December 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ASoC: fsl_sai: Has anybody attempted to run the i.MX6UL in TDM mode with channels > 2?
by Caleb Crome 19 Jan '17

19 Jan '17

Hi all, Now that the MX6 fsl_ssi seems to be getting stable for TDM with channels > 2 (thanks all!), the MX6 UL and MX7 have only the SAI port. Does anybody have experience with running the SAI with more than 2 channels? Any thoughts or comments would be welcome! Thanks, -Caleb

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[alsa-devel] [PATCH 00/12] ASoC: Intel: Skylake: Driver updates
by jeeja.kp＠intel.com 19 Jan '17

19 Jan '17

From: Jeeja KP <jeeja.kp(a)intel.com> This patch series includes driver updates for the handling scenarios during System S0/S3 transitions when audio stream is stopped during system. o move DMA configuration in DSP Gateway Module widget event handler to ensure DMA is configured after resume. o Enable SSP MLCK using Supply widget. This will ensure the MCLK is enabled when the widget is power on. o Check DMA decouple register state before coupling/decoupling the DMA. o In case of HDMI pass-through pipe, reset only the FE Pipe when stream in in XRUN state. o Resume the stream from HW renderer position. o HDMI Codec pin and converter state need to be restored when stream is resumed. Jeeja KP (12): ASoC: Intel: Skylake: Add helper function to setup host/link dma ASoC: Intel: Skylake: Configure DMA in PRE_PMD handler of Mixer ASoC: Intel: Skylake: Removed unused skl_get_format() ALSA: hda: check stream decoupled register state ASoC: Intel: Skylake: Add set_tristate DAI ops to enable SSP MCLK ASoC: Intel: Skylake: Remove unused SSP BE prepare DAI ops ASoC: Intel: Skylake: Add supply widget as non DSP widget ASoC: Intel: Skylake: Add supply widget in skl_nau_max machine ASoC: Intel: Skylake: Add supply widget in bxt_da_max machine ASoC: Intel: Skylake: Don't reset pass-through pipe in BE prepare ASoC: Intel: Skylake: set the resume point to LPIB ASoC: hdac_hdmi: Enable pin and converter in prepare sound/hda/ext/hdac_ext_stream.c | 15 +- sound/soc/codecs/hdac_hdmi.c | 28 ++-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 34 ++++ sound/soc/intel/boards/skl_nau88l25_max98357a.c | 34 ++++ sound/soc/intel/skylake/skl-pcm.c | 210 +++++++++++++----------- sound/soc/intel/skylake/skl-topology.c | 27 ++- sound/soc/intel/skylake/skl-topology.h | 6 + 7 files changed, 240 insertions(+), 114 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH v2 0/3] ucm: Add support for component devices
by mengdong.lin＠linux.intel.com 18 Jan '17

18 Jan '17

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Sound cards are defined by machines. And off-soc codecs and DSPs embedded in DSP can be taken as the components and resued by different machines/ sound cards. This series allows codec and SOC vendors to install UCM configuration files for a specific codec or DSP, and sound card verb files can include these conf files of components. Component devices will not be exposed to applications for backward compatibility. So audio servers like PulseAudio and CRAS still only see the machine devices. History: v2: Use an array to define component directories. Document syntax to enable/disable a component device. Hide cdev defined by the parent device in ucm manager for executing a component sequence. Mengdong Lin (3): ucm: Skip component directories when scanning sound card configuration files ucm: Parse sequence of component devices ucm: Execute sequence of component devices src/ucm/main.c | 88 +++++++++++++++++++++++++++++- src/ucm/parser.c | 154 +++++++++++++++++++++++++++++++++++++++++++++++++++- src/ucm/ucm_local.h | 19 +++++++ src/ucm/utils.c | 1 + 4 files changed, 259 insertions(+), 3 deletions(-) -- 2.7.4

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[alsa-devel] [RFC PATCH 0/2] ASoC: Use DMI name in sound card long name
by mengdong.lin＠linux.intel.com 17 Jan '17

17 Jan '17

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series is based on Liam's idea to use DMI name as the sound card short name in the audio workshop. After testing on serveral machines, we changed the design a little: - Not change the card name (short name), keep it simple. So the card ID won't change as well. - Use short name and DMI name together to make a readable and unqique long name. Machine drivers can use a new API to get this feature. We hope to get aligned on the kernel side at first, then we'll update UCM code in alsa-lib: when the sound server ask UCM to open a sound card, UCM will try to find the best configuration file that matches the card long name, and only if not avaiable, fallback to the default configuration file that matches the card name (short name). Liam Girdwood (1): ASoC: core: Add API to use DMI name in sound card long name Mengdong Lin (1): ASoC: Intel: Use DMI name for sound card long name in Broadwell machine driver include/sound/soc.h | 5 ++ sound/soc/intel/boards/broadwell.c | 2 + sound/soc/soc-core.c | 134 +++++++++++++++++++++++++++++++++++++ 3 files changed, 141 insertions(+) -- 2.5.0

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Re: [alsa-devel] [PATCH 2/2] ASoC: cx2072x Add driver for CX2072X CODEC
by Mark Brown 11 Jan '17

