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

[alsa-devel] [PATCH v3 1/2] ASoC: rt5645: Add dmi for Broadwell
by John Lin 22 Sep '15

22 Sep '15

Add DMI data for Buddy project. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5645.c | 28 +++++++++++++++++++++++++++- 1 file changed, 27 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index dbc1d76..e830688 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3209,6 +3209,31 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { { } }; +static struct rt5645_platform_data buddy_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, + .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, + .jd_mode = 3, +}; + +static int buddy_quirk_cb(const struct dmi_system_id *id) +{ + rt5645_pdata = &buddy_platform_data; + + return 1; +} + +static struct dmi_system_id dmi_platform_intel_broadwell[] __initdata = { + { + .ident = "Chrome Buddy", + .callback = buddy_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Buddy"), + }, + }, + { } +}; + + static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) { rt5645->pdata.in2_diff = device_property_read_bool(dev, @@ -3241,7 +3266,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (pdata) rt5645->pdata = *pdata; - else if (dmi_check_system(dmi_platform_intel_braswell)) + else if (dmi_check_system(dmi_platform_intel_braswell) || + dmi_check_system(dmi_platform_intel_broadwell)) rt5645->pdata = *rt5645_pdata; else rt5645_parse_dt(rt5645, &i2c->dev); -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 1/2] ASoC: rt5645: Add dmi for Broadwell
by Bard Liao 22 Sep '15

22 Sep '15

Add DMI data for Buddy project. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 28 +++++++++++++++++++++++++++- 1 file changed, 27 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 66e9ce6..2fc437f 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3210,6 +3210,31 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { { } }; +static struct rt5645_platform_data buddy_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, + .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, + .jd_mode = 3, +}; + +static int buddy_quirk_cb(const struct dmi_system_id *id) +{ + rt5645_pdata = &buddy_platform_data; + + return 1; +} + +static struct dmi_system_id dmi_platform_intel_broadwell[] __initdata = { + { + .ident = "Chrome Buddy", + .callback = buddy_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Buddy"), + }, + }, + { } +}; + + static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) { rt5645->pdata.in2_diff = device_property_read_bool(dev, @@ -3242,7 +3267,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (pdata) rt5645->pdata = *pdata; - else if (dmi_check_system(dmi_platform_intel_braswell)) + else if (dmi_check_system(dmi_platform_intel_braswell) || + dmi_check_system(dmi_platform_intel_broadwel) rt5645->pdata = *rt5645_pdata; else rt5645_parse_dt(rt5645, &i2c->dev); -- 1.8.1.1.439.g50a6b54

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[alsa-devel] Applied "MAINTAINERS: Update website and git repo for Wolfson Microelectronics" to the asoc tree
by Mark Brown 22 Sep '15

22 Sep '15

The patch MAINTAINERS: Update website and git repo for Wolfson Microelectronics has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f0e03dbd2d61d991bdd2d76b4e84681fe3077176 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Date: Tue, 22 Sep 2015 15:30:33 +0100 Subject: [PATCH] MAINTAINERS: Update website and git repo for Wolfson Microelectronics Support for Wolfson Microelectronics devices is now part of Cirrus Logic and the relevant parts of the old opensource.wolfsonmicro.com site have moved to the Cirrus Logic GitHub area. This patch updates the website and git repo links, and also removes an obsolete website link for the voltage and current drivers. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- MAINTAINERS | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 7ba7ab7..dffea21 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -11239,7 +11239,6 @@ VOLTAGE AND CURRENT REGULATOR FRAMEWORK M: Liam Girdwood <lgirdwood(a)gmail.com> M: Mark Brown <broonie(a)kernel.org> L: linux-kernel(a)vger.kernel.org -W: http://opensource.wolfsonmicro.com/node/15 W: http://www.slimlogic.co.uk/?p=48 T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regulator.git S: Supported @@ -11368,17 +11367,15 @@ WM97XX TOUCHSCREEN DRIVERS M: Mark Brown <broonie(a)kernel.org> M: Liam Girdwood <lrg(a)slimlogic.co.uk> L: linux-input(a)vger.kernel.org -T: git git://opensource.wolfsonmicro.com/linux-2.6-touch -W: http://opensource.wolfsonmicro.com/node/7 +W: https://github.com/CirrusLogic/linux-drivers/wiki S: Supported F: drivers/input/touchscreen/*wm97* F: include/linux/wm97xx.h WOLFSON MICROELECTRONICS DRIVERS L: patches(a)opensource.wolfsonmicro.com -T: git git://opensource.wolfsonmicro.com/linux-2.6-asoc -T: git git://opensource.wolfsonmicro.com/linux-2.6-audioplus -W: http://opensource.wolfsonmicro.com/content/linux-drivers-wolfson-devices +T: git https://github.com/CirrusLogic/linux-drivers.git +W: https://github.com/CirrusLogic/linux-drivers/wiki S: Supported F: Documentation/hwmon/wm83?? F: arch/arm/mach-s3c64xx/mach-crag6410* -- 2.5.0

