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

[alsa-devel] [PATCH v7 0/9] ASoC: add OF-graph base simple-card
by Kuninori Morimoto 17 May '17

17 May '17

Hi Mark, Rob These are v7 of OF-graph sound card support. Main change from v6 "label" doesn't allow [prefix]label (in [6/9]) 1) - 4) : OF-graph base functions 5) - 9) : OF-graph + simple-card (= audio_graph_card) Kuninori Morimoto (9): of-graph: export symbol of_phandle_iterator_init/next of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_endpoint_count() ASoC: soc-core: enable "dai-format" on snd_soc_of_parse_daifmt() ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() ASoC: add audio-graph-card document ASoC: add audio-graph-card support .../devicetree/bindings/sound/audio-graph-card.txt | 124 +++++++++ drivers/of/base.c | 62 ++++- include/linux/of_graph.h | 21 ++ include/sound/simple_card_utils.h | 10 + sound/soc/generic/Kconfig | 8 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-card.c | 308 +++++++++++++++++++++ sound/soc/generic/simple-card-utils.c | 73 ++++- sound/soc/soc-core.c | 10 +- 9 files changed, 600 insertions(+), 18 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 sound/soc/generic/audio-graph-card.c -- 1.9.1

3 16

[alsa-devel] [PATCH v5 0/9] ASoC: add OF-graph base simple-card
by Kuninori Morimoto 17 May '17

17 May '17

Hi Rob, Mark These are v5 of OF-graph base audio card patch-set. It fixed small things which was pointed by Rob. 1) - 3) : new OF-graph helper functions 4) - 5) : expand "label", "dai-format" 6) - 9) : OF-graph base simple-card Kuninori Morimoto (9): of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_endpoint_count() ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name ASoC: soc-core: enable "dai-name" on snd_soc_of_parse_daifmt() ASoC: add snd_soc_get_dai_id() ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() ASoC: add audio-graph-card document ASoC: add audio-graph-card support .../devicetree/bindings/sound/audio-graph-card.txt | 124 +++++++++ drivers/of/base.c | 60 +++- include/linux/of_graph.h | 21 ++ include/sound/simple_card_utils.h | 10 + include/sound/soc.h | 1 + sound/soc/generic/Kconfig | 8 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-card.c | 308 +++++++++++++++++++++ sound/soc/generic/simple-card-utils.c | 54 +++- sound/soc/soc-core.c | 32 ++- 10 files changed, 604 insertions(+), 16 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 sound/soc/generic/audio-graph-card.c -- 1.9.1

2 12

[alsa-devel] [PATCH 0/2] ASoC: rsnd: tidyup Document
by Kuninori Morimoto 17 May '17

17 May '17

Hi Mark Cc Geert These patches tidyup Document for Renesas sound driver Kuninori Morimoto (2): ASoC: rsnd: move "renesas,rsrc-card" to "simple-scu-audio-card" on Document ASoC: rsnd: add missing clocks/clock-names on Document .../devicetree/bindings/sound/renesas,rsnd.txt | 34 ++++++++++++---------- 1 file changed, 19 insertions(+), 15 deletions(-) -- 1.9.1

3 6

[alsa-devel] [PATCH] ASoC: codec: wm9860: avoid maybe-uninitialized warning
by Arnd Bergmann 15 May '17

