diff options
author | Linus Torvalds <torvalds@linux-foundation.org> | 2018-10-25 09:00:15 -0700 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2018-10-25 09:00:15 -0700 |
commit | 3acbd2de6bc3af215c6ed7732dfc097d1e238503 (patch) | |
tree | 5152e90a4d2d586dd6ad1cf0b8f28c4de2e46e66 /sound/soc/codecs/tas5720.c | |
parent | d49f8a52b15bf35db778035340d8a673149f9f93 (diff) | |
parent | de7d83da84bdf0b5ec50b3b09249e608c0e4b81d (diff) |
Merge tag 'sound-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"There have been little changes in ALSA core stuff, but ASoC core still
kept rolling for the continued restructuring. The rest are lots of
small driver-specific changes and some minor API updates. Here are
highlights:
General:
- Appropriate fall-through annotations everywhere
- Some code cleanup in memalloc code, handling non-cacahed pages more
commonly in the helper
- Deployment of SNDRV_PCM_INFO_SYNC_APPLPTR flag consistently
Drivers:
- More HD-audio CA0132 codec improvement for supporting other Creative
boards
- Plumbing legacy HD-audio codecs as ASoC BE on Intel SST; this will
give move support of existing HD-audio devices with DSP
- A few device-specific HD-audio quirks as usual
- New quirk for RME CC devices and correction for B&W PX for USB-audio
- FireWire: code refactoring including devres usages
ASoC Core:
- Continued componentization works; it's almost done!
- A bunch of new for_each_foo macros
- Cleanups and fixes in DAPM code
ASoC Drivers:
- MCLK support for several different devices, including CS42L51, STM32
SAI, and MAX98373
- Support for Allwinner A64 CODEC analog, Intel boards with DA7219 and
MAX98927, Meson AXG PDM inputs, Nuvoton NAU8822, Renesas R8A7744 and
TI PCM3060"
* tag 'sound-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (299 commits)
ASoC: stm32: sai: fix master clock naming
ASoC: stm32: add clock dependency for sai
ALSA: hda/ca0132 - Actually fix microphone issue
ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks
ASoC: wm2000: Remove wm2000_read helper function
ASoC: cs42l51: fix mclk support
ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate
ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer
ASoC: cs42l51: add mclk support
ASoC: stm32: sai: set sai as mclk clock provider
ASoC: dt-bindings: add mclk support to cs42l51
ASoC: dt-bindings: add mclk provider support to stm32 sai
ASoC: soc-core: fix trivial checkpatch issues
ASoC: dapm: Add support for hw_free on CODEC to CODEC links
ASoC: Intel: kbl_da7219_max98927: minor white space clean up
ALSA: i2c/cs8427: Fix int to char conversion
ALSA: doc: Brush up the old writing-an-alsa-driver
ASoC: rsnd: tidyup SSICR::SWSP for TDM
ASoC: rsnd: enable TDM settings for SSI parent
ASoC: pcm3168a: add hw constraint for capture channel
...
Diffstat (limited to 'sound/soc/codecs/tas5720.c')
-rw-r--r-- | