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

[alsa-devel] [PATCH 3/5] staging: max98927: Updated volatile register list
by Ryan Lee 19 May '17

19 May '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98927.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index cdee3a3..e787a03 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -586,6 +586,12 @@ static bool max98927_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case MAX98927_R0001_INT_RAW1 ... MAX98927_R0009_INT_FLAG3: + case MAX98927_R004C_MEAS_ADC_CH0_READ: + case MAX98927_R004D_MEAS_ADC_CH1_READ: + case MAX98927_R004E_MEAS_ADC_CH2_READ: + case MAX98927_R0051_BROWNOUT_STATUS: + case MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ: + case MAX98927_R01FF_REV_ID: return true; default: return false; -- 2.7.4

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[alsa-devel] [PATCH 2/5] staging: max98927: Added controls for Envelope tracking
by Ryan Lee 19 May '17

19 May '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98927.c | 20 ++++++++++++++++---- sound/soc/codecs/max98927.h | 4 ++++ 2 files changed, 20 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index 99d6e41..cdee3a3 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -614,6 +614,18 @@ static SOC_ENUM_SINGLE_DECL(max98927_current_limit, MAX98927_R0042_BOOST_CTRL1, 1, max98927_current_limit_text); +static const char * const max98927_env_track_headroom_text[] = { + "0.000V", "0.125V", "0.250V", "0.375V", "0.500V", "0.625V", + "0.750V", "0.875V", "1.000V", "1.125V", "1.250V", "1.375V", + "1.500V", "1.625V", "1.750V", "1.875V", "2.000V", "2.125V", + "2.250V", "2.375V", "2.500V", "2.625V", "2.750V", "2.875V", + "3.000V", "3.125V", "3.250V", "3.375V", "3.500V" +}; + +static SOC_ENUM_SINGLE_DECL(max98927_env_track_headroom, + MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM, 0, + max98927_env_track_headroom_text); + static const struct snd_kcontrol_new max98927_snd_controls[] = { SOC_SINGLE_TLV("Speaker Volume", MAX98927_R003C_SPK_GAIN, 0, 6, 0, @@ -631,6 +643,9 @@ static const struct snd_kcontrol_new max98927_snd_controls[] = { MAX98927_AMP_VOL_SEL_SHIFT, 1, 0), SOC_ENUM("Boost Output Voltage", max98927_boost_voltage), SOC_ENUM("Current Limit", max98927_current_limit), + SOC_SINGLE("EnvTrack Switch", MAX98927_R0086_ENV_TRACK_CTRL, + MAX98927_ENV_TRACKER_EN_SHIFT, 1, 0), + SOC_ENUM("EnvTrack Headroom", max98927_env_track_headroom), }; static const struct snd_soc_dapm_route max98927_audio_map[] = { @@ -726,13 +741,10 @@ static int max98927_probe(struct snd_soc_codec *codec) /* Envelope Tracking configuration */ regmap_write(max98927->regmap, MAX98927_R0082_ENV_TRACK_VOUT_HEADROOM, - 0x08); + 0x0A); regmap_write(max98927->regmap, MAX98927_R0086_ENV_TRACK_CTRL, 0x01); - regmap_write(max98927->regmap, - MAX98927_R0087_ENV_TRACK_BOOST_VOUT_READ, - 0x10); /* voltage, current slot configuration */ regmap_write(max98927->regmap, diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index bf7a6f92..3069a09 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -250,6 +250,10 @@ #define MAX98927_BROWNOUT_DSP_EN (0x1 << 2) #define MAX98927_BROWNOUT_DSP_SHIFT (2) +/* MAX98927_R0086_ENV_TRACK_CTRL */ +#define MAX98927_ENV_TRACKER_EN (0x1 << 0) +#define MAX98927_ENV_TRACKER_EN_SHIFT (0) + /* MAX98927_R0100_SOFT_RESET */ #define MAX98927_SOFT_RESET (0x1 << 0) -- 2.7.4

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[alsa-devel] [PATCH 1/5] staging: max98927: Added TDM support
by Ryan Lee 19 May '17

19 May '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98927.c | 141 ++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/max98927.h | 6 +- 2 files changed, 120 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index b5ee294..99d6e41 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -1,7 +1,7 @@ /* * max98927.c -- MAX98927 ALSA Soc Audio driver * - * Copyright (C) 2016 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -146,6 +146,7 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); unsigned int mode = 0; unsigned int format = 0; + bool use_pdm = false; unsigned int invert = 0; dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); @@ -187,22 +188,27 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - max98927->iface |= SND_SOC_DAIFMT_I2S; format = MAX98927_PCM_FORMAT_I2S; break; case SND_SOC_DAIFMT_LEFT_J: - max98927->iface |= SND_SOC_DAIFMT_LEFT_J; format = MAX98927_PCM_FORMAT_LJ; break; + case SND_SOC_DAIFMT_DSP_A: + format = MAX98927_PCM_FORMAT_TDM_MODE1; + break; + case SND_SOC_DAIFMT_DSP_B: + format = MAX98927_PCM_FORMAT_TDM_MODE0; + break; case SND_SOC_DAIFMT_PDM: - max98927->iface |= SND_SOC_DAIFMT_PDM; + use_pdm = true; break; default: return -EINVAL; } + max98927->iface = fmt & SND_SOC_DAIFMT_FORMAT_MASK; - /* pcm channel configuration */ - if (max98927->iface & (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J)) { + if (!use_pdm) { + /* pcm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0018_PCM_RX_EN_A, MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, @@ -217,13 +223,12 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 0); - } else regmap_update_bits(max98927->regmap, - MAX98927_R0018_PCM_RX_EN_A, - MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + MAX98927_R0035_PDM_RX_CTRL, + MAX98927_PDM_RX_EN_MASK, 0); - /* pdm channel configuration */ - if (max98927->iface & SND_SOC_DAIFMT_PDM) { + } else { + /* pdm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0035_PDM_RX_CTRL, MAX98927_PDM_RX_EN_MASK, 1); @@ -231,10 +236,12 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) regmap_update_bits(max98927->regmap, MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 3); - } else + regmap_update_bits(max98927->regmap, - MAX98927_R0035_PDM_RX_CTRL, - MAX98927_PDM_RX_EN_MASK, 0); + MAX98927_R0018_PCM_RX_EN_A, + MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + + } return 0; } @@ -245,6 +252,21 @@ static const int rate_table[] = { 13000000, 19200000, }; +/* BCLKs per LRCLK */ +static const int bclk_sel_table[] = { + 32, 48, 64, 96, 128, 192, 256, 384, 512, +}; + +static int max98927_get_bclk_sel(int bclk) +{ + int i; + /* match BCLKs per LRCLK */ + for (i = 0; i < ARRAY_SIZE(bclk_sel_table); i++) { + if (bclk_sel_table[i] == bclk) + return i + 2; + } + return 0; +} static int max98927_set_clock(struct max98927_priv *max98927, struct snd_pcm_hw_params *params) { @@ -270,19 +292,17 @@ static int max98927_set_clock(struct max98927_priv *max98927, i << MAX98927_PCM_MASTER_MODE_MCLK_RATE_SHIFT); } - switch (blr_clk_ratio) { - case 32: - value = 2; - break; - case 48: - value = 3; - break; - case 64: - value = 4; - break; - default: + if ((max98927->iface == SND_SOC_DAIFMT_DSP_A) || + (max98927->iface == SND_SOC_DAIFMT_DSP_B)) + return 0; + + /* BCLK configuration */ + value = max98927_get_bclk_sel(blr_clk_ratio); + if (!value) { + dev_err(codec->dev, "BCLK %d not supported\n", blr_clk_ratio); return -EINVAL; } + regmap_update_bits(max98927->regmap, MAX98927_R0022_PCM_CLK_SETUP, MAX98927_PCM_CLK_SETUP_BSEL_MASK, @@ -386,6 +406,76 @@ static int max98927_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } +static int max98927_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + int bsel = 0; + unsigned int chan_sz = 0; + + /* BCLK configuration */ + bsel = max98927_get_bclk_sel(slots * slot_width); + if (bsel == 0) { + dev_err(codec->dev, "BCLK %d not supported\n", + slots * slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0022_PCM_CLK_SETUP, + MAX98927_PCM_CLK_SETUP_BSEL_MASK, + bsel); + + /* Channel size configuration */ + switch (slot_width) { + case 16: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(codec->dev, "format unsupported %d\n", + slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + /* Rx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R0018_PCM_RX_EN_A, + rx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R0019_PCM_RX_EN_B, + (rx_mask & 0xFF00) >> 8); + + /* Tx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R001A_PCM_TX_EN_A, + tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001B_PCM_TX_EN_B, + (tx_mask & 0xFF00) >> 8); + + /* Tx slot Hi-Z configuration */ + regmap_write(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + ~tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + (~tx_mask & 0xFF00) >> 8); + + return 0; +} + #define MAX98927_RATES SNDRV_PCM_RATE_8000_48000 #define MAX98927_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ @@ -405,6 +495,7 @@ static const struct snd_soc_dai_ops max98927_dai_ops = { .set_sysclk = max98927_dai_set_sysclk, .set_fmt = max98927_dai_set_fmt, .hw_params = max98927_dai_hw_params, + .set_tdm_slot = max98927_dai_tdm_slot, }; static int max98927_dac_event(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index ece6a60..bf7a6f92 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -1,7 +1,7 @@ /* * max98927.h -- MAX98927 ALSA Soc Audio driver * - * Copyright 2013-15 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -161,7 +161,9 @@ #define MAX98927_PCM_MODE_CFG_FORMAT_SHIFT (3) #define MAX98927_PCM_FORMAT_I2S (0x0 << 0) #define MAX98927_PCM_FORMAT_LJ (0x1 << 0) - +#define MAX98927_PCM_FORMAT_TDM_MODE0 (0x3 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE1 (0x4 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE2 (0x5 << 0) #define MAX98927_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) -- 2.7.4

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[alsa-devel] [bug report] ASoC: add audio-graph-card support
by Dan Carpenter 19 May '17

19 May '17

Hello Kuninori Morimoto, The patch 2692c1c63c29: "ASoC: add audio-graph-card support" from Apr 20, 2017, leads to the following static checker warning: sound/soc/generic/audio-graph-card.c:275 asoc_graph_card_probe() info: return a literal instead of 'ret' sound/soc/generic/audio-graph-card.c 268 goto err; 269 } 270 271 snd_soc_card_set_drvdata(card, priv); 272 273 ret = devm_snd_soc_register_card(dev, card); 274 if (ret >= 0) 275 return ret; This code is confusing. I'm pretty sure that devm_snd_soc_register_card() returns zero on success or negative error codes but why are we checking >= 0 here? It's better to do error handling than success handling. People often get a bit creative with the last condition in a function... :/ In other words, it's cleaner like this: ret = devm_snd_soc_register_card(dev, card); if (ret) goto err; return 0; err: asoc_simple_card_clean_reference(card); return ret; 276 err: 277 asoc_simple_card_clean_reference(card); 278 279 return ret; 280 } regards, dan carpenter

