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

[alsa-devel] [PATCH] ALSA: pcm: Revert capture stream behavior change in blocking mode
by Takashi Iwai 07 Feb '19

07 Feb '19

In the commit 62ba568f7aef ("ALSA: pcm: Return 0 when size < start_threshold in capture"), we changed the behavior of __snd_pcm_lib_xfer() to return immediately with 0 when a capture stream has a high start_threshold. This was intended to be a correction of the behavior consistency and looked harmless, but this was the culprit of the recent breakage reported by syzkaller, which was fixed by the commit e190161f96b8 ("ALSA: pcm: Fix tight loop of OSS capture stream"). At the time for the OSS fix, I didn't touch the behavior for ALSA native API, as assuming that this behavior actually is good. But this turned out to be also broken actually for a similar deployment, e.g. one thread goes to a write loop in blocking mode while another thread controls the start/stop of the stream manually. Overall, the original commit is harmful, and it brings less merit to keep that behavior. Let's revert it. Fixes: 62ba568f7aef ("ALSA: pcm: Return 0 when size < start_threshold in capture") Fixes: e190161f96b8 ("ALSA: pcm: Fix tight loop of OSS capture stream") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 20 ++++---------------- 1 file changed, 4 insertions(+), 16 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 5957aeb1099e..bcb06bd3d81d 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2112,13 +2112,6 @@ int pcm_lib_apply_appl_ptr(struct snd_pcm_substream *substream, return 0; } -/* allow waiting for a capture stream that hasn't been started */ -#if IS_ENABLED(CONFIG_SND_PCM_OSS) -#define wait_capture_start(substream) ((substream)->oss.oss) -#else -#define wait_capture_start(substream) false -#endif - /* the common loop for read/write data */ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, void *data, bool interleaved, @@ -2184,16 +2177,11 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, snd_pcm_update_hw_ptr(substream); if (!is_playback && - runtime->status->state == SNDRV_PCM_STATE_PREPARED) { - if (size >= runtime->start_threshold) { - err = snd_pcm_start(substream); - if (err < 0) - goto _end_unlock; - } else if (!wait_capture_start(substream)) { - /* nothing to do */ - err = 0; + runtime->status->state == SNDRV_PCM_STATE_PREPARED && + size >= runtime->start_threshold) { + err = snd_pcm_start(substream); + if (err < 0) goto _end_unlock; - } } avail = snd_pcm_avail(substream); -- 2.16.4

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[alsa-devel] [PATCH] UCM: bytcr/PlatformEnableSeq.conf update some comments
by Hans de Goede 07 Feb '19

07 Feb '19

Commit f91cc3c7d6b7 ("Update chtrt5645 ucm variants to use bytcr/PlatformEnableSeq.conf component") updated the following 2 comments: # codec0_out settings (used if SSP2 is connected to aif1) # modem_out settings (used if SSP0 is connected to aif2) Specifically it added the " to aif1" resp. " to aif2" part of the comments. This is not correct, AIF1 / AIF2 are something which is present on Realtek codecs only, and either one can be used indepedent of SSP0 or SSP2 being used (the comments in the chtrt5645 UCM profile before this change were wrong / outdated). Besides there not being any relationship between SSP0 or SSP2 being used, bytcr/PlatformEnableSeq.conf is also used with other codecs, e.g. the ESS8316 codec where this is not applicable at all. Therefor this commit removes the " to aif?" part of the comments again to avoid confusing people reading this in the future. Cc: Russell Parker <russell.parker7(a)gmail.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- src/conf/ucm/platforms/bytcr/PlatformEnableSeq.conf | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/conf/ucm/platforms/bytcr/PlatformEnableSeq.conf b/src/conf/ucm/platforms/bytcr/PlatformEnableSeq.conf index 6f5e899c..b5ee2b41 100644 --- a/src/conf/ucm/platforms/bytcr/PlatformEnableSeq.conf +++ b/src/conf/ucm/platforms/bytcr/PlatformEnableSeq.conf @@ -29,7 +29,7 @@ cset "name='pcm0_in Gain 0 Volume' 0" cset "name='pcm1_in Gain 0 Switch' off" cset "name='pcm1_in Gain 0 Volume' 0%" -# codec0_out settings (used if SSP2 is connected to aif1) +# codec0_out settings (used if SSP2 is connected) cset "name='codec_out0 mix 0 codec_in0 Switch' off" cset "name='codec_out0 mix 0 codec_in1 Switch' off" cset "name='codec_out0 mix 0 media_loop1_in Switch' off" @@ -40,7 +40,7 @@ cset "name='codec_out0 mix 0 sprot_loop_in Switch' off" cset "name='codec_out0 Gain 0 Switch' on" cset "name='codec_out0 Gain 0 Volume' 0" -# modem_out settings (used if SSP0 is connected to aif2) +# modem_out settings (used if SSP0 is connected) cset "name='modem_out mix 0 codec_in0 Switch' off" cset "name='modem_out mix 0 codec_in1 Switch' off" cset "name='modem_out mix 0 media_loop1_in Switch' off" -- 2.20.1

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[alsa-devel] [PATCH v2 2/2] ASoC: cs35l36: Add device tree documentation for CS35L36
by James Schulman 06 Feb '19

