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

[alsa-devel] [PATCH] alsatplg: fix topology compiler long option parsing
by Liam Girdwood 14 Jun '17

14 Jun '17

verbose, compile and output options all have a parameter. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- topology/topology.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/topology/topology.c b/topology/topology.c index ffe2f05..097c255 100644 --- a/topology/topology.c +++ b/topology/topology.c @@ -54,10 +54,10 @@ int main(int argc, char *argv[]) snd_tplg_t *snd_tplg; static const char short_options[] = "hc:v:o:"; static const struct option long_options[] = { - {"help", 0, 0, 'h'}, - {"verbose", 0, 0, 'v'}, - {"compile", 0, 0, 'c'}, - {"output", 0, 0, 'o'}, + {"help", 0, NULL, 'h'}, + {"verbose", 1, NULL, 'v'}, + {"compile", 1, NULL, 'c'}, + {"output", 1, NULL, 'o'}, {0, 0, 0, 0}, }; char *source_file = NULL, *output_file = NULL; -- 1.9.1

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[alsa-devel] [PATCH 0/2] ALSA: pcm: remove unused local commands
by Takashi Sakamoto 14 Jun '17

14 Jun '17

Hi, A callback of 'struct snd_pcm_ops.ioctl' is internally used for ALSA pcm core to communicate to each drivers for some purposes; e.g. ask the size of FIFO in handled devices. However, in 'include/sound/pcm.h', there're some commands for the callback, which has never been used. This commit removes the unused commands to prevent developers from undocumented and unreferenced features. Takashi Sakamoto (2): ALSA: pcm: remove SNDRV_PCM_IOCTL1_INFO internal command ALSA: pcm: remove SNDRV_PCM_IOCTL1_GSTATE internal command include/sound/pcm.h | 4 ++-- sound/core/pcm_lib.c | 2 -- sound/core/pcm_native.c | 6 +----- 3 files changed, 3 insertions(+), 9 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH] ASoC: simple_card_utils: add EXPORT_SYMBOL_GPL() for asoc_simple_card_clk_xxx()
by Kuninori Morimoto 14 Jun '17

14 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 891caea41746 ("ASoC: simple_card_utils: add asoc_simple_card_clk_xxx()") added new asoc_simple_card_clk_xxx(), but, it didn't have EXPORT_SYMBOL_GPL(). This patch adds it. Otherwise, we will get below error ERROR: "asoc_simple_card_clk_enable" [sound/soc/generic/snd-soc-simple-scu-card.ko] undefined! ERROR: "asoc_simple_card_clk_disable" [sound/soc/generic/snd-soc-simple-scu-card.ko] undefined! ERROR: "asoc_simple_card_clk_enable" [sound/soc/generic/snd-soc-simple-card.ko] undefined! ERROR: "asoc_simple_card_clk_disable" [sound/soc/generic/snd-soc-simple-card.ko] undefined! ERROR: "asoc_simple_card_clk_enable" [sound/soc/generic/snd-soc-audio-graph-scu-card.ko] undefined! ERROR: "asoc_simple_card_clk_disable" [sound/soc/generic/snd-soc-audio-graph-scu-card.ko] undefined! ERROR: "asoc_simple_card_clk_enable" [sound/soc/generic/snd-soc-audio-graph-card.ko] undefined! ERROR: "asoc_simple_card_clk_disable" [sound/soc/generic/snd-soc-audio-graph-card.ko] undefined! Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card-utils.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index b3fc8ce..26d64fa 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -160,11 +160,13 @@ int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai) { return clk_prepare_enable(dai->clk); } +EXPORT_SYMBOL_GPL(asoc_simple_card_clk_enable); void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai) { clk_disable_unprepare(dai->clk); } +EXPORT_SYMBOL_GPL(asoc_simple_card_clk_disable); int asoc_simple_card_parse_clk(struct device *dev, struct device_node *node, -- 1.9.1

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[alsa-devel] [PATCH v4] ASoC: sgtl5000: add avc support
by Richard Leitner 14 Jun '17