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Re: [alsa-devel] WM5102 - Help to make baytrail machine driver work
by Paulo Sergio 18 May '17

18 May '17

Adding alsa list + intel devs (Sorry for that ) Inline answers for Charles Keepax questions bellow: 2017-05-08 5:24 GMT-03:00 Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com>: > On Sun, May 07, 2017 at 10:42:31PM -0300, Paulo Sergio wrote: >> Hi, >> >> >> I'm trying to make wm5102 work on Lenovo Yoga 2 1051F (Baytrail) again >> and would like some tips. >> >> With a revised machine driver code [1] ( based on bytcr_rt5640 and >> some ports from Lenovo kernel [2]) I can initialize >> and setting routes with alsa mixer [3], I still don't have sound. >> >> However now if I connect 'HPOUT1' to 'Tone Generator 1' I can hear the 1k tone. >> Question: When I do this association, it is using my route? (in my >> case CODEC OUT -> SSP0 TX -> AIF1RX) >> Or this internal tone generator has a direct link to HPOUTs? >> >> Here's some lines from my dmesg output while attemptig to play >> something (complete dmesg - see [4]): >> >> [ 211.801176] arizona spi-WM510204:00: FLL1: Fref=25000000 Fout=49152000 >> [ 211.801184] arizona spi-WM510204:00: FLL1: Fvco=98304000Hz >> [ 211.801190] arizona spi-WM510204:00: FLL1: GCD=4000 >> [ 211.801195] arizona spi-WM510204:00: FLL1: N=7 THETA=a8d LAMBDA=c35 >> [ 211.801200] arizona spi-WM510204:00: FLL1: FRATIO=0(0) OUTDIV=2 REFCLK_DIV=1 >> [ 211.801204] arizona spi-WM510204:00: FLL1: GAIN=4 >> [ 211.801210] arizona spi-WM510204:00: 173 <= a8d >> [ 211.801227] arizona spi-WM510204:00: 174 <= c35 >> [ 211.801234] arizona spi-WM510204:00: 176 <= 40 >> [ 211.801241] arizona spi-WM510204:00: 179 <= 10 >> [ 211.813301] arizona spi-WM510204:00: 172 <= 8007 >> [ 211.813353] arizona spi-WM510204:00: 171 <= 1 >> [ 211.813366] arizona spi-WM510204:00: FLL1: Waiting for FLL lock... >> [ 211.814098] arizona spi-WM510204:00: d23 => 100 >> [ 211.814641] arizona spi-WM510204:00: d23 => 101 >> [ 211.814651] arizona spi-WM510204:00: FLL1: FLL locked (1 polls) >> [ 211.814660] arizona spi-WM510204:00: SYSCLK set to 49152000Hz >> [ 211.814668] arizona spi-WM510204:00: SYSCLK set to 49152000Hz >> [ 211.814675] arizona spi-WM510204:00: 101 <= 300 >> [ 211.814882] wm5102-codec wm5102-codec: AIF1: BCLK 1536000Hz LRCLK 48000Hz >> [ 211.814892] arizona spi-WM510204:00: 80 <= 3 >> [ 211.815191] arizona spi-WM510204:00: 4dc <= 0 >> [ 211.815339] arizona spi-WM510204:00: 4dd <= 0 >> [ 211.815460] arizona spi-WM510204:00: 80 <= 0 >> [ 211.815662] sst-mfld-platform sst-mfld-platform: Enter: enable=1 >> port_name=ssp0-port >> [ 211.816833] wm5102-codec wm5102-codec: AIF1: BCLK 1536000Hz LRCLK 48000Hz >> [ 211.816845] arizona spi-WM510204:00: 80 <= 3 >> [ 211.817030] arizona spi-WM510204:00: 4dc <= 0 >> [ 211.817287] arizona spi-WM510204:00: 4dd <= 0 >> >> Thanks in advance! >> >> >> [1] - https://github.com/pstglia/linux/tree/lenovo_yoga2_returns >> [2] - https://github.com/lenovo-yt2-dev/android_kernel_lenovo_baytrail/blob/cm-12… >> [3] - https://drive.google.com/file/d/0BxO6THtB865fQnQyZkx6c05Nbjg/view?usp=shari… >> [4] - https://drive.google.com/file/d/0BxO6THtB865fRy0xV3F0elRrUHM/view?usp=shari… > > OK good news you are getting sound from the Tone generator, that > likely implies things are setup sensibly on the CODEC side. My > suspicion would be that data is not making it over the I2S. > Probalby worth copying the Intel guys on the email, as I don't > know know much about that side of things. Copied alsa list and Intel guys (hit reply instead of reply all - my apologies) > > How are you setting up the I2S? As in which side is master, what > format are you using? Also do you see any over/underflow error, > does the system complain with "Error playing sample". The only part where I2S is explicity used on machine driver is this setting on backend DAI (SSP0-Codec) dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, And after setting the routes using routes mentioned on [3], I have no errors, but no sound Note: Based on Lenovo code, I concluded this device uses ssp0, so I'm using a fw file "fw_sst_0f28_ssp0.bin" posted another thread. Could use the hacks used on kernel 4.11, but as I'm still testing with kernel 4.4, using this fw would make things easier x86_64:/data # alsa_aplay -v TNT.mp3 Playing raw data 'TNT.mp3' : Unsigned 8 bit, Rate 8000 Hz, Mono Plug PCM: Rate conversion PCM (48000, sformat=U8) Converter: linear-interpolation Protocol version: 10002 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 8 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 262144000 Slave: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 48000 exact rate : 48000 (48000/1) msbits : 8 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 24000 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 1572864000 Slave: Hardware PCM card 0 'baytrailcraudio' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 24000 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 1572864000 appl_ptr : 0 hw_ptr : 0 Regards, Pstglia

