March 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASoC: rockchip: i2s: separate capture and playback
by John Keeping 29 Apr '16

29 Apr '16

If we only clear the tx/rx state when both are disabled it is not possible to start/stop one multiple times while the other is running. Since the two are independently controlled, treat them as such and remove the false dependency between capture and playback. Signed-off-by: John Keeping <john(a)metanate.com> --- sound/soc/rockchip/rockchip_i2s.c | 72 +++++++++++++++++---------------------- 1 file changed, 32 insertions(+), 40 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 83b1b9c..acc6225 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -82,8 +82,8 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | I2S_XFER_RXS_START, - I2S_XFER_TXS_START | I2S_XFER_RXS_START); + I2S_XFER_TXS_START, + I2S_XFER_TXS_START); i2s->tx_start = true; } else { @@ -92,27 +92,23 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); - if (!i2s->rx_start) { - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | - I2S_XFER_RXS_START, - I2S_XFER_TXS_STOP | - I2S_XFER_RXS_STOP); + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START, + I2S_XFER_TXS_STOP); - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC | I2S_CLR_RXC, - I2S_CLR_TXC | I2S_CLR_RXC); + regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC, + I2S_CLR_TXC); - regmap_read(i2s->regmap, I2S_CLR, &val); + regmap_read(i2s->regmap, I2S_CLR, &val); - /* Should wait for clear operation to finish */ - while (val) { - regmap_read(i2s->regmap, I2S_CLR, &val); - retry--; - if (!retry) { - dev_warn(i2s->dev, "fail to clear\n"); - break; - } + /* Should wait for clear operation to finish */ + while (val & I2S_CLR_TXC) { + regmap_read(i2s->regmap, I2S_CLR, &val); + retry--; + if (!retry) { + dev_warn(i2s->dev, "fail to clear\n"); + break; } } } @@ -128,8 +124,8 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | I2S_XFER_RXS_START, - I2S_XFER_TXS_START | I2S_XFER_RXS_START); + I2S_XFER_RXS_START, + I2S_XFER_RXS_START); i2s->rx_start = true; } else { @@ -138,27 +134,23 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); - if (!i2s->tx_start) { - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | - I2S_XFER_RXS_START, - I2S_XFER_TXS_STOP | - I2S_XFER_RXS_STOP); + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_RXS_START, + I2S_XFER_RXS_STOP); - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC | I2S_CLR_RXC, - I2S_CLR_TXC | I2S_CLR_RXC); + regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_RXC, + I2S_CLR_RXC); - regmap_read(i2s->regmap, I2S_CLR, &val); + regmap_read(i2s->regmap, I2S_CLR, &val); - /* Should wait for clear operation to finish */ - while (val) { - regmap_read(i2s->regmap, I2S_CLR, &val); - retry--; - if (!retry) { - dev_warn(i2s->dev, "fail to clear\n"); - break; - } + /* Should wait for clear operation to finish */ + while (val & I2S_CLR_RXC) { + regmap_read(i2s->regmap, I2S_CLR, &val); + retry--; + if (!retry) { + dev_warn(i2s->dev, "fail to clear\n"); + break; } } } -- 2.6.3.462.gbe2c914

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[alsa-devel] [PATCH] ASoC: Make aux_dev more like a generic component
by mengdong.lin＠linux.intel.com 25 Apr '16