14 Jun '17

The sgtl5000 features an automatic volume control block (AVC), which reduces loud signals and amplifies low level signals for easier listening. This patch adds support for this AVC block to the driver. Apart from the "AVC Switch" control which enables the block following controls for the configuration of AVC are added: + AVC Threshold Volume: threshold where audio is compressed when the measured level is above or expanded when below + AVC Max Gain Volume: maximum gain which can be applied when the measured audio level is below threshold + AVC Hard Limiter Switch: when enabled the signal is limited to the programmed threshold. + AVC Integrator Response: response time of the integrator The AVC block is enabled and configured using the DAP_AVC_CTRL and DAP_AVC_THRESHOLD registers. Following 2 checkpatch.pl strict checks are ignored because the indentation style is different for the struct snd_kcontrol_new definition: patch:147: CHECK: Alignment should match open parenthesis patch:150: CHECK: Alignment should match open parenthesis Signed-off-by: Richard Leitner <richard.leitner(a)skidata.com> --- CHANGES v4: - reject out-of-bounds avc threshold values with EINVAL CHANGES v3: - rename "AVC Enable Switch" to "AVC Switch" - add AVC configuration controls CHANGES v2: - added trailing "Switch" as covered in ControlNames.txt --- sound/soc/codecs/sgtl5000.c | 89 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 89 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 5a2702e..8f6814c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -74,6 +74,20 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_AVC_DECAY, 0x0050 }, }; +/* AVC: Threshold dB -> register: pre-calculated values */ +static const u16 avc_thr_db2reg[97] = { + 0x5168, 0x488E, 0x40AA, 0x39A1, 0x335D, 0x2DC7, 0x28CC, 0x245D, 0x2068, + 0x1CE2, 0x19BE, 0x16F1, 0x1472, 0x1239, 0x103E, 0x0E7A, 0x0CE6, 0x0B7F, + 0x0A3F, 0x0922, 0x0824, 0x0741, 0x0677, 0x05C3, 0x0522, 0x0493, 0x0414, + 0x03A2, 0x033D, 0x02E3, 0x0293, 0x024B, 0x020B, 0x01D2, 0x019F, 0x0172, + 0x014A, 0x0126, 0x0106, 0x00E9, 0x00D0, 0x00B9, 0x00A5, 0x0093, 0x0083, + 0x0075, 0x0068, 0x005D, 0x0052, 0x0049, 0x0041, 0x003A, 0x0034, 0x002E, + 0x0029, 0x0025, 0x0021, 0x001D, 0x001A, 0x0017, 0x0014, 0x0012, 0x0010, + 0x000E, 0x000D, 0x000B, 0x000A, 0x0009, 0x0008, 0x0007, 0x0006, 0x0005, + 0x0005, 0x0004, 0x0004, 0x0003, 0x0003, 0x0002, 0x0002, 0x0002, 0x0002, + 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0001, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000}; + /* regulator supplies for sgtl5000, VDDD is an optional external supply */ enum sgtl5000_regulator_supplies { VDDA, @@ -382,6 +396,65 @@ static int dac_put_volsw(struct snd_kcontrol *kcontrol, return 0; } +/* + * custom function to get AVC threshold + * + * The threshold dB is calculated by rearranging the calculation from the + * avc_put_threshold function: register_value = 10^(dB/20) * 0.636 * 2^15 ==> + * dB = ( fls(register_value) - 14.347 ) * 6.02 + * + * As this calculation is expensive and the threshold dB values may not exeed + * 0 to 96 we use pre-calculated values. + */ +static int avc_get_threshold(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int db, i; + u16 reg = snd_soc_read(codec, SGTL5000_DAP_AVC_THRESHOLD); + + /* register value 0 => -96dB */ + if (!reg) { + ucontrol->value.integer.value[0] = 96; + ucontrol->value.integer.value[1] = 96; + return 0; + } + + /* get dB from register value (rounded down) */ + for (i = 0; avc_thr_db2reg[i] > reg; i++) + ; + db = i; + + ucontrol->value.integer.value[0] = db; + ucontrol->value.integer.value[1] = db; + + return 0; +} + +/* + * custom function to put AVC threshold + * + * The register value is calculated by following formula: + * register_value = 10^(dB/20) * 0.636 * 2^15 + * As this calculation is expensive and the threshold dB values may not exeed + * 0 to 96 we use pre-calculated values. + */ +static int avc_put_threshold(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int db; + u16 reg; + + db = (int)ucontrol->value.integer.value[0]; + if (db < 0 || db > 96) + return -EINVAL; + reg = avc_thr_db2reg[db]; + snd_soc_write(codec, SGTL5000_DAP_AVC_THRESHOLD, reg); + + return 0; +} + static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0); /* tlv for mic gain, 0db 20db 30db 40db */ @@ -396,6 +469,12 @@ static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0); /* tlv for lineout volume, 31 steps of .5db each */ static const DECLARE_TLV_DB_SCALE(lineout_volume, -1550, 50, 0); +/* tlv for dap avc max gain, 0db, 6db, 12db */ +static const DECLARE_TLV_DB_SCALE(avc_max_gain, 0, 600, 0); + +/* tlv for dap avc threshold, */ +static const DECLARE_TLV_DB_MINMAX(avc_threshold, 0, 9600); + static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { /* SOC_DOUBLE_S8_TLV with invert */ { @@ -434,6 +513,16 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 0x1f, 1, lineout_volume), SOC_SINGLE("Lineout Playback Switch", SGTL5000_CHIP_ANA_CTRL, 8, 1, 1), + + /* Automatic Volume Control (DAP AVC) */ + SOC_SINGLE("AVC Switch", SGTL5000_DAP_AVC_CTRL, 0, 1, 0), + SOC_SINGLE("AVC Hard Limiter Switch", SGTL5000_DAP_AVC_CTRL, 5, 1, 0), + SOC_SINGLE_TLV("AVC Max Gain Volume", SGTL5000_DAP_AVC_CTRL, 12, 2, 0, + avc_max_gain), + SOC_SINGLE("AVC Integrator Response", SGTL5000_DAP_AVC_CTRL, 8, 3, 0), + SOC_SINGLE_EXT_TLV("AVC Threshold Volume", SGTL5000_DAP_AVC_THRESHOLD, + 0, 96, 0, avc_get_threshold, avc_put_threshold, + avc_threshold), }; /* mute the codec used by alsa core */ -- 2.1.4

