November 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Fix kerneldoc errors
by Takashi Iwai 11 Nov '14

11 Nov '14

Lots of typos and missing parameter descriptions. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/soc.h | 11 ++++++++--- sound/soc/soc-core.c | 21 ++++++++++++++------- sound/soc/soc-dapm.c | 2 +- sound/soc/soc-devres.c | 2 +- sound/soc/soc-jack.c | 2 +- 5 files changed, 25 insertions(+), 13 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130037a0..25d0bdbbeb11 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -583,6 +583,7 @@ int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, /** * struct snd_soc_jack_pin - Describes a pin to update based on jack detection * + * @list: linked list of assigned pins to the same jack * @pin: name of the pin to update * @mask: bits to check for in reported jack status * @invert: if non-zero then pin is enabled when status is not reported @@ -602,7 +603,7 @@ struct snd_soc_jack_pin { * @jack_type: type of jack that is expected for this voltage * @debounce_time: debounce_time for jack, codec driver should wait for this * duration before reading the adc for voltages - * @:list: list container + * @list: list container */ struct snd_soc_jack_zone { unsigned int min_mv; @@ -618,16 +619,20 @@ struct snd_soc_jack_zone { * @gpio: legacy gpio number * @idx: gpio descriptor index within the function of the GPIO * consumer device - * @gpiod_dev GPIO consumer device + * @gpiod_dev: GPIO consumer device * @name: gpio name. Also as connection ID for the GPIO consumer * device function name lookup * @report: value to report when jack detected * @invert: report presence in low state - * @debouce_time: debouce time in ms + * @debounce_time: debouce time in ms * @wake: enable as wake source + * @jack: assigned jack + * @work: delayed_work for GPIO detection + * @desc: GPIO descriptor * @jack_status_check: callback function which overrides the detection * to provide more complex checks (eg, reading an * ADC). + * @data: arbitrary data passed to jack_status_check callback */ struct snd_soc_jack_gpio { unsigned int gpio; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c8f8a23a0e9..74a8a80e2d9a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2113,6 +2113,8 @@ EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops); /** * snd_soc_set_ac97_ops_of_reset - Set ac97 ops with generic ac97 reset functions + * @ops: AC97 opts to set + * @pdev: platform to deal with * * This function sets the reset and warm_reset properties of ops and parses * the device node of pdev to get pinctrl states and gpio numbers to use. @@ -2685,7 +2687,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); /** * snd_soc_put_volsw_sx - double mixer set callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to set the value of a double mixer control that spans 2 registers. * @@ -3458,7 +3460,7 @@ EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); /** * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. * @dai: DAI - * @ratio Ratio of BCLK to Sample rate. + * @ratio: Ratio of BCLK to Sample rate. * * Configures the DAI for a preset BCLK to sample rate ratio. */ @@ -4082,6 +4084,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_component); /** * snd_soc_unregister_component - Unregister a component from the ASoC core + * @dev: device to unregister * */ void snd_soc_unregister_component(struct device *dev) @@ -4119,7 +4122,7 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component) * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform * @platform: The platform to add - * @platform_driver: The driver for the platform + * @platform_drv: The driver for the platform */ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, const struct snd_soc_platform_driver *platform_drv) @@ -4158,7 +4161,8 @@ EXPORT_SYMBOL_GPL(snd_soc_add_platform); /** * snd_soc_register_platform - Register a platform with the ASoC core * - * @platform: platform to register + * @dev: platform to register + * @platform_drv: the driver for the platform */ int snd_soc_register_platform(struct device *dev, const struct snd_soc_platform_driver *platform_drv) @@ -4215,7 +4219,7 @@ EXPORT_SYMBOL_GPL(snd_soc_lookup_platform); /** * snd_soc_unregister_platform - Unregister a platform from the ASoC core * - * @platform: platform to unregister + * @dev: platform to unregister */ void snd_soc_unregister_platform(struct device *dev) { @@ -4306,7 +4310,10 @@ static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, /** * snd_soc_register_codec - Register a codec with the ASoC core * - * @codec: codec to register + * @dev: codec to register + * @codec_drv: the codec driver to register + * @dai_drv: the array of DAI drivers to register + * @num_dai: the number of DAIs to register */ int snd_soc_register_codec(struct device *dev, const struct snd_soc_codec_driver *codec_drv, @@ -4406,7 +4413,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); /** * snd_soc_unregister_codec - Unregister a codec from the ASoC core * - * @codec: codec to unregister + * @dev: codec to unregister */ void snd_soc_unregister_codec(struct device *dev) { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c61cb9cedbcd..7050ec1d5b72 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2680,7 +2680,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); /** * snd_soc_dapm_new_widgets - add new dapm widgets - * @dapm: DAPM context + * @card: card object containing widgets * * Checks the codec for any new dapm widgets and creates them if found. * diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 057e5ef7dcce..a57921eeee81 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -60,7 +60,7 @@ static void devm_platform_release(struct device *dev, void *res) /** * devm_snd_soc_register_platform - resource managed platform registration * @dev: Device used to manage platform - * @platform: platform to register + * @platform_drv: platform to register * * Register a platform driver with automatic unregistration when the device is * unregistered. diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ab47fea997a3..ef1d42d7c6f6 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -116,7 +116,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_report); * * @jack: ASoC jack * @count: Number of zones - * @zone: Array of zones + * @zones: Array of zones * * After this function has been called the zones specified in the * array will be associated with the jack. -- 2.1.3