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[alsa-devel] [PATCH 1/2] ALSA: pcm: use "do {} while (0)" for empty macro
by Arnd Bergmann 18 May '17

18 May '17

Recent compilers produce a harmless warning for the new pcm_call_notify() macro when CONFIG_SND_PCM_OSS is disabled: sound/core/pcm.c: In function 'snd_pcm_free': sound/core/pcm.c:905:37: error: suggest braces around empty body in an 'if' statement [-Werror=empty-body] This turns the empty macro into a 'do {} while (0)' statement to avoid the warning. Fixes: 58f30d650c7f ("ALSA: pcm: Build pcm notifier code conditionally") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/core/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 24acbfb5c531..d30dba0ee688 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -894,7 +894,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) _notify->call(pcm); \ } while (0) #else -#define pcm_call_notify(pcm, call) /* NOP */ +#define pcm_call_notify(pcm, call) do {} while (0) #endif static int snd_pcm_free(struct snd_pcm *pcm) -- 2.9.0

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[alsa-devel] Question about tsched=0 on PulseAudio
by Kuninori Morimoto 18 May '17

18 May '17

Hi ALSA ML We noticed that our PulseAudio has crackling / skipping noise which doesn't happen on normal aplay. And it was removed if we do below load-module module-hal-detect tsched=0 According to google, it say tsched Since 0.9.11. Use system-timer based model (aka glitch-free). Defaults to 1 (enabled). If your hardware does not return accurate timing information (e.g. Creative sound cards) you can try to set tsched=0 to enable the interupt based timing which was used in 0.9.10 and before. Here, what does this "return accurate timing information" mean ? Is it struct snd_pcm_ops::pointer ?? Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ALSA: mixart: constify snd_kcontrol_new structures
by Bhumika Goyal 17 May '17

17 May '17

Declare snd_kcontrol_new structures as const as they are only passed an argument to the function snd_ctl_new1. This argument is of type const, so snd_kcontrol_new structures having this property can be made const. Done using Coccinelle: @r disable optional_qualifier@ identifier x; position p; @@ static struct snd_kcontrol_new x@p={...}; @ok@ identifier r.x; position p; @@ snd_ctl_new1(&x@p,...) @bad@ position p != {r.p,ok.p}; identifier r.x; @@ x@p @depends on !bad disable optional_qualifier@ identifier r.x; @@ +const struct snd_kcontrol_new x; Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/pci/mixart/mixart_mixer.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/mixart/mixart_mixer.c b/sound/pci/mixart/mixart_mixer.c index 4a4616a..2b9496a 100644 --- a/sound/pci/mixart/mixart_mixer.c +++ b/sound/pci/mixart/mixart_mixer.c @@ -404,7 +404,7 @@ static int mixart_analog_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e static const DECLARE_TLV_DB_SCALE(db_scale_analog, -9600, 50, 0); -static struct snd_kcontrol_new mixart_control_analog_level = { +static const struct snd_kcontrol_new mixart_control_analog_level = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ), @@ -897,7 +897,7 @@ static int mixart_pcm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem static const DECLARE_TLV_DB_SCALE(db_scale_digital, -10950, 50, 0); -static struct snd_kcontrol_new snd_mixart_pcm_vol = +static const struct snd_kcontrol_new snd_mixart_pcm_vol = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | @@ -951,7 +951,7 @@ static int mixart_pcm_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ return changed; } -static struct snd_kcontrol_new mixart_control_pcm_switch = { +static const struct snd_kcontrol_new mixart_control_pcm_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* name will be filled later */ .count = MIXART_PLAYBACK_STREAMS, -- 2.7.4

1 0

[alsa-devel] [PATCH] ALSA: cs46xx: constify snd_kcontrol_new structures
by Bhumika Goyal 17 May '17

17 May '17

Declare snd_kcontrol_new structures as const as they are only passed an argument to the function snd_ctl_new1. This argument is of type const, so snd_kcontrol_new structures having this property can be made const. Done using Coccinelle: @r disable optional_qualifier@ identifier x; position p; @@ static struct snd_kcontrol_new x@p={...}; @ok@ identifier r.x; position p; @@ snd_ctl_new1(&x@p,...) @bad@ position p != {r.p,ok.p}; identifier r.x; @@ x@p @depends on !bad disable optional_qualifier@ identifier r.x; @@ +const struct snd_kcontrol_new x; Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/pci/cs46xx/cs46xx_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index e4cf318..00fa52e 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2371,7 +2371,7 @@ static int snd_cs46xx_front_dup_put(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] ? 0 : 0x200); } -static struct snd_kcontrol_new snd_cs46xx_front_dup_ctl = { +static const struct snd_kcontrol_new snd_cs46xx_front_dup_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Duplicate Front", .info = snd_mixer_boolean_info, -- 2.7.4

1 0

[alsa-devel] [PATCH] ALSA: ice1712: constify snd_kcontrol_new structures
by Bhumika Goyal 17 May '17

