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

[alsa-devel] [PATCH alsa-lib 0/2] pcm: minor code cleanup
by Takashi Sakamoto 23 Jun '17

23 Jun '17

Hi, This includes two patches from my previous RFC for minor code cleanup: [alsa-devel] [alsa-lib][RFC][PATCH 0/9] pcm: handle status/control mapping independently http://mailman.alsa-project.org/pipermail/alsa-devel/2017-June/122130.html Takashi Sakamoto (2): pcm: obsolete 'mmap_emulation' parameter of snd_pcm_hw_open_fd() pcm: minor code cleanup for ioctl call src/pcm/pcm_direct.c | 2 +- src/pcm/pcm_hw.c | 9 +++------ src/pcm/pcm_local.h | 3 ++- 3 files changed, 6 insertions(+), 8 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH v2 0/2] Add ZTE zx-aud96p22 audio codec driver
by Shawn Guo 23 Jun '17

23 Jun '17

From: Shawn Guo <shawn.guo(a)linaro.org> It adds the ASoC driver for zx-aud96p22 audio codec found on ZTE ZX family SoCs. Changes for v2: - Follow document control-names.rst to name controls in a standard way, so that user space mixer UIs know what to do with them. - Drop the GPIO support for an optional external amplifier, as it belongs to machine driver rather than codec driver. - Support overall power bit with a DAPM supply widget and attach it to both DAC and ADC components. - Instead of reset DAC/ADC data path in startup/shutdown hooks, handle it with DAPM event, so that startup/shutdown hooks get dropped completely. Baoyou Xie (2): ASoC: add bindings for ZTE zx-aud96p22 audio codec ASoC: zx_aud96p22: add ZTE ZX AUD96P22 codec driver .../devicetree/bindings/sound/zte,zx-aud96p22.txt | 24 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/zx_aud96p22.c | 403 +++++++++++++++++++++ 4 files changed, 434 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt create mode 100644 sound/soc/codecs/zx_aud96p22.c -- 1.9.1

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[alsa-devel] [PATCH] ASoC: nau8825: default value for property
by John Hsu 23 Jun '17

23 Jun '17

Assign default value for codec private data when property not given. If without those default value and property, the codec will work abnormally. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- sound/soc/codecs/nau8825.c | 53 +++++++++++++++++++++++++++++++++++----------- 1 file changed, 41 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 97fbeba..c00b86d 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2429,6 +2429,7 @@ static void nau8825_print_device_properties(struct nau8825 *nau8825) static int nau8825_read_device_properties(struct device *dev, struct nau8825 *nau8825) { + int ret; nau8825->jkdet_enable = device_property_read_bool(dev, "nuvoton,jkdet-enable"); @@ -2436,30 +2437,58 @@ static int nau8825_read_device_properties(struct device *dev, "nuvoton,jkdet-pull-enable"); nau8825->jkdet_pull_up = device_property_read_bool(dev, "nuvoton,jkdet-pull-up"); - device_property_read_u32(dev, "nuvoton,jkdet-polarity", + ret = device_property_read_u32(dev, "nuvoton,jkdet-polarity", &nau8825->jkdet_polarity); - device_property_read_u32(dev, "nuvoton,micbias-voltage", + if (ret) + nau8825->jkdet_polarity = 1; + ret = device_property_read_u32(dev, "nuvoton,micbias-voltage", &nau8825->micbias_voltage); - device_property_read_u32(dev, "nuvoton,vref-impedance", + if (ret) + nau8825->micbias_voltage = 6; + ret = device_property_read_u32(dev, "nuvoton,vref-impedance", &nau8825->vref_impedance); - device_property_read_u32(dev, "nuvoton,sar-threshold-num", + if (ret) + nau8825->vref_impedance = 2; + ret = device_property_read_u32(dev, "nuvoton,sar-threshold-num", &nau8825->sar_threshold_num); - device_property_read_u32_array(dev, "nuvoton,sar-threshold", + if (ret) + nau8825->sar_threshold_num = 4; + ret = device_property_read_u32_array(dev, "nuvoton,sar-threshold", nau8825->sar_threshold, nau8825->sar_threshold_num); - device_property_read_u32(dev, "nuvoton,sar-hysteresis", + if (ret) { + nau8825->sar_threshold[0] = 0x08; + nau8825->sar_threshold[1] = 0x12; + nau8825->sar_threshold[2] = 0x26; + nau8825->sar_threshold[3] = 0x73; + } + ret = device_property_read_u32(dev, "nuvoton,sar-hysteresis", &nau8825->sar_hysteresis); - device_property_read_u32(dev, "nuvoton,sar-voltage", + if (ret) + nau8825->sar_hysteresis = 0; + ret = device_property_read_u32(dev, "nuvoton,sar-voltage", &nau8825->sar_voltage); - device_property_read_u32(dev, "nuvoton,sar-compare-time", + if (ret) + nau8825->sar_voltage = 6; + ret = device_property_read_u32(dev, "nuvoton,sar-compare-time", &nau8825->sar_compare_time); - device_property_read_u32(dev, "nuvoton,sar-sampling-time", + if (ret) + nau8825->sar_compare_time = 1; + ret = device_property_read_u32(dev, "nuvoton,sar-sampling-time", &nau8825->sar_sampling_time); - device_property_read_u32(dev, "nuvoton,short-key-debounce", + if (ret) + nau8825->sar_sampling_time = 1; + ret = device_property_read_u32(dev, "nuvoton,short-key-debounce", &nau8825->key_debounce); - device_property_read_u32(dev, "nuvoton,jack-insert-debounce", + if (ret) + nau8825->key_debounce = 3; + ret = device_property_read_u32(dev, "nuvoton,jack-insert-debounce", &nau8825->jack_insert_debounce); - device_property_read_u32(dev, "nuvoton,jack-eject-debounce", + if (ret) + nau8825->jack_insert_debounce = 7; + ret = device_property_read_u32(dev, "nuvoton,jack-eject-debounce", &nau8825->jack_eject_debounce); + if (ret) + nau8825->jack_eject_debounce = 0; nau8825->mclk = devm_clk_get(dev, "mclk"); if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { -- 2.6.4

