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

[alsa-devel] [PATCH] ALSA: hda: Fix resource leak issue in snd_hda_codec_build_controls and snd_hda_codec_parse_pcms
by Wang YanQing 03 Sep '17

03 Sep '17

When patch_ops.init, patch_ops.build_pcms and patch_ops.build_controls return failure, we need to free resource with patch_ops.free, or we will get resource leak. Signed-off-by: Wang YanQing <udknight(a)gmail.com> --- sound/pci/hda/hda_codec.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index df6b57e..4e3e613 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2973,8 +2973,11 @@ int snd_hda_codec_build_controls(struct hda_codec *codec) err = codec->patch_ops.init(codec); if (!err && codec->patch_ops.build_controls) err = codec->patch_ops.build_controls(codec); - if (err < 0) + if (err < 0) { + if (codec->patch_ops.free) + codec->patch_ops.free(codec); return err; + } /* we create chmaps here instead of build_pcms */ err = add_std_chmaps(codec); @@ -3170,6 +3173,8 @@ int snd_hda_codec_parse_pcms(struct hda_codec *codec) if (err < 0) { codec_err(codec, "cannot build PCMs for #%d (error %d)\n", codec->core.addr, err); + if (codec->patch_ops.free) + codec->patch_ops.free(codec); return err; } -- 1.8.5.6.2.g3d8a54e.dirty

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[alsa-devel] [PATCH] ALSA: hda: Fix forget to free resource in error handling code path in hda_codec_driver_probe
by Wang YanQing 03 Sep '17

03 Sep '17

When hda_codec_driver_probe meet error and return failure, we need to free resource with patch_ops.free, or we will get resource leak. Signed-off-by: Wang YanQing <udknight(a)gmail.com> --- sound/pci/hda/hda_bind.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 6efadbf..f1b1b7e 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -123,6 +123,8 @@ static int hda_codec_driver_probe(struct device *dev) module_put(owner); error: + if (codec->patch_ops.free) + codec->patch_ops.free(codec); snd_hda_codec_cleanup_for_unbind(codec); return err; } -- 1.8.5.6.2.g3d8a54e.dirty

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[alsa-devel] [PATCH] ALSA: hda: Fix assign invalid fixed slots pcm numbers to HDA_PCM_TYPE_HDMI in get_empty_pcm_device
by Wang YanQing 03 Sep '17

03 Sep '17

The length of pcm_dev_bits in hda_bus is SNDRV_PCM_DEVICES, and the definiton of SNDRV_PCM_DEVICES is: " #if defined(CONFIG_SND_DYNAMIC_MINORS) #define SNDRV_PCM_DEVICES (SNDRV_OS_MINORS-2) #else #define SNDRV_PCM_DEVICES 8 #endif " So the max valid fixed slots pcm number is 7, and the first non-fixed slot is 8. Signed-off-by: Wang YanQing <udknight(a)gmail.com> --- sound/pci/hda/hda_codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3db26c4..df6b57e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3118,7 +3118,7 @@ static int get_empty_pcm_device(struct hda_bus *bus, unsigned int type) static int audio_idx[HDA_PCM_NTYPES][5] = { [HDA_PCM_TYPE_AUDIO] = { 0, 2, 4, 5, -1 }, [HDA_PCM_TYPE_SPDIF] = { 1, -1 }, - [HDA_PCM_TYPE_HDMI] = { 3, 7, 8, 9, -1 }, + [HDA_PCM_TYPE_HDMI] = { 3, 7, -1 }, [HDA_PCM_TYPE_MODEM] = { 6, -1 }, }; int i; @@ -3139,7 +3139,7 @@ static int get_empty_pcm_device(struct hda_bus *bus, unsigned int type) #ifdef CONFIG_SND_DYNAMIC_MINORS /* non-fixed slots starting from 10 */ - for (i = 10; i < 32; i++) { + for (i = 8; i < 32; i++) { if (!test_and_set_bit(i, bus->pcm_dev_bits)) return i; } -- 1.8.5.6.2.g3d8a54e.dirty

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[alsa-devel] [PATCH] ALSA: hda: Fix regression of hdmi eld control created based on invalid pcm
by Wang YanQing 03 Sep '17

