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

[alsa-devel] [PATCH] Revert "ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members"
by Takashi Sakamoto 26 Sep '17

26 Sep '17

This reverts commit 275353bb684e to fix a regression which can abort 'alsactl' program in alsa-utils due to assertion in alsa-lib. alsactl: control.c:2513: snd_ctl_elem_value_get_integer: Assertion `idx < sizeof(obj->value.integer.value) / sizeof(obj->value.integer.value[0])' failed. alsactl: control.c:2976: snd_ctl_elem_value_get_integer: Assertion `idx < ARRAY_SIZE(obj->value.integer.value)' failed. This commit is a band-aid. In a point of usage of ALSA control interface, the drivers still bring an issue that they prevent userspace applications to have a consistent way to parse each levels of the dimension information via ALSA control interface. Let me investigate this issue. Current implementation of the drivers have three control element sets with dimension information: * 'Monitor Mixer Volume' (type: integer) * 'VMixer Volume' (type: integer) * 'VU-meters' (type: boolean) Although the number of elements named as 'Monitor Mixer Volume' differs depending on drivers in this group, it can be calculated by macros defined by each driver (= (BX_NUM - BX_ANALOG_IN) * BX_ANALOG_IN). Each of the elements has one member for value and has dimension information with 2 levels (= BX_ANALOG_IN * (BX_NUM - BX_ANALOG_IN)). For these elements, userspace applications are expected to handle the dimension information so that all of the elements construct a matrix where the number of rows and columns are represented by the dimension information. The same way is applied to elements named as 'VMixer Volume'. The number of these elements can also be calculated by macros defined by each drivers (= PX_ANALOG_IN * BX_ANALOG_IN). Each of the element has one member for value and has dimension information with 2 levels (= BX_ANALOG_IN * PX_ANALOG_IN). All of the elements construct a matrix with the dimension information. An element named as 'VU-meters' gets a different way in a point of dimension information. The element includes 96 members for value. The element has dimension information with 3 levels (= 3 or 2 * 16 * 2). For this element, userspace applications are expected to handle the dimension information so that all of the members for value construct a matrix where the number of rows and columns are represented by the dimension information. This is different from the way for the former. As a summary, the drivers were not designed to produce a consistent way to parse the dimension information. This makes it hard for general userspace applications such as amixer to parse the information by a consistent way, and actually no userspace applications except for 'echomixer' utilize the dimension information. Additionally, no drivers excluding this group use the information. The reverted commit was written based on the latter way. A commit 860c1994a70a ('ALSA: control: add dimension validator for userspace elements') is written based on the latter way, too. The patch should be reconsider too in the same time to re-define a consistent way to parse the dimension information. Reported-by: Mark Hills <mark(a)xwax.org> Reported-by: S. Christian Collins <s.chriscollins(a)gmail.com> Fixes: 275353bb684e ('ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members') Cc: <stable(a)vger.kernel.org> # v4.8+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/pci/echoaudio/echoaudio.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 7326695bca33..d68f99e076a8 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1272,11 +1272,11 @@ static int snd_echo_mixer_info(struct snd_kcontrol *kcontrol, chip = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; uinfo->value.integer.min = ECHOGAIN_MINOUT; uinfo->value.integer.max = ECHOGAIN_MAXOUT; uinfo->dimen.d[0] = num_busses_out(chip); uinfo->dimen.d[1] = num_busses_in(chip); - uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1]; return 0; } @@ -1344,11 +1344,11 @@ static int snd_echo_vmixer_info(struct snd_kcontrol *kcontrol, chip = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; uinfo->value.integer.min = ECHOGAIN_MINOUT; uinfo->value.integer.max = ECHOGAIN_MAXOUT; uinfo->dimen.d[0] = num_busses_out(chip); uinfo->dimen.d[1] = num_pipes_out(chip); - uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1]; return 0; } @@ -1728,6 +1728,7 @@ static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 96; uinfo->value.integer.min = ECHOGAIN_MINOUT; uinfo->value.integer.max = 0; #ifdef ECHOCARD_HAS_VMIXER @@ -1737,7 +1738,6 @@ static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol, #endif uinfo->dimen.d[1] = 16; /* 16 channels */ uinfo->dimen.d[2] = 2; /* 0=level, 1=peak */ - uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1] * uinfo->dimen.d[2]; return 0; } -- 2.11.0