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[alsa-devel] ASoC: nau8825: fix jack type detection issue after resume
by John Hsu 23 Jun '17

23 Jun '17

Fix the issue that mic type detection error after resume. The patch help to fix the issue as follows: Google issue 37973093: CTIA/OMTP jack type detection failure after resume Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- sound/soc/codecs/nau8825.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index fd307f5..fb9f1d4 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1612,7 +1612,6 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) snd_soc_dapm_sync(dapm); break; case 2: - case 3: dev_dbg(nau8825->dev, "CTIA (micgnd2) mic connected\n"); type = SND_JACK_HEADSET; @@ -1632,6 +1631,11 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) snd_soc_dapm_force_enable_pin(dapm, "SAR"); snd_soc_dapm_sync(dapm); break; + case 3: + /* detect error case */ + dev_err(nau8825->dev, "detection error; disable mic function\n"); + type = SND_JACK_HEADPHONE; + break; } /* Leaving HPOL/R grounded after jack insert by default. They will be @@ -2328,6 +2332,13 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: + /* Reset the configuration of jack type for detection */ + /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, 0); + /* ground HPL/HPR, MICGRND1/2 */ + regmap_update_bits(nau8825->regmap, + NAU8825_REG_HSD_CTRL, 0xf, 0xf); /* Cancel and reset cross talk detection funciton */ nau8825_xtalk_cancel(nau8825); /* Turn off all interruptions before system shutdown. Keep the @@ -2351,6 +2362,10 @@ static int __maybe_unused nau8825_suspend(struct snd_soc_codec *codec) disable_irq(nau8825->irq); snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); + /* Power down codec power; don't suppoet button wakeup */ + snd_soc_dapm_disable_pin(nau8825->dapm, "SAR"); + snd_soc_dapm_disable_pin(nau8825->dapm, "MICBIAS"); + snd_soc_dapm_sync(nau8825->dapm); regcache_cache_only(nau8825->regmap, true); regcache_mark_dirty(nau8825->regmap); -- 2.6.4

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[alsa-devel] ASoC: nau8825: make crosstalk function optional
by John Hsu 23 Jun '17

23 Jun '17

Make crosstalk functoin optional. The jack detection can speed up without crosstalk detection. Let the decision of function usage to platform design. The patch helps the issue concern as follows: Google issue 35574278: Chell_headphone pop back from S3 Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- sound/soc/codecs/nau8825.c | 6 +++++- sound/soc/codecs/nau8825.h | 1 + 2 files changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index ff827bb..fd307f5 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1682,7 +1682,7 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { if (nau8825_is_jack_inserted(regmap)) { event |= nau8825_jack_insert(nau8825); - if (!nau8825->high_imped) { + if (!nau8825->xtalk_bypass && !nau8825->high_imped) { /* Apply the cross talk suppression in the * headset without high impedance. */ @@ -2498,6 +2498,10 @@ static int nau8825_read_device_properties(struct device *dev, &nau8825->jack_eject_debounce); if (ret) nau8825->jack_eject_debounce = 0; + ret = device_property_read_u32(dev, "nuvoton,crosstalk-bypass", + &nau8825->xtalk_bypass); + if (ret) + nau8825->xtalk_bypass = 1; nau8825->mclk = devm_clk_get(dev, "mclk"); if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 514fd13..8aee5c86 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -476,6 +476,7 @@ struct nau8825 { int xtalk_event_mask; bool xtalk_protect; int imp_rms[NAU8825_XTALK_IMM]; + int xtalk_bypass; }; int nau8825_enable_jack_detect(struct snd_soc_codec *codec, -- 2.6.4