11 Jan '17

On Sat, Dec 17, 2016 at 03:52:33PM +0800, simon.ho.cnxt(a)gmail.com wrote: > +/*FIXME: need to move the default settings to device tree*/ No, we'd expect things to be configured from userspace via a binary control. > +static unsigned char cx2072x_eq_coeff_array[MAX_EQ_BAND][MAC_EQ_COEFF] = { > + {0x77, 0x26, 0x13, 0xb3, 0x76, 0x26, 0x0a, 0x3d, 0xd4, 0xe2, 0x04}, > + {0x97, 0x3e, 0xb3, 0x86, 0xc2, 0x3b, 0x4d, 0x79, 0xa7, 0xc5, 0x03}, > + {0x0f, 0x39, 0x76, 0xa3, 0x1b, 0x2b, 0x89, 0x5c, 0xd7, 0xdb, 0x03}, > + {0x21, 0x46, 0xfe, 0xa6, 0xec, 0x24, 0x01, 0x59, 0xf4, 0xd4, 0x03}, > + {0xe9, 0x78, 0x9c, 0xb0, 0x8a, 0x56, 0x64, 0x4f, 0x8d, 0xb0, 0x02}, > + {0x60, 0x6e, 0x57, 0xee, 0xec, 0x18, 0xa8, 0x11, 0xb5, 0xf8, 0x02}, > + {0x5a, 0x14, 0x68, 0xe9, 0x1d, 0x06, 0xb9, 0x5f, 0x68, 0xdc, 0x03}, > +}; > + > +static unsigned char cx2072x_drc_array[MAX_DRC_REGS] = { > + 0x65, 0x55, 0x3C, 0x01, 0x05, 0x39, 0x76, 0x1A, 0x00 > +}; Just use the chip defaults, this avoids any confusion or debate about the values in the kernel. > +#define get_cx2072x_priv(_codec_) \ > + ((struct cx2072x_priv *) snd_soc_codec_get_drvdata(_codec_)) There is no need to cast void pointers, this will at most hide actual errors, and there's no real need for the macro at all really. > + { 36864000, 7 },/* Don't use div 7*/ Does something enforce that? Also coding style, missing space. > +static const struct reg_default cx2072x_reg_defaults[] = { > + { 0x0414, 0x00000003 }, /*2072X_AFG_POWER_STATE */ We have defines for the register names which are helpfully commented here, why not just actually use the defines? > + if (reg == CX2072X_UM_INTERRUPT_CRTL_E) { > + /* Update the MSB byte only */ > + reg += 3; > + size = 1; > + value >>= 24; > + } Use a switch here in case you find any more magic registers. > + ret = i2c_master_send(client, buf, size + 2); > + if (ret == size + 2) { > + ret = 0; > + } else if (ret < 0) { > + dev_err(dev, > + "I2C write address failed\n"); Please print the error code, it makes life easier for people looking at logs with errors. > +static unsigned int get_div_from_mclk(unsigned int mclk) > +{ > + unsigned int div = 8; > + int i = 0; > + > + for (i = 0; i < ARRAY_SIZE(MCLK_PRE_DIV); i++) { > + if (mclk <= MCLK_PRE_DIV[i].mclk) { > + div = MCLK_PRE_DIV[i].div; > + break; > + } > + } > + return div; > +} Why is this function in the middle of all the register I/O functions? > +static int cx2072x_config_headset_det(struct cx2072x_priv *cx2072x) > +{ > + const int interrupt_gpio_pin = 1; > + > + dev_dbg(cx2072x->dev, > + "Configure interrupt pin: %d\n", interrupt_gpio_pin); > + /*No-sticky input type*/ Coding style. There's *lots* of really obvious and bad coding style problems throughout the driver, this is really not good - as well as making things harder to read poor coding style is normally a good indicator that there are other problems. > + /* support both nokia and apple headset set. Monitor time = 275 ms*/ > + regmap_write(cx2072x->regmap, CX2072X_DIGITAL_TEST15, 0x73); > + > + /* Disable TIP detection*/ > + regmap_write(cx2072x->regmap, CX2072X_ANALOG_TEST12, 0x300); > + > + /* Switch MusicD3Live pin to GPIO */ > + regmap_write(cx2072x->regmap, CX2072X_DIGITAL_TEST1, 0); These look awfully like they should be configured by the system integrator via DT. > + regmap_write(cx2072x->regmap, CX2072X_ANALOG_TEST8, > + (unsigned char)int_div & 0xffff); This cast is worrying, why is it needed? > + > + /* clock inversion */ > + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { > + case SND_SOC_DAIFMT_NB_NF: > + is_frame_inv = is_i2s ? 1 : 0; > + is_bclk_inv = is_i2s ? 1 : 0; Please don't abuse the ternery operator like this, write normal logic statements with straightforward assignments. Readers will thank you. > + regmap_write(cx2072x->regmap, CX2072X_I2SPCM_CONTROL1, reg1.ulVal); Please don't use Hungarian notation. > + /*Enables bclk and EAPD pin*/ > + if (cx2072x->rev_id == CX2072X_REV_A2) > + regmap_update_bits(cx2072x->regmap, CX2072X_DIGITAL_BIOS_TEST2, > + 0x84, 0xFF); > + else > + regmap_write(cx2072x->regmap, CX2072X_DIGITAL_BIOS_TEST2, > + regDBT2.ulVal); Again, switch statement. > +static int portg_power_ev(struct snd_soc_dapm_widget *w, > + struct snd_kcontrol *kcontrol, int event) > +{ > + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); > + > + switch (event) { > + case SND_SOC_DAPM_POST_PMU: > + cx2072x_update_dsp(codec); > + break; > + default: > + break; > + } > + return 0; > +} > +static int cx2072x_plbk_dsp_info(struct snd_kcontrol *kcontrol, Blank lines between functions. > + mutex_lock(&cx2072x->eq_coeff_lock); > + memcpy(cache, param, CX2072X_PLBK_EQ_COEF_LEN); > + cx2072x->plbk_dsp_changed = true; Shouldn't this involve a memcmp() to check if the new value is different? > +static const struct snd_kcontrol_new cx2072x_snd_controls[] = { > + > + SOC_DOUBLE_R_TLV("PortD Boost", CX2072X_PORTD_GAIN_LEFT, > + CX2072X_PORTD_GAIN_RIGHT, 0, 3, 0, boost_tlv), All volume controls should end in Volume so userspace knows how to handle them. Also please use normal kernel indentation for the continuation lines. > + SOC_DOUBLE_R("DAC1 Mute", CX2072X_DAC1_AMP_GAIN_LEFT, > + CX2072X_DAC1_AMP_GAIN_RIGHT, 7, 1, 0), All boolean controls should end in Switch so userspace knows how to handle them. > + CX2072X_PLBK_EQ_COEF("DACL EQ 0", 0, 0), > + CX2072X_PLBK_EQ_COEF("DACL EQ 1", 0, 1), > + CX2072X_PLBK_EQ_COEF("DACL EQ 2", 0, 2), > + CX2072X_PLBK_EQ_COEF("DACL EQ 3", 0, 3), > + CX2072X_PLBK_EQ_COEF("DACL EQ 4", 0, 4), > + CX2072X_PLBK_EQ_COEF("DACL EQ 5", 0, 5), > + CX2072X_PLBK_EQ_COEF("DACL EQ 6", 0, 6), > + CX2072X_PLBK_EQ_COEF("DACR EQ 0", 1, 0), > + CX2072X_PLBK_EQ_COEF("DACR EQ 1", 1, 1), > + CX2072X_PLBK_EQ_COEF("DACR EQ 2", 1, 2), > + CX2072X_PLBK_EQ_COEF("DACR EQ 3", 1, 3), > + CX2072X_PLBK_EQ_COEF("DACR EQ 4", 1, 4), > + CX2072X_PLBK_EQ_COEF("DACR EQ 5", 1, 5), > + CX2072X_PLBK_EQ_COEF("DACR EQ 6", 1, 6), This looks like it should be two binary controls, one per EQ. > +int cx2072x_hs_jack_report(struct snd_soc_codec *codec) > +{ > +} > +EXPORT_SYMBOL_GPL(cx2072x_hs_jack_report); We have a standard jack detection interface in the kernel, the driver should be using it rather than writing its own. > +static int cx2072x_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, > + unsigned int rx_mask, int slots, int slot_width) > +{ > + struct snd_soc_codec *codec = dai->codec; > + struct cx2072x_priv *cx2072x = snd_soc_codec_get_drvdata(codec); > + > + if (slots == 0) > + goto out; > + > + > + switch (rx_mask) { > + case 1 ... 0xf: > + default: > + return -EINVAL; > + } So all possible values are invalid? That doesn't seemw right... > + switch (clk_id) { > + case CX2072X_MCLK_EXTERNAL_PLL: > + if (cx2072x->mclk && clk_set_rate(cx2072x->mclk, freq)) > + return -EINVAL; Please write sensible error handling, log what went wrong if you get an error and pass back error codes you get from other APIs. This will make it a lot easier for users to figure out what's happened. > +#define CX2072X_DAPM_SUPPLY_S(wname, wsubseq, wreg, wshift, wmask, won_val, \ > + woff_val, wevent, wflags) \ > + {.id = snd_soc_dapm_supply, .name = wname, .kcontrol_news = NULL, \ > + .num_kcontrols = 0, .reg = wreg, .shift = wshift, .mask = wmask, \ > + .on_val = won_val, .off_val = woff_val, \ > + .subseq = wsubseq, .event = wevent, .event_flags = wflags} There appears to be nothing device specific in here, why is this not being added as a generic supply type? > +static void cx2072x_sw_reset(struct cx2072x_priv *cx2072x) > +{ > + > + regmap_write(cx2072x->regmap, CX2072X_AFG_FUNCTION_RESET, 0x01); Coding style, random blank line hre. > + /* reduce the jack monitor time*/ > + regmap_update_bits(cx2072x->regmap, CX2072X_DIGITAL_TEST15, > + 0x00, 0x06); Coding style, please intent contiuation lines... > + cx2072x_config_headset_det(cx2072x); > + /* configre PortC as input device */ > + regmap_update_bits(cx2072x->regmap, CX2072X_PORTC_PIN_CTRL, > + 0x20, 0x20); ...this isn't even consistent with the rest of the same function :( > + switch (level) { > + case SND_SOC_BIAS_ON: > + /* Enable Headset Mic Bias */ > + if (cx2072x->is_biason == 0) > + cx2072x->is_biason = 1; > + break; This looks worrying but since nothing else uses that driver data I'm guessing it doesn't actually do anything. > + case SND_SOC_BIAS_PREPARE: > + if (old_level == SND_SOC_BIAS_STANDBY) { > + if (cx2072x->mclk) { > + dev_dbg(cx2072x->dev, "Turn on MCLK\n"); > + ret = clk_prepare_enable(cx2072x->mclk); > + if (ret) > + return ret; > + } > + } > + break; > + case SND_SOC_BIAS_STANDBY: > + > + if (old_level == SND_SOC_BIAS_OFF) { > + if (cx2072x->mclk) { > + dev_dbg(cx2072x->dev, "Turn on MCLK\n"); > + ret = clk_prepare_enable(cx2072x->mclk); > + if (ret) > + return ret; > + } So we enable MCLK both when coming out of _OFF and when coming out of _STANDBY? Why (and note that we're missing some disables here... > + regmap_read(cx2072x->regmap, CX2072X_VENDOR_ID, &ven_id); > + regmap_read(cx2072x->regmap, CX2072X_REVISION_ID, &cx2072x->rev_id); > + dev_dbg(codec->dev, "codec version: %08x,%08x\n", > + ven_id, cx2072x->rev_id); Log this at info level. > +#ifdef CONFIG_PM > +static int cx2072x_runtime_suspend(struct device *dev) > +{ > + struct cx2072x_priv *cx2072x = dev_get_drvdata(dev); > + > + cx2072x_set_bias_level(cx2072x->codec, SND_SOC_BIAS_OFF); Let the framework take care of managing bias level for you. > +static bool cx2072x_readable_register(struct device *dev, unsigned int reg) > +{ > + switch (reg) { Why is this nowhere near the rest of the regmap code? > +static int cx2072x_i2c_probe(struct i2c_client *i2c, > + const struct i2c_device_id *id) > +{ > + int ret = -1; > + struct cx2072x_priv *cx2072x; > + > + cx2072x = (struct cx2072x_priv *)devm_kzalloc( > + &i2c->dev, sizeof(struct cx2072x_priv), GFP_KERNEL); No need to cast away from void. > + if (cx2072x == NULL) { > + dev_err(&i2c->dev, "Out of memory!\n"); kzalloc() will already be very noisy if this happens. > +static const struct i2c_device_id cx2072x_i2c_id[] = { > + { "cx20721", 0 }, > + { "cx20723", 0 }, > + { "14F10720", 0 }, > + {} > +}; > +MODULE_DEVICE_TABLE(i2c, cx2072x_i2c_id); One of those looks like an ACPI ID not an I2C ID. > + > +static const struct of_device_id cx2072x_of_match[] = { > + { .compatible = "cnxt,cx20721", }, > + { .compatible = "cnxt,cx20723", }, > + { .compatible = "cnxt,cx7601", }, > + {} Why is cx7601 not an I2C ID? > +#ifdef CONFIG_ACPI > + .acpi_match_table = ACPI_PTR(cx2072x_acpi_match), > +#endif No need for the defines here.

