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

[alsa-devel] [PATCH v3 0/8] Add support for voice control on Arizona ADSP
by Charles Keepax 06 Jan '16

06 Jan '16

This series add support for voice control on wm8280/5110. This is done by opening a compressed record channel, upon which data will be available once the voice control functionality has been triggered on the DSP. Changes since v2: - Fix specified using in printf for size_t in wm_adsp_compr_read, as reported by the kbuild test robot. Thanks, Charles Charles Keepax (8): ASoC: wm5110: Provide basic hookup for voice control ASoC: wm_adsp: Factor out finding the location of an algorithm region ALSA: compress: Add SND_AUDIOCODEC_BESPOKE ASoC: wm_adsp: Add support for opening a compressed stream ASoC: wm_adsp: Add code to locate and initialise compressed buffer ASoC: wm_adsp: Attach buffers and streams together ASoC: wm_adsp: Add a handler for the compressed IRQ ASoC: wm_adsp: Pull data through compressed read include/uapi/sound/compress_params.h | 5 +- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/arizona.h | 2 +- sound/soc/codecs/wm5110.c | 95 +++- sound/soc/codecs/wm_adsp.c | 907 ++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 21 + 6 files changed, 1010 insertions(+), 21 deletions(-) -- 2.1.4

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[alsa-devel] Applied "Linux 4.4-rc8" to the asoc tree
by Mark Brown 06 Jan '16

06 Jan '16

The patch Linux 4.4-rc8 has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 168309855a7d1e16db751e9c647119fe2d2dc878 Mon Sep 17 00:00:00 2001 From: Linus Torvalds <torvalds(a)linux-foundation.org> Date: Sun, 3 Jan 2016 15:15:37 -0800 Subject: [PATCH] Linux 4.4-rc8 --- Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Makefile b/Makefile index 1122433a5cd5..9d94adeceab6 100644 --- a/Makefile +++ b/Makefile @@ -1,7 +1,7 @@ VERSION = 4 PATCHLEVEL = 4 SUBLEVEL = 0 -EXTRAVERSION = -rc7 +EXTRAVERSION = -rc8 NAME = Blurry Fish Butt # *DOCUMENTATION* -- 2.6.2

2 2

[alsa-devel] [PATCH] ASoC: rt5645: Constify ACPI device ids
by Mathias Krause 06 Jan '16

06 Jan '16

Constify the ACPI device ID array, no need to have it writable at runtime. Also drop the unused RT5645_INIT_REG_LEN define. Signed-off-by: Mathias Krause <minipli(a)googlemail.com> Cc: Bard Liao <bardliao(a)realtek.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> Cc: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5645.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index ef76940f9dcb..38a0b60a7f6e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -64,7 +64,6 @@ static const struct reg_sequence init_list[] = { {RT5645_PR_BASE + 0x21, 0x4040}, {RT5645_PR_BASE + 0x23, 0x0004}, }; -#define RT5645_INIT_REG_LEN ARRAY_SIZE(init_list) static const struct reg_sequence rt5650_init_list[] = { {0xf6, 0x0100}, @@ -3326,7 +3325,7 @@ static const struct i2c_device_id rt5645_i2c_id[] = { MODULE_DEVICE_TABLE(i2c, rt5645_i2c_id); #ifdef CONFIG_ACPI -static struct acpi_device_id rt5645_acpi_match[] = { +static const struct acpi_device_id rt5645_acpi_match[] = { { "10EC5645", 0 }, { "10EC5650", 0 }, {}, -- 1.7.10.4

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[alsa-devel] [PATCH] ALSA: emux: constify nrpn_conv_table structures
by Julia Lawall 06 Jan '16

06 Jan '16

The nrpn_conv_table structures are never modified, so declare them as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/synth/emux/emux_nrpn.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/synth/emux/emux_nrpn.c b/sound/synth/emux/emux_nrpn.c index 00fc005..9729a15 100644 --- a/sound/synth/emux/emux_nrpn.c +++ b/sound/synth/emux/emux_nrpn.c @@ -48,7 +48,8 @@ struct nrpn_conv_table { * convert NRPN/control values */ -static int send_converted_effect(struct nrpn_conv_table *table, int num_tables, +static int send_converted_effect(const struct nrpn_conv_table *table, + int num_tables, struct snd_emux_port *port, struct snd_midi_channel *chan, int type, int val, int mode) @@ -179,7 +180,7 @@ static int fx_conv_Q(int val) } -static struct nrpn_conv_table awe_effects[] = +static const struct nrpn_conv_table awe_effects[] = { { 0, EMUX_FX_LFO1_DELAY, fx_lfo1_delay}, { 1, EMUX_FX_LFO1_FREQ, fx_lfo1_freq}, @@ -266,7 +267,7 @@ static int gs_vib_delay(int val) return -(val - 64) * gs_sense[FX_VIBDELAY] / 50; } -static struct nrpn_conv_table gs_effects[] = +static const struct nrpn_conv_table gs_effects[] = { {32, EMUX_FX_CUTOFF, gs_cutoff}, {33, EMUX_FX_FILTERQ, gs_filterQ}, @@ -350,7 +351,7 @@ static int xg_release(int val) return -(val - 64) * xg_sense[FX_RELEASE] / 64; } -static struct nrpn_conv_table xg_effects[] = +static const struct nrpn_conv_table xg_effects[] = { {71, EMUX_FX_CUTOFF, xg_cutoff}, {74, EMUX_FX_FILTERQ, xg_filterQ},

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[alsa-devel] PCM3168 alsa driver for beaglebone black
by Pablo Fonovich 06 Jan '16

06 Jan '16

Hi: I'm working in a project of making an audio cape for beaglebone black with pcm3168 codec... so far i've written the driver i've attached based on pcm1681.c available in linux kernel. Im a noob in this subjets, so i would like to know how can now make it work in my custom system for the beaglebone. I followed the guide in this link http://processors.wiki.ti.com/index.php/Sitara_Linux_Audio_DAC_Example, but it makes me write the ops and hw structures that i already have in my pcm3168.c file, in the davinci_evm.c file. The drivers compiles and is detected at boot time, as it says in the guide. But when i type aplay command to play some wav, the message "ALSA lib pcm.c:2267:(snd_pcm_open_noupdate) Unknown PCM default aplay: main:722: audio open error: No such file or directory" appears, so i'm missing something... I don't knwo if this post belongs to this list, but any help would be great... If some docummentation about writing a driver for an embedded system is available, please tell me... Also, if someone can suggest me some guidelines of work i would be grateful, cause i just can't get organized with the information i found Thanks very Much... Pablo

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[alsa-devel] Applied "ASoC: da7218: Correct BCLK inversion for DSP DAI format mode" to the asoc tree
by Mark Brown 05 Jan '16

05 Jan '16

The patch ASoC: da7218: Correct BCLK inversion for DSP DAI format mode has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f644eb62fe88a2414e5e5c1bd80e20c881f5d86a Mon Sep 17 00:00:00 2001 From: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Date: Tue, 5 Jan 2016 18:15:33 +0000 Subject: [PATCH] ASoC: da7218: Correct BCLK inversion for DSP DAI format mode By default the device latches data on the falling edge of the BCLK in DSP mode, whereas the expectation for normal BCLK is to latch on the rising edge. This updates the driver to invert the BCLK configuration for DSP mode, to align with expected behaviour. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/da7218.c | 47 +++++++++++++++++++++++++++++++++++++---------- 1 file changed, 37 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index 72686517ff54..93575f251866 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -1954,17 +1954,44 @@ static int da7218_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - break; - case SND_SOC_DAIFMT_NB_IF: - dai_clk_mode |= DA7218_DAI_WCLK_POL_INV; - break; - case SND_SOC_DAIFMT_IB_NF: - dai_clk_mode |= DA7218_DAI_CLK_POL_INV; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + dai_clk_mode |= DA7218_DAI_WCLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + dai_clk_mode |= DA7218_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_IF: + dai_clk_mode |= DA7218_DAI_WCLK_POL_INV | + DA7218_DAI_CLK_POL_INV; + break; + default: + return -EINVAL; + } break; - case SND_SOC_DAIFMT_IB_IF: - dai_clk_mode |= DA7218_DAI_WCLK_POL_INV | DA7218_DAI_CLK_POL_INV; + case SND_SOC_DAIFMT_DSP_B: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + dai_clk_mode |= DA7218_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + dai_clk_mode |= DA7218_DAI_WCLK_POL_INV | + DA7218_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + dai_clk_mode |= DA7218_DAI_WCLK_POL_INV; + break; + default: + return -EINVAL; + } break; default: return -EINVAL; -- 2.7.0.rc3

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[alsa-devel] [PATCH] ASoC: da7218: Correct BCLK inversion for DSP DAI format mode
by Adam Thomson 05 Jan '16

05 Jan '16

By default the device latches data on the falling edge of the BCLK in DSP mode, whereas the expectation for normal BCLK is to latch on the rising edge. This updates the driver to invert the BCLK configuration for DSP mode, to align with expected behaviour. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- This patch is based on the following commit: 'ASoC: da7218: Enable mic level detection reporting to user-space' (Commit e05c25a1af29d65260ed1458f2cc4a959030ebd2, kernel/git/broonie/sound.git) sound/soc/codecs/da7218.c | 47 +++++++++++++++++++++++++++++++++++++---------- 1 file changed, 37 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index 752ed04..c7cc88a 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -1954,17 +1954,44 @@ static int da7218_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - break; - case SND_SOC_DAIFMT_NB_IF: - dai_clk_mode |= DA7218_DAI_WCLK_POL_INV; - break; - case SND_SOC_DAIFMT_IB_NF: - dai_clk_mode |= DA7218_DAI_CLK_POL_INV; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + dai_clk_mode |= DA7218_DAI_WCLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + dai_clk_mode |= DA7218_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_IF: + dai_clk_mode |= DA7218_DAI_WCLK_POL_INV | + DA7218_DAI_CLK_POL_INV; + break; + default: + return -EINVAL; + } break; - case SND_SOC_DAIFMT_IB_IF: - dai_clk_mode |= DA7218_DAI_WCLK_POL_INV | DA7218_DAI_CLK_POL_INV; + case SND_SOC_DAIFMT_DSP_B: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + dai_clk_mode |= DA7218_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + dai_clk_mode |= DA7218_DAI_WCLK_POL_INV | + DA7218_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + dai_clk_mode |= DA7218_DAI_WCLK_POL_INV; + break; + default: + return -EINVAL; + } break; default: return -EINVAL; -- 1.9.3

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[alsa-devel] [PATCH 00/19] Baytrail/Cherrytrail cleanups and fixes
by Pierre-Louis Bossart 05 Jan '16

05 Jan '16

Set of patches to provide better upstream support for BYT/CHT platforms (24 bit, Deep-buffer, DPCM). This was tested on Asus T100 and Minnowboard. A set of UCM files is available at https://github.com/plbossart/UCM.git for better integration with Ubuntu/pulseaudio (no need for asound.state or asound.conf) Acknowledgements: Keyon Jie for his MinnowBoard validation Sebastien Guiriec for pointing out the switch logic issue and fixing SSP configs Vinod Koul for providing codec/mixer configurations Francisco Mendez for his help with the MCLK and jack detection Jack Yu and Bard Liao for their codec support Known issues: noise on startup when using speakers Once this batch is merged, the following updates will be provided: MCLK support to fix noise issues BYT-CR fixes jack detection Jack Yu (1): ASoC: rt5640: add ASRC support Jorge Fernandez Monteagudo (1): ASoC: Intel: Atom: Add support for HP ElitePad 1000 G2 Pierre-Louis Bossart (16): ASoC: Intel: fix ACPI probe regression with Atom DPCM driver ASoC: Intel: boards: align pin names between byt-rt5640 drivers ASoC: Intel: boards: start merging byt-rt5640 drivers ASoC: Intel: boards: merge DMI-based quirks in bytcr-rt5640 driver ASoC: Intel: tag byt-rt5640 machine driver as deprecated ASoc: Intel: Atom: add deep buffer definitions for atom platforms ASoC: Intel: boards: add DEEP_BUFFER support for BYT/CHT/BSW ASoC: Intel: Atom: add 24-bit support for media playback and capture ASoC: Intel: Atom: clean-up compressed DAI definition ASoC: Intel: Atom: flip logic for gain Switch ASoC: codec: rt5651: add ACPI and OF support ASoC: Intel: add bytct-rt5651 machine driver ASoC: intel: boards: add card for MinnowBoard I2S access ASoC: Intel: bytcr-rt5640: enable ASRC ASoC: Intel: bytcr_rt5640: fixup DAI codec_name with HID ASoC: Intel: Atom: add support for RT5642 Sebastien Guiriec (1): ASoC: Intel: bytcr_rt5640: set SSP to I2S mode 2ch arch/x86/include/asm/platform_sst_audio.h | 1 + sound/soc/codecs/rt5640.c | 103 ++++++++ sound/soc/codecs/rt5640.h | 17 ++ sound/soc/codecs/rt5651.c | 19 ++ sound/soc/intel/Kconfig | 35 ++- sound/soc/intel/atom/sst-atom-controls.c | 5 +- sound/soc/intel/atom/sst-atom-controls.h | 1 + sound/soc/intel/atom/sst-mfld-platform-pcm.c | 20 +- sound/soc/intel/atom/sst/sst_acpi.c | 25 +- sound/soc/intel/boards/Makefile | 4 + sound/soc/intel/boards/bytcr_nocodec.c | 233 ++++++++++++++++++ sound/soc/intel/boards/bytcr_rt5640.c | 266 ++++++++++++++++----- sound/soc/intel/boards/bytcr_rt5651.c | 338 +++++++++++++++++++++++++++ sound/soc/intel/boards/cht_bsw_max98090_ti.c | 12 + sound/soc/intel/boards/cht_bsw_rt5645.c | 12 + sound/soc/intel/boards/cht_bsw_rt5672.c | 12 + sound/soc/intel/common/Makefile | 5 + 17 files changed, 1036 insertions(+), 72 deletions(-) create mode 100644 sound/soc/intel/boards/bytcr_nocodec.c create mode 100644 sound/soc/intel/boards/bytcr_rt5651.c -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: fsl_asrc: old version asrc only support even numbered channels
by Zidan Wang 05 Jan '16

