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b=Rez2it9i9c3Jm/+28Wk0JP8Dw149dG9EldB04BbkIXSMKKQGGUODTRA12L3Et8T634vo8R n9vWEAtehtzxwn1tbhFUauMrmkTyZBtZkHixVwBuaBv61REOKWxcKAnfo/HoVtT1DvakJU U0cTay5kmIxbtyKJfdCeFioAhyX6izI= Received: from mimecast-mx01.redhat.com (mimecast-mx01.redhat.com [209.132.183.4]) (Using TLS) by relay.mimecast.com with ESMTP id us-mta-190-AE2bjIGkPNSpSXG2NTPowQ-1; Fri, 26 Feb 2021 09:38:23 -0500 X-MC-Unique: AE2bjIGkPNSpSXG2NTPowQ-1 Received: from smtp.corp.redhat.com (int-mx04.intmail.prod.int.phx2.redhat.com [10.5.11.14]) (using TLSv1.2 with cipher AECDH-AES256-SHA (256/256 bits)) (No client certificate requested) by mimecast-mx01.redhat.com (Postfix) with ESMTPS id 52CEF10066EE; Fri, 26 Feb 2021 14:38:22 +0000 (UTC) Received: from x1.localdomain (ovpn-112-36.ams2.redhat.com [10.36.112.36]) by smtp.corp.redhat.com (Postfix) with ESMTP id CA29E5D9E2; Fri, 26 Feb 2021 14:38:20 +0000 (UTC) From: Hans de Goede To: Cezary Rojewski , Pierre-Louis Bossart , Liam Girdwood , Jie Yang , Mark Brown Subject: [PATCH 1/5] ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor of 10 Date: Fri, 26 Feb 2021 15:38:13 +0100 Message-Id: <20210226143817.84287-2-hdegoede@redhat.com> In-Reply-To: <20210226143817.84287-1-hdegoede@redhat.com> References: <20210226143817.84287-1-hdegoede@redhat.com> MIME-Version: 1.0 X-Scanned-By: MIMEDefang 2.79 on 10.5.11.14 Authentication-Results: relay.mimecast.com; auth=pass smtp.auth=CUSA124A263 smtp.mailfrom=hdegoede@redhat.com X-Mimecast-Spam-Score: 0 X-Mimecast-Originator: redhat.com Cc: Oder Chiou , Hans de Goede , alsa-devel@alsa-project.org X-BeenThere: alsa-devel@alsa-project.org X-Mailman-Version: 2.1.15 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Errors-To: alsa-devel-bounces@alsa-project.org Sender: "Alsa-devel" The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede --- sound/soc/codecs/rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 1414ad15d01c..a5674c227b3a 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -339,9 +339,9 @@ static bool rt5640_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ From patchwork Fri Feb 26 14:38:14 2021 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Hans de Goede X-Patchwork-Id: 12106751 Return-Path: X-Spam-Checker-Version: SpamAssassin 3.4.0 (2014-02-07) on aws-us-west-2-korg-lkml-1.web.codeaurora.org X-Spam-Level: X-Spam-Status: No, score=-13.8 required=3.0 tests=BAYES_00,DKIM_SIGNED, DKIM_VALID,HEADER_FROM_DIFFERENT_DOMAINS,INCLUDES_CR_TRAILER,INCLUDES_PATCH, MAILING_LIST_MULTI,SPF_HELO_NONE,SPF_PASS autolearn=ham autolearn_force=no version=3.4.0 Received: from mail.kernel.org (mail.kernel.org [198.145.29.99]) by smtp.lore.kernel.org (Postfix) with ESMTP id ACAC1C433E0 for ; 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Fri, 26 Feb 2021 14:38:24 +0000 (UTC) Received: from x1.localdomain (ovpn-112-36.ams2.redhat.com [10.36.112.36]) by smtp.corp.redhat.com (Postfix) with ESMTP id 97AE35D9D2; Fri, 26 Feb 2021 14:38:22 +0000 (UTC) From: Hans de Goede To: Cezary Rojewski , Pierre-Louis Bossart , Liam Girdwood , Jie Yang , Mark Brown Subject: [PATCH 2/5] ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor of 10 Date: Fri, 26 Feb 2021 15:38:14 +0100 Message-Id: <20210226143817.84287-3-hdegoede@redhat.com> In-Reply-To: <20210226143817.84287-1-hdegoede@redhat.com> References: <20210226143817.84287-1-hdegoede@redhat.com> MIME-Version: 1.0 X-Scanned-By: MIMEDefang 2.79 on 10.5.11.14 Authentication-Results: relay.mimecast.com; auth=pass smtp.auth=CUSA124A263 smtp.mailfrom=hdegoede@redhat.com X-Mimecast-Spam-Score: 0 X-Mimecast-Originator: redhat.com Cc: Oder Chiou , Hans de Goede , alsa-devel@alsa-project.org X-BeenThere: alsa-devel@alsa-project.org X-Mailman-Version: 2.1.15 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Errors-To: alsa-devel-bounces@alsa-project.org Sender: "Alsa-devel" The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede --- sound/soc/codecs/rt5651.