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[alsa-devel] [PATCH v4 0/3] support DP MST audio
by libin.yang＠intel.com 11 Jan '17

11 Jan '17

From: Libin Yang <libin.yang(a)intel.com> This patchset starts to support DP MST audio. This patchset is based on drm-tip on git://anongit.freedesktop.org/drm-tip change log: v4: refine some comments in 0001 and 0002 patches. refine the 0003 patch for the new ALSA-Configuration document Libin Yang (3): ALSA: hda - add DP mst verb support ALSA: hda - add DP MST audio support ALSA: Documentation about HDA DP MST pin init and connection Documentation/sound/hd-audio/dp-mst.rst | 17 +++ sound/pci/hda/hda_codec.c | 76 +++++++++- sound/pci/hda/hda_codec.h | 3 + sound/pci/hda/patch_hdmi.c | 245 +++++++++++++++++++++++++------- 4 files changed, 285 insertions(+), 56 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH 0/6] ASoC: platform cleanup
by Kuninori Morimoto 11 Jan '17

11 Jan '17

Hi Mark, Lars-Peter These removes un-used platform feature, and move last platform feature to component. If these are OK, all platform feature goes to component. Last platform piece is platform_list on soc-core.c Kuninori Morimoto (6): ASoC: remove .delay from snd_soc_platform_driver ASoC: remove .bespoke_trigger from snd_soc_platform_driver ASoC: remove snd_soc_platform_trigger() ASoC: add for_each_component{_safe} ASoC: add Component level struct snd_compr_ops ASoC: add Component level struct snd_pcm_ops include/sound/soc.h | 23 +++-- sound/soc/soc-compress.c | 228 ++++++++++++++++++++++++++++--------------- sound/soc/soc-core.c | 14 +-- sound/soc/soc-pcm.c | 244 +++++++++++++++++++++++++++++++++-------------- 4 files changed, 339 insertions(+), 170 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/7] Move dell-led to drivers/platform/x86
by Michał Kępień 10 Jan '17

10 Jan '17

This patch series moves the dell-led driver from the LED subsystem to the x86 platform driver subsystem. I decided to also CC the sound subsystem contacts for the whole series as sound/pci/hda/dell_wmi_helper.c is also affected. The original motivation behind this effort was to move all code using the dell-smbios module to the x86 platform driver subsystem. While I was investigating the possibilites to do that, it quickly emerged that dell-led can and in fact should be moved to the x86 platform driver subsystem in its entirety. dell-led consists of two major parts: - the part exposing a microphone mute LED interface, introduced in db6d8cc ("dell-led: add mic mute led interface"); this interface is used by sound/pci/hda/dell_wmi_helper.c; while the original implementation used a WMI interface, it was changed to use dell-smbios in cf0d7ea ("dell-led: use dell_smbios_find_token() for finding mic DMI tokens") and 0c41a08 ("dell-led: use dell_smbios_send_request() for performing SMBIOS calls"), - the part handling an activity LED present in Dell Latitude 2100 netbooks, introduced in 72dcd8d ("leds: Add Dell Business Class Netbook LED driver"); it binds to a specific WMI GUID and then registers a LED device which is controlled using WMI (i.e. it is basically a WMI driver). Patches 1-4 clean up the microphone mute LED interface to minimize the amount of code moved around. Patch 5 moves the microphone mute LED interface to drivers/platform/x86/dell-laptop.c, effectively causing sound/pci/hda/dell_wmi_helper.c to depend on CONFIG_DELL_LAPTOP instead of CONFIG_LEDS_DELL_NETBOOKS. Patch 6 reverts dell-led to the state it was in after its initial commit 72dcd8d ("leds: Add Dell Business Class Netbook LED driver") by removing all remnants of the microphone mute LED handling code. Patch 7 moves all that is left of dell-led (i.e. the activity LED part, as originally implemented), to a new module which is placed in drivers/platform/x86/dell-wmi-led.c. This patch series is based on linux-leds/for-4.11 as the LED subsystem is affected by all patches except patch 3. If anyone reading this has access to a Dell device which has an activity LED and/or a microphone mute LED currently supported by dell-led, I would love to hear from you as I do not have the hardware needed to practically test this patch series. drivers/leds/Kconfig | 9 --- drivers/leds/Makefile | 1 - drivers/platform/x86/Kconfig | 8 +++ drivers/platform/x86/Makefile | 1 + drivers/platform/x86/dell-laptop.c | 28 ++++++++ .../dell-led.c => platform/x86/dell-wmi-led.c} | 75 +++------------------- include/linux/dell-led.h | 6 +- sound/pci/hda/dell_wmi_helper.c | 18 +++--- 8 files changed, 55 insertions(+), 91 deletions(-) rename drivers/{leds/dell-led.c => platform/x86/dell-wmi-led.c} (73%) -- 2.10.2

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[alsa-devel] [PATCH] ASoC: ak4642: Replace mdelay function to msleep
by Kuninori Morimoto 10 Jan '17

10 Jan '17

From: Harunobu Kurokawa <harunobu.kurokawa.dn(a)renesas.com> Replace mdelay to msleep to avoid busy loop on ak4642_lout_event(). Otherwise, sometimes playback doesn't work correctly when pulseaudio was used. Signed-off-by: Harunobu Kurokawa <harunobu.kurokawa.dn(a)renesas.com> --- sound/soc/codecs/ak4642.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 40500cd..0205ae1 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -186,7 +186,7 @@ static int ak4642_lout_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: /* after widget power up */ /* Power save mode OFF */ - mdelay(popup_wait); + msleep(popup_wait); snd_soc_update_bits(codec, SG_SL2, LOPS, 0); break; case SND_SOC_DAPM_PRE_PMD: /* before widget power down */ -- 2.9.2

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[alsa-devel] [PATCH 1/2] ASoC: rt5645: set sel_i2s_pre_div1 to 2
by Bard Liao 09 Jan '17

09 Jan '17

The i2s clock pre-divider 1 is used for both i2s1 and sysclk. The i2s1 is usually used for the main i2s and the pre-divider will be set in hw_params function. However, if i2s2 is used, the pre-divider is not set in the hw_params function and the default value of i2s clock pre-divider 1 is too high for sysclk and DMIC usage. Fix by overriding default divider value to 2. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=95681 Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 10c2a56..1ac96ef 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3833,6 +3833,9 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } } + regmap_update_bits(rt5645->regmap, RT5645_ADDA_CLK1, + RT5645_I2S_PD1_MASK, RT5645_I2S_PD1_2); + if (rt5645->pdata.jd_invert) { regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 1/2] ASoC: dwc: Add record capability in PIO mode
by Jose Abreu 09 Jan '17

