October 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: pcm_dmaengine: Remove hardcoded PCM formats
by Takashi Iwai 27 Oct '13

27 Oct '13

Use the standard PCM helper function to figure out the sample bytes instead of hardcodec PCM format checks in snd_hwparams_to_dma_slave_config(). The patch also extends the format check for 8 bytes formats although no one should match so far. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- I stumbled upon this, but am not 100% sure whether the current hardcoded formats are the designed limitation... sound/core/pcm_dmaengine.c | 22 +++++++++------------- 1 file changed, 9 insertions(+), 13 deletions(-) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index aa924d9..94d0873 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -63,23 +63,19 @@ int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream, struct dma_slave_config *slave_config) { enum dma_slave_buswidth buswidth; + int bits; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: + bits = snd_pcm_format_physical_width(params_format(params)); + if (bits < 8 || bits > 64) + return -EINVAL; + else if (bits == 8) buswidth = DMA_SLAVE_BUSWIDTH_1_BYTE; - break; - case SNDRV_PCM_FORMAT_S16_LE: + else if (bits == 16) buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; - break; - case SNDRV_PCM_FORMAT_S18_3LE: - case SNDRV_PCM_FORMAT_S20_3LE: - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + else if (bits <= 32) buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; - break; - default: - return -EINVAL; - } + else + buswidth = DMA_SLAVE_BUSWIDTH_8_BYTES; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { slave_config->direction = DMA_MEM_TO_DEV; -- 1.8.4

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[alsa-devel] [PATCH] ASoC: fsl_spdif: fix return value check in fsl_spdif_probe()
by Wei Yongjun 26 Oct '13

26 Oct '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> In case of error, the function platform_get_resource() returns NULL pointer not ERR_PTR(). The IS_ERR() test in the return value check should be replaced with NULL test. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/fsl/fsl_spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 3920c3e..e5bfafe 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1107,9 +1107,9 @@ static int fsl_spdif_probe(struct platform_device *pdev) /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (IS_ERR(res)) { + if (!res) { dev_err(&pdev->dev, "could not determine device resources\n"); - return PTR_ERR(res); + return -ENXIO; } regs = devm_ioremap_resource(&pdev->dev, res);

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[alsa-devel] [PATCH v2] ASoC: wm5110: Add missing routes for AEC Loopback
by Charles Keepax 26 Oct '13

26 Oct '13

Reported-by: Nariman Poushin <nariman.poushin(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index bbd6438..8c91be5 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -983,24 +983,36 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), + { "AEC Loopback", "HPOUT1L", "OUT1L" }, + { "AEC Loopback", "HPOUT1R", "OUT1R" }, { "HPOUT1L", NULL, "OUT1L" }, { "HPOUT1R", NULL, "OUT1R" }, + { "AEC Loopback", "HPOUT2L", "OUT2L" }, + { "AEC Loopback", "HPOUT2R", "OUT2R" }, { "HPOUT2L", NULL, "OUT2L" }, { "HPOUT2R", NULL, "OUT2R" }, + { "AEC Loopback", "HPOUT3L", "OUT3L" }, + { "AEC Loopback", "HPOUT3R", "OUT3R" }, { "HPOUT3L", NULL, "OUT3L" }, { "HPOUT3R", NULL, "OUT3L" }, + { "AEC Loopback", "SPKOUTL", "OUT4L" }, { "SPKOUTLN", NULL, "OUT4L" }, { "SPKOUTLP", NULL, "OUT4L" }, + { "AEC Loopback", "SPKOUTR", "OUT4R" }, { "SPKOUTRN", NULL, "OUT4R" }, { "SPKOUTRP", NULL, "OUT4R" }, + { "AEC Loopback", "SPKDAT1L", "OUT5L" }, + { "AEC Loopback", "SPKDAT1R", "OUT5R" }, { "SPKDAT1L", NULL, "OUT5L" }, { "SPKDAT1R", NULL, "OUT5R" }, + { "AEC Loopback", "SPKDAT2L", "OUT6L" }, + { "AEC Loopback", "SPKDAT2R", "OUT6R" }, { "SPKDAT2L", NULL, "OUT6L" }, { "SPKDAT2R", NULL, "OUT6R" }, -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: wm5110: Add missing routes for AEC Loopback
by Charles Keepax 26 Oct '13

26 Oct '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index bbd6438..8c91be5 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -983,24 +983,36 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), + { "AEC Loopback", "HPOUT1L", "OUT1L" }, + { "AEC Loopback", "HPOUT1R", "OUT1R" }, { "HPOUT1L", NULL, "OUT1L" }, { "HPOUT1R", NULL, "OUT1R" }, + { "AEC Loopback", "HPOUT2L", "OUT2L" }, + { "AEC Loopback", "HPOUT2R", "OUT2R" }, { "HPOUT2L", NULL, "OUT2L" }, { "HPOUT2R", NULL, "OUT2R" }, + { "AEC Loopback", "HPOUT3L", "OUT3L" }, + { "AEC Loopback", "HPOUT3R", "OUT3R" }, { "HPOUT3L", NULL, "OUT3L" }, { "HPOUT3R", NULL, "OUT3L" }, + { "AEC Loopback", "SPKOUTL", "OUT4L" }, { "SPKOUTLN", NULL, "OUT4L" }, { "SPKOUTLP", NULL, "OUT4L" }, + { "AEC Loopback", "SPKOUTR", "OUT4R" }, { "SPKOUTRN", NULL, "OUT4R" }, { "SPKOUTRP", NULL, "OUT4R" }, + { "AEC Loopback", "SPKDAT1L", "OUT5L" }, + { "AEC Loopback", "SPKDAT1R", "OUT5R" }, { "SPKDAT1L", NULL, "OUT5L" }, { "SPKDAT1R", NULL, "OUT5R" }, + { "AEC Loopback", "SPKDAT2L", "OUT6L" }, + { "AEC Loopback", "SPKDAT2R", "OUT6R" }, { "SPKDAT2L", NULL, "OUT6L" }, { "SPKDAT2R", NULL, "OUT6R" }, -- 1.7.2.5

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[alsa-devel] [PATCH] ALSA: hda - Sync EAPD with vmaster on AD1984A Thinkpads
by Takashi Iwai 26 Oct '13

26 Oct '13

As EAPD on NID 0x12 (speaker pin) is used as the master amp on Thinkpads with AD1984A codec, we can hook this to vmaster for saving a bit more power at master mute state. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2aa2f57..87d2e03 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -973,8 +973,11 @@ static void ad1884_fixup_thinkpad(struct hda_codec *codec, { struct ad198x_spec *spec = codec->spec; - if (action == HDA_FIXUP_ACT_PRE_PROBE) + if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.keep_eapd_on = 1; + spec->gen.vmaster_mute.hook = ad_vmaster_eapd_hook; + spec->eapd_nid = 0x12; + } } /* set magic COEFs for dmic */ -- 1.8.4

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[alsa-devel] [PATCH] ALSA: hda - Fix silent headphone on Thinkpads with AD1984A codec
by Takashi Iwai 26 Oct '13

26 Oct '13

AD1984A codec has a couple of pins with EAPD controls, and the generic codec driver tries to turn each of them on/off depending on the pin active state. However, Thinkpads seem to use EAPD of the speaker pin as a master EAPD for controlling the mute of all outputs, including the headphone. This results in the dead headphone output via the headphone plugging because it mutes the speaker and turns off EAPD. The fix is to simply add spec->gen.keep_on_eapd flag. [This is a regression fix on 3.12 where we moved the AD codec parser to the generic parser. 3.11 and earlier didn't show this problem because still static quirks have been used.] Reported-and-tested-by: Vito Caputo <vcaputo(a)gnugeneration.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 0cbdd87..2aa2f57 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -968,6 +968,15 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec, } } +static void ad1884_fixup_thinkpad(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct ad198x_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->gen.keep_eapd_on = 1; +} + /* set magic COEFs for dmic */ static const struct hda_verb ad1884_dmic_init_verbs[] = { {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, @@ -979,6 +988,7 @@ enum { AD1884_FIXUP_AMP_OVERRIDE, AD1884_FIXUP_HP_EAPD, AD1884_FIXUP_DMIC_COEF, + AD1884_FIXUP_THINKPAD, AD1884_FIXUP_HP_TOUCHSMART, }; @@ -997,6 +1007,12 @@ static const struct hda_fixup ad1884_fixups[] = { .type = HDA_FIXUP_VERBS, .v.verbs = ad1884_dmic_init_verbs, }, + [AD1884_FIXUP_THINKPAD] = { + .type = HDA_FIXUP_FUNC, + .v.func = ad1884_fixup_thinkpad, + .chained = true, + .chain_id = AD1884_FIXUP_DMIC_COEF, + }, [AD1884_FIXUP_HP_TOUCHSMART] = { .type = HDA_FIXUP_VERBS, .v.verbs = ad1884_dmic_init_verbs, @@ -1008,7 +1024,7 @@ static const struct hda_fixup ad1884_fixups[] = { static const struct snd_pci_quirk ad1884_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a82, "HP Touchsmart", AD1884_FIXUP_HP_TOUCHSMART), SND_PCI_QUIRK_VENDOR(0x103c, "HP", AD1884_FIXUP_HP_EAPD), - SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1884_FIXUP_DMIC_COEF), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1884_FIXUP_THINKPAD), {} }; -- 1.8.4

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[alsa-devel] [PATCH] ALSA: hda - Add missing initial vmaster hook at build_controls callback
by Takashi Iwai 26 Oct '13

26 Oct '13

The generic parser has a support of vmaster hook, but this is initialized only in the init callback with the check of the presence of the corresponding kctl. However, since kctl is NULL at the very first init callback that is called before build_controls callback, the vmaster hook sync is skipped there. Eventually this leads to the uninitialized state depending on the hook implementation. This patch adds a simple workaround, just calling the sync function explicitly at build_controls callback. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 26ad4f0..b7c89df 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4475,9 +4475,11 @@ int snd_hda_gen_build_controls(struct hda_codec *codec) true, &spec->vmaster_mute.sw_kctl); if (err < 0) return err; - if (spec->vmaster_mute.hook) + if (spec->vmaster_mute.hook) { snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, spec->vmaster_mute_enum); + snd_hda_sync_vmaster_hook(&spec->vmaster_mute); + } } free_kctls(spec); /* no longer needed */ -- 1.8.4

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[alsa-devel] [PATCH RESEND v2 0/4] Cleanups/Bugfixes and DT Prep for CS42L52
by Brian Austin 25 Oct '13

25 Oct '13

This series fixes a bug and cleans up the platform data so the driver can nicely move to DT. Convert platform data initialization to regmap_update_bits in i2c_probe. Add GPIO for CODEC Reset. Add print message for Rev ID. Version 2 splits the regmap_patch bug with the MAX_REGISTER increase. Series based off of a clean for-next branch Brian Austin (4): ASoC: cs42l52: Add platform data for reset gpio ASoC: cs42l52: increase MAX_REGISTER for regmap_register_patch ASoC: cs42l52: convert pdata config to regmap_update_bits ASoC: cs42l52: Add chip rev id message include/sound/cs42l52.h | 2 + sound/soc/codecs/cs42l52.c | 93 +++++++++++++++++++++++++++------------------- sound/soc/codecs/cs42l52.h | 2 +- 3 files changed, 57 insertions(+), 40 deletions(-) -- 1.8.4.rc3

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[alsa-devel] [PATCH] ASoC: dmaengine: Use SNDRV_PCM_STREAM_LAST for array size
by Takashi Iwai 25 Oct '13

25 Oct '13

... to make the meaning more obvious. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 0c469cb..ee07903 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -25,7 +25,7 @@ #include <sound/dmaengine_pcm.h> struct dmaengine_pcm { - struct dma_chan *chan[SNDRV_PCM_STREAM_CAPTURE + 1]; + struct dma_chan *chan[SNDRV_PCM_STREAM_LAST + 1]; const struct snd_dmaengine_pcm_config *config; struct snd_soc_platform platform; unsigned int flags; -- 1.8.4

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[alsa-devel] Enable display power well management in HD-A driver on Baytrail platform (Valleyview SOC)
by Lin, Mengdong 25 Oct '13

25 Oct '13

Hi Takashi, We want to enable display power well management on Batrail platform (Valleyview SOC), similar to what we did on Haswell. Would you please check if our ideas are ok and give some advice? For Valleyview/VLV the display controller has a power well island. When display is off, the display island can be power gated. Unlike Haswell, the VLV HD-A controller is not in the display power well. The controller is in south bridge and shared by multiple codecs, same as the predecessors of Haswell. To assure audio is not disturbed by power well off, the Gfx i915 driver will provide the same API as Haswell: i915_request_power_well & i915_release_power_well So audio driver can indicate whether audio codec needs the power well. In HD-A driver, the display codec can suspend to D3 when there this no active streaming or ioctls which will actually trigger transactions to the display codec on HD-A bus. So our basic idea is to request power before codec resumes to D0 and release power after codec suspends to D3. And there can be two options for this: 1. Overwrite the 'set_power_state' ops for the VLV display codec, and request/release Gfx power well there via the i915 API in this ops. And future VLV specific fixes may also go to this ops. 2. Add two new codec ops like 'pre_resume/post_suspend', and request/release Gfx power well here. Which way do you think is better? Personally I don't want to add new ops and prefer 1st option, and VLV-like processors can use the same ops. And there is an open question: Can we assume that middleware (e.g. pulseaudio or audio flinger) will close the PCM device if there is no need to play audio for power saving? Is this a usual way for middleware to work with ALSA? Now in HD-A driver, once the PCM device is open, the codec will resume to D0. And so the codec will request display power well. If middleware opens PCM device but doesn't play, there will be power waste. We want to avoid this. Thanks Mengdong

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[alsa-devel] ASoC updates for v3.13
by Mark Brown 25 Oct '13

