November 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 00/15] ASoC: soc-core cleanup step6
by Kuninori Morimoto 02 Dec '19

02 Dec '19

Hi Mark These are v2 of soc-core cleanup step6. These doesn't add new features. Kuninori Morimoto (15): ASoC: soc-core: remove dai_link_list ASoC: soc-core: remove snd_soc_disconnect_sync() ASoC: soc-core: remove snd_soc_get_dai_substream() ASoC: soc-core: move snd_soc_get_pcm_runtime() ASoC: soc-core: find rtd via dai_link pointer at snd_soc_get_pcm_runtime() ASoC: soc-core: move snd_soc_find_dai_link() ASoC: soc-core: rename snd_soc_add_dai_link() to snd_soc_add_pcm_runtime() ASoC: soc-core: rename snd_soc_remove_dai_link() to snd_soc_remove_pcm_runtime() ASoC: soc-core: move soc_link_dai_pcm_new() ASoC: soc-core: rename soc_link_dai_pcm_new() to soc_dai_pcm_new() ASoC: soc-core: move soc_link_init() ASoC: soc-core: add missing return value check for soc_link_init() ASoC: soc-core: rename soc_link_init() to soc_init_pcm_runtime() ASoC: soc-core: soc_set_name_prefix(): tidyup loop condition ASoC: soc-core: soc_set_name_prefix(): get component device_node at out of loop include/sound/soc.h | 22 +-- sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/pxa/mioa701_wm9713.c | 2 +- sound/soc/samsung/bells.c | 12 +- sound/soc/samsung/littlemill.c | 10 +- sound/soc/samsung/snow.c | 2 +- sound/soc/samsung/speyside.c | 4 +- sound/soc/samsung/tm2_wm5110.c | 6 +- sound/soc/samsung/tobermory.c | 6 +- sound/soc/sh/rcar/core.c | 2 - sound/soc/soc-core.c | 339 ++++++++++++++++------------------------- sound/soc/soc-topology.c | 47 +++++- sound/soc/tegra/tegra_wm8903.c | 2 +- 13 files changed, 204 insertions(+), 252 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ALSA: usb-audio: Fix Scarlett 6i6 Gen 2 port data
by Geoffrey D. Bennett 02 Dec '19