09 Jan '17

Up until now PIO mode offered only playback support. With this patch we add support for record mode. The PCM was refactored so that we could reuse the existing infrastructure without many changes. We have support for 16 and 32 bits of sample size using only 2 channels. Tested in a x86_64 platform and in ARC AXS101 SDP platform. Signed-off-by: Jose Abreu <joabreu(a)synopsys.com> Cc: Carlos Palminha <palminha(a)synopsys.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/dwc/designware_i2s.c | 9 +++- sound/soc/dwc/designware_pcm.c | 97 +++++++++++++++++++++++++++++++++--------- sound/soc/dwc/local.h | 9 +++- 3 files changed, 92 insertions(+), 23 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index bdf8398..9c46e41 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -121,9 +121,14 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) irq_valid = true; } - /* Data available. Record mode not supported in PIO mode */ - if (isr[i] & ISR_RXDA) + /* + * Data available. Retrieve samples from FIFO + * NOTE: Only two channels supported + */ + if ((isr[i] & ISR_RXDA) && (i == 0) && dev->use_pio) { + dw_pcm_pop_rx(dev); irq_valid = true; + } /* Error Handling: TX */ if (isr[i] & ISR_TXFO) { diff --git a/sound/soc/dwc/designware_pcm.c b/sound/soc/dwc/designware_pcm.c index 4a83a22..b063c86 100644 --- a/sound/soc/dwc/designware_pcm.c +++ b/sound/soc/dwc/designware_pcm.c @@ -41,10 +41,33 @@ return tx_ptr; \ } +#define dw_pcm_rx_fn(sample_bits) \ +static unsigned int dw_pcm_rx_##sample_bits(struct dw_i2s_dev *dev, \ + struct snd_pcm_runtime *runtime, unsigned int rx_ptr, \ + bool *period_elapsed) \ +{ \ + u##sample_bits (*p)[2] = (void *)runtime->dma_area; \ + unsigned int period_pos = rx_ptr % runtime->period_size; \ + int i; \ +\ + for (i = 0; i < dev->fifo_th; i++) { \ + p[rx_ptr][0] = ioread32(dev->i2s_base + LRBR_LTHR(0)); \ + p[rx_ptr][1] = ioread32(dev->i2s_base + RRBR_RTHR(0)); \ + period_pos++; \ + if (++rx_ptr >= runtime->buffer_size) \ + rx_ptr = 0; \ + } \ + *period_elapsed = period_pos >= runtime->period_size; \ + return rx_ptr; \ +} + dw_pcm_tx_fn(16); dw_pcm_tx_fn(32); +dw_pcm_rx_fn(16); +dw_pcm_rx_fn(32); #undef dw_pcm_tx_fn +#undef dw_pcm_rx_fn static const struct snd_pcm_hardware dw_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | @@ -68,27 +91,51 @@ .fifo_size = 16, }; -void dw_pcm_push_tx(struct dw_i2s_dev *dev) +static void dw_pcm_transfer(struct dw_i2s_dev *dev, bool push) { - struct snd_pcm_substream *tx_substream; - bool tx_active, period_elapsed; + struct snd_pcm_substream *substream; + bool active, period_elapsed; rcu_read_lock(); - tx_substream = rcu_dereference(dev->tx_substream); - tx_active = tx_substream && snd_pcm_running(tx_substream); - if (tx_active) { - unsigned int tx_ptr = READ_ONCE(dev->tx_ptr); - unsigned int new_tx_ptr = dev->tx_fn(dev, tx_substream->runtime, - tx_ptr, &period_elapsed); - cmpxchg(&dev->tx_ptr, tx_ptr, new_tx_ptr); + if (push) + substream = rcu_dereference(dev->tx_substream); + else + substream = rcu_dereference(dev->rx_substream); + active = substream && snd_pcm_running(substream); + if (active) { + unsigned int ptr; + unsigned int new_ptr; + + if (push) { + ptr = READ_ONCE(dev->tx_ptr); + new_ptr = dev->tx_fn(dev, substream->runtime, ptr, + &period_elapsed); + cmpxchg(&dev->tx_ptr, ptr, new_ptr); + } else { + ptr = READ_ONCE(dev->rx_ptr); + new_ptr = dev->rx_fn(dev, substream->runtime, ptr, + &period_elapsed); + cmpxchg(&dev->rx_ptr, ptr, new_ptr); + } if (period_elapsed) - snd_pcm_period_elapsed(tx_substream); + snd_pcm_period_elapsed(substream); } rcu_read_unlock(); } + +void dw_pcm_push_tx(struct dw_i2s_dev *dev) +{ + dw_pcm_transfer(dev, true); +} EXPORT_SYMBOL_GPL(dw_pcm_push_tx); +void dw_pcm_pop_rx(struct dw_i2s_dev *dev) +{ + dw_pcm_transfer(dev, false); +} +EXPORT_SYMBOL_GPL(dw_pcm_pop_rx); + static int dw_pcm_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -126,20 +173,17 @@ static int dw_pcm_hw_params(struct snd_pcm_substream *substream, switch (params_format(hw_params)) { case SNDRV_PCM_FORMAT_S16_LE: dev->tx_fn = dw_pcm_tx_16; + dev->rx_fn = dw_pcm_rx_16; break; case SNDRV_PCM_FORMAT_S32_LE: dev->tx_fn = dw_pcm_tx_32; + dev->rx_fn = dw_pcm_rx_32; break; default: dev_err(dev->dev, "invalid format\n"); return -EINVAL; } - if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) { - dev_err(dev->dev, "only playback is available\n"); - return -EINVAL; - } - ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (ret < 0) @@ -163,13 +207,21 @@ static int dw_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - WRITE_ONCE(dev->tx_ptr, 0); - rcu_assign_pointer(dev->tx_substream, substream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + WRITE_ONCE(dev->tx_ptr, 0); + rcu_assign_pointer(dev->tx_substream, substream); + } else { + WRITE_ONCE(dev->rx_ptr, 0); + rcu_assign_pointer(dev->rx_substream, substream); + } break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - rcu_assign_pointer(dev->tx_substream, NULL); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rcu_assign_pointer(dev->tx_substream, NULL); + else + rcu_assign_pointer(dev->rx_substream, NULL); break; default: ret = -EINVAL; @@ -183,7 +235,12 @@ static snd_pcm_uframes_t dw_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct dw_i2s_dev *dev = runtime->private_data; - snd_pcm_uframes_t pos = READ_ONCE(dev->tx_ptr); + snd_pcm_uframes_t pos; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + pos = READ_ONCE(dev->tx_ptr); + else + pos = READ_ONCE(dev->rx_ptr); return pos < runtime->buffer_size ? pos : 0; } diff --git a/sound/soc/dwc/local.h b/sound/soc/dwc/local.h index 68afd75..91dc70a 100644 --- a/sound/soc/dwc/local.h +++ b/sound/soc/dwc/local.h @@ -105,20 +105,27 @@ struct dw_i2s_dev { struct i2s_clk_config_data config; int (*i2s_clk_cfg)(struct i2s_clk_config_data *config); - /* data related to PIO transfers (TX) */ + /* data related to PIO transfers */ bool use_pio; struct snd_pcm_substream __rcu *tx_substream; + struct snd_pcm_substream __rcu *rx_substream; unsigned int (*tx_fn)(struct dw_i2s_dev *dev, struct snd_pcm_runtime *runtime, unsigned int tx_ptr, bool *period_elapsed); + unsigned int (*rx_fn)(struct dw_i2s_dev *dev, + struct snd_pcm_runtime *runtime, unsigned int rx_ptr, + bool *period_elapsed); unsigned int tx_ptr; + unsigned int rx_ptr; }; #if IS_ENABLED(CONFIG_SND_DESIGNWARE_PCM) void dw_pcm_push_tx(struct dw_i2s_dev *dev); +void dw_pcm_pop_rx(struct dw_i2s_dev *dev); int dw_pcm_register(struct platform_device *pdev); #else void dw_pcm_push_tx(struct dw_i2s_dev *dev) { } +void dw_pcm_pop_rx(struct dw_i2s_dev *dev) { } int dw_pcm_register(struct platform_device *pdev) { return -EINVAL; -- 1.9.1

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[alsa-devel] [PATCH v3 0/6] ASoC: sun4i-codec: Add support for A31 Codec
by Chen-Yu Tsai 09 Jan '17