25 Oct '13

The following changes since commit 31d141e3a666269a3b6fcccddb0351caf7454240: Linux 3.12-rc6 (2013-10-19 12:28:15 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13 for you to fetch changes up to 9645083ca5ef365b7b750cf219bb20b61bb925f8: Merge remote-tracking branch 'asoc/topic/wm8962' into asoc-next (2013-10-24 11:24:18 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.13 - Further work on the dmaengine helpers, including support for configuring the parameters for DMA by reading the capabilities of the DMA controller which removes some guesswork and magic numbers fromm drivers. - A refresh of the documentation. - Conversions of many drivers to direct regmap API usage in order to allow the ASoC level register I/O code to be removed, this will hopefully be completed by v3.14. - Support for using async register I/O in DAPM, reducing the time taken to implement power transitions on systems that support it. ---------------------------------------------------------------- Andrew Morton (1): ASoC: atmel-pcm: fix warning Brian Austin (3): ASoC: cs42l73: Add platform data support for cs42l73 codec ASoC: cs42l73: Namespace defines for cs42l73 codec ASoC: cs42l73: Add Device Tree support for CS42L73 Charles Keepax (3): ASoC: samsung: Allow mono in i2s driver ASoC: bells: Add missing route to power up DSP clock ASoC: arizona: Improve handling of setting REFCLK to 0 Chen Gang (1): ASoC: fsl_ssi: let check zero instead of check NO_IRQ Daniel Mack (5): ASoC: ak4104: provide a module device table ASoC: tas5086: move two variables into private struct ASoC: tas5086: move initialization code to own functions ASoC: tas5086: add suspend callback ASoC: davinci-mcasp: add support for suspend and resume Dmitry Eremin-Solenikov (3): ALSA: pxa: slightly refactor reset handling ALSA: ASoC: pxa: fix pxa2xx-ac97 DAI initialization order ALSA: ASoC: pxa: add asoc pm callbacks to pxa audio drivers Fabio Estevam (3): ASoC: mxs: mxs-sgtl5000: Simplify probe function ASoC: fsl: fsl_ssi: Fix simultaneous capture and playback ASoC: fsl: imx-sgtl5000: Add .remove back Felipe Pena (1): ASoC: fsl: Fix memory leak in imx-audmux.c Hebbar, Gururaja (2): ASoC: davinci: Add support for AM33xx SoC Audio ASoC: davinci-evm: Add device tree binding Jarkko Nikula (1): ASoC: pcm: Remove extra spaces from dev_ prints Jean-Francois Moine (4): ASoC: kirkwood: fix loss of external clock at probe time ASoc: kirkwood: Extend the min and max number of bytes per period ASoC: kirkwood: prefer external clock over internal clock ASoC: kirkwood: add S/PDIF support Jyri Sarha (8): ASoC: davinci-evm: Move sysclk logic away from evm_hw_params ASoC: tlv320aic3x: Add regulators to DT bindings document ASoC: tlv320aic3x: Add codec pins to DT bindings document ASoC: davinci-mcasp: Add location for data port registers to DT ASoC: davinci-mcasp: Extract DMA channels directly from DT ASoC: davinci-mcasp: Change compatible property model to more accurate ASoC: davinci-mcasp: Improve DT bindings document ASoC: davinci-mcasp: Remove last reference to num-serializer in DT doc Kuninori Morimoto (14): regmap: add regmap_field_update_bits() ASoC: snd_soc_codec includes snd_soc_component ASoC: add .of_xlate_dai_name on snd_soc_component_driver regmap: Add regmap_fields APIs ASoC: rsnd: gen: rsnd_gen_ops cares .probe and .remove ASoC: rcar: remove unnecessary mach/clock.h ASoC: rsnd: remove rsnd_priv_read/write/bset() ASoC: rsnd: use regmap instead of original register mapping method ASoC: rcar: fixup rsnd_platform_call() return value ASoC: rcar: add ID check on rsnd_dai_get() ASoC: add snd_soc_of_get_dai_name() default of_xlate ASoC: simple-card: un-implemented set_fmt is not error ASoC: rcar: add rsnd_scu_hpbif_is_enable() ASoC: rcar: remove RSND_SSI_CLK_FROM_ADG Lars-Peter Clausen (15): ASoC: Remove support for reg_access_defaults ASoC: Remove snd_soc_bulk_write_raw() ASoC: Remove reg_def_copy ASoC: Remove 'reg_size' field from snd_soc_codec struct ASoC: Remove infrastructure for supporting multiple cache types ASoC: adau1373: Convert to direct regmap usage ASoC: adau1373: Remove ADAU1373_PLL_CTRL7 register definition ASoC: adau1373: Issue soft reset on probe ASoC: adav80x: Convert to direct regmap usage ASoC: dapm: Ignore VMID widgets for target bias ASoC: dapm: Add support for virtual mixer controls ASoC: twl6040: Use virtual DAPM mixer controls ASoC: twl4030: Use virtual DAPM mixer controls ASoC: dmaengine-pcm: Add support for querying DMA capabilities ASoC: dmaengine-pcm: Provide default config Liam Girdwood (10): ASoC: trace: Make sure trace header doesnt depend on any headers ASoC: core utils: Dont set DMA params for BE substreams ASoC: rt5640: Provide more useful hw_params error reasons. ASoC: rt5640: Add ACPI probing support. ASoC: core: Add API for configuration of DAI BCLK ratio ASoC: Docs: Update codec documentation ASoC: Docs: Platform update ASoC: Docs: update DAPM ASoC: Docs: Machine update ASoC: Docs: Add documentation for Dynamic PCM Mark Brown (122): ASoC: cq93vc: Don't use control data for core driver data ASoC: wm8400: Use supplies to manage input power ASoC: wm8400: Use regmap for I/O ASoC: core: Implement devm_snd_soc_register_component() ASoC: core: Add devm_snd_soc_register_card() Merge remote-tracking branch 'asoc/topic/devm' into asoc-samsung ASoC: smdk_wm8994: Use devm_snd_soc_unregister_card() ASoC: samsung-i2s: Use devm_snd_soc_register_component() Merge remote-tracking branch 'asoc/topic/component' into asoc-core ASoC: 88pm60x: Don't use control data for i2c ASoC: 88pm860x: Use regmap for I/O ASoC: ab8500: Downgrade noisy log message ASoC: ab8500: Use ASoC I/O functions ASoC: ab8500: Convert register I/O to regmap ASoC: max98088: Fix indentation ASoC: max98088: Use table based control init ASoC: max98088: Convert to direct regmap API usage ASoC: ab8500: Explicitly set I/O up ASoC: max98095: Remove custom hw_write() implementation ASoC: max90895: Convert to table based control init ASoC: max98095: Convert to direct regmap API usage ASoC: tlv320aic23: Remove #defines for I2C ASoC: tlv320aic23: Convert to table based control init ASoC: tlv320aic3x: Remove nonsense comment for register cache ASoC: tlv320aic3x: Move resource acquisition to I2C probe ASoC: tlv320aic3x: Move to table based control init ASoC: tlv320aic3x: Move to table based DAPM init ASoC: tlv320aic3x: Don't reference cache datastructure directly ASoC: tlv320aic3x: Convert to direct regmap API usage ASoC: max98088: Set max_register ASoC: max9850: Convert to direct regmap API usage ASoC: cq93vc: Use table based control registration ASoC: cq93vc: Use core I/O functions ASoC: cq93vc: Use regmap for I/O Merge tag 'regmap-fields' of git://git.kernel.org/.../broonie/regmap into asoc-rcar ASoC: tlv320aic23: Convert to direct regmap API usage ASoC: tlv320aic26: Convert to table based control init ASoC: tlv320aic26: Use snd_soc_update_bits() ASoC: tlv320aic26: Convert to direct regmap API usage mfd: mc13xxx: Don't require lock for simple register I/O mfd: mc13xxx: Move SPI erratum workaround into SPI I/O function ASoC: mc13783: Use regmap directly from ASoC ASoC: sn95031: Convert to regmap ASoC: tlv320aic32x4: Move GPIO acquisition to I2C probe ASoC: tlv320aic32x4: Convert to direct regmap API usage Merge remote-tracking branch 'asoc/fix/fsl' into asoc-devm ASoC: si476x: Remove custom register I/O implementation Merge remote-tracking branch 'asoc/fix/fsl' into asoc-fsl Merge remote-tracking branch 'asoc/topic/dapm' into asoc-twl6040 regmap: Cache async work structures regmap: Don't generate gather writes for single register raw writes regmap: Simplify the initiation of async I/O regmap: Provide asynchronous write and update bits operations Merge tag 'regmap-async-reg' of git://git.kernel.org/.../broonie/regmap into asoc-dapm ASoC: dapm: Use async I/O for DAPM sequences ASoC: dapm: Run clock and regulator events separately to other supplies ASoC: wm8962: Move interrupt initalisation to probe() ASoC: wm8962: Move register initialisation to I2C probe() ASoC: cirrus: Enable compile test builds ASoC: rt5640: Power down LDO while suspended ASoC: rt5640: Don't go to standby on resume ASoC: dai: Provide interface for setting DMA data at probe time ASoC: samsung: Initialise DMA data at device probe time ASoC: si476x: Fix locking of core ASoC: ep93xx: Open code dma channel request ASoC: tegra: Remove redundant initialisation of compat_filter_fn ASoC: dmaengine: Support custom channel names ASoC: dmaengine: Use filter_data rather than dma_data for compat requests Merge remote-tracking branch 'asoc/fix/fsl' into asoc-linus Merge remote-tracking branch 'asoc/fix/omap' into asoc-linus Merge remote-tracking branch 'asoc/fix/pcm1681' into asoc-linus Merge remote-tracking branch 'asoc/fix/pcm1792a' into asoc-linus Merge remote-tracking branch 'asoc/fix/rcar' into asoc-linus Merge remote-tracking branch 'asoc/fix/tlv320aic3x' into asoc-linus Merge remote-tracking branch 'asoc/topic/88pm860x' into asoc-next Merge remote-tracking branch 'asoc/topic/ab8500' into asoc-next Merge remote-tracking branch 'asoc/topic/adau1373' into asoc-next Merge remote-tracking branch 'asoc/topic/adav80x' into asoc-next Merge remote-tracking branch 'asoc/topic/ak4104' into asoc-next Merge remote-tracking branch 'asoc/topic/ak4642' into asoc-next Merge remote-tracking branch 'asoc/topic/arizona' into asoc-next Merge remote-tracking branch 'asoc/topic/atmel' into asoc-next Merge remote-tracking branch 'asoc/topic/bclk' into asoc-next Merge remote-tracking branch 