October 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/4 v5] i915 to call hda driver on HDMI plug/unplug
by David Henningsson 27 Nov '15

27 Nov '15

Hopefully last version? :-) * Added commit text about duplicate events (patch 4/4) * Added locks in bind/unbind on i915 side (patch 2/4) * Fixed docbook comments in i915 struct (patch 1/4) * Removed port_mst_streaming parameter (patch 1/4) * Added Reviewed-by - 1 & 2 by Jani, 3 & 4 by Takashi. Hopefully this was correct, otherwise feel free to adjust yourself before committing. Both Jani and Takashi seem okay with this going into 4.3. Nobody has said which tree you prefer to take it through, so how about Takashi merging it? David Henningsson (4): drm/i915: Add audio pin sense / ELD callback drm/i915: Call audio pin/ELD notify function ALSA: hda - allow codecs to access the i915 pin/ELD callback ALSA: hda - Wake the codec up on pin/ELD notify events drivers/gpu/drm/i915/i915_drv.h | 1 + drivers/gpu/drm/i915/intel_audio.c | 27 ++++++++++++++++++++++++--- include/drm/i915_component.h | 16 ++++++++++++++++ include/sound/hda_i915.h | 7 +++++++ sound/hda/hdac_i915.c | 10 ++++++++++ sound/pci/hda/patch_hdmi.c | 22 +++++++++++++++++++++- 6 files changed, 79 insertions(+), 4 deletions(-) -- 1.9.1

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[alsa-devel] [RFC PATCH 02/10] ASoC: Define soc_init_dai_link() to wrap link intialization.
by mengdong.lin＠intel.com 25 Nov '15

