP_TPT440_DOCK, + ALC292_FIXUP_TPX240_DOCK, ALC283_FIXUP_BXBT2807_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, @@ -4856,6 +4873,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST }, + [ALC292_FIXUP_TPX240_DOCK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_tpx240_dacs, + .chained = true, + .chain_id = ALC292_FIXUP_TPT440_DOCK, + }, [ALC283_FIXUP_BXBT2807_MIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5026,7 +5049,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), - SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPX240_DOCK), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), -- 2.2.2

2 1

[alsa-devel] Can we remove un-used Sound driver ?
by Kuninori Morimoto 09 Feb '15

09 Feb '15

Hi SH ML, ALSA ML We have below sound drivers in upstream kernel. But, these are not used from upstreamed board. And, these are for very old SoC. I guess Renesas doesn't/can't support these SoC/board anymore. Can we remove these sound drivers ? Because maintenance cost for these is not zero... - ${LINUX}/sound/soc/sh/ssi.c - SH7760/SH7780 - CONFIG_SND_SOC_SH4_SSI - ${LINUX}/linux/sound/soc/sh/hac.c - SH7760/SH7780 - CONFIG_SND_SOC_SH4_HAC - ${LINUX}/sound/soc/sh/sh7760-ac97.c - SH7760 - CONFIG_SND_SH7760_AC97 - ${LINUX}/sound/soc/sh/dma-sh7760.c - SH7760 - CONFIG_SND_SOC_PCM_SH7760 - ${LINUX}/sound/soc/sh/siu_dai.c *1 ${LINUX}/sound/soc/sh/siu_pcm.c ${LINUX}/sound/soc/sh/siu.h - SH7343, SH7722 - CONFIG_SND_SOC_SH4_SIU - ${LINUX}/sound/soc/sh/migor.c *2 - SH7722 - CONFIG_SND_SIU_MIGOR *1 it is registered from ${LINUX}/arch/sh/kernel/cpu/sh4a/setup-sh7722.c but, no one enabled CONFIG_SND_SOC_SH4_SIU in defconfig *2 it uses *1, but no one enabled CONFIG_SND_SIU_MIGOR Best regards --- Kuninori Morimoto

4 12

[alsa-devel] [PATCH 0/2] ASoC: rsnd: tidyup .remove method
by Kuninori Morimoto 09 Feb '15

09 Feb '15

Hi Mark These patches call missing remove/unregister functions on rsnd driver. Kuninori Morimoto (2): ASoC: rsnd: call missing snd_ctl_remove() ASoC: rsnd: call missing snd_soc_unregiter_component/platform() sound/soc/sh/rcar/core.c | 14 +++++++++++++- sound/soc/sh/rcar/dvc.c | 15 +++++++++++++++ sound/soc/sh/rcar/rsnd.h | 5 +++++ 3 files changed, 33 insertions(+), 1 deletion(-) BTW, we found ALSA SoC bind/unbind issue on latest upstream kernel. ALSA SoC is using like this. [CPU] - [CARD] - [CODEC] If we unbind CPU here, it still has CARD infomation. And, kernel doesn't call initialize function when re-bind, and it will panic when playback ------------------------- In our case, CPU : rsnd CARD : simple-card CODEC : ak4642 Current list / # 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 echo "ec500000.rcar_sound" > /sys/bus/platform/drivers/rcar_sound/unbind But, it still has unbinded 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 Re-bind. .probe function was called, but no mapping message "asoc-simple-card sound: ak4642-hifi <-> ec500000.rcar_sound mapping ok" echo "ec500000.rcar_sound" > /sys/bus/platform/drivers/rcar_sound/bind rcar_sound ec500000.rcar_sound: probed Kernel panic when playback after rebind / # aplay /home/xxx.wav ASoC: ak4642-hifi <-> No matching rates Unable to handle kernel paging request at virtual address ee196100 pgd = eebbc000 [ee196100] *pgd=6e01141e(bad) Internal error: Oops: 8000000d [#1] SMP ARM CPU: 0 PID: 936 Comm: aplay Tainted: G W 3.19.0-rc6-02875-gf8eac9a #1541 Hardware name: Generic R8A7790 (Flattened Device Tree) task: edc2adc0 ti: ee08a000 task.ti: ee08a000 PC is at 0xee196100 LR is at soc_pcm_open+0x6e8/0x7ac pc : [<ee196100>] lr : [<c0386e18>] psr: a00f0013 sp : ee08bc90 ip : ee08bc90 fp : ee08bcec r10: eeb30800 r9 : eeae7a00 r8 : ffffffea r7 : eeb30800 r6 : ee947480 r5 : ee328a00 r4 : edbf5010 r3 : ee196100 r2 : ee0dc290 r1 : ee947480 r0 : ee328a00 Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 10c5307d Table: 6ebbc06a DAC: 00000015 Process aplay (pid: 936, stack limit = 0xee08a238) Stack: (0xee08bc90 to 0xee08c000) bc80: 00000001 00000002 edbf5010 00000010 bca0: ee91d520 edbf501c 0000000c 00000000 00000000 00000001 ffffffff eeae7a00 bcc0: ee08bcf4 00000000 ee08bd10 eea3ad00 ee328ce8 ee328d00 00000000 00000001 bce0: ee08bd0c ee08bcf0 c0376a60 c038673c c0045098 ee328a00 00000000 ee328c00 bd00: ee08bd54 ee08bd10 c0376b38 c0376a18 ee08bd2c 00000000 edc2adc0 c0045098 bd20: ee328d04 ee328d04 c024dd10 ee328c00 eea3ad00 eebb3638 00000000 c0661848 bd40: eea3ad00 eea3ad00 ee08bd6c ee08bd58 c0376d04 c0376ab4 c04f179c eea3ad00 bd60: ee08bd8c ee08bd70 c0367f18 c0376ccc c0367e9c 00000000 eebb3638 ee13bec0 bd80: ee08bdbc ee08bd90 c00c8b94 c0367ea8 00000000 00000010 eea3ad00 eebb3638 bda0: ee08be88 00000000 c00c8a5c eea3ad08 ee08bde4 ee08bdc0 c00c303c c00c8a68 bdc0: ee08bed0 eea3ad00 ee08be88 00000000 ee3f0000 ee08be84 ee08bdf4 ee08bde8 bde0: c00c3ff8 c00c2ea0 ee08be6c ee08bdf8 c00d0c78 c00c3fb8 00000101 ee4264c8 be00: ee03e890 ee425770 eebb26b8 00000026 00000000 ee425770 ee08bee8 00000000 be20: 00000802 ee08bf5c 00000000 00000000 ee08a000 eebb3638 ee03e890 ee425770 be40: c00cef24 ee08bed0 eea3ad00 ee08be90 ee08bf5c ee3f0000 ee08be88 00000000 be60: ee08bec4 ee08be70 c00d10d4 c00d0400 ee08be84 ee9a1380 ee08bf0c ee08be88 be80: c036cddc 00000000 ee03e890 ee4274c8 ee08bec4 ee08bea0 c01aa17c ee08bf5c bea0: 00000001 ee3f0000 ffffff9c c000eb24 ee08a000 00000000 ee08bf4c ee08bec8 bec0: c00d1474 c00d0f00 00000041 ee08bed8 ee03e890 ee4274c8 474edc20 00000008 bee0: ee3f0019 eea3a700 ee03e7d0 ee4264c8 eebb3638 00000101 00000000 00000034 bf00: 00000000 00000000 00000000 c00de444 00080802 00080802 be8b4ea4 ee3f0000 bf20: ffffff9c c000eb24 ee08a000 00000000 00080802 00000004 ee3f0000 ffffff9c bf40: ee08bf94 ee08bf50 c00c4430 c00d144c c00d38e0 c00def38 eea3a700 00000802 bf60: be8b0000 00000026 00000100 00000001 be8b466c be8b4ea4 00000002 00000005 bf80: c000eb24 ee08a000 ee08bfa4 ee08bf98 c00c44f8 c00c431c 00000000 ee08bfa8 bfa0: c000e980 c00c44dc be8b466c be8b4ea4 be8b4794 00080802 be8b4628 00080802 bfc0: be8b466c be8b4ea4 00000002 00000005 001255f8 00129264 000dd264 00124478 bfe0: 00000000 be8b4634 00049257 00083036 600f0030 be8b4794 03433c61 0717419f Backtrace: [<c0386730>] (soc_pcm_open) from [<c0376a60>] (snd_pcm_open_substream+0x54/0x9c) r10:00000001 r9:00000000 r8:ee328d00 r7:ee328ce8 r6:eea3ad00 r5:ee08bd10 r4:00000000 [<c0376a0c>] (snd_pcm_open_substream) from [<c0376b38>] (snd_pcm_open+0x90/0x1b4) r5:ee328c00 r4:00000000 [<c0376aa8>] (snd_pcm_open) from [<c0376d04>] (snd_pcm_playback_open+0x44/0x64) r10:eea3ad00 r9:eea3ad00 r8:c0661848 r7:00000000 r6:eebb3638 r5:eea3ad00 r4:ee328c00 [<c0376cc0>] (snd_pcm_playback_open) from [<c0367f18>] (snd_open+0x7c/0x90) r5:eea3ad00 r4:c04f179c [<c0367e9c>] (snd_open) from [<c00c8b94>] (chrdev_open+0x138/0x164) r6:ee13bec0 r5:eebb3638 r4:00000000 r3:c0367e9c [<c00c8a5c>] (chrdev_open) from [<c00c303c>] (do_dentry_open.isra.17+0x1a8/0x2b8) r9:eea3ad08 r8:c00c8a5c r7:00000000 r6:ee08be88 r5:eebb3638 r4:eea3ad00 [<c00c2e94>] (do_dentry_open.isra.17) from [<c00c3ff8>] (vfs_open+0x4c/0x50) r9:ee08be84 r8:ee3f0000 r7:00000000 r6:ee08be88 r5:eea3ad00 r4:ee08bed0 [<c00c3fac>] (vfs_open) from [<c00d0c78>] (do_last.isra.47+0x884/0xb00) [<c00d03f4>] (do_last.isra.47) from [<c00d10d4>] (path_openat+0x1e0/0x54c) r10:00000000 r9:ee08be88 r8:ee3f0000 r7:ee08bf5c r6:ee08be90 r5:eea3ad00 r4:ee08bed0 [<c00d0ef4>] (path_openat) from [<c00d1474>] (do_filp_open+0x34/0x80) r10:00000000 r9:ee08a000 r8:c000eb24 r7:ffffff9c r6:ee3f0000 r5:00000001 r4:ee08bf5c [<c00d1440>] (do_filp_open) from [<c00c4430>] (do_sys_open+0x120/0x1c0) r7:ffffff9c r6:ee3f0000 r5:00000004 r4:00080802 [<c00c4310>] (do_sys_open) from [<c00c44f8>] (SyS_open+0x28/0x2c) r9:ee08a000 r8:c000eb24 r7:00000005 r6:00000002 r5:be8b4ea4 r4:be8b466c [<c00c44d0>] (SyS_open) from [<c000e980>] (ret_fast_syscall+0x0/0x34) Code: 00000000 00000000 00000000 00000000 (0000004f) ---[ end trace bd44069e52a540ae ]--- Segmentation fault Best regards --- Kuninori Morimoto

2 28

[alsa-devel] [GIT PULL] ASoC updates for v3.20
by Mark Brown 09 Feb '15

09 Feb '15

2 1

[alsa-devel] [PATCH] ASoC: rt5670: Set use_single_rw flag for regmap
by Bard Liao 09 Feb '15

09 Feb '15

RT5670 doesn't support auto incrementing writes so driver should set the use_single_rw flag for regmap. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0632b74..6aa4ab2 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2614,6 +2614,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5670 = { static const struct regmap_config rt5670_regmap = { .reg_bits = 8, .val_bits = 16, + .use_single_rw = true, .max_register = RT5670_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5670_ranges) * RT5670_PR_SPACING), .volatile_reg = rt5670_volatile_register, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v2] ASoC: rt286: Add rt288 codec support
by Bard Liao 09 Feb '15