05 Jan '16

The latest asrc's channel bits is 4 bit, the old asrc's channel bits is 3. For the old version asrc, channel should be a even number. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v1->v2: 1. add an extra else if for the new condition. sound/soc/fsl/fsl_asrc.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index be78dc9..d7d81cf 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -116,6 +116,9 @@ static int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair) } else if (asrc_priv->channel_avail < channels) { dev_err(dev, "can't afford required channels: %d\n", channels); ret = -EINVAL; + } else if (asrc_priv->channel_bits < 4 && channels % 2 != 0) { + dev_err(dev, "old version asrc can't support an odd channel number"); + ret = -EINVAL; } else { asrc_priv->channel_avail -= channels; asrc_priv->pair[index] = pair; -- 1.9.1

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[alsa-devel] Applied "ASoC: da7219: Correct BCLK inversion for DSP DAI format mode" to the asoc tree
by Mark Brown 05 Jan '16

05 Jan '16

The patch ASoC: da7219: Correct BCLK inversion for DSP DAI format mode has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4acfa36be618eb8ac3aa39f473e7550710216435 Mon Sep 17 00:00:00 2001 From: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Date: Tue, 5 Jan 2016 15:05:36 +0000 Subject: [PATCH] ASoC: da7219: Correct BCLK inversion for DSP DAI format mode By default the device latches data on the falling edge of the BCLK in DSP mode, whereas the expectation for normal BCLK is to latch on the rising edge. This updates the driver to invert the BCLK configuration for DSP mode, to align with expected behaviour. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/da7219.c | 48 ++++++++++++++++++++++++++++++++++++----------- 1 file changed, 37 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 9c7e8ec68b94..81c0708b85c1 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1156,18 +1156,44 @@ static int da7219_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - break; - case SND_SOC_DAIFMT_NB_IF: - dai_clk_mode |= DA7219_DAI_WCLK_POL_INV; - break; - case SND_SOC_DAIFMT_IB_NF: - dai_clk_mode |= DA7219_DAI_CLK_POL_INV; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + dai_clk_mode |= DA7219_DAI_WCLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + dai_clk_mode |= DA7219_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_IF: + dai_clk_mode |= DA7219_DAI_WCLK_POL_INV | + DA7219_DAI_CLK_POL_INV; + break; + default: + return -EINVAL; + } break; - case SND_SOC_DAIFMT_IB_IF: - dai_clk_mode |= DA7219_DAI_WCLK_POL_INV | - DA7219_DAI_CLK_POL_INV; + case SND_SOC_DAIFMT_DSP_B: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + dai_clk_mode |= DA7219_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + dai_clk_mode |= DA7219_DAI_WCLK_POL_INV | + DA7219_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + dai_clk_mode |= DA7219_DAI_WCLK_POL_INV; + break; + default: + return -EINVAL; + } break; default: return -EINVAL; -- 2.7.0.rc3

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