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[alsa-devel] [Resend] [PATCH v1 3/4] ASoC: dapm: Avoid creating kcontrol for params
by yesanishhere＠gmail.com 25 Sep '17

25 Sep '17

From: anish kumar <yesanishhere(a)gmail.com> Currently in codec to codec dai link if there are multiple params defined then dapm can use created kcontrol to decide which param to apply at runtime. However, in case there is only single param configuration then there is no point in creating the kcontrol and also there is no point in allocating memory for kcontrol. In the snd_soc_dapm_new_pcm function, there is memory allocation happening for kcontrol which is later used or not used based on num_param. It is better to not allocate memory when there is only a single configuration. This change is to remedy that anomaly. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- Changes in v1: - Included the review comments from charles regarding releasing memory in error path. sound/soc/soc-dapm.c | 130 ++++++++++++++++++++++++++++++--------------------- 1 file changed, 78 insertions(+), 52 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d55cac6..e659c74 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3778,18 +3778,13 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, return 0; } -int snd_soc_dapm_new_pcm(struct snd_soc_card *card, - const struct snd_soc_pcm_stream *params, - unsigned int num_params, - struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) +static struct snd_kcontrol_new * +snd_soc_dapm_alloc_kcontrol(struct snd_soc_card *card, + char *link_name, + const struct snd_soc_pcm_stream *params, + int num_params, const char **w_param_text, + unsigned long *private_value) { - struct snd_soc_dapm_widget template; - struct snd_soc_dapm_widget *w; - char *link_name; - int ret, count; - unsigned long private_value; - const char **w_param_text; struct soc_enum w_param_enum[] = { SOC_ENUM_SINGLE(0, 0, 0, NULL), }; @@ -3798,19 +3793,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, snd_soc_dapm_dai_link_get, snd_soc_dapm_dai_link_put), }; + struct snd_kcontrol_new *kcontrol_news; const struct snd_soc_pcm_stream *config = params; - - w_param_text = devm_kcalloc(card->dev, num_params, - sizeof(char *), GFP_KERNEL); - if (!w_param_text) - return -ENOMEM; - - link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", - source->name, sink->name); - if (!link_name) { - ret = -ENOMEM; - goto outfree_w_param; - } + int count; for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { @@ -3827,47 +3812,90 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, strlen(config->stream_name) + 1, GFP_KERNEL); } - if (!w_param_text[count]) { - ret = -ENOMEM; - goto outfree_link_name; - } + if (!w_param_text[count]) + goto outfree_w_param; config++; } + w_param_enum[0].items = num_params; w_param_enum[0].texts = w_param_text; - memset(&template, 0, sizeof(template)); - template.reg = SND_SOC_NOPM; - template.id = snd_soc_dapm_dai_link; - template.name = link_name; - template.event = snd_soc_dai_link_event; - template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD; - template.num_kcontrols = 1; - /* duplicate w_param_enum on heap so that memory persists */ - private_value = + *private_value = (unsigned long) devm_kmemdup(card->dev, (void *)(kcontrol_dai_link[0].private_value), sizeof(struct soc_enum), GFP_KERNEL); - if (!private_value) { + if (!*private_value) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; - goto outfree_link_name; + goto outfree_w_param; } - kcontrol_dai_link[0].private_value = private_value; + kcontrol_dai_link[0].private_value = *private_value; /* duplicate kcontrol_dai_link on heap so that memory persists */ - template.kcontrol_news = - devm_kmemdup(card->dev, &kcontrol_dai_link[0], - sizeof(struct snd_kcontrol_new), - GFP_KERNEL); - if (!template.kcontrol_news) { + kcontrol_news = + devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + if (!kcontrol_news) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; goto outfree_private_value; } + return kcontrol_news; + +outfree_private_value: + devm_kfree(card->dev, (void *)*private_value); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + devm_kfree(card->dev, (void *)w_param_text[count]); + return NULL; +} +int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + unsigned int num_params, + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget template; + struct snd_soc_dapm_widget *w; + const char **w_param_text; + unsigned long private_value; + char *link_name; + int ret, count; + + link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", + source->name, sink->name); + if (!link_name) + return -ENOMEM; + + memset(&template, 0, sizeof(template)); + template.reg = SND_SOC_NOPM; + template.id = snd_soc_dapm_dai_link; + template.name = link_name; + template.event = snd_soc_dai_link_event; + template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD; + template.kcontrol_news = NULL; + + w_param_text = devm_kcalloc(card->dev, num_params, + sizeof(char *), GFP_KERNEL); + if (!w_param_text) { + ret = -ENOMEM; + goto param_fail; + } + + /* allocate memory for control, only in case of multiple configs */ + if (num_params > 1) { + template.num_kcontrols = 1; + template.kcontrol_news = + snd_soc_dapm_alloc_kcontrol(card, + link_name, params, num_params, + w_param_text, &private_value); + if (!template.kcontrol_news) { + ret = -ENOMEM; + goto outfree_link_name; + } + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); @@ -3899,15 +3927,13 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, devm_kfree(card->dev, w); outfree_kcontrol_news: devm_kfree(card->dev, (void *)template.kcontrol_news); -outfree_private_value: devm_kfree(card->dev, (void *)private_value); -outfree_link_name: - devm_kfree(card->dev, link_name); for (count = 0 ; count < num_params; count++) devm_kfree(card->dev, (void *)w_param_text[count]); -outfree_w_param: +outfree_link_name: devm_kfree(card->dev, w_param_text); - +param_fail: + devm_kfree(card->dev, link_name); return ret; } -- 2.5.4 (Apple Git-61)