25 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> aux_dev is mainly used by the machine driver to specify analog devices, which are registered as codecs. Making it more like a generic component can help the machine driver to use it to specify any component with topology info by name. Details: - Remove the stub 'rtd_aux' array from the soc card. - Add a list 'aux_comp_list' to store the components of aux_devs. And add a list head 'list_aux' to struct snd_soc_component, for adding such components to the above list. - Add a 'init' ops to a component for machine specific init. soc_bind_aux_dev() will set it to be aux_dev's init. And it will be called when probing the component. - soc_bind_aux_dev() will also search components by name of an aux_dev, since it may not be a codec. - Move probing of aux_devs before checking new DAI links brought by topology. - Move removal of aux_devs later than removal of links. Because topology of aux components may register DAIs and the DAI drivers will go with removal of the aux components, we want soc_remove_link_dais() to remove the DAIs at first. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/include/sound/soc.h b/include/sound/soc.h index 49011c3..c03e07a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -798,6 +798,7 @@ struct snd_soc_component { unsigned int registered_as_component:1; struct list_head list; + struct list_head list_aux; /* for auxiliary component of the card */ struct snd_soc_dai_driver *dai_drv; int num_dai; @@ -841,6 +842,9 @@ struct snd_soc_component { int (*probe)(struct snd_soc_component *); void (*remove)(struct snd_soc_component *); + /* machine specific init */ + int (*init)(struct snd_soc_component *component); + #ifdef CONFIG_DEBUG_FS void (*init_debugfs)(struct snd_soc_component *component); const char *debugfs_prefix; @@ -1144,8 +1148,7 @@ struct snd_soc_card { */ struct snd_soc_aux_dev *aux_dev; int num_aux_devs; - struct snd_soc_pcm_runtime *rtd_aux; - int num_aux_rtd; + struct list_head aux_comp_list; const struct snd_kcontrol_new *controls; int num_controls; @@ -1553,6 +1556,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->widgets); INIT_LIST_HEAD(&card->paths); INIT_LIST_HEAD(&card->dapm_list); + INIT_LIST_HEAD(&card->aux_comp_list); } static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4e03e82..790ee2b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1413,6 +1413,16 @@ static int soc_probe_component(struct snd_soc_card *card, component->name); } + /* machine specific init */ + if (component->init) { + ret = component->init(component); + if (ret < 0) { + dev_err(component->dev, + "Failed to do machine specific init %d\n", ret); + goto err_probe; + } + } + if (component->controls) snd_soc_add_component_controls(component, component->controls, component->num_controls); @@ -1657,65 +1667,81 @@ static int soc_probe_link_dais(struct snd_soc_card *card, static int soc_bind_aux_dev(struct snd_soc_card *card, int num) { - struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; - const char *name = aux_dev->codec_name; - - rtd->component = soc_find_component(aux_dev->codec_of_node, name); - if (!rtd->component) { - if (aux_dev->codec_of_node) - name = of_node_full_name(aux_dev->codec_of_node); - - dev_err(card->dev, "ASoC: %s not registered\n", name); - return -EPROBE_DEFER; + struct snd_soc_component *component; + const char *name; + struct device_node *codec_of_node; + + if (aux_dev->codec_of_node || aux_dev->codec_name) { + /* codecs, usually analog devices */ + name = aux_dev->codec_name; + codec_of_node = aux_dev->codec_of_node; + component = soc_find_component(codec_of_node, name); + if (!component) { + if (codec_of_node) + name = of_node_full_name(codec_of_node); + goto err_defer; + } + } else if (aux_dev->name) { + /* generic components */ + name = aux_dev->name; + component = soc_find_component(NULL, name); + if (!component) + goto err_defer; + } else { + dev_err(card->dev, "ASoC: Invalid auxiliary device\n"); + return -EINVAL; } - /* - * Some places still reference rtd->codec, so we have to keep that - * initialized if the component is a CODEC. Once all those references - * have been