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index d198e191fb0c..e59fdc81dbd4 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -285,9 +285,9 @@ static bool rt5651_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ From patchwork Fri Feb 26 14:38:15 2021 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Hans de Goede X-Patchwork-Id: 12106753 Return-Path: X-Spam-Checker-Version: SpamAssassin 3.4.0 (2014-02-07) on aws-us-west-2-korg-lkml-1.web.codeaurora.org X-Spam-Level: X-Spam-Status: No, score=-13.8 required=3.0 tests=BAYES_00,DKIM_SIGNED, DKIM_VALID,HEADER_FROM_DIFFERENT_DOMAINS,INCLUDES_CR_TRAILER,INCLUDES_PATCH, MAILING_LIST_MULTI,SPF_HELO_NONE,SPF_PASS autolearn=ham autolearn_force=no version=3.4.0 Received: from mail.kernel.org (mail.kernel.org [198.145.29.99]) by smtp.lore.kernel.org (Postfix) with ESMTP id 7A514C433DB for ; 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Fri, 26 Feb 2021 14:38:28 +0000 (UTC) Received: from x1.localdomain (ovpn-112-36.ams2.redhat.com [10.36.112.36]) by smtp.corp.redhat.com (Postfix) with ESMTP id 6C92D5D9D2; Fri, 26 Feb 2021 14:38:24 +0000 (UTC) From: Hans de Goede To: Cezary Rojewski , Pierre-Louis Bossart , Liam Girdwood , Jie Yang , Mark Brown Subject: [PATCH 3/5] ASoC: rt5640: Add emulated 'DAC1 Playback Switch' control Date: Fri, 26 Feb 2021 15:38:15 +0100 Message-Id: <20210226143817.84287-4-hdegoede@redhat.com> In-Reply-To: <20210226143817.84287-1-hdegoede@redhat.com> References: <20210226143817.84287-1-hdegoede@redhat.com> MIME-Version: 1.0 X-Scanned-By: MIMEDefang 2.79 on 10.5.11.14 Authentication-Results: relay.mimecast.com; auth=pass smtp.auth=CUSA124A263 smtp.mailfrom=hdegoede@redhat.com X-Mimecast-Spam-Score: 0 X-Mimecast-Originator: redhat.com Cc: Oder Chiou , Hans de Goede , alsa-devel@alsa-project.org X-BeenThere: alsa-devel@alsa-project.org X-Mailman-Version: 2.1.15 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Errors-To: alsa-devel-bounces@alsa-project.org Sender: "Alsa-devel" When using AIF1 the 'DAC1 Playback Volume' control will be used as the PlaybackMasterElem in UCM. We need a matching 'DAC1 Playback Switch' for 2 reasons: 1. To be able to truely fully mute the output (the softest volume setting is not fully muted). 2. For reliable output-mute LED control. The path from the IF1_DAC data input to the 'Stereo DAC MIXL' / 'Stereo DAC MIXR' digital mixer has a 'DAC MIXL' / 'DAC MIXR' digital mixer with IF1_DAC data as one of its inputs direclty after the 'DAC1 Playback Volume' control. This commit adds an emulated "DAC1 Playback Switch" control by: 1. Declaring the enable flag for the mixers IF1_DAC input as well as the "DAC1 Playback Switch" control both as SND_SOC_NOPM controls. 2. Storing the settings of both controls as driver-private data. 3. Only clearing the mute flag for the IF1_DAC input of that mixer if the stored values indicate both controls are enabled. Signed-off-by: Hans de Goede --- sound/soc/codecs/rt5640.c | 96 +++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/rt5640.h | 4 ++ 2 files changed, 96 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index a5674c227b3a..c143ca174921 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -378,6 +378,56 @@ static const char * const rt5640_clsd_spk_ratio[] = {"1.66x", "1.83x", "1.94x", static SOC_ENUM_SINGLE_DECL(rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT, RT5640_CLSD_RATIO_SFT, rt5640_clsd_spk_ratio); +/* + * For reliable output-mute LED control we need a "DAC1 Playback Switch" control. + * We emulate this by only