02 Dec '19

The s6i6_gen2_info.ports[] array had the Mixer and PCM port type entries in the wrong place. Use designators to explicitly specify the array elements being set. Fixes: 9e4d5c1be21f ("ALSA: usb-audio: Scarlett Gen 2 mixer interface") Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> Reported-by: Jonas Berlin <kernel(a)xkr47.space> Tested-by: Alex Fellows <alex.fellows(a)gmail.com> --- Hi Takashi, Is there something special I need to do to mark this as a bug fix and request this go into 5.4? Thanks, Geoffrey. sound/usb/mixer_scarlett_gen2.c | 36 ++++++++++++++++----------------- 1 file changed, 18 insertions(+), 18 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 7d460b1f1735..94b903d95afa 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -261,34 +261,34 @@ static const struct scarlett2_device_info s6i6_gen2_info = { }, .ports = { - { + [SCARLETT2_PORT_TYPE_NONE] = { .id = 0x000, .num = { 1, 0, 8, 8, 8 }, .src_descr = "Off", .src_num_offset = 0, }, - { + [SCARLETT2_PORT_TYPE_ANALOGUE] = { .id = 0x080, .num = { 4, 4, 4, 4, 4 }, .src_descr = "Analogue %d", .src_num_offset = 1, .dst_descr = "Analogue Output %02d Playback" }, - { + [SCARLETT2_PORT_TYPE_SPDIF] = { .id = 0x180, .num = { 2, 2, 2, 2, 2 }, .src_descr = "S/PDIF %d", .src_num_offset = 1, .dst_descr = "S/PDIF Output %d Playback" }, - { + [SCARLETT2_PORT_TYPE_MIX] = { .id = 0x300, .num = { 10, 18, 18, 18, 18 }, .src_descr = "Mix %c", .src_num_offset = 65, .dst_descr = "Mixer Input %02d Capture" }, - { + [SCARLETT2_PORT_TYPE_PCM] = { .id = 0x600, .num = { 6, 6, 6, 6, 6 }, .src_descr = "PCM %d", @@ -317,44 +317,44 @@ static const struct scarlett2_device_info s18i8_gen2_info = { }, .ports = { - { + [SCARLETT2_PORT_TYPE_NONE] = { .id = 0x000, .num = { 1, 0, 8, 8, 4 }, .src_descr = "Off", .src_num_offset = 0, }, - { + [SCARLETT2_PORT_TYPE_ANALOGUE] = { .id = 0x080, .num = { 8, 6, 6, 6, 6 }, .src_descr = "Analogue %d", .src_num_offset = 1, .dst_descr = "Analogue Output %02d Playback" }, - { + [SCARLETT2_PORT_TYPE_SPDIF] = { + .id = 0x180, /* S/PDIF outputs aren't available at 192KHz * but are included in the USB mux I/O * assignment message anyway */ - .id = 0x180, .num = { 2, 2, 2, 2, 2 }, .src_descr = "S/PDIF %d", .src_num_offset = 1, .dst_descr = "S/PDIF Output %d Playback" }, - { + [SCARLETT2_PORT_TYPE_ADAT] = { .id = 0x200, .num = { 8, 0, 0, 0, 0 }, .src_descr = "ADAT %d", .src_num_offset = 1, }, - { + [SCARLETT2_PORT_TYPE_MIX] = { .id = 0x300, .num = { 10, 18, 18, 18, 18 }, .src_descr = "Mix %c", .src_num_offset = 65, .dst_descr = "Mixer Input %02d Capture" }, - { + [SCARLETT2_PORT_TYPE_PCM] = { .id = 0x600, .num = { 20, 18, 18, 14, 10 }, .src_descr = "PCM %d", @@ -387,20 +387,20 @@ static const struct scarlett2_device_info s18i20_gen2_info = { }, .ports = { - { + [SCARLETT2_PORT_TYPE_NONE] = { .id = 0x000, .num = { 1, 0, 8, 8, 6 }, .src_descr = "Off", .src_num_offset = 0, }, - { + [SCARLETT2_PORT_TYPE_ANALOGUE] = { .id = 0x080, .num = { 8, 10, 10, 10, 10 }, .src_descr = "Analogue %d", .src_num_offset = 1, .dst_descr = "Analogue Output %02d Playback" }, - { + [SCARLETT2_PORT_TYPE_SPDIF] = { /* S/PDIF outputs aren't available at 192KHz * but are included in the USB mux I/O * assignment message anyway @@ -411,21 +411,21 @@ static const struct scarlett2_device_info s18i20_gen2_info = { .src_num_offset = 1, .dst_descr = "S/PDIF Output %d Playback" }, - { + [SCARLETT2_PORT_TYPE_ADAT] = { .id = 0x200, .num = { 8, 8, 8, 4, 0 }, .src_descr = "ADAT %d", .src_num_offset = 1, .dst_descr = "ADAT Output %d Playback" }, - { + [SCARLETT2_PORT_TYPE_MIX] = { .id = 0x300, .num = { 10, 18, 18, 18, 18 }, .src_descr = "Mix %c", .src_num_offset = 65, .dst_descr = "Mixer Input %02d Capture" }, - { + [SCARLETT2_PORT_TYPE_PCM] = { .id = 0x600, .num = { 20, 18, 18, 14, 10 }, .src_descr = "PCM %d", -- 2.20.1

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[alsa-devel] More USB audio quirks
by Damien Zammit 30 Nov '19

30 Nov '19

Hi all, I asked on a forum for people to post their lsusb -vv output for usb audio devices that were not being detected correctly, and I tried to write some quirks for them. I don't know the current status on whether support has been added, but please someone take a look at: https://discourse.ardour.org/t/providing-assistance-to-devs-for-new-alsa-us… for some more output and my attempts at writing new quirks. Apologies if this is now out of date. Damien

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[alsa-devel] [PATCH alsa-lib] ucm: Use strncmp to avoid access-out-of-boundary
by Chih-Yang Hsia 29 Nov '19

29 Nov '19

From: paulhsia <paulhsia(a)chromium.org> If the length of the identifier is less than the length of the prefix, access-out-of-boundary will occur in memcmp(). Signed-off-by: paulhsia <paulhsia(a)chromium.org> --- src/ucm/main.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index b0b6ffb3..252e50d9 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -61,11 +61,13 @@ static int check_identifier(const char *identifier, const char *prefix) { int len; - if (strcmp(identifier, prefix) == 0) - return 1; len = strlen(prefix); - if (memcmp(identifier, prefix, len) == 0 && identifier[len] == '/') + if (strncmp(identifier, prefix, len) != 0) + return 0; + + if (identifier[len] == 0 || identifier[len] == '/') return 1; + return 0; } -- 2.24.0.393.g34dc348eaf-goog

