April 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v3 0/2] Add STM32 SAI support
by olivier moysan 02 May '17

02 May '17

This patch-set handles the Serial Audio Interface (SAI) IP on STM32 platforms. The SAI IP is composed of two Sub-block A and B. (see the figure below) Each sub-block makes use of individual and shared resources. Shared resources: - Reset line. Reset common and sub-block registers. - Bus interface clocks. This clock is not exposed in device as it it managed by clock driver, according to kernel clocks gating. - Common register. SAI IP exhibits a common configuration register to manage synchronization modes. NB: These modes are not yet implemented in this version of the SAI driver. - Interrupt. Sub-blocks have their own interrupt status registers but they share the same interrupt line. Individual resources: - Register set - DMA request line. - Communication interface. Each sub-block has its own GPIOs and associated bus lines. - Kernel clock. Each sub-block has its own dedicated clock for its communication interface. To reflect this architecture Sub-block A and B are handled by 2 child devices. Sub-block A and B devices can be configured independently either as transmitter or receiver. A PCM device is associated to each sub-block. A sub-block has to select the appropriated parent clock at runtime, depending on the audio stream sampling rate to be played or captured. Two parent clocks must be provided to support sampling rates multiples of 8 kHz or 11.025kHz. Interface clock +--------------------------------+ +---------------> | SAI IP | Reset | +-----------------+ | +---------------> | |common registers | | | +-----------------+ | | | | +----------------------+ | | | Sub-block A | | Kernel clock A | | | | Bus A +--------------------> | +----------------+ +----------------> | | |A registers | | | | | +----------------+ | | | +----------------------+ | | | | +----------------------+ | | | Sub-block B | | Kernel clock B | | | | Bus B +--------------------> | +----------------+ +-----------------> | | |B registers | | | | | +----------------+ | | | +----------------------+ | | | +--------------------------------+ v3: correct DT bindings - remove aliases - use audio graph card olivier moysan (2): ASoC: stm32: add bindings for SAI ASoC: stm32: add SAI driver .../devicetree/bindings/sound/st,stm32-sai.txt | 89 +++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/stm/Kconfig | 8 + sound/soc/stm/Makefile | 6 + sound/soc/stm/stm32_sai.c | 115 +++ sound/soc/stm/stm32_sai.h | 200 +++++ sound/soc/stm/stm32_sai_sub.c | 884 +++++++++++++++++++++ 8 files changed, 1304 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.txt create mode 100644 sound/soc/stm/Kconfig create mode 100644 sound/soc/stm/Makefile create mode 100644 sound/soc/stm/stm32_sai.c create mode 100644 sound/soc/stm/stm32_sai.h create mode 100644 sound/soc/stm/stm32_sai_sub.c -- 1.9.1

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[alsa-devel] new codec tda1543
by George Makdessi 02 May '17

02 May '17

Commit 9f79c0289fa4b02544c6a51c789c84edd6a479fc ("audio: add support for philips tda1543 dac chip") new simple driver for the old well known philips tda1543 dac chip. Hello, We would like to contribute with an audio codec file for the well known Philips TDA1543 dac chip. We have already made a PR for Raspbian with Raspberry Pi specific driver that is well suited together with this codec. We have already got OK confirmation for our PR request and we just need to upstream this codec to finalize the system for our new commersial dac module for Raspberyy PI. https://github.com/McDaisy/linux-1/commit/9f79c0289fa4b02544c6a51c789c84edd… Thanks! BR George The code: /* * Driver for the tda1543 codec * * Author: Georgios F <georgios(a)gdis.se> * Copyright 2017 * based on code by Florian Meier <florian.meier(a)koalo.de> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License * version 2 as published by the Free Software Foundation. * * 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. */ #include <linux/init.h> #include <linux/module.h> #include <linux/platform_device.h> #include <sound/soc.h> static struct snd_soc_dai_driver tda1543_dai = { .name = "tda1543-hifi", .playback = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE }, }; static struct snd_soc_codec_driver soc_codec_dev_tda1543; static int tda1543_probe(struct platform_device *pdev) { return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_tda1543, &tda1543_dai, 1); } static int tda1543_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; } static const struct of_device_id tda1543_of_match[] = { { .compatible = "ti,tda1543", }, { } }; MODULE_DEVICE_TABLE(of, tda1543_of_match); static struct platform_driver tda1543_codec_driver = { .probe = tda1543_probe, .remove = tda1543_remove, .driver = { .name = "tda1543-codec", .owner = THIS_MODULE, .of_match_table = of_match_ptr(tda1543_of_match), }, }; module_platform_driver(tda1543_codec_driver); MODULE_AUTHOR("Georgios F <georgios(a)gdis.se>"); MODULE_DESCRIPTION("ASoC tda1543 codec driver"); MODULE_LICENSE("GPL v2");