clearing the RT5640_M_IF1_DAC_L/_R AD_DA_MIXER register + * bits when both our emulated DAC1 Playback Switch control and the DAC1 MIXL/R + * DAPM-mixer DAC1 input are enabled. + */ +static void rt5640_update_ad_da_mixer_if1_dac_m_bits(struct rt5640_priv *rt5640) +{ + int val = RT5640_M_IF1_DAC_L | RT5640_M_IF1_DAC_R; + + if (rt5640->dac1_mixl_if1_switch && rt5640->dac1_playback_switch_l) + val &= ~RT5640_M_IF1_DAC_L; + + if (rt5640->dac1_mixr_if1_switch && rt5640->dac1_playback_switch_r) + val &= ~RT5640_M_IF1_DAC_R; + + regmap_update_bits(rt5640->regmap, RT5640_AD_DA_MIXER, + RT5640_M_IF1_DAC_L | RT5640_M_IF1_DAC_R, val); +} + +static int rt5640_dac1_playback_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = rt5640->dac1_playback_switch_l; + ucontrol->value.integer.value[1] = rt5640->dac1_playback_switch_r; + + return 0; +} + +static int rt5640_dac1_playback_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); + + if (rt5640->dac1_playback_switch_l == ucontrol->value.integer.value[0] && + rt5640->dac1_playback_switch_r == ucontrol->value.integer.value[1]) + return 0; + + rt5640->dac1_playback_switch_l = ucontrol->value.integer.value[0]; + rt5640->dac1_playback_switch_r = ucontrol->value.integer.value[1]; + + rt5640_update_ad_da_mixer_if1_dac_m_bits(rt5640); + + return 1; +} + static const struct snd_kcontrol_new rt5640_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL, @@ -400,6 +450,8 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE("DAC2 Playback Switch", RT5640_DAC2_CTRL, RT5640_M_DAC_L2_VOL_SFT, RT5640_M_DAC_R2_VOL_SFT, 1, 1), + SOC_DOUBLE_EXT("DAC1 Playback Switch", SND_SOC_NOPM, 0, 1, 1, 0, + rt5640_dac1_playback_switch_get, rt5640_dac1_playback_switch_put), SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5640_DAC1_DIG_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 175, 0, dac_vol_tlv), @@ -515,18 +567,44 @@ static const struct snd_kcontrol_new rt5640_mono_adc_r_mix[] = { RT5640_M_MONO_ADC_R2_SFT, 1, 1), }; +/* See comment above rt5640_update_ad_da_mixer_if1_dac_m_bits() */ +static int rt5640_put_dac1_mix_if1_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); + int ret; + + if (mc->shift == 0) + rt5640->dac1_mixl_if1_switch = ucontrol->value.integer.value[0]; + else + rt5640->dac1_mixr_if1_switch = ucontrol->value.integer.value[0]; + + /* Apply the update (if any) */ + ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); + if (ret == 0) + return 0; + + rt5640_update_ad_da_mixer_if1_dac_m_bits(rt5640); + + return 1; +} + +#define SOC_DAPM_SINGLE_RT5640_IF1_SW(name, shift) \ + SOC_SINGLE_EXT(name, SND_SOC_NOPM, shift, 1, 0, \ + snd_soc_dapm_get_volsw, rt5640_put_dac1_mix_if1_switch) + static const struct snd_kcontrol_new rt5640_dac_l_mix[] = { SOC_DAPM_SINGLE("Stereo ADC Switch", RT5640_AD_DA_MIXER, RT5640_M_ADCMIX_L_SFT, 1, 1), - SOC_DAPM_SINGLE("INF1 Switch", RT5640_AD_DA_MIXER, - RT5640_M_IF1_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE_RT5640_IF1_SW("INF1 Switch", 0), }; static const struct snd_kcontrol_new rt5640_dac_r_mix[] = { SOC_DAPM_SINGLE("Stereo ADC Switch", RT5640_AD_DA_MIXER, RT5640_M_ADCMIX_R_SFT, 1, 1), - SOC_DAPM_SINGLE("INF1 Switch", RT5640_AD_DA_MIXER, - RT5640_M_IF1_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE_RT5640_IF1_SW("INF1 Switch", 1), }; static const struct snd_kcontrol_new rt5640_sto_dac_l_mix[] = { @@ -2831,6 +2909,16 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, INIT_DELAYED_WORK(&rt5640->bp_work, rt5640_button_press_work); INIT_WORK(&rt5640->jack_work, rt5640_jack_work); + /* + * Enable the emulated "DAC1 Playback Switch" by default to avoid + * muting the output with older UCM profiles. + */ + rt5640->dac1_playback_switch_l = true; + rt5640->dac1_playback_switch_r = true; + /* The Power-On-Reset