'asoc/topic/component' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/cq93vc' into asoc-next Merge remote-tracking branch 'asoc/topic/cs4271' into asoc-next Merge remote-tracking branch 'asoc/topic/cs42l73' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/davinci' into asoc-next Merge remote-tracking branch 'asoc/topic/devm' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/doc' into asoc-next Merge remote-tracking branch 'asoc/topic/ep93xx' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl' into asoc-next Merge remote-tracking branch 'asoc/topic/kirkwood' into asoc-next Merge remote-tracking branch 'asoc/topic/max98088' into asoc-next Merge remote-tracking branch 'asoc/topic/max98095' into asoc-next Merge remote-tracking branch 'asoc/topic/max9850' into asoc-next Merge remote-tracking branch 'asoc/topic/mc13783' into asoc-next Merge remote-tracking branch 'asoc/topic/mxs' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm1681' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm1792a' into asoc-next Merge remote-tracking branch 'asoc/topic/pxa' into asoc-next Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5640' into asoc-next Merge remote-tracking branch 'asoc/topic/samsung' into asoc-next Merge remote-tracking branch 'asoc/topic/si476x' into asoc-next Merge remote-tracking branch 'asoc/topic/simple' into asoc-next Merge remote-tracking branch 'asoc/topic/sn95031' into asoc-next Merge remote-tracking branch 'asoc/topic/spear' into asoc-next Merge remote-tracking branch 'asoc/topic/tas5086' into asoc-next Merge remote-tracking branch 'asoc/topic/tegra' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320aic23' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320aic26' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320aic32x4' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320aic3x' into asoc-next Merge remote-tracking branch 'asoc/topic/tpa6130a2' into asoc-next Merge remote-tracking branch 'asoc/topic/twl4030' into asoc-next Merge remote-tracking branch 'asoc/topic/twl6040' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8400' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8962' into asoc-next Markus Pargmann (3): ASoC: snd_soc_dai_ops trigger function description ASoC: mxs-saif: Store saif state ASoC: mxs-saif: Handle errors in trigger function Nicolin Chen (1): ASoC: fsl: Add missing pm to current machine drivers Peter Ujfalusi (1): ASoC: davinci-mcasp: Remove redundant num-serializer DT parameter Russell King (1): ASoC: kirkwood-dma: remove IEC958_SUBFRAME formats Sachin Kamat (25): ASoC: fsl_ssi: Staticize local symbols ASoC: rt5640: Staticize hp_amp_power_on ASoC: twl6040: Remove redundant semicolon ASoC: ak4642: Remove redundant break ASoC: fsl_spdif: Remove redundant semicolon ASoC: fsl_ssi: Remove redundant dev_set_drvdata ASoC: SPEAr spdif_in: Use devm_snd_soc_register_component ASoC: SPEAr spdif_out: Use devm_snd_soc_register_component ASoC: SPEAr spdif_out: Remove redundant variable ASoC: imx-sgtl5000: Use devm_snd_soc_register_card ASoC: imx-spdif: Use devm_snd_soc_register_card ASoC: imx-wm8962: Use devm_snd_soc_register_card ASoC: mfld: Use devm_snd_soc_register_card ASoC: omap-twl4030: Use devm_snd_soc_register_card ASoC: fsl_spdif: Use devm_snd_soc_register_component ASoC: mxs-saif: Use devm_snd_soc_register_component ASoC: omap-mcpdm: Use devm_snd_soc_register_component ASoC: mmp-sspa: Use devm_snd_soc_register_component ASoC: smdk_wm8994: Add .pm to struct smdk_audio_driver ASoC: cs4271: Include linux/of.h header ASoC: pcm1681: Include linux/of.h header ASoC: pcm1792a: Include linux/of.h header ASoC: tas5086: Include linux/of.h header ASoC: tlv320aic3x: Include linux/of.h header ASoC: atmel: Include linux/of.h header Sebastian Reichel (1): ASoC: tpa6130a2: Add device tree support Steffen Trumtrar (2): ASoC: mc13783: add mixer controls ASoC: mc13783: add more DAPM routes Stephen Warren (1): ASoC: tegra: support new register layouts in Tegra124 Thierry Reding (1): ASoC: rt5640: Omit ACPI match table only if !ACPI Vinod Koul (1): ASoC: dont call dapm_sync while reporting jack always Vladimir Murzin (1): ASoC: kirkwood: fix compilation warning in kirkwood_dma_open Wolfram Sang (2): ASoC: atmel: don't use devm_pinctrl_get_select_default() in probe ASoC: fsl: remove leftover release_mem_region .../devicetree/bindings/sound/cs42l73.txt | 22 + .../bindings/sound/davinci-evm-audio.txt | 42 ++ .../bindings/sound/davinci-mcasp-audio.txt | 41 +- .../devicetree/bindings/sound/tlv320aic3x.txt | 26 + .../devicetree/bindings/sound/tpa6130a2.txt | 27 + Documentation/sound/alsa/soc/DPCM.txt | 380 +++++++++++++ Documentation/sound/alsa/soc/codec.txt | 46 +- Documentation/sound/alsa/soc/dapm.txt | 71 ++- Documentation/sound/alsa/soc/machine.txt | 6 +- Documentation/sound/alsa/soc/platform.txt | 19 +- drivers/base/regmap/internal.h | 8 +- drivers/base/regmap/regcache.c | 3 +- drivers/base/regmap/regmap.c | 294 ++++++++-- drivers/mfd/mc13xxx-core.c | 5 - drivers/mfd/mc13xxx-spi.c | 5 + include/linux/mfd/mc13xxx.h | 7 + include/linux/platform_data/davinci_asp.h | 2 + include/linux/regmap.h | 44 ++ include/sound/cs42l73.h | 22 + include/sound/dmaengine_pcm.h | 8 + include/sound/rcar_snd.h | 2 +- include/sound/soc-dai.h | 17 + include/sound/soc-dapm.h | 4 + include/sound/soc.h | 100 ++-- include/trace/events/asoc.h | 1 + sound/arm/pxa2xx-ac97-lib.c | 27 +- sound/soc/Makefile | 2 +- sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/atmel/atmel_wm8904.c | 8 - sound/soc/atmel/sam9g20_wm8731.c | 1 + sound/soc/cirrus/Kconfig | 2 +- sound/soc/cirrus/ep93xx-pcm.c | 13 + sound/soc/codecs/88pm860x-codec.c | 75 +-- sound/soc/codecs/88pm860x-codec.h | 117 ++-- sound/soc/codecs/ab8500-codec.c | 78 ++- sound/soc/codecs/adau1373.c | 298 ++++++++-- sound/soc/codecs/adav80x.c | 147 +++-- sound/soc/codecs/ak4104.c | 11 +- sound/soc/codecs/ak4642.c | 2 - sound/soc/codecs/arizona.c | 23 +- sound/soc/codecs/cq93vc.c | 46 +- sound/soc/codecs/cs4271.c | 1 + sound/soc/codecs/cs42l73.c | 114 ++-- sound/soc/codecs/cs42l73.h | 105 ++-- sound/soc/codecs/max98088.c | 624 +++++++++------------ sound/soc/codecs/max98095.c | 466 ++++++--------- sound/soc/codecs/max9850.c | 39 +- sound/soc/codecs/mc13783.c | 135 +++-- sound/soc/codecs/pcm1681.c | 3 +- sound/soc/codecs/pcm1792a.c | 3 +- sound/soc/codecs/rt5640.c | 24 +- sound/soc/codecs/si476x.c | 64 +-- sound/soc/codecs/sn95031.c | 35 +- sound/soc/codecs/tas5086.c | 171 ++++-- sound/soc/codecs/tlv320aic23.c | 84 ++- sound/soc/codecs/tlv320aic26.c | 139 +---- sound/soc/codecs/tlv320aic26.h | 5 +- sound/soc/codecs/tlv320aic32x4.c | 101 +--- sound/soc/codecs/tlv320aic3x.c | 238 ++++---- sound/soc/codecs/tpa6130a2.c | 32 +- sound/soc/codecs/twl4030.c | 80 ++- sound/soc/codecs/twl6040.c | 26 +- sound/soc/codecs/wm8400.c | 95 +--- sound/soc/codecs/wm8962.c | 218 ++++--- sound/soc/davinci/Kconfig | 18 +- sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 188 ++++++- sound/soc/davinci/davinci-mcasp.c | 169 ++++-- sound/soc/davinci/davinci-mcasp.h | 12 + sound/soc/fsl/fsl_spdif.c | 18 +- sound/soc/fsl/fsl_ssi.c | 22 +- sound/soc/fsl/imx-audmux.c | 9 +- sound/soc/fsl/imx-mc13783.c | 3 +- sound/soc/fsl/imx-pcm-dma.c | 4 +- sound/soc/fsl/imx-sgtl5000.c | 4 +- sound/soc/fsl/imx-spdif.c | 4 +- sound/soc/fsl/imx-ssi.c | 26 +- sound/soc/fsl/imx-ssi.h | 2 + sound/soc/fsl/imx-wm8962.c | 4 +- sound/soc/generic/simple-card.c | 5 + sound/soc/kirkwood/kirkwood-dma.c | 6 +- sound/soc/kirkwood/kirkwood-i2s.c | 106 +++- sound/soc/kirkwood/kirkwood-openrd.c | 2 +- sound/soc/kirkwood/kirkwood-t5325.c | 2 +- sound/soc/kirkwood/kirkwood.h | 4 +- sound/soc/mid-x86/mfld_machine.c | 10 +- sound/soc/mxs/mxs-saif.c | 42 +- sound/soc/mxs/mxs-saif.h | 5 + sound/soc/mxs/mxs-sgtl5000.c | 20 +- sound/soc/omap/Kconfig | 4 +- sound/soc/omap/omap-mcpdm.c | 12 +- sound/soc/omap/omap-twl4030.c | 5 +- sound/soc/pxa/brownstone.c | 1 + sound/soc/pxa/corgi.c | 1 + sound/soc/pxa/e740_wm9705.c | 1 + sound/soc/pxa/e750_wm9705.c | 1 + sound/soc/pxa/e800_wm9712.c | 1 + sound/soc/pxa/imote2.c | 1 + sound/soc/pxa/mioa701_wm9713.c | 1 + sound/soc/pxa/mmp-sspa.c | 5 +- sound/soc/pxa/palm27x.c | 1 + sound/soc/pxa/poodle.c | 1 + sound/soc/pxa/pxa2xx-ac97.c | 56 +- sound/soc/pxa/tosa.c | 1 + sound/soc/pxa/ttc-dkb.c | 1 + sound/soc/samsung/bells.c | 1 + sound/soc/samsung/i2s.c | 23 +- sound/soc/samsung/smdk_wm8994.c | 14 +- sound/soc/sh/rcar/adg.c | 11 +- sound/soc/sh/rcar/core.c | 50 +- sound/soc/sh/rcar/gen.c | 261 +++++---- sound/soc/sh/rcar/rsnd.h | 9 +- sound/soc/sh/rcar/scu.c | 12 +- sound/soc/sh/rcar/ssi.c | 52 +- sound/soc/soc-cache.c | 263 ++------- sound/soc/soc-core.c | 338 ++++++----- sound/soc/soc-dapm.c | 120 ++-- sound/soc/soc-devres.c | 86 +++ sound/soc/soc-generic-dmaengine-pcm.c | 101 +++- sound/soc/soc-io.c | 26 - sound/soc/soc-jack.c | 7 +- sound/soc/soc-pcm.c | 8 +- sound/soc/soc-utils.c | 6 +- sound/soc/spear/spdif_in.c | 12 +- sound/soc/spear/spdif_out.c | 14 +- sound/soc/tegra/tegra30_ahub.c | 115 +++- sound/soc/tegra/tegra30_ahub.h | 38 +- sound/soc/tegra/tegra30_i2s.c | 51 +- sound/soc/tegra/tegra30_i2s.h | 7 + sound/soc/tegra/tegra_asoc_utils.c | 2 + sound/soc/tegra/tegra_asoc_utils.h | 1 + sound/soc/tegra/tegra_pcm.c | 1 - 132 files changed, 4222 insertions(+), 3042 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cs42l73.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/tpa6130a2.txt create mode 100644 Documentation/sound/alsa/soc/DPCM.txt create mode 100644 include/sound/cs42l73.h create mode 100644 sound/soc/soc-devres.c