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Re: [alsa-devel] [PATCH 1/3] ASoC: codecs: Add da7219 codec driver
by Mark Brown 22 Sep '15

22 Sep '15

On Thu, Sep 17, 2015 at 05:01:14PM +0100, Adam Thomson wrote: > + do { > + statusa = snd_soc_read(codec, DA7219_ACCDET_STATUS_A); > + if (statusa & DA7219_MICBIAS_UP_STS_MASK) > + micbias_up = true; > + } while (!micbias_up); This could go into an inifinite loop. > +static void da7219_aad_hptest_work(struct work_struct *work) > +{ > + struct da7219_aad_priv *da7219_aad = > + container_of(work, struct da7219_aad_priv, hptest_work); > + struct snd_soc_codec *codec = da7219_aad->codec; > + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); > + > + u8 tonegen_cfg1, tonegen_cfg2, tonegen_onper; > + u16 tonegen_freq1, tonegen_freq_hptest; > + u8 hpl_gain, hpr_gain, dacl_gain, dacr_gain, dac_filters1, dac_filters4; > + u8 dac_filters5, cp_ctrl, routing_dac, dacl_ctrl, dacr_ctrl; > + u8 mixoutl_sel, mixoutr_sel, st_outfilt_1l, st_outfilt_1r; > + u8 mixoutl_ctrl, mixoutr_ctrl, hpl_ctrl, hpr_ctrl, accdet_cfg8; > + int report = 0; > + > + /* Save current settings */ This is obviously a massive reconfiguration of the device. I'm not seeing anything here which prevents userspace coming in and change the configuration while we're in this function, that would obviously have serious issues. I'm also wondering if this might be more elegantly implemented by going into cache bypass mode, doing the test and then using a cache resync to restore the initial configuration. That will at least avoid issues with updates adding a new register but not modifying it. > + if (statusa & DA7219_JACK_TYPE_STS_MASK) { > + report |= SND_JACK_HEADSET; > + mask |= SND_JACK_HEADSET | SND_JACK_LINEOUT; > + schedule_work(&da7219_aad->btn_det_work); > + } else { > + schedule_work(&da7219_aad->hptest_work); > + } Why are we scheduling work - we're already in thread context? > + /* Un-drive headphones/lineout */ > + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, > + DA7219_HP_R_AMP_OE_MASK, 0); > + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, > + DA7219_HP_L_AMP_OE_MASK, 0); This looks like DAPM? > +static enum da7219_aad_jack_ins_deb da7219_aad_of_jack_ins_deb(u32 val) > +{ > + switch (val) { > + case 5: > + return DA7219_AAD_JACK_INS_DEB_5MS; > + case 10: > + return DA7219_AAD_JACK_INS_DEB_10MS; > + case 20: > + return DA7219_AAD_JACK_INS_DEB_20MS; > + case 50: > + return DA7219_AAD_JACK_INS_DEB_50MS; > + case 100: > + return DA7219_AAD_JACK_INS_DEB_100MS; > + case 200: > + return DA7219_AAD_JACK_INS_DEB_200MS; > + case 500: > + return DA7219_AAD_JACK_INS_DEB_500MS; > + case 1000: > + return DA7219_AAD_JACK_INS_DEB_1S; > + default: > + return DA7219_AAD_JACK_INS_DEB_20MS; This isn't an error? > +/* Input/Output Enums */ > +static const char * const da7219_gain_ramp_rate_txt[] = { > + "nominal rate * 8", "nominal rate", "nominal rate / 8", > + "nominal rate / 16" > +}; The ALSA ABI generally capitalises words. > +/* ToneGen */ > +static int da7219_tonegen_freq_get(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); > + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); > + struct soc_mixer_control *mixer_ctrl = > + (struct soc_mixer_control *) kcontrol->private_value; > + unsigned int reg = mixer_ctrl->reg; > + u16 val; > + int ret; > + > + ret = regmap_bulk_read(da7219->regmap, reg, &val, sizeof(val)); > + if (ret) > + return ret; > + > + ucontrol->value.integer.value[0] = le16_to_cpu(val); This is *weird*. We do a bulk read for a single register using an API that returns CPU endian data then make it CPU endian (without any annotations on variables...). Why not use regmap_read()? Why swap? Why not use raw I/O? > +static int da7219_hpf_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); > + struct soc_enum *enum_ctrl = (struct soc_enum *)kcontrol->private_value; > + unsigned int reg = enum_ctrl->reg; > + unsigned int sel = ucontrol->value.integer.value[0]; > + unsigned int bits; > + > + switch (sel) { > + case DA7219_HPF_MODE_DISABLED: > + bits = DA7219_HPF_DISABLED; > + break; > + case DA7219_HPF_MODE_AUDIO: > + bits = DA7219_HPF_AUDIO_EN; > + break; > + case DA7219_HPF_MODE_VOICE: > + bits = DA7219_HPF_VOICE_EN; > + break; > + default: > + return -EINVAL; > + } > + > + snd_soc_update_bits(codec, reg, DA7219_HPF_MODE_MASK, bits); > + > + return 0; > +} This looks like a standard enumeration with a simple mapping to the register map? > + > + /* ADCs */ > + SOC_SINGLE_TLV("ADC Volume", DA7219_ADC_L_GAIN, > + DA7219_ADC_L_DIGITAL_GAIN_SHIFT, > + DA7219_ADC_L_DIGITAL_GAIN_MAX, DA7219_NO_INVERT, > + da7219_adc_dig_gain_tlv), > + SOC_SINGLE("ADC Switch", DA7219_ADC_L_CTRL, > + DA7219_ADC_L_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, > + DA7219_INVERT), > + SOC_SINGLE("ADC Gain Ramp Switch", DA7219_ADC_L_CTRL, > + DA7219_ADC_L_RAMP_EN_SHIFT, DA7219_SWITCH_EN_MAX, > + DA7219_NO_INVERT), Capture Digital rather than ADC. > + SOC_SINGLE_TLV("ALC Max Attenuation", DA7219_ALC_GAIN_LIMITS, > + DA7219_ALC_ATTEN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX, > + DA7219_NO_INVERT, da7219_alc_gain_tlv), > + SOC_SINGLE_TLV("ALC Max Gain", DA7219_ALC_GAIN_LIMITS, > + DA7219_ALC_GAIN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX, > + DA7219_NO_INVERT, da7219_alc_gain_tlv), > + SOC_SINGLE_RANGE_TLV("ALC Min Analog Gain", DA7219_ALC_ANA_GAIN_LIMITS, > + DA7219_ALC_ANA_GAIN_MIN_SHIFT, > + DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX, > + DA7219_NO_INVERT, da7219_alc_ana_gain_tlv), > + SOC_SINGLE_RANGE_TLV("ALC Max Analog Gain", DA7219_ALC_ANA_GAIN_LIMITS, > + DA7219_ALC_ANA_GAIN_MAX_SHIFT, > + DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX, > + DA7219_NO_INVERT, da7219_alc_ana_gain_tlv), Volume not Gain. > + SOC_SINGLE("ToneGen DTMF Key", DA7219_TONE_GEN_CFG1, > + DA7219_DTMF_REG_SHIFT, DA7219_DTMF_REG_MAX, > + DA7219_NO_INVERT), Should