17 May '17

Declare snd_kcontrol_new structures as const as they are only passed an argument to the function snd_ctl_new1. This argument is of type const, so snd_kcontrol_new structures having this property can be made const. Done using Coccinelle: @r disable optional_qualifier@ identifier x; position p; @@ static struct snd_kcontrol_new x@p={...}; @ok@ identifier r.x; position p; @@ snd_ctl_new1(&x@p,...) @bad@ position p != {r.p,ok.p}; identifier r.x; @@ x@p @depends on !bad disable optional_qualifier@ identifier r.x; @@ +const struct snd_kcontrol_new x; Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/pci/ice1712/ice1712.c | 6 +++--- sound/pci/ice1712/ice1724.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 1d8612c..6b650ab 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -1401,7 +1401,7 @@ static struct snd_kcontrol_new snd_ice1712_multi_playback_ctrls[] = { }, }; -static struct snd_kcontrol_new snd_ice1712_multi_capture_analog_switch = { +static const struct snd_kcontrol_new snd_ice1712_multi_capture_analog_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "H/W Multi Capture Switch", .info = snd_ice1712_pro_mixer_switch_info, @@ -1420,7 +1420,7 @@ static const struct snd_kcontrol_new snd_ice1712_multi_capture_spdif_switch = { .count = 2, }; -static struct snd_kcontrol_new snd_ice1712_multi_capture_analog_volume = { +static const struct snd_kcontrol_new snd_ice1712_multi_capture_analog_volume = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ), @@ -2165,7 +2165,7 @@ static int snd_ice1712_pro_route_spdif_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new snd_ice1712_mixer_pro_analog_route = { +static const struct snd_kcontrol_new snd_ice1712_mixer_pro_analog_route = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "H/W Playback Route", .info = snd_ice1712_pro_route_info, diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 9cd6e55..7851881 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2142,7 +2142,7 @@ static int snd_vt1724_pro_route_spdif_put(struct snd_kcontrol *kcontrol, digital_route_shift(idx)); } -static struct snd_kcontrol_new snd_vt1724_mixer_pro_analog_route = +static const struct snd_kcontrol_new snd_vt1724_mixer_pro_analog_route = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "H/W Playback Route", -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: rt5665: Fix uninitialized warning in rt5665_i2s_pin_event()
by Geert Uytterhoeven 17 May '17

17 May '17

With gcc 4.1.2: sound/soc/codecs/rt5665.c: In function ‘rt5665_i2s_pin_event’: sound/soc/codecs/rt5665.c:2610: warning: ‘mask1’ may be used uninitialized in this function sound/soc/codecs/rt5665.c:2610: warning: ‘val2’ may be used uninitialized in this function sound/soc/codecs/rt5665.c:2610: warning: ‘val1’ may be used uninitialized in this function The first one is currently a false positive, as rt5665_i2s_pin_event() is never called with snd_soc_dapm_widget.shift set to a value not handled by the switch() statement. But that may change, so preinitialize mask1 to fix this, like is already done for mask2. The last two are false-positives, the compiler is just not smart enough to notice the mask and val variables are always used together. Fixes: 9b5d3865b3b410d2 ("ASoC: rt5665: set i2s pin share configuration") Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- sound/soc/codecs/rt5665.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 8cd22307f5b6e6ab..5e42f4ee51ba5b20 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -2607,7 +2607,7 @@ static int rt5665_i2s_pin_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - unsigned int val1, val2, mask1, mask2 = 0; + unsigned int val1, val2, mask1 = 0, mask2 = 0; switch (w->shift) { case RT5665_PWR_I2S2_1_BIT: @@ -2635,13 +2635,17 @@ static int rt5665_i2s_pin_event(struct snd_soc_dapm_widget *w, } switch (event) { case SND_SOC_DAPM_PRE_PMU: - snd_soc_update_bits(codec, RT5665_GPIO_CTRL_1, mask1, val1); + if (mask1) + snd_soc_update_bits(codec, RT5665_GPIO_CTRL_1, + mask1, val1); if (mask2) snd_soc_update_bits(codec, RT5665_GPIO_CTRL_2, mask2, val2); break; case SND_SOC_DAPM_POST_PMD: - snd_soc_update_bits(codec, RT5665_GPIO_CTRL_1, mask1, 0); + if (mask1) + snd_soc_update_bits(codec, RT5665_GPIO_CTRL_1, + mask1, 0); if (mask2) snd_soc_update_bits(codec, RT5665_GPIO_CTRL_2, mask2, 0); -- 2.7.4

2 1

[alsa-devel] [PATCH] base: export symbol of_phandle_iterator_init/next
by Kuninori Morimoto 17 May '17

17 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> of_for_each_phandle() uses of_phandle_iterator_init/next but these aren't exported. So kernel module complile will say ERROR: "of_phandle_iterator_init" [xxx.ko] undefined! ERROR: "of_phandle_iterator_next" [xxx.ko] undefined! This patch solves this issue Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This patch is related to this patch-set Subject: [PATCH v5 0/9] ASoC: add OF-graph base simple-card Date: Tue, 21 Mar 2017 14:17:03 +0900 drivers/of/base.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/of/base.c b/drivers/of/base.c index 812edb9..bc42f91 100644 --- a/drivers/of/base.c +++ b/drivers/of/base.c @@ -1570,6 +1570,7 @@ int of_phandle_iterator_init(struct of_phandle_iterator *it, return 0; } +EXPORT_SYMBOL_GPL(of_phandle_iterator_init); int of_phandle_iterator_next(struct of_phandle_iterator *it) { @@ -1639,6 +1640,7 @@ int of_phandle_iterator_next(struct of_phandle_iterator *it) return -EINVAL; } +EXPORT_SYMBOL_GPL(of_phandle_iterator_next); int of_phandle_iterator_args(struct of_phandle_iterator *it, uint32_t *args, -- 1.9.1