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[alsa-devel] Applied "ASoC: max98927: Added support for DSP_A and DSP_B format" to the asoc tree
by Mark Brown 25 Sep '17

25 Sep '17

The patch ASoC: max98927: Added support for DSP_A and DSP_B format has been applied to the asoc tree at git://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 4eee20246c0fa77a7a7b189ac806d5f78d675546 Mon Sep 17 00:00:00 2001 From: Ryan Lee <ryans.lee(a)maximintegrated.com> Date: Thu, 14 Sep 2017 17:30:38 -0700 Subject: [PATCH] ASoC: max98927: Added support for DSP_A and DSP_B format Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98927.c | 35 ++++++++++++++++++++--------------- sound/soc/codecs/max98927.h | 6 ++++-- 2 files changed, 24 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index d9dbbe72f8ad..6f7b3ef48e25 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 +223,11 @@ 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); - - /* pdm channel configuration */ - if (max98927->iface & SND_SOC_DAIFMT_PDM) { + MAX98927_R0035_PDM_RX_CTRL, + MAX98927_PDM_RX_EN_MASK, 0); + } else { + /* pdm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0035_PDM_RX_CTRL, MAX98927_PDM_RX_EN_MASK, 1); @@ -231,10 +235,11 @@ 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; } diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index ece6a608cbe1..bf7a6f92562a 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) -- 2.14.1

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[alsa-devel] Applied "ASoC: max98927: Added max98927_dai_tdm_slot function" to the asoc tree
by Mark Brown 25 Sep '17