values for the DAC1 mixer have the INF1 input enabled. */ + rt5640->dac1_mixl_if1_switch = true; + rt5640->dac1_mixr_if1_switch = true; + /* Make sure work is stopped on probe-error / remove */ ret = devm_add_action_or_reset(&i2c->dev, rt5640_cancel_work, rt5640); if (ret) diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 4fd47f2b936b..0d029f5dbb61 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2135,6 +2135,10 @@ struct rt5640_priv { bool hp_mute; bool asrc_en; + bool dac1_mixl_if1_switch; + bool dac1_mixr_if1_switch; + bool dac1_playback_switch_l; + bool dac1_playback_switch_r; /* Jack and button detect data */ bool ovcd_irq_enabled; From patchwork Fri Feb 26 14:38:16 2021 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Hans de Goede X-Patchwork-Id: 12106755 Return-Path: X-Spam-Checker-Version: SpamAssassin 3.4.0 (2014-02-07) on aws-us-west-2-korg-lkml-1.web.codeaurora.org X-Spam-Level: X-Spam-Status: No, score=-13.8 required=3.0 tests=BAYES_00,DKIM_SIGNED, DKIM_VALID,HEADER_FROM_DIFFERENT_DOMAINS,INCLUDES_CR_TRAILER,INCLUDES_PATCH, MAILING_LIST_MULTI,SPF_HELO_NONE,SPF_PASS autolearn=ham autolearn_force=no version=3.4.0 Received: from mail.kernel.org (mail.kernel.org [198.145.29.99]) by smtp.lore.kernel.org (Postfix) with ESMTP id C8AE9C43381 for ; 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Fri, 26 Feb 2021 14:38:30 +0000 (UTC) Received: from x1.localdomain (ovpn-112-36.ams2.redhat.com [10.36.112.36]) by smtp.corp.redhat.com (Postfix) with ESMTP id 69B0D5D9D2; Fri, 26 Feb 2021 14:38:28 +0000 (UTC) From: Hans de Goede To: Cezary Rojewski , Pierre-Louis Bossart , Liam Girdwood , Jie Yang , Mark Brown Subject: [PATCH 4/5] ASoC: rt5640: Rename 'Mono DAC Playback Volume' to 'DAC2 Playback Volume' Date: Fri, 26 Feb 2021 15:38:16 +0100 Message-Id: <20210226143817.84287-5-hdegoede@redhat.com> In-Reply-To: <20210226143817.84287-1-hdegoede@redhat.com> References: <20210226143817.84287-1-hdegoede@redhat.com> MIME-Version: 1.0 X-Scanned-By: MIMEDefang 2.79 on 10.5.11.14 Authentication-Results: relay.mimecast.com; auth=pass smtp.auth=CUSA124A263 smtp.mailfrom=hdegoede@redhat.com X-Mimecast-Spam-Score: 0 X-Mimecast-Originator: redhat.com Cc: Oder Chiou , Hans de Goede , alsa-devel@alsa-project.org X-BeenThere: alsa-devel@alsa-project.org X-Mailman-Version: 2.1.15 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Errors-To: alsa-devel-bounces@alsa-project.org Sender: "Alsa-devel" Rename 'Mono DAC Playback Volume' to 'DAC2 Playback Volume' and move it from rt5640_specific_snd_controls[] to rt5640_snd_controls[]. The RT5640_DAC2_DIG_VOL register controlled by this mixer-element has nothing to do with the Mono (Amplified) output which is only available on the ALC5640 chip and not on the ALC5642 chip. The RT5640_DAC2_DIG_VOL volume-control is the main volume control for audio coming from the I2S2 / AIF2 input of the chip and as such is also available on the ALC5642. This commit results in the following userspace visible changes: 1. On devices with an ACL5640 codec, the 'Mono DAC Playback Volume' control is renamed to 'DAC2 Playback Volume' allowing the alsa-lib mixer code to properly group it with the 'DAC2 Playback Switch' which is controlling the mute bits in the RT5640_DAC2_DIG_VOL register. Note the removal of the 'Mono DAC Playback Volume' is not an issue for userspace because the UCM profiles do not use it (the UCM profiles are shared betweent the 5640 and 5642 and only the 5640 had this control). 2. On devices with an ACL5642 codec, there now will be a new 'DAC2 Playback Volume', grouped with the 'DAC2 Playback Switch' Having a complete 'DAC2 Playback Volume' / 'DAC2 Playback Switch' pair on both variants will allow enabling hardware-volume control by setting the UCM PlaybackMasterElem to "DAC2" on devices where the I2S2/AIF2 interface of the codec is used. Signed-off-by: Hans de Goede --- sound/soc/codecs/rt5640.