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[alsa-devel] [PATCH v2 0/2] Relax bitclk computation when using PLL
by Daniel Baluta 30 Apr '17

30 Apr '17

Using strict bitclk requirements we cannot support all promised rates and formats. For this reason we relax bitclk computation by choosing the best available bitclk. First patch in the series is based on Arnd's patch: http://mailman.alsa-project.org/pipermail/alsa-devel/2017-April/119899.html Second one does the actual bitclk relaxation. Daniel Baluta (2): ASoC: codec: wm9860: avoid maybe-uninitialized warning ASoC: codec: wm8960: Relax bit clock computation when using PLL sound/soc/codecs/wm8960.c | 36 +++++++++++++++++++++--------------- 1 file changed, 21 insertions(+), 15 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v2] ASoC: dwc: Disallow building designware_pcm as a module
by Jose Abreu 30 Apr '17

30 Apr '17

Designware PCM is an extension to Designware I2S and they are dependent on each other. For this reason, make Designware PCM a boolean which will compile with Desigwnare I2S module. The name of the module is not changed but the name of the files need to be changed. Also, without this commit we get errors when probbing designware_i2s module because of unspecified license: designware_pcm: module license 'unspecified' taints kernel. Disabling lock debugging due to kernel taint designware_pcm: Unknown symbol __rcu_read_lock (err 0) designware_pcm: Unknown symbol devm_snd_soc_register_platform (err 0) designware_pcm: Unknown symbol synchronize_rcu (err 0) designware_pcm: Unknown symbol __rcu_read_unlock (err 0) designware_pcm: Unknown symbol snd_soc_set_runtime_hwparams (err 0) So, this is really needed as a fix. Fixes: 79361b2b98b7 ("ASoC: dwc: Add PIO PCM extension") Signed-off-by: Lubomir Rintel <lkundrak(a)v3.sk> Signed-off-by: Jose Abreu <joabreu(a)synopsys.com> Cc: Carlos Palminha <palminha(a)synopsys.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Lubomir Rintel <lkundrak(a)v3.sk> --- Changes v1->v2: - Refrased commit message - Removed EXPORT_SYMBOL_GPL as this isn't needed anymore - Added my self (Jose Abreu) as commiter - Changed name of files to dwc-i2s and dwc-pcm sound/soc/dwc/Kconfig | 4 ++-- sound/soc/dwc/Makefile | 6 +++--- sound/soc/dwc/{designware_i2s.c => dwc-i2s.c} | 0 sound/soc/dwc/{designware_pcm.c => dwc-pcm.c} | 3 --- 4 files changed, 5 insertions(+), 8 deletions(-) rename sound/soc/dwc/{designware_i2s.c => dwc-i2s.c} (100%) rename sound/soc/dwc/{designware_pcm.c => dwc-pcm.c} (98%) diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig index c297efe..c6fd95f 100644 --- a/sound/soc/dwc/Kconfig +++ b/sound/soc/dwc/Kconfig @@ -8,10 +8,10 @@ config SND_DESIGNWARE_I2S maximum of 8 channels each for play and record. config SND_DESIGNWARE_PCM - tristate "PCM PIO extension for I2S driver" + bool "PCM PIO extension for I2S driver" depends on SND_DESIGNWARE_I2S help - Say Y, M or N if you want to add a custom ALSA extension that registers + Say Y or N if you want to add a custom ALSA extension that registers a PCM and uses PIO to transfer data. This functionality is specially suited for I2S devices that don't have diff --git a/sound/soc/dwc/Makefile b/sound/soc/dwc/Makefile index 38f1ca3..3e24c0f 100644 --- a/sound/soc/dwc/Makefile +++ b/sound/soc/dwc/Makefile @@ -1,5 +1,5 @@ # SYNOPSYS Platform Support obj-$(CONFIG_SND_DESIGNWARE_I2S) += designware_i2s.o -ifdef CONFIG_SND_DESIGNWARE_PCM -obj-$(CONFIG_SND_DESIGNWARE_I2S) += designware_pcm.o -endif + +designware_i2s-y := dwc-i2s.o +designware_i2s-$(CONFIG_SND_DESIGNWARE_PCM) += dwc-pcm.o diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/dwc-i2s.c similarity index 100% rename from sound/soc/dwc/designware_i2s.c rename to sound/soc/dwc/dwc-i2s.c diff --git a/sound/soc/dwc/designware_pcm.c b/sound/soc/dwc/dwc-pcm.c similarity index 98% rename from sound/soc/dwc/designware_pcm.c rename to sound/soc/dwc/dwc-pcm.c index 459ec86..406fd867 100644 --- a/sound/soc/dwc/designware_pcm.c +++ b/sound/soc/dwc/dwc-pcm.c @@ -129,13 +129,11 @@ void dw_pcm_push_tx(struct dw_i2s_dev *dev) { dw_pcm_transfer(dev, true); } -EXPORT_SYMBOL_GPL(dw_pcm_push_tx); void dw_pcm_pop_rx(struct dw_i2s_dev *dev) { dw_pcm_transfer(dev, false); } -EXPORT_SYMBOL_GPL(dw_pcm_pop_rx); static int dw_pcm_open(struct snd_pcm_substream *substream) { @@ -281,4 +279,3 @@ int dw_pcm_register(struct platform_device *pdev) { return devm_snd_soc_register_platform(&pdev->dev, &dw_pcm_platform); } -EXPORT_SYMBOL_GPL(dw_pcm_register); -- 1.9.1