this be an enumeration with the DTMF digits in it (# and * aren't numbers)? > + /* DACs */ > + SOC_DOUBLE_R_TLV("DAC Volume", DA7219_DAC_L_GAIN, DA7219_DAC_R_GAIN, > + DA7219_DAC_L_DIGITAL_GAIN_SHIFT, > + DA7219_DAC_DIGITAL_GAIN_MAX, DA7219_NO_INVERT, > + da7219_dac_dig_gain_tlv), > + SOC_DOUBLE_R("DAC Switch", DA7219_DAC_L_CTRL, DA7219_DAC_R_CTRL, > + DA7219_DAC_L_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, > + DA7219_INVERT), > + SOC_DOUBLE_R("DAC Gain Ramp Switch", DA7219_DAC_L_CTRL, > + DA7219_DAC_R_CTRL, DA7219_DAC_L_RAMP_EN_SHIFT, > + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), All the DAC controls should probably be Playback Digital instead. > + > +static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai, > + int clk_id, unsigned int freq, int dir) > +{ > + struct snd_soc_codec *codec = codec_dai->codec; > + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); > + > + if (da7219->mclk_rate == freq) > + return 0; Given that we also have source selection is this safe - should we also be checking the source? > + if (da7219->mclk) { > + freq = clk_round_rate(da7219->mclk, freq); > + clk_set_rate(da7219->mclk, freq); > + } Missing error checking. > + /* Internal LDO */ > + if (da7219->use_int_ldo) > + snd_soc_update_bits(codec, DA7219_LDO_CTRL, > + DA7219_LDO_EN_MASK, > + DA7219_LDO_EN_MASK); If there is an option to use an external supply I would expect to see the regulator API used to discover the external LDO (and ideally also to configure the integrated LDO). If the driver works outside the frameworks then it is likely this will lead to integration issues later on. > + /* Internal LDO */ > + da7219->use_int_ldo = pdata->use_internal_ldo; This is likely to lead to surprises for example... > + /* Check if MCLK provided, if not the clock is NULL */ > + da7219->mclk = devm_clk_get(codec->dev, "mclk"); > + if (IS_ERR(da7219->mclk)) { > + if (PTR_ERR(da7219->mclk) == -EPROBE_DEFER) > + return -EPROBE_DEFER; > + da7219->mclk = NULL; > + } This should specifically look for the error code that corresponds to a clock not being mapped and only continue silently in that case rather than silently accepting all failures to obtain a clock - if we silently accept all failures that will mean that actual errors will be ignored. > + /* > + * There are multiple control bits for the input mixer. > + * The following can be enabled now as it's not power related. > + */ > + snd_soc_update_bits(codec, DA7219_MIXIN_L_CTRL, > + DA7219_MIXIN_L_MIX_EN_MASK, > + DA7219_MIXIN_L_MIX_EN_MASK); So the chip designers just put these in for randomness? Fun. It'd be more idiomatic to do something like making these supply widgets so they're controlled via DAPM even if they don't matter much. > + else if (device_may_wakeup(codec->dev)) > + disable_irq_wake(da7219->aad->irq); You can use dev_pm_set_wake_irq() and skip having to manage this stuff explicitly in the driver.