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index c143ca174921..38cc3155bf25 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -450,6 +450,9 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE("DAC2 Playback Switch", RT5640_DAC2_CTRL, RT5640_M_DAC_L2_VOL_SFT, RT5640_M_DAC_R2_VOL_SFT, 1, 1), + SOC_DOUBLE_TLV("DAC2 Playback Volume", RT5640_DAC2_DIG_VOL, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, + 175, 0, dac_vol_tlv), SOC_DOUBLE_EXT("DAC1 Playback Switch", SND_SOC_NOPM, 0, 1, 1, 0, rt5640_dac1_playback_switch_get, rt5640_dac1_playback_switch_put), SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5640_DAC1_DIG_VOL, @@ -495,9 +498,6 @@ static const struct snd_kcontrol_new rt5640_specific_snd_controls[] = { /* MONO Output Control */ SOC_SINGLE("Mono Playback Switch", RT5640_MONO_OUT, RT5640_L_MUTE_SFT, 1, 1), - - SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5640_DAC2_DIG_VOL, - RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 175, 0, dac_vol_tlv), }; 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auth=pass smtp.auth=CUSA124A263 smtp.mailfrom=hdegoede@redhat.com X-Mimecast-Spam-Score: 0 X-Mimecast-Originator: redhat.com Cc: Oder Chiou , Hans de Goede , alsa-devel@alsa-project.org X-BeenThere: alsa-devel@alsa-project.org X-Mailman-Version: 2.1.15 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Errors-To: alsa-devel-bounces@alsa-project.org Sender: "Alsa-devel" Depending on which AIF is used the UCM profile needs to setup a different path through the rt5640's "Digital Mixer Path" graph. ATM the UCM profiles solve this by just enabling paths to the outputs / from the input from both AIF1 and AIF2 and then relying on the DAPM framework to power-down the parts of the graph connected to the unused AIF. But in order to be able to use hardware-volumecontrol and to use the hardware mute controls, which are necessary for mute LED control, the UCM profiles need to know which AIF is actually being used. Add a new "aif:1" or "aif:2" part to the component string to provide info about the used AIF to userspace / to the UCM profiles. Note the size of byt_rt5640_components is not increased because the size of 32 chars already is big enough. Signed-off-by: Hans de Goede --- sound/soc/intel/boards/bytcr_rt5640.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 5d48cc359c3d..1f6a636571c2 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1252,6 +1252,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) int ret_val = 0; int dai_index = 0; int i, cfg_spk; + int aif; is_bytcr = false; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); @@ -1363,8 +1364,12 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) log_quirks(&pdev->dev); if ((byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2) || - (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) + (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) { byt_rt5640_dais[dai_index].codecs->dai_name = "rt5640-aif2"; + aif = 2; + } else { + aif = 1; + } if ((byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) || (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) @@ -1402,8 +1407,8 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) } snprintf(byt_rt5640_components, sizeof(byt_rt5640_components), - "cfg-spk:%d cfg-mic:%s", cfg_spk, - map_name[BYT_RT5640_MAP(byt_rt5640_quirk)]); + "cfg-spk:%d cfg-mic:%s aif:%d", cfg_spk, + map_name[BYT_RT5640_MAP(byt_rt5640_quirk)], aif); byt_rt5640_card.components = byt_rt5640_components; #if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES) snprintf(byt_rt5640_long_name, sizeof(byt_rt5640_long_name),