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[alsa-devel] [PATCH 1/2] ASoC: rt5645: Add ASRC support
by bardliao＠realtek.com 11 Nov '14

11 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patch add ASRC support for rt5645 codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 144 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 144 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index bc442f4..3131e91 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -443,6 +443,65 @@ static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_sel_enum, RT5645_TDM_CTRL_1, 8, rt5645_tdm_adc_data_select); +static int rt5645_clk_sel_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + unsigned int u_bit = 0, p_bit = 0; + struct soc_enum *em = + (struct soc_enum *)kcontrol->private_value; + + switch (em->reg) { + case RT5645_ASRC_2: + switch (em->shift_l) { + case 0: + u_bit = 0x8; + p_bit = RT5645_PWR_ADC_S1F; + break; + case 4: + u_bit = 0x100; + p_bit = RT5645_PWR_DAC_MF_R; + break; + case 8: + u_bit = 0x200; + p_bit = RT5645_PWR_DAC_MF_L; + break; + case 12: + u_bit = 0x400; + p_bit = RT5645_PWR_DAC_S1F; + break; + } + break; + case RT5645_ASRC_3: + switch (em->shift_l) { + case 0: + u_bit = 0x1; + p_bit = RT5645_PWR_ADC_MF_R; + break; + case 4: + u_bit = 0x2; + p_bit = RT5645_PWR_ADC_MF_L; + break; + } + break; + } + + if (u_bit || p_bit) { + switch (ucontrol->value.integer.value[0]) { + case 1 ... 4: /*enable*/ + if (snd_soc_read(codec, RT5645_PWR_DIG2) & p_bit) + snd_soc_update_bits(codec, + RT5645_ASRC_1, u_bit, u_bit); + break; + default: /*disable*/ + snd_soc_update_bits(codec, RT5645_ASRC_1, u_bit, 0); + break; + } + } + + return snd_soc_put_enum_double(kcontrol, ucontrol); +} + static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, @@ -554,6 +613,53 @@ static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, return 0; } +static int is_using_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, shift, val; + + switch (source->shift) { + case 0: + reg = RT5645_ASRC_3; + shift = 0; + break; + case 1: + reg = RT5645_ASRC_3; + shift = 4; + break; + case 3: + reg = RT5645_ASRC_2; + shift = 0; + break; + case 8: + reg = RT5645_ASRC_2; + shift = 4; + break; + case 9: + reg = RT5645_ASRC_2; + shift = 8; + break; + case 10: + reg = RT5645_ASRC_2; + shift = 12; + break; + default: + return 0; + } + + val = (snd_soc_read(source->codec, reg) >> shift) & 0xf; + switch (val) { + case 1: + case 2: + case 3: + case 4: + return 1; + default: + return 0; + } + +} + /* Digital Mixer */ static const struct snd_kcontrol_new rt5645_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5645_STO1_ADC_MIXER, @@ -1246,6 +1352,30 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5645_PWR_VOL, RT5645_PWR_MIC_DET_BIT, 0, NULL, 0), + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5645_ASRC_1, + 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("I2S2 ASRC", 1, RT5645_ASRC_1, + 12, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC STO ASRC", 1, RT5645_ASRC_1, + 10, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO L ASRC", 1, RT5645_ASRC_1, + 9, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO R ASRC", 1, RT5645_ASRC_1, + 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC STO1 ASRC", 1, RT5645_ASRC_1, + 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC MONO L ASRC", 1, RT5645_ASRC_1, + 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC MONO R ASRC", 1, RT5645_ASRC_1, + 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5645_ASRC_1, + 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO L ASRC", 1, RT5645_ASRC_1, + 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO R ASRC", 1, RT5645_ASRC_1, + 0, 0, NULL, 0), + /* Input Side */ /* micbias */ SND_SOC_DAPM_MICBIAS("micbias1", RT5645_PWR_ANLG2, @@ -1504,6 +1634,17 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { }; static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { + { "adc stereo1 filter", NULL, "ADC STO1 ASRC", is_using_asrc }, + { "adc stereo2 filter", NULL, "ADC STO2 ASRC", is_using_asrc }, + { "adc mono left filter", NULL, "ADC MONO L ASRC", is_using_asrc }, + { "adc mono right filter", NULL, "ADC MONO R ASRC", is_using_asrc }, + { "dac mono left filter", NULL, "DAC MONO L ASRC", is_using_asrc }, + { "dac mono right filter", NULL, "DAC MONO R ASRC", is_using_asrc }, + { "dac stereo1 filter", NULL, "DAC STO ASRC", is_using_asrc }, + + { "I2S1", NULL, "I2S1 ASRC" }, + { "I2S2", NULL, "I2S2 ASRC" }, + { "IN1P", NULL, "LDO2" }, { "IN2P", NULL, "LDO2" }, @@ -1550,12 +1691,15 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "Stereo1 DMIC Mux", "DMIC1", "DMIC1" }, { "Stereo1 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo1 DMIC Mux", NULL, "DMIC STO1 ASRC" }, { "Mono DMIC L Mux", "DMIC1", "DMIC L1" }, { "Mono DMIC L Mux", "DMIC2", "DMIC L2" }, + { "Mono DMIC L Mux", NULL, "DMIC MONO L ASRC" }, { "Mono DMIC R Mux", "DMIC1", "DMIC R1" }, { "Mono DMIC R Mux", "DMIC2", "DMIC R2" }, + { "Mono DMIC R Mux", NULL, "DMIC MONO R ASRC" }, { "Stereo1 ADC L2 Mux", "DMIC", "Stereo1 DMIC Mux" }, { "Stereo1 ADC L2 Mux", "DAC MIX", "DAC MIXL" }, -- 1.8.1.1.439.g50a6b54