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[alsa-devel] [PATCH v5 0/2] ASoC: mt8183: fix audio playback slowly after playback
by Jiaxin Yu 29 Nov '19

29 Nov '19

This series patches add reset controller for MT8183, and audio will use it in machine driver during bootup, they depend on the for-next. v5 changes: 1. Add Signed-off-by tag and Reviewed-by tag. v4 changes: 1. Fixed wrong signed-off as correct mail suffix. 2. Fixed patch subject that add patch version. v3 changes: 1. https://patchwork.kernel.org/patch/11164283/ and https://patchwork.kernel.org/patch/11164305/ has been merged. 2. Change the name of mtk_wdt_compatible to mtk_wdt_data. 3. Remove toprgu_reset struct and use mtk_wdt_dev instead. 4. Get the value of sw_rst_num from .h file. 5. Adddd mt2712-resets.h for mt2712. 6. Improve commit message. v2 changes: 1. remove "WIP" that in the title of patches 2. add hyper link for the patch that depends on 3. patchwork list: https://patchwork.kernel.org/cover/11164285/ https://patchwork.kernel.org/patch/11164295/ https://patchwork.kernel.org/patch/11164299/ https://patchwork.kernel.org/patch/11164283/ https://patchwork.kernel.org/patch/11164305/ v1 changes: 1. patchwork list: https://patchwork.kernel.org/cover/11164173/ https://patchwork.kernel.org/patch/11164181/ https://patchwork.kernel.org/patch/11164185/ https://patchwork.kernel.org/patch/11164187/ https://patchwork.kernel.org/patch/11164175/ yong.liang (2): dt-bindings: mediatek: mt8183: Add #reset-cells watchdog: mtk_wdt: mt8183: Add reset controller .../devicetree/bindings/watchdog/mtk-wdt.txt | 10 +- drivers/watchdog/Kconfig | 1 + drivers/watchdog/mtk_wdt.c | 111 +++++++++++++++++- .../reset-controller/mt2712-resets.h | 22 ++++ .../reset-controller/mt8183-resets.h | 15 +++ 5 files changed, 155 insertions(+), 4 deletions(-) create mode 100644 include/dt-bindings/reset-controller/mt2712-resets.h -- 2.18.0

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[alsa-devel] [PATCH] ALSA: hda/realtek - Fix inverted bass GPIO pin on Acer 8951G
by Takashi Iwai 28 Nov '19

28 Nov '19

We've added the bass speaker support on Acer 8951G by the commit 00066e9733f6 ("Add Acer Aspire Ethos 8951G model quirk"), but it seems that the GPIO pin was wrongly set: while the commit turns off the bit to power up the amp, the actual hardware reacts other way round, i.e. GPIO bit on = amp on. So this patch fixes the bug, turning on the GPIO bit 0x02 as default. Since turning on the GPIO bit can be more easily managed with alc_setup_gpio() call, we simplify the quirk code by integrating the GPIO setup into the existing alc662_fixup_aspire_ethos_hp() and dropping the whole ALC669_FIXUP_ACER_ASPIRE_ETHOS_SUBWOOFER quirk. Fixes: 00066e9733f6 ("Add Acer Aspire Ethos 8951G model quirk") Reported-and-tested-by: Sergey 'Jin' Bostandzhyan <jin(a)mediatomb.cc> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 17 +++-------------- 1 file changed, 3 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9f355b2f7d7b..c7a232fd5a5b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8427,6 +8427,8 @@ static void alc662_fixup_aspire_ethos_hp(struct hda_codec *codec, case HDA_FIXUP_ACT_PRE_PROBE: snd_hda_jack_detect_enable_callback(codec, 0x1b, alc662_aspire_ethos_mute_speakers); + /* subwoofer needs an extra GPIO setting to become audible */ + alc_setup_gpio(codec, 0x02); break; case HDA_FIXUP_ACT_INIT: /* Make sure to start in a correct state, i.e. if @@ -8509,7 +8511,6 @@ enum { ALC662_FIXUP_USI_HEADSET_MODE, ALC662_FIXUP_LENOVO_MULTI_CODECS, ALC669_FIXUP_ACER_ASPIRE_ETHOS, - ALC669_FIXUP_ACER_ASPIRE_ETHOS_SUBWOOFER, ALC669_FIXUP_ACER_ASPIRE_ETHOS_HEADSET, }; @@ -8841,18 +8842,6 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc662_fixup_aspire_ethos_hp, }, - [ALC669_FIXUP_ACER_ASPIRE_ETHOS_SUBWOOFER] = { - .type = HDA_FIXUP_VERBS, - /* subwoofer needs an extra GPIO setting to become audible */ - .v.verbs = (const struct hda_verb[]) { - {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, - { } - }, - .chained = true, - .chain_id = ALC669_FIXUP_ACER_ASPIRE_ETHOS_HEADSET - }, [ALC669_FIXUP_ACER_ASPIRE_ETHOS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -8862,7 +8851,7 @@ static const struct hda_fixup alc662_fixups[] = { { } }, .chained = true, - .chain_id = ALC669_FIXUP_ACER_ASPIRE_ETHOS_SUBWOOFER + .chain_id = ALC669_FIXUP_ACER_ASPIRE_ETHOS_HEADSET }, }; -- 2.16.4