2 1

[alsa-devel] [PATCH 0/7] of_graph: prepare for ALSA graph support
by Kuninori Morimoto 17 May '17

17 May '17

Hi Rob, Mark Cc Guennadi, Laurent Now OF graph is mainly used by V4L2 SoC, and ALSA SoC is using different style for SoC <-> Codec binding. But, for example, HDMI case, V4L2 <-> ALSA need to collaborate, and then ALSA SoC needs to adjust to OF graph. OTOH, V4L2's "OF graph" position is same as ALSA SoC "sound card" position. And ALSA SoC side want to keep existing supported feature on new OF graph style. I'm posting this on ALSA SoC ML now. Now, current of_graph is indicating port/endpoint, but there is no way to understand that it is for video port ? or sound port ? or other device port ? For example, HDMI has video port, and sound port. Because of this reason, ALSA SoC side can't handle OF graph correctly. Thus, this patch-set tries to add new "type" on OF graph. And this patch-set includes small feature which are useful for ALSA SoC side OF graph support. Kuninori Morimoto (7): Documentation: of: add type property of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_port/endpoint_type_is() of_graph: add of_graph_get_endpoint_count() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_top_port() of_graph: add for_each_of_port() / for_each_of_endpoint_in_port() Documentation/devicetree/bindings/graph.txt | 26 ++++ drivers/of/base.c | 196 ++++++++++++++++++++++++++-- include/linux/of_graph.h | 67 ++++++++++ 3 files changed, 279 insertions(+), 10 deletions(-) -- 1.9.1

3 12

[alsa-devel] [PATCH v4 0/9] ASoC: add OF-graph base simple-card
by Kuninori Morimoto 17 May '17

17 May '17

Hi Rob, Mark These are v4 of OF-graph base audio card patch-set. Main difference from v3 is that it doesn't have prefix on DT, and have new "label", "dai-format". 1) - 3) : new OF-graph helper functions 4) - 5) : expand "label", "dai-format" 6) - 9) : OF-graph base simple-card Kuninori Morimoto (9): 1) of_graph: add of_graph_get_remote_endpoint() 2) of_graph: add of_graph_get_port_parent() 3) of_graph: add of_graph_get_endpoint_count() 4) ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name 5) ASoC: soc-core: enable "dai-name" on snd_soc_of_parse_daifmt() 6) ASoC: add snd_soc_get_dai_id() 7) ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() 8) ASoC: add audio-graph-card document 9) ASoC: add audio-graph-card support .../devicetree/bindings/sound/audio-graph-card.txt | 124 +++++++++ .../devicetree/bindings/sound/simple-card.txt | 3 +- .../devicetree/bindings/sound/simple-scu-card.txt | 4 +- drivers/of/base.c | 60 +++- include/linux/of_graph.h | 21 ++ include/sound/simple_card_utils.h | 10 + include/sound/soc.h | 1 + sound/soc/generic/Kconfig | 8 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-card.c | 301 +++++++++++++++++++++ sound/soc/generic/simple-card-utils.c | 54 +++- sound/soc/soc-core.c | 32 ++- 12 files changed, 602 insertions(+), 18 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 sound/soc/generic/audio-graph-card.c -- 1.9.1

4 25

[alsa-devel] [PATCH v6 0/9] ASoC: add OF-graph base simple-card
by Kuninori Morimoto 17 May '17

17 May '17

Hi Mark, Rob These are v6 of OF-graph sound card support. Main change is asoc_simple_card_get_dai_id() becames simple-card-utils's static function. 1) - 4) : OF-graph base functions 5) - 9) : OF-graph + simple-card (= audio_graph_card) Kuninori Morimoto (9): 1) of-graph: export symbol of_phandle_iterator_init/next 2) of_graph: add of_graph_get_remote_endpoint() 3) of_graph: add of_graph_get_port_parent() 4) of_graph: add of_graph_get_endpoint_count() 5) ASoC: soc-core: enable "dai-format" on snd_soc_of_parse_daifmt() 6) ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name 7) ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() 8) ASoC: add audio-graph-card document 9) ASoC: add audio-graph-card support .../devicetree/bindings/sound/audio-graph-card.txt | 124 +++++++++ drivers/of/base.c | 62 ++++- include/linux/of_graph.h | 21 ++ include/sound/simple_card_utils.h | 10 + sound/soc/generic/Kconfig | 8 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-card.c | 308 +++++++++++++++++++++ sound/soc/generic/simple-card-utils.c | 75 ++++- sound/soc/soc-core.c | 10 +- 9 files changed, 603 insertions(+), 17 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 sound/soc/generic/audio-graph-card.c -- 1.9.1

3 13

[alsa-devel] [PATCH v7 0/9] ASoC: add OF-graph base simple-card
by Kuninori Morimoto 17 May '17

17 May '17

Hi Mark, Rob These are v7 of OF-graph sound card support. Main change from v6 "label" doesn't allow [prefix]label (in [6/9]) 1) - 4) : OF-graph base functions 5) - 9) : OF-graph + simple-card (= audio_graph_card) Kuninori Morimoto (9): of-graph: export symbol of_phandle_iterator_init/next of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_endpoint_count() ASoC: soc-core: enable "dai-format" on snd_soc_of_parse_daifmt() ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() ASoC: add audio-graph-card document ASoC: add audio-graph-card support .../devicetree/bindings/sound/audio-graph-card.txt | 124 +++++++++ drivers/of/base.c | 62 ++++- include/linux/of_graph.h | 21 ++ include/sound/simple_card_utils.h | 10 + sound/soc/generic/Kconfig | 8 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-card.c | 308 +++++++++++++++++++++ sound/soc/generic/simple-card-utils.c | 73 ++++- sound/soc/soc-core.c | 10 +- 9 files changed, 600 insertions(+), 18 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 sound/soc/generic/audio-graph-card.c -- 1.9.1