2 1

[alsa-devel] Is that reasonable to support pinctrl PM in ASoC core?
by Nicolin Chen 25 Oct '13

25 Oct '13

Hi all, It's quite popular that more drivers are using pinctrl PM, for example: (Documentation/devicetree/bindings/arm/primecell.txt). Just like what runtime PM does, it would de-active and en-active pin group depending on whether it's being used or not. And I think this pinctrl PM might be also beneficial to cpu dai drivers since they might have actual pins, and they can hypnotize/wake them up along with runtime PM: @@ -183,6 +183,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; int ret = 0; + pinctrl_pm_select_default_state(cpu_dai->dev); pm_runtime_get_sync(cpu_dai->dev); pm_runtime_get_sync(codec_dai->dev); pm_runtime_get_sync(platform->dev); As pinctrl PM is also reference counted and would return 0 if there is no pinctrl settings, this would not break current ASoC subsystem and any cpu dai driver. So I just want to ask if it's reasonable to add it. Or another one: is there anything similar to sleep pins? Although I've searched the whole ./sound directory, it seems no drivers under it is using this pinctrl PM. Best regards, Nicolin Chen

3 5

[alsa-devel] [PATCH] ASoC: Samsung: I2S: fix return value check in i2s_alloc_dai()
by Wei Yongjun 25 Oct '13

25 Oct '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> In case of error, the function platform_device_alloc() returns NULL pointer not ERR_PTR(). The IS_ERR() test in the return value check should be replaced with NULL test. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/samsung/i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index b302f3b..b721443 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1073,7 +1073,7 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) dev_set_drvdata(&i2s->pdev->dev, i2s); } else { /* Create a new platform_device for Secondary */ i2s->pdev = platform_device_alloc("samsung-i2s-sec", -1); - if (IS_ERR(i2s->pdev)) + if (!i2s->pdev) return NULL; i2s->pdev->dev.parent = &pdev->dev;

2 1

[alsa-devel] [PATCH RESEND 1/2] ASoC: cs42l52: Add support for platform data reset GPIO
by Brian Austin 25 Oct '13

25 Oct '13

This patch adds support for grabbing a reset GPIO from platform data and resetting the codec Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- include/sound/cs42l52.h | 3 +++ sound/soc/codecs/cs42l52.c | 25 ++++++++++++++++++++----- sound/soc/codecs/cs42l52.h | 2 +- 3 files changed, 24 insertions(+), 6 deletions(-) diff --git a/include/sound/cs42l52.h b/include/sound/cs42l52.h index a902e51..853df80 100644 --- a/include/sound/cs42l52.h +++ b/include/sound/cs42l52.h @@ -33,6 +33,9 @@ struct cs42l52_platform_data { /* MCLK Freq */ unsigned int mclk_freq; + + /* !RST GPIO */ + unsigned int reset_gpio; }; #endif /* __CS42L52_H */ diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index be2ba1b..6782d4875 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -17,6 +17,7 @@ #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/gpio.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/input.h> @@ -1205,6 +1206,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { struct cs42l52_private *cs42l52; + struct cs42l52_platform_data *pdata = dev_get_platdata(&i2c_client->dev); int ret; unsigned int devid = 0; unsigned int reg; @@ -1222,12 +1224,23 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, return ret; } + if (pdata) + cs42l52->pdata = *pdata; + + if (cs42l52->pdata.reset_gpio) { + ret = gpio_request_one(cs42l52->pdata.reset_gpio, + GPIOF_OUT_INIT_HIGH, "CS42L52 /RST"); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to request /RST %d: %d\n", + cs42l52->pdata.reset_gpio, ret); + return ret; + } + gpio_set_value_cansleep(cs42l52->pdata.reset_gpio, 0); + gpio_set_value_cansleep(cs42l52->pdata.reset_gpio, 1); + } + i2c_set_clientdata(i2c_client, cs42l52); - - if (dev_get_platdata(&i2c_client->dev)) - memcpy(&cs42l52->pdata, dev_get_platdata(&i2c_client->dev), - sizeof(cs42l52->pdata)); - + ret = regmap_register_patch(cs42l52->regmap, cs42l52_threshold_patch, ARRAY_SIZE(cs42l52_threshold_patch)); if (ret != 0) @@ -1243,6 +1256,8 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, devid, CS42L52_CHIP_ID); return ret; } + dev_info(&i2c_client->dev, + "Cirrus Logic CS42L52, Revision: %02X\n", reg & 0xFF); regcache_cache_only(cs42l52->regmap, true); diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h index 4277012..1a9412d 100644 --- a/sound/soc/codecs/cs42l52.h +++ b/sound/soc/codecs/cs42l52.h @@ -269,6 +269,6 @@ #define CS42L52_FIX_BITS1 0x3E #define CS42L52_FIX_BITS2 0x47 -#define CS42L52_MAX_REGISTER 0x34 +#define CS42L52_MAX_REGISTER 0x47 #endif -- 1.8.4.rc3

2 3

[alsa-devel] [PATCH] ALSA: hda - fix DigEn for Valleyview2 display codec
by mengdong.lin＠intel.com 25 Oct '13

25 Oct '13

From: Mengdong Lin <mengdong.lin(a)intel.com> On some Baytrail/Vallleyview platforms, DigEn is disabled on convertors. Thus audio data is blocked from passing through the node and no sound can be heard over HDMI/DP. This patch fixes the covertor's DigEn when setting up a stream. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e332db8..f619732 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -45,6 +45,7 @@ module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); #define is_haswell(codec) ((codec)->vendor_id == 0x80862807) +#define is_valleyview(codec) ((codec)->vendor_id == 0x80862882) struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; @@ -1061,6 +1062,20 @@ static void haswell_verify_D0(struct hda_codec *codec, } } +static void vlv_verify_digen(struct hda_codec *codec, hda_nid_t cvt_nid) +{ + unsigned int digi1; + + digi1 = snd_hda_codec_read(codec, cvt_nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0); + + if (!(digi1 & AC_DIG1_ENABLE)) + snd_hda_codec_write_cache(codec, cvt_nid, 0, + AC_VERB_SET_DIGI_CONVERT_1, + (digi1 | AC_DIG1_ENABLE) & 0xff); + +} + /* * Callbacks */ @@ -1078,6 +1093,9 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, if (is_haswell(codec)) haswell_verify_D0(codec, cvt_nid, pin_nid); + if (is_valleyview(codec)) + vlv_verify_digen(codec, cvt_nid); + if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) { pinctl = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: kirkwood: Fix compile error due to patch 'add S/PDIF support'
by Jean-Francois Moine 25 Oct '13

25 Oct '13

This patch fixes the compilation error of kirkwood-i2s.c introduced by the commit 75b9b65ee5a80e99e 'ASoC: kirkwood: add S/PDIF support'. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- Mark, sorry for this error. This patch applies on your repository broonie/sound.git/ topic/kirkwood. I may give you an other patch to apply with the commit removed. --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 9ec38d1..d34d917 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -568,7 +568,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } else { dev_info(&pdev->dev, "found external clock\n"); clk_prepare_enable(priv->extclk); - soc_dai = &kirkwood_i2s_dai_extclk; + soc_dai = kirkwood_i2s_dai_extclk; } } -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] Variable Delay through ALSA driver
by Brandon Yates 25 Oct '13