25 Sep '17

The patch ASoC: max98927: Added max98927_dai_tdm_slot function has been applied to the asoc tree at git://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 d4a8bce81cbd53420988f5db0d096ad04960f189 Mon Sep 17 00:00:00 2001 From: Ryan Lee <ryans.lee(a)maximintegrated.com> Date: Thu, 14 Sep 2017 17:30:39 -0700 Subject: [PATCH] ASoC: max98927: Added max98927_dai_tdm_slot function Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98927.c | 120 ++++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/max98927.h | 1 + 2 files changed, 105 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index 6f7b3ef48e25..a1d39353719d 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -250,6 +250,21 @@ 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) { @@ -275,23 +290,20 @@ 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: - return -EINVAL; + if (!max98927->tdm_mode) { + /* BCLK configuration */ + value = max98927_get_bclk_sel(blr_clk_ratio); + if (!value) { + dev_err(codec->dev, "format unsupported %d\n", + params_format(params)); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0022_PCM_CLK_SETUP, + MAX98927_PCM_CLK_SETUP_BSEL_MASK, + value); } - regmap_update_bits(max98927->regmap, - MAX98927_R0022_PCM_CLK_SETUP, - MAX98927_PCM_CLK_SETUP_BSEL_MASK, - value); return 0; } @@ -391,6 +403,78 @@ 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; + + max98927->tdm_mode = true; + + /* 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 | \ @@ -410,6 +494,7 @@ 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, @@ -419,6 +504,9 @@ static int max98927_dac_event(struct snd_soc_dapm_widget *w, struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); switch (event) { + case SND_SOC_DAPM_PRE_PMU: + max98927->tdm_mode = 0; + break; case SND_SOC_DAPM_POST_PMU: regmap_update_bits(max98927->regmap, MAX98927_R003A_AMP_EN, diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index bf7a6f92562a..9ea839735433 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -270,5 +270,6 @@ struct max98927_priv { unsigned int iface; unsigned int master; unsigned int digital_gain; + bool tdm_mode; }; #endif -- 2.14.1

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[alsa-devel] [PATCH 1/2] ASoC: max98927: Added support for DSP_A and DSP_B format
by Ryan Lee 25 Sep '17

25 Sep '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98927.c | 35 ++++++++++++++++++++--------------- sound/soc/codecs/max98927.h | 6 ++++-- 2 files changed, 24 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index d9dbbe7..6f7b3ef 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 +223,11 @@ 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); - - /* pdm channel configuration */ - if (max98927->iface & SND_SOC_DAIFMT_PDM) { + MAX98927_R0035_PDM_RX_CTRL, + MAX98927_PDM_RX_EN_MASK, 0); + } else { + /* pdm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0035_PDM_RX_CTRL, MAX98927_PDM_RX_EN_MASK, 1); @@ -231,10 +235,11 @@ 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; } 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] [PATCH] ASoC: dapm: Make snd_soc_dapm_add/del_routes use runtime mutex subclass
by Stuart Henderson 25 Sep '17

25 Sep '17

Make snd_soc_dapm_add_routes and snd_soc_dapm_del_routes use the SND_SOC_DAPM_CLASS_RUNTIME dapm subclass to allow dynamic dapm route updates. Signed-off-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dcef67a..e6b6011 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2884,7 +2884,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, { int i, r, ret = 0; - mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); for (i = 0; i < num; i++) { r = snd_soc_dapm_add_route(dapm, route); if (r < 0) { @@ -2915,7 +2915,7 @@ int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm, { int i; - mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); for (i = 0; i < num; i++) { snd_soc_dapm_del_route(dapm, route); route++; -- 2.1.4

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[alsa-devel] [PATCH 0/4][RESEND] ASoC: rtd NULL check and unneeded ops check
by Kuninori Morimoto 25 Sep '17

25 Sep '17

Hi Mark These are posted few weeks ago, and nothting happen. Thus, I resend these. I'm posting "Convert Codec/Platform to Component" patch-set too, but I know it is very big patch-set, need more review. OTOH, These are very light patches. Thus, I will repost these 1st. Then, I will repost "Convert ..." patch-set again on top of these if these are accepted. Kuninori Morimoto (4): ASoC: soc-core: remove rtd NULL check on soc_free_pcm_runtime() ASoC: remove unneeded dai->driver check ASoC: remove unneeded dai->driver->ops check ASoC: add null_snd_soc_ops and reduce NULL ops check sound/soc/soc-core.c | 24 ++++++++++++++---------- sound/soc/soc-dapm.c | 8 ++++---- sound/soc/soc-pcm.c | 47 +++++++++++++++++++++++------------------------ 3 files changed, 41 insertions(+), 38 deletions(-) -- 1.9.1 Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: rt5663: Change the dev getting function in rt5663_irq
by Oder Chiou 25 Sep '17

