June 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH RESEND] ASoC: pxa: remove the dmaengine compat need
by Daniel Mack 29 Jun '18

29 Jun '18

From: Robert Jarzmik <robert.jarzmik(a)free.fr> As the pxa architecture switched towards the dmaengine slave map, the old compatibility mechanism to acquire the dma requestor line number and priority are not needed anymore. This patch simplifies the dma resource acquisition, using the more generic function dma_request_slave_channel(). Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> Signed-off-by: Daniel Mack <daniel(a)zonque.org> Acked-by: Mark Brown <broonie(a)kernel.org> --- sound/arm/pxa2xx-ac97.c | 14 ++------------ sound/arm/pxa2xx-pcm-lib.c | 6 +++--- sound/soc/pxa/pxa2xx-ac97.c | 32 +++++--------------------------- sound/soc/pxa/pxa2xx-i2s.c | 6 ++---- 4 files changed, 12 insertions(+), 46 deletions(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 4bc244c40f80..236a63cdaf9f 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -63,28 +63,18 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_legacy_reset, }; -static struct pxad_param pxa2xx_ac97_pcm_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 12, -}; - static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_out", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_out_req, -}; - -static struct pxad_param pxa2xx_ac97_pcm_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 11, }; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_in", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_in_req, }; static struct snd_pcm *pxa2xx_ac97_pcm; diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index e8da3b8ee721..dcbe7ecc1835 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -125,9 +125,9 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) if (ret < 0) return ret; - return snd_dmaengine_pcm_open_request_chan(substream, - pxad_filter_fn, - dma_params->filter_data); + return snd_dmaengine_pcm_open( + substream, dma_request_slave_channel(rtd->cpu_dai->dev, + dma_params->chan_name)); } EXPORT_SYMBOL(__pxa2xx_pcm_open); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 803818aabee9..1b41c0f2a8fb 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -68,61 +68,39 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_cold_reset, }; -static struct pxad_param pxa2xx_ac97_pcm_stereo_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 11, -}; - static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_in", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_stereo_in_req, -}; - -static struct pxad_param pxa2xx_ac97_pcm_stereo_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 12, }; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_out", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_stereo_out_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mono_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 10, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mono_out", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mono_out_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mono_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 9, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mono_in", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mono_in_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mic_mono_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 8, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { .addr = __PREG(MCDR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mic_mono", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req, }; static int pxa2xx_ac97_hifi_startup(struct snd_pcm_substream *substream, diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 3fb60baf6eab..e7184de0de04 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -82,20 +82,18 @@ static struct pxa_i2s_port pxa_i2s; static struct clk *clk_i2s; static int clk_ena = 0; -static unsigned long pxa2xx_i2s_pcm_stereo_out_req = 3; static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = { .addr = __PREG(SADR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "tx", .maxburst = 32, - .filter_data = &pxa2xx_i2s_pcm_stereo_out_req, }; -static unsigned long pxa2xx_i2s_pcm_stereo_in_req = 2; static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = { .addr = __PREG(SADR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "rx", .maxburst = 32, - .filter_data = &pxa2xx_i2s_pcm_stereo_in_req, }; static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: snd-aoa: add of_node_put() in error path
by Nicholas Mc Guire 29 Jun '18

29 Jun '18

Both calls to of_find_node_by_name() and of_get_next_child() return a node pointer with refcount incremented thus it must be explicidly decremented here after the last usage. As we are assured to have a refcounted np either from the initial of_find_node_by_name(NULL, name); or from the of_get_next_child(gpio, np) in the while loop if we reached the error code path below, an x of_node_put(np) is needed. Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> Fixes: commit f3d9478b2ce4 ("[ALSA] snd-aoa: add snd-aoa") --- Problem located by an experimental coccinelle script Patch was compiletested with: ppc64_defconfig (implies CONFIG_SND_AOA=m) Patch is against 4.18-rc2 (localversion-next is next-20180629) sound/aoa/core/gpio-feature.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/aoa/core/gpio-feature.c b/sound/aoa/core/gpio-feature.c index 7196008..6555742 100644 --- a/sound/aoa/core/gpio-feature.c +++ b/sound/aoa/core/gpio-feature.c @@ -88,8 +88,10 @@ static struct device_node *get_gpio(char *name, } reg = of_get_property(np, "reg", NULL); - if (!reg) + if (!reg) { + of_node_put(np); return NULL; + } *gpioptr = *reg; -- 2.1.4