09 Feb '15

This patch adds support for rt288 codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 81bdd276..7a72eac 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -34,6 +34,7 @@ #include "rt286.h" #define RT286_VENDOR_ID 0x10ec0286 +#define RT288_VENDOR_ID 0x10ec0288 struct rt286_priv { struct regmap *regmap; @@ -1168,6 +1169,7 @@ static const struct regmap_config rt286_regmap = { static const struct i2c_device_id rt286_i2c_id[] = { {"rt286", 0}, + {"rt288", 0}, {} }; MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); @@ -1210,7 +1212,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_read(rt286->regmap, RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret); - if (ret != RT286_VENDOR_ID) { + if (ret != RT286_VENDOR_ID && ret != RT288_VENDOR_ID) { dev_err(&i2c->dev, "Device with ID register %x is not rt286\n", ret); return -ENODEV; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: max98357a: Fix build in !CONFIG_OF case
by Mark Brown 09 Feb '15

09 Feb '15

Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 98b9153..1806333 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -120,6 +120,7 @@ static const struct of_device_id max98357a_device_id[] = { { .compatible = "maxim," DRV_NAME, }, {} }; +MODULE_DEVICE_TABLE(of, max98357a_device_id); #endif static struct platform_driver max98357a_platform_driver = { @@ -135,4 +136,3 @@ module_platform_driver(max98357a_platform_driver); MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:" DRV_NAME); -MODULE_DEVICE_TABLE(of, max98357a_device_id); -- 2.1.4

1 0

[alsa-devel] [RESEND PATCH V3 13/15] ARM: dts: Exynos4 and Odroid X2/U3 sound device nodes update
by Sylwester Nawrocki 09 Feb '15

09 Feb '15

Clock related properties are added to the Exynos4 I2S device nodes so they can be referred to as clock providers. Missing i2s_opclk1 clock is added to the I2S0 node and clock properties are added to the MAX98090 codec node to allow it to control/read frequency of the MCLK clock directly. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- arch/arm/boot/dts/exynos4.dtsi | 6 ++++++ arch/arm/boot/dts/exynos4412-odroid-common.dtsi | 8 ++++++-- 2 files changed, 12 insertions(+), 2 deletions(-) diff --git a/arch/arm/boot/dts/exynos4.dtsi b/arch/arm/boot/dts/exynos4.dtsi index b8168f1..38d8f68 100644 --- a/arch/arm/boot/dts/exynos4.dtsi +++ b/arch/arm/boot/dts/exynos4.dtsi @@ -61,6 +61,8 @@ reg = <0x03830000 0x100>; clocks = <&clock_audss EXYNOS_I2S_BUS>; clock-names = "iis"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk0"; dmas = <&pdma0 12>, <&pdma0 11>, <&pdma0 10>; dma-names = "tx", "rx", "tx-sec"; samsung,idma-addr = <0x03000000>; @@ -372,6 +374,8 @@ reg = <0x13960000 0x100>; clocks = <&clock CLK_I2S1>; clock-names = "iis"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk1"; dmas = <&pdma1 12>, <&pdma1 11>; dma-names = "tx", "rx"; status = "disabled"; @@ -382,6 +386,8 @@ reg = <0x13970000 0x100>; clocks = <&clock CLK_I2S2>; clock-names = "iis"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk2"; dmas = <&pdma0 14>, <&pdma0 13>; dma-names = "tx", "rx"; status = "disabled"; diff --git a/arch/arm/boot/dts/exynos4412-odroid-common.dtsi b/arch/arm/boot/dts/exynos4412-odroid-common.dtsi index 3fbf588..c26b9fb 100644 --- a/arch/arm/boot/dts/exynos4412-odroid-common.dtsi +++ b/arch/arm/boot/dts/exynos4412-odroid-common.dtsi @@ -7,6 +7,7 @@ * published by the Free Software Foundation. */ +#include <dt-bindings/sound/samsung-i2s.h> #include <dt-bindings/input/input.h> #include "exynos4412.dtsi" @@ -37,8 +38,9 @@ pinctrl-names = "default"; status = "okay"; clocks = <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_DOUT_AUD_BUS>; - clock-names = "iis", "i2s_opclk0"; + <&clock_audss EXYNOS_DOUT_AUD_BUS>, + <&clock_audss EXYNOS_SCLK_I2S>; + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; }; sound: sound { @@ -373,6 +375,8 @@ reg = <0x10>; interrupt-parent = <&gpx0>; interrupts = <0 0>; + clocks = <&i2s0 CLK_I2S_CDCLK>; + clock-names = "mclk"; }; }; -- 1.7.9.5

2 7

[alsa-devel] [PATCH 0/3] ASoC: atmel_ssc_dai: fix and refactor
by Bo Shen 09 Feb '15

09 Feb '15

This patch set do following things: - Fix the DSP mode A setting. - Refactor the startup and shutdown function. - Only clean PDC when use PDC. Bo Shen (3): ASoC: atmel_ssc_dai: fix the setting for DSP mode ASoC: atmel_ssc_dai: refactor the startup and shutdown ASoC: atmel_ssc_dai: only clean PDC when using PDC sound/soc/atmel/atmel_ssc_dai.c | 54 +++++++++++++++++++---------------------- 1 file changed, 25 insertions(+), 29 deletions(-) -- 2.3.0.rc0

2 5

[alsa-devel] [PATCH] ALSA: control: obsolete 'already present' messaging because it's caller's fault
by Takashi Sakamoto 08 Feb '15

08 Feb '15

When drivers or userspace applications add a duplicated control, snd module outputs error message and return -EBUSY. This state is a bug in driver or application, therefore the messaging should be done by them, not by snd module. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index d482997..8906b66 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -357,18 +357,10 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) down_write(&card->controls_rwsem); if (snd_ctl_find_id(card, &id)) { - up_write(&card->controls_rwsem); - dev_err(card->dev, "control %i:%i:%i:%s:%i is already present\n", - id.iface, - id.device, - id.subdevice, - id.name, - id.index); err = -EBUSY; goto error; } if (snd_ctl_find_hole(card, kcontrol->count) < 0) { - up_write(&card->controls_rwsem); err = -ENOMEM; goto error; } @@ -388,6 +380,7 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) return 0; error: + up_write(&card->controls_rwsem); snd_ctl_free_one(kcontrol); return err; } -- 2.1.0

2 2

[alsa-devel] [PATCH v2] ALSA: control: fix failure to return numerical ID in 'add' event
by Takashi Sakamoto 08 Feb '15

08 Feb '15

Currently when adding a new control, the assigned numerical ID is not set for event data, thus userspace applications cannot realize it just by event data. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/control.c b/sound/core/control.c index 60caba1..8aae6ea 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -373,6 +373,7 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) card->controls_count += kcontrol->count; kcontrol->id.numid = card->last_numid + 1; card->last_numid += kcontrol->count; + id = kcontrol->id; count = kcontrol->count; up_write(&card->controls_rwsem); for (idx = 0; idx < count; idx++, id.index++, id.numid++) -- 2.1.0

2 1

[alsa-devel] [PATCH] ALSA: control: fix failure to return numerical ID in event data
by Takashi Sakamoto 08 Feb '15

08 Feb '15

Currently an instance of control has zero as its member for numerical ID even if any IDs are assigned to. According to this bug, any userspace applications cannot identify controls by the ID when handling any events. On the other hand, the other members such as name are still valid, therefore applications can identify controls without fixing this. This is not preferable because the ID has an advantage to allow userspace applications to distinguish each controls by itself, without any combinations. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 60caba1..d482997 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -352,6 +352,9 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) if (id.index > UINT_MAX - kcontrol->count) goto error; + /* Any ID numbers are not assigned to this control yet. */ + id.numid = 0; + down_write(&card->controls_rwsem); if (snd_ctl_find_id(card, &id)) { up_write(&card->controls_rwsem); @@ -369,12 +372,17 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) err = -ENOMEM; goto error; } - list_add_tail(&kcontrol->list, &card->controls); - card->controls_count += kcontrol->count; + + /* Now this control is allowed to have ID number. */ kcontrol->id.numid = card->last_numid + 1; card->last_numid += kcontrol->count; - count = kcontrol->count; + card->controls_count += kcontrol->count; + list_add_tail(&kcontrol->list, &card->controls); + up_write(&card->controls_rwsem); + + id = kcontrol->id; + count = kcontrol->count; for (idx = 0; idx < count; idx++, id.index++, id.numid++) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_ADD, &id); return 0; -- 2.1.0

2 2

[alsa-devel] Some questions about userspace control elements
by Takashi Sakamoto 08 Feb '15

08 Feb '15

Dear all, I have some questions about ALSA control interface. Currently I'm working for writing control applications in userspace. The application adds some control elements to a certain control device, then execute interprocess communication via the device by ALSA control interface. To help this work, I'm preparing for alsa-gir, with partly implementation of ALSA Control/Timer/Sequencer interface. (Currently I have no plan to support the other interfaces, such as PCM.) https://github.com/takaswie/alsa-gir Then I have six questions. 1.The 'numid' is unique number for each element or not. 2.The 'numid' is always zero in SND_CTL_EVENT_MASK_ADD event. 3.The way to change permissions of each control element. 4.Added elements are not removed when closing fd. 5.Processes can delete elements which the other processes added. 6.The operation to replace element generates separate events. In detail: 1.The 'numid' is unique number for each element or not? I want to use it to indicate each element, if possible. But there're more members in snd_ctl_elem_id structure. 2.The 'numid' can be notified in SND_CTL_EVENT_MASK_ADD event? As long as I test, it's always zero when userspace applications add any elements. Is it an expected behaviour? Or a bug? 3.The way to change permissions of each control element. Is it possible after adding the elements? If possible, how to notify it in the other processes? 4.Any elements added by a process are not removed at closing snd_ctl_t. They're remained after closing the process' snd_ctl_t. Is it an expected behaviour? Or a bug? 5.Processes can delete elements which the other processes added. Is it an expected behaviour? Or a bug? 6.The operation to replace elements generates separate events. ADD/REMOVE events occurs sequencially. Userspace application cannot distinguish replacement from adding/removing, therefore cannot process appropriate action. Is it an expected behaviour? Or a bug? Regards Takashi Sakamoto

2 8

[alsa-devel] [PATCH v2 0/3] More Line 6 cleanup
by Chris Rorvick 08 Feb '15

08 Feb '15

Changes in v2: * return true/false instead of 1/0 * do not include spaces in driver name * drop patch altering struct names (unnecessary) Chris Rorvick (3): ALSA: line6: Add toneport_has_source_select() ALSA: line6: Pass toneport pointer to toneport_has_led() ALSA: line6: Pass driver name to line6_probe() sound/usb/line6/driver.c | 3 ++- sound/usb/line6/driver.h | 3 +-- sound/usb/line6/pod.c | 2 +- sound/usb/line6/podhd.c | 2 +- sound/usb/line6/toneport.c | 53 +++++++++++++++++++++++++--------------------- sound/usb/line6/variax.c | 2 +- 6 files changed, 35 insertions(+), 30 deletions(-) -- 2.1.0

2 4

[alsa-devel] [PATCH] ASoC: Intel: sst: Fix firmware name size handling
by Christian Engelmayer 08 Feb '15

08 Feb '15

Function sst_acpi_probe() uses plain strcpy for setting member firmware_name of a struct intel_sst_drv from member firmware of a struct sst_machines. Thereby the destination array has got a length of 20 byte while the source may hold 32 byte. Since eg. commit 64b9c90b8600 ("ASoC: Intel: Fix BYTCR firmware name") increased strings from "fw_sst_0f28.bin" to "intel/fw_sst_0f28.bin" there is an actual possibility that the 20 byte array at the end of struct intel_sst_drv is overrun. Thus increase the size of the destination and use the same define for both structs. Detected by Coverity CID 1260087. Signed-off-by: Christian Engelmayer <cengelma(a)gmx.at> --- Compile tested only. Applies against linux-next. --- sound/soc/intel/sst/sst.h | 3 ++- sound/soc/intel/sst/sst_acpi.c | 2 +- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index 7f4bbfc..562bc48 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -58,6 +58,7 @@ enum sst_algo_ops { #define SST_BLOCK_TIMEOUT 1000 #define FW_SIGNATURE_SIZE 4 +#define FW_NAME_SIZE 32 /* stream states */ enum sst_stream_states { @@ -426,7 +427,7 @@ struct intel_sst_drv { * Holder for firmware name. Due to async call it needs to be * persistent till worker thread gets called */ - char firmware_name[20]; + char firmware_name[FW_NAME_SIZE]; }; /* misc definitions */ diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 43bc1c4..0542442 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -47,7 +47,7 @@ struct sst_machines { char board[32]; char machine[32]; void (*machine_quirk)(void); - char firmware[32]; + char firmware[FW_NAME_SIZE]; struct sst_platform_info *pdata; }; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: sirf: atlas7: fix platform_no_drv_owner.cocci warnings
by kbuild test robot 08 Feb '15