removed, this code can be removed as well. - */ - rtd->codec = rtd->component->codec; - + component->init = aux_dev->init; + list_add(&component->list_aux, &card->aux_comp_list); return 0; + +err_defer: + dev_err(card->dev, "ASoC: %s not registered\n", name); + return -EPROBE_DEFER; } -static int soc_probe_aux_dev(struct snd_soc_card *card, int num) +static int soc_probe_aux_devices(struct snd_soc_card *card) { - struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; - struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; + struct snd_soc_component *comp; + int order; int ret; - ret = soc_probe_component(card, rtd->component); - if (ret < 0) - return ret; - - /* do machine specific initialization */ - if (aux_dev->init) { - ret = aux_dev->init(rtd->component); - if (ret < 0) { - dev_err(card->dev, "ASoC: failed to init %s: %d\n", - aux_dev->name, ret); - return ret; + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + list_for_each_entry(comp, &card->aux_comp_list, list_aux) { + if (comp->driver->probe_order == order) { + ret = soc_probe_component(card, comp); + if (ret < 0) { + dev_err(card->dev, + "ASoC: failed to probe aux component %s %d\n", + comp->name, ret); + return ret; + } + } } } - return soc_post_component_init(rtd, aux_dev->name); + return 0; } -static void soc_remove_aux_dev(struct snd_soc_card *card, int num) +static void soc_remove_aux_devices(struct snd_soc_card *card) { - struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; - struct snd_soc_component *component = rtd->component; + struct snd_soc_component *comp, *_comp; + int order; - /* unregister the rtd device */ - if (rtd->dev_registered) { - device_unregister(rtd->dev); - rtd->dev_registered = 0; + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + list_for_each_entry_safe(comp, _comp, + &card->aux_comp_list, list_aux) { + if (comp->driver->remove_order == order) { + soc_remove_component(comp); + /* remove it from the card's aux_comp_list */ + list_del(&comp->list_aux); + } + } } - - if (component) - soc_remove_component(component); } static int snd_soc_init_codec_cache(struct snd_soc_codec *codec) @@ -1894,6 +1920,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } } + /* probe auxiliary components */ + ret = soc_probe_aux_devices(card); + if (ret < 0) + goto probe_dai_err; + /* Find new DAI links added during probing components and bind them. * Components with topology may bring new DAIs and DAI links. */ @@ -1923,16 +1954,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } } - for (i = 0; i < card->num_aux_devs; i++) { - ret = soc_probe_aux_dev(card, i); - if (ret < 0) { - dev_err(card->dev, - "ASoC: failed to add auxiliary devices %d\n", - ret); - goto probe_aux_dev_err; - } - } - snd_soc_dapm_link_dai_widgets(card); snd_soc_dapm_connect_dai_link_widgets(card); @@ -1992,8 +2013,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) return 0; probe_aux_dev_err: - for (i = 0; i < card->num_aux_devs; i++) - soc_remove_aux_dev(card, i); + soc_remove_aux_devices(card); probe_dai_err: soc_remove_dai_links(card); @@ -2039,20 +2059,18 @@ static int soc_probe(struct platform_device *pdev) static int soc_cleanup_card_resources(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; - int i; /* make sure any delayed work runs */ list_for_each_entry(rtd, &card->rtd_list, list) flush_delayed_work(&rtd->delayed_work); - /* remove auxiliary devices */ - for (i = 0; i < card->num_aux_devs; i++) - soc_remove_aux_dev(card, i); - /* remove and free each DAI */ soc_remove_dai_links(card); soc_remove_pcm_runtimes(card); + /* remove auxiliary devices */ + soc_remove_aux_devices(card); + soc_cleanup_card_debugfs(card); /* remove the card */ @@ -2608,16 +2626,6 @@ int snd_soc_register_card(struct snd_soc_card *card) INIT_LIST_HEAD(&card->rtd_list); card->num_rtd = 0; - card->rtd_aux = devm_kzalloc(card->dev, - sizeof(struct snd_soc_pcm_runtime) * - card->num_aux_devs, - GFP_KERNEL); - if (card->rtd_aux == NULL) - return -ENOMEM; - - for (i = 0; i < card->num_aux_devs; i++) - card->rtd_aux[i].card = card; - INIT_LIST_HEAD(&card->dapm_dirty); INIT_LIST_HEAD(&card->dobj_list); card->instantiated = 0; -- 2.5.0