25 Sep '17

In the irq handler "rt5663_irq", while the codec is not initialized, rt5663->codec will be null, and it will cause the kernel panic in the debug print enabled. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 767f219..fcd1643 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -1827,7 +1827,8 @@ static irqreturn_t rt5663_irq(int irq, void *data) { struct rt5663_priv *rt5663 = data; - dev_dbg(rt5663->codec->dev, "%s IRQ queue work\n", __func__); + dev_dbg(regmap_get_device(rt5663->regmap), "%s IRQ queue work\n", + __func__); queue_delayed_work(system_wq, &rt5663->jack_detect_work, msecs_to_jiffies(250)); -- 2.7.4

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[alsa-devel] [PATCH 0/3] Intel-rt5670 updates
by Pierre-Louis Bossart 25 Sep '17

25 Sep '17

Instead of submitting all my patches at ones, I broke them by platforms/codecs. This series is only for Intel-rt5670 platforms. It fixes a mistake on the Dell Venue 5585 - corrected with the help from Realtek and Dell who kindly provided schematics and data sheet information needed to fix non-working DMIC capture. Since there are still some reports of non-functional audio on some platforms, I added quirks that can be changed from the kernel command line (and later added as DMI-based quirks). The last two patches are minor corrections to fix ACPI HID naming issues and use a more meaningful card name. Pierre-Louis Bossart (3): ASoC: rt5670: refactor DMI quirks and fix Dell Venue settings ASoC: Intel: cht_bsw_rt5672: use actual HID in suspend/resume ASoC: Intel: cht_bsw_rt5672: fix card name sound/soc/codecs/rt5670.c | 124 +++++++++++++++++++++++++++----- sound/soc/intel/boards/cht_bsw_rt5672.c | 10 ++- 2 files changed, 112 insertions(+), 22 deletions(-) -- 2.9.3

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[alsa-devel] [PATCH 00/14] BYT/CHT codec/machine drivers update
by Pierre-Louis Bossart 25 Sep '17

25 Sep '17

Finally took the time to cleanup known fixes for codec and machine drivers. I tested on Baytrail/Cherrytrail and w/ RT5640, 45 and 72. This should be the last batch of fixes before we move on to the integration of SOF (Sound Open Firmware) support in the mainline. This batch contains two patches already submitted by Thierry Escande on alsa-devel but never merged. Fang, Yang A (1): ASoC: ts3a227e: add acpi table Pierre-Louis Bossart (12): ASoC: Intel: boards: use devm_clk_get() unconditionally ASoC: intel: cht_bsw_max98090: add support for clock framework ASoC: Intel: cht_bsw_max98090: Fix I2S config + unused code ASoC: Intel: atom: add support for Baytrail Chromebooks ASoC: Intel: bytcr-rt5651: fix capture routes ASoC: Intel: bytcr_rt5651: add MCLK support ASoC: Intel: bytcr_rt5651: filter codec name ASoC: codecs: rt5645: add quirks for Asus T100HA ASoC: Intel: bytcr_rt5640: simplify MCLK quirk tests ASoC: codecs: rt5670: add jack detection quirk for Dell Venue 5585 ASoC: Intel: cht_bsw_rt5672: use actual HID in suspend/resume ASoC: Intel: cht_bsw_rt5672: fix card name Thierry Escande (1): ASoc: cht_bsw_max98090_ti: Fix jack initialization sound/soc/codecs/rt5645.c | 20 ++ sound/soc/codecs/rt5670.c | 16 ++ sound/soc/codecs/ts3a227e.c | 10 + sound/soc/intel/atom/sst/sst_acpi.c | 8 + sound/soc/intel/boards/bytcr_rt5640.c | 8 +- sound/soc/intel/boards/bytcr_rt5651.c | 278 ++++++++++++++++++++++++--- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 157 ++++++++++++--- sound/soc/intel/boards/cht_bsw_rt5645.c | 14 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 36 ++-- 9 files changed, 460 insertions(+), 87 deletions(-) -- 2.9.3

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