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[alsa-devel] snd_pcm_hw_params_set_buffer_time_near fails with "Invalid argument" when var_periodsize is enabled
by Wischer, Timo (ADITG/ESB) 29 Jun '18

29 Jun '18

Hi all, at the moment I am investigating in the issue described in [1]. This issue is already avoided by a workaround [2]. I am able to reproduce the issue with a more simple aplay use case using the following configuration: pcm_slave.adr3_tdm_8ch { pcm { type hw card "Loopback" device 0 } rate 48000 period_size 128 buffer_size 1024 channels 2 } pcm.dshare_Playback_3 { type dmix ipc_key 600 ipc_perm 0660 ipc_gid audio var_periodsize true slave adr3_tdm_8ch } pcm.AdevAcousticoutSpeech { type rate slave.pcm dshare_Playback_3 slave.rate 48000 } $ modprobe snd_aloop $ aplay -v --period-size=352 -c2 -fS16_LE -r22500 -D AdevAcousticoutSpeech /dev/urandom aplay: ../../alsa-utils-1.1.5/aplay/aplay.c:1390: set_params: Assertion `err >= 0' failed. Aborted by signal Aborted... It looks like the refinement fails. Now, I need your help because the intervals seems not to be handled in the same way in each function and I want to understand what is the correct interpretation of an interval. For example snd_interval_checkempty() and snd_interval_single() conflicting with each other. snd_interval_checkempty() interprets (x x] and [x x) as an empty set but snd_interval_single() does not. (x x) seems not to be an empty set as implemented by snd_interval_single(). Therefore could someone please explain me what is the correct interpretation of the snd_interval_t in combination with openmin and openmax. The following seems to be consistent (x x+1) and [x x+1) → x (see snd_interval_single() and snd_interval_value()) but it is different to what I have learnt at university. So it would be very helpful if someone could direct me to the right definition. So I hope I could easier solve this issue. [1] https://bugzilla.opensuse.org/show_bug.cgi?id=1033179 [2] http://git.alsa-project.org/?p=alsa-lib.git;a=commit;h=e736715a8ed4c8690059… Best regards Timo Wischer Advanced Driver Information Technology GmbH Engineering Software Base (ADITG/ESB) Robert-Bosch-Str. 200 31139 Hildesheim Germany Tel. +49 5121 49 6938 Fax +49 5121 49 6999 twischer(a)de.adit-jv.com ADIT is a joint venture company of Robert Bosch GmbH/Robert Bosch Car Multimedia GmbH and DENSO Corporation Sitz: Hildesheim, Registergericht: Amtsgericht Hildesheim HRB 3438 Geschäftsführung: Wilhelm Grabow, Ken Yaguchi

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[alsa-devel] [PATCH v2] ALSA: hda - Handle pm failure during hotplug
by Takashi Iwai 29 Jun '18