25 Oct '13

I have an application using the ALSA API that writes to playback device and simultaneously reads from capture device. My processing loop is : 1)Read from microphone (blocking) 2)Process Data 3)Write to speaker I am measuring the delay from the time I play a series of frames to the speaker until I detect the echo of this coming back in through the microphone. This delay seems to be changing every time I initialize the devices with ALSA. I have measured in the range of 60-160ms delay. I know things like period size effect delay, but using the exact same software and hardware configuration I would expect this to be constant across runs. What are some parameters that can help me make the delay deterministic? Thanks, Brandon

3 2

[alsa-devel] [PATCH] ALSA: hda - hdmi: Re-setup pin and infoframe on plug-in on all codecs
by Anssi Hannula 25 Oct '13

25 Oct '13

hdmi_setup_audio_infoframe() does not set up pin and infoframe if there is no connected sink. If a sink is connected while audio playback is already in progress, the pin and infoframe will not be properly set up, causing no audio or wrongly mapped audio. On Intel Haswell codecs the hdmi_setup_audio_infoframe() is already called again from hdmi_present_sense() when an ELD appears because transcoder:port mapping may have changed. Make the call non-Haswell-specific so that audio will be properly set up if the playback was started before a sink was connected. Tested on non-Haswell Intel HDMI codec by plugging sink in during playback. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- sound/pci/hda/patch_hdmi.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 804adb8..a1fe4bb 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1462,11 +1462,12 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) pin_eld->eld_size = eld->eld_size; pin_eld->info = eld->info; - /* Haswell-specific workaround: re-setup when the transcoder is - * changed during the stream playback + /* + * Re-setup pin and infoframe. This is needed e.g. when + * - sink is first plugged-in (infoframe is not set up if !monitor_present) + * - transcoder can change during stream playback on Haswell */ - if (is_haswell(codec) && - eld->eld_valid && !old_eld_valid && per_pin->setup) + if (eld->eld_valid && !old_eld_valid && per_pin->setup) hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); } -- 1.8.1.5

2 1

[alsa-devel] Bug report Realtek ALC887-VD HDA Intel tone://15000
by Mihai Moise 24 Oct '13

24 Oct '13

I am writing to report a bug with ALSA that affects XMMS and Audacious. I have a Realtek ALC887-VD HDA Intel. I tested with kernel 3.12-rc4 and alsa-libs 1.0.27.2.10.gc1fbd and alsa-oss-1.0.25.2.g39df1 library. I also compiled and installed alsa-utils-1.0.27.2.6.gf1e99 which made me realize I had to move the libraries from /usr/lib to /usr/lib64. Playing tone://15000 in Audacious or XMMS with the ALSA driver without resampling sounds bad (in XMMS there is no resampling option). How to reproduce the bug: Play in XMMS tone://10 In Audacious you would have to lower the volume by 6 dB to get the same sound volume as in XMMS because it pre-amplifies sounds. Lower the PC-volume until you stop hearing clicks. To hear the clicks you need headphones and to raise the real-life amplifier volume all the way up. For me it is, Master: 91% PCM: 100% Front: 60% Select the ALSA driver and no resampling in Audacious (I think Audacious comes with no resampling as the default setting which is why this is so important). Play tone://15000 It is supposed to sound like this: perl -e '$|++;binmode(STDOUT);$_ = 0;while(1){print pack("v",32767.0*sin(($_&(~1))*3.14159265*15000.0/48000.0));++$_;}' | aplay -f dat but it sounds like this: perl -e '$|++;binmode(STDOUT);$_ = 0;while(1){print pack("v",32767.0*sin(($_&(~1))*3.14159265*15000.0/44100.0));++$_;}' | aplay -f cd This is how it sounds like if the OSS driver is invoked: perl -e '$|++;binmode(STDOUT);$_ = 0;while(1){print pack("v",32767.0*sin(($_&(~1))*3.14159265*15000.0/44100.0));++$_;}' | sox -V4 -b 16 -c 2 -r 44100 -e signed-integer -t raw - -r 44100 -t oss -d On my computer, the first and third command lines sound good, the second command line sounds bad. When using the OSS driver in XMMS it sounds like the first and third command lines. To go into more details: when using the ALSA driver in XMMS, for frequencies higher than tone://10000 the main frequency gets swamped by lower harmonics, so the frequency heard seems to drop the higher it increases. At tone://15000 what is heard is a louder lower harmonic. My suggestion is: could there be a way to force ALSA to resample the sound when using an ALC887-VD and/or a Realtek chipset and/or HDA Intel...

1 3

[alsa-devel] [PATCH] ALSA: hda - hdmi: Fix channel map switch not taking effect
by Anssi Hannula 24 Oct '13

24 Oct '13

Currently hdmi_setup_audio_infoframe() reprograms the HDA channel mapping only when the infoframe is not up-to-date or the non-PCM flag has changed. However, when just the channel map has been changed, the infoframe may still be up-to-date and non-PCM flag may not have changed, so the new channel map is not actually programmed into the HDA codec. Notably, this failing case is also always triggered when the device is already in a prepared state and a new channel map is configured while changing only the channel positions (for example, plain "speaker-test -c2 -m FR,FL"). Fix that by keeping track of the last programmed channel map and making hdmi_setup_audio_infoframe() always update the hardware mapping if the channel map has changed. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: <stable(a)vger.kernel.org> --- BTW, is there some mechanism preventing hdmi_chmap_ctl_put() to be called at the same time with generic_hdmi_playback_pcm_prepare() that I didn't see with a quick look? Both of them call hdmi_setup_audio_infoframe() which is not thread-safe. sound/pci/hda/patch_hdmi.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7ea0245..09f38a1 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -74,6 +74,8 @@ struct hdmi_spec_per_pin { bool non_pcm; bool chmap_set; /* channel-map override by ALSA API? */ unsigned char chmap[8]; /* ALSA API channel-map */ + bool last_chmap_set; /* channel-map override programmed? */ + unsigned char last_chmap[8]; /* last programmed channel-map */ char pcm_name[8]; /* filled in build_pcm callbacks */ }; @@ -955,15 +957,21 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, ai.bytes, sizeof(ai)); hdmi_start_infoframe_trans(codec, pin_nid); } else { - /* For non-pcm audio switch, setup new channel mapping - * accordingly */ - if (per_pin->non_pcm != non_pcm) + /* For non-pcm audio switch or channel-map switch, + * setup new channel mapping accordingly */ + if (per_pin->non_pcm != non_pcm + || per_pin->chmap_set != per_pin->last_chmap_set + || (per_pin->chmap_set && memcmp(per_pin->chmap, per_pin->last_chmap, + ARRAY_SIZE(per_pin->chmap)) != 0)) hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, channels, per_pin->chmap, per_pin->chmap_set); } per_pin->non_pcm = non_pcm; + per_pin->last_chmap_set = per_pin->chmap_set; + if (per_pin->chmap_set) + memcpy(per_pin->last_chmap, per_pin->chmap, ARRAY_SIZE(per_pin->chmap)); } -- 1.8.1.5

2 12

[alsa-devel] [PATCH 1/2] ASoC: cs42l52: Add support for platform data reset GPIO
by Brian Austin 24 Oct '13

24 Oct '13

This patch adds support for grabbing a reset GPIO from platform data and resetting the codec Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- include/sound/cs42l52.h | 3 +++ sound/soc/codecs/cs42l52.c | 25 ++++++++++++++++++++----- sound/soc/codecs/cs42l52.h | 2 +- 3 files changed, 24 insertions(+), 6 deletions(-) diff --git a/include/sound/cs42l52.h b/include/sound/cs42l52.h index a902e51..853df80 100644 --- a/include/sound/cs42l52.h +++ b/include/sound/cs42l52.h @@ -33,6 +33,9 @@ struct cs42l52_platform_data { /* MCLK Freq */ unsigned int mclk_freq; + + /* !RST GPIO */ + unsigned int reset_gpio; }; #endif /* __CS42L52_H */ diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index be2ba1b..6782d4875 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -17,6 +17,7 @@ #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/gpio.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/input.h> @@ -1205,6 +1206,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { struct cs42l52_private *cs42l52; + struct cs42l52_platform_data *pdata = dev_get_platdata(&i2c_client->dev); int ret; unsigned int devid = 0; unsigned int reg; @@ -1222,12 +1224,23 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, return ret; } + if (pdata) + cs42l52->pdata = *pdata; + + if ( cs42l52->pdata.reset_gpio) { + ret = gpio_request_one(cs42l52->pdata.reset_gpio, + GPIOF_OUT_INIT_HIGH, "CS42L52 /RST"); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to request /RST %d: %d\n", + cs42l52->pdata.reset_gpio, ret); + return ret; + } + gpio_set_value_cansleep(cs42l52->pdata.reset_gpio, 0); + gpio_set_value_cansleep(cs42l52->pdata.reset_gpio, 1); + } + i2c_set_clientdata(i2c_client, cs42l52); - - if (dev_get_platdata(&i2c_client->dev)) - memcpy(&cs42l52->pdata, dev_get_platdata(&i2c_client->dev), - sizeof(cs42l52->pdata)); - + ret = regmap_register_patch(cs42l52->regmap, cs42l52_threshold_patch, ARRAY_SIZE(cs42l52_threshold_patch)); if (ret != 0) @@ -1243,6 +1256,8 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, devid, CS42L52_CHIP_ID); return ret; } + dev_info(&i2c_client->dev, + "Cirrus Logic CS42L52, Revision: %02X\n", reg & 0xFF); regcache_cache_only(cs42l52->regmap, true); diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h index 4277012..1a9412d 100644 --- a/sound/soc/codecs/cs42l52.h +++ b/sound/soc/codecs/cs42l52.h @@ -269,6 +269,6 @@ #define CS42L52_FIX_BITS1 0x3E #define CS42L52_FIX_BITS2 0x47 -#define CS42L52_MAX_REGISTER 0x34 +#define CS42L52_MAX_REGISTER 0x47 #endif -- 1.8.4.rc3