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[alsa-devel] Applied "ASoC: cs4271: configure reset GPIO as output" to the asoc tree
by Mark Brown 30 Apr '17

30 Apr '17

The patch ASoC: cs4271: configure reset GPIO as output has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 49b2e27ab9f66b0a22c21980ad8118a4038324ae Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin <alexander.sverdlin(a)gmail.com> Date: Sat, 29 Apr 2017 12:19:33 +0200 Subject: [PATCH] ASoC: cs4271: configure reset GPIO as output During reset "refactoring" the output configuration was lost. This commit repairs sound on EDB93XX boards. Fixes: 9a397f4 ("ASoC: cs4271: add regulator consumer support") Signed-off-by: Alexander Sverdlin <alexander.sverdlin(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: <stable(a)vger.kernel.org> # 4.6+ --- sound/soc/codecs/cs4271.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 8c0f3b89b5bc..e78b5f055f25 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -498,7 +498,7 @@ static int cs4271_reset(struct snd_soc_codec *codec) struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); if (gpio_is_valid(cs4271->gpio_nreset)) { - gpio_set_value(cs4271->gpio_nreset, 0); + gpio_direction_output(cs4271->gpio_nreset, 0); mdelay(1); gpio_set_value(cs4271->gpio_nreset, 1); mdelay(1); -- 2.11.0

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[alsa-devel] [bug report] ASoC: sti: Add CPU DAI driver for playback
by Dan Carpenter 30 Apr '17