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[alsa-devel] unable to create IPC shm instance; SND_PCM_DIRECT_MAGIC
by Mike Bjorge 22 Sep '15

22 Sep '15

I was wondering if anyone could explain the point of SND_PCM_DIRECT_MAGIC and dmix->shmptr->magic? Specifically, what it's doing in pcm_direct.c from this commit? http://git.alsa-project.org/?p=alsa-lib.git;a=blobdiff;f=src/pcm/pcm_direct… I am trying to use pcm_dmix.c, but this is occasionally failing first_instance = ret = snd_pcm_direct_shm_create_or_connect(dmix); if (ret < 0) { SNDERR("unable to create IPC shm instance"); goto _err; } because dmix->shmptr->magic is not correctly set to SND_PCM_DIRECT_MAGIC. Thanks for any information! -Mike

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[alsa-devel] [PATCH v2 0/9] Add DSP topology management for SKL
by Vinod Koul 21 Sep '15

21 Sep '15

The SKL driver does not code DSP topology in driver. With this series the ASoC topology framework is used to parse the topology information to create widgets, controls and route map for the FW topology. It adds routines for SKL DSP module configuration, compute resources for modules, initialize and bind the pipe modules. It uses the SKL IPC library added earlier to send IPC for initialize the module, bind/unbind modules. Last patch in this series removes the unused dais. changes in v2: - add more explanation in changelog and code - add explanation of MCPS - fix whitespace issues - remove skl widget check on event handler as it is redundant - add switch for depth - add else for direction based code - remove cast for NHLT blob query - make debug prints to error - remove dump info - clean some debug prints Jeeja KP (9): ASoC: Intel: Skylake: Add pipe and modules handlers ASoC: Intel: Skylake: Add module configuration helpers ASoC: Intel: Skylake: add DSP platform widget event handlers ASoC: Intel: Skylake: Add FE and BE hw_params handling ASoC: Intel: Skylake: Add topology core init and handlers ASoC: Intel: Skylake: Initialize and load DSP controls ASoC: Intel: Skylake: Add DSP support and enable it ASoC: Intel: Skylake: Initialize NHLT table ASoC: Intel: Skylake: Remove unused CPU dai's sound/soc/intel/skylake/Makefile | 3 +- sound/soc/intel/skylake/skl-pcm.c | 171 ++-- sound/soc/intel/skylake/skl-topology.c | 1262 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-topology.h | 21 + sound/soc/intel/skylake/skl-tplg-interface.h | 78 ++ sound/soc/intel/skylake/skl.c | 29 +- sound/soc/intel/skylake/skl.h | 11 + 7 files changed, 1507 insertions(+), 68 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-topology.c -- 1.9.1

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[alsa-devel] Intel SST on a Bay Trail tablet
by Michele Curti 21 Sep '15

21 Sep '15

> On Thu, 25 Jun 2015 22:17:17 +0530 Vinod Koul wrote: > > > On Thu, Jun 25, 2015 at 12:21:24PM +0200, Antonio Ospite wrote: > > > On Thu, 25 Jun 2015 11:20:46 +0530 > > > Vinod Koul <vinod.koul at intel.com> wrote: > > > > > > > On Wed, Jun 24, 2015 at 03:46:13PM +0530, Vinod Koul wrote: > > > > > On Tue, Apr 14, 2015 at 05:06:27PM +0300, Jarkko Nikula wrote: > > > [...] > > > > > > > > > > > > Which makes me thinking how does those strings describe the SSP port > > > > > > setup? E.g. do they reflect what port is actually used or could it > > > > > > be possible that those are just driver strings but firmware could > > > > > > have been tuned for SSP0? If I looked at earlier right, Teclast has > > > > > > the low pin-count Baytrail without SSP2 but I'm not sure about that. > > > > > The FW doesnt provide way to changes ports from driver in this, so this > > > > > doesnt mean much. If we are sure it is not ssp2, I can provide ssp1 fw for > > > > > test > > > > I ahve pushed latest BYT version we have as well as added binary for second > > > > SSP port. Please give it a try: > > > > > > > > https://git.kernel.org/cgit/linux/kernel/git/vkoul/firmware.git/commit/?h=b… > > > > > > > > > > Thanks Vinod, but I see these are in the ELF format, and the mainline > > > driver does not support this format, at least the last time I checked it > > > didn't. > > My bad, updated the patch now with right format > > > > https://git.kernel.org/cgit/linux/kernel/git/vkoul/firmware.git/commit/?h=b… > > > > (Adding Michele to CC) > > OK, so Jarkko was right: after a quick test I confirm that with > fw_sst_0f28_ssp0.bin I get sound; this is quite an improvement, > thanks! Damn, completely missed.. Hi Vinod, if it's possible, could you please generate a fw_sst_0f28_ssp1.bin also, so I can test my driver (mine was for the Asus X205TA laptop, where the codec is on SSP1 port)? Thanks, Michele

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[alsa-devel] [PATCH BAT V1 0/7] BAT: Add Basic Audio Tester command line tool
by han.lu＠intel.com 21 Sep '15