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[alsa-devel] [PATCH] ASoC: audio-graph-card: tidyup asoc_simple_card_canonicalize_cpu() parameter
by Kuninori Morimoto 23 Jun '17

23 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> asoc_simple_card_canonicalize_cpu() 2nd param is asking CPU component's DAI links, not Card links. This patch fixup it. Otherwise, audio-graph-card can't handle CPU component correctly if CPU has mult-DAIs and Card uses only one of them Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 885b405..ee752f62 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -100,7 +100,6 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, struct graph_dai_props *dai_props = graph_priv_to_props(priv, idx); struct asoc_simple_dai *cpu_dai = &dai_props->cpu_dai; struct asoc_simple_dai *codec_dai = &dai_props->codec_dai; - struct snd_soc_card *card = graph_priv_to_card(priv); struct device_node *cpu_ep = of_get_next_child(cpu_port, NULL); struct device_node *codec_ep = of_graph_get_remote_endpoint(cpu_ep); struct device_node *rcpu_ep = of_graph_get_remote_endpoint(codec_ep); @@ -162,7 +161,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, dai_link->init = asoc_graph_card_dai_init; asoc_simple_card_canonicalize_cpu(dai_link, - card->num_links == 1); + of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); dai_link_of_err: of_node_put(cpu_ep); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: audio-graph-scu-card: tidyup asoc_simple_card_canonicalize_cpu() parameter
by Kuninori Morimoto 23 Jun '17

23 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> asoc_simple_card_canonicalize_cpu() 2nd param is asking CPU component's DAI links, not Card links. This patch fixup it. Otherwise, audio-graph-card can't handle CPU component correctly if CPU has mult-DAIs and Card uses only one of them Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-scu-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 05934b2..061c7a6 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -125,7 +125,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep, /* card->num_links includes Codec */ asoc_simple_card_canonicalize_cpu(dai_link, - (card->num_links - 1) == 1); + of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); } else { /* FE is dummy */ dai_link->cpu_of_node = NULL; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: audio-graph-scu-card: support 2nd codec endpoint on DT
by Kuninori Morimoto 23 Jun '17

23 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> audio-graph-scu-card can handle below connection which is mainly for sound mixing purpose. +----------+ +-------+ | CPU0--+--|-->| Codec | | | | +-------+ | CPU1--+ | +----------+ From OF-graph point of view, it should have CPU0 <-> Codec, and CPU1 <-> Codec on DT. But current driver doesn't care about 2nd connection of Codec, because it is dummy from DPCM point of view. This patch can care 2nd Codec connection, and it should be supported from OF-graph point of view. It still have backward compatibility. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../bindings/sound/audio-graph-scu-card.txt | 9 +++++-- sound/soc/generic/audio-graph-scu-card.c | 28 +++++++++++++++++++--- 2 files changed, 32 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt index b63c559..8b8afe9 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt +++ b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt @@ -90,9 +90,12 @@ Example 2. 2 CPU 1 Codec (Mixing) ... port { - codec_endpoint: endpoint { + codec_endpoint0: endpoint { remote-endpoint = <&cpu_endpoint0>; }; + codec_endpoint1: endpoint { + remote-endpoint = <&cpu_endpoint1>; + }; }; }; @@ -101,7 +104,7 @@ Example 2. 2 CPU 1 Codec (Mixing) ports { cpu_port0: port { cpu_endpoint0: endpoint { - remote-endpoint = <&codec_endpoint>; + remote-endpoint = <&codec_endpoint0>; dai-format = "left_j"; ... @@ -109,6 +112,8 @@ Example 2. 2 CPU 1 Codec (Mixing) }; cpu_port1: port { cpu_endpoint1: endpoint { + remote-endpoint = <&codec_endpoint1>; + dai-format = "left_j"; ... }; diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 061c7a6..dcd2df3 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -183,6 +183,8 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) struct device_node *cpu_ep; struct device_node *codec_ep; struct device_node *rcpu_ep; + struct device_node *codec_port; + struct device_node *codec_port_old; unsigned int daifmt = 0; int dai_idx, ret; int rc, codec; @@ -235,6 +237,7 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) } dai_idx = 0; + codec_port_old = NULL; for (codec = 0; codec < 2; codec++) { /* * To listup valid sounds continuously, @@ -245,15 +248,22 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) cpu_port = it.node; cpu_ep = of_get_next_child(cpu_port, NULL); codec_ep = of_graph_get_remote_endpoint(cpu_ep); + codec_port = of_graph_get_port_parent(codec_ep); of_node_put(cpu_port); of_node_put(cpu_ep); of_node_put(codec_ep); + of_node_put(codec_port); if (codec) { - if (!codec_ep) + if (!codec_port) continue; + if (codec_port_old == codec_port) + continue; + + codec_port_old = codec_port; + /* Back-End (= Codec) */ ret = asoc_graph_card_dai_link_of(codec_ep, priv, daifmt, dai_idx++, 0); if (ret < 0) @@ -284,22 +294,34 @@ static int asoc_graph_get_dais_count(struct device *dev) struct device_node *cpu_port; struct device_node *cpu_ep; struct device_node *codec_ep; + struct device_node *codec_port; + struct device_node *codec_port_old; int count = 0; int rc; + codec_port_old = NULL; of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { cpu_port = it.node; cpu_ep = of_get_next_child(cpu_port, NULL); codec_ep = of_graph_get_remote_endpoint(cpu_ep); + codec_port = of_graph_get_port_parent(codec_ep); of_node_put(cpu_port); of_node_put(cpu_ep); of_node_put(codec_ep); + of_node_put(codec_port); if (cpu_ep) count++; - if (codec_ep) - count++; + + if (!codec_port) + continue; + + if (codec_port_old == codec_port) + continue; + + count++; + codec_port_old = codec_port; } return count; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rt5645: enable speaker protection features
by Bard Liao 23 Jun '17