03 Sep '17

Commit fb087eaaef72 ("ALSA: hda - hdmi eld control created based on pcm") forget to filter out invalid pcm numbers, if there is only one invalid pcm number, then this issue causes we create eld control for invalid pcm silently, but when there are more than one invalid pcm numbers, then this issue bring probe error looks like below dmesg: " kernel: [ 1.647283] snd_hda_intel 0000:00:03.0: bound 0000:00:02.0 (ops 0xc2967540) kernel: [ 1.651192] snd_hda_intel 0000:00:03.0: Too many HDMI devices kernel: [ 1.651195] snd_hda_intel 0000:00:03.0: Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y kernel: [ 1.651197] snd_hda_intel 0000:00:03.0: Too many HDMI devices kernel: [ 1.651199] snd_hda_intel 0000:00:03.0: Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y kernel: [ 1.651201] snd_hda_intel 0000:00:03.0: Too many HDMI devices kernel: [ 1.651203] snd_hda_intel 0000:00:03.0: Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y kernel: [ 1.651676] snd_hda_intel 0000:00:03.0: control 3:0:0:ELD:0 is already present kernel: [ 1.651787] snd_hda_codec_hdmi: probe of hdaudioC0D0 failed with error -16 " This patch add invalid pcm number filter before calling hdmi_create_eld_ctl. Fixes: fb087eaaef72 ("ALSA: hda - hdmi eld control created based on pcm") Signed-off-by: Wang YanQing <udknight(a)gmail.com> --- sound/pci/hda/hda_codec.c | 4 +++- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/patch_hdmi.c | 14 ++++++++------ 3 files changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 821aad3..3db26c4 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3213,8 +3213,10 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) continue; /* no substreams assigned */ dev = get_empty_pcm_device(bus, cpcm->pcm_type); - if (dev < 0) + if (dev < 0) { + cpcm->device = SNDRV_PCM_INVALID_DEVICE; continue; /* no fatal error */ + } cpcm->device = dev; err = snd_hda_attach_pcm_stream(bus, codec, cpcm); if (err < 0) { diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 60ce1cf..681c360 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -164,6 +164,7 @@ enum { HDA_PCM_NTYPES }; +#define SNDRV_PCM_INVALID_DEVICE (-1) /* for PCM creation */ struct hda_pcm { char *name; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 53f9311..2b64fab 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2126,7 +2126,7 @@ static int generic_hdmi_build_jack(struct hda_codec *codec, int pcm_idx) static int generic_hdmi_build_controls(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; - int err; + int dev, err; int pin_idx, pcm_idx; @@ -2154,11 +2154,13 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) return err; snd_hda_spdif_ctls_unassign(codec, pcm_idx); - /* add control for ELD Bytes */ - err = hdmi_create_eld_ctl(codec, pcm_idx, - get_pcm_rec(spec, pcm_idx)->device); - if (err < 0) - return err; + dev = get_pcm_rec(spec, pcm_idx)->device; + if (dev != SNDRV_PCM_INVALID_DEVICE) { + /* add control for ELD Bytes */ + err = hdmi_create_eld_ctl(codec, pcm_idx, dev); + if (err < 0) + return err; + } } for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { -- 1.8.5.6.2.g3d8a54e.dirty

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[alsa-devel] Raspberry PI reports no channels, period and buffer
by Petr Kulhavy 03 Sep '17