15 May '17

The new PLL configuration code triggers a harmless warning: sound/soc/codecs/wm8960.c: In function 'wm8960_configure_clocking': sound/soc/codecs/wm8960.c:735:3: error: 'best_freq_out' may be used uninitialized in this function [-Werror=maybe-uninitialized] wm8960_set_pll(codec, freq_in, best_freq_out); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/wm8960.c:699:12: note: 'best_freq_out' was declared here I think the warning was introduced by Daniel's bugfix. I've come up with a way to simplify the code in a way that is more readable to both humans and to gcc, which gets us rid of the warning. Fixes: 84fdc00d519f ("ASoC: codec: wm9860: Refactor PLL out freq search") Fixes: 303e8954af8d ("ASoC: codec: wm8960: Stop when a matching PLL freq is found") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/wm8960.c | 25 ++++++++----------------- 1 file changed, 8 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index ace69da97cb8..ba0f80504c6b 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -696,14 +696,12 @@ int wm8960_configure_pll(struct snd_soc_codec *codec, int freq_in, { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); int sysclk, bclk, lrclk, freq_out; - int diff, best_freq_out; + int diff; int i, j, k; bclk = wm8960->bclk; lrclk = wm8960->lrclk; - *bclk_idx = -1; - for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { if (sysclk_divs[i] == -1) continue; @@ -720,22 +718,14 @@ int wm8960_configure_pll(struct snd_soc_codec *codec, int freq_in, *sysclk_idx = i; *dac_idx = j; *bclk_idx = k; - best_freq_out = freq_out; - break; + return freq_out; } } - if (k != ARRAY_SIZE(bclk_divs)) - break; } - if (j != ARRAY_SIZE(dac_divs)) - break; } - - if (*bclk_idx != -1) - wm8960_set_pll(codec, freq_in, best_freq_out); - - return *bclk_idx; + return -EINVAL; } + static int wm8960_configure_clocking(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); @@ -783,11 +773,12 @@ static int wm8960_configure_clocking(struct snd_soc_codec *codec) } } - ret = wm8960_configure_pll(codec, freq_in, &i, &j, &k); - if (ret < 0) { + freq_out = wm8960_configure_pll(codec, freq_in, &i, &j, &k); + if (freq_out < 0) { dev_err(codec->dev, "failed to configure clock via PLL\n"); - return -EINVAL; + return freq_out; } + wm8960_set_pll(codec, freq_in, freq_out); configure_clock: /* configure sysclk clock */ -- 2.9.0

4 7

[alsa-devel] [PATCH] ASoC: ak4613: disable asymmetric audio interface format
by Kuninori Morimoto 08 May '17

08 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Asymmetric audio interface formats exist in ak4613 by same register settings. Capture Playback 24bit LEFT_J 16bit RIGHT_J 24bit LEFT_J 20bit RIGHT_J 24bit LEFT_J 24bit RIGHT_J 24bit LEFT_J 24bit LEFT_J 24bit I2S 24bit I2S These asymmetric formats makes driver / behavior difficult. It is not HW limitation, but SW limitation. To makes code reading easy, this patch doesn't remove code, but just comment out. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4613.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index b2dfdde..9d2be7c 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -139,9 +139,9 @@ struct ak4613_priv { #define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt } static const struct ak4613_interface ak4613_iface[] = { /* capture */ /* playback */ - [0] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(16, RIGHT_J) }, - [1] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(20, RIGHT_J) }, - [2] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, RIGHT_J) }, +/* [0] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(16, RIGHT_J) }, */ +/* [1] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(20, RIGHT_J) }, */ +/* [2] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, RIGHT_J) }, */ [3] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, LEFT_J) }, [4] = { AUDIO_IFACE(24, I2S), AUDIO_IFACE(24, I2S) }, }; @@ -262,7 +262,7 @@ static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) fmt &= SND_SOC_DAIFMT_FORMAT_MASK; switch (fmt) { - case SND_SOC_DAIFMT_RIGHT_J: +/* case SND_SOC_DAIFMT_RIGHT_J:*/ case SND_SOC_DAIFMT_LEFT_J: case SND_SOC_DAIFMT_I2S: priv->fmt = fmt; @@ -420,7 +420,7 @@ static int ak4613_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_96000 |\ SNDRV_PCM_RATE_176400 |\ SNDRV_PCM_RATE_192000) -#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S16_LE |\ +#define AK4613_PCM_FMTBIT (/*SNDRV_PCM_FMTBIT_S16_LE |*/ \ SNDRV_PCM_FMTBIT_S24_LE) static struct snd_soc_dai_driver ak4613_dai = { -- 1.9.1

2 2

[alsa-devel] [PATCH v2 00/11] drm/i915: LPE audio runtime PM and multipipe (v2)
by ville.syrjala＠linux.intel.com 03 May '17