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[alsa-devel] Issue with hda_intel on i7+Z97 chipset
by Daniel Silverstone 10 Nov '14

10 Nov '14

Hi, Firstly, I hope this is a suitable way to report this issue. I am using Debian/Wheezy on a reasonably recent i7 based system which has a Z97 motherboard chipset. In Windows, sound is perfect, but in Linux it's very crackly (direct through alsa) and if I bring pulseaudio into the mix then it reports buffer/interrupt issues on the microphone along the lines of: Oct 19 16:03:33 listless pulseaudio[3780]: [alsa-source] alsa-source.c: ALSA woke us up to read new data from the device, but there was actually nothing to read! Oct 19 16:03:33 listless pulseaudio[3780]: [alsa-source] alsa-source.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Oct 19 16:03:33 listless pulseaudio[3780]: [alsa-source] alsa-source.c: We were woken up with POLLIN set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Similar messages are shown for issues with playback: user.log.2.gz:Oct 18 21:50:25 listless pulseaudio[3780]: [alsa-sink] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! user.log.2.gz:Oct 18 21:50:25 listless pulseaudio[3780]: [alsa-sink] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. user.log.2.gz:Oct 18 21:50:25 listless pulseaudio[3780]: [alsa-sink] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. In both cases, sound exhibts crackles, pops, crunches, stalls, repeats etc. I can mitigate the issue by working one or more of the CPU cores hard with bzip2 -c </dev/urandom > /dev/null Of course, this makes my system fans run harder etc, which is a bit of a pain for me, and it's only masking the issue, not solving it. For reference, my motherboard's dmi shows: Manufacturer: Gigabyte Technology Co., Ltd. Product Name: Z97-D3H-CF And the CPU reports as: Family: Core i7 Manufacturer: Intel ID: C3 06 03 00 FF FB EB BF Signature: Type 0, Family 6, Model 60, Stepping 3 Version: Intel(R) Core(TM) i7-4790K CPU @ 4.00GHz The relevant PCI devices are: 00:03.0 0403: 8086:0c0c (rev 06) Subsystem: 8086:2010 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 51 Region 0: Memory at f7834000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel and 00:1b.0 0403: 8086:8ca0 Subsystem: 1458:a182 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 50 Region 0: Memory at f7830000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel and it's possible that this kernel message is relevant: [ 17.248493] hda-intel: IRQ timing workaround is activated for card #1. Suggest a bigger bdl_pos_adj. [ 17.251735] hda-intel: Invalid position buffer, using LPIB read method instead. I have tried a number of bdl_pos_adj between 1 and 32 and I have tried enable_msi=1 Neither seem to have helped significantly. I am currently running 3.12-0.bpo.1-amd64 from Debian's wheezy-backports repository. Unfortunately anything newer seems to lock up during boot and I've not managed to track down what or why as yet. I was hoping that you may have some idea of what is going on and either some suggestions for improving my situation on 3.12 or else some desire for further debug data such that a newer kernel (once I can persuade one to boot) may perform better for me. Please let me know if you require further details from me in order to address this. Regards, Daniel. -- Daniel Silverstone http://www.digital-scurf.org/ PGP mail accepted and encouraged. Key Id: 3CCE BABE 206C 3B69