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[alsa-devel] [PATCH] ASoC: intel/skl/hda - export number of digital microphones via control components
by Jaroslav Kysela 28 Nov '19

28 Nov '19

It is required for the auto-detection in the user space (for UCM). Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 8 ++++++++ sound/soc/sof/intel/hda.c | 3 ++- 2 files changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 4e45901e3a2f..11eaee9ae41f 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -100,6 +100,8 @@ static struct snd_soc_card hda_soc_card = { .late_probe = skl_hda_card_late_probe, }; +static char hda_soc_components[30]; + #define IDISP_DAI_COUNT 3 #define HDAC_DAI_COUNT 2 #define DMIC_DAI_COUNT 2 @@ -183,6 +185,12 @@ static int skl_hda_audio_probe(struct platform_device *pdev) hda_soc_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&hda_soc_card, ctx); + if (mach->mach_params.dmic_num > 0) { + snprintf(hda_soc_components, sizeof(hda_soc_components), + "cfg-dmics:%d", mach->mach_params.dmic_num); + hda_soc_card.components = hda_soc_components; + } + return devm_snd_soc_register_card(&pdev->dev, &hda_soc_card); } diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 7dc0018dc4c3..92fffca76af5 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -351,7 +351,7 @@ static int hda_init_caps(struct snd_sof_dev *sdev) const char *tplg_filename; const char *idisp_str; const char *dmic_str; - int dmic_num; + int dmic_num = 0; int codec_num = 0; int i; #endif @@ -465,6 +465,7 @@ static int hda_init_caps(struct snd_sof_dev *sdev) mach_params->codec_mask = bus->codec_mask; mach_params->platform = dev_name(sdev->dev); mach_params->common_hdmi_codec_drv = hda_codec_use_common_hdmi; + mach_params->dmic_num = dmic_num; } /* create codec instances */ -- 2.20.1

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[alsa-devel] Surround speaker connection on Acer 8951G
by Sergey 'Jin' Bostandzhyan 28 Nov '19

28 Nov '19

Hi, I am not sure if this is the right place to ask or if I should just submit a bugreport, but you guys helped me out with a similar problem about 10 years ago, so I'll try here first. If I'm wrong here, then I apologize, just let me know. I have an Acer Aspire Ethos 8951G notebook which has 6 built in speakers for 5.1 surround sound, it's a Realtek ALC669X. The problem is that only two speakers are recognized by default. I tried fiddling around with hdajackretast and it does make a difference, but I somehow could not find a step-by-step approach on how to figure out what is what and how to map everything correctly, so some guidance would be very much appreciated. I even ended up in situations where I got channels supporting only mute which would influence muting of other channels, I really got lost there.. Here's my alsa-info output: http://alsa-project.org/db/?f=3adde87164f5fc349c3c78b211ee63e416ebf10b It would be great if we could figure it out and make it work by default. Basically that's what we did 10 years ago, back then I had an Acer Aspire 8920G with a ALC889 which had the same problem that got fixed by you and these days it just works out of the box :) Kind regards, Jin

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[alsa-devel] [PATCH] ASoC: rt5682: fix i2c arbitration lost issue
by shumingf＠realtek.com 28 Nov '19