03 Sep '17

Hi, I'm writing an audio decoding program for ALSA, which among others detects audio capabilities. It uses the snd_pcm_hw_params_get_XXX() functions for that. This works well on my Ubuntu PC but on Raspberry PI 3 these calls return no information. They return success but the min and max values are meaningless. For instance the period: err = snd_pcm_hw_params_get_period_size_min(hw_params, &m_period_min, NULL); if (err < 0) { LOG(LOG_ERROR, LOG_OBJ "Failed to get minimum period time: %s\n", snd_strerror(err)); goto fail; } err = snd_pcm_hw_params_get_period_size_max(hw_params, &m_period_max, NULL); if (err < 0) { LOG(LOG_ERROR, LOG_OBJ "Failed to get maximum period time: %s\n", snd_strerror(err)); goto fail; } err = snd_pcm_hw_params_get_period_time_min(hw_params, &m_period_min_us, NULL); if (err < 0) { LOG(LOG_ERROR, LOG_OBJ "Failed to get minimum period time: %s\n", snd_strerror(err)); goto fail; } err = snd_pcm_hw_params_get_period_time_max(hw_params, &m_period_max_us, NULL); if (err < 0) { LOG(LOG_ERROR, LOG_OBJ "Failed to get maximum period time: %s\n", snd_strerror(err)); goto fail; } LOG(LOG_DEBUG, LOG_OBJ "Supported period: %lu to %lu frames (%u to %u us)\n", (unsigned long)m_period_min, (unsigned long)m_period_max, m_period_min_us, m_period_max_us); This prints: Supported period: 0 to 0 frames (1 to 0 us) // query buffer size err = snd_pcm_hw_params_get_buffer_size_min(hw_params, &m_buffer_min); if (err < 0) { LOG(LOG_ERROR, LOG_OBJ "Failed get minimum buffer time: %s\n", snd_strerror(err)); goto fail; } err = snd_pcm_hw_params_get_buffer_size_max(hw_params, &m_buffer_max); if (err < 0) { LOG(LOG_ERROR, LOG_OBJ "Failed get maximum buffer time: %s\n", snd_strerror(err)); goto fail; } err = snd_pcm_hw_params_get_buffer_time_min(hw_params, &m_buffer_min_us, NULL); if (err < 0) { LOG(LOG_ERROR, LOG_OBJ "Failed get minimum buffer time: %s\n", snd_strerror(err)); goto fail; } err = snd_pcm_hw_params_get_buffer_time_max(hw_params, &m_buffer_max_us, NULL); if (err < 0) { LOG(LOG_ERROR, LOG_OBJ "Failed get maximum buffer time: %s\n", snd_strerror(err)); goto fail; } LOG(LOG_DEBUG, LOG_OBJ "Supported buffer size: %lu to %lu frames (%u to %u us)\n", (unsigned long)m_buffer_min, (unsigned long)m_buffer_max, m_buffer_min_us, m_buffer_max_us); This prints: Supported buffer size: 0 to 0 frames (1 to 0 us) And so on. Same for the number of periods and channels. On the other hand the snd_pcm_hw_params_test_XXX() calls do work and that way I can detect the rates, formats, accesses. Why is that? Am I doing something wrong? And more importantly, how can I get the buffering and channel information? Thanks Petr

2 1

[alsa-devel] [patch 1/4] ASoC: dapm: fix error path in snd_soc_dapm_new_pcm
by anish kumar 02 Sep '17

02 Sep '17

Before freeing the memory, check if it was allcoated. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- sound/soc/soc-dapm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dcef67a..6e10a9a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3907,9 +3907,10 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, devm_kfree(card->dev, (void *)private_value); outfree_link_name: devm_kfree(card->dev, link_name); -outfree_w_param: for (count = 0 ; count < num_params; count++) - devm_kfree(card->dev, (void *)w_param_text[count]); + if (w_param_text[count]) + devm_kfree(card->dev, (void *)w_param_text[count]); +outfree_w_param: devm_kfree(card->dev, w_param_text); return ret; -- 2.10.1 (Apple Git-78)

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[alsa-devel] [PATCH 1/9] ASoC: max98927: Added support for DSP_A and DSP_B format
by Ryan Lee 01 Sep '17

01 Sep '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 b0380b5..951cc05 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 int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) 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 int max98927_dai_set_sysclk(struct snd_soc_dai *dai, .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) -- 1.9.1

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[alsa-devel] [PATCH 1/3] ASoC: max98927: Added support for DSP_A and DSP_B format
by Ryan Lee 01 Sep '17

01 Sep '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changelog: Added support for DSP_A and DSP_B format Added 'max98927_dai_tdm_slot' function to configure for proper slot configuration. Moved max98927->iface out of switch statement to avoid line duplication. Added variable 'tdm_mode' to avoid BCLK overwrite after 'max98927_dai_tdm_slot' configure right BCLK value. sound/soc/codecs/max98927.c | 155 +++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/max98927.h | 7 +- 2 files changed, 129 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index d9dbbe7..a1d3935 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; } @@ -245,6 +250,21 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) 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) { @@ -270,23 +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; } @@ -386,6 +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 | \ @@ -405,6 +494,7 @@ static int max98927_dai_set_sysclk(struct snd_soc_dai *dai, .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, @@ -414,6 +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 ece6a60..9ea8397 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) @@ -268,5 +270,6 @@ struct max98927_priv { unsigned int iface; unsigned int master; unsigned int digital_gain; + bool tdm_mode; }; #endif -- 1.9.1

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[alsa-devel] [PATCH v1] ALSA: atmel: Remove leftovers of AVR32 removal
by Andy Shevchenko 01 Sep '17