29 Jun '18

From: Chris Wilson <chris(a)chris-wilson.co.uk> Obtaining the runtime pm wakeref can fail, especially in a hotplug scenario where i915.ko has been unloaded. If we do not catch the failure, we end up with an unbalanced pm. [ Additional changes: hdmi_present_sense() checks the return value and handle only a negative error case and bails out only if it's really still suspended. Also, snd_hda_power_down() is called at the error path so that the refcount is balanced. Along with it, the spec->pcm_lock is taken outside hdmi_present_sense() in the caller side, so that it won't cause deadlock at reentrace via runtime resume. -- tiwai ] References: 222bde03881c ("ALSA: hda - Fix mutex deadlock at HDMI/DP hotplug") Signed-off-by: Chris Wilson <chris(a)chris-wilson.co.uk> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 18 +++++++++++++----- 1 file changed, 13 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 98e1c411c56a..9e3916ed2588 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -764,8 +764,10 @@ static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid, if (pin_idx < 0) return; + mutex_lock(&spec->pcm_lock); if (hdmi_present_sense(get_pin(spec, pin_idx), 1)) snd_hda_jack_report_sync(codec); + mutex_unlock(&spec->pcm_lock); } static void jack_callback(struct hda_codec *codec, @@ -1628,21 +1630,23 @@ static void sync_eld_via_acomp(struct hda_codec *codec, static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) { struct hda_codec *codec = per_pin->codec; - struct hdmi_spec *spec = codec->spec; int ret; /* no temporary power up/down needed for component notifier */ - if (!codec_has_acomp(codec)) - snd_hda_power_up_pm(codec); + if (!codec_has_acomp(codec)) { + ret = snd_hda_power_up_pm(codec); + if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) { + snd_hda_power_down_pm(codec); + return false; + } + } - mutex_lock(&spec->pcm_lock); if (codec_has_acomp(codec)) { sync_eld_via_acomp(codec, per_pin); ret = false; /* don't call snd_hda_jack_report_sync() */ } else { ret = hdmi_present_sense_via_verbs(per_pin, repoll); } - mutex_unlock(&spec->pcm_lock); if (!codec_has_acomp(codec)) snd_hda_power_down_pm(codec); @@ -1654,12 +1658,16 @@ static void hdmi_repoll_eld(struct work_struct *work) { struct hdmi_spec_per_pin *per_pin = container_of(to_delayed_work(work), struct hdmi_spec_per_pin, work); + struct hda_codec *codec = per_pin->codec; + struct hdmi_spec *spec = codec->spec; if (per_pin->repoll_count++ > 6) per_pin->repoll_count = 0; + mutex_lock(&spec->pcm_lock); if (hdmi_present_sense(per_pin, per_pin->repoll_count)) snd_hda_jack_report_sync(per_pin->codec); + mutex_unlock(&spec->pcm_lock); } static void intel_haswell_fixup_connect_list(struct hda_codec *codec, -- 2.17.1

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[alsa-devel] Applied "ASoC: mediatek: sub dai use list_head" to the asoc tree
by Mark Brown 29 Jun '18

29 Jun '18

The patch ASoC: mediatek: sub dai use list_head has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f11c5db770ab675d270cb4d5a2bb90923066ef49 Mon Sep 17 00:00:00 2001 From: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> Date: Fri, 29 Jun 2018 20:29:44 +0800 Subject: [PATCH] ASoC: mediatek: sub dai use list_head use list_head for sub_dais, since original sub_dais array is sparsely occupied Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../mediatek/common/mtk-afe-platform-driver.c | 64 +++++++------------ sound/soc/mediatek/common/mtk-base-afe.h | 6 +- 2 files changed, 28 insertions(+), 42 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index 