08 Feb '15

sound/soc/sirf/atlas7-iacc.c:625:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci CC: Rongjun Ying <rongjun.ying(a)csr.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- atlas7-iacc.c | 1 - 1 file changed, 1 deletion(-) --- a/sound/soc/sirf/atlas7-iacc.c +++ b/sound/soc/sirf/atlas7-iacc.c @@ -622,7 +622,6 @@ static const struct dev_pm_ops atlas7_ia static struct platform_driver atlas7_iacc_driver = { .driver = { .name = "sirf-atlas7-iacc", - .owner = THIS_MODULE, .of_match_table = atlas7_iacc_of_match, .pm = &atlas7_iacc_pm_ops, },

1 0

[alsa-devel] [PATCH 0/4] More Line 6 cleanup
by Chris Rorvick 07 Feb '15

07 Feb '15

Not much to say about these other than what's in the commit log. I was looking into what appeared to be a regression that was only reproducible outside of QEMU, but it turned out to be me pulling in the test/snd-device changes but testing against vanilla v3.19-rc7 ALSA modules. Fixing my process. :-) Regards, Chris Chris Rorvick (4): ALSA: line6: Add toneport_has_source_select() ALSA: line6: Pass toneport pointer to toneport_has_led() ALSA: line6: Pass driver name to line6_probe() ALSA: line6: Remove `usb_' prefix from structs sound/usb/line6/capture.c | 2 +- sound/usb/line6/driver.c | 45 +++++++++++----------- sound/usb/line6/driver.h | 25 ++++++------ sound/usb/line6/midi.c | 24 +++++------- sound/usb/line6/midi.h | 6 +-- sound/usb/line6/pcm.c | 5 +-- sound/usb/line6/pcm.h | 4 +- sound/usb/line6/playback.c | 2 +- sound/usb/line6/pod.c | 52 ++++++++++++------------- sound/usb/line6/podhd.c | 7 ++-- sound/usb/line6/toneport.c | 94 ++++++++++++++++++++++++---------------------- sound/usb/line6/variax.c | 41 ++++++++++---------- 12 files changed, 151 insertions(+), 156 deletions(-) -- 2.1.0

2 11

[alsa-devel] [Intel HDA][ALC3226] Too much white noise from microphone
by Shawn Starr 07 Feb '15

07 Feb '15

Hello ALSA devs, I spoke to patrakov on Freenode from the #Alsa IRC channel and they mentioned that the noise background I am experiencing is not normal. I tried two different headsets, both headsets have two separate jacks, the same result. I will test any patches/quirks for kernel as needed. Here is the alsa-info: http://www.alsa-project.org/db/?f=29d3c45e2d66c58be03a10bd0186067a9d735a11 Here is a 10 second recording: http://turl.ca/zrhbcs Here is the amixer information: Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 87 [100%] [0.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 87 [100%] [0.00dB] [on] Front Right: Playback 87 [100%] [0.00dB] [on] Simple mixer control 'Speaker',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [off] Front Right: Playback [off] Simple mixer control 'Speaker+LO',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 65 [75%] [-16.50dB] Front Right: Playback 65 [75%] [-16.50dB] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Mono Limits: Playback 0 - 31 Mono: Capture [on] Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Front Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 2 [67%] [20.00dB] Front Right: 2 [67%] [20.00dB] Simple mixer control 'Line Out',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [off] Front Right: Playback [off] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 63 Front Left: Capture 63 [100%] [30.00dB] [on] Front Right: Capture 63 [100%] [30.00dB] [on] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Speaker Only' 'Line Out+Speaker' Item0: 'Line Out+Speaker' Simple mixer control 'Dock Mic',0 Capabilities: pvolume pswitch cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Mono Limits: Playback 0 - 31 Mono: Capture [off] Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Dock Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right

3 5

[alsa-devel] [PATCH v2 0/3] ASoC: add audio card creation from graph of ports in DT
by Jean-Francois Moine 07 Feb '15

07 Feb '15

This patchset adds a way to build audio cards from the description of a graph of ports in a devicetree. v2: - move the soc-core and dt-card from the patchset "ASoC: tda998x: add a codec to the HDMI transmitter" - add DT documentation (Mark Brown) Jean-Francois Moine (3): ASoC: core: export snd_soc_get_dai_name Documentation: of: Document audio graph bindings ASoC: add generic dt-card support .../devicetree/bindings/sound/audio-interfaces.txt | 101 ++++++++ include/sound/soc.h | 2 + sound/soc/generic/Kconfig | 2 + sound/soc/generic/Makefile | 2 + sound/soc/generic/dt-card.c | 275 +++++++++++++++++++++ sound/soc/soc-core.c | 5 +- 6 files changed, 385 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-interfaces.txt create mode 100644 sound/soc/generic/dt-card.c -- 2.1.4

3 20

[alsa-devel] [PATCH v3 0/15] Those patches is used for dw_hdmi audio support
by Yakir Yang 07 Feb '15

07 Feb '15

We found Designware hdmi driver only support audio clock config, we can not play sound through it. To add Designware HDMI Audio support, we make those patch set: 1): fixed dw_hdmi irq bug, add irq control to suspend/resume interfaces. 2): add suspend/resume callback for dw_hdmi rockchip driver. 3): Warp irq control in functions. 4): Combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() 5): Adjust n/cts config order. 6): Set ncts_atomic_write & cts_manual, according to dw_hdmi document. 7): Add audio support for more display resolutions(eg. 800x600). 8): Add audio support for No-CEA display resolutions. 9): Add Audio Sample Channel Status config interfaces to dw_hdmi driver. 10): add audio clock control interfaces o dw_hdmi driver. 11): creat "dw_hdmi-audio" platform device in dw_hdmi driver. 12): add codec driver for hdmi audio, callback dw_hdmi audio config functions. 13): add sound driver for hdmi audio, creat hdmi audio sound card. 14): add dt-bings file and add hdmi_audio node to corresponding dt file. Changes in v3: - Clear Hotplug interrupts before dw_hdmi_fb_register - Wrap irq control in fucntions - Setting the .pm member instead of suspend/resume - Add ID registers parse and record - Combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() - Only adjust the n/cts setting order - Set ncts_atomic_write & cts_manual - Determine whether sample channel should set by desig_id. - Delete hdmi_audio_config interface and modify audio clock control functions - Remove audio_config & get_connect_status callback functions and add write/read/mod register callback functions - Keep audio format config function in dw-hdmi-audio driver and remove audio_config & get_connect_status functions, move jack control to dw-hdmi-audio completely. - Delete the operation of jack in rockchip-hdmi-audio driver, get ready to switch to simple-audio-card driver. - modify cpu-of-node to i2s-controller Changes in v2: - Add irq control to suspend/resume interfaces - Add suspend/resume support for dw_hdmi_rockchip driver - add more n/cts combinations for more display resolutions - enable audio support for No-CEA display mode - Add audio sample channel status setting - Add audio config interfaces to dw_hdmi driver - Update the audio control interfaces - Update dw_hdmi audio control interfaces, and adjust jack report process - give "codec-name" & "codec-dai-name" an const name - remove codec-name and codec-dai-name - rename rockchip,rockchip-hdmi-audio.txt to rockchip,rockchip-dw-hdmi-audio.txt Yakir Yang (15): drm: bridge/dw_hdmi: add irq control to suspend/resume drm: bridge/dw_hdmi: wrap irq control in fucntions drm: rockchip/dw_hdmi_rockchip: add resume/suspend support drm: bridge/dw_hdmi: add indentification registers parse and record drm: bridge/dw_hdmi: combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() drm: bridge/dw_hdmi: adjust n/cts setting order drm: bridge/dw_hdmi: set ncts_atomic_write & cts_manual drm: bridge/dw_hdmi: add audio support for more display resolutions drm: bridge/dw_hdmi: enable audio support for No-CEA display resolutions drm: bridge/dw_hdmi: add audio sample channel status setting drm: bridge/dw_hdmi: add audio clock control interfaces drm: bridge/dw_hdmi: creat dw-hdmi-audio platform device ASoC: dw-hdmi-audio: add codec driver for dw hdmi audio ASoC: rockchip-hdmi-audio: add sound driver for hdmi audio dt-bindings: Add documentation for Rockchip dw-hdmi-audio .../sound/rockchip,rockchip-dw-hdmi-audio.txt | 12 + drivers/gpu/drm/bridge/dw_hdmi.c | 361 ++++++++++++++++++--- drivers/gpu/drm/bridge/dw_hdmi.h | 49 +++ drivers/gpu/drm/rockchip/dw_hdmi-rockchip.c | 16 + include/drm/bridge/dw_hdmi.h | 17 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/dw-hdmi-audio.c | 333 +++++++++++++++++++ sound/soc/codecs/dw-hdmi-audio.h | 79 +++++ sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_hdmi_audio.c | 168 ++++++++++ 12 files changed, 1013 insertions(+), 39 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt create mode 100644 sound/soc/codecs/dw-hdmi-audio.c create mode 100644 sound/soc/codecs/dw-hdmi-audio.h create mode 100644 sound/soc/rockchip/rockchip_hdmi_audio.c -- 2.1.2

3 19

[alsa-devel] [PATCH 1/4] ASoC: samsung: Remove goni or aquila with the WM8994
by Paul Bolle 07 Feb '15

07 Feb '15

Commit 28c8331d386a ("ARM: S5PV210: Remove support for board files") removed the Kconfig symbols MACH_GONI and MACH_AQUILA. As a result the dependencies of SND_SOC_GONI_AQUILA_WM8994 can never be met. So remove the unbuildable "SoC I2S Audio support for AQUILA/GONI - WM8994". Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Suggested by Arnd Bergmann (see https://lkml.org/lkml/2014/9/4/381 ). Previously submitted in https://lkml.org/lkml/2014/9/18/128 , but in an awkward way. So this is basically a resend. sound/soc/samsung/Kconfig | 11 -- sound/soc/samsung/Makefile | 2 - sound/soc/samsung/goni_wm8994.c | 289 ---------------------------------------- 3 files changed, 302 deletions(-) delete mode 100644 sound/soc/samsung/goni_wm8994.c diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index fc67f97f19f6..da6d0229cfa2 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -146,17 +146,6 @@ config SND_SOC_SMARTQ select SND_SAMSUNG_I2S select SND_SOC_WM8750 -config SND_SOC_GONI_AQUILA_WM8994 - tristate "SoC I2S Audio support for AQUILA/GONI - WM8994" - depends on SND_SOC_SAMSUNG && (MACH_GONI || MACH_AQUILA) - depends on I2C=y - select SND_SAMSUNG_I2S - select MFD_WM8994 - select SND_SOC_WM8994 - help - Say Y if you want to add support for SoC audio on goni or aquila - with the WM8994. - config SND_SOC_SAMSUNG_SMDK_SPDIF tristate "SoC S/PDIF Audio support for SMDK" depends on SND_SOC_SAMSUNG diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 31e3dba7e3b5..052fe71be518 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -35,7 +35,6 @@ snd-soc-smdk-wm8994-objs := smdk_wm8994.o snd-soc-snow-objs := snow.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o -snd-soc-goni-wm8994-objs := goni_wm8994.o snd-soc-smdk-spdif-objs := smdk_spdif.o snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o @@ -63,7 +62,6 @@ obj-$(CONFIG_SND_SOC_SNOW) += snd-soc-snow.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o -obj-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c deleted file mode 100644 index fad56b9e7369..000000000000 --- a/sound/soc/samsung/goni_wm8994.c +++ /dev/null @@ -1,289 +0,0 @@ -/* - * goni_wm8994.c - * - * Copyright (C) 2010 Samsung Electronics Co.Ltd - * Author: Chanwoo Choi <cw00.choi(a)samsung.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 the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include <linux/module.h> -#include <sound/soc.h> -#include <sound/jack.h> - -#include <asm/mach-types.h> -#include <mach/gpio-samsung.h> - -#include "../codecs/wm8994.h" - -#define MACHINE_NAME 0 -#define CPU_VOICE_DAI 1 - -static const char *aquila_str[] = { - [MACHINE_NAME] = "aquila", - [CPU_VOICE_DAI] = "aquila-voice-dai", -}; - -static struct snd_soc_card goni; -static struct platform_device *goni_snd_device; - -/* 3.5 pie jack */ -static struct snd_soc_jack jack; - -/* 3.5 pie jack detection DAPM pins */ -static struct snd_soc_jack_pin jack_pins[] = { - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, { - .pin = "Headset Stereophone", - .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | - SND_JACK_AVOUT, - }, -}; - -/* 3.5 pie jack detection gpios */ -static struct snd_soc_jack_gpio jack_gpios[] = { - { - .gpio = S5PV210_GPH0(6), - .name = "DET_3.5", - .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | - SND_JACK_AVOUT, - .debounce_time = 200, - }, -}; - -static const struct snd_soc_dapm_widget goni_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Ext Left Spk", NULL), - SND_SOC_DAPM_SPK("Ext Right Spk", NULL), - SND_SOC_DAPM_SPK("Ext Rcv", NULL), - SND_SOC_DAPM_HP("Headset Stereophone", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Main Mic", NULL), - SND_SOC_DAPM_MIC("2nd Mic", NULL), - SND_SOC_DAPM_LINE("Radio In", NULL), -}; - -static const struct snd_soc_dapm_route goni_dapm_routes[] = { - {"Ext Left Spk", NULL, "SPKOUTLP"}, - {"Ext Left Spk", NULL, "SPKOUTLN"}, - - {"Ext Right Spk", NULL, "SPKOUTRP"}, - {"Ext Right Spk", NULL, "SPKOUTRN"}, - - {"Ext Rcv", NULL, "HPOUT2N"}, - {"Ext Rcv", NULL, "HPOUT2P"}, - - {"Headset Stereophone", NULL, "HPOUT1L"}, - {"Headset Stereophone", NULL, "HPOUT1R"}, - - {"IN1RN", NULL, "Headset Mic"}, - {"IN1RP", NULL, "Headset Mic"}, - - {"IN1RN", NULL, "2nd Mic"}, - {"IN1RP", NULL, "2nd Mic"}, - - {"IN1LN", NULL, "Main Mic"}, - {"IN1LP", NULL, "Main Mic"}, - - {"IN2LN", NULL, "Radio In"}, - {"IN2RN", NULL, "Radio In"}, -}; - -static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - /* set endpoints to not connected */ - snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN"); - snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); - 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"); - - if (machine_is_aquila()) { - snd_soc_dapm_nc_pin(dapm, "SPKOUTRN"); - snd_soc_dapm_nc_pin(dapm, "SPKOUTRP"); - } - - /* Headset jack detection */ - ret = snd_soc_jack_new(codec, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, - &jack); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); - if (ret) - return ret; - - ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); - if (ret) - return ret; - - return 0; -} - -static int goni_hifi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - unsigned int pll_out = 24000000; - int ret = 0; - - /* set the codec FLL */ - ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, - params_rate(params) * 256); - if (ret < 0) - return ret; - - /* set the codec system clock */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, - params_rate(params) * 256, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops goni_hifi_ops = { - .hw_params = goni_hifi_hw_params, -}; - -static int goni_voice_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - unsigned int pll_out = 24000000; - int ret = 0; - - if (params_rate(params) != 8000) - return -EINVAL; - - /* set the codec FLL */ - ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, - params_rate(params) * 256); - if (ret < 0) - return ret; - - /* set the codec system clock */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, - params_rate(params) * 256, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_dai_driver voice_dai = { - .name = "goni-voice-dai", - .id = 0, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, -}; - -static const struct snd_soc_component_driver voice_component = { - .name = "goni-voice", -}; - -static struct snd_soc_ops goni_voice_ops = { - .hw_params = goni_voice_hw_params, -}; - -static struct snd_soc_dai_link goni_dai[] = { -{ - .name = "WM8994", - .stream_name = "WM8994 HiFi", - .cpu_dai_name = "samsung-i2s.0", - .codec_dai_name = "wm8994-aif1", - .platform_name = "samsung-i2s.0", - .codec_name = "wm8994-codec.0-001a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .init = goni_wm8994_init, - .ops = &goni_hifi_ops, -}, { - .name = "WM8994 Voice", - .stream_name = "Voice", - .cpu_dai_name = "goni-voice-dai", - .codec_dai_name = "wm8994-aif2", - .codec_name = "wm8994-codec.0-001a", - .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_IB_IF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &goni_voice_ops, -}, -}; - -static struct snd_soc_card goni = { - .name = "goni", - .owner = THIS_MODULE, - .dai_link = goni_dai, - .num_links = ARRAY_SIZE(goni_dai), - - .dapm_widgets = goni_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(goni_dapm_widgets), - .dapm_routes = goni_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(goni_dapm_routes), -}; - -static int __init goni_init(void) -{ - int ret; - - if (machine_is_aquila()) { - voice_dai.name = aquila_str[CPU_VOICE_DAI]; - goni_dai[1].cpu_dai_name = aquila_str[CPU_VOICE_DAI]; - goni.name = aquila_str[MACHINE_NAME]; - } else if (!machine_is_goni()) - return -ENODEV; - - goni_snd_device = platform_device_alloc("soc-audio", -1); - if (!goni_snd_device) - return -ENOMEM; - - /* register voice DAI here */ - ret = devm_snd_soc_register_component(&goni_snd_device->dev, - &voice_component, &voice_dai, 1); - if (ret) { - platform_device_put(goni_snd_device); - return ret; - } - - platform_set_drvdata(goni_snd_device, &goni); - ret = platform_device_add(goni_snd_device); - - if (ret) - platform_device_put(goni_snd_device); - - return ret; -} - -static void __exit goni_exit(void) -{ - platform_device_unregister(goni_snd_device); -} - -module_init(goni_init); -module_exit(goni_exit); - -/* Module information */ -MODULE_DESCRIPTION("ALSA SoC WM8994 GONI(S5PV210)"); -MODULE_AUTHOR("Chanwoo Choi <cw00.choi(a)samsung.com>"); -MODULE_LICENSE("GPL"); -- 1.9.3

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[alsa-devel] [PATCH 4/4] ASoC: samsung: remove outdated Kconfig references
by Paul Bolle 07 Feb '15

07 Feb '15

The Kconfig symbols MACH_SMDKC110 and MACH_SMDKV210 were removed in v3.17. But the Kconfig entries for SND_SOC_SAMSUNG_SMDK_WM8580 and SND_SOC_SAMSUNG_SMDK_WM9713 still reference them. Perhaps these references to those removed symbols should be changed into references to an existing symbol (such as MACH_S5PV210_DT). But apparently no one noticed that these references were pointless ever since v3.17. So it seems they can simply be removed. Let's do that now. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- sound/soc/samsung/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 5f983d064920..01980bb2c6e7 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -53,7 +53,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750 config SND_SOC_SAMSUNG_SMDK_WM8580 tristate "SoC I2S Audio support for WM8580 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKV210 || MACH_SMDKC110) + depends on SND_SOC_SAMSUNG && MACH_SMDK6410 depends on REGMAP_I2C select SND_SOC_WM8580 select SND_SAMSUNG_I2S @@ -134,7 +134,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKV210 || MACH_SMDKC110) + depends on SND_SOC_SAMSUNG && MACH_SMDK6410 select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help -- 1.9.3

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[alsa-devel] [PATCH 3/4] ASoC: samsung: MACH_SMDKC100 isn't supported anymore
by Paul Bolle 07 Feb '15

07 Feb '15

Commit b8529ec1c1b0 ("ARM: S5PC100: no more support S5PC100 SoC") removed Kconfig symbol MACH_SMDKC100. But there are still two references to that symbol left. These are now pointless. Remove them too. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Apparently Kukjin had a fixup queued (see http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081482.… ). But that fixup never entered the tree, so let's try again. sound/soc/samsung/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 3be649098b30..5f983d064920 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -53,7 +53,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750 config SND_SOC_SAMSUNG_SMDK_WM8580 tristate "SoC I2S Audio support for WM8580 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKV210 || MACH_SMDKC110) depends on REGMAP_I2C select SND_SOC_WM8580 select SND_SAMSUNG_I2S @@ -134,7 +134,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKV210 || MACH_SMDKC110) select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help -- 1.9.3

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[alsa-devel] [PATCH 2/4] ASoC: samsung: Remove PCM support for WM8580 on SMDK
by Paul Bolle 07 Feb '15