01 Sep '17

The ALSA related include header files are left overs after the commit 020c5260c2b1 ("ALSA: atmel: Remove AVR32 bits from the driver") Fixes: 020c5260c2b1 ("ALSA: atmel: Remove AVR32 bits from the driver") Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- include/sound/atmel-abdac.h | 23 ----------------------- include/sound/atmel-ac97c.h | 38 -------------------------------------- sound/atmel/ac97c.c | 1 - 3 files changed, 62 deletions(-) delete mode 100644 include/sound/atmel-abdac.h delete mode 100644 include/sound/atmel-ac97c.h diff --git a/include/sound/atmel-abdac.h b/include/sound/atmel-abdac.h deleted file mode 100644 index a8f735d677fa..000000000000 --- a/include/sound/atmel-abdac.h +++ /dev/null @@ -1,23 +0,0 @@ -/* - * Driver for the Atmel Audio Bitstream DAC (ABDAC) - * - * Copyright (C) 2009 Atmel Corporation - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License version 2 as published - * by the Free Software Foundation. - */ -#ifndef __INCLUDE_SOUND_ATMEL_ABDAC_H -#define __INCLUDE_SOUND_ATMEL_ABDAC_H - -#include <linux/platform_data/dma-dw.h> - -/** - * struct atmel_abdac_pdata - board specific ABDAC configuration - * @dws: DMA slave interface to use for sound playback. - */ -struct atmel_abdac_pdata { - struct dw_dma_slave dws; -}; - -#endif /* __INCLUDE_SOUND_ATMEL_ABDAC_H */ diff --git a/include/sound/atmel-ac97c.h b/include/sound/atmel-ac97c.h deleted file mode 100644 index f2a1cdc37661..000000000000 --- a/include/sound/atmel-ac97c.h +++ /dev/null @@ -1,38 +0,0 @@ -/* - * Driver for the Atmel AC97C controller - * - * Copyright (C) 2005-2009 Atmel Corporation - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License version 2 as published - * by the Free Software Foundation. - */ -#ifndef __INCLUDE_SOUND_ATMEL_AC97C_H -#define __INCLUDE_SOUND_ATMEL_AC97C_H - -#include <linux/platform_data/dma-dw.h> - -#define AC97C_CAPTURE 0x01 -#define AC97C_PLAYBACK 0x02 -#define AC97C_BOTH (AC97C_CAPTURE | AC97C_PLAYBACK) - -/** - * struct atmel_ac97c_pdata - board specific AC97C configuration - * @rx_dws: DMA slave interface to use for sound capture. - * @tx_dws: DMA slave interface to use for sound playback. - * @reset_pin: GPIO pin wired to the reset input on the external AC97 codec, - * optional to use, set to -ENODEV if not in use. AC97 layer will - * try to do a software reset of the external codec anyway. - * - * If the user do not want to use a DMA channel for playback or capture, i.e. - * only one feature is required on the board. The slave for playback or capture - * can be set to NULL. The AC97C driver will take use of this when setting up - * the sound streams. - */ -struct ac97c_platform_data { - struct dw_dma_slave rx_dws; - struct dw_dma_slave tx_dws; - int reset_pin; -}; - -#endif /* __INCLUDE_SOUND_ATMEL_AC97C_H */ diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 5ffefac2fa8f..6d085e1d1d0a 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -29,7 +29,6 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/ac97_codec.h> -#include <sound/atmel-ac97c.h> #include <sound/memalloc.h> #include "ac97c.h" -- 2.14.1

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[alsa-devel] [asoc:topic/cs43130 1/2] sound/soc/codecs/cs43130.c:2089:18: warning: 'hpload_seq' may be used uninitialized in this function
by kbuild test robot 01 Sep '17