06 Feb '19

Add device tree documentation for Cirrus Logic CS35L36 speaker amplifier Signed-off-by: James Schulman <james.schulman(a)cirrus.com> Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Acked-by: Brian Austin <brian.austin(a)cirrus.com> --- .../devicetree/bindings/sound/cs35l36.txt | 168 +++++++++++++++++++++ 1 file changed, 168 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l36.txt diff --git a/Documentation/devicetree/bindings/sound/cs35l36.txt b/Documentation/devicetree/bindings/sound/cs35l36.txt new file mode 100644 index 0000000..912bd16 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l36.txt @@ -0,0 +1,168 @@ +CS35L36 Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l36" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + - cirrus,boost-ctl-millivolt : Boost Voltage Value. Configures the boost + converter's output voltage in mV. The range is from 2550mV to 12000mV with + increments of 50mV. + (Default) VP + + - cirrus,boost-peak-milliamp : Boost-converter peak current limit in mA. + Configures the peak current by monitoring the current through the boost FET. + Range starts at 1600mA and goes to a maximum of 4500mA with increments of + 50mA. + (Default) 4.50 Amps + + - cirrus,boost-ind-nanohenry : Inductor estimation LBST reference value. + Seeds the digital boost converter's inductor estimation block with the initial + inductance value to reference. + + 1000 = 1uH (Default) + 1200 = 1.2uH + +Optional properties: + - cirrus,multi-amp-mode : Boolean to determine if there are more than + one amplifier in the system. If more than one it is best to Hi-Z the ASP + port to prevent bus contention on the output signal + + - cirrus,boost-ctl-select : Boost conerter control source selection. + Selects the source of the BST_CTL target VBST voltage for the boost + converter to generate. + 0x00 - Control Port Value + 0x01 - Class H Tracking (Default) + 0x10 - MultiDevice Sync Value + + - cirrus,amp-pcm-inv : Boolean to determine Amplifier will invert incoming + PCM data + + - cirrus,imon-pol-inv : Boolean to determine Amplifier will invert the + polarity of outbound IMON feedback data + + - cirrus,vmon-pol-inv : Boolean to determine Amplifier will invert the + polarity of outbound VMON feedback data + + - cirrus,dcm-mode-enable : Boost converter automatic DCM Mode enable. + This enables the digital boost converter to operate in a low power + (Discontinuous Conduction) mode during low loading conditions. + + - cirrus,weak-fet-disable : Boolean : The strength of the output drivers is + reduced when operating in a Weak-FET Drive Mode and must not be used to drive + a large load. + + - cirrus,classh-wk-fet-delay : Weak-FET entry delay. Controls the delay + (in ms) before the Class H algorithm switches to the weak-FET voltage + (after the audio falls and remains below the value specified in WKFET_AMP_THLD). + + 0 = 0ms + 1 = 5ms + 2 = 10ms + 3 = 50ms + 4 = 100ms (Default) + 5 = 200ms + 6 = 500ms + 7 = 1000ms + + - cirrus,classh-weak-fet-thld-millivolt : Weak-FET amplifier drive threshold. + Configures the signal threshold at which the PWM output stage enters + weak-FET operation. The range is 50mV to 700mV in 50mV increments. + + - cirrus,temp-warn-threshold : Amplifier overtemperature warning threshold. + Configures the threshold at which the overtemperature warning condition occurs. + When the threshold is met, the overtemperature warning attenuation is applied + and the TEMP_WARN_EINT interrupt status bit is set. + If TEMP_WARN_MASK = 0, INTb is asserted. + + 0 = 105C + 1 = 115C + 2 = 125C (Default) + 3 = 135C + + - cirrus,irq-drive-select : Selects the driver type of the selected interrupt + output. + + 0 = Open-drain + 1 = Push-pull (Default) + + - cirrus,irq-gpio-select : Selects the pin to serve as the programmable + interrupt output. + + 0 = PDM_DATA / SWIRE_SD / INT (Default) + 1 = GPIO + +Optional properties for the "cirrus,vpbr-config" Sub-node + + - cirrus,vpbr-en : VBST brownout prevention enable. Configures whether the + VBST brownout prevention algorithm is enabled or disabled. + + 0 = VBST brownout prevention disabled (default) + 1 = VBST brownout prevention enabled + + See Section 7.31.1 VPBR Config for configuration options & further details + + - cirrus,vpbr-thld : Initial VPBR threshold. Configures the VP brownout + threshold voltage + + - cirrus,cirrus,vpbr-atk-rate : Attenuation attack step rate. Configures the + amount delay between consecutive volume attenuation steps when a brownout + condition is present and the VP brownout condition is in an attacking state. + + - cirrus,vpbr-atk-vol : VP brownout prevention step size. Configures the VP + brownout prevention attacking attenuation step size when operating in either + digital volume or analog gain modes. + + - cirrus,vpbr-max-attn : Maximum attenuation that the VP brownout prevention + can apply to the audio signal. + + - cirrus,vpbr-wait : Configures the delay time between a brownout condition + no longer being present and the VP brownout prevention entering an attenuation + release state. + + - cirrus,vpbr-rel-rate : Attenuation release step rate. Configures the delay + between consecutive volume attenuation release steps when a brownout condition + is not longer present and the VP brownout is in an attenuation release state. + + - cirrus,vpbr-mute-en : During the attack state, if the vpbr-max-attn value + is reached, the error condition still remains, and this bit is set, the audio + is muted. + +Example: + +cs35l36: cs35l36@40 { + compatible = "cirrus,cs35l36"; + reg = <0x40>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&gpio0 54 0>; + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + + cirrus,boost-ind-nanohenry = <1000>; + cirrus,boost-ctl-millivolt = <10000>; + cirrus,boost-peak-milliamp = <4500>; + cirrus,boost-ctl-select = <0x00>; + cirrus,weak-fet-delay = <0x04>; + cirrus,weak-fet-thld = <0x01>; + cirrus,temp-warn-threshold = <0x01>; + cirrus,multi-amp-mode; + cirrus,irq-drive-select = <0x01>; + cirrus,irq-gpio-select = <0x01>; + + cirrus,vpbr-config { + cirrus,vpbr-en = <0x00>; + cirrus,vpbr-thld = <0x05>; + cirrus,vpbr-atk-rate = <0x02>; + cirrus,vpbr-atk-vol = <0x01>; + cirrus,vpbr-max-attn = <0x09>; + cirrus,vpbr-wait = <0x01>; + cirrus,vpbr-rel-rate = <0x05>; + cirrus,vpbr-mute-en = <0x00>; + }; +}; -- 1.9.1

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[alsa-devel] Applied "ASoC: ssm2602: switch to SPDX identifier" to the asoc tree
by Mark Brown 06 Feb '19

06 Feb '19

The patch ASoC: ssm2602: switch to SPDX identifier has been applied to the asoc tree at https://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 47306401835a095843df8d87a51848f7e5e5099e Mon Sep 17 00:00:00 2001 From: Marco Felsch <m.felsch(a)pengutronix.de> Date: Wed, 6 Feb 2019 18:00:03 +0100 Subject: [PATCH] ASoC: ssm2602: switch to SPDX identifier Drop old license header and switch to SPDX-License-Identifier. Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ssm2602.c | 39 ++++++++++++-------------------------- 1 file changed, 12 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 6f79cb0fbc63..464a4d7873bb 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -1,30 +1,15 @@ -/* - * File: sound/soc/codecs/ssm2602.c - * Author: Cliff Cai <Cliff.Cai(a)analog.com> - * - * Created: Tue June 06 2008 - * Description: Driver for ssm2602 sound chip - * - * Modified: - * Copyright 2008 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// File: sound/soc/codecs/ssm2602.c +// Author: Cliff Cai <Cliff.Cai(a)analog.com> +// +// Created: Tue June 06 2008 +// Description: Driver for ssm2602 sound chip +// +// Modified: +// Copyright 2008 Analog Devices Inc. +// +// Bugs: Enter bugs at http://blackfin.uclinux.org/ #include <linux/delay.h> #include <linux/module.h> -- 2.20.1

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[alsa-devel] Applied "ASoC: dapm: fix out-of-bounds accesses to DAPM lookup tables" to the asoc tree
by Mark Brown 06 Feb '19

06 Feb '19

The patch ASoC: dapm: fix out-of-bounds accesses to DAPM lookup tables has been applied to the asoc tree at https://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 c16e12010060c6c7a31f08b4a99513064cb53b7d Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Tue, 5 Feb 2019 10:22:27 -0600 Subject: [PATCH] ASoC: dapm: fix out-of-bounds accesses to DAPM lookup tables KASAN reports and additional traces point to out-of-bounds accesses to the dapm_up_seq and dapm_down_seq lookup tables. The indices used are larger than the array definition. Fix by adding missing entries for the new widget types in these two lookup tables, and align them with PGA values. Also the sequences for the following widgets were not defined. Since their values defaulted to zero, assign them explicitly snd_soc_dapm_input snd_soc_dapm_output snd_soc_dapm_vmid snd_soc_dapm_siggen snd_soc_dapm_sink Fixes: 8a70b4544ef4 ('ASoC: dapm: Add new widget type for constructing DAPM graphs on DSPs.'). Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2c4c13419539..20bad755888b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -70,12 +70,16 @@ static int dapm_up_seq[] = { [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_supply] = 2, [snd_soc_dapm_micbias] = 3, + [snd_soc_dapm_vmid] = 3, [snd_soc_dapm_dai_link] = 2, [snd_soc_dapm_dai_in] = 4, [snd_soc_dapm_dai_out] = 4, [snd_soc_dapm_aif_in] = 4, [snd_soc_dapm_aif_out] = 4, [snd_soc_dapm_mic] = 5, + [snd_soc_dapm_siggen] = 5, + [snd_soc_dapm_input] = 5, + [snd_soc_dapm_output] = 5, [snd_soc_dapm_mux] = 6, [snd_soc_dapm_demux] = 6, [snd_soc_dapm_dac] = 7, @@ -83,11 +87,19 @@ static int dapm_up_seq[] = { [snd_soc_dapm_mixer] = 8, [snd_soc_dapm_mixer_named_ctl] = 8, [snd_soc_dapm_pga] = 9, + [snd_soc_dapm_buffer] = 9, + [snd_soc_dapm_scheduler] = 9, + [snd_soc_dapm_effect] = 9, + [snd_soc_dapm_src] = 9, + [snd_soc_dapm_asrc] = 9, + [snd_soc_dapm_encoder] = 9, + [snd_soc_dapm_decoder] = 9, [snd_soc_dapm_adc] = 10, [snd_soc_dapm_out_drv] = 11, [snd_soc_dapm_hp] = 11, [snd_soc_dapm_spk] = 11, [snd_soc_dapm_line] = 11, + [snd_soc_dapm_sink] = 11, [snd_soc_dapm_kcontrol] = 12, [snd_soc_dapm_post] = 13, }; @@ -100,13 +112,25 @@ static int dapm_down_seq[] = { [snd_soc_dapm_spk] = 3, [snd_soc_dapm_line] = 3, [snd_soc_dapm_out_drv] = 3, + [snd_soc_dapm_sink] = 3, [snd_soc_dapm_pga] = 4, + [snd_soc_dapm_buffer] = 4, + [snd_soc_dapm_scheduler] = 4, + [snd_soc_dapm_effect] = 4, + [snd_soc_dapm_src] = 4, + [snd_soc_dapm_asrc] = 4, + [snd_soc_dapm_encoder] = 4, + [snd_soc_dapm_decoder] = 4, [snd_soc_dapm_switch] = 5, [snd_soc_dapm_mixer_named_ctl] = 5, [snd_soc_dapm_mixer] = 5, [snd_soc_dapm_dac] = 6, [snd_soc_dapm_mic] = 7, + [snd_soc_dapm_siggen] = 7, + [snd_soc_dapm_input] = 7, + [snd_soc_dapm_output] = 7, [snd_soc_dapm_micbias] = 8, + [snd_soc_dapm_vmid] = 8, [snd_soc_dapm_mux] = 9, [snd_soc_dapm_demux] = 9, [snd_soc_dapm_aif_in] = 10, -- 2.20.1

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[alsa-devel] [PATCH v4 1/2] ASoC: dapm: fix out-of-bounds accesses to DAPM lookup tables
by Pierre-Louis Bossart 06 Feb '19