09 Jan '17

Hi everyone, This is v3 of my Allwinner A31 audio codec support series. These are the remaining driver patches not yet merged, with 1 new fix, and the dts file patches. The A31's internal codec is similar (in terms of DMA, interface and control layouts) to the one found in the A10/A13/A20 SoCs. However it has more external inputs and outputs, the mixer controls are now stereo (left/right separated), and it also has some audio processing features (not supported yet). Changes since v2: - Added patch """ASoC: sun4i-codec: Add "Right Mixer" to "Line Out Mono Diff." route""" which fixes line out when all the sources have their left channel muted. - Dropped unused codec_pa_pin from A31 hummingbird dts patch. In the past we always provided a pinmux setting for GPIOs used, to avoid accidentally muxing a pin that was already used for GPIO, or vice versa. The pinctrl subsystem now supports exclusive use of a pin for GPIO, which the sunxi platform wants to migrate to. This drops the requirement of the explicit pinmux setting for GPIOs. - Added Maxime's Acked-by for the two existing patches from v2. After these are merged I plan to submit patches to support the audio codec found in Allwinner's A23 and H3 SoCs. Regards ChenYu Chen-Yu Tsai (6): ASoC: sun4i-codec: Add support for optional reset control to quirks ASoC: sun4i-codec: Add support for A31 ADC capture path ASoC: sun4i-codec: Add "Right Mixer" to "Line Out Mono Diff." route ARM: dts: sun6i: Add audio codec device node ARM: dts: sun6i: hummingbird: Enable internal audio codec ARM: dts: sun6i: sina31s: Enable internal audio codec arch/arm/boot/dts/sun6i-a31-hummingbird.dts | 13 ++++ arch/arm/boot/dts/sun6i-a31.dtsi | 13 ++++ arch/arm/boot/dts/sun6i-a31s-sina31s.dts | 8 +++ sound/soc/sunxi/sun4i-codec.c | 94 ++++++++++++++++++++++++++++- 4 files changed, 127 insertions(+), 1 deletion(-) -- 2.10.2

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[alsa-devel] [PATCH] ASoC: nau8825: Add Class G function options
by Ben Zhang 09 Jan '17

09 Jan '17

Add a DAPM_MUX for userspace to control the Class G function: "Switching": Use +-1.8V or +-0.9V supply, based on signal amplitude. "1.8V": Always use +-1.8V. The default is "Switching" which is the existing behavior without this patch, so no effect on current users. On some devices, the switching may cause a clicking noise at low volume. "1.8V" eliminates the noise at the expense of a few additional mW. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/nau8825.c | 24 +++++++++++++++++++++--- 1 file changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index efe3a44..ddd15a0 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1062,6 +1062,10 @@ static const char * const nau8825_dac_src[] = { "DACL", "DACR", }; +static const char * const nau8825_class_g_src[] = { + "Switching", "1.8V" +}; + static SOC_ENUM_SINGLE_DECL( nau8825_dacl_enum, NAU8825_REG_DACL_CTRL, NAU8825_DACL_CH_SEL_SFT, nau8825_dac_src); @@ -1070,12 +1074,17 @@ static SOC_ENUM_SINGLE_DECL( nau8825_dacr_enum, NAU8825_REG_DACR_CTRL, NAU8825_DACR_CH_SEL_SFT, nau8825_dac_src); +static SOC_ENUM_SINGLE_DECL( + nau8825_class_g_enum, SND_SOC_NOPM, 0, nau8825_class_g_src); + static const struct snd_kcontrol_new nau8825_dacl_mux = SOC_DAPM_ENUM("DACL Source", nau8825_dacl_enum); static const struct snd_kcontrol_new nau8825_dacr_mux = SOC_DAPM_ENUM("DACR Source", nau8825_dacr_enum); +static const struct snd_kcontrol_new nau8825_class_g_mux = + SOC_DAPM_ENUM("Class G Source", nau8825_class_g_enum); static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, NAU8825_REG_I2S_PCM_CTRL2, @@ -1154,9 +1163,16 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_PGA_S("HP Boost Driver", 9, NAU8825_REG_BOOST, 9, 1, NULL, 0), - /* Class G operation control*/ - SND_SOC_DAPM_PGA_S("Class G", 10, + /* Class G operation control. "Class G" MUX options: + * "Switching": Use +-1.8V or +-0.9V supply, based on signal amplitude. + * "1.8V": Always use +-1.8V. + * The PGA_S widget is used to ensure the Class G control bit is set at + * the correct time w.r.t the other widgets. + */ + SND_SOC_DAPM_PGA_S("Class G Driver", 10, NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_MUX("Class G", SND_SOC_NOPM, 0, 0, + &nau8825_class_g_mux), SND_SOC_DAPM_OUTPUT("HPOL"), SND_SOC_DAPM_OUTPUT("HPOR"), @@ -1197,7 +1213,9 @@ static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { {"HPOR Pulldown", NULL, "Output DACR"}, {"HP Boost Driver", NULL, "HPOL Pulldown"}, {"HP Boost Driver", NULL, "HPOR Pulldown"}, - {"Class G", NULL, "HP Boost Driver"}, + {"Class G Driver", NULL, "HP Boost Driver"}, + {"Class G", "1.8V", "HP Boost Driver"}, + {"Class G", "Switching", "Class G Driver"}, {"HPOL", NULL, "Class G"}, {"HPOR", NULL, "Class G"}, }; -- 2.8.0.rc3.226.g39d4020

2 3

[alsa-devel] [PATCH] Revert "ASoC: core: replace aux_comp_list to component_dev_list"
by Nikita Yushchenko 09 Jan '17

09 Jan '17

This reverts commit 1a653aa44725668590b36bbe2d7fe4736a69f055 ("ASoC: core: replace aux_comp_list to component_dev_list"). That commit tries to remove card->aux_comp_list, using flagged entries in card->component_dev_list instead. However, components are added to card->component_dev_list in soc_probe_component(), which is called for aux devices by soc_probe_aux_devices(). Before that commit, it traversed card->aux_comp_list and found aux devices added by soc_bind_aux_dev(). After that commit, it traverses card->component_dev_list and finds no aux devices, because soc_probe_component() was not yet called for them. Thus all aux devices are lost and any setup that needs them no longer works. Signed-off-by: Nikita Yushchenko <nikita.yoush(a)cogentembedded.com> --- Fixed lost line in commit message ;) include/sound/soc.h | 4 +++- sound/soc/soc-core.c | 17 +++++------------ 2 files changed, 8 insertions(+), 13 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2b502f6cc6d0..8172a512632a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -809,10 +809,10 @@ struct snd_soc_component { unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */ unsigned int registered_as_component:1; - unsigned int auxiliary:1; /* for auxiliary component of the card */ unsigned int suspended:1; /* is in suspend PM state */ struct list_head list; + struct list_head list_aux; /* for auxiliary component of the card */ struct list_head card_list; struct snd_soc_dai_driver *dai_drv; @@ -1152,6 +1152,7 @@ struct snd_soc_card { */ struct snd_soc_aux_dev *aux_dev; int num_aux_devs; + struct list_head aux_comp_list; const struct snd_kcontrol_new *controls; int num_controls; @@ -1547,6 +1548,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->widgets); INIT_LIST_HEAD(&card->paths); INIT_LIST_HEAD(&card->dapm_list); + INIT_LIST_HEAD(&card->aux_comp_list); INIT_LIST_HEAD(&card->component_dev_list); } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f1901bb1466e..15657a65e4ca 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1747,8 +1747,7 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) } component->init = aux_dev->init; - component->auxiliary = 1; - + list_add(&component->list_aux, &card->aux_comp_list); return 0; err_defer: @@ -1764,10 +1763,7 @@ static int soc_probe_aux_devices(struct snd_soc_card *card) for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - list_for_each_entry(comp, &card->component_dev_list, card_list) { - if (!comp->auxiliary) - continue; - + list_for_each_entry(comp, &card->aux_comp_list, list_aux) { if (comp->driver->probe_order == order) { ret = soc_probe_component(card, comp); if (ret < 0) { @@ -1791,14 +1787,11 @@ static void soc_remove_aux_devices(struct snd_soc_card *card) for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { list_for_each_entry_safe(comp, _comp, - &card->component_dev_list, card_list) { - - if (!comp->auxiliary) - continue; - + &card->aux_comp_list, list_aux) { if (comp->driver->remove_order == order) { soc_remove_component(comp); - comp->auxiliary = 0; + /* remove it from the card's aux_comp_list */ + list_del(&comp->list_aux); } } } -- 2.1.4

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[alsa-devel] Alsamixer: missing Speakers entry in analog output choice (chipset AV200)
by Germano Massullo 08 Jan '17

08 Jan '17

I have a Asus Xonar Essence ST soundboard (Chip: AV200) on Fedora 23 kernel 4.4.5-300.fc23.x86_64 The audiocard has 3 different outputs - Speakers - Headphones - FP Headphones that have mutual exclusion, so that when you are using one of them, you cannot use the others. Sometimes the Speakers output disappears from alsamixer. By restarting the computer, the system gets back to its normal behaviour. I attach 3 logs taken when the problem was occurring. The problem happened also in past years, but in last a 1-2 years it happened less frequently