51ec4ff6ed95..697aa50aff9a 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -15,20 +15,12 @@ int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) { - struct snd_soc_dai_driver *sub_dai_drivers; + struct mtk_base_afe_dai *dai; size_t num_dai_drivers = 0, dai_idx = 0; - int i; - - if (!afe->sub_dais) { - dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); - return -EINVAL; - } /* calcualte total dai driver size */ - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].dai_drivers && - afe->sub_dais[i].num_dai_drivers != 0) - num_dai_drivers += afe->sub_dais[i].num_dai_drivers; + list_for_each_entry(dai, &afe->sub_dais, list) { + num_dai_drivers += dai->num_dai_drivers; } dev_info(afe->dev, "%s(), num of dai %zd\n", __func__, num_dai_drivers); @@ -42,19 +34,14 @@ int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) if (!afe->dai_drivers) return -ENOMEM; - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].dai_drivers && - afe->sub_dais[i].num_dai_drivers != 0) { - sub_dai_drivers = afe->sub_dais[i].dai_drivers; - /* dai driver */ - memcpy(&afe->dai_drivers[dai_idx], - sub_dai_drivers, - afe->sub_dais[i].num_dai_drivers * - sizeof(struct snd_soc_dai_driver)); - dai_idx += afe->sub_dais[i].num_dai_drivers; - } + list_for_each_entry(dai, &afe->sub_dais, list) { + /* dai driver */ + memcpy(&afe->dai_drivers[dai_idx], + dai->dai_drivers, + dai->num_dai_drivers * + sizeof(struct snd_soc_dai_driver)); + dai_idx += dai->num_dai_drivers; } - return 0; } EXPORT_SYMBOL_GPL(mtk_afe_combine_sub_dai); @@ -62,28 +49,25 @@ EXPORT_SYMBOL_GPL(mtk_afe_combine_sub_dai); int mtk_afe_add_sub_dai_control(struct snd_soc_component *component) { struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); - int i; + struct mtk_base_afe_dai *dai; - if (!afe->sub_dais) { - dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); - return -EINVAL; - } - - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].controls) + list_for_each_entry(dai, &afe->sub_dais, list) { + if (dai->controls) snd_soc_add_component_controls(component, - afe->sub_dais[i].controls, - afe->sub_dais[i].num_controls); + dai->controls, + dai->num_controls); - if (afe->sub_dais[i].dapm_widgets) + if (dai->dapm_widgets) snd_soc_dapm_new_controls(&component->dapm, - afe->sub_dais[i].dapm_widgets, - afe->sub_dais[i].num_dapm_widgets); - - if (afe->sub_dais[i].dapm_routes) + dai->dapm_widgets, + dai->num_dapm_widgets); + } + /* add routes after all widgets are added */ + list_for_each_entry(dai, &afe->sub_dais, list) { + if (dai->dapm_routes) snd_soc_dapm_add_routes(&component->dapm, - afe->sub_dais[i].dapm_routes, - afe->sub_dais[i].num_dapm_routes); + dai->dapm_routes, + dai->num_dapm_routes); } snd_soc_dapm_new_widgets(component->dapm.card); diff --git a/sound/soc/mediatek/common/mtk-base-afe.h b/sound/soc/mediatek/common/mtk-base-afe.h index bcf562f029b6..bd8d5e0c6843 100644 --- a/sound/soc/mediatek/common/mtk-base-afe.h +++ b/sound/soc/mediatek/common/mtk-base-afe.h @@ -46,6 +46,7 @@ struct mtk_base_irq_data { }; struct device; +struct list_head; struct mtk_base_afe_memif; struct mtk_base_afe_irq; struct mtk_base_afe_dai; @@ -72,8 +73,7 @@ struct mtk_base_afe { struct mtk_base_afe_irq *irqs; int irqs_size; - struct mtk_base_afe_dai *sub_dais; - int num_sub_dais; + struct list_head sub_dais; struct snd_soc_dai_driver *dai_drivers; unsigned int num_dai_drivers; @@ -110,6 +110,8 @@ struct mtk_base_afe_dai { unsigned int num_dapm_widgets; const struct snd_soc_dapm_route *dapm_routes; unsigned int num_dapm_routes; + + struct list_head list; }; #endif -- 2.18.0.rc2