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[alsa-devel] [PATCH v2] ASoC: intel: bxt: Constify hw_constraints
by Takashi Iwai 14 Jun '17

14 Jun '17

snd_pcm_hw_constraint_list(), *_ratnums() and *_ratdens() receive the const pointers. Constify the corresponding static objects for better hardening. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: rebased to topic/intel branch sound/soc/intel/boards/bxt_da7219_max98357a.c | 12 ++++++------ sound/soc/intel/boards/bxt_rt298.c | 12 ++++++------ 2 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 3a8c4d954a91..1429eb391da8 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -238,31 +238,31 @@ static int broxton_da7219_fe_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static unsigned int rates[] = { +static const unsigned int rates[] = { 48000, }; -static struct snd_pcm_hw_constraint_list constraints_rates = { +static const struct snd_pcm_hw_constraint_list constraints_rates = { .count = ARRAY_SIZE(rates), .list = rates, .mask = 0, }; -static unsigned int channels[] = { +static const unsigned int channels[] = { DUAL_CHANNEL, }; -static struct snd_pcm_hw_constraint_list constraints_channels = { +static const struct snd_pcm_hw_constraint_list constraints_channels = { .count = ARRAY_SIZE(channels), .list = channels, .mask = 0, }; -static unsigned int channels_quad[] = { +static const unsigned int channels_quad[] = { QUAD_CHANNEL, }; -static struct snd_pcm_hw_constraint_list constraints_channels_quad = { +static const struct snd_pcm_hw_constraint_list constraints_channels_quad = { .count = ARRAY_SIZE(channels_quad), .list = channels_quad, .mask = 0, diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index 36ee7480e9f1..0c3a3cbcb884 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -207,11 +207,11 @@ static const struct snd_soc_ops broxton_rt298_ops = { .hw_params = broxton_rt298_hw_params, }; -static unsigned int rates[] = { +static const unsigned int rates[] = { 48000, }; -static struct snd_pcm_hw_constraint_list constraints_rates = { +static const struct snd_pcm_hw_constraint_list constraints_rates = { .count = ARRAY_SIZE(rates), .list = rates, .mask = 0, @@ -227,11 +227,11 @@ static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static unsigned int channels_dmic[] = { +static const unsigned int channels_dmic[] = { 1, 2, 3, 4, }; -static struct snd_pcm_hw_constraint_list constraints_dmic_channels = { +static const struct snd_pcm_hw_constraint_list constraints_dmic_channels = { .count = ARRAY_SIZE(channels_dmic), .list = channels_dmic, .mask = 0, @@ -253,11 +253,11 @@ static const struct snd_soc_ops broxton_dmic_ops = { .startup = broxton_dmic_startup, }; -static unsigned int channels[] = { +static const unsigned int channels[] = { 2, }; -static struct snd_pcm_hw_constraint_list constraints_channels = { +static const struct snd_pcm_hw_constraint_list constraints_channels = { .count = ARRAY_SIZE(channels), .list = channels, .mask = 0, -- 2.13.1