21 Sep '15

From: "Lu, Han" <han.lu(a)intel.com> BAT (Basic Audio Tester) is a simple command line utility intended to automate audio driver and sound server QA testing with no human interaction. BAT works by playing an audio stream and capturing the same stream in either a digital or analog loopback. It then compares the captured stream to the original to determine if the testcase passes or fails. The main idea of frequency detecting is: The analysis function reads data from wav file, run fft against the data to get magnitude of frequency vectors, and then calculates the average value and standard deviation of frequency vectors. After that, we define a threshold: threshold = 3 * standard_deviation + average_value Frequencies with amplitude larger than threshold will be recognized as a peak, and the frequency with largest peak value will be recognized as a detected frequency. BAT then compares the detected frequency to target frequency, to decide if the detecting passes or fails. BAT supports 4 working modes: 1. single line playback; 2. single line capture and analysis; 3. playback and capture in loop, and analyze captured data; 4. local analyze without actual playback or capture. BAT will check devices input by user to decide which mode to use. BAT will create threads for playback and record, and will run spectrum analysis for mode 2, 3 or 4. Lu, Han (7): BAT: Add initial functions BAT: Add common definitions and functions BAT: Add playback and record functions BAT: Add signal generator BAT: Add converting functions BAT: Add spectrum analysis functions BAT: Add Makefile and configures Makefile.am | 3 + bat/Makefile.am | 14 ++ bat/alsa.c | 618 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/alsa.h | 20 ++ bat/analyze.c | 314 ++++++++++++++++++++++++++++ bat/analyze.h | 16 ++ bat/bat.c | 608 +++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/common.c | 198 ++++++++++++++++++ bat/common.h | 169 ++++++++++++++++ bat/convert.c | 113 +++++++++++ bat/convert.h | 23 +++ bat/signal.c | 88 ++++++++ configure.ac | 19 +- 13 files changed, 2202 insertions(+), 1 deletion(-) create mode 100644 bat/Makefile.am create mode 100644 bat/alsa.c create mode 100644 bat/alsa.h create mode 100644 bat/analyze.c create mode 100644 bat/analyze.h create mode 100644 bat/bat.c create mode 100644 bat/common.c create mode 100644 bat/common.h create mode 100644 bat/convert.c create mode 100644 bat/convert.h create mode 100644 bat/signal.c -- 1.9.1

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[alsa-devel] [PATCH - 2/2] HDA - add headset-sonicmaster and headset-multi-sonicmaster models
by Raymond Yau 21 Sep '15