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[alsa-devel] Applied "ASoC: mt6797: sub dai use list_head" to the asoc tree
by Mark Brown 29 Jun '18

29 Jun '18

The patch ASoC: mt6797: sub dai use list_head has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c1d9b4196ba6b311bd48a9320cd46aa125e0b034 Mon Sep 17 00:00:00 2001 From: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> Date: Fri, 29 Jun 2018 20:29:45 +0800 Subject: [PATCH] ASoC: mt6797: sub dai use list_head Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/mediatek/mt6797/mt6797-afe-common.h | 1 + sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 65 +++++++++++++------ sound/soc/mediatek/mt6797/mt6797-dai-adda.c | 20 ++++-- .../soc/mediatek/mt6797/mt6797-dai-hostless.c | 16 +++-- sound/soc/mediatek/mt6797/mt6797-dai-pcm.c | 19 ++++-- 5 files changed, 81 insertions(+), 40 deletions(-) diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-common.h b/sound/soc/mediatek/mt6797/mt6797-afe-common.h index 22eb7b455cf1..4eac9977b2b0 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-common.h +++ b/sound/soc/mediatek/mt6797/mt6797-afe-common.h @@ -10,6 +10,7 @@ #define _MT_6797_AFE_COMMON_H_ #include <sound/soc.h> +#include <linux/list.h> #include <linux/regmap.h> #include "../common/mtk-base-afe.h" diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index 6c5dd9fc9976..192f4d7b37b6 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -733,6 +733,34 @@ static const struct snd_soc_component_driver mt6797_afe_component = { .probe = mt6797_afe_component_probe, }; +static int mt6797_dai_memif_register(struct mtk_base_afe *afe) +{ + struct mtk_base_afe_dai *dai; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mt6797_memif_dai_driver; + dai->num_dai_drivers = ARRAY_SIZE(mt6797_memif_dai_driver); + + dai->dapm_widgets = mt6797_memif_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mt6797_memif_widgets); + dai->dapm_routes = mt6797_memif_routes; + dai->num_dapm_routes = ARRAY_SIZE(mt6797_memif_routes); + return 0; +} + +typedef int (*dai_register_cb)(struct mtk_base_afe *); +static const dai_register_cb dai_register_cbs[] = { + mt6797_dai_adda_register, + mt6797_dai_pcm_register, + mt6797_dai_hostless_register, + mt6797_dai_memif_register, +}; + static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) { struct mtk_base_afe *afe; @@ -811,29 +839,24 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) } /* init sub_dais */ - afe->num_sub_dais = MT6797_DAI_NUM; - afe->sub_dais = devm_kcalloc(dev, afe->num_sub_dais, - sizeof(*afe->sub_dais), - GFP_KERNEL); - if (!afe->sub_dais) - return -ENOMEM; - - mt6797_dai_adda_register(afe); - mt6797_dai_pcm_register(afe); - mt6797_dai_hostless_register(afe); - - afe->sub_dais[MT6797_MEMIF_DL1].dai_drivers = mt6797_memif_dai_driver; - afe->sub_dais[MT6797_MEMIF_DL1].num_dai_drivers = - ARRAY_SIZE(mt6797_memif_dai_driver); - afe->sub_dais[MT6797_MEMIF_DL1].dapm_widgets = mt6797_memif_widgets; - afe->sub_dais[MT6797_MEMIF_DL1].num_dapm_widgets = - ARRAY_SIZE(mt6797_memif_widgets); - afe->sub_dais[MT6797_MEMIF_DL1].dapm_routes = mt6797_memif_routes; - afe->sub_dais[MT6797_MEMIF_DL1].num_dapm_routes = - ARRAY_SIZE(mt6797_memif_routes); + INIT_LIST_HEAD(&afe->sub_dais); + + for (i = 0; i < ARRAY_SIZE(dai_register_cbs); i++) { + ret = dai_register_cbs[i](afe); + if (ret) { + dev_warn(afe->dev, "dai register i %d fail, ret %d\n", + i, ret); + return ret; + } + } /* init dai_driver and component_driver */ - mtk_afe_combine_sub_dai(afe); + ret = mtk_afe_combine_sub_dai(afe); + if (ret) { + dev_warn(afe->dev, "mtk_afe_combine_sub_dai fail, ret %d\n", + ret); + return ret; + } afe->mtk_afe_hardware = &mt6797_afe_hardware; afe->memif_fs = mt6797_memif_fs; diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c index ad083265ce94..0ac6409c6d61 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c @@ -383,14 +383,20 @@ static struct snd_soc_dai_driver mtk_dai_adda_driver[] = { int mt6797_dai_adda_register(struct mtk_base_afe *afe) { - int id = MT6797_DAI_ADDA; + struct mtk_base_afe_dai *dai; - afe->sub_dais[id].dai_drivers = mtk_dai_adda_driver; - afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver); + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; - afe->sub_dais[id].dapm_widgets = mtk_dai_adda_widgets; - afe->sub_dais[id].num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets); - afe->sub_dais[id].dapm_routes = mtk_dai_adda_routes; - afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes); + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_adda_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver); + + dai->dapm_widgets = mtk_dai_adda_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets); + dai->dapm_routes = mtk_dai_adda_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes); return 0; } diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c b/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c index 4cf985b15a11..ed23e6a53b08 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-hostless.c @@ -100,13 +100,19 @@ static struct snd_soc_dai_driver mtk_dai_hostless_driver[] = { int mt6797_dai_hostless_register(struct mtk_base_afe *afe) { - int id = MT6797_DAI_HOSTLESS_LPBK; + struct mtk_base_afe_dai *dai; - afe->sub_dais[id].dai_drivers = mtk_dai_hostless_driver; - afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver); + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; - afe->sub_dais[id].dapm_routes = mtk_dai_hostless_routes; - afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes); + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_hostless_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_hostless_driver); + + dai->dapm_routes = mtk_dai_hostless_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_hostless_routes); return 0; } diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c index 16d5b5067204..3136f0bc7827 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-pcm.c @@ -298,15 +298,20 @@ static struct snd_soc_dai_driver mtk_dai_pcm_driver[] = { int mt6797_dai_pcm_register(struct mtk_base_afe *afe) { - int id = MT6797_DAI_PCM_1; + struct mtk_base_afe_dai *dai; - afe->sub_dais[id].dai_drivers = mtk_dai_pcm_driver; - afe->sub_dais[id].num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver); + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; - afe->sub_dais[id].dapm_widgets = mtk_dai_pcm_widgets; - afe->sub_dais[id].num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets); - afe->sub_dais[id].dapm_routes = mtk_dai_pcm_routes; - afe->sub_dais[id].num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes); + list_add(&dai->list, &afe->sub_dais); + dai->dai_drivers = mtk_dai_pcm_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver); + + dai->dapm_widgets = mtk_dai_pcm_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets); + dai->dapm_routes = mtk_dai_pcm_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes); return 0; } -- 2.18.0.rc2