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[alsa-devel] [PATCH 1/7] ASoC: wm9705: Remove unused DAI ID defines
by Lars-Peter Clausen 06 Jan '17

06 Jan '17

The DAI ID defines are back from the time when DAIs were referenced by a numerical ID. These days a string is used for matching instead and the defines are unused. The last user of these defines was removed in commit f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support"). So remove the defines as well. This also means the wm9705.h file no longer has any content and can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm9705.c | 2 -- sound/soc/codecs/wm9705.h | 11 ----------- sound/soc/pxa/e740_wm9705.c | 1 - sound/soc/pxa/e750_wm9705.c | 1 - 4 files changed, 15 deletions(-) delete mode 100644 sound/soc/codecs/wm9705.h diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index dcdd055..266b6ec 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -20,8 +20,6 @@ #include <sound/initval.h> #include <sound/soc.h> -#include "wm9705.h" - #define WM9705_VENDOR_ID 0x574d4c05 #define WM9705_VENDOR_ID_MASK 0xffffffff diff --git a/sound/soc/codecs/wm9705.h b/sound/soc/codecs/wm9705.h deleted file mode 100644 index 23ea9ce..0000000 --- a/sound/soc/codecs/wm9705.h +++ /dev/null @@ -1,11 +0,0 @@ -/* - * wm9705.h -- WM9705 Soc Audio driver - */ - -#ifndef _WM9705_H -#define _WM9705_H - -#define WM9705_DAI_AC97_HIFI 0 -#define WM9705_DAI_AC97_AUX 1 - -#endif diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index 1de8765..086c37a 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -22,7 +22,6 @@ #include <asm/mach-types.h> -#include "../codecs/wm9705.h" #include "pxa2xx-ac97.h" diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index b7eb7cd..7823278 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -22,7 +22,6 @@ #include <asm/mach-types.h> -#include "../codecs/wm9705.h" #include "pxa2xx-ac97.h" static int e750_spk_amp_event(struct snd_soc_dapm_widget *w, -- 2.1.4

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[alsa-devel] [PATCH v3 00/13] ASoC: Intel: Skylake: Driver updates
by jeeja.kp＠intel.com 06 Jan '17

06 Jan '17

From: Jeeja KP <jeeja.kp(a)intel.com> This patch series includes driver updates for the handling scenarios during System S0/S3 transitions when audio stream is stopped during system. o move DMA configuration in DSP Gateway Module widget event handler to ensure DMA is configured after resume. o Enable SSP MLCK using Supply widget. This will ensure the MCLK is enabled when the widget is power on. o Check DMA decouple register state before coupling/decoupling the DMA. o In case of HDMI pass-through pipe, reset only the FE Pipe when stream in in XRUN state. o Resume the stream from HW renderer position. o HDMI Codec pin and converter state need to be restored when stream is resumed. Changes in v2: - Addressed Mark's comment on patch 1, split this into 2 as this had additional changes other than configuring the DMA. - Addressed Takashi's comment on the usage of snd_hdac_updatel() Changes in v3: - Addressed Takashi's comment on superfluous shit used in reading the value of PPCTL register and also cleaned the condition for updating the reg value. Jeeja KP (13): ASoC: Intel: Skylake: Update link_index and format in pipe params ASoC: Intel: Skylake: Add helper function to setup host/link dma ASoC: Intel: Skylake: Configure DMA in PRE_PMD handler of Mixer ASoC: Intel: Skylake: Removed unused skl_get_format() ALSA: hda: check stream decoupled register state ASoC: Intel: Skylake: Add set_tristate DAI ops to enable SSP MCLK ASoC: Intel: Skylake: Remove unused SSP BE prepare DAI ops ASoC: Intel: Skylake: Add supply widget as non DSP widget ASoC: Intel: Skylake: Add supply widget in skl_nau_max machine ASoC: Intel: Skylake: Add supply widget in bxt_da_max machine ASoC: Intel: Skylake: Don't reset pass-through pipe in BE prepare ASoC: Intel: Skylake: set the resume point to LPIB ASoC: hdac_hdmi: Enable pin and converter in prepare sound/hda/ext/hdac_ext_stream.c | 15 +- sound/soc/codecs/hdac_hdmi.c | 28 ++-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 34 ++++ sound/soc/intel/boards/skl_nau88l25_max98357a.c | 34 ++++ sound/soc/intel/skylake/skl-pcm.c | 210 +++++++++++++----------- sound/soc/intel/skylake/skl-topology.c | 27 ++- sound/soc/intel/skylake/skl-topology.h | 6 + 7 files changed, 240 insertions(+), 114 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH] ASoC: pcm5102a: Provide a menu selection text
by Jörg Krause 06 Jan '17

06 Jan '17

Provide a menu selection text so that users can enable, disable or mark it as module in menuconfig. Signed-off-by: Jörg Krause <joerg.krause(a)embedded.rocks> --- 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 084c2aa..9ee9591 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -624,7 +624,7 @@ config SND_SOC_PCM3168A_SPI select REGMAP_SPI config SND_SOC_PCM5102A - tristate + tristate "Texas Instruments PCM5102A CODEC" config SND_SOC_PCM512x tristate -- 2.10.2

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Re: [alsa-devel] Stream format conversion in SoC driver
by Clemens Ladisch 05 Jan '17

05 Jan '17

Giedrius Trainavičius wrote: >It uses I2S for sending / receiving audio. However, the master clock on our >sound card is generated by the ADC, which supports only 24 bit data, so >getting other sample formats is not possible. The I2S sample format need not be the same as what the DMA controller reads/writes from/to memory. >A question on 'plug' - it accepts 'nearest' as the argument for format, >rate, what happens in case the slave pcm stream supports the exact same >format as requested by user software? The plug plugin instantiates the actual conversion plugins only as needed; with no conversion, it vanishes entirely. Regards, Clemens

2 4

[alsa-devel] [PATCH v6 0/3] Add platform clock for BayTrail platforms
by Irina Tirdea 05 Jan '17

05 Jan '17

These patches specifically enable the audio MCLK required by Baytrail CR devices. It is the remaining part of a bigger set of patches (already merged in Mark's tree) that enable sound for Baytrail CR devices (especially Asus T100TAF) [1]. They include the clock driver and enabling the clock in the pmc_atom code (along with moving of the non-architectural pmc_atom driver code into drivers/platform/x86). [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/111704.html Changes from v5: - fix build error reported by kbuild test robot - split the clk driver code from x86 platform changes Changes from v4: - move the pmc_atom driver from arch/x86/platform/atom to drivers/platform/x86 Changes from v3: - replace devm_kzalloc with devm_kcalloc - add x86 architecture maintainers Changes from v2: - move clk platform data structures to a separate include file - store clk_hw pointer for the fixed rate clocks Changes from v1: - register the clk device as child of pmc device - pass iomem pointer from pmc driver to clk driver to avoid using pmc_atom_read()/write() and use readl/writel API instead - use devm_clk_hw_register/clkdev_hw_create instead of clk_register/clkdev_create Irina Tirdea (3): clk: x86: Add Atom PMC platform clocks arch/x86/platform/atom: Move pmc_atom to drivers/platform/x86 platform/x86: Enable Atom PMC platform clocks arch/x86/Kconfig | 4 - arch/x86/platform/atom/Makefile | 1 - drivers/acpi/acpi_lpss.c | 2 +- drivers/clk/x86/Makefile | 3 + drivers/clk/x86/clk-byt-plt.c | 380 +++++++++++++++++++++ drivers/platform/x86/Kconfig | 5 + drivers/platform/x86/Makefile | 1 + .../atom => drivers/platform/x86}/pmc_atom.c | 81 ++++- include/linux/platform_data/x86/clk-byt-plt.h | 31 ++ .../linux/platform_data/x86}/pmc_atom.h | 3 + 10 files changed, 500 insertions(+), 11 deletions(-) create mode 100644 drivers/clk/x86/clk-byt-plt.c rename {arch/x86/platform/atom => drivers/platform/x86}/pmc_atom.c (88%) create mode 100644 include/linux/platform_data/x86/clk-byt-plt.h rename {arch/x86/include/asm => include/linux/platform_data/x86}/pmc_atom.h (98%) -- 1.9.1

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[alsa-devel] [PATCH] drm: sti: allow audio playback on HDMI even if disabled.
by Arnaud Pouliquen 04 Jan '17