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[alsa-devel] [PATCH 0/7] ALSA: firewire: apply deleyed registration to all drivers
by Takashi Sakamoto 25 Apr '16

25 Apr '16

Hi, Commit b59fb1900b4fee('ALSA: dice: postpone card registration') applied delayed registration[1] of sound card instance to ALSA dice driver. This idea is also nice for drivers in ALSA firewire stack because in IEEE 1394 bus adding units is corresponding to bus reset and the actual units tend not to work well during bus-reset state. This patchset adds a new function to firewire-lib module to schedule a delayed work for sound card registration. Currently, the work is commonly scheduled 2 seconds after time of the last bus-reset recorded by Linux firewire core. Then, take all drivers to use it as Dice driver does. Related to this patchset, I'd like to cancel a discussion about my proposed patchset to remove unpractical module parameters from ALSA bebob/fireworks[2]. I'm sorry for reviewers but my work priority to this patchset is prior to the former one. [1]: [alsa-devel] [PATCH 0/4 v5] ALSA: dice: improve card registration processing http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/102562.h… [2]: [alsa-devel] [PATCH 0/2] ALSA: bebob/fireworks: remove module parameters to reduce maintenance work http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/106219.html Regards Takashi Sakamoto (8): ALSA: firewire-lib: suppress kernel warnings when releasing uninitialized stream data ALSA: dice: simplify unit probe processing ALSA: firewire-lib: add new function to schedule a work for sound card registration ALSA: bebob: delayed registration of sound card ALSA: fireworks: delayed registration of sound card ALSA: oxfw: delayed registration of sound card ALSA: firewire-digi00x: delayed registration of sound card ALSA: firewire-tascam: deleyed registration of sound card sound/firewire/amdtp-stream.c | 4 + sound/firewire/bebob/bebob.c | 215 ++++++++++++++++----------- sound/firewire/bebob/bebob.h | 5 +- sound/firewire/dice/dice.c | 34 +---- sound/firewire/digi00x/digi00x-transaction.c | 7 +- sound/firewire/digi00x/digi00x.c | 107 +++++++++---- sound/firewire/digi00x/digi00x.h | 3 + sound/firewire/fireworks/fireworks.c | 147 ++++++++++++------ sound/firewire/fireworks/fireworks.h | 3 + sound/firewire/lib.c | 32 ++++ sound/firewire/lib.h | 3 + sound/firewire/oxfw/oxfw.c | 151 ++++++++++++------- sound/firewire/oxfw/oxfw.h | 3 + sound/firewire/tascam/tascam.c | 118 ++++++++++----- sound/firewire/tascam/tascam.h | 2 + 15 files changed, 551 insertions(+), 283 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH 0/4] ASoC: simple-card/davinci-mcasp: System clock configuration support
by Peter Ujfalusi 22 Apr '16