25 Nov '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Define soc_init_dai_link() to wrap link initialization, to reuse it later by snd_soc_instantiate_card() when adding new DAI links from topology in component probing phase. Move static func snd_soc_init_multicodec(), so that it can be reused by soc_init_dai_link(). This saves adding a function declaration for snd_soc_init_multicodec(). Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fb78674..e9e8956 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1141,6 +1141,100 @@ static void soc_remove_dai_links(struct snd_soc_card *card) } } +static int snd_soc_init_multicodec(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) +{ + /* Legacy codec/codec_dai link is a single entry in multicodec */ + if (dai_link->codec_name || dai_link->codec_of_node || + dai_link->codec_dai_name) { + dai_link->num_codecs = 1; + + dai_link->codecs = devm_kzalloc(card->dev, + sizeof(struct snd_soc_dai_link_component), + GFP_KERNEL); + if (!dai_link->codecs) + return -ENOMEM; + + dai_link->codecs[0].name = dai_link->codec_name; + dai_link->codecs[0].of_node = dai_link->codec_of_node; + dai_link->codecs[0].dai_name = dai_link->codec_dai_name; + } + + if (!dai_link->codecs) { + dev_err(card->dev, "ASoC: DAI link has no CODECs\n"); + return -EINVAL; + } + + return 0; +} + +static int soc_init_dai_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link) +{ + int i, ret; + + ret = snd_soc_init_multicodec(card, link); + if (ret) { + dev_err(card->dev, "ASoC: failed to init multicodec\n"); + return ret; + } + + for (i = 0; i < link->num_codecs; i++) { + /* + * Codec must be specified by 1 of name or OF node, + * not both or neither. + */ + if (!!link->codecs[i].name == + !!link->codecs[i].of_node) { + dev_err(card->dev, "ASoC: Neither/both codec name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + /* Codec DAI name must be specified */ + if (!link->codecs[i].dai_name) { + dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n", + link->name); + return -EINVAL; + } + } + + /* + * Platform may be specified by either name or OF node, but + * can be left unspecified, and a dummy platform will be used. + */ + if (link->platform_name && link->platform_of_node) { + dev_err(card->dev, + "ASoC: Both platform name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + + /* + * CPU device may be specified by either name or OF node, but + * can be left unspecified, and will be matched based on DAI + * name alone.. + */ + if (link->cpu_name && link->cpu_of_node) { + dev_err(card->dev, + "ASoC: Neither/both cpu name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + /* + * At least one of CPU DAI name or CPU device name/node must be + * specified + */ + if (!link->cpu_dai_name && + !(link->cpu_name || link->cpu_of_node)) { + dev_err(card->dev, + "ASoC: Neither cpu_dai_name nor cpu_name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + + return 0; +} + static void soc_set_name_prefix(struct snd_soc_card *card, struct snd_soc_component *component) { @@ -2363,33 +2457,6 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); -static int snd_soc_init_multicodec(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_link) -{ - /* Legacy codec/codec_dai link is a single entry in multicodec */ - if (dai_link->codec_name || dai_link->codec_of_node || - dai_link->codec_dai_name) { - dai_link->num_codecs = 1; - - dai_link->codecs = devm_kzalloc(card->dev, - sizeof(struct snd_soc_dai_link_component), - GFP_KERNEL); - if (!dai_link->codecs) - return -ENOMEM; - - dai_link->codecs[0].name = dai_link->codec_name; - dai_link->codecs[0].of_node = dai_link->codec_of_node; - dai_link->codecs[0].dai_name = dai_link->codec_dai_name; - } - - if (!dai_link->codecs) { - dev_err(card->dev, "ASoC: DAI link has no CODECs\n"); - return -EINVAL; - } - - return 0; -} - /** * snd_soc_register_card - Register a card with the ASoC core * @@ -2398,7 +2465,7 @@ static int snd_soc_init_multicodec(struct snd_soc_card *card, */ int snd_soc_register_card(struct snd_soc_card *card) { - int i, j, ret; + int i, ret; struct snd_soc_pcm_runtime *rtd; if (!card->name || !card->dev) @@ -2407,63 +2474,11 @@ int snd_soc_register_card(struct snd_soc_card *card) for (i = 0; i < card->num_links; i++) { struct snd_soc_dai_link *link = &card->dai_link[i]; - ret = snd_soc_init_multicodec(card, link); + ret = soc_init_dai_link(card, link); if (ret) { - dev_err(card->dev, "ASoC: failed to init multicodec\n"); - return ret; - } - - for (j = 0; j < link->num_codecs; j++) { - /* - * Codec must be specified by 1 of name or OF node, - * not both or neither. - */ - if (!!link->codecs[j].name == - !!link->codecs[j].of_node) { - dev_err(card->dev, "ASoC: Neither/both codec name/of_node are set for %s\n", - link->name); - return -EINVAL; - } - /* Codec DAI name must be specified */ - if (!link->codecs[j].dai_name) { - dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n", - link->name); - return -EINVAL; - } - } - - /* - * Platform may be specified by either name or OF node, but - * can be left unspecified, and a dummy platform will be used. - */ - if (link->platform_name && link->platform_of_node) { - dev_err(card->dev, - "ASoC: Both platform name/of_node are set for %s\n", + dev_err(card->dev, "ASoC: failed to init link %s\n", link->name); - return -EINVAL; - } - - /* - * CPU device may be specified by either name or OF node, but - * can be left unspecified, and will be matched based on DAI - * name alone.. - */ - if (link->cpu_name && link->cpu_of_node) { - dev_err(card->dev, - "ASoC: Neither/both cpu name/of_node are set for %s\n", - link->name); - return -EINVAL; - } - /* - * At least one of CPU DAI name or CPU device name/node must be - * specified - */ - if (!link->cpu_dai_name && - !(link->cpu_name || link->cpu_of_node)) { - dev_err(card->dev, - "ASoC: Neither cpu_dai_name nor cpu_name/of_node are set for %s\n", - link->name); - return -EINVAL; + return ret; } } -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Fix typo in kernel doc comment for snd_soc_put_volsw_sx()
by Takashi Iwai 25 Nov '15