04 Jan '17

This fix allows to play audio while HDMI is disconnected. When HDMI is disable, audio configuration is stored and samples are dropped (by HDMI IP). When HDMI is enabled, audio HDMI configuration is applied and samples are outputted on HDMI wire. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- drivers/gpu/drm/sti/sti_hdmi.c | 205 ++++++++++++++++++++--------------------- 1 file changed, 101 insertions(+), 104 deletions(-) diff --git a/drivers/gpu/drm/sti/sti_hdmi.c b/drivers/gpu/drm/sti/sti_hdmi.c index 376b076..9c0025e 100644 --- a/drivers/gpu/drm/sti/sti_hdmi.c +++ b/drivers/gpu/drm/sti/sti_hdmi.c @@ -788,6 +788,95 @@ static void sti_hdmi_disable(struct drm_bridge *bridge) hdmi->enabled = false; } +/** + * sti_hdmi_audio_get_non_coherent_n() - get N parameter for non-coherent + * clocks. None-coherent clocks means that audio and TMDS clocks have not the + * same source (drifts between clocks). In this case assumption is that CTS is + * automatically calculated by hardware. + * + * @audio_fs: audio frame clock frequency in Hz + * + * Values computed are based on table described in HDMI specification 1.4b + * + * Returns n value. + */ +static int sti_hdmi_audio_get_non_coherent_n(unsigned int audio_fs) +{ + unsigned int n; + + switch (audio_fs) { + case 32000: + n = 4096; + break; + case 44100: + n = 6272; + break; + case 48000: + n = 6144; + break; + case 88200: + n = 6272 * 2; + break; + case 96000: + n = 6144 * 2; + break; + case 176400: + n = 6272 * 4; + break; + case 192000: + n = 6144 * 4; + break; + default: + /* Not pre-defined, recommended value: 128 * fs / 1000 */ + n = (audio_fs * 128) / 1000; + } + + return n; +} + +static int hdmi_audio_configure(struct sti_hdmi *hdmi) +{ + int audio_cfg, n; + struct hdmi_audio_params *params = &hdmi->audio; + struct hdmi_audio_infoframe *info = &params->cea; + + DRM_DEBUG_DRIVER("\n"); + + if (!hdmi->enabled) + return 0; + + /* update N parameter */ + n = sti_hdmi_audio_get_non_coherent_n(params->sample_rate); + + DRM_DEBUG_DRIVER("Audio rate = %d Hz, TMDS clock = %d Hz, n = %d\n", + params->sample_rate, hdmi->mode.clock * 1000, n); + hdmi_write(hdmi, n, HDMI_AUDN); + + /* update HDMI registers according to configuration */ + audio_cfg = HDMI_AUD_CFG_SPDIF_DIV_2 | HDMI_AUD_CFG_DTS_INVALID | + HDMI_AUD_CFG_ONE_BIT_INVALID; + + switch (info->channels) { + case 8: + audio_cfg |= HDMI_AUD_CFG_CH78_VALID; + case 6: + audio_cfg |= HDMI_AUD_CFG_CH56_VALID; + case 4: + audio_cfg |= HDMI_AUD_CFG_CH34_VALID | HDMI_AUD_CFG_8CH; + case 2: + audio_cfg |= HDMI_AUD_CFG_CH12_VALID; + break; + default: + DRM_ERROR("ERROR: Unsupported number of channels (%d)!\n", + info->channels); + return -EINVAL; + } + + hdmi_write(hdmi, audio_cfg, HDMI_AUDIO_CFG); + + return hdmi_audio_infoframe_config(hdmi); +} + static void sti_hdmi_pre_enable(struct drm_bridge *bridge) { struct sti_hdmi *hdmi = bridge->driver_private; @@ -826,9 +915,12 @@ static void sti_hdmi_pre_enable(struct drm_bridge *bridge) if (hdmi_avi_infoframe_config(hdmi)) DRM_ERROR("Unable to configure AVI infoframe\n"); - /* Program AUDIO infoframe */ - if (hdmi_audio_infoframe_config(hdmi)) - DRM_ERROR("Unable to configure AUDIO infoframe\n"); + if (hdmi->audio.enabled) { + if (hdmi_audio_configure(hdmi)) + DRM_ERROR("Unable to configure audio\n"); + } else { + hdmi_audio_infoframe_config(hdmi); + } /* Program VS infoframe */ if (hdmi_vendor_infoframe_config(hdmi)) @@ -1078,97 +1170,6 @@ static struct drm_encoder *sti_hdmi_find_encoder(struct drm_device *dev) return NULL; } -/** - * sti_hdmi_audio_get_non_coherent_n() - get N parameter for non-coherent - * clocks. None-coherent clocks means that audio and TMDS clocks have not the - * same source (drifts between clocks). In this case assumption is that CTS is - * automatically calculated by hardware. - * - * @audio_fs: audio frame clock frequency in Hz - * - * Values computed are based on table described in HDMI specification 1.4b - * - * Returns n value. - */ -static int sti_hdmi_audio_get_non_coherent_n(unsigned int audio_fs) -{ - unsigned int n; - - switch (audio_fs) { - case 32000: - n = 4096; - break; - case 44100: - n = 6272; - break; - case 48000: - n = 6144; - break; - case 88200: - n = 6272 * 2; - break; - case 96000: - n = 6144 * 2; - break; - case 176400: - n = 6272 * 4; - break; - case 192000: - n = 6144 * 4; - break; - default: - /* Not pre-defined, recommended value: 128 * fs / 1000 */ - n = (audio_fs * 128) / 1000; - } - - return n; -} - -static int hdmi_audio_configure(struct sti_hdmi *hdmi, - struct hdmi_audio_params *params) -{ - int audio_cfg, n; - struct hdmi_audio_infoframe *info = &params->cea; - - DRM_DEBUG_DRIVER("\n"); - - if (!hdmi->enabled) - return 0; - - /* update N parameter */ - n = sti_hdmi_audio_get_non_coherent_n(params->sample_rate); - - DRM_DEBUG_DRIVER("Audio rate = %d Hz, TMDS clock = %d Hz, n = %d\n", - params->sample_rate, hdmi->mode.clock * 1000, n); - hdmi_write(hdmi, n, HDMI_AUDN); - - /* update HDMI registers according to configuration */ - audio_cfg = HDMI_AUD_CFG_SPDIF_DIV_2 | HDMI_AUD_CFG_DTS_INVALID | - HDMI_AUD_CFG_ONE_BIT_INVALID; - - switch (info->channels) { - case 8: - audio_cfg |= HDMI_AUD_CFG_CH78_VALID; - case 6: - audio_cfg |= HDMI_AUD_CFG_CH56_VALID; - case 4: - audio_cfg |= HDMI_AUD_CFG_CH34_VALID | HDMI_AUD_CFG_8CH; - case 2: - audio_cfg |= HDMI_AUD_CFG_CH12_VALID; - break; - default: - DRM_ERROR("ERROR: Unsupported number of channels (%d)!\n", - info->channels); - return -EINVAL; - } - - hdmi_write(hdmi, audio_cfg, HDMI_AUDIO_CFG); - - hdmi->audio = *params; - - return hdmi_audio_infoframe_config(hdmi); -} - static void hdmi_audio_shutdown(struct device *dev, void *data) { struct sti_hdmi *hdmi = dev_get_drvdata(dev); @@ -1192,17 +1193,9 @@ static int hdmi_audio_hw_params(struct device *dev, { struct sti_hdmi *hdmi = dev_get_drvdata(dev); int ret; - struct hdmi_audio_params audio = { - .sample_width = params->sample_width, - .sample_rate = params->sample_rate, - .cea = params->cea, - }; DRM_DEBUG_DRIVER("\n"); - if (!hdmi->enabled) - return 0; - if ((daifmt->fmt != HDMI_I2S) || daifmt->bit_clk_inv || daifmt->frame_clk_inv || daifmt->bit_clk_master || daifmt->frame_clk_master) { @@ -1213,9 +1206,13 @@ static int hdmi_audio_hw_params(struct device *dev, return -EINVAL; } - audio.enabled = true; + hdmi->audio.sample_width = params->sample_width; + hdmi->audio.sample_rate = params->sample_rate; + hdmi->audio.cea = params->cea; + + hdmi->audio.enabled = true; - ret = hdmi_audio_configure(hdmi, &audio); + ret = hdmi_audio_configure(hdmi); if (ret < 0) return ret; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC : fsl_ssi : Correct the condition to check AC97 mode
by Harisangam, Sharvari (S.) 04 Jan '17