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[alsa-devel] [PATCH 0/3] ASoC: twl4030: support routine to external VOICE source.
by NeilBrown 10 Nov '14

10 Nov '14

The 'voice' port of the twl4030 on my board is connected to a GSM modem, not to the CPU. As such it is not visible to ASLA and normal approaches to configuring the interface (as one end of a DAI) don't apply. I need a way to tell the twl4030 that the connected device will be master of 'clk' and 'FRM', and whether they are inverted. Using device tree seems the correct approach - I am describing the properties of the hardware attached the 'voice' port. Using magic numbers from include/sound/soc-dai.h almost certainly is not the right way to represent this information, but it isn't clear to me what a good way would be. So the current patch uses magic numbers as a proof of concept, and I'm asking if there is any establish precedent, or any suggestions on how to record in device tree the 'master' and 'polarity' of CLK and FRM signals for an external device. [I currently use (SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS)] This is in the second patch. The first silences a pointless warning (when it is pointless) and the last allowed the voice linked to be turned on or off via ALSA controls. If the first and last can be applied without finalizing the device-tree part yet, that would be great. Thanks, NeilBrown --- NeilBrown (3): ASoC: twl4030: don't report EBUSY if no change requested. ASoC: twl4030: allow voice port to be connected externally. ASoC: twl4030: enable routing audio to 'voice' interface. .../devicetree/bindings/mfd/twl4030-audio.txt | 7 ++ include/linux/i2c/twl.h | 3 + sound/soc/codecs/twl4030.c | 78 ++++++++++++++++++-- 3 files changed, 82 insertions(+), 6 deletions(-) -- Signature

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[alsa-devel] [PATCH] ASoC: mxs: mxs-saif: Register the irq with the device name
by Fabio Estevam 10 Nov '14

10 Nov '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Instead of registering the irq name with the driver name, it's better to pass the device name so that we have a more explicit indication as to what saif instance the irq is related: $ cat /proc/interrupts CPU0 ... 214: 4 - 59 80042000.saif 215: 0 - 58 80046000.saif Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-saif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index eade2742..d986508 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -773,7 +773,7 @@ static int mxs_saif_probe(struct platform_device *pdev) saif->dev = &pdev->dev; ret = devm_request_irq(&pdev->dev, saif->irq, mxs_saif_irq, 0, - "mxs-saif", saif); + dev_name(&pdev->dev), saif); if (ret) { dev_err(&pdev->dev, "failed to request irq\n"); return ret; -- 1.9.1

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[alsa-devel] [PATCH v2 0/2] Fix some 'const' problems
by Jean-Francois Moine 10 Nov '14