1 2

[alsa-devel] [PATCH v7] ALSA: Add SoC on-chip internal ram support for DMA buffer allocation
by Nicolin Chen 24 Oct '13

24 Oct '13

Now it's quite common that an SoC contains its on-chip internal RAM. By using this RAM space for DMA buffer during audio playback/record, we can shutdown the voltage for external RAM to save power. So add new DEV type with iram malloc()/free() and accordingly modify current default mmap() for the iram circumstance. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v6->v7: * Assigned pool to private_data field and accordingly refined helper functions. v5->v6: * Dropped remaining OF dependency bacause no need to add them: of_node doesn't care about OF; of_get_named_gen_pool() returns NULL if !OF. v4->v5: * Minimized the OF dependency. v3->v4: * Appropriately placed '#ifdef CONFIG_OF' * Tested by adding '#undef CONFIG_OF' to modified files, and passed compiling. v2->v3: * Moved iram specific mmap procedure out of ifdef ARCH_HAS_DMA_MMAP_COHERENT v1->v2: * Added of_node check * Dropped noops and unused physical addr in snd_free_dev_iram() --- include/sound/memalloc.h | 1 + sound/core/memalloc.c | 52 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/core/pcm_native.c | 6 ++++++ 3 files changed, 59 insertions(+) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index cf15b82..510aec4 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -52,6 +52,7 @@ struct snd_dma_device { #else #define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ #endif +#define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */ /* * info for buffer allocation diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index bdf826f..18c1d47 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -30,6 +30,7 @@ #include <linux/seq_file.h> #include <asm/uaccess.h> #include <linux/dma-mapping.h> +#include <linux/genalloc.h> #include <linux/moduleparam.h> #include <linux/mutex.h> #include <sound/memalloc.h> @@ -157,6 +158,46 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, dec_snd_pages(pg); dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); } + +/** + * snd_malloc_dev_iram - allocate memory from on-chip internal ram + * @dmab: buffer allocation record to store the allocated data + * @size: number of bytes to allocate from the iram + * + * This function requires iram phandle provided via of_node + */ +void snd_malloc_dev_iram(struct snd_dma_buffer *dmab, size_t size) +{ + struct device *dev = dmab->dev.dev; + struct gen_pool *pool = NULL; + + if (dev->of_node) + pool = of_get_named_gen_pool(dev->of_node, "iram", 0); + + if (!pool) + return; + + /* Assign the pool into private_data field */ + dmab->private_data = pool; + + dmab->area = (void *)gen_pool_alloc(pool, size); + if (!dmab->area) + return; + + dmab->addr = gen_pool_virt_to_phys(pool, (unsigned long)dmab->area); +} + +/** + * snd_free_dev_iram - free allocated specific memory from on-chip internal ram + * @dmab: buffer allocation record to store the allocated data + */ +void snd_free_dev_iram(struct snd_dma_buffer *dmab) +{ + struct gen_pool *pool = dmab->private_data; + + if (pool && dmab->area) + gen_pool_free(pool, (unsigned long)dmab->area, dmab->bytes); +} #endif /* CONFIG_HAS_DMA */ /* @@ -197,6 +238,14 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->addr = 0; break; #ifdef CONFIG_HAS_DMA + case SNDRV_DMA_TYPE_DEV_IRAM: + snd_malloc_dev_iram(dmab, size); + if (dmab->area) + break; + /* Internal memory might have limited size and no enough space, + * so if we fail to malloc, try to fetch memory traditionally. + */ + dmab->dev.type = SNDRV_DMA_TYPE_DEV; case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); break; @@ -269,6 +318,9 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) snd_free_pages(dmab->area, dmab->bytes); break; #ifdef CONFIG_HAS_DMA + case SNDRV_DMA_TYPE_DEV_IRAM: + snd_free_dev_iram(dmab); + break; case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); break; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a68d4c6..513f095 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3199,6 +3199,12 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *area) { area->vm_flags |= VM_DONTEXPAND | VM_DONTDUMP; + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_IRAM) { + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return remap_pfn_range(area, area->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + area->vm_end - area->vm_start, area->vm_page_prot); + } #ifdef ARCH_HAS_DMA_MMAP_COHERENT if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) -- 1.8.4

3 6

[alsa-devel] [PATCH v4] ASoC: kirkwood: add S/PDIF support
by Jean-Francois Moine 24 Oct '13

24 Oct '13

This patch adds S/PDIF input/output for mvebu DT boards. Three output DAI's are created: I2S only, S/PDIF only, both I2S and S/PDIF. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v4: declare 3 DAI's v3: (Mark Brown) - giving a name to the DAI's impacts the Openrd and HP t5325 clients - test i2s or spdif from the DAI's ID instead of its name v2: declare 2 DAI's (Mark Brown) --- sound/soc/kirkwood/kirkwood-i2s.c | 119 ++++++++++++--------- sound/soc/kirkwood/kirkwood-openrd.c | 2 +- sound/soc/kirkwood/kirkwood-t5325.c | 2 +- 3 file changed, 72 insertions(+), 51 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index d0504a2..c3a5a3e 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -160,9 +160,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_16; ctl_play = KIRKWOOD_PLAYCTL_SIZE_16_C | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_16_C | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN; break; /* * doesn't work... S20_3LE != kirkwood 20bit format ? @@ -178,9 +180,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S24_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_24; ctl_play = KIRKWOOD_PLAYCTL_SIZE_24 | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_24 | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN; break; case SNDRV_PCM_FORMAT_S32_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_32; @@ -240,6 +244,11 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, ctl); } + if (dai->id == 0) + ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s only */ + else if (dai->id == 1) + ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif only */ + switch (cmd) { case SNDRV_PCM_TRIGGER_START: /* configure */ @@ -258,7 +267,8 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: /* stop audio, disable interrupts */ - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_INT_MASK); @@ -272,13 +282,15 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE); + ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE); writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; @@ -301,7 +313,13 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: /* configure */ ctl = priv->ctl_rec; - value = ctl & ~KIRKWOOD_RECCTL_I2S_EN; + if (dai->id == 0) + ctl &= ~KIRKWOOD_RECCTL_SPDIF_EN; /* i2s only */ + else if (dai->id == 1) + ctl &= ~KIRKWOOD_RECCTL_I2S_EN; /* spdif only */ + + value = ctl & ~(KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN); writel(value, priv->io + KIRKWOOD_RECCTL); /* enable interrupts */ @@ -361,9 +379,8 @@ static int kirkwood_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int kirkwood_i2s_probe(struct snd_soc_dai *dai) +static int kirkwood_i2s_init(struct kirkwood_dma_data *priv) { - struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai); unsigned long value; unsigned int reg_data; @@ -404,45 +421,46 @@ static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { .set_fmt = kirkwood_i2s_set_fmt, }; +#define DAI_DEF(name_p, id_p, rates_p) {\ + .name = name_p,\ + .id = id_p,\ + .playback = {\ + .channels_min = 1,\ + .channels_max = 2,\ + .rates = rates_p,\ + .formats = KIRKWOOD_I2S_FORMATS,\ + },\ + .capture = {\ + .channels_min = 1,\ + .channels_max = 2,\ + .rates = rates_p,\ + .formats = KIRKWOOD_I2S_FORMATS,\ + },\ + .ops = &kirkwood_i2s_dai_ops,\ +} -static struct snd_soc_dai_driver kirkwood_i2s_dai = { - .probe = kirkwood_i2s_probe, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_96000, - .formats = KIRKWOOD_I2S_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_96000, - .formats = KIRKWOOD_I2S_FORMATS, - }, - .ops = &kirkwood_i2s_dai_ops, +static struct snd_soc_dai_driver kirkwood_i2s_dai[3] = { + DAI_DEF("i2s", 0, + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000), + DAI_DEF("spdif", 1, + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000), + DAI_DEF("i2s/spdif", 2, + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000), }; -static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { - .probe = kirkwood_i2s_probe, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, - .formats = KIRKWOOD_I2S_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, - .formats = KIRKWOOD_I2S_FORMATS, - }, - .ops = &kirkwood_i2s_dai_ops, +static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[3] = { + DAI_DEF("i2s", 0, + SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT), + DAI_DEF("spdif", 1, + SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT), + DAI_DEF("i2s/spdif", 2, + SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT), }; static const struct snd_soc_component_driver kirkwood_i2s_component = { @@ -452,7 +470,7 @@ static const struct snd_soc_component_driver kirkwood_i2s_component = { static int kirkwood_i2s_dev_probe(struct platform_device *pdev) { struct kirkwood_asoc_platform_data *data = pdev->dev.platform_data; - struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; + struct snd_soc_dai_driver *soc_dai = kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; struct device_node *np = pdev->dev.of_node; @@ -506,7 +524,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } else { dev_info(&pdev->dev, "found external clock\n"); clk_prepare_enable(priv->extclk); - soc_dai = &kirkwood_i2s_dai_extclk; + soc_dai = kirkwood_i2s_dai_extclk; } } @@ -524,7 +542,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component, - soc_dai, 1); + soc_dai, 3); if (err) { dev_err(&pdev->dev, "snd_soc_register_component failed\n"); goto err_component; @@ -535,6 +553,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "snd_soc_register_platform failed\n"); goto err_platform; } + + kirkwood_i2s_init(priv); + return 0; err_platform: snd_soc_unregister_component(&pdev->dev); diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index 025be0e..65f2a5b 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -52,7 +52,7 @@ static struct snd_soc_dai_link openrd_client_dai[] = { { .name = "CS42L51", .stream_name = "CS42L51 HiFi", - .cpu_dai_name = "mvebu-audio", + .cpu_dai_name = "i2s", .platform_name = "mvebu-audio", .codec_dai_name = "cs42l51-hifi", .codec_name = "cs42l51-codec.0-004a", diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 27545b0..d213832 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -68,7 +68,7 @@ static struct snd_soc_dai_link t5325_dai[] = { { .name = "ALC5621", .stream_name = "ALC5621 HiFi", - .cpu_dai_name = "mvebu-audio", + .cpu_dai_name = "i2s", .platform_name = "mvebu-audio", .codec_dai_name = "alc5621-hifi", .codec_name = "alc562x-codec.0-001a", -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] [PATCH v2] ASoC: kirkwood: add S/PDIF support
by Jean-Francois Moine 24 Oct '13