07 Feb '15

Commit 28c8331d386a ("ARM: S5PV210: Remove support for board files") removed the Kconfig symbols MACH_SMDKC110 and MACH_SMDKV210. As a result the dependencies of SND_SOC_SMDK_WM8580_PCM can never be met. So remove the unbuildable "SoC PCM Audio support for WM8580 on SMDK". Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Suggested by Arnd Bergmann (see https://lkml.org/lkml/2014/9/4/381 ). Previously submitted in https://lkml.org/lkml/2014/9/18/147 , but again in an awkward way. So this is a resend too. sound/soc/samsung/Kconfig | 9 -- sound/soc/samsung/Makefile | 2 - sound/soc/samsung/smdk_wm8580pcm.c | 176 ------------------------------------- 3 files changed, 187 deletions(-) delete mode 100644 sound/soc/samsung/smdk_wm8580pcm.c diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index da6d0229cfa2..3be649098b30 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -153,15 +153,6 @@ config SND_SOC_SAMSUNG_SMDK_SPDIF help Say Y if you want to add support for SoC S/PDIF audio on the SMDK. -config SND_SOC_SMDK_WM8580_PCM - tristate "SoC PCM Audio support for WM8580 on SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKV210 || MACH_SMDKC110) - depends on REGMAP_I2C - select SND_SOC_WM8580 - select SND_SAMSUNG_PCM - help - Say Y if you want to add support for SoC audio on the SMDK. - config SND_SOC_SMDK_WM8994_PCM tristate "SoC PCM Audio support for WM8994 on SMDK" depends on SND_SOC_SAMSUNG diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 052fe71be518..2b32c371da55 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -36,7 +36,6 @@ snd-soc-snow-objs := snow.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o snd-soc-smdk-spdif-objs := smdk_spdif.o -snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o snd-soc-speyside-objs := speyside.o snd-soc-tobermory-objs := tobermory.o @@ -62,7 +61,6 @@ obj-$(CONFIG_SND_SOC_SNOW) += snd-soc-snow.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o -obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o obj-$(CONFIG_SND_SOC_TOBERMORY) += snd-soc-tobermory.o diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c deleted file mode 100644 index 6deec5234c92..000000000000 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ /dev/null @@ -1,176 +0,0 @@ -/* - * sound/soc/samsung/smdk_wm8580pcm.c - * - * Copyright (c) 2011 Samsung Electronics Co. Ltd - * - * 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. - */ -#include <linux/module.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> -#include <sound/pcm.h> - -#include <asm/mach-types.h> - -#include "../codecs/wm8580.h" -#include "dma.h" -#include "pcm.h" - -/* - * Board Settings: - * o '1' means 'ON' - * o '0' means 'OFF' - * o 'X' means 'Don't care' - * - * SMDK6410 Base B/D: CFG1-0000, CFG2-1111 - * SMDKC110, SMDKV210: CFGB11-100100, CFGB12-0000 - */ - -#define SMDK_WM8580_EXT_OSC 12000000 -#define SMDK_WM8580_EXT_MCLK 4096000 -#define SMDK_WM8580_EXT_VOICE 2048000 - -static unsigned long mclk_freq; -static unsigned long xtal_freq; - -/* - * If MCLK clock directly gets from XTAL, we don't have to use PLL - * to make MCLK, but if XTAL clock source connects with other codec - * pin (like XTI), we should have to set codec's PLL to make MCLK. - * Because Samsung SoC does not support pcmcdclk output like I2S. - */ - -static int smdk_wm8580_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int rfs, ret; - - switch (params_rate(params)) { - case 8000: - break; - default: - printk(KERN_ERR "%s:%d Sampling Rate %u not supported!\n", - __func__, __LINE__, params_rate(params)); - return -EINVAL; - } - - rfs = mclk_freq / params_rate(params) / 2; - - if (mclk_freq == xtal_freq) { - ret = snd_soc_dai_set_sysclk(codec_dai, WM8580_CLKSRC_MCLK, - mclk_freq, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, - WM8580_CLKSRC_MCLK); - if (ret < 0) - return ret; - } else { - ret = snd_soc_dai_set_sysclk(codec_dai, WM8580_CLKSRC_PLLA, - mclk_freq, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, - WM8580_CLKSRC_PLLA); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, - xtal_freq, mclk_freq); - if (ret < 0) - return ret; - } - - /* Set PCM source clock on CPU */ - ret = snd_soc_dai_set_sysclk(cpu_dai, S3C_PCM_CLKSRC_MUX, - mclk_freq, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* Set SCLK_DIV for making bclk */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_PCM_SCLK_PER_FS, rfs); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops smdk_wm8580_pcm_ops = { - .hw_params = smdk_wm8580_pcm_hw_params, -}; - -#define SMDK_DAI_FMT (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | \ - SND_SOC_DAIFMT_CBS_CFS) - -static struct snd_soc_dai_link smdk_dai[] = { - { - .name = "WM8580 PAIF PCM RX", - .stream_name = "Playback", - .cpu_dai_name = "samsung-pcm.0", - .codec_dai_name = "wm8580-hifi-playback", - .platform_name = "samsung-audio", - .codec_name = "wm8580.0-001b", - .dai_fmt = SMDK_DAI_FMT, - .ops = &smdk_wm8580_pcm_ops, - }, { - .name = "WM8580 PAIF PCM TX", - .stream_name = "Capture", - .cpu_dai_name = "samsung-pcm.0", - .codec_dai_name = "wm8580-hifi-capture", - .platform_name = "samsung-pcm.0", - .codec_name = "wm8580.0-001b", - .dai_fmt = SMDK_DAI_FMT, - .ops = &smdk_wm8580_pcm_ops, - }, -}; - -static struct snd_soc_card smdk_pcm = { - .name = "SMDK-PCM", - .owner = THIS_MODULE, - .dai_link = smdk_dai, - .num_links = 2, -}; - -/* - * After SMDKC110 Base Board's Rev is '0.1', 12MHz External OSC(X1) - * is absent (or not connected), so we connect EXT_VOICE_CLK(OSC4), - * 2.0484Mhz, directly with MCLK both Codec and SoC. - */ -static int snd_smdk_probe(struct platform_device *pdev) -{ - int ret = 0; - - xtal_freq = SMDK_WM8580_EXT_OSC; - mclk_freq = SMDK_WM8580_EXT_MCLK; - - if (machine_is_smdkc110() || machine_is_smdkv210()) - xtal_freq = mclk_freq = SMDK_WM8580_EXT_VOICE; - - smdk_pcm.dev = &pdev->dev; - ret = devm_snd_soc_register_card(&pdev->dev, &smdk_pcm); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret); - - return ret; -} - -static struct platform_driver snd_smdk_driver = { - .driver = { - .name = "samsung-smdk-pcm", - }, - .probe = snd_smdk_probe, -}; - -module_platform_driver(snd_smdk_driver); - -MODULE_AUTHOR("Sangbeom Kim, <sbkim73(a)samsung.com>"); -MODULE_DESCRIPTION("ALSA SoC SMDK WM8580 for PCM"); -MODULE_LICENSE("GPL"); -- 1.9.3

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[alsa-devel] [PATCH v2] soc: intel: sst-haswell-ipc: Remove unused functions
by Rickard Strandqvist 07 Feb '15

07 Feb '15

Removes some functions that are not used anywhere: sst_hsw_stream_unmute() sst_hsw_stream_mute() msg_set_stage_type() sst_hsw_dx_get_state() sst_hsw_stream_set_write_position() sst_hsw_stream_get_vol_reg() sst_hsw_stream_get_peak_reg() sst_hsw_stream_get_pointer_reg() sst_hsw_stream_get_read_reg() sst_hsw_stream_get_mixer_id() sst_hsw_stream_get_hw_id() sst_hsw_mixer_set_volume_curve() sst_hsw_mixer_unmute() sst_hsw_mixer_mute() sst_hsw_stream_set_volume_curve() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/soc/intel/sst-haswell-ipc.c | 168 ------------------------------------- sound/soc/intel/sst-haswell-ipc.h | 31 ------- 2 files changed, 199 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 3f8c482..d693d4a 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -337,12 +337,6 @@ static inline u32 msg_get_stage_type(u32 msg) return (msg & IPC_STG_TYPE_MASK) >> IPC_STG_TYPE_SHIFT; } -static inline u32 msg_set_stage_type(u32 msg, u32 type) -{ - return (msg & ~IPC_STG_TYPE_MASK) + - (type << IPC_STG_TYPE_SHIFT); -} - static inline u32 msg_get_stream_id(u32 msg) { return (msg & IPC_STR_ID_MASK) >> IPC_STR_ID_SHIFT; @@ -969,45 +963,6 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw, } /* Mixer Controls */ -int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_stream_get_volume(hsw, stream, stage_id, channel, - &stream->mute_volume[channel]); - if (ret < 0) - return ret; - - ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, 0); - if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", - stream->reply.stream_hw_id, channel); - return ret; - } - - stream->mute[channel] = 1; - return 0; -} - -int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel) - -{ - int ret; - - stream->mute[channel] = 0; - ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, - stream->mute_volume[channel]); - if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", - stream->reply.stream_hw_id, channel); - return ret; - } - - return 0; -} - int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume) { @@ -1021,17 +976,6 @@ int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream return 0; } -int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u64 curve_duration, - enum sst_hsw_volume_curve curve) -{ - /* curve duration in steps of 100ns */ - stream->vol_req.curve_duration = curve_duration; - stream->vol_req.curve_type = curve; - - return 0; -} - /* stream volume */ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume) @@ -1083,42 +1027,6 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, return 0; } -int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_mixer_get_volume(hsw, stage_id, channel, - &hsw->mute_volume[channel]); - if (ret < 0) - return ret; - - ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, 0); - if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", - channel); - return ret; - } - - hsw->mute[channel] = 1; - return 0; -} - -int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, - hsw->mixer_info.volume_register_address[channel]); - if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", - channel); - return ret; - } - - hsw->mute[channel] = 0; - return 0; -} - int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 *volume) { @@ -1132,16 +1040,6 @@ int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, return 0; } -int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, - u64 curve_duration, enum sst_hsw_volume_curve curve) -{ - /* curve duration in steps of 100ns */ - hsw->curve_duration = curve_duration; - hsw->curve_type = curve; - - return 0; -} - /* global mixer volume */ int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 volume) @@ -1434,48 +1332,6 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) /* Stream Information - these calls could be inline but we want the IPC ABI to be opaque to client PCM drivers to cope with any future ABI changes */ -int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.stream_hw_id; -} - -int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.mixer_hw_id; -} - -u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.read_position_register_address; -} - -u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.presentation_position_register_address; -} - -u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel) -{ - if (channel >= 2) - return 0; - - return stream->reply.peak_meter_register_address[channel]; -} - -u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel) -{ - if (channel >= 2) - return 0; - - return stream->reply.volume_register_address[channel]; -} - int sst_hsw_mixer_get_info(struct sst_hsw *hsw) { struct sst_hsw_ipc_stream_info_reply *reply; @@ -1598,30 +1454,6 @@ u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw, return ppos; } -int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 stage_id, u32 position) -{ - u32 header; - int ret; - - trace_stream_write_position(stream->reply.stream_hw_id, position); - - header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | - IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE); - header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT); - header |= (IPC_STG_SET_WRITE_POSITION << IPC_STG_TYPE_SHIFT); - header |= (stage_id << IPC_STG_ID_SHIFT); - stream->wpos.position = position; - - ret = ipc_tx_message_nowait(hsw, header, &stream->wpos, - sizeof(stream->wpos)); - if (ret < 0) - dev_err(hsw->dev, "error: stream %d set position %d failed\n", - stream->reply.stream_hw_id, position); - - return ret; -} - /* physical BE config */ int sst_hsw_device_set_config(struct sst_hsw *hsw, enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk, diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 138e894..c1ad901 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -376,32 +376,17 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw, u32 create_channel_map(enum sst_hsw_channel_config config); /* Stream Mixer Controls - */ -int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel); -int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel); - int sst_hsw_stream_set_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume); int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume); -int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u64 curve_duration, - enum sst_hsw_volume_curve curve); - /* Global Mixer Controls - */ -int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel); -int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel); - int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 volume); int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 *volume); -int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, - u64 curve_duration, enum sst_hsw_volume_curve curve); - /* Stream API */ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, u32 (*get_write_position)(struct sst_hsw_stream *stream, void *data), @@ -440,18 +425,6 @@ int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); -int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel); -u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel); int sst_hsw_mixer_get_info(struct sst_hsw *hsw); /* Stream ALSA trigger operations */ @@ -466,8 +439,6 @@ int sst_hsw_stream_get_read_pos(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 *position); int sst_hsw_stream_get_write_pos(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 *position); -int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 stage_id, u32 position); u32 sst_hsw_get_dsp_position(struct sst_hsw *hsw, struct sst_hsw_stream *stream); u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw, @@ -481,8 +452,6 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw, /* DX Config */ int sst_hsw_dx_set_state(struct sst_hsw *hsw, enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx); -int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item, - u32 *offset, u32 *size, u32 *source); /* init */ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata); -- 1.7.10.4

2 1

[alsa-devel] [PATCH v3 alsa-lib 1/3] snd_pcm_start_at and friends.
by Tim Cussins 06 Feb '15