10 Nov '14

A cast was often needed to call some OF functions, removing a 'const' attribute. This patchset removes 'const' from the device_node pointers of the sound/soc structures and also removes some useless casts in the sound/soc tree. v2: remove const from the pointers instead of modifying the OF functions (Russell King) Jean-Francois Moine (2): ASoC: Remove 'const' from the device_node pointers ASoC: simple-card: Remove useless casts include/sound/soc.h | 6 +++--- sound/soc/generic/simple-card.c | 7 ++----- sound/soc/samsung/odroidx2_max98090.c | 4 ++-- sound/soc/ux500/mop500.c | 6 ++---- 4 files changed, 9 insertions(+), 14 deletions(-) -- 2.1.3

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[alsa-devel] [PATCH 0/8] ASoC: Replace remaining users of the CODEC mutex
by Lars-Peter Clausen 10 Nov '14

10 Nov '14

We've stopped using the snd_soc_codec mutex in the ASoC core itself a while ago. There is still a handful of drivers which use the mutex to protect some of their data structures. This patch moves the lock for the remaining users to the drivers private data struct and then removes it from the snd_soc_codec struct. - Lars Lars-Peter Clausen (8): ASoC: ab8500-codec: Move control lock to the driver level ASoC: max98095: Move mutex to the driver level ASoC: wm5102: Move ultrasonic response settings lock to the driver level ASoC: wm8731: Move the deemph lock to the driver level ASoC: wm8903: Move the deemph lock to the driver level ASoC: wm8958: Move DSP firmware lock to driver level ASoC: wm8962: Move DSP enable lock to the driver level ASoC: Remove CODEC mutex include/linux/mfd/arizona/core.h | 1 + include/sound/soc.h | 1 - sound/soc/codecs/ab8500-codec.c | 32 ++++++++++++++++---------------- sound/soc/codecs/arizona.c | 4 ++-- sound/soc/codecs/max98095.c | 12 ++++++++---- sound/soc/codecs/wm5102.c | 16 ++++++++-------- sound/soc/codecs/wm8731.c | 9 +++++++-- sound/soc/codecs/wm8903.c | 8 ++++++-- sound/soc/codecs/wm8958-dsp2.c | 12 ++++++------ sound/soc/codecs/wm8962.c | 8 ++++++-- sound/soc/codecs/wm8994.c | 2 ++ sound/soc/codecs/wm8994.h | 2 ++ sound/soc/soc-core.c | 1 - 13 files changed, 64 insertions(+), 44 deletions(-) -- 1.8.0

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[alsa-devel] [PATCH 0/4] Yet more kenreldoc fixes
by Takashi Iwai 10 Nov '14

10 Nov '14

Hi, these are a series of patches to fix kerneldoc errors / warnings. These aren't actaully compiled from DocBook directory, so they are harmless, but better to be fixed, of course. [PATCH 1/4] ALSA: vx: Fix missing kenreldoc parameter descriptions [PATCH 2/4] ALSA: hda - Fix kerneldoc errors in patch_ca0132.c [PATCH 3/4] ALSA: mixart: Fix kerneldoc comments [PATCH 4/4] ALSA: Fix invalid kerneldoc markers Takashi

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[alsa-devel] [PATCH 0/5 v2] ASoC: rsnd: Add Volume Ramp support
by Kuninori Morimoto 10 Nov '14

10 Nov '14

Hi Mark These are v2 of Renesas R-Car sound Volume Ramp support. It uses "Volume Ramp Up/Down Rate" on last patch instead of "Period". Because it is good match naming, and it was difficult to calculate real period time on this HW. Kuninori Morimoto (5): ASoC: rsnd: control DVC_DVUCR under rsnd_dvc_volume_update() ASoC: rsnd: move DVC_DVUER settings under rsnd_dvc_volume_update() ASoC: rsnd: enable multiple DVC valume settings ASoC: rsnd: enable single DVC valume settings ASoC: rsnd: Add Volume Ramp support sound/soc/sh/rcar/dvc.c | 126 +++++++++++++++++++++++++++++++++++++++------- sound/soc/sh/rcar/gen.c | 3 ++ sound/soc/sh/rcar/rsnd.h | 6 +++ 3 files changed, 116 insertions(+), 19 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: rsnd: fallback to PIO mode if DMA mode was failed
by Kuninori Morimoto 10 Nov '14