22 Apr '16

Hi, Currently davinci-mcasp is hardwired for two configuration regarding to reference clock: McASP is master: AHCLKX is selected as source of the reference clock used to generate the FS and BCLK McASP is slave: AHCLKX is selected as output and the default AUXCLK is routed there. This works in most of the boards, but not all configuration can be supported with this limitation. McASP uses it's internal AUXCLK as a reference clock. AUXCLK can be sourced from the external AHCLKX pin or from internal functional clock. If the board uses McASP as master and AHCLKX is not receiving clock, we need to be able to select the internal clock source to be able to generate the FS and BCLK. Since we try to move all of our boards to simple-card instead of the davinci-evm custom machine driver, I have included Jyri's two patch to add the needed support for it to be able to select clocks and directions. I remember the discussion regarding to clock IDs, and to try to move to common clock framework. I have taken a look at that path and I'm not convinced that by trying to add ASoC level common clock handling will simplify or makes things cleaner/simpler. Regards, Peter --- Jyri Sarha (2): ASoC: simple-card: Add system-clock-direction DT parameter to dai nodes ASoC: simple-card: Support for selecting system clocks by ID Peter Ujfalusi (2): ASoC: davinci-mcasp: Use defines for clkdiv IDs via DT binding header ASoC: davinci-mcasp: Improve the sysclk selection .../devicetree/bindings/sound/simple-card.txt | 3 ++ include/dt-bindings/sound/ti-mcasp.h | 13 +++++++ include/sound/simple_card.h | 2 ++ sound/soc/davinci/davinci-mcasp.c | 42 ++++++++++++++++------ sound/soc/generic/simple-card.c | 17 ++++++++- 5 files changed, 66 insertions(+), 11 deletions(-) create mode 100644 include/dt-bindings/sound/ti-mcasp.h -- 2.7.1

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[alsa-devel] [PATCH v6 00/10] ASoC: Add mediatek HDMI codec support
by Philipp Zabel 21 Apr '16

21 Apr '16

these patches add an interface for Jyri's generic hdmi-codec driver [1] to the mediatek hdmi bridge driver. This version fixes the hdmi-codec notification patch and splits the jack support out of the ASoC machine driver patches. Changes since v5: - select SND_SOC_HDMI_CODEC and HDMI_NOTIFIERS where appropriate - split out jack support from machine driver patches - fix include of hdmi-notifier.h in hdmi-codec notification support - Drop applied patch: commit 05f5afd3f0d9 ("ASoC: mediatek: address dai link array entries by enum") The mediatek drm patches apply on top of https://patchwork.kernel.org/patch/8534251/ ("drm/mediatek: Add HDMI support"), and need this patch: [2] https://patchwork.kernel.org/patch/8091531/ ("video: hdmi: add helper function for N and CTS") The asoc patches depend on these two patches: https://patchwork.kernel.org/patch/7215121/ ("ALSA: pcm: add IEC958 channel status helper for hw_params") [1] https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") regards Philipp Koro Chen (1): ASoC: mediatek: Add HDMI dai-links in the mt8173-rt5650-rt5676 machine driver PC Liao (1): ASoC: mediatek: Add HDMI dai-links to the mt8173-rt5650 machine driver Philipp Zabel (8): drm/mediatek: hdmi: Add audio interface to the hdmi-codec driver video: rmk's HDMI notification prototype drm/mediatek: hdmi: issue notifications ASoC: hdmi-codec: Use HDMI notifications to add jack support ASoC: hdmi-codec: Add ELD control drm/mediatek: hdmi: use helper function for N and CTS calculation ASoC: mediatek: Add jack detection support to mt8173-rt5650-rt5676 machine driver ASoC: mediatek: Add jack detection support to the mt8173-rt5650 machine driver .../bindings/sound/mt8173-rt5650-rt5676.txt | 5 +- .../devicetree/bindings/sound/mt8173-rt5650.txt | 5 +- drivers/gpu/drm/mediatek/Kconfig | 1 + drivers/gpu/drm/mediatek/mtk_cec.c | 11 ++ drivers/gpu/drm/mediatek/mtk_drm_hdmi_drv.c | 148 +++++++++++++++++ drivers/gpu/drm/mediatek/mtk_hdmi.c | 89 +++++----- drivers/gpu/drm/mediatek/mtk_hdmi.h | 10 +- drivers/gpu/drm/mediatek/mtk_hdmi_hw.c | 179 ++++----------------- drivers/gpu/drm/mediatek/mtk_hdmi_hw.h | 4 +- drivers/video/Kconfig | 3 + drivers/video/Makefile | 1 + drivers/video/hdmi-notifier.c | 61 +++++++ include/linux/hdmi-notifier.h | 44 +++++ include/sound/hdmi-codec.h | 6 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdmi-codec.c | 123 +++++++++++++- sound/soc/mediatek/Kconfig | 2 + sound/soc/mediatek/mt8173-rt5650-rt5676.c | 48 ++++++ sound/soc/mediatek/mt8173-rt5650.c | 47 ++++++ 19 files changed, 576 insertions(+), 212 deletions(-) create mode 100644 drivers/video/hdmi-notifier.c create mode 100644 include/linux/hdmi-notifier.h -- 2.7.0