04 Jan '17

Corrected the condition to check if ssi is configured for AC97 mode. Other modes like dsp_a also satisfy the ANDing condition. Signed-off-by: Sharvari Harisangam <sharvari.harisangam(a)visteon.com> --- sound/soc/fsl/fsl_ssi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ca6f6b1..4fbb7a2 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -317,7 +317,8 @@ MODULE_DEVICE_TABLE(of, fsl_ssi_ids); static bool fsl_ssi_is_ac97(struct fsl_ssi_private *ssi_private) { - return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97); + return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97) == + SND_SOC_DAIFMT_AC97; } static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private) @@ -1016,7 +1017,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, CCSR_SSI_SCR_TCH_EN); } - if (fmt & SND_SOC_DAIFMT_AC97) + if ((fmt & SND_SOC_DAIFMT_AC97) == SND_SOC_DAIFMT_AC97) fsl_ssi_setup_ac97(ssi_private); return 0; -- 2.7.4

5 5

[alsa-devel] [PATCH - alsa-lib 1/1] Change card/pid get functions to return -ENOSYS if the kernel is too old
by Adam Goode 03 Jan '17

03 Jan '17

When trying to get the pid or card of a client, the get functions will return -1 if there is no pid or card for a client. Clients have zero or one of these, so -1 is correct for these cases. But we also need to detect the case where the kernel cannot tell us if there is a card or pid, so that userspace can fallback to probing this information in the old way. Signed-off-by: Adam Goode <agoode(a)google.com> diff --git a/src/seq/seq.c b/src/seq/seq.c index 4405e68..7de1e81 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -1522,9 +1522,10 @@ int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info) } /** - * \brief Get the sound card number. + * \brief Get the sound card number if the kernel supports this. * \param info client_info container - * \return card number or -1 if value is not available. + * \return card number, -1 if there is no card for this client, + * or \c -ENOSYS if the kernel does not have support for this operation * * Only available for SND_SEQ_KERNEL_CLIENT clients. */ @@ -1535,9 +1536,10 @@ int snd_seq_client_info_get_card(const snd_seq_client_info_t *info) } /** - * \brief Get the owning PID. + * \brief Get the owning PID if the kernel supports this. * \param info client_info container - * \return pid or -1 if value is not available. + * \return pid, -1 if there is no PID for this client, + * or \c -ENOSYS if the kernel does not have support for this operation * * Only available for SND_SEQ_USER_CLIENT clients. */ diff --git a/src/seq/seq_hw.c b/src/seq/seq_hw.c index 578ef12..a1d1e4a 100644 --- a/src/seq/seq_hw.c +++ b/src/seq/seq_hw.c @@ -102,8 +102,8 @@ static int snd_seq_hw_get_client_info(snd_seq_t *seq, snd_seq_client_info_t * in return -errno; } if (hw->version < SNDRV_PROTOCOL_VERSION(1, 0, 2)) { - info->card = -1; - info->pid = -1; + info->card = -ENOSYS; + info->pid = -ENOSYS; } return 0; } @@ -374,8 +374,8 @@ static int snd_seq_hw_query_next_client(snd_seq_t *seq, snd_seq_client_info_t *i return -errno; } if (hw->version < SNDRV_PROTOCOL_VERSION(1, 0, 2)) { - info->card = -1; - info->pid = -1; + info->card = -ENOSYS; + info->pid = -ENOSYS; } return 0; } -- 2.8.0.rc3.226.g39d4020

5 16

[alsa-devel] [PATCH v4] ASoC: omap-mcbsp: Add PM QoS support for McBSP to prevent glitches
by Matt Ranostay 03 Jan '17

03 Jan '17

We can get audio errors if hitting deeper idle states on omaps: [alsa.c:230] error: Fatal problem with alsa output, error -5. [audio.c:614] error: Error in writing audio (Input/output error?)! This seems to happen with off mode idle enabled as power for the whole SoC may get cut off between filling the McBSP fifo using DMA. While active DMA blocks deeper idle states in hardware, McBSP activity does not seem to do so. Basing the QoS latency calculation on the FIFO size, threshold, sample rate, and channels. Based on the original patch by Tony Lindgren Link: https://patchwork.kernel.org/patch/9305867/ Cc: Tony Lindgren <tony(a)atomide.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Matt Ranostay <matt(a)ranostay.consulting> --- Changes from v1: * add calculations for latency per number of FIFO locations Changes from v2: * add missing mcbsp.h header change Changes from v3: * base the latency calculations on threshold, buffer size, sample rate, and channels sound/soc/omap/mcbsp.c | 19 +++++++++++++++++++ sound/soc/omap/mcbsp.h | 3 +++ sound/soc/omap/omap-mcbsp.c | 21 ++++++++++++++++++++- 3 files changed, 42 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 06fec5699cc8..8f792e714894 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -25,6 +25,7 @@ #include <linux/io.h> #include <linux/slab.h> #include <linux/pm_runtime.h> +#include <linux/pm_qos.h> #include <linux/platform_data/asoc-ti-mcbsp.h> @@ -640,9 +641,19 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) */ void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx) { + struct pm_qos_request *pm_qos_req = &mcbsp->pm_qos_req; int enable_srg = 0; u16 w; + /* Prevent omap hardware from hitting off between fifo fills */ + if (mcbsp->latency) { + if (pm_qos_request_active(pm_qos_req)) + pm_qos_update_request(pm_qos_req, mcbsp->latency); + else + pm_qos_add_request(pm_qos_req, PM_QOS_CPU_DMA_LATENCY, + mcbsp->latency); + } + if (mcbsp->st_data) omap_st_start(mcbsp); @@ -731,6 +742,11 @@ void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx) if (mcbsp->st_data) omap_st_stop(mcbsp); + + if (mcbsp->active == 1 && pm_qos_request_active(&mcbsp->pm_qos_req)) { + pm_qos_remove_request(&mcbsp->pm_qos_req); + mcbsp->latency = 0; + } } int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) @@ -1098,6 +1114,9 @@ int omap_mcbsp_init(struct platform_device *pdev) void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp) { + if (pm_qos_request_active(&mcbsp->pm_qos_req)) + pm_qos_remove_request(&mcbsp->pm_qos_req); + if (mcbsp->pdata->buffer_size) sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index 61e93b1c185d..d312e72f7979 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -323,8 +323,11 @@ struct omap_mcbsp { unsigned int fmt; unsigned int in_freq; + unsigned int latency; int clk_div; int wlen; + + struct pm_qos_request pm_qos_req; }; void omap_mcbsp_config(struct omap_mcbsp *mcbsp, diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d018e966e533..ddaec37fb4c3 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -226,6 +226,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, int wlen, channels, wpf; int pkt_size = 0; unsigned int format, div, framesize, master; + unsigned int buffer_size = mcbsp->pdata->buffer_size; dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); channels = params_channels(params); @@ -240,7 +241,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } - if (mcbsp->pdata->buffer_size) { + if (buffer_size) { + int latency; + if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) { int period_words, max_thrsh; int divider = 0; @@ -271,6 +274,22 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, /* Use packet mode for non mono streams */ pkt_size = channels; } + + latency = ((((buffer_size - pkt_size) / channels) * 1000) + / (params->rate_num / params->rate_den)); + + /* + * Use the lowest QoS latency value for active streams, but + * this has the disadvantage of using lower than needed latency + * if an stream with lower QoS ends before one with a higher + * one. Latency setting only gets reset when all streams + * complete. + */ + if (latency <= mcbsp->latency) + mcbsp->latency = latency; + else + mcbsp->latency = 0; + omap_mcbsp_set_threshold(substream, pkt_size); } -- 2.7.4

2 2

[alsa-devel] [PATCH][alsa-lib] topology: fix unused-const-variable warning
by Takashi Sakamoto 02 Jan '17

02 Jan '17

Last year, unused static const variable was added, then compiler generates a below warning. dapm.c:43:30: warning: ‘widget_control_map’ defined but not used [-Wunused-const-variable=] static const struct map_elem widget_control_map[] = { ^~~~~~~~~~~~~~~~~~ This commit removes it. Fixes: 01a0e1a1c219 ("topology: Add DAPM object parser") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- src/topology/dapm.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/src/topology/dapm.c b/src/topology/dapm.c index e830751..8c585a7 100644 --- a/src/topology/dapm.c +++ b/src/topology/dapm.c @@ -39,14 +39,6 @@ static const struct map_elem widget_map[] = { {"dai_link", SND_SOC_TPLG_DAPM_DAI_LINK}, }; -/* mapping of widget kcontrol text names to types */ -static const struct map_elem widget_control_map[] = { - {"volsw", SND_SOC_TPLG_DAPM_CTL_VOLSW}, - {"enum_double", SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE}, - {"enum_virt", SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT}, - {"enum_value", SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE}, -}; - static int lookup_widget(const char *w) { unsigned int i; -- 2.9.3

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