06 Feb '15

Upcoming audio hardware has the ability to begin playback when the value of a high-precision free-running counter matches some programmable value. This patch exposes such capability in a general way, allowing for several clock sources, including both system clocks and audio hardware clocks. We define 3 new methods that operate on a snd_pcm_t, and 1 that operates on a snd_pcm_status_t. - snd_pcm_start_at() allows client code to delegate the starting of the stream to the kernel. - snd_pcm_start_at_abort() allows client code to cancel a previous call to snd_pcm_start_at(). - snd_pcm_start_at_gettime() allows client code to query the current time for a given clock. - snd_pcm_status_get_startat_state() allows client code to extract information about any pending start_at timer from a snd_pcm_status_t object. We define a new enum: snd_pcm_startat_clock_type_t, which reinforces a clean boundary between the capabilities of the timestamping system and the start_at system. The clock and audio hw sources are grouped together because the difference is irrelevant to the use of snd_pcm_start_at(). Signed-off-by: Tim Cussins <timcussins(a)eml.cc> diff --git a/include/pcm.h b/include/pcm.h index 0655e7f..4f4834b 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -330,6 +330,17 @@ typedef enum _snd_pcm_tstamp_type { SND_PCM_TSTAMP_TYPE_LAST = SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, } snd_pcm_tstamp_type_t; +/** PCM start_at clock types */ +typedef enum _snd_pcm_startat_clock_type { + /** POSIX CLOCK_REALTIME equivalent */ + SND_PCM_STARTAT_CLOCK_TYPE_GETTIMEOFDAY = 0, + /** POSIX CLOCK_MONOTONIC equivalent */ + SND_PCM_STARTAT_CLOCK_TYPE_MONOTONIC, + /** Default link clock */ + SND_PCM_STARTAT_CLOCK_TYPE_LINK, + SND_PCM_STARTAT_CLOCK_TYPE_LAST = SND_PCM_STARTAT_CLOCK_TYPE_LINK, +} snd_pcm_startat_clock_type_t; + /** Unsigned frames quantity */ typedef unsigned long snd_pcm_uframes_t; /** Signed frames quantity */ @@ -478,6 +489,9 @@ int snd_pcm_prepare(snd_pcm_t *pcm); int snd_pcm_reset(snd_pcm_t *pcm); int snd_pcm_status(snd_pcm_t *pcm, snd_pcm_status_t *status); int snd_pcm_start(snd_pcm_t *pcm); +int snd_pcm_start_at(snd_pcm_t *pcm, snd_pcm_startat_clock_type_t clock_type, const snd_htimestamp_t* start_time); +int snd_pcm_start_at_abort(snd_pcm_t *pcm); +int snd_pcm_start_at_gettime(snd_pcm_t *pcm, snd_pcm_startat_clock_type_t clock_type, snd_htimestamp_t* current_time); int snd_pcm_drop(snd_pcm_t *pcm); int snd_pcm_drain(snd_pcm_t *pcm); int snd_pcm_pause(snd_pcm_t *pcm, int enable); @@ -984,6 +998,7 @@ snd_pcm_sframes_t snd_pcm_status_get_delay(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail_max(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_overrange(const snd_pcm_status_t *obj); +void snd_pcm_status_get_startat_state(const snd_pcm_status_t *obj, int *pending, snd_pcm_startat_clock_type_t *clock_type, snd_htimestamp_t *start_time); /** \} */ diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index e74e02f..f2cbd8e 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1085,6 +1085,74 @@ int snd_pcm_start(snd_pcm_t *pcm) } /** + * \brief Start a PCM at a specified point in the future + * \param pcm PCM handle + * \param clock_type Specifies the start_at clock with which to interpret \p start_time + * \param start_time Absolute time at which to start the stream + * \return 0 on success otherwise a negative error code + * \retval -ENOSYS operation not supported for the current timestamp type + * \retval -EINVAL timespec, tstamp_class or tstamp_type is invalid + * \retval -ETIME requested start_time cannot be satisfied + * + * This method is non-blocking: It establishes an appropriate timer in the kernel + * that will start the stream on expiry. + * + * Any pending timer is unconditionally cancelled. + */ +int snd_pcm_start_at(snd_pcm_t *pcm, snd_pcm_startat_clock_type_t clock_type, const snd_htimestamp_t *start_time) +{ + assert(pcm); + assert(start_time); + if (CHECK_SANITY(! pcm->setup)) { + SNDMSG("PCM not set up"); + return -EIO; + } + if (pcm->fast_ops->start_at) { + return pcm->fast_ops->start_at(pcm->fast_op_arg, clock_type, start_time); + } + return -EINVAL; +} + +/** + * \brief Abort the pending PCM start_at timer + * \param pcm PCM handle + * \return 0 on success otherwise a negative error code + * + * If a start_at timer is pending, it is cancelled. + */ +int snd_pcm_start_at_abort(snd_pcm_t *pcm) +{ + assert(pcm); + if (CHECK_SANITY(! pcm->setup)) { + SNDMSG("PCM not set up"); + return -EIO; + } + if (pcm->fast_ops->start_at_abort) { + return pcm->fast_ops->start_at_abort(pcm->fast_op_arg); + } + return -EINVAL; +} + +/** + * \brief Get current time for a given start_at clock + * \param pcm PCM handle + * \param clock_type Start_at clock type e.g. SND_PCM_STARTAT_CLOCK_TYPE_GETTIMEOFDAY + * \return 0 on success otherwise a negative error code + */ +int snd_pcm_start_at_gettime(snd_pcm_t *pcm, snd_pcm_startat_clock_type_t clock_type, snd_htimestamp_t *current_time) +{ + assert(pcm); + if (CHECK_SANITY(! pcm->setup)) { + SNDMSG("PCM not set up"); + return -EIO; + } + if (pcm->fast_ops->start_at_gettime) { + return pcm->fast_ops->start_at_gettime(pcm->fast_op_arg, clock_type, current_time); + } + return -EINVAL; +} + +/** * \brief Stop a PCM dropping pending frames * \param pcm PCM handle * \return 0 on success otherwise a negative error code @@ -6380,6 +6448,20 @@ snd_pcm_uframes_t snd_pcm_status_get_overrange(const snd_pcm_status_t *obj) } /** + * \brief Get status of a pending start_at timer, if any. + * \param pending output param: 1 if start_at timer pending, 0 otherwise + * \param clock_type output param: if pending == 1, is updated with current start_at timer clock_type + * \param start_time output param: if pending == 1, is updated with current start_at timer start_time + */ +void snd_pcm_status_get_startat_state(const snd_pcm_status_t *obj, int *pending, snd_pcm_startat_clock_type_t *clock_type, snd_htimestamp_t *start_time) +{ + assert(obj); + *pending = obj->start_at_pending; + *clock_type = obj->start_at_clock_type; + *start_time = obj->start_at_start_time; +} + +/** * \brief get size of #snd_pcm_info_t * \return size in bytes */ -- 1.9.1

1 2

[alsa-devel] [PATCH 2/2] davinci-mcasp: add option to use AHCLKX pin as clock in
by Urs Fässler 06 Feb '15

06 Feb '15

Signed-off-by: Urs Fässler <urs.fassler(a)bytesatwork.ch> --- .../bindings/sound/davinci-mcasp-audio.txt | 1 + sound/soc/davinci/davinci-mcasp.c | 26 +++++++++++++++++++--- 2 files changed, 24 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index 46bc982..22a91d2 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -39,6 +39,7 @@ Optional properties: please refer to pinctrl-bindings.txt - fck_parent : Should contain a valid clock name which will be used as parent for the McASP fck +- ahclkx-pin : if set use AHCLKX pin for clock in instead of ACLKX Example: diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 396cdec..3730288 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -85,6 +85,7 @@ struct davinci_mcasp { int sysclk_freq; bool bclk_master; + bool ahclkx_pin; /* McASP FIFO related */ u8 txnumevt; @@ -429,7 +430,12 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ - mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + if (mcasp->ahclkx_pin) + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE); + else + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE); mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); @@ -441,7 +447,12 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is clock and frame master */ - mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + if (mcasp->ahclkx_pin) + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE); + else + mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); @@ -542,7 +553,12 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); if (mcasp->bclk_master) { - mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); + if (mcasp->ahclkx_pin) + mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); + else + mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX); } else { @@ -1340,6 +1356,8 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of( if (ret >= 0) pdata->sram_size_capture = val; + pdata->clk_input_pin = of_property_read_bool(np, "ahclkx-pin"); + return pdata; nodata: @@ -1412,6 +1430,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) goto err; } + mcasp->ahclkx_pin = pdata->clk_input_pin; + mcasp->op_mode = pdata->op_mode; /* sanity check for tdm slots parameter */ if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) { -- 2.1.4

2 1

[alsa-devel] [PATCH v3] ASoC: rt286: add jack detection enable/disable APIs and export
by Jie Yang 06 Feb '15

06 Feb '15

Some platforms, e.g. WSB, don't need jack detection when system is in Suspend, for power save reason. Here add jack detection enable/disable APIs and export them, when disabled, it will ignore interrupt, and disable LDO1, which is used for jack detection when headphone is plugged in. Signed-off-by: Jie Yang <yang.jie(a)intel.com> Reviewed-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 29 +++++++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 2 ++ 2 files changed, 31 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index d5cc1bc..1a0c250f 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -402,6 +402,33 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(rt286_mic_detect); +int rt286_enable_jack_detection(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if (!jack) + return -EINVAL; + rt286->jack = jack; + if (jack->status | SND_JACK_HEADPHONE) + snd_soc_dapm_force_enable_pin(&rt286->codec->dapm, "LDO1"); + snd_soc_dapm_sync(&rt286->codec->dapm); + return 0; +} +EXPORT_SYMBOL_GPL(rt286_enable_jack_detection); + +int rt286_disable_jack_detection(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if (!jack) + return -EINVAL; + rt286->jack = NULL; + snd_soc_dapm_disable_pin(&rt286->codec->dapm, "LDO1"); + snd_soc_dapm_sync(&rt286->codec->dapm); + return 0; +} +EXPORT_SYMBOL_GPL(rt286_disable_jack_detection); + static int is_mclk_mode(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -1012,6 +1039,8 @@ static irqreturn_t rt286_irq(int irq, void *data) bool mic = false; int status = 0; + if (!rt286->jack) + return IRQ_HANDLED; rt286_jack_detect(rt286, &hp, &mic); /* Clear IRQ */ diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h index 7130edb..d793185 100644 --- a/sound/soc/codecs/rt286.h +++ b/sound/soc/codecs/rt286.h @@ -200,6 +200,8 @@ enum { }; int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); +int rt286_enable_jack_detection(struct snd_soc_codec *codec, struct snd_soc_jack *jack); +int rt286_disable_jack_detection(struct snd_soc_codec *codec, struct snd_soc_jack *jack); #endif /* __RT286_H__ */ -- 1.9.1

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[alsa-devel] [PATCH 1/2] davinci-mcasp: use bclk_master flag to decide if the SOC is clock master
by Urs Fässler 06 Feb '15

06 Feb '15

Signed-off-by: Urs Fässler <urs.fassler(a)bytesatwork.ch> --- 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 30b94d4..396cdec 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -541,7 +541,7 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - if (dir == SND_SOC_CLOCK_OUT) { + if (mcasp->bclk_master) { mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX); -- 2.1.4

1 0

[alsa-devel] [PATCH] ALSA: hda - Set up GPIO for Toshiba Satellite S50D
by Takashi Iwai 05 Feb '15

05 Feb '15

Toshiba Satellite S50D laptop with an IDT codec uses the GPIO4 (0x10) as the master EAPD. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=915858 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 605d14003d25..6d36c5b78805 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -99,6 +99,7 @@ enum { STAC_HP_ENVY_BASS, STAC_HP_BNB13_EQ, STAC_HP_ENVY_TS_BASS, + STAC_92HD83XXX_GPIO10_EAPD, STAC_92HD83XXX_MODELS }; @@ -2141,6 +2142,19 @@ static void stac92hd83xxx_fixup_headset_jack(struct hda_codec *codec, spec->headset_jack = 1; } +static void stac92hd83xxx_fixup_gpio10_eapd(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct sigmatel_spec *spec = codec->spec; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = + spec->gpio_data = 0x10; + spec->eapd_switch = 0; +} + static const struct hda_verb hp_bnb13_eq_verbs[] = { /* 44.1KHz base */ { 0x22, 0x7A6, 0x3E }, @@ -2656,6 +2670,10 @@ static const struct hda_fixup stac92hd83xxx_fixups[] = { {} }, }, + [STAC_92HD83XXX_GPIO10_EAPD] = { + .type = HDA_FIXUP_FUNC, + .v.func = stac92hd83xxx_fixup_gpio10_eapd, + }, }; static const struct hda_model_fixup stac92hd83xxx_models[] = { @@ -2861,6 +2879,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x148a, "HP Mini", STAC_92HD83XXX_HP_LED), SND_PCI_QUIRK_VENDOR(PCI_VENDOR_ID_HP, "HP", STAC_92HD83XXX_HP), + SND_PCI_QUIRK(PCI_VENDOR_ID_TOSHIBA, 0xfa91, + "Toshiba Satellite S50D", STAC_92HD83XXX_GPIO10_EAPD), {} /* terminator */ }; -- 2.2.2

1 0

[alsa-devel] ASoC fix for v3.19
by Mark Brown 05 Feb '15

05 Feb '15

The following changes since commit 97fa99a3b8aa83c8ff17ef380dd377efac373812: MAINTAINERS: ASoC: add maintainer for Intel BDW/HSW ASoC driver (2015-02-02 12:00:20 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-intel-v3.19-rc7 for you to fetch changes up to 5c2b06369dafd796ebb4f17dab543d3da500245e: ASoC: Intel: fix sst firmware path for cht-bsw-rt5672 (2015-02-05 20:16:21 +0000) ---------------------------------------------------------------- ASoC: Fix for Intel firmware name Another one liner that arrived after the earlier pull request. There's a trivial conflict with my -next branch, I'll send a pull request with that resolution and some other new stuff before Monday. ---------------------------------------------------------------- Kevin Strasser (1): ASoC: Intel: fix sst firmware path for cht-bsw-rt5672 sound/soc/intel/sst/sst_acpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)