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[alsa-devel] [PATCH v8 0/2] ASoc: sun4i-codec: Add FM, Line and Mic inputs
by Danny Milosavljevic 21 Apr '16

21 Apr '16

Hi, this is the eighth version of the patch set that adds inputs to sun4i-codec. Changes compared to v7 are: - fix the routes for line and mic capturing. The patches are on top of the branch "sunxi-next" in <git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git>. Regards, Danny --- Danny Milosavljevic (2): sun4i-codec: Add FM, Line and Mic inputs sun4i-codec: make it possible to use different codec_widgets for A10 and A20. b/sound/soc/sunxi/sun4i-codec.c | 230 +++++++++++++++++++++++++++++++++++++--- 1 file changed, 215 insertions(+), 15 deletions(-)

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[alsa-devel] [PATCH v9 0/7] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 19 Apr '16

19 Apr '16

Couple of minor changes since v8 to ALSA/ASoC side patches. The series is based on top of drm-next. However, it does not look like there is enough attention to the DRM side so that the tda998x patches would get in any time soon. So I am happy to rebase the first ALSA-side patches on top of Mark Brown's for-next tree, for instance, if they would have a change to go in on their own. The first two IEC958 helper patches should apply as such. There is currently two other patch series[2][3] that depend on the first three (ALSA-) patches of this series. The third ("ASoC: hdmi-codec: Add audio abort() callback for video side to use") is currently not used and can be dropped if so decided. The rest depends on those the first patches and adds hdmi-audio support for Beaglebone-black. Best regards, Jyri [1] "[PATCH v6 0/6] Implement generic ASoC HDMI codec and use it in tda998x" http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105524.html [2] "[PATCH v6 00/10] ASoC: Add mediatek HDMI codec support" by Philipp Zabel http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105509.html [3] "[RFC v2 0/6] sti: add audio interface to the hdmi driver" http://mailman.alsa-project.org/pipermail/alsa-devel/2016-January/103374.ht… and "[PATCH v4 0/6] add IEC958 channel status control helpers" http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105502.html by Arnaud Pouliquen Changes since v8 * "ALSA: pcm: add IEC958 channel status helper for hw_params" - Split out the 32 bit support - add: "ALSA: pcm: Allow 32 bit sample format in IEC958 channel status helper" * "ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders" - Set hw_params->msbits to 24 in hw_params() callback if sample width is greater than 24 bits Changes since v7 * "drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding" - Remove the unused "int ret" variable from tda998x_configure_audio() Changes since v6 * "ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders" - Add "Acked-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com>" and "Tested-by: Philipp Zabel <p.zabel(a)pengutronix.de>", no other changes * "ALSA: pcm: add IEC958 channel status helper for hw_param" - Added kernel-doc for snd_pcm_create_iec958_consumer_hw_params() * "drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding" - Use PTR_ERR_OR_ZERO() in tda998x_audio_codec_init() - Fix possible uninitialized use of 'size' in tda998x_get_audio_ports() - Store audio configuration in tda998x_audio_hw_params() instead of tda998x_configure_audio(). Changes since v5 - Rebased on top of the latest drm-next branch - Allow 32 bit samplewidth in snd_pcm_create_iec958_consumer() and snd_pcm_create_iec958_consumer_hw_params() - Propose new simpler DT binding for tda998x audio - Squash tda998x audio DT binding together with hdmi-codec integration Changes since RFC v4, - Rebased on top of the latest drm-next branch - Split the hdmi-codec abort functionality into a separate patch for better visibility of what it is all about - This does not affect the tda998x patches as the abort functionality is not used - Drop S18_3* formats from I2S_FORMATS and add a comment about formats not supported by HDMI Changes since RFC v3, ASoC side: - Add "ALSA: pcm: add IEC958 channel status helper for hw_params" - Add "tda998x: Improve tda998x_configure_audio() audio related pdata" - use snd_pcm_create_iec958_consumer_hw_params() to construct the stream header - Remove set_clk() callback from hdmi-codec. It is not needed for now. - Refer to stream header in AIF as specified in HDMI standard - Set current_stream to NULL only after video side audio_shutdown() has been called. Avoid potential race if video side attempts to abort audio at the same time. - No need to have video side device pointer in the hdmi codec's pdata as it is found from dev->parent. - Fix hdmi-codec enum: DAI_ID_I2C > DAI_ID_I2S - Improve audio_startup API comment - Make improved checkpatch happy - BUG_ON > WARN_ON - put */ ending the block comment to a separate line DRM side: - Fix tda998x get_eld() locking - Change tda998x audio parameters in pdata to more generic, that can be readily used in tda998x_audio_config() - Rename and restructure audio port related private data members to be more descriptive - Require audio configuration trough ASoC hdmi-codec if HDMI audio is configured trough DT binding. DTS: - Increase McASP fifo usage form 1 to 32 Jyri Sarha (7): ALSA: pcm: add IEC958 channel status helper for hw_params ALSA: pcm: Allow 32 bit sample format in IEC958 channel status helper ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders ASoC: hdmi-codec: Add audio abort() callback for video side to use drm/i2c: tda998x: Improve tda998x_configure_audio() audio related pdata drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding ARM: dts: am335x-boneblack: Add HDMI audio support .../devicetree/bindings/display/bridge/tda998x.txt | 18 + arch/arm/boot/dts/am335x-boneblack.dts | 71 +++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 275 ++++++++++++-- include/drm/i2c/tda998x.h | 24 +- include/dt-bindings/display/tda998x.h | 7 + include/sound/hdmi-codec.h | 104 ++++++ include/sound/pcm_iec958.h | 2 + sound/core/pcm_iec958.c | 65 +++- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdmi-codec.c | 414 +++++++++++++++++++++ 12 files changed, 917 insertions(+), 72 deletions(-) create mode 100644 include/dt-bindings/display/tda998x.h create mode 100644 include/sound/hdmi-codec.h create mode 100644 sound/soc/codecs/hdmi-codec.c -- 1.9.1

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[alsa-devel] ASoC: TLV320AIC3x: Adding additional functionality for 3106 with [Patch] for discuss
by Timur Karaldin 18 Apr '16