10 Nov '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current Renesas R-Car sound driver probe will be failed if it try to use DMA mode and it couldn't use for some reasons. But PIO mode works even though in such case. This patch try to fallback to PIO mode if DMA mode probing was failed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 74 +++++++++++++++++++++++++++++++++++++++++++--- sound/soc/sh/rcar/ssi.c | 15 ++++++++++ 2 files changed, 85 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 5205618..110b99d 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -349,7 +349,7 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, dma_name); if (!dma->chan) { dev_err(dev, "can't get dma channel\n"); - return -EIO; + goto rsnd_dma_channel_err; } ret = dmaengine_slave_config(dma->chan, &cfg); @@ -363,8 +363,15 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, rsnd_dma_init_err: rsnd_dma_quit(priv, dma); +rsnd_dma_channel_err: - return ret; + /* + * DMA failed. try to PIO mode + * see + * rsnd_ssi_dma_remove() + * rsnd_rdai_continuance_probe() + */ + return -EAGAIN; } void rsnd_dma_quit(struct rsnd_priv *priv, @@ -456,6 +463,13 @@ static int rsnd_dai_connect(struct rsnd_mod *mod, return 0; } +static void rsnd_dai_disconnect(struct rsnd_mod *mod, + struct rsnd_dai_stream *io) +{ + mod->io = NULL; + io->mod[mod->type] = NULL; +} + int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai) { int id = rdai - priv->rdai; @@ -686,6 +700,20 @@ static const struct snd_soc_dai_ops rsnd_soc_dai_ops = { ret; \ }) +#define rsnd_path_break(priv, io, type) \ +{ \ + struct rsnd_mod *mod; \ + int id = -1; \ + \ + if (rsnd_is_enable_path(io, type)) { \ + id = rsnd_info_id(priv, io, type); \ + if (id >= 0) { \ + mod = rsnd_##type##_mod_get(priv, id); \ + rsnd_dai_disconnect(mod, io); \ + } \ + } \ +} + static int rsnd_path_init(struct rsnd_priv *priv, struct rsnd_dai *rdai, struct rsnd_dai_stream *io) @@ -977,6 +1005,44 @@ static const struct snd_soc_component_driver rsnd_soc_component = { .name = "rsnd", }; +static int rsnd_rdai_continuance_probe(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + int is_play) +{ + struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture; + int ret; + + ret = rsnd_dai_call(probe, io, rdai); + if (ret == -EAGAIN) { + /* + * Fallback to PIO mode + */ + + /* + * call "remove" for SSI/SRC/DVC + * SSI will be switch to PIO mode if it was DMA mode + * see + * rsnd_dma_init() + * rsnd_ssi_dma_remove() + */ + rsnd_dai_call(remove, io, rdai); + + /* + * remove SRC/DVC from DAI, + */ + rsnd_path_break(priv, io, src); + rsnd_path_break(priv, io, dvc); + + /* + * retry to "probe". + * DAI has SSI which is PIO mode only now. + */ + ret = rsnd_dai_call(probe, io, rdai); + } + + return ret; +} + /* * rsnd probe */ @@ -1038,11 +1104,11 @@ static int rsnd_probe(struct platform_device *pdev) } for_each_rsnd_dai(rdai, priv, i) { - ret = rsnd_dai_call(probe, &rdai->playback, rdai); + ret = rsnd_rdai_continuance_probe(priv, rdai, 1); if (ret) goto exit_snd_probe; - ret = rsnd_dai_call(probe, &rdai->capture, rdai); + ret = rsnd_rdai_continuance_probe(priv, rdai, 0); if (ret) goto exit_snd_probe; } diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index cae08b7..346d3dc 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -465,8 +465,23 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, static int rsnd_ssi_dma_remove(struct rsnd_mod *mod, struct rsnd_dai *rdai) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + rsnd_dma_quit(rsnd_mod_to_priv(mod), rsnd_mod_to_dma(mod)); + /* + * fallback to PIO + * + * SSI .probe might be called again. + * see + * rsnd_rdai_continuance_probe() + */ + mod->ops = &rsnd_ssi_pio_ops; + + dev_info(dev, "%s[%d] fallback to PIO mode\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + return 0; } -- 1.7.9.5

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