25 Nov '15

Spotted by kbuild bot: sound/soc/soc-ops.c:415: warning: No description found for parameter 'ucontrol' sound/soc/soc-ops.c:415: warning: Excess function parameter 'uinfo' description in 'snd_soc_put_volsw_sx' Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index ecd38e52285a..2f67ba6d7a8f 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -404,7 +404,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); /** * snd_soc_put_volsw_sx - double mixer set callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to set the value of a double mixer control that spans 2 registers. * -- 2.6.2

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[alsa-devel] [PATCH v1 1/1] sst: replace custom implementation of readq / writeq
by Andy Shevchenko 24 Nov '15

24 Nov '15

The readq() and writeq() helpers are available in the asm-generic/io-64-nonatomic-hi-lo.h and asm-generic/io-64-nonatomic-lo-hi.h headers. Replace custom implementation by the generic helpers. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/atom/sst/sst_pvt.c | 16 ++++++---------- sound/soc/intel/common/sst-dsp.c | 9 ++++----- 2 files changed, 10 insertions(+), 15 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index adb32fe..6cc222d 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -31,7 +31,10 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <sound/compress_driver.h> + +#include <asm-generic/io-64-nonatomic-lo-hi.h> #include <asm/platform_sst_audio.h> + #include "../sst-mfld-platform.h" #include "sst.h" #include "../../common/sst-dsp.h" @@ -49,25 +52,18 @@ u32 sst_shim_read(void __iomem *addr, int offset) u64 sst_reg_read64(void __iomem *addr, int offset) { - u64 val = 0; - - memcpy_fromio(&val, addr + offset, sizeof(val)); - - return val; + return readq(addr + offset); } int sst_shim_write64(void __iomem *addr, int offset, u64 value) { - memcpy_toio(addr + offset, &value, sizeof(value)); + writeq(value, addr + offset); return 0; } u64 sst_shim_read64(void __iomem *addr, int offset) { - u64 val = 0; - - memcpy_fromio(&val, addr + offset, sizeof(val)); - return val; + return readq(addr + offset); } void sst_set_fw_state_locked( diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c index a627236..5274482 100644 --- a/sound/soc/intel/common/sst-dsp.c +++ b/sound/soc/intel/common/sst-dsp.c @@ -22,6 +22,8 @@ #include <linux/io.h> #include <linux/delay.h> +#include <asm-generic/io-64-nonatomic-lo-hi.h> + #include "sst-dsp.h" #include "sst-dsp-priv.h" @@ -43,16 +45,13 @@ EXPORT_SYMBOL_GPL(sst_shim32_read); void sst_shim32_write64(void __iomem *addr, u32 offset, u64 value) { - memcpy_toio(addr + offset, &value, sizeof(value)); + writeq(value, addr + offset); } EXPORT_SYMBOL_GPL(sst_shim32_write64); u64 sst_shim32_read64(void __iomem *addr, u32 offset) { - u64 val; - - memcpy_fromio(&val, addr + offset, sizeof(val)); - return val; + return readq(addr + offset); } EXPORT_SYMBOL_GPL(sst_shim32_read64); -- 2.5.0

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[alsa-devel] [PATCH v2 0/6] ASoC: Add HDA HDMI codec driver
by Vinod Koul 21 Nov '15

21 Nov '15

Hi all, This is second series of the HDA HDMI patches. This attempts to add HDA based HDMI driver and uses i915 component framework to interact with display for power up/down. The communication to codec is over HDA link If at all we need a change in interface, we can i915 interface which can be made transparent to users Due to dependency on already merged patches in ALSA tree, it would be best if this get merged thru ALSA tree Changes in v2: - remove infoframe open code and use video one - add pm on/off i915 triggers - add ext_dma_params struct Subhransu S. Prusty (6): ALSA: hdac: structure definition for ext_dma_params ASoC: hdac-hdmi: Add hdmi driver ASoC: hdac_hdmi: Add PM support for HDMI ASoC: hdac_hdmi: Add hdac hdmi dai ops ASoC: hdac_hdmi: Setup and start infoframe ASoC: hdac_hdmi: Use i915 component framework for PM include/sound/hdaudio_ext.h | 6 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_hdmi.c | 659 +++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 672 insertions(+) create mode 100644 sound/soc/codecs/hdac_hdmi.c -- 2.4.3