30 Apr '17

Hello Arnaud Pouliquen, The patch 76c2145ded6b: "ASoC: sti: Add CPU DAI driver for playback" from Jun 22, 2015, leads to the following static checker warning: sound/soc/sti/uniperif_player.c:1077 uni_player_init() warn: unused return: ret = PTR_ERR() sound/soc/sti/uniperif_player.c 1047 int uni_player_init(struct platform_device *pdev, 1048 struct uniperif *player) 1049 { 1050 int ret = 0; 1051 1052 player->dev = &pdev->dev; 1053 player->state = UNIPERIF_STATE_STOPPED; 1054 player->dai_ops = &uni_player_dai_ops; 1055 1056 /* Get PCM_CLK_SEL & PCMP_VALID_SEL from audio-glue-ctrl SoC reg */ 1057 ret = uni_player_parse_dt_audio_glue(pdev, player); 1058 1059 if (ret < 0) { 1060 dev_err(player->dev, "Failed to parse DeviceTree\n"); 1061 return ret; 1062 } 1063 1064 /* Underflow recovery is only supported on later ip revisions */ 1065 if (player->ver >= SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) 1066 player->underflow_enabled = 1; 1067 1068 if (UNIPERIF_TYPE_IS_TDM(player)) 1069 player->hw = &uni_tdm_hw; 1070 else 1071 player->hw = &uni_player_pcm_hw; 1072 1073 /* Get uniperif resource */ 1074 player->clk = of_clk_get(pdev->dev.of_node, 0); 1075 if (IS_ERR(player->clk)) { 1076 dev_err(player->dev, "Failed to get clock\n"); 1077 ret = PTR_ERR(player->clk); ^^^^^^^^^^^^^^^^^^^^^^^^^^ This should probably be "return PTR_ERR(player->clk);? Can we run without a clk? 1078 } 1079 1080 /* Select the frequency synthesizer clock */ 1081 if (player->clk_sel) { 1082 ret = regmap_field_write(player->clk_sel, 1); 1083 if (ret) { 1084 dev_err(player->dev, 1085 "%s: Failed to select freq synth clock\n", 1086 __func__); 1087 return ret; 1088 } 1089 } regards, dan carpenter

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[alsa-devel] [PATCH 1/3] ASoC: Intel: Skylake: Fix uninitialized pointer usage
by Vinod Koul 30 Apr '17

30 Apr '17

Due to recent code move, one instance dev_err() was passed a wrong uninitialized pointer for the device, so fix it. sound/soc/intel/skylake/bxt-sst.c: In function 'bxt_sst_dsp_init': sound/soc/intel/skylake/bxt-sst.c:567:14: warning: 'skl' may be used uninitialized in this function [-Wmaybe-uninitialized] dev_err(skl->dev, "%s: no device\n", __func__); ^ Fixes: 9fe9c7119283 ("ASoC: Intel: Skylake: Move sst common initialization to a helper function") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/bxt-sst.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index fde4bc0f35b0..f5e7dbb1ba39 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -564,7 +564,7 @@ int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, ret = skl_sst_ctx_init(dev, irq, fw_name, dsp_ops, dsp, &skl_dev); if (ret < 0) { - dev_err(skl->dev, "%s: no device\n", __func__); + dev_err(dev, "%s: no device\n", __func__); return ret; } -- 1.9.1

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[alsa-devel] [PATCH] ASoC: nau8824: rename controls to match DAPM controls
by John Hsu 30 Apr '17