03 May '17

From: Ville Syrjälä <ville.syrjala(a)linux.intel.com> Okay, here's the second attempt at getting multiple pipes playing back audio on the VLV/CHV HDMI LPE audio device. The main change from v1 is that now the PCM devices are associated with ports instead of pipes, so the audio from one device always gets output on the same display. I've also tacked on the alsa-lib conf update. No clue whether it's really correct or not (the config language isn't a close friend of mine). BTW I did notice that with LPE audio all the controls say iface=PCM, whereas on HDA a bunch of them say iface=MIXER. No idea if that's OK or not, just something I spotted when I was comparing the results with HDA. Entire series available here: git://github.com/vsyrjala/linux.git lpe_audio_multipipe_2 Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Ville Syrjälä (11): drm/i915: Fix runtime PM for LPE audio ALSA: x86: Clear the pdata.notify_lpe_audio pointer before teardown drm/i915: Stop pretending to mask/unmask LPE audio interrupts drm/i915: Remove the unused pending_notify from LPE platform data drm/i915: Replace tmds_clock_speed and link_rate with just ls_clock drm/i915: Remove hdmi_connected from LPE audio pdata drm/i915: Reorganize intel_lpe_audio_notify() arguments drm/i915: Clean up the LPE audio platform data ALSA: x86: Prepare LPE audio ctls for multiple PCMs ALSA: x86: Split snd_intelhad into card and PCM specific structures ALSA: x86: Register multiple PCM devices for the LPE audio card drivers/gpu/drm/i915/i915_drv.h | 4 +- drivers/gpu/drm/i915/i915_irq.c | 15 +- drivers/gpu/drm/i915/intel_audio.c | 19 +- drivers/gpu/drm/i915/intel_lpe_audio.c | 99 ++++------ include/drm/intel_lpe_audio.h | 22 +-- sound/x86/intel_hdmi_audio.c | 328 ++++++++++++++++++++------------- sound/x86/intel_hdmi_audio.h | 20 +- 7 files changed, 271 insertions(+), 236 deletions(-) -- 2.10.2

4 24

[alsa-devel] [PATCH] ASoC: snd_soc_set_dmi_name will use DMI_SYS_VENDOR if available
by mengdong.lin＠linux.intel.com 02 May '17

02 May '17

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This fix is based on Daniel Drake's suggestion: most x86 machines have a meaningfull DMI_SYS_VENDOR rather than DMI_BOARD_VENDOR. So snd_soc_set_dmi_name will try to get vendor name from DMI_SYS_VENDOR at first, and if that's null, then use DMI_BOARD_VENDOR. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 525f2f3..45f0a34 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1974,7 +1974,9 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) return 0; /* long name already set by driver or from DMI */ /* make up dmi long name as: vendor.product.version.board */ - vendor = dmi_get_system_info(DMI_BOARD_VENDOR); + vendor = dmi_get_system_info(DMI_SYS_VENDOR); + if (!vendor) + vendor = dmi_get_system_info(DMI_BOARD_VENDOR); if (!vendor) { dev_warn(card->dev, "ASoC: no DMI vendor name!\n"); return 0; -- 2.5.0

4 3

[alsa-devel] snd_soc_set_dmi_name - Shouldn't it use SYS_VENDOR?
by Daniel Drake 02 May '17

02 May '17

Hi, Nice to see DMI info being attached to SoC sound devices so that we can differentiate between differerent products in userspace, thanks for the effort there. However I just noticed that it using DMI_BOARD_VENDOR. Is this intentional? I have several examples of products here where DMI_BOARD_VENDOR is not set to a meaningful value, but DMI_SYS_VENDOR is just fine. DMI quirks in the kernel ordinarily tend to match products by combining DMI_SYS_VENDOR with DMI_PRODUCT_NAME. In the SMBIOS data format, Type 1 (System information) includes DMI_SYS_VENDOR and DMI_PRODUCT_NAME. Type 2 (Base board information) includes DMI_BOARD_VENDOR and DMI_BOARD_NAME. See dmi_decode() in drivers/firmware/dmi_scan.c It seems strange that ASoC is pulling the product name from Type 1 but the vendor from type 2. Can we make it instead just use Type 1 data, DMI_SYS_VENDOR and DMI_PRODUCT_NAME? Thanks Daniel

6 10

[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

4 7

[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");

2 2