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[alsa-devel] [PATCH] ASoC: rsnd: reduce confusable macro parameter magic
by Kuninori Morimoto 14 Jun '17

14 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd_dai_call() macro is using "priv" inside. Thus, if caller function doesn't have "priv" related operation, strange phenomenon occur which code is using "priv", but compiler indicates "unused variable 'priv'". From code point of view, it is not problem, but it is very confusable. This patch removes "priv" from rsnd_dai_call() macro, and adds "priv" on caller function. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index e2d554f3..9b9e898 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -469,8 +469,7 @@ static int rsnd_status_update(u32 *status, #define rsnd_dai_call(fn, io, param...) \ ({ \ - struct rsnd_priv *priv = rsnd_io_to_priv(io); \ - struct device *dev = rsnd_priv_to_dev(priv); \ + struct device *dev = rsnd_priv_to_dev(rsnd_io_to_priv(io)); \ struct rsnd_mod *mod; \ int is_play = rsnd_io_is_play(io); \ int ret = 0, i; \ @@ -741,6 +740,7 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); + struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); int ret; @@ -758,6 +758,7 @@ static void rsnd_soc_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); + struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); /* -- 1.9.1

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[alsa-devel] [PATCH] ASoC: ak4613: add missing 64000 in ak4613_dai_hw_params()
by Kuninori Morimoto 14 Jun '17

14 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4613.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index c87ec49..27c44c7 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -321,6 +321,7 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, case 48000: ctrl2 = DFS_NORMAL_SPEED; break; + case 64000: case 88200: case 96000: ctrl2 = DFS_DOUBLE_SPEED; -- 1.9.1

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[alsa-devel] [PATCH 0/5] ASoC: simple/graph card: use asoc_simple_card_of_parse_tdm()
by Kuninori Morimoto 14 Jun '17

14 Jun '17

Hi Mark These are not a big deal, but current simple/graph card drivers are using asoc_simple_dai's tx_slot_mask, rx_slot_mask, slots, slot_width directly to parse TDM. Encapsulation is one of simple card util's purpose. These are cleanup for it. Kuninori Morimoto (5): ASoC: simple-card-utils: add asoc_simple_card_of_parse_tdm() ASoC: simple-card: use asoc_simple_card_of_parse_tdm() ASoC: simple-scu-card: use asoc_simple_card_of_parse_tdm() ASoC: audio-graph-card: use asoc_simple_card_of_parse_tdm() ASoC: audio-graph-scu-card: use asoc_simple_card_of_parse_tdm() include/sound/simple_card_utils.h | 6 ++++++ sound/soc/generic/audio-graph-card.c | 12 ++---------- sound/soc/generic/audio-graph-scu-card.c | 6 +----- sound/soc/generic/simple-card.c | 10 ++-------- sound/soc/generic/simple-scu-card.c | 6 +----- 5 files changed, 12 insertions(+), 28 deletions(-) -- 1.9.1

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[alsa-devel] MEDION laptop full of Intel components
by fulgor fulgor 14 Jun '17

14 Jun '17

Hello, I bought a MEDION laptop in December 2016 I had some problems like crashes but with manjaro I still have no sound I believe the device is not detected I tried to get hardware informations in a file

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[alsa-devel] [asoc:topic/of-graph 19/22] ERROR: "asoc_simple_card_clk_enable" [sound/soc/generic/snd-soc-simple-card.ko] undefined!
by kbuild test robot 14 Jun '17

14 Jun '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/of-graph head: d471d55934ca8b4f38535207589df4e3cc8b1484 commit: 3ab50c4f98434080c1f73fc56d8d8b38364c6cd8 [19/22] ASoC: simple-card: use asoc_simple_card_clk_xxx() config: x86_64-allmodconfig (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout 3ab50c4f98434080c1f73fc56d8d8b38364c6cd8 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): >> ERROR: "asoc_simple_card_clk_enable" [sound/soc/generic/snd-soc-simple-card.ko] undefined! >> ERROR: "asoc_simple_card_clk_disable" [sound/soc/generic/snd-soc-simple-card.ko] undefined! --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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