3 16

[alsa-devel] [PATCH v5 0/9] ASoC: add OF-graph base simple-card
by Kuninori Morimoto 17 May '17

17 May '17

Hi Rob, Mark These are v5 of OF-graph base audio card patch-set. It fixed small things which was pointed by Rob. 1) - 3) : new OF-graph helper functions 4) - 5) : expand "label", "dai-format" 6) - 9) : OF-graph base simple-card Kuninori Morimoto (9): of_graph: add of_graph_get_remote_endpoint() of_graph: add of_graph_get_port_parent() of_graph: add of_graph_get_endpoint_count() ASoC: simple-card-utils: enable "label" on asoc_simple_card_parse_card_name ASoC: soc-core: enable "dai-name" on snd_soc_of_parse_daifmt() ASoC: add snd_soc_get_dai_id() ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() ASoC: add audio-graph-card document ASoC: add audio-graph-card support .../devicetree/bindings/sound/audio-graph-card.txt | 124 +++++++++ drivers/of/base.c | 60 +++- include/linux/of_graph.h | 21 ++ include/sound/simple_card_utils.h | 10 + include/sound/soc.h | 1 + sound/soc/generic/Kconfig | 8 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-card.c | 308 +++++++++++++++++++++ sound/soc/generic/simple-card-utils.c | 54 +++- sound/soc/soc-core.c | 32 ++- 10 files changed, 604 insertions(+), 16 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 sound/soc/generic/audio-graph-card.c -- 1.9.1

2 12

[alsa-devel] [PATCH 0/2] ASoC: rsnd: tidyup Document
by Kuninori Morimoto 17 May '17

17 May '17

Hi Mark Cc Geert These patches tidyup Document for Renesas sound driver Kuninori Morimoto (2): ASoC: rsnd: move "renesas,rsrc-card" to "simple-scu-audio-card" on Document ASoC: rsnd: add missing clocks/clock-names on Document .../devicetree/bindings/sound/renesas,rsnd.txt | 34 ++++++++++++---------- 1 file changed, 19 insertions(+), 15 deletions(-) -- 1.9.1

3 6

[alsa-devel] [PATCH] ASoC: hdmi-codec: fix spelling mistake: "deteced" -> "detected"
by Colin King 17 May '17

17 May '17

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/hdmi-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 8c5ae1fc23a9..a3f15149afcf 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -819,7 +819,7 @@ static int hdmi_codec_probe(struct platform_device *pdev) mutex_unlock(&hdmi_mutex); if (hd->cnt >= ARRAY_SIZE(hdmi_dai_name)) { - dev_err(dev, "too many hdmi codec are deteced\n"); + dev_err(dev, "too many hdmi codec are detected\n"); return -EINVAL; } -- 2.11.0

3 2

[alsa-devel] [PATCH] ASoC: rsnd: remove un-used OUF_SRCO/I macro
by Kuninori Morimoto 17 May '17

17 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/src.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 20b5b2e..a8ab26d 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -12,10 +12,6 @@ #define SRC_NAME "src" -/* SRCx_STATUS */ -#define OUF_SRCO ((1 << 12) | (1 << 13)) -#define OUF_SRCI ((1 << 9) | (1 << 8)) - /* SCU_SYSTEM_STATUS0/1 */ #define OUF_SRC(id) ((1 << (id + 16)) | (1 << id)) -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: remove mismatch explain comment from src.c
by Kuninori Morimoto 17 May '17

17 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Before, src.c cares SRC and SSIU, but now SSIU is cared by ssiu.c by commit c7f69ab5364d ("ASoC: rsnd: use mod base common method on SSIU"). This patch removes mismatched explanation for SRC/SSIU from src.c Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/src.c | 14 -------------- 1 file changed, 14 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index a8ab26d..e4bde0c 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -51,20 +51,6 @@ struct rsnd_src { * */ -/* - * src.c is caring... - * - * Gen1 - * - * [mem] -> [SRU] -> [SSI] - * |--------| - * - * Gen2 - * - * [mem] -> [SRC] -> [SSIU] -> [SSI] - * |-----------------| - */ - static void rsnd_src_activation(struct rsnd_mod *mod) { rsnd_mod_write(mod, SRC_SWRSR, 0); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: clkout-lr-asynchronous is AUDIO_CLKOUT related property
by Kuninori Morimoto 17 May '17

17 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> clkout-lr-asynchronous is AUDIO_CLKOUT related property, thus, it should be handled by rsnd_adg_get_clkout(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 66203d1..cb4eab0 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -480,6 +480,9 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, if (req_rate[0] % 48000 == 0) adg->flags = AUDIO_OUT_48; + if (of_get_property(np, "clkout-lr-asynchronous", NULL)) + adg->flags = LRCLK_ASYNC; + /* * This driver is assuming that AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC * have 44.1kHz or 48kHz base clocks for now. @@ -578,7 +581,6 @@ int rsnd_adg_probe(struct rsnd_priv *priv) { struct rsnd_adg *adg; struct device *dev = rsnd_priv_to_dev(priv); - struct device_node *np = dev->of_node; int ret; adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL); @@ -595,9 +597,6 @@ int rsnd_adg_probe(struct rsnd_priv *priv) rsnd_adg_get_clkin(priv, adg); rsnd_adg_get_clkout(priv, adg); - if (of_get_property(np, "clkout-lr-asynchronous", NULL)) - adg->flags = LRCLK_ASYNC; - priv->adg = adg; rsnd_adg_clk_enable(priv); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: remove error pointer for adg->clkout[i]
by Kuninori Morimoto 17 May '17