2 1

[alsa-devel] ASoC AC'97 fix for v3.19
by Mark Brown 05 Feb '15

05 Feb '15

The following changes since commit 97bf6af1f928216fd6c5a66e8a57bfa95a659672: Linux 3.19-rc1 (2014-12-20 17:08:50 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-ac97-v3.19-rc7 for you to fetch changes up to f8d71be5553a3fbc20363243ecfc11dcdd1e18fe: ASoC: wm97xx: Reset AC'97 device before registering it (2015-01-26 19:15:06 +0000) ---------------------------------------------------------------- ASoC: AC'97 fixes These are rather too large for this late in the release cycle but they're clear, well understood and have been tested to fix a regression which was introduced for v3.19. The details are all in Lars' changelog and they've been cooking in -next for a while, to a large extent out of conservatism about the size. ---------------------------------------------------------------- Lars-Peter Clausen (2): ASoC: Add support for allocating AC'97 device before registering it ASoC: wm97xx: Reset AC'97 device before registering it include/sound/soc.h | 1 + sound/soc/codecs/wm9705.c | 16 ++++++++++------ sound/soc/codecs/wm9712.c | 12 ++++++++---- sound/soc/codecs/wm9713.c | 12 ++++++++---- sound/soc/soc-ac97.c | 36 ++++++++++++++++++++++++++++++------ 5 files changed, 57 insertions(+), 20 deletions(-)

2 1

[alsa-devel] ASoC fixes for v3.19
by Mark Brown 05 Feb '15

05 Feb '15

The following changes since commit e36f014edff70fc02b3d3d79cead1d58f289332e: Linux 3.19-rc7 (2015-02-01 20:07:21 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v3.19-rc7 for you to fetch changes up to c503ca52a57cfe24f8937ca145a96ba94f6bb112: Merge remote-tracking branch 'asoc/fix/samsung' into asoc-linus (2015-02-05 19:36:29 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.19 A few last minute fixes for v3.19, all driver specific. None of them stand out particularly - it's all the standard people who are affected will care stuff. The Samsung fix is a DT only fix for the audio controller, it's being merged via the ASoC tree due to process messups (the submitter sent it at the end of a tangentally related series rather than separately to the ARM folks) in order to make sure that it gets to people sooner. ---------------------------------------------------------------- Bo Shen (2): ASoC: atmel_ssc_dai: fix start event for I2S mode ASoC: atmel_ssc_dai: fix the setting for DSP mode Eric Nelson (1): ASoC: sgtl5000: add delay before first I2C access Filip Brozovic (1): ASoC: sgtl5000: Use shift mask when setting codec mode Jarkko Nikula (1): ASoC: rt5640: Add RT5642 ACPI ID for Intel Baytrail Jie Yang (2): ASoC: Intel: Used lock version to update shim registers MAINTAINERS: ASoC: add maintainer for Intel BDW/HSW ASoC driver Manuel Lauss (1): ASoC: wm8731: init mutex in i2c init path Mark Brown (7): Merge remote-tracking branch 'asoc/fix/atmel' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5640' into asoc-linus Merge remote-tracking branch 'asoc/fix/sgtl5000' into asoc-linus Merge remote-tracking branch 'asoc/fix/tlv320aic3x' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8731' into asoc-linus Merge remote-tracking branch 'asoc/fix/samsung' into asoc-linus Peter Ujfalusi (1): ASoC: tlv320aic3x: Fix data delay configuration Sylwester Nawrocki (1): ARM: dts: Fix I2S1, I2S2 compatible for exynos4 SoCs MAINTAINERS | 10 ++++++++++ arch/arm/boot/dts/exynos4.dtsi | 4 ++-- sound/soc/atmel/atmel_ssc_dai.c | 24 +++++++----------------- sound/soc/codecs/rt5640.c | 1 + sound/soc/codecs/sgtl5000.c | 13 ++++++++----- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/wm8731.c | 2 ++ sound/soc/intel/sst-haswell-ipc.c | 4 ++-- 8 files changed, 33 insertions(+), 27 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: Intel: fix sst firmware path
by Kevin Strasser 05 Feb '15

05 Feb '15

All sst firmware is provided under the intel directory of the linux-firmware tree. By default this directory structure is kept when installing on a target system. Change the path to expect a default linux-firmware installation. Signed-off-by: Kevin Strasser <kevin.strasser(a)linux.intel.com> --- sound/soc/intel/sst/sst_acpi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 21b22e6..378ef3c 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -343,16 +343,16 @@ static int sst_acpi_remove(struct platform_device *pdev) } static struct sst_machines sst_acpi_bytcr[] = { - {"10EC5640", "T100", "bytt100_rt5640", NULL, "fw_sst_0f28.bin", + {"10EC5640", "T100", "bytt100_rt5640", NULL, "intel/fw_sst_0f28.bin", &byt_rvp_platform_data }, {}, }; /* Cherryview-based platforms: CherryTrail and Braswell */ static struct sst_machines sst_acpi_chv[] = { - {"10EC5670", "cht-bsw", "cht-bsw-rt5672", NULL, "fw_sst_22a8.bin", + {"10EC5670", "cht-bsw", "cht-bsw-rt5672", NULL, "intel/fw_sst_22a8.bin", &chv_platform_data }, - {"10EC5645", "cht-bsw", "cht-bsw-rt5645", NULL, "fw_sst_22a8.bin", + {"10EC5645", "cht-bsw", "cht-bsw-rt5645", NULL, "intel/fw_sst_22a8.bin", &chv_platform_data }, {}, }; -- 1.9.1

5 9

[alsa-devel] [PATCH 1/2] ASoC: rt286: Fix potencial crash in jd function
by Bard Liao 05 Feb '15

05 Feb '15

We assign rt286->codec in rt286_probe. If rt286_jack_detect is invoked before rt286_probe, rt286->codec will be NULL and cause a kernel panic. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 2eafbcf..3d69cb4 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -306,6 +306,8 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) *hp = false; *mic = false; + if (!rt286->codec) + return -EINVAL; if (rt286->pdata.cbj_en) { regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf); *hp = buf & 0x80000000; -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH v2] ASoC: rt286: add jack detection enable/disable APIs and export
by Jie Yang 05 Feb '15

05 Feb '15

Some platforms, e.g. WSB, don't need jack detection when system is in Suspend, for power save reason. Here add jack detection enable/disable APIs and export them, when disabled, it will ignore interrupt, and disable LDO1, which is used for jack detection when headphone is plugged in. Signed-off-by: Jie Yang <yang.jie(a)intel.com> Reviewed-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 24 ++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 2 ++ 2 files changed, 26 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 8104d22..9ea01d0 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -400,6 +400,30 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(rt286_mic_detect); +int rt286_enable_jack_detection(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if (rt286->jack->status | SND_JACK_HEADPHONE) + snd_soc_dapm_force_enable_pin(&rt286->codec->dapm, "LDO1"); + snd_soc_dapm_sync(&rt286->codec->dapm); + enable_irq(rt286->i2c->irq); + return 0; +} +EXPORT_SYMBOL_GPL(rt286_enable_jack_detection); + +int rt286_disable_jack_detection(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + disable_irq(rt286->i2c->irq); + + snd_soc_dapm_disable_pin(&rt286->codec->dapm, "LDO1"); + snd_soc_dapm_sync(&rt286->codec->dapm); + return 0; +} +EXPORT_SYMBOL_GPL(rt286_disable_jack_detection); + static int is_mclk_mode(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h index b539b73..c958182 100644 --- a/sound/soc/codecs/rt286.h +++ b/sound/soc/codecs/rt286.h @@ -193,6 +193,8 @@ enum { }; int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); +int rt286_enable_jack_detection(struct snd_soc_codec *codec); +int rt286_disable_jack_detection(struct snd_soc_codec *codec); #endif /* __RT286_H__ */ -- 1.9.1

3 3

[alsa-devel] [PATCH 1/2] ASoC: Intel:Add Cherrytrail & Braswell machine driver cht_bsw_rt5645
by yang.a.fang＠intel.com 05 Feb '15

05 Feb '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> Add machine driver for two Intel Cherryview-based platforms, Cherrytrail and Braswell, with RT5645 codec Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/intel/Kconfig | 11 ++ sound/soc/intel/Makefile | 2 + sound/soc/intel/cht_bsw_rt5645.c | 317 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 330 insertions(+) create mode 100644 sound/soc/intel/cht_bsw_rt5645.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index f06fcf1..12093fd 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -110,3 +110,14 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH platforms with RT5672 audio codec. Say Y if you have such a device 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 + 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. + If unsure select "N". diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index e928ec3..a8e53c4 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -28,6 +28,7 @@ snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bytcr-dpcm-rt5640-objs := bytcr_dpcm_rt5640.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o +snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -35,6 +36,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-dpcm-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o # DSP driver obj-$(CONFIG_SND_SST_IPC) += sst/ diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/cht_bsw_rt5645.c new file mode 100644 index 0000000..7a6f1e5 --- /dev/null +++ b/sound/soc/intel/cht_bsw_rt5645.c @@ -0,0 +1,317 @@ +/* + * cht-bsw-rt5645.c - ASoc Machine driver for Intel Cherryview-based platforms + * Cherrytrail and Braswell, with RT5645 codec. + * + * Copyright (C) 2014 Intel Corp + * Author: Fang, Yang A <yang.a.fang(a)intel.com> + * N,Harshapriya <harshapriya.n(a)intel.com> + * This file is modified from cht_bsw_rt5672.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; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../codecs/rt5645.h" +#include "sst-atom-controls.h" + +#define CHT_PLAT_CLK_3_HZ 19200000 +#define CHT_CODEC_DAI "rt5645-aif1" + +struct cht_mc_private { + struct snd_soc_jack hp_jack; + struct snd_soc_jack mic_jack; +}; + +static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) +{ + int i; + + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd; + + rtd = card->rtd + i; + if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI, + strlen(CHT_CODEC_DAI))) + return rtd->codec_dai; + } + return NULL; +} + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + + codec_dai = cht_get_codec_dai(card); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n"); + return -EIO; + } + if (!SND_SOC_DAPM_EVENT_OFF(event)) + return 0; + + /* Set codec sysclk source to its internal clock because codec PLL will + * be off when idle and MCLK will also be off by ACPI when codec is + * runtime suspended. Codec needs clock for jack detection and button + * press. + */ + snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_RCCLK, + 0, SND_SOC_CLOCK_IN); + + return 0; +} + +static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route cht_audio_map[] = { + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + {"DMIC L1", NULL, "Int Mic"}, + {"DMIC R1", 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"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static int cht_aif1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK, + CHT_PLAT_CLK_3_HZ, params_rate(params) * 512); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1, + params_rate(params) * 512, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + 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); + + /* Select clk_i2s1_asrc as ASRC clock source */ + snd_soc_write(codec, RT5645_ASRC_2, 0x1111); + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); + if (ret < 0) { + dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + ret = snd_soc_jack_new(codec, "Headphone Jack", + SND_JACK_HEADPHONE, + &ctx->hp_jack); + if (ret) { + dev_err(runtime->dev, "Headphone jack creation failed\n"); + return ret; + } + + ret = snd_soc_jack_new(codec, "Mic Jack", + SND_JACK_MICROPHONE, + &ctx->mic_jack); + if (ret) { + dev_err(runtime->dev, "Mic jack creation failed\n"); + return ret; + } + + rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack); + + return ret; +} + +static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + 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); + return 0; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int cht_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops cht_aif1_ops = { + .startup = cht_aif1_startup, +}; + +static struct snd_soc_ops cht_be_ssp2_ops = { + .hw_params = cht_aif1_hw_params, +}; + +static struct snd_soc_dai_link cht_dailink[] = { + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .ignore_suspend = 1, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + /* CODEC<->CODEC link */ + /* back ends */ + { + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "rt5645-aif1", + .codec_name = "i2c-10EC5645:00", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .init = cht_codec_init, + .be_hw_params_fixup = cht_codec_fixup, + .ignore_suspend = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_be_ssp2_ops, + }, +}; + +/* 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 int snd_cht_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + struct cht_mc_private *drv; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + if (!drv) + return -ENOMEM; + + /* register the soc card */ + 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); + 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); + return ret_val; +} + +static struct platform_driver snd_cht_mc_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "cht-bsw-rt5645", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_cht_mc_probe, +}; + +module_platform_driver(snd_cht_mc_driver) + +MODULE_DESCRIPTION("ASoC Intel(R) Braswell Machine driver"); +MODULE_AUTHOR("Fang, Yang A,N,Harshapriya"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht-bsw-rt5645"); -- 1.7.9.5