21 Sep '15

>From e97383136359c5d3ef93282888540bbb20939a9e Mon Sep 17 00:00:00 2001 From: Raymond Yau <superquad.vortex2(a)gmail.com> Date: Sun, 20 Sep 2015 22:18:56 +0800 Subject: [PATCH - 2/2] HDA - add headset-sonicmaster and headset-multi-sonicmaster models add headset only model for ALC668 with Sonic Master Subwoofer add 625ms delay after headset is plugged for correct detection of headset type for ALC668 add headset multi model with Sonice Master Subwoofer update HD-Audio-Models.txt Signed-off-by: Raymond Yau <superquad.vortex2(a)gmail.com> diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index ec099d4..8baddac 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -55,7 +55,12 @@ ALC66x/67x/892 asus-mode7 ASUS asus-mode8 ASUS inv-dmic Inverted internal mic workaround + +ALC668 +====== dell-headset-multi Headset jack, which can also be used as mic-in + headset-sonicmaster Headset jack and SonicMaster Subwoofer + headset-multi-sonicmaster Headset, HP or Mic and SonicMaster Subwoofer ALC680 ====== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a75b561..91de00c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4029,7 +4029,13 @@ static void alc_update_headset_mode(struct hda_codec *codec) PIN_VREFHIZ); } spec->current_headset_mode = new_headset_mode; - + if (codec->core.vendor_id == 0x10ec0668) { + if (spec->gen.hs_mic_use_hp_sense && spec->gen.auto_mic) { + if (new_headset_mode == ALC_HEADSET_MODE_HEADPHONE) + msleep(625); + snd_hda_gen_mic_autoswitch(codec, NULL); + }; + } snd_hda_gen_update_outputs(codec); } @@ -6071,6 +6077,42 @@ static void alc662_fixup_led_gpio1(struct hda_codec *codec, } } +static void alc668_fixup_asus_headset_only(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + const struct hda_pintbl hs_mic_pin[] = { + { 0x1b, 0x03a1113c }, /* headset mic */ + { } + }; + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, hs_mic_pin); + spec->gen.hs_mic_use_hp_sense = 1; + alc_fixup_disable_aamix(codec, fix, action); + break; + } +} + +static void alc668_fixup_sonicmaster(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + const struct hda_pintbl bass_pin[] = { + { 0x1a, 0x04110011 }, /* external subwoofer */ + { } + }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, bass_pin); + break; + case HDA_FIXUP_ACT_BUILD: + alc_fixup_bass_chmap(codec, fix, action); + break; + } +} + static struct coef_fw alc668_coefs[] = { WRITE_COEF(0x01, 0xbebe), WRITE_COEF(0x02, 0xaaaa), WRITE_COEF(0x03, 0x0), WRITE_COEF(0x04, 0x0180), WRITE_COEF(0x06, 0x0), WRITE_COEF(0x07, 0x0f80), @@ -6130,6 +6172,9 @@ enum { ALC668_FIXUP_AUTO_MUTE, ALC668_FIXUP_DELL_DISABLE_AAMIX, ALC668_FIXUP_DELL_XPS13, + ALC668_FIXUP_HEADSET_ONLY, + ALC668_FIXUP_HEADSET_SONICMASTER, + ALC668_FIXUP_HEADSET_MULTI_SONICMASTER, }; static const struct hda_fixup alc662_fixups[] = { @@ -6370,6 +6415,24 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_bass_chmap, }, + [ALC668_FIXUP_HEADSET_ONLY] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc668_fixup_asus_headset_only, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_MODE, + }, + [ALC668_FIXUP_HEADSET_SONICMASTER] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc668_fixup_sonicmaster, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_ONLY, + }, + [ALC668_FIXUP_HEADSET_MULTI_SONICMASTER] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc668_fixup_sonicmaster, + .chained = true, + .chain_id = ALC668_FIXUP_DELL_DISABLE_AAMIX, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -6476,6 +6539,9 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {.id = ALC662_FIXUP_ASUS_MODE8, .name = "asus-mode8"}, {.id = ALC662_FIXUP_INV_DMIC, .name = "inv-dmic"}, {.id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, + {.id = ALC668_FIXUP_HEADSET_SONICMASTER, .name = "headset-sonicmaster"}, + {.id = ALC668_FIXUP_HEADSET_MULTI_SONICMASTER, + .name = "headset-multi-sonicmaster"}, {} }; -- 2.5.0

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[alsa-devel] [PATCH v2 1/2] ASoC: rt5645: Add dmi for Broadwell
by Bard Liao 20 Sep '15

20 Sep '15

Add DMI data for Buddy project. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 28 +++++++++++++++++++++++++++- 1 file changed, 27 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 66e9ce6..2fc437f 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3210,6 +3210,31 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { { } }; +static struct rt5645_platform_data buddy_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, + .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, + .jd_mode = 3, +}; + +static int buddy_quirk_cb(const struct dmi_system_id *id) +{ + rt5645_pdata = &buddy_platform_data; + + return 1; +} + +static struct dmi_system_id dmi_platform_intel_broadwell[] __initdata = { + { + .ident = "Chrome Buddy", + .callback = buddy_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Buddy"), + }, + }, + { } +}; + + static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) { rt5645->pdata.in2_diff = device_property_read_bool(dev, @@ -3242,7 +3267,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (pdata) rt5645->pdata = *pdata; - else if (dmi_check_system(dmi_platform_intel_braswell)) + else if (dmi_check_system(dmi_platform_intel_braswell) || + dmi_check_system(dmi_platform_intel_broadwel) rt5645->pdata = *rt5645_pdata; else rt5645_parse_dt(rt5645, &i2c->dev); -- 1.8.1.1.439.g50a6b54

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