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[alsa-devel] [PATCH 1/2] ASoC: mediatek: sub dai use list_head
by KaiChieh Chuang 29 Jun '18

29 Jun '18

use list_head for sub_dais, since original sub_dais array is sparsely occupied Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- .../soc/mediatek/common/mtk-afe-platform-driver.c | 64 ++++++++-------------- sound/soc/mediatek/common/mtk-base-afe.h | 6 +- 2 files changed, 28 insertions(+), 42 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index 0061858..3a5bde6 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -15,20 +15,12 @@ int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) { - struct snd_soc_dai_driver *sub_dai_drivers; + struct mtk_base_afe_dai *dai; size_t num_dai_drivers = 0, dai_idx = 0; - int i; - - if (!afe->sub_dais) { - dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); - return -EINVAL; - } /* calcualte total dai driver size */ - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].dai_drivers && - afe->sub_dais[i].num_dai_drivers != 0) - num_dai_drivers += afe->sub_dais[i].num_dai_drivers; + list_for_each_entry(dai, &afe->sub_dais, list) { + num_dai_drivers += dai->num_dai_drivers; } dev_info(afe->dev, "%s(), num of dai %zd\n", __func__, num_dai_drivers); @@ -42,47 +34,39 @@ int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) if (!afe->dai_drivers) return -ENOMEM; - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].dai_drivers && - afe->sub_dais[i].num_dai_drivers != 0) { - sub_dai_drivers = afe->sub_dais[i].dai_drivers; - /* dai driver */ - memcpy(&afe->dai_drivers[dai_idx], - sub_dai_drivers, - afe->sub_dais[i].num_dai_drivers * - sizeof(struct snd_soc_dai_driver)); - dai_idx += afe->sub_dais[i].num_dai_drivers; - } + list_for_each_entry(dai, &afe->sub_dais, list) { + /* dai driver */ + memcpy(&afe->dai_drivers[dai_idx], + dai->dai_drivers, + dai->num_dai_drivers * + sizeof(struct snd_soc_dai_driver)); + dai_idx += dai->num_dai_drivers; } - return 0; } int mtk_afe_add_sub_dai_control(struct snd_soc_component *component) { struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); - int i; + struct mtk_base_afe_dai *dai; - if (!afe->sub_dais) { - dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); - return -EINVAL; - } - - for (i = 0; i < afe->num_sub_dais; i++) { - if (afe->sub_dais[i].controls) + list_for_each_entry(dai, &afe->sub_dais, list) { + if (dai->controls) snd_soc_add_component_controls(component, - afe->sub_dais[i].controls, - afe->sub_dais[i].num_controls); + dai->controls, + dai->num_controls); - if (afe->sub_dais[i].dapm_widgets) + if (dai->dapm_widgets) snd_soc_dapm_new_controls(&component->dapm, - afe->sub_dais[i].dapm_widgets, - afe->sub_dais[i].num_dapm_widgets); - - if (afe->sub_dais[i].dapm_routes) + dai->dapm_widgets, + dai->num_dapm_widgets); + } + /* add routes after all widgets are added */ + list_for_each_entry(dai, &afe->sub_dais, list) { + if (dai->dapm_routes) snd_soc_dapm_add_routes(&component->dapm, - afe->sub_dais[i].dapm_routes, - afe->sub_dais[i].num_dapm_routes); + dai->dapm_routes, + dai->num_dapm_routes); } snd_soc_dapm_new_widgets(component->dapm.card); diff --git a/sound/soc/mediatek/common/mtk-base-afe.h b/sound/soc/mediatek/common/mtk-base-afe.h index bcf562f0..bd8d5e0 100644 --- a/sound/soc/mediatek/common/mtk-base-afe.h +++ b/sound/soc/mediatek/common/mtk-base-afe.h @@ -46,6 +46,7 @@ struct mtk_base_irq_data { }; struct device; +struct list_head; struct mtk_base_afe_memif; struct mtk_base_afe_irq; struct mtk_base_afe_dai; @@ -72,8 +73,7 @@ struct mtk_base_afe { struct mtk_base_afe_irq *irqs; int irqs_size; - struct mtk_base_afe_dai *sub_dais; - int num_sub_dais; + struct list_head sub_dais; struct snd_soc_dai_driver *dai_drivers; unsigned int num_dai_drivers; @@ -110,6 +110,8 @@ struct mtk_base_afe_dai { unsigned int num_dapm_widgets; const struct snd_soc_dapm_route *dapm_routes; unsigned int num_dapm_routes; + + struct list_head list; }; #endif -- 1.9.1