3 13

[alsa-devel] [PATCH v3] ASoC: Intel: add a status for runtime suspend/resume
by Jie Yang 05 Feb '15

05 Feb '15

For runtime suspend/resume, it is some different with suspend/resume, e.g. codec power supply won't be switch off, codec jack detection still working(to wake up system from Jack event), won't call call snd_soc_suspend/resume, etc. So here, we add a platform PM status, HSW_PM_STATE_RTD3, to make the status more clear, when in idle, it will enter this status, to transfer from HSW_PM_STATE_RTD3 to HSW_PM_STATE_D3, we will do those extra jobs, and vice versa for resuming. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 70 ++++++++++++++++++++------------------- 1 file changed, 36 insertions(+), 34 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index ad7f4a5..78fa01b 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -119,8 +119,9 @@ struct hsw_pcm_data { }; enum hsw_pm_state { - HSW_PM_STATE_D3 = 0, - HSW_PM_STATE_D0 = 1, + HSW_PM_STATE_D0 = 0, + HSW_PM_STATE_RTD3 = 1, + HSW_PM_STATE_D3 = 2, }; /* private data for the driver */ @@ -1035,12 +1036,12 @@ static int hsw_pcm_runtime_suspend(struct device *dev) struct hsw_priv_data *pdata = dev_get_drvdata(dev); struct sst_hsw *hsw = pdata->hsw; - if (pdata->pm_state == HSW_PM_STATE_D3) + if (pdata->pm_state >= HSW_PM_STATE_RTD3) return 0; sst_hsw_dsp_runtime_suspend(hsw); sst_hsw_dsp_runtime_sleep(hsw); - pdata->pm_state = HSW_PM_STATE_D3; + pdata->pm_state = HSW_PM_STATE_RTD3; return 0; } @@ -1051,7 +1052,7 @@ static int hsw_pcm_runtime_resume(struct device *dev) struct sst_hsw *hsw = pdata->hsw; int ret; - if (pdata->pm_state == HSW_PM_STATE_D0) + if (pdata->pm_state != HSW_PM_STATE_RTD3) return 0; ret = sst_hsw_dsp_load(hsw); @@ -1091,7 +1092,7 @@ static void hsw_pcm_complete(struct device *dev) struct hsw_pcm_data *pcm_data; int i, err; - if (pdata->pm_state == HSW_PM_STATE_D0) + if (pdata->pm_state != HSW_PM_STATE_D3) return; err = sst_hsw_dsp_load(hsw); @@ -1139,41 +1140,42 @@ static int hsw_pcm_prepare(struct device *dev) if (pdata->pm_state == HSW_PM_STATE_D3) return 0; - /* suspend all active streams */ - for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; + else if (pdata->pm_state == HSW_PM_STATE_D0) { + /* suspend all active streams */ + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; + + if (!pcm_data->substream) + continue; + dev_dbg(dev, "suspending pcm %d\n", i); + snd_pcm_suspend_all(pcm_data->hsw_pcm); + + /* We need to wait until the DSP FW stops the streams */ + msleep(2); + } - if (!pcm_data->substream) - continue; - dev_dbg(dev, "suspending pcm %d\n", i); - snd_pcm_suspend_all(pcm_data->hsw_pcm); + /* preserve persistent memory */ + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; + + if (!pcm_data->substream) + continue; - /* We need to wait until the DSP FW stops the streams */ - msleep(2); + dev_dbg(dev, "saving context pcm %d\n", i); + err = sst_module_runtime_save(pcm_data->runtime, + &pcm_data->context); + if (err < 0) + dev_err(dev, "failed to save context for PCM %d\n", i); + } + /* enter D3 state and stall */ + sst_hsw_dsp_runtime_suspend(hsw); + /* put the DSP to sleep */ + sst_hsw_dsp_runtime_sleep(hsw); } snd_soc_suspend(pdata->soc_card->dev); snd_soc_poweroff(pdata->soc_card->dev); - /* enter D3 state and stall */ - sst_hsw_dsp_runtime_suspend(hsw); - - /* preserve persistent memory */ - for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; - - if (!pcm_data->substream) - continue; - - dev_dbg(dev, "saving context pcm %d\n", i); - err = sst_module_runtime_save(pcm_data->runtime, - &pcm_data->context); - if (err < 0) - dev_err(dev, "failed to save context for PCM %d\n", i); - } - - /* put the DSP to sleep */ - sst_hsw_dsp_runtime_sleep(hsw); pdata->pm_state = HSW_PM_STATE_D3; return 0; -- 1.9.1

2 1

[alsa-devel] [PATCH] linux-firmware: update firmware for Intel Broadwell SST DSP
by Jie Yang 05 Feb '15

05 Feb '15

Update firmware intel/IntcSST2.bin from Version 8.4.1.68 to 8.4.1.77 Fix for freeing capture stream just after resetting it. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- WHENCE | 1 + intel/IntcSST2.bin | Bin 260256 -> 260320 bytes 2 files changed, 1 insertion(+) diff --git a/WHENCE b/WHENCE index 7f2d6bf..6cb0778 100644 --- a/WHENCE +++ b/WHENCE @@ -2646,6 +2646,7 @@ Licence: Redistributable. See LICENCE.it913x for details Driver: snd_soc_sst_acpi File: intel/IntcSST2.bin +Version: 8.4.1.77 License: Redistributable. See LICENCE.IntcSST2 for details diff --git a/intel/IntcSST2.bin b/intel/IntcSST2.bin index 69ac37e78582dc5971b56eb598f43821958a9a59..e11cb61f4a00bf32a304781e3e9b1f32775285e1 100644 GIT binary patch delta 35726 zcmZTQ30zZ0^VuW>2oK~8w?O16AS%HV#UrQ)-guu?g3ntk+QXZpEk&iRuwVPY+R95C z1q3z014NXdASf!}g=a;5UU+~gf};7)CeW|_{``KAw>vv~%<RnU?Cic|Hg&nz)WzC& z>eOi^ts;SiKp^;p6Ocoyw;0_^rH9*B-$H^}w~@f{4iX$*9w7+5tNm(@$Zq{cp&-i~ zu;DM%P9V64>qXs!wu$)?u?(R@*wd`f{Px^PfyI;vfjBQxFbm%6;XyBj0t)i!khX<) zE4(S_)naOdz!CB;@LsbrLJ$Gz$>9-#5%7$Mv>(9Ziv)trP__v2$vE4rXGk5S46h6Z z!}kWmJxFIl=}veaLRuh*k83<85EPJ_&`Jfo4P69+c7s6h1Y1i6+C3+=4T8qQP-FeH z2*DdXUD7w89iZt8A_Q5hA_RWZBLuBbLK?F;5Ghz=hoFXGI6S9Zu4A1fcxWJi1qTcU zk-!kw7$6kX!BYZH7CbxQSr1S8Yk^=Pyd&TVg2x3O0X$EzkNH5<56?F56Fmspv6ul} z#X>o}VPFEkNx%Sr`$3*sV;qLSGZE^Jh!Ct?1dM{GWQFNT&WaG+6Tw^o?jU$<0e=+K zjexWxq$BWC^L3Wv0b($GG8ACOq6eCcf3UcOX5smkGtokP#L~;ZdOjk=B7wI+un2G+ zV}TK$MF^Hcx&q!S;q3y<S%W`XPK{a*xkau5K?=P67DE3KFviFTfj2zK<A4|8LP5nS 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[alsa-devel] [PATCH v2] ASoC: Intel: add a status for runtime suspend/resume
by Jie Yang 05 Feb '15

05 Feb '15

For runtime suspend/resume, it is some different with suspend/resume, e.g. codec power supply won't be switch off, codec jack detection still working(to wake up system from Jack event), won't call call snd_soc_suspend/resume, etc. So here, we add a platform PM status, HSW_PM_STATE_RTD3, to make the status more clear, when in idle, it will enter this status, to transfer from HSW_PM_STATE_RTD3 to HSW_PM_STATE_D3, we will do those extra jobs, and vice versa for resuming. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 70 ++++++++++++++++++++------------------- 1 file changed, 36 insertions(+), 34 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index ad7f4a5..edf8eaf 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -119,8 +119,9 @@ struct hsw_pcm_data { }; enum hsw_pm_state { - HSW_PM_STATE_D3 = 0, - HSW_PM_STATE_D0 = 1, + HSW_PM_STATE_D0 = 0, + HSW_PM_STATE_RTD3 = 1, + HSW_PM_STATE_D3 = 2, }; /* private data for the driver */ @@ -1035,12 +1036,12 @@ static int hsw_pcm_runtime_suspend(struct device *dev) struct hsw_priv_data *pdata = dev_get_drvdata(dev); struct sst_hsw *hsw = pdata->hsw; - if (pdata->pm_state == HSW_PM_STATE_D3) + if (pdata->pm_state >= HSW_PM_STATE_RTD3) return 0; sst_hsw_dsp_runtime_suspend(hsw); sst_hsw_dsp_runtime_sleep(hsw); - pdata->pm_state = HSW_PM_STATE_D3; + pdata->pm_state = HSW_PM_STATE_RTD3; return 0; } @@ -1051,7 +1052,7 @@ static int hsw_pcm_runtime_resume(struct device *dev) struct sst_hsw *hsw = pdata->hsw; int ret; - if (pdata->pm_state == HSW_PM_STATE_D0) + if (pdata->pm_state != HSW_PM_STATE_RTD3) return 0; ret = sst_hsw_dsp_load(hsw); @@ -1091,7 +1092,7 @@ static void hsw_pcm_complete(struct device *dev) struct hsw_pcm_data *pcm_data; int i, err; - if (pdata->pm_state == HSW_PM_STATE_D0) + if (pdata->pm_state != HSW_PM_STATE_D3) return; err = sst_hsw_dsp_load(hsw); @@ -1139,41 +1140,42 @@ static int hsw_pcm_prepare(struct device *dev) if (pdata->pm_state == HSW_PM_STATE_D3) return 0; - /* suspend all active streams */ - for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; + else if (pdata->pm_state == HSW_PM_STATE_D0){ + /* suspend all active streams */ + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; + + if (!pcm_data->substream) + continue; + dev_dbg(dev, "suspending pcm %d\n", i); + snd_pcm_suspend_all(pcm_data->hsw_pcm); + + /* We need to wait until the DSP FW stops the streams */ + msleep(2); + } - if (!pcm_data->substream) - continue; - dev_dbg(dev, "suspending pcm %d\n", i); - snd_pcm_suspend_all(pcm_data->hsw_pcm); + /* preserve persistent memory */ + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; + + if (!pcm_data->substream) + continue; - /* We need to wait until the DSP FW stops the streams */ - msleep(2); + dev_dbg(dev, "saving context pcm %d\n", i); + err = sst_module_runtime_save(pcm_data->runtime, + &pcm_data->context); + if (err < 0) + dev_err(dev, "failed to save context for PCM %d\n", i); + } + /* enter D3 state and stall */ + sst_hsw_dsp_runtime_suspend(hsw); + /* put the DSP to sleep */ + sst_hsw_dsp_runtime_sleep(hsw); } snd_soc_suspend(pdata->soc_card->dev); snd_soc_poweroff(pdata->soc_card->dev); - /* enter D3 state and stall */ - sst_hsw_dsp_runtime_suspend(hsw); - - /* preserve persistent memory */ - for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; - - if (!pcm_data->substream) - continue; - - dev_dbg(dev, "saving context pcm %d\n", i); - err = sst_module_runtime_save(pcm_data->runtime, - &pcm_data->context); - if (err < 0) - dev_err(dev, "failed to save context for PCM %d\n", i); - } - - /* put the DSP to sleep */ - sst_hsw_dsp_runtime_sleep(hsw); pdata->pm_state = HSW_PM_STATE_D3; return 0; -- 1.9.1

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