18 Apr '16

Hi, colleagues! I'm newbie with ALSA devel. I need to get extra functionality from codec through ALSA mixer. Task: 1. making possibility to set-up gain for each input (XXX_2_LADCCTRL and XXX_2_RADCCTRL registers) 2. making possibility to set-up debounce for button and headset detection ( HEADSET_DETECT_CTRL_A register) 3. making possibility to enable high-power AC-coupled output (HEADSET_DETECT_CTRL_A register) 4. making possibility to enable stereo pseudo differential output (HEADSET_DETECT_CTRL_B register) 5. enable headset detection (HEADSET_DETECT_CTRL_A) 6. getting headset detect status (HEADSET_DETECT_CTRL_A) Well, my patch works in generally, but I don't like it very much, because: 1. All controls for input amplifiers' gain are shown in Alsamixer-Playback section. But not only mine, capture controls from base codec's version are shown in playback section too, for example (Right or Left) PGA Mixer Line1(L or R)/2(L or R)/MIC3(L or R), which is definitely capture settings. Should I worried about it? because, as I understand, it could not be setted correctly during changing playback/capture mode for power-saving reason. 2. "Right PGA Mixer Line1R" control (and similar) using the same bits in LINE1R_2_RADCCTRL register as my "Line1R RADC Gain" control (and similar) and put there b"1111" when user switch disconnect it from ADC. So my control is not shown when user disconnect it from ADC. 3. When I start/stop arecord some of my options were reset (debounces, headset enable, high-power AC-coupled, pseudo differential output). When I set up it in shell and then start arecord, it's reset again. Is it the reason because of displaying controls in playback mixer section? I will be very pleased for any advise, thank you! here is my patch for tlv320aic3x.c --- sound/soc/codecs/tlv320aic3x.c | 55 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 52 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index d7349bc..27d031e 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -122,11 +122,24 @@ static const struct reg_default aic3x_reg[] = { { 108, 0x00 }, { 109, 0x00 }, }; +static bool aic3106_volatile(struct device *dev, unsigned int reg) +{ + switch(reg) + { + case AIC3X_HEADSET_DETECT_CTRL_A: + case AIC3X_HEADSET_DETECT_CTRL_B: + return true; + default: + return false; + } +} + static const struct regmap_config aic3x_regmap = { .reg_bits = 8, .val_bits = 8, .max_register = DAC_ICC_ADJ, + .volatile_reg = aic3106_volatile, .reg_defaults = aic3x_reg, .num_reg_defaults = ARRAY_SIZE(aic3x_reg), .cache_type = REGCACHE_RBTREE, @@ -196,7 +209,7 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -270,6 +283,26 @@ static const struct soc_enum aic3x_agc_decay_enum[] = { SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), }; +static const char *aic3x_adc_gain[] = { "0dB", "-1.5dB", "-3dB", "-4.5dB", "-6dB", "-7.5dB", "-9dB", "-10.5dB", "-12dB" }; +static const struct soc_enum aic3x_adc_gain_enum[] = { + SOC_ENUM_SINGLE(MIC3LR_2_LADC_CTRL, 4, 9, aic3x_adc_gain), + SOC_ENUM_SINGLE(MIC3LR_2_RADC_CTRL, 0, 9, aic3x_adc_gain), + SOC_ENUM_SINGLE(LINE1L_2_LADC_CTRL, 3, 9, aic3x_adc_gain), + SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 3, 9, aic3x_adc_gain), + SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 3, 9, aic3x_adc_gain), + SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 3, 9, aic3x_adc_gain), +}; +static const char *aic3x_headset_debounce[] = { "16ms", "32ms", "64ms", "128ms", "256ms", "512ms" }; +static const struct soc_enum aic3x_headset_debounce_enum[] = { + SOC_ENUM_SINGLE(AIC3X_HEADSET_DETECT_CTRL_A, 2, 6, aic3x_headset_debounce), +}; +static const char *aic3x_button_debounce[] = { "0ms", "8ms", "16ms", "32ms" }; +static const struct soc_enum aic3x_button_debounce_enum[] = { + SOC_ENUM_SINGLE(AIC3X_HEADSET_DETECT_CTRL_A, 0, 4, aic3x_button_debounce), +}; + + + /* * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps */ @@ -399,6 +432,19 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), + /* Additional controls */ + SOC_ENUM("Mic3L LADC Gain", aic3x_adc_gain_enum[0]), + SOC_ENUM("Mic3R RADC Gain", aic3x_adc_gain_enum[1]), + SOC_ENUM("Line1L LADC Gain", aic3x_adc_gain_enum[2]), + SOC_ENUM("Line1R RADC Gain", aic3x_adc_gain_enum[3]), + SOC_ENUM("Line2L LADC Gain", aic3x_adc_gain_enum[4]), + SOC_ENUM("Line2R RADC Gain", aic3x_adc_gain_enum[5]), + SOC_ENUM("Headset Debounce", aic3x_headset_debounce_enum), + SOC_ENUM("Button Debounce", aic3x_button_debounce_enum), + SOC_SINGLE("Headset Detected", AIC3X_HEADSET_DETECT_CTRL_A, AIC3X_HEADSET_DETECT_SHIFT, AIC3X_HEADSET_DETECT_MASK, 0), + SOC_SINGLE("Headset Detect Enable", AIC3X_HEADSET_DETECT_CTRL_A, 7, 1, 0), + SOC_SINGLE("High power output Ac-coupled", AIC3X_HEADSET_DETECT_CTRL_B, 7, 1, 0), + SOC_SINGLE("Stereo pseudodifferential output", AIC3X_HEADSET_DETECT_CTRL_B, 3, 1, 0), }; static const struct snd_kcontrol_new aic3x_mono_controls[] = { @@ -614,7 +660,7 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 32", AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0), - /* Mic Bias */ + /* Mic Bias */ SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0, mic_bias_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -692,7 +738,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left Line2L Mux", "single-ended", "LINE2L"}, {"Left Line2L Mux", "differential", "LINE2L"}, - {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"}, + {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"}, {"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"}, {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"}, {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, @@ -814,6 +860,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right HPCOM Mux", "external feedback", "Right HPCOM Mixer"}, {"Right HP Com", NULL, "Right HPCOM Mux"}, {"HPRCOM", NULL, "Right HP Com"}, + }; static const struct snd_soc_dapm_route intercon_mono[] = { @@ -918,7 +965,9 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, data = (LDAC2LCH | RDAC2RCH); data |= (fsref == 44100) ? FSREF_44100 : FSREF_48000; if (params_rate(params) >= 64000) + { data |= DUAL_RATE_MODE; + } snd_soc_write(codec, AIC3X_CODEC_DATAPATH_REG, data); /* codec sample rate select */ -- and tlv320aic3x.h --- sound/soc/codecs/tlv320aic3x.h | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index e521ac3..8ff42d2 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -271,6 +271,12 @@ enum { AIC3X_BUTTON_DEBOUNCE_32MS = 3 }; +typedef struct { + struct platform_device* pdev; + struct proc_dir_entry *proc_value; + struct snd_soc_codec *codec; +} aic3106_detect_t; + #define AIC3X_HEADSET_DETECT_ENABLED 0x80 #define AIC3X_HEADSET_DETECT_SHIFT 5 #define AIC3X_HEADSET_DETECT_MASK 3 -- -- С уважением, Руководитель проектов Морские комплексы и системы плюс Каралдин Т.М. ---------------------------------------------------------------------------------- Best regards, Projects director Marine complexes and systems plus Timur Karaldin