17 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit d7f298197a22f11b38059f257842dac7c30a564c ("ASoC: rcar: fixup of_clk_add_provider() usage for multi clkout") added adg->clkout[i] = ERR_PTR(-ENOENT), but, adg->clkout[i] are assumed NULL or clk pointer. This patch fix it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index cb4eab0..36bece8 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -556,7 +556,6 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, clk = clk_register_fixed_rate(dev, clkout_name[i], parent_clk_name, 0, req_rate[0]); - adg->clkout[i] = ERR_PTR(-ENOENT); if (!IS_ERR(clk)) adg->clkout[i] = clk; } -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rsnd: need to call nolock_stop if nolock_start was failed
by Kuninori Morimoto 17 May '17

17 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd_dai_call() is counting the called number of start/stop functions. nolock_stop should be called if nolock_start was failed. Otherwise, nolock_start doesn't called in 2nd play. This patch solved this issue. 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 | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 8c1f4e2..e16f723 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -766,11 +766,16 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, { struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); + int ret; /* * call rsnd_dai_call without spinlock */ - return rsnd_dai_call(nolock_start, io, priv); + ret = rsnd_dai_call(nolock_start, io, priv); + if (ret < 0) + rsnd_dai_call(nolock_stop, io, priv); + + return ret; } static void rsnd_soc_dai_shutdown(struct snd_pcm_substream *substream, -- 1.9.1

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Re: [alsa-devel] ASoC: rsnd: fix sound route path when using SRC6/SRC9
by Geert Uytterhoeven 17 May '17

17 May '17

Hi Morimoto-san, On Fri, Mar 31, 2017 at 9:32 PM, Linux Kernel Mailing List <linux-kernel(a)vger.kernel.org> wrote: > Web: https://git.kernel.org/torvalds/c/a1c2ff53726907aff5feb37e4cfd45c1ff626431 > Commit: a1c2ff53726907aff5feb37e4cfd45c1ff626431 > Parent: 4b30eebfc35c67771b5f58d9274d3e321b72d7a8 > Refname: refs/heads/master > Author: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> > AuthorDate: Wed Mar 1 03:51:00 2017 +0000 > Committer: Mark Brown <broonie(a)kernel.org> > CommitDate: Mon Mar 13 12:58:07 2017 +0000 > > ASoC: rsnd: fix sound route path when using SRC6/SRC9 > > This patch fixes the problem that the missing value of the route path > setting table and incorrect values are set in the CMD_ROUTE_SELECT > register. > > Signed-off-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> > [Kuninori: shared data on MIX and non-MIX case] > Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > Signed-off-by: Mark Brown <broonie(a)kernel.org> > --- > sound/soc/sh/rcar/cmd.c | 36 ++++++++++++++++++++---------------- > 1 file changed, 20 insertions(+), 16 deletions(-) > > diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c > index abb5eaac854a..7d92a24b7cfa 100644 > --- a/sound/soc/sh/rcar/cmd.c > +++ b/sound/soc/sh/rcar/cmd.c > @@ -31,23 +31,24 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, > struct rsnd_mod *mix = rsnd_io_to_mod_mix(io); > struct device *dev = rsnd_priv_to_dev(priv); > u32 data; > + u32 path[] = { > + [1] = 1 << 0, > + [5] = 1 << 8, > + [6] = 1 << 12, > + [9] = 1 << 15, > + }; > > if (!mix && !dvc) > return 0; > > + if (ARRAY_SIZE(path) < rsnd_mod_id(mod) + 1) > + return -ENXIO; > + > if (mix) { > struct rsnd_dai *rdai; > struct rsnd_mod *src; > struct rsnd_dai_stream *tio; > int i; > - u32 path[] = { > - [0] = 0, > - [1] = 1 << 0, > - [2] = 0, > - [3] = 0, > - [4] = 0, > - [5] = 1 << 8 > - }; > > /* > * it is assuming that integrater is well understanding about > @@ -70,16 +71,19 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, > } else { > struct rsnd_mod *src = rsnd_io_to_mod_src(io); > > - u32 path[] = { > - [0] = 0x30000, > - [1] = 0x30001, > - [2] = 0x40000, > - [3] = 0x10000, > - [4] = 0x20000, > - [5] = 0x40100 > + u8 cmd_case[] = { > + [0] = 0x3, > + [1] = 0x3, > + [2] = 0x4, > + [3] = 0x1, > + [4] = 0x2, > + [5] = 0x4, > + [6] = 0x1, > + [9] = 0x2, > }; > > - data = path[rsnd_mod_id(src)]; > + data = path[rsnd_mod_id(src)] | With gcc 4.9.0: sound/soc/sh/rcar/cmd.c: In function 'rsnd_cmd_init': sound/soc/sh/rcar/cmd.c:85:14: warning: array subscript is below array bounds [-Warray-bounds] data = path[rsnd_mod_id(src)] | > + cmd_case[rsnd_mod_id(src)] << 16; > } > > dev_dbg(dev, "ctu/mix path = 0x%08x", data); Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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