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Re: [alsa-devel] [PATCH] vgaswitchroo: set audio client id according to bound gpu client id
by Lukas Wunner 29 Jun '18

29 Jun '18

On Fri, Jun 29, 2018 at 08:55:40AM +0000, Qu, Jim wrote: > When our dGPU does suspend by runtime pm. amdgpu driver for dgpu will > also call vgaswtichroo to power off its audio. vgaswitchroo driver will > find audio codec by vgaswitchroo dgpu client id(VGA_SWITCHEROO_DIS). That is no longer the case since v4.17. The HDA controller now runtime suspends autonomously, see commit 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller"). Your patch appears to be geared towards an older kernel version. Please retest on the laptop in question with a v4.17+ kernel. > I think the issue should be observed on both Intel+AMD or AMD+AMD > platform which has the same HW configuration. > > 1.if dGPU has no audio codec. the issue should be always observed. > 2.if both iGPU and dGPU has audio codecs, the issue should be random, > it depends on the first audio found by vgaswitchroo driver is on iGPU > or dGPU. On discrete AMD and Nvidia GPUs, the HDA controller is function 1 and the GPU is function 0 in the same PCI slot. On Intel chipsets, the HDA controller and the GPU have completely different PCI device numbers, e.g. the GPU might be 0000:00:02.0 and the HDA controller might be 0000:00:1b.0. get_bound_vga() checks that the HDA controller is function 1 and there's a GPU in function 0 of the same slot. Thus get_bound_vga() always returns NULL for an Intel HDA controller and the controller is never registered with vga_switcheroo (which is fine). Thanks, Lukas

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[alsa-devel] [PATCH v2 00/13] ARM: pxa: switch to DMA slave maps
by Robert Jarzmik 29 Jun '18