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[alsa-devel] [PATCH 0/4] ASoC: sti: tdm mode
by Moise Gergaud 18 Apr '16

18 Apr '16

This patchset enables tdm mode for ASoC sti. Moise Gergaud (4): ASoC: sti: correct typo errors ASoC: sti: tdm: uni tdm mode ASoC: sti: select player for I2S/TDM TX bus Documentation: sti-asoc-card: update tdm mode .../devicetree/bindings/sound/st,sti-asoc-card.txt | 48 ++++- sound/soc/sti/sti_uniperif.c | 144 ++++++++++++- sound/soc/sti/uniperif.h | 220 ++++++++++++++++++-- sound/soc/sti/uniperif_player.c | 182 +++++++++++----- sound/soc/sti/uniperif_reader.c | 229 +++++++++++++++------ 5 files changed, 687 insertions(+), 136 deletions(-) -- 1.9.1

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Re: [alsa-devel] ALSA: lx6464es - driver for the digigram lx6464es interface
by Dan Carpenter 15 Apr '16

15 Apr '16

Hello Tim Blechmann, The patch 02bec4904508: "ALSA: lx6464es - driver for the digigram lx6464es interface" from Mar 24, 2009, leads to the following static checker warning: sound/pci/lx6464es/lx_core.c:647 lx_pipe_wait_for_state() error: uninitialized symbol'current_state'. sound/pci/lx6464es/lx_core.c 612 int lx_pipe_state(struct lx6464es *chip, u32 pipe, int is_capture, u16 *rstate) 613 { 614 int err; 615 u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); 616 617 mutex_lock(&chip->msg_lock); 618 lx_message_init(&chip->rmh, CMD_0A_GET_PIPE_SPL_COUNT); 619 620 chip->rmh.cmd[0] |= pipe_cmd; 621 622 err = lx_message_send_atomic(chip, &chip->rmh); lx_message_send_atomic() returns either negative error codes or a bit mask. 623 624 if (err != 0) ^^^^^^^^ This code assumes that all non-zero values are errors which seems not true since it could be a bit mask. 625 dev_err(chip->card->dev, "could not query pipe's state\n"); 626 else 627 *rstate = (chip->rmh.stat[0] >> PSTATE_OFFSET) & 0x0F; 628 629 mutex_unlock(&chip->msg_lock); 630 return err; 631 } 632 633 static int lx_pipe_wait_for_state(struct lx6464es *chip, u32 pipe, 634 int is_capture, u16 state) 635 { 636 int i; 637 638 /* max 2*PCMOnlyGranularity = 2*1024 at 44100 = < 50 ms: 639 * timeout 50 ms */ 640 for (i = 0; i != 50; ++i) { 641 u16 current_state; 642 int err = lx_pipe_state(chip, pipe, is_capture, &current_state); 643 644 if (err < 0) ^^^^^^^ This code assume that only negatives are errors. 645 return err; 646 647 if (current_state == state) ^^^^^^^^^^^^^^^^^^^^^^ This is potentially slightly buggy because we returned a bit mask so we didn't initialize it. It not terribly harmful but the warning message is annoying. 648 return 0; 649 650 mdelay(1); 651 } 652 653 return -ETIMEDOUT; 654 } regards, dan carpenter

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