06 Feb '19

KASAN reports and additional traces point to out-of-bounds accesses to the dapm_up_seq and dapm_down_seq lookup tables. The indices used are larger than the array definition. Fix by adding missing entries for the new widget types in these two lookup tables, and align them with PGA values. Also the sequences for the following widgets were not defined. Since their values defaulted to zero, assign them explicitly snd_soc_dapm_input snd_soc_dapm_output snd_soc_dapm_vmid snd_soc_dapm_siggen snd_soc_dapm_sink Fixes: 8a70b4544ef4 ('ASoC: dapm: Add new widget type for constructing DAPM graphs on DSPs.'). Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/soc-dapm.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5b74dffc9c11..2bf5bab579de 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -70,12 +70,16 @@ static int dapm_up_seq[] = { [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_supply] = 2, [snd_soc_dapm_micbias] = 3, + [snd_soc_dapm_vmid] = 3, [snd_soc_dapm_dai_link] = 2, [snd_soc_dapm_dai_in] = 4, [snd_soc_dapm_dai_out] = 4, [snd_soc_dapm_aif_in] = 4, [snd_soc_dapm_aif_out] = 4, [snd_soc_dapm_mic] = 5, + [snd_soc_dapm_siggen] = 5, + [snd_soc_dapm_input] = 5, + [snd_soc_dapm_output] = 5, [snd_soc_dapm_mux] = 6, [snd_soc_dapm_demux] = 6, [snd_soc_dapm_dac] = 7, @@ -83,11 +87,19 @@ static int dapm_up_seq[] = { [snd_soc_dapm_mixer] = 8, [snd_soc_dapm_mixer_named_ctl] = 8, [snd_soc_dapm_pga] = 9, + [snd_soc_dapm_buffer] = 9, + [snd_soc_dapm_scheduler] = 9, + [snd_soc_dapm_effect] = 9, + [snd_soc_dapm_src] = 9, + [snd_soc_dapm_asrc] = 9, + [snd_soc_dapm_encoder] = 9, + [snd_soc_dapm_decoder] = 9, [snd_soc_dapm_adc] = 10, [snd_soc_dapm_out_drv] = 11, [snd_soc_dapm_hp] = 11, [snd_soc_dapm_spk] = 11, [snd_soc_dapm_line] = 11, + [snd_soc_dapm_sink] = 11, [snd_soc_dapm_kcontrol] = 12, [snd_soc_dapm_post] = 13, }; @@ -100,13 +112,25 @@ static int dapm_down_seq[] = { [snd_soc_dapm_spk] = 3, [snd_soc_dapm_line] = 3, [snd_soc_dapm_out_drv] = 3, + [snd_soc_dapm_sink] = 3, [snd_soc_dapm_pga] = 4, + [snd_soc_dapm_buffer] = 4, + [snd_soc_dapm_scheduler] = 4, + [snd_soc_dapm_effect] = 4, + [snd_soc_dapm_src] = 4, + [snd_soc_dapm_asrc] = 4, + [snd_soc_dapm_encoder] = 4, + [snd_soc_dapm_decoder] = 4, [snd_soc_dapm_switch] = 5, [snd_soc_dapm_mixer_named_ctl] = 5, [snd_soc_dapm_mixer] = 5, [snd_soc_dapm_dac] = 6, [snd_soc_dapm_mic] = 7, + [snd_soc_dapm_siggen] = 7, + [snd_soc_dapm_input] = 7, + [snd_soc_dapm_output] = 7, [snd_soc_dapm_micbias] = 8, + [snd_soc_dapm_vmid] = 8, [snd_soc_dapm_mux] = 9, [snd_soc_dapm_demux] = 9, [snd_soc_dapm_aif_in] = 10, -- 2.17.1

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[alsa-devel] Applied "ASoC: rsnd: fixup rsnd_ssi_master_clk_start() user count check" to the asoc tree
by Mark Brown 06 Feb '19

06 Feb '19

The patch ASoC: rsnd: fixup rsnd_ssi_master_clk_start() user count check has been applied to the asoc tree at https://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 d9111d36024de07784f2e1ba2ccf70b16035f378 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 5 Feb 2019 09:46:43 +0900 Subject: [PATCH] ASoC: rsnd: fixup rsnd_ssi_master_clk_start() user count check commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") added new rsnd_ssi_prepare() and moved rsnd_ssi_master_clk_start() to .prepare. But, ssi user count (= ssi->usrcnt) is incremented at .init (= rsnd_ssi_init()). Because of these timing exchange, ssi->usrcnt check at rsnd_ssi_master_clk_start() should be adjusted. Otherwise, 2nd master clock setup will be no check. This patch fixup this issue. Fixes: commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") Reported-by: Yusuke Goda <yusuke.goda.sx(a)renesas.com> Reported-by: Valentine Barshak <valentine.barshak(a)cogentembedded.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Yusuke Goda <yusuke.goda.sx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 45ef295743ec..f5afab631abb 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -286,7 +286,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_ssi_is_multi_slave(mod, io)) return 0; - if (ssi->usrcnt > 1) { + if (ssi->usrcnt > 0) { if (ssi->rate != rate) { dev_err(dev, "SSI parent/child should use same rate\n"); return -EINVAL; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: codecs: Add jz4725b-codec driver" to the asoc tree
by Mark Brown 06 Feb '19