28 Nov '19

From: Shuming Fan <shumingf(a)realtek.com> This patch modified the HW initial setting to fix i2c arbitration lost issue. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index b1713fffa3eb..ae6f6121bc1b 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -73,6 +73,7 @@ struct rt5682_priv { static const struct reg_sequence patch_list[] = { {RT5682_HP_IMP_SENS_CTRL_19, 0x1000}, {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, + {RT5682_I2C_CTRL, 0x000f}, }; static const struct reg_default rt5682_reg[] = { @@ -2474,6 +2475,7 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) mutex_lock(&rt5682->calibrate_mutex); rt5682_reset(rt5682->regmap); + regmap_write(rt5682->regmap, RT5682_I2C_CTRL, 0x000f); regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af); usleep_range(15000, 20000); regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2af); -- 2.24.0

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[alsa-devel] [PATCH] ASoC: AMD: Enable clk in startup intead of hw_params
by Yu-Hsuan Hsu 28 Nov '19

28 Nov '19

Some usages only call startup and shutdown without setting hw_params (e.g. arecord --dump-hw-params). If we don't enable clk in startup, it will cause ref count error because the clk will be disabled in shutdown. For this reason, we should move enabling clk from hw_params to startup. In addition, the hw_params is fixed in this driver(48000 rate, 2 channels, S16_LE format) so we don't need to change the clk rate after the hw_params is set. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/amd/acp-da7219-max98357a.c | 46 +++++++++------------------- 1 file changed, 14 insertions(+), 32 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index f4ee6798154af5..7a5621e5e2330d 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -96,14 +96,19 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int da7219_clk_enable(struct snd_pcm_substream *substream, - int wclk_rate, int bclk_rate) +static int da7219_clk_enable(struct snd_pcm_substream *substream) { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - clk_set_rate(da7219_dai_wclk, wclk_rate); - clk_set_rate(da7219_dai_bclk, bclk_rate); + /* + * Set wclk to 48000 because the rate constraint of this driver is + * 48000. ADAU7002 spec: "The ADAU7002 requires a BCLK rate that is + * minimum of 64x the LRCLK sample rate." DA7219 is the only clk + * source so for all codecs we have to limit bclk to 64X lrclk. + */ + clk_set_rate(da7219_dai_wclk, 48000); + clk_set_rate(da7219_dai_bclk, 48000 * 64); ret = clk_prepare_enable(da7219_dai_bclk); if (ret < 0) { dev_err(rtd->dev, "can't enable master clock %d\n", ret); @@ -156,7 +161,7 @@ static int cz_da7219_play_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->play_i2s_instance = I2S_SP_INSTANCE; - return 0; + return da7219_clk_enable(substream); } static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) @@ -178,7 +183,7 @@ static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; - return 0; + return da7219_clk_enable(substream); } static int cz_max_startup(struct snd_pcm_substream *substream) @@ -199,7 +204,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->play_i2s_instance = I2S_BT_INSTANCE; - return 0; + return da7219_clk_enable(substream); } static int cz_dmic0_startup(struct snd_pcm_substream *substream) @@ -220,7 +225,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->cap_i2s_instance = I2S_BT_INSTANCE; - return 0; + return da7219_clk_enable(substream); } static int cz_dmic1_startup(struct snd_pcm_substream *substream) @@ -242,25 +247,7 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; - return 0; -} - -static int cz_da7219_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - int wclk, bclk; - - wclk = params_rate(params); - bclk = wclk * params_channels(params) * - snd_pcm_format_width(params_format(params)); - /* ADAU7002 spec: "The ADAU7002 requires a BCLK rate - * that is minimum of 64x the LRCLK sample rate." - * DA7219 is the only clk source so for all codecs - * we have to limit bclk to 64X lrclk. - */ - if (bclk < (wclk * 64)) - bclk = wclk * 64; - return da7219_clk_enable(substream, wclk, bclk); + return da7219_clk_enable(substream); } static void cz_da7219_shutdown(struct snd_pcm_substream *substream) @@ -271,31 +258,26 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream) static const struct snd_soc_ops cz_da7219_play_ops = { .startup = cz_da7219_play_startup, .shutdown = cz_da7219_shutdown, - .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_da7219_cap_ops = { .startup = cz_da7219_cap_startup, .shutdown = cz_da7219_shutdown, - .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_max_play_ops = { .startup = cz_max_startup, .shutdown = cz_da7219_shutdown, - .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_dmic0_cap_ops = { .startup = cz_dmic0_startup, .shutdown = cz_da7219_shutdown, - .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_dmic1_cap_ops = { .startup = cz_dmic1_startup, .shutdown = cz_da7219_shutdown, - .hw_params = cz_da7219_params, }; SND_SOC_DAILINK_DEF(designware1, -- 2.24.0.432.g9d3f5f5b63-goog

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