29 Jun '18

This v1 cover letter is quoted in [1]. For maintainers the table below should help you focus on the patches targetted at you, and ignore the other noise. The differences since v1 is by maintainers / topic / patch : - Arnd and Daniel / PXA topic / 0002 devices.c split into pxa25x.c, pxa27x.c and pxa3xx.c - Boris and Daniel / MTD topic / 0005 Review and ack of this one - Arnd and netdev / NET topic / 0006 and 0007 Arnd comment taken, review and ack of these one - Mark and alsa-devel / ASoC topic / 0008 Mark, I couldn't keep your former Ack because : - I changed one line so that the cpu device provides the DMA - I added pxa2xx-i2s which was forgotten Therefore I need a new ack - Arnd and Daniel / SSP topic / 0013 Review and ack of this one Happy review. -- Robert Robert Jarzmik (13): dmaengine: pxa: use a dma slave map ARM: pxa: add dma slave map mmc: pxamci: remove the dmaengine compat need media: pxa_camera: remove the dmaengine compat need mtd: rawnand: marvell: remove the dmaengine compat need net: smc911x: remove the dmaengine compat need net: smc91x: remove the dmaengine compat need ASoC: pxa: remove the dmaengine compat need ata: pata_pxa: remove the dmaengine compat need dmaengine: pxa: document pxad_param dmaengine: pxa: make the filter function internal ARM: pxa: remove the DMA IO resources ARM: pxa: change SSP DMA channels allocation arch/arm/mach-pxa/devices.c | 148 +--------------------------------- arch/arm/mach-pxa/devices.h | 6 +- arch/arm/mach-pxa/pxa25x.c | 41 +++++++++- arch/arm/mach-pxa/pxa27x.c | 42 +++++++++- arch/arm/mach-pxa/pxa3xx.c | 44 +++++++++- arch/arm/plat-pxa/ssp.c | 47 ----------- drivers/ata/pata_pxa.c | 10 +-- drivers/dma/pxa_dma.c | 13 ++- drivers/media/platform/pxa_camera.c | 22 +---- drivers/mmc/host/pxamci.c | 29 +------ drivers/mtd/nand/raw/marvell_nand.c | 17 +--- drivers/net/ethernet/smsc/smc911x.c | 16 +--- drivers/net/ethernet/smsc/smc91x.c | 12 +-- drivers/net/ethernet/smsc/smc91x.h | 1 - include/linux/dma/pxa-dma.h | 20 +++-- include/linux/platform_data/mmp_dma.h | 4 + include/linux/pxa2xx_ssp.h | 2 - sound/arm/pxa2xx-ac97.c | 14 +--- sound/arm/pxa2xx-pcm-lib.c | 6 +- sound/soc/pxa/pxa-ssp.c | 5 +- sound/soc/pxa/pxa2xx-ac97.c | 32 ++------ sound/soc/pxa/pxa2xx-i2s.c | 6 +- 22 files changed, 180 insertions(+), 357 deletions(-) -- 2.11.0 --- [1] Former v1 cover letter This serie is aimed at removing the dmaengine slave compat use, and transfer knowledge of the DMA requestors into architecture code. This was discussed/advised by Arnd a couple of years back, it's almost time. The serie is divided in 3 phasees : - phase 1 : patch 1/15 and patch 2/15 => this is the preparation work - phase 2 : patches 3/15 .. 10/15 => this is the switch of all the drivers => this one will require either an Ack of the maintainers or be taken by them once phase 1 is merged - phase 3 : patches 11/15 => this is the last part, cleanup and removal of export of the DMA filter function As this looks like a patch bomb, each maintainer expressing for his tree either an Ack or "I want to take through my tree" will be spared in the next iterations of this serie. Several of these changes have been tested on actual hardware, including : - pxamci - pxa_camera - smc* - ASoC and SSP

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[alsa-devel] [PATCH 0/7] ASoC: pxa: code refactoring
by Daniel Mack 29 Jun '18

29 Jun '18

This is a series to clean up the code base in the ASoC/PXA area a bit. It folds some implementations, removes some files and brings pcm ops to the CPU DAIS, so they can be used on DT platforms without platform drivers. They are a preparation for some more patches that will follow on top. Daniel Mack (7): ASoC: fold pxa2xx-pcm into its only user, pxa2xx-ac97 ASoC: pxa: clean up function names in pxa2xx-lib ASoC: pxa: move some functions to pxa2xx-lib ASoC: pxa2xx-pcm-lib: fix indenting ASoC: pxa: provide PCM ops for ssp, i2s and ac97 components ASoC: pxa: remove bindings from pxa2xx-pcm ASoC: pxa: make SND_PXA2XX_SOC_I2S selectable .../bindings/sound/mrvl,pxa2xx-pcm.txt | 15 -- include/sound/pxa2xx-lib.h | 13 +- sound/arm/Kconfig | 5 - sound/arm/Makefile | 3 - sound/arm/pxa2xx-ac97.c | 114 ++++++++++------ sound/arm/pxa2xx-pcm-lib.c | 69 ++++++++-- sound/arm/pxa2xx-pcm.c | 129 ------------------ sound/arm/pxa2xx-pcm.h | 27 ---- sound/soc/pxa/Kconfig | 2 +- sound/soc/pxa/pxa-ssp.c | 4 +- sound/soc/pxa/pxa2xx-ac97.c | 3 + sound/soc/pxa/pxa2xx-i2s.c | 3 + sound/soc/pxa/pxa2xx-pcm.c | 73 ---------- 13 files changed, 148 insertions(+), 312 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt delete mode 100644 sound/arm/pxa2xx-pcm.c delete mode 100644 sound/arm/pxa2xx-pcm.h -- 2.17.1

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