06 Feb '19

The patch ASoC: codecs: Add jz4725b-codec driver has been applied to the asoc tree at https://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 e9d97b05a80f27d5ba7379b108db19b0d93cf267 Mon Sep 17 00:00:00 2001 From: Paul Cercueil <paul(a)crapouillou.net> Date: Tue, 5 Feb 2019 00:11:08 -0300 Subject: [PATCH] ASoC: codecs: Add jz4725b-codec driver Add jz4725b-codec driver to support the internal CODEC found in the JZ4725B SoC from Ingenic. Signed-off-by: Paul Cercueil <paul(a)crapouillou.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/jz4725b.c | 599 +++++++++++++++++++++++++++++++++++++ 3 files changed, 613 insertions(+) create mode 100644 sound/soc/codecs/jz4725b.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a15710c8a95f..fec894c725d3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -90,6 +90,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_INNO_RK3036 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC + select SND_SOC_JZ4725B_CODEC select SND_SOC_LM4857 if I2C select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C @@ -581,6 +582,17 @@ config SND_SOC_JZ4740_CODEC select REGMAP_MMIO tristate +config SND_SOC_JZ4725B_CODEC + depends on MIPS || COMPILE_TEST + select REGMAP + tristate "Ingenic JZ4725B internal CODEC" + help + Enable support for the internal CODEC found in the JZ4725B SoC + from Ingenic. + + This driver can also be built as a module. If so, the module + will be called snd-soc-jz4725b-codec. + config SND_SOC_L3 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3d7a59761c08..b07dfb5fa700 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -86,6 +86,7 @@ snd-soc-ics43432-objs := ics43432.o snd-soc-inno-rk3036-objs := inno_rk3036.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o +snd-soc-jz4725b-codec-objs := jz4725b.o snd-soc-l3-objs := l3.o snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o @@ -357,6 +358,7 @@ obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_INNO_RK3036) += snd-soc-inno-rk3036.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o +obj-$(CONFIG_SND_SOC_JZ4725B_CODEC) += snd-soc-jz4725b-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c new file mode 100644 index 000000000000..e3dba92f30bd --- /dev/null +++ b/sound/soc/codecs/jz4725b.c @@ -0,0 +1,599 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * JZ4725B CODEC driver + * + * Copyright (C) 2019, Paul Cercueil <paul(a)crapouillou.net> + */ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/io.h> +#include <linux/iopoll.h> +#include <linux/regmap.h> +#include <linux/clk.h> + +#include <linux/delay.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#define ICDC_RGADW_OFFSET 0x00 +#define ICDC_RGDATA_OFFSET 0x04 + +/* ICDC internal register access control register(RGADW) */ +#define ICDC_RGADW_RGWR BIT(16) + +#define ICDC_RGADW_RGADDR_OFFSET 8 +#define ICDC_RGADW_RGADDR_MASK GENMASK(14, ICDC_RGADW_RGADDR_OFFSET) + +#define ICDC_RGADW_RGDIN_OFFSET 0 +#define ICDC_RGADW_RGDIN_MASK GENMASK(7, ICDC_RGADW_RGDIN_OFFSET) + +/* ICDC internal register data output register (RGDATA)*/ +#define ICDC_RGDATA_IRQ BIT(8) + +#define ICDC_RGDATA_RGDOUT_OFFSET 0 +#define ICDC_RGDATA_RGDOUT_MASK GENMASK(7, ICDC_RGDATA_RGDOUT_OFFSET) + +/* JZ internal register space */ +enum { + JZ4725B_CODEC_REG_AICR, + JZ4725B_CODEC_REG_CR1, + JZ4725B_CODEC_REG_CR2, + JZ4725B_CODEC_REG_CCR1, + JZ4725B_CODEC_REG_CCR2, + JZ4725B_CODEC_REG_PMR1, + JZ4725B_CODEC_REG_PMR2, + JZ4725B_CODEC_REG_CRR, + JZ4725B_CODEC_REG_ICR, + JZ4725B_CODEC_REG_IFR, + JZ4725B_CODEC_REG_CGR1, + JZ4725B_CODEC_REG_CGR2, + JZ4725B_CODEC_REG_CGR3, + JZ4725B_CODEC_REG_CGR4, + JZ4725B_CODEC_REG_CGR5, + JZ4725B_CODEC_REG_CGR6, + JZ4725B_CODEC_REG_CGR7, + JZ4725B_CODEC_REG_CGR8, + JZ4725B_CODEC_REG_CGR9, + JZ4725B_CODEC_REG_CGR10, + JZ4725B_CODEC_REG_TR1, + JZ4725B_CODEC_REG_TR2, + JZ4725B_CODEC_REG_CR3, + JZ4725B_CODEC_REG_AGC1, + JZ4725B_CODEC_REG_AGC2, + JZ4725B_CODEC_REG_AGC3, + JZ4725B_CODEC_REG_AGC4, + JZ4725B_CODEC_REG_AGC5, +}; + +#define REG_AICR_CONFIG1_OFFSET 0 +#define REG_AICR_CONFIG1_MASK (0xf << REG_AICR_CONFIG1_OFFSET) + +#define REG_CR1_SB_MICBIAS_OFFSET 7 +#define REG_CR1_MONO_OFFSET 6 +#define REG_CR1_DAC_MUTE_OFFSET 5 +#define REG_CR1_HP_DIS_OFFSET 4 +#define REG_CR1_DACSEL_OFFSET 3 +#define REG_CR1_BYPASS_OFFSET 2 + +#define REG_CR2_DAC_DEEMP_OFFSET 7 +#define REG_CR2_DAC_ADWL_OFFSET 5 +#define REG_CR2_DAC_ADWL_MASK (0x3 << REG_CR2_DAC_ADWL_OFFSET) +#define REG_CR2_ADC_ADWL_OFFSET 3 +#define REG_CR2_ADC_ADWL_MASK (0x3 << REG_CR2_ADC_ADWL_OFFSET) +#define REG_CR2_ADC_HPF_OFFSET 2 + +#define REG_CR3_SB_MIC1_OFFSET 7 +#define REG_CR3_SB_MIC2_OFFSET 6 +#define REG_CR3_SIDETONE1_OFFSET 5 +#define REG_CR3_SIDETONE2_OFFSET 4 +#define REG_CR3_MICDIFF_OFFSET 3 +#define REG_CR3_MICSTEREO_OFFSET 2 +#define REG_CR3_INSEL_OFFSET 0 +#define REG_CR3_INSEL_MASK (0x3 << REG_CR3_INSEL_OFFSET) + +#define REG_CCR1_CONFIG4_OFFSET 0 +#define REG_CCR1_CONFIG4_MASK (0xf << REG_CCR1_CONFIG4_OFFSET) + +#define REG_CCR2_DFREQ_OFFSET 4 +#define REG_CCR2_DFREQ_MASK (0xf << REG_CCR2_DFREQ_OFFSET) +#define REG_CCR2_AFREQ_OFFSET 0 +#define REG_CCR2_AFREQ_MASK (0xf << REG_CCR2_AFREQ_OFFSET) + +#define REG_PMR1_SB_DAC_OFFSET 7 +#define REG_PMR1_SB_OUT_OFFSET 6 +#define REG_PMR1_SB_MIX_OFFSET 5 +#define REG_PMR1_SB_ADC_OFFSET 4 +#define REG_PMR1_SB_LIN_OFFSET 3 +#define REG_PMR1_SB_IND_OFFSET 0 + +#define REG_PMR2_LRGI_OFFSET 7 +#define REG_PMR2_RLGI_OFFSET 6 +#define REG_PMR2_LRGOD_OFFSET 5 +#define REG_PMR2_RLGOD_OFFSET 4 +#define REG_PMR2_GIM_OFFSET 3 +#define REG_PMR2_SB_MC_OFFSET 2 +#define REG_PMR2_SB_OFFSET 1 +#define REG_PMR2_SB_SLEEP_OFFSET 0 + +#define REG_IFR_RAMP_UP_DONE_OFFSET 3 +#define REG_IFR_RAMP_DOWN_DONE_OFFSET 2 + +#define REG_CGR1_GODL_OFFSET 4 +#define REG_CGR1_GODL_MASK (0xf << REG_CGR1_GODL_OFFSET) +#define REG_CGR1_GODR_OFFSET 0 +#define REG_CGR1_GODR_MASK (0xf << REG_CGR1_GODR_OFFSET) + +#define REG_CGR2_GO1R_OFFSET 0 +#define REG_CGR2_GO1R_MASK (0x1f << REG_CGR2_GO1R_OFFSET) + +#define REG_CGR3_GO1L_OFFSET 0 +#define REG_CGR3_GO1L_MASK (0x1f << REG_CGR3_GO1L_OFFSET) + +struct jz_icdc { + struct regmap *regmap; + void __iomem *base; + struct clk *clk; +}; + +static const SNDRV_CTL_TLVD_DECLARE_DB_LINEAR(jz4725b_dac_tlv, -2250, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_LINEAR(jz4725b_line_tlv, -1500, 600); + +static const struct snd_kcontrol_new jz4725b_codec_controls[] = { + SOC_DOUBLE_TLV("Master Playback Volume", + JZ4725B_CODEC_REG_CGR1, + REG_CGR1_GODL_OFFSET, + REG_CGR1_GODR_OFFSET, + 0xf, 1, jz4725b_dac_tlv), + SOC_DOUBLE_R_TLV("Master Capture Volume", + JZ4725B_CODEC_REG_CGR3, + JZ4725B_CODEC_REG_CGR2, + REG_CGR2_GO1R_OFFSET, + 0x1f, 1, jz4725b_line_tlv), + + SOC_SINGLE("Master Playback Switch", JZ4725B_CODEC_REG_CR1, + REG_CR1_DAC_MUTE_OFFSET, 1, 1), + + SOC_SINGLE("Deemphatize Filter Playback Switch", + JZ4725B_CODEC_REG_CR2, + REG_CR2_DAC_DEEMP_OFFSET, 1, 0), + + SOC_SINGLE("High-Pass Filter Capture Switch", + JZ4725B_CODEC_REG_CR2, + REG_CR2_ADC_HPF_OFFSET, 1, 0), +}; + +static const char * const jz4725b_codec_adc_src_texts[] = { + "Mic 1", "Mic 2", "Line In", "Mixer", +}; +static const unsigned int jz4725b_codec_adc_src_values[] = { 0, 1, 2, 3, }; +static const SOC_VALUE_ENUM_SINGLE_DECL(jz4725b_codec_adc_src_enum, + JZ4725B_CODEC_REG_CR3, + REG_CR3_INSEL_OFFSET, + REG_CR3_INSEL_MASK, + jz4725b_codec_adc_src_texts, + jz4725b_codec_adc_src_values); +static const struct snd_kcontrol_new jz4725b_codec_adc_src_ctrl = + SOC_DAPM_ENUM("Route", jz4725b_codec_adc_src_enum); + +static const struct snd_kcontrol_new jz4725b_codec_mixer_controls[] = { + SOC_DAPM_SINGLE("Line In Bypass", JZ4725B_CODEC_REG_CR1, + REG_CR1_BYPASS_OFFSET, 1, 0), +}; + +static int jz4725b_out_stage_enable(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *codec = snd_soc_dapm_to_component(w->dapm); + struct jz_icdc *icdc = snd_soc_component_get_drvdata(codec); + struct regmap *map = icdc->regmap; + unsigned int val; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + return regmap_update_bits(map, JZ4725B_CODEC_REG_IFR, + BIT(REG_IFR_RAMP_UP_DONE_OFFSET), 0); + case SND_SOC_DAPM_POST_PMU: + return regmap_read_poll_timeout(map, JZ4725B_CODEC_REG_IFR, + val, val & BIT(REG_IFR_RAMP_UP_DONE_OFFSET), + 100000, 500000); + case SND_SOC_DAPM_PRE_PMD: + return regmap_update_bits(map, JZ4725B_CODEC_REG_IFR, + BIT(REG_IFR_RAMP_DOWN_DONE_OFFSET), 0); + case SND_SOC_DAPM_POST_PMD: + return regmap_read_poll_timeout(map, JZ4725B_CODEC_REG_IFR, + val, val & BIT(REG_IFR_RAMP_DOWN_DONE_OFFSET), + 100000, 500000); + default: + return -EINVAL; + } +} + +static const struct snd_soc_dapm_widget jz4725b_codec_dapm_widgets[] = { + /* DAC */ + SND_SOC_DAPM_DAC("DAC", "Playback", + JZ4725B_CODEC_REG_PMR1, REG_PMR1_SB_DAC_OFFSET, 1), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC", "Capture", + JZ4725B_CODEC_REG_PMR1, REG_PMR1_SB_ADC_OFFSET, 1), + + SND_SOC_DAPM_MUX("ADC Source", SND_SOC_NOPM, 0, 0, + &jz4725b_codec_adc_src_ctrl), + + /* Mixer */ + SND_SOC_DAPM_MIXER("Mixer", JZ4725B_CODEC_REG_PMR1, + REG_PMR1_SB_MIX_OFFSET, 1, + jz4725b_codec_mixer_controls, + ARRAY_SIZE(jz4725b_codec_mixer_controls)), + SND_SOC_DAPM_MIXER("DAC to Mixer", JZ4725B_CODEC_REG_CR1, + REG_CR1_DACSEL_OFFSET, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Line In", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("HP Out", JZ4725B_CODEC_REG_CR1, + REG_CR1_HP_DIS_OFFSET, 1, NULL, 0), + + SND_SOC_DAPM_MIXER("Mic 1", JZ4725B_CODEC_REG_CR3, + REG_CR3_SB_MIC1_OFFSET, 1, NULL, 0), + SND_SOC_DAPM_MIXER("Mic 2", JZ4725B_CODEC_REG_CR3, + REG_CR3_SB_MIC2_OFFSET, 1, NULL, 0), + + SND_SOC_DAPM_MIXER_E("Out Stage", JZ4725B_CODEC_REG_PMR1, + REG_PMR1_SB_OUT_OFFSET, 1, NULL, 0, + jz4725b_out_stage_enable, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER("Mixer to ADC", JZ4725B_CODEC_REG_PMR1, + REG_PMR1_SB_IND_OFFSET, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Mic Bias", JZ4725B_CODEC_REG_CR1, + REG_CR1_SB_MICBIAS_OFFSET, 1, NULL, 0), + + /* Pins */ + SND_SOC_DAPM_INPUT("MIC1P"), + SND_SOC_DAPM_INPUT("MIC1N"), + SND_SOC_DAPM_INPUT("MIC2P"), + SND_SOC_DAPM_INPUT("MIC2N"), + + SND_SOC_DAPM_INPUT("LLINEIN"), + SND_SOC_DAPM_INPUT("RLINEIN"), + + SND_SOC_DAPM_OUTPUT("LHPOUT"), + SND_SOC_DAPM_OUTPUT("RHPOUT"), +}; + +static const struct snd_soc_dapm_route jz4725b_codec_dapm_routes[] = { + {"Mic 1", NULL, "MIC1P"}, + {"Mic 1", NULL, "MIC1N"}, + {"Mic 2", NULL, "MIC2P"}, + {"Mic 2", NULL, "MIC2N"}, + + {"Line In", NULL, "LLINEIN"}, + {"Line In", NULL, "RLINEIN"}, + + {"Mixer", "Line In Bypass", "Line In"}, + {"DAC to Mixer", NULL, "DAC"}, + {"Mixer", NULL, "DAC to Mixer"}, + + {"Mixer to ADC", NULL, "Mixer"}, + {"ADC Source", "Mixer", "Mixer to ADC"}, + {"ADC Source", "Line In", "Line In"}, + {"ADC Source", "Mic 1", "Mic 1"}, + {"ADC Source", "Mic 2", "Mic 2"}, + {"ADC", NULL, "ADC Source"}, + + {"Out Stage", NULL, "Mixer"}, + {"HP Out", NULL, "Out Stage"}, + {"LHPOUT", NULL, "HP Out"}, + {"RHPOUT", NULL, "HP Out"}, +}; + +static int jz4725b_codec_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct jz_icdc *icdc = snd_soc_component_get_drvdata(component); + struct regmap *map = icdc->regmap; + + switch (level) { + case SND_SOC_BIAS_ON: + regmap_update_bits(map, JZ4725B_CODEC_REG_PMR2, + BIT(REG_PMR2_SB_SLEEP_OFFSET), 0); + break; + case SND_SOC_BIAS_PREPARE: + /* Enable sound hardware */ + regmap_update_bits(map, JZ4725B_CODEC_REG_PMR2, + BIT(REG_PMR2_SB_OFFSET), 0); + msleep(224); + break; + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(map, JZ4725B_CODEC_REG_PMR2, + BIT(REG_PMR2_SB_SLEEP_OFFSET), + BIT(REG_PMR2_SB_SLEEP_OFFSET)); + break; + case SND_SOC_BIAS_OFF: + regmap_update_bits(map, JZ4725B_CODEC_REG_PMR2, + BIT(REG_PMR2_SB_OFFSET), + BIT(REG_PMR2_SB_OFFSET)); + break; + } + + return 0; +} + +static int jz4725b_codec_dev_probe(struct snd_soc_component *component) +{ + struct jz_icdc *icdc = snd_soc_component_get_drvdata(component); + struct regmap *map = icdc->regmap; + + clk_prepare_enable(icdc->clk); + + /* Write CONFIGn (n=1 to 8) bits. + * The value 0x0f is specified in the datasheet as a requirement. + */ + regmap_write(map, JZ4725B_CODEC_REG_AICR, + 0xf << REG_AICR_CONFIG1_OFFSET); + regmap_write(map, JZ4725B_CODEC_REG_CCR1, + 0x0 << REG_CCR1_CONFIG4_OFFSET); + + return 0; +} + +static void jz4725b_codec_dev_remove(struct