30 Apr '17

Rename the name of kcontrols to match up the DAPM widget controls. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- sound/soc/codecs/nau8824.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 18552ed..cd358be0 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -362,27 +362,27 @@ static const struct snd_kcontrol_new nau8824_snd_controls[] = { SOC_ENUM("ADC Decimation Rate", nau8824_adc_decimation_enum), SOC_ENUM("DAC Oversampling Rate", nau8824_dac_oversampl_enum), - SOC_SINGLE_TLV("Speaker Right from DACR Volume", + SOC_SINGLE_TLV("Speaker Right DACR Volume", NAU8824_REG_CLASSD_GAIN_1, 8, 0x1f, 0, spk_vol_tlv), - SOC_SINGLE_TLV("Speaker Left from DACL Volume", + SOC_SINGLE_TLV("Speaker Left DACL Volume", NAU8824_REG_CLASSD_GAIN_2, 0, 0x1f, 0, spk_vol_tlv), - SOC_SINGLE_TLV("Speaker Left from DACR Volume", + SOC_SINGLE_TLV("Speaker Left DACR Volume", NAU8824_REG_CLASSD_GAIN_1, 0, 0x1f, 0, spk_vol_tlv), - SOC_SINGLE_TLV("Speaker Right from DACL Volume", + SOC_SINGLE_TLV("Speaker Right DACL Volume", NAU8824_REG_CLASSD_GAIN_2, 8, 0x1f, 0, spk_vol_tlv), - SOC_SINGLE_TLV("Headphone Right from DACR Volume", + SOC_SINGLE_TLV("Headphone Right DACR Volume", NAU8824_REG_ATT_PORT0, 8, 0x1f, 0, hp_vol_tlv), - SOC_SINGLE_TLV("Headphone Left from DACL Volume", + SOC_SINGLE_TLV("Headphone Left DACL Volume", NAU8824_REG_ATT_PORT0, 0, 0x1f, 0, hp_vol_tlv), - SOC_SINGLE_TLV("Headphone Right from DACL Volume", + SOC_SINGLE_TLV("Headphone Right DACL Volume", NAU8824_REG_ATT_PORT1, 8, 0x1f, 0, hp_vol_tlv), - SOC_SINGLE_TLV("Headphone Left from DACR Volume", + SOC_SINGLE_TLV("Headphone Left DACR Volume", NAU8824_REG_ATT_PORT1, 0, 0x1f, 0, hp_vol_tlv), - SOC_SINGLE_TLV("Mic1 Volume", NAU8824_REG_FEPGA_II, + SOC_SINGLE_TLV("MIC1 Volume", NAU8824_REG_FEPGA_II, NAU8824_FEPGA_GAINL_SFT, 0x12, 0, mic_vol_tlv), - SOC_SINGLE_TLV("Mic2 Volume", NAU8824_REG_FEPGA_II, + SOC_SINGLE_TLV("MIC2 Volume", NAU8824_REG_FEPGA_II, NAU8824_FEPGA_GAINR_SFT, 0x12, 0, mic_vol_tlv), SOC_SINGLE_TLV("DMIC1 Volume", NAU8824_REG_ADC_CH0_DGAIN_CTRL, -- 2.6.4

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[alsa-devel] [PATCH] ASoC: nau8824: leave Class D gain at chip default
by John Hsu 30 Apr '17

30 Apr '17

Remove initial configuration of Class D gain for 1R and 2L. Leave them at the chip default. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- sound/soc/codecs/nau8824.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index cd358be0..cca974d 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1626,12 +1626,6 @@ static void nau8824_init_regs(struct nau8824 *nau8824) regmap_update_bits(regmap, NAU8824_REG_DAC_FILTER_CTRL_1, NAU8824_DAC_CICCLP_OFF | NAU8824_DAC_OVERSAMPLE_MASK, NAU8824_DAC_CICCLP_OFF | NAU8824_DAC_OVERSAMPLE_64); - /* Class D gain 9db for 1R and 2L */ - regmap_update_bits(regmap, NAU8824_REG_CLASSD_GAIN_1, - NAU8824_CLASSD_GAIN_1R_MASK, - (0xa << NAU8824_CLASSD_GAIN_1R_SFT)); - regmap_update_bits(regmap, NAU8824_REG_CLASSD_GAIN_2, - NAU8824_CLASSD_GAIN_2L_MASK, 0xa); /* DAC clock delay 2ns, VREF */ regmap_update_bits(regmap, NAU8824_REG_RDAC, NAU8824_RDAC_CLK_DELAY_MASK | NAU8824_RDAC_VREF_MASK, -- 2.6.4

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[alsa-devel] [PATCH] ALSA: ali5451: fix spelling mistake in "ali_capture_preapre"
by Colin King 30 Apr '17

30 Apr '17

From: Colin Ian King <colin.king(a)canonical.com> trivial fix to spelling mistake in dev_warn message, "ali_capture_preapre" should be "ali_capture_prepare" Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/ali5451/ali5451.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 92b819e4f729..34bbc2e730a6 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -1339,7 +1339,7 @@ static int snd_ali_prepare(struct snd_pcm_substream *substream) rate = snd_ali_get_spdif_in_rate(codec); if (rate == 0) { dev_warn(codec->card->dev, - "ali_capture_preapre: spdif rate detect err!\n"); + "ali_capture_prepare: spdif rate detect err!\n"); rate = 48000; } spin_lock_irq(&codec->reg_lock); -- 2.11.0

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