01 Sep '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/cs43130 head: 465c925695593b43695e2396f7279879a72275eb commit: 8f1e5bf9b4408e91942b94c76583eaf098f19382 [1/2] ASoC: cs43130: Add support for CS43130 codec config: i386-allmodconfig (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout 8f1e5bf9b4408e91942b94c76583eaf098f19382 # save the attached .config to linux build tree make ARCH=i386 Note: it may well be a FALSE warning. FWIW you are at least aware of it now. http://gcc.gnu.org/wiki/Better_Uninitialized_Warnings All warnings (new ones prefixed by >>): sound/soc/codecs/cs43130.c: In function 'cs43130_imp_meas': >> sound/soc/codecs/cs43130.c:2089:18: warning: 'hpload_seq' may be used uninitialized in this function [-Wmaybe-uninitialized] hpload_seq[i].msk, ac_idx); ^ vim +/hpload_seq +2089 sound/soc/codecs/cs43130.c 2042 2043 static void cs43130_imp_meas(struct work_struct *wk) 2044 { 2045 unsigned int reg, seq_size; 2046 int i, ret, ac_idx; 2047 struct cs43130_private *cs43130; 2048 struct snd_soc_codec *codec; 2049 struct reg_sequences *hpload_seq; 2050 2051 cs43130 = container_of(wk, struct cs43130_private, work); 2052 codec = cs43130->codec; 2053 2054 if (!cs43130->mclk) 2055 return; 2056 2057 cs43130->hpload_done = false; 2058 2059 mutex_lock(&cs43130->clk_mutex); 2060 if (!cs43130->clk_req) { 2061 /* clk not in use */ 2062 cs43130_set_pll(codec, 0, 0, cs43130->mclk, CS43130_MCLK_22M); 2063 if (cs43130->pll_bypass) 2064 cs43130_change_clksrc(codec, CS43130_MCLK_SRC_EXT); 2065 else 2066 cs43130_change_clksrc(codec, CS43130_MCLK_SRC_PLL); 2067 } 2068 2069 cs43130->clk_req++; 2070 mutex_unlock(&cs43130->clk_mutex); 2071 2072 regmap_read(cs43130->regmap, CS43130_INT_STATUS_4, &reg); 2073 2074 switch (cs43130->dev_id) { 2075 case CS43130_CHIP_ID: 2076 hpload_seq = hpload_seq1; 2077 seq_size = ARRAY_SIZE(hpload_seq1); 2078 break; 2079 case CS43131_CHIP_ID: 2080 hpload_seq = hpload_seq2; 2081 seq_size = ARRAY_SIZE(hpload_seq2); 2082 } 2083 2084 i = 0; 2085 ac_idx = 0; 2086 while (i < seq_size) { 2087 ret = cs43130_hpload_proc(cs43130, hpload_seq[i].seq, 2088 hpload_seq[i].size, > 2089 hpload_seq[i].msk, ac_idx); 2090 if (ret < 0) 2091 goto exit; 2092 2093 cs43130_update_hpload(hpload_seq[i].msk, ac_idx, cs43130); 2094 2095 if (cs43130->ac_meas && 2096 hpload_seq[i].msk == CS43130_HPLOAD_AC_INT && 2097 ac_idx < CS43130_AC_FREQ - 1) { 2098 ac_idx++; 2099 } else { 2100 ac_idx = 0; 2101 i++; 2102 } 2103 } 2104 cs43130->hpload_done = true; 2105 2106 if (cs43130->hpload_dc[HP_LEFT] >= CS43130_LINEOUT_LOAD) 2107 snd_soc_jack_report(&cs43130->jack, CS43130_JACK_LINEOUT, 2108 CS43130_JACK_MASK); 2109 else 2110 snd_soc_jack_report(&cs43130->jack, CS43130_JACK_HEADPHONE, 2111 CS43130_JACK_MASK); 2112 2113 dev_dbg(codec->dev, "Set HP output control. DC threshold\n"); 2114 for (i = 0; i < CS43130_DC_THRESHOLD; i++) 2115 dev_dbg(codec->dev, "DC threshold[%d]: %u.\n", i, 2116 cs43130->dc_threshold[i]); 2117 2118 cs43130_set_hv(cs43130->regmap, cs43130->hpload_dc[HP_LEFT], 2119 cs43130->dc_threshold); 2120 2121 exit: 2122 switch (cs43130->dev_id) { 2123 case CS43130_CHIP_ID: 2124 cs43130_hpload_proc(cs43130, hp_dis_cal_seq, 2125 ARRAY_SIZE(hp_dis_cal_seq), 2126 CS43130_HPLOAD_OFF_INT, ac_idx); 2127 break; 2128 case CS43131_CHIP_ID: 2129 cs43130_hpload_proc(cs43130, hp_dis_cal_seq2, 2130 ARRAY_SIZE(hp_dis_cal_seq2), 2131 CS43130_HPLOAD_OFF_INT, ac_idx); 2132 } 2133 2134 regmap_multi_reg_write(cs43130->regmap, hp_cln_seq, 2135 ARRAY_SIZE(hp_cln_seq)); 2136 2137 mutex_lock(&cs43130->clk_mutex); 2138 cs43130->clk_req--; 2139 /* clk not in use */ 2140 if (!cs43130->clk_req) 2141 cs43130_change_clksrc(codec, CS43130_MCLK_SRC_RCO); 2142 mutex_unlock(&cs43130->clk_mutex); 2143 } 2144 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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