23 Jun '17

This patch is uploaded for enabling the speaker protection features of the audio codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 162044d..9344af2 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -59,10 +59,11 @@ static const struct regmap_range_cfg rt5645_ranges[] = { static const struct reg_sequence init_list[] = { {RT5645_PR_BASE + 0x3d, 0x3600}, - {RT5645_PR_BASE + 0x1c, 0xfd20}, + {RT5645_PR_BASE + 0x1c, 0xfd70}, {RT5645_PR_BASE + 0x20, 0x611f}, {RT5645_PR_BASE + 0x21, 0x4040}, {RT5645_PR_BASE + 0x23, 0x0004}, + {RT5645_CLSD_OUT_CTRL, 0xa0e8}, {RT5645_ASRC_4, 0x0120}, }; @@ -171,7 +172,7 @@ static const struct reg_default rt5645_reg[] = { { 0x9c, 0x0aea }, { 0x9d, 0x000c }, { 0x9e, 0x0400 }, - { 0xa0, 0xa0a8 }, + { 0xa0, 0xa0e8 }, { 0xa1, 0x0059 }, { 0xa2, 0x0001 }, { 0xae, 0x6000 }, @@ -328,7 +329,7 @@ static const struct reg_default rt5650_reg[] = { { 0x9c, 0x0aea }, { 0x9d, 0x000c }, { 0x9e, 0x0400 }, - { 0xa0, 0xa0a8 }, + { 0xa0, 0xa0e8 }, { 0xa1, 0x0059 }, { 0xa2, 0x0001 }, { 0xae, 0x6000 }, -- 2.7.4

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[alsa-devel] [PATCH 0/3] ALSA: Add the explicit appl_ptr sync support
by Takashi Iwai 23 Jun '17

23 Jun '17

Hi, this is a patchset for the issues we've discussed recently lengthily, the explicit sync of appl_ptr by disabling the control mmap. The first patch achieves it for a driver setting the new INFO flag. The second patch is for a generic ABI extension I wanted to implement for long time. It allows user-space reporting its supporting protocol version, so that the kernel can behave differently. It's especially useful for extending the ABI struct to use the reserved bits. The third patch is the optimization by the second patch; it simply skips the previous workaround when the user-space declares itself being new enough to support the feature. The corresponding alsa-lib patchset will follow in another thread. thanks, Takashi === Takashi Iwai (3): ALSA: pcm: Add the explicit appl_ptr sync support ALSA: pcm: Add an ioctl to specify the supported protocol version ALSA: pcm: Limit the appl_ptr sync workaround only for old user-space include/sound/pcm.h | 1 + include/uapi/sound/asound.h | 4 +++- sound/core/pcm_compat.c | 1 + sound/core/pcm_native.c | 28 ++++++++++++++++++++++++++++ 4 files changed, 33 insertions(+), 1 deletion(-) -- 2.13.1

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