snd_soc_component *component) +{ + struct jz_icdc *icdc = snd_soc_component_get_drvdata(component); + + clk_disable_unprepare(icdc->clk); +} + +static const struct snd_soc_component_driver jz4725b_codec = { + .probe = jz4725b_codec_dev_probe, + .remove = jz4725b_codec_dev_remove, + .set_bias_level = jz4725b_codec_set_bias_level, + .controls = jz4725b_codec_controls, + .num_controls = ARRAY_SIZE(jz4725b_codec_controls), + .dapm_widgets = jz4725b_codec_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(jz4725b_codec_dapm_widgets), + .dapm_routes = jz4725b_codec_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(jz4725b_codec_dapm_routes), + .suspend_bias_off = 1, + .use_pmdown_time = 1, +}; + +static const unsigned int jz4725b_codec_sample_rates[] = { + 96000, 48000, 44100, 32000, + 24000, 22050, 16000, 12000, + 11025, 9600, 8000, +}; + +static int jz4725b_codec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct jz_icdc *icdc = snd_soc_component_get_drvdata(dai->component); + unsigned int rate, bit_width; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + bit_width = 0; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + bit_width = 1; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + bit_width = 2; + break; + case SNDRV_PCM_FORMAT_S24_3LE: + bit_width = 3; + break; + default: + return -EINVAL; + } + + for (rate = 0; rate < ARRAY_SIZE(jz4725b_codec_sample_rates); rate++) { + if (jz4725b_codec_sample_rates[rate] == params_rate(params)) + break; + } + + if (rate == ARRAY_SIZE(jz4725b_codec_sample_rates)) + return -EINVAL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(icdc->regmap, + JZ4725B_CODEC_REG_CR2, + REG_CR2_DAC_ADWL_MASK, + bit_width << REG_CR2_DAC_ADWL_OFFSET); + + regmap_update_bits(icdc->regmap, + JZ4725B_CODEC_REG_CCR2, + REG_CCR2_DFREQ_MASK, + rate << REG_CCR2_DFREQ_OFFSET); + } else { + regmap_update_bits(icdc->regmap, + JZ4725B_CODEC_REG_CR2, + REG_CR2_ADC_ADWL_MASK, + bit_width << REG_CR2_ADC_ADWL_OFFSET); + + regmap_update_bits(icdc->regmap, + JZ4725B_CODEC_REG_CCR2, + REG_CCR2_AFREQ_MASK, + rate << REG_CCR2_AFREQ_OFFSET); + } + + return 0; +} + +static const struct snd_soc_dai_ops jz4725b_codec_dai_ops = { + .hw_params = jz4725b_codec_hw_params, +}; + +#define JZ_ICDC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_3LE) + +static struct snd_soc_dai_driver jz4725b_codec_dai = { + .name = "jz4725b-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = JZ_ICDC_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = JZ_ICDC_FORMATS, + }, + .ops = &jz4725b_codec_dai_ops, +}; + +static bool jz4725b_codec_volatile(struct device *dev, unsigned int reg) +{ + return reg == JZ4725B_CODEC_REG_IFR; +} + +static bool jz4725b_codec_can_access_reg(struct device *dev, unsigned int reg) +{ + return (reg != JZ4725B_CODEC_REG_TR1) && (reg != JZ4725B_CODEC_REG_TR2); +} + +static int jz4725b_codec_io_wait(struct jz_icdc *icdc) +{ + u32 reg; + + return readl_poll_timeout(icdc->base + ICDC_RGADW_OFFSET, reg, + !(reg & ICDC_RGADW_RGWR), 1000, 10000); +} + +static int jz4725b_codec_reg_read(void *context, unsigned int reg, + unsigned int *val) +{ + struct jz_icdc *icdc = context; + unsigned int i; + u32 tmp; + int ret; + + ret = jz4725b_codec_io_wait(icdc); + if (ret) + return ret; + + tmp = readl(icdc->base + ICDC_RGADW_OFFSET); + tmp = (tmp & ~ICDC_RGADW_RGADDR_MASK) + | (reg << ICDC_RGADW_RGADDR_OFFSET); + writel(tmp, icdc->base + ICDC_RGADW_OFFSET); + + /* wait 6+ cycles */ + for (i = 0; i < 6; i++) + *val = readl(icdc->base + ICDC_RGDATA_OFFSET) & + ICDC_RGDATA_RGDOUT_MASK; + + return 0; +} + +static int jz4725b_codec_reg_write(void *context, unsigned int reg, + unsigned int val) +{ + struct jz_icdc *icdc = context; + int ret; + + ret = jz4725b_codec_io_wait(icdc); + if (ret) + return ret; + + writel(ICDC_RGADW_RGWR | (reg << ICDC_RGADW_RGADDR_OFFSET) | val, + icdc->base + ICDC_RGADW_OFFSET); + + ret = jz4725b_codec_io_wait(icdc); + if (ret) + return ret; + + return 0; +} + +static const u8 jz4725b_codec_reg_defaults[] = { + 0x0c, 0xaa, 0x78, 0x00, 0x00, 0xff, 0x03, 0x51, + 0x3f, 0x00, 0x00, 0x04, 0x04, 0x04, 0x04, 0x04, + 0x04, 0x0a, 0x0a, 0x00, 0x00, 0x00, 0xc0, 0x34, + 0x07, 0x44, 0x1f, 0x00, +}; + +static const struct regmap_config jz4725b_codec_regmap_config = { + .reg_bits = 7, + .val_bits = 8, + + .max_register = JZ4725B_CODEC_REG_AGC5, + .volatile_reg = jz4725b_codec_volatile, + .readable_reg = jz4725b_codec_can_access_reg, + .writeable_reg = jz4725b_codec_can_access_reg, + + .reg_read = jz4725b_codec_reg_read, + .reg_write = jz4725b_codec_reg_write, + + .reg_defaults_raw = jz4725b_codec_reg_defaults, + .num_reg_defaults_raw = ARRAY_SIZE(jz4725b_codec_reg_defaults), + .cache_type = REGCACHE_FLAT, +}; + +static int jz4725b_codec_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct jz_icdc *icdc; + struct resource *mem; + int ret; + + icdc = devm_kzalloc(dev, sizeof(*icdc), GFP_KERNEL); + if (!icdc) + return -ENOMEM; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + icdc->base = devm_ioremap_resource(dev, mem); + if (IS_ERR(icdc->base)) + return PTR_ERR(icdc->base); + + icdc->regmap = devm_regmap_init(dev, NULL, icdc, + &jz4725b_codec_regmap_config); + if (IS_ERR(icdc->regmap)) + return PTR_ERR(icdc->regmap); + + icdc->clk = devm_clk_get(&pdev->dev, "aic"); + if (IS_ERR(icdc->clk)) + return PTR_ERR(icdc->clk); + + platform_set_drvdata(pdev, icdc); + + ret = devm_snd_soc_register_component(dev, &jz4725b_codec, + &jz4725b_codec_dai, 1); + if (ret) + dev_err(dev, "Failed to register codec\n"); + + return ret; +} + +#ifdef CONFIG_OF +static const struct of_device_id jz4725b_codec_of_matches[] = { + { .compatible = "ingenic,jz4725b-codec", }, + { } +}; +MODULE_DEVICE_TABLE(of, jz4725b_codec_of_matches); +#endif + +static struct platform_driver jz4725b_codec_driver = { + .probe = jz4725b_codec_probe, + .driver = { + .name = "jz4725b-codec", + .of_match_table = of_match_ptr(jz4725b_codec_of_matches), + }, +}; +module_platform_driver(jz4725b_codec_driver); + +MODULE_DESCRIPTION("JZ4725B SoC internal codec driver"); +MODULE_AUTHOR("Paul Cercueil <paul(a)crapouillou.net>"); +MODULE_LICENSE("GPL v2"); -- 2.20.1