24 Oct '13

This patch adds S/PDIF input/output for mvebu DT boards. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v3: (Mark Brown) - giving a name to the DAI's impacts the Openrd and HP t5325 clients - test i2s or spdif from the DAI's ID instead of its name v2: declare 2 DAI's (Mark Brown) --- sound/soc/kirkwood/kirkwood-i2s.c | 99 ++++++++++++++++++---- sound/soc/kirkwood/kirkwood-openrd.c | 2 +- sound/soc/kirkwood/kirkwood-t5325.c | 2 +- 3 file changed, 84 insertions(+), 19 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index c2433ea..f3e1547 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -160,9 +160,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_16; ctl_play = KIRKWOOD_PLAYCTL_SIZE_16_C | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_16_C | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN; break; /* * doesn't work... S20_3LE != kirkwood 20bit format ? @@ -178,9 +180,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S24_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_24; ctl_play = KIRKWOOD_PLAYCTL_SIZE_24 | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_24 | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN; break; case SNDRV_PCM_FORMAT_S32_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_32; @@ -240,6 +244,11 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, ctl); } + if (dai->id == 0) + ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */ + else + ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */ + switch (cmd) { case SNDRV_PCM_TRIGGER_START: /* configure */ @@ -258,7 +267,8 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: /* stop audio, disable interrupts */ - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_INT_MASK); @@ -272,13 +282,15 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE); + ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE); writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; @@ -301,7 +313,13 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: /* configure */ ctl = priv->ctl_rec; - value = ctl & ~KIRKWOOD_RECCTL_I2S_EN; + if (dai->id == 0) + ctl &= ~KIRKWOOD_RECCTL_SPDIF_EN; /* i2s */ + else + ctl &= ~KIRKWOOD_RECCTL_I2S_EN; /* spdif */ + + value = ctl & ~(KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN); writel(value, priv->io + KIRKWOOD_RECCTL); /* enable interrupts */ @@ -361,9 +379,8 @@ static int kirkwood_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int kirkwood_i2s_probe(struct snd_soc_dai *dai) +static int kirkwood_i2s_init(struct kirkwood_dma_data *priv) { - struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai); unsigned long value; unsigned int reg_data; @@ -404,9 +421,29 @@ static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { .set_fmt = kirkwood_i2s_set_fmt, }; - -static struct snd_soc_dai_driver kirkwood_i2s_dai = { - .probe = kirkwood_i2s_probe, +static struct snd_soc_dai_driver kirkwood_i2s_dai[2] = { + { + .name = "i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .ops = &kirkwood_i2s_dai_ops, + }, + { + .name = "spdif", + .id = 1, .playback = { .channels_min = 1, .channels_max = 2, @@ -422,10 +459,34 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai = { .formats = KIRKWOOD_I2S_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, + }, }; -static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { - .probe = kirkwood_i2s_probe, +static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { + { + .name = "i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000 | + SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000 | + SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .ops = &kirkwood_i2s_dai_ops, + }, + { + .name = "spdif", + .id = 1, .playback = { .channels_min = 1, .channels_max = 2, @@ -443,6 +504,7 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { .formats = KIRKWOOD_I2S_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, + }, }; static const struct snd_soc_component_driver kirkwood_i2s_component = { @@ -452,7 +514,7 @@ static const struct snd_soc_component_driver kirkwood_i2s_component = { static int kirkwood_i2s_dev_probe(struct platform_device *pdev) { struct kirkwood_asoc_platform_data *data = pdev->dev.platform_data; - struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; + struct snd_soc_dai_driver *soc_dai = kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; struct device_node *np = pdev->dev.of_node; @@ -521,7 +583,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component, - soc_dai, 1); + soc_dai, 2); if (err) { dev_err(&pdev->dev, "snd_soc_register_component failed\n"); goto err_component; @@ -532,6 +594,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "snd_soc_register_platform failed\n"); goto err_platform; } + + kirkwood_i2s_init(priv); + return 0; err_platform: snd_soc_unregister_component(&pdev->dev); diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index 025be0e..65f2a5b 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -52,7 +52,7 @@ static struct snd_soc_dai_link openrd_client_dai[] = { { .name = "CS42L51", .stream_name = "CS42L51 HiFi", - .cpu_dai_name = "mvebu-audio", + .cpu_dai_name = "i2s", .platform_name = "mvebu-audio", .codec_dai_name = "cs42l51-hifi", .codec_name = "cs42l51-codec.0-004a", diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 27545b0..d213832 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -68,7 +68,7 @@ static struct snd_soc_dai_link t5325_dai[] = { { .name = "ALC5621", .stream_name = "ALC5621 HiFi", - .cpu_dai_name = "mvebu-audio", + .cpu_dai_name = "i2s", .platform_name = "mvebu-audio", .codec_dai_name = "alc5621-hifi", .codec_name = "alc562x-codec.0-001a", -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] [PATCH] ASoC: fsl: Add missing pm to current machine drivers
by Nicolin Chen 24 Oct '13

24 Oct '13

Add missing pm to current machine drivers so that all of them would correctly do suspend/resume. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/imx-mc13783.c | 1 + sound/soc/fsl/imx-sgtl5000.c | 1 + sound/soc/fsl/imx-wm8962.c | 1 + 3 files changed, 3 insertions(+) diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index a3d60d4..8ded5ad 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -160,6 +160,7 @@ static struct platform_driver imx_mc13783_audio_driver = { .driver = { .name = "imx_mc13783", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = imx_mc13783_probe, .remove = imx_mc13783_remove diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 6f4bdc8..4b65c04 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -201,6 +201,7 @@ static struct platform_driver imx_sgtl5000_driver = { .driver = { .name = "imx-sgtl5000", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, .of_match_table = imx_sgtl5000_dt_ids, }, .probe = imx_sgtl5000_probe, diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 6c60666..72064e9 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -310,6 +310,7 @@ static struct platform_driver imx_wm8962_driver = { .driver = { .name = "imx-wm8962", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, .of_match_table = imx_wm8962_dt_ids, }, .probe = imx_wm8962_probe, -- 1.8.4

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