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[alsa-devel] [PATCH 00/13] dmaengine + omap drivers: support fro deferred probing
by Peter Ujfalusi 20 Nov '15

20 Nov '15

Hi, Vinod: as I promised: https://lkml.org/lkml/2015/5/8/80 With this series it is possible to put omap-dma or edma to even late_initcall and the drivers will figure out the load order fine(ish). We need to add dma_request_slave_channel_compat_reason() which is the equivalent of dma_request_slave_channel_compat() but returning with error codes in case of failure instead of NULL pointer. The rest of the series just converts the OMAP/daVinci drivers to use this new function to get the channel(s) and to handle the deferred probing. I did not moved the omap-dma, edma or ti-dma-crossbar from arch_initcall. If I do so UART will only comes up after the DMA driver is loaded since we are using, or going to use 8250 with DAM support. This delays the kernel messages. Other issue is the MMC/SD cards. On board with eMMC and SD card slot the mmcblk0/1 might get swapped due to different probe order for the MMC/SD drivers. For example in omap5-uevm: 1. omap-dma in arch_initcall the SD card is mmcblk1 (4809c000.mmc) and eMMC is mmcblk0 (480b4000.mmc) 2. omap-dma in late_initcall the SD card is mmcblk0 (4809c000.mmc) and eMMC is mmcblk1 (480b4000.mmc) Because in case 1 the 4809c000.mmc got deferred by missing regulator so 480b4000.mmc got registered first. In case 2 both deferring because of DMA and at the end 4809c000.mmc get registered first. So far I have not found a way to bind mmcblk0/1 to a specific node... Regards, Peter --- Peter Ujfalusi (13): dmaengine: of_dma: Correct return code for of_dma_request_slave_channel in case !CONFIG_OF dmaengine: Introduce dma_request_slave_channel_compat_reason() serial: 8250_dma: Support for deferred probing when requesting DMA channels mmc: omap_hsmmc: No need to check DMA channel validity at module remove mmc: omap_hsmmc: Support for deferred probing when requesting DMA channels mmc: omap: Support for deferred probing when requesting DMA channels mmc: davinci_mmc: Support for deferred probing when requesting DMA channels crypto: omap-aes - Support for deferred probing when requesting DMA channels crypto: omap-des - Support for deferred probing when requesting DMA channels crypto: omap-sham - Support for deferred probing when requesting DMA channel spi: omap2-mcspi: Support for deferred probing when requesting DMA channels [media] omap3isp: Support for deferred probing when requesting DMA channel ASoC: omap-pcm: Switch to use dma_request_slave_channel_compat_reason() drivers/crypto/omap-aes.c | 38 ++++++++++++++++--------------- drivers/crypto/omap-des.c | 38 ++++++++++++++++--------------- drivers/crypto/omap-sham.c | 15 ++++++++---- drivers/media/platform/omap3isp/isphist.c | 12 +++++++--- drivers/mmc/host/davinci_mmc.c | 26 ++++++++++++--------- drivers/mmc/host/omap.c | 20 ++++++++++++---- drivers/mmc/host/omap_hsmmc.c | 28 ++++++++++------------- drivers/spi/spi-omap2-mcspi.c | 36 +++++++++++++++++------------ drivers/tty/serial/8250/8250_dma.c | 18 +++++++-------- include/linux/dmaengine.h | 22 ++++++++++++++++++ include/linux/of_dma.h | 2 +- sound/soc/omap/omap-pcm.c | 16 ++++++++----- 12 files changed, 164 insertions(+), 107 deletions(-) -- 2.3.5