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[alsa-devel] [PATCH 1/6] dt-bindings: sound: Document jz4740-codec bindings
by Paul Cercueil 06 Feb '19

06 Feb '19

Add documentation about how to probe the jz4740-codec driver from devicetree. Signed-off-by: Paul Cercueil <paul(a)crapouillou.net> --- .../bindings/sound/ingenic,jz4740-codec.txt | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt diff --git a/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt b/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt new file mode 100644 index 000000000000..1ffcade87e7b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt @@ -0,0 +1,20 @@ +Ingenic JZ4740 codec controller + +Required properties: +- compatible : "ingenic,jz4740-codec" +- reg : codec registers location and length +- clocks : phandle to the AIC clock. +- clock-names: must be set to "aic". +- #sound-dai-cells: Must be set to 0. + +Example: + +codec: audio-codec@10020080 { + compatible = "ingenic,jz4740-codec"; + reg = <0x10020080 0x8>; + + #sound-dai-cells = <0>; + + clocks = <&cgu JZ4740_CLK_AIC>; + clock-names = "aic"; +}; -- 2.11.0

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[alsa-devel] [PATCH 0/3] ALSA: Remove superfluous snd_info_register() calls
by Takashi Iwai 06 Feb '19

06 Feb '19

Hi, this is a fallout of the previous patchset, covering more places that call snd_info_register() unnecessarily. These should be applied on top of my previous patchset. The complete patches are found in topic/proc-fixes branch of sound git tree. thanks, Takashi === Takashi Iwai (3): ALSA: firewire: Remove superfluous snd_info_register() calls ALSA: opl4: Remove superfluous snd_info_register() calls ALSA: emux: Remove superfluous snd_info_register() calls sound/drivers/opl4/opl4_proc.c | 4 ---- sound/firewire/bebob/bebob_proc.c | 12 ++---------- sound/firewire/dice/dice-proc.c | 12 ++---------- sound/firewire/digi00x/digi00x-proc.c | 16 ++-------------- sound/firewire/fireface/ff-proc.c | 12 ++---------- sound/firewire/fireworks/fireworks_proc.c | 12 ++---------- sound/firewire/motu/motu-proc.c | 12 ++---------- sound/firewire/oxfw/oxfw-proc.c | 12 ++---------- sound/firewire/tascam/tascam-proc.c | 12 ++---------- sound/synth/emux/emux_proc.c | 4 ---- 10 files changed, 16 insertions(+), 92 deletions(-) -- 2.16.4

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