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[alsa-devel] [PATCH] ASoC: adau1701: Add 44.1 KHz to supported sample rates
by pascal.huerst＠gmail.com 19 Nov '15

19 Nov '15

From: Pascal Huerst <pascal.huerst(a)gmail.com> According to the datasheet, this device does also support 44.1 KHz as sample rate. This adds the necessary changes to support this. See Datasheet: Rev. B / page 39 / SR[1:0] Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- sound/soc/codecs/adau1701.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index de53c0d..8b3ba51 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -459,6 +459,7 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream, case 96000: val = ADAU1701_DSPCTRL_SR_96; break; + case 44100: case 48000: val = ADAU1701_DSPCTRL_SR_48; break; @@ -623,8 +624,8 @@ static int adau1701_startup(struct snd_pcm_substream *substream, return sigmadsp_restrict_params(adau1701->sigmadsp, substream); } -#define ADAU1701_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \ - SNDRV_PCM_RATE_192000) +#define ADAU1701_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) #define ADAU1701_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -- 2.4.3

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[alsa-devel] [PATCH RFT] ASoC: wm8904: Make undocumented registers non-readable
by Axel Lin 18 Nov '15

18 Nov '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- I think the intention of wm8904_readable_register is to return false for undocumented registers, but current code returns true for all cases in wm8904_readable_register. Please review if this patch is correct or not. Thanks. sound/soc/codecs/wm8904.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index b783743..f48b9cb 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -312,7 +312,7 @@ static bool wm8904_readable_register(struct device *dev, unsigned int reg) case WM8904_FLL_NCO_TEST_1: return true; default: - return true; + return false; } } -- 2.1.4

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[alsa-devel] [PATCH] ARM: sun7i: Fix HDMI Audio driver's Alsa interface and do 8CH IEC60958 output
by thomas schorpp 13 Nov '15

13 Nov '15

Version 0.2 Users doc: All You need to get a HDMI- Pulseaudio Server on Your sunxi board to output Multichannel PCM is this line in default.pa, ttable pcms in asound.conf get ignored: +load-module module-alsa-sink device="hw:1,0" channels=8 channel_map="front-left,front-right,lfe,front-center,rear-left,rear-right,side-left,side-right" ### Automatically load driver modules depending on the hardware available .ifexists module-udev-detect.so load-module module-udev-detect Probe with speaker-test, You may need other channel mapping than I. If you run PA in system mode please read the security documentation before, see PA log for link. y tom

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[alsa-devel] DP1.2 MST audio support discussion
by Yang, Libin 13 Nov '15

13 Nov '15

Hi Takashi and all, We are planning to enable DP1.2 MST (Multi-Stream Transport) audio. Based on the previous discussion, we will extend the struct hdmi_spec_per_pin to support MST audio device entry. So the struct hdmi_spec_per_pin can be a real pin or a device entry in the pin. The idea is to add a member dev_idx in the struct hdmi_spec_per_pin. Dev_idx, together with pin_nid, can represent a device entry. 1. Dynamic PCM assignment We will use dynamic PCM assignment for MST audio. This means we will create a fixed number of PCMs (the number is the same convertor number). All the created PCMs will not be assigned to any pin (device entry). When there is a monitor connected, an available PCM will be assigned to the pin. And it will be de-assigned when the monitor is disconnected. Userspace can fetch the HW param when monitor connection status is changed. I'm not sure how to notify the userspace, such as notifying pulseaudio the PCM is assigned or de-assigned. Any ideas? 2. Compatibility. We will patch patch_hdmi.c to support the MST audio. Will we use mst audio driver to support the old mode or we use a flag, when HW doesn't support MST audio, we will use the old code? Suppose MST audio driver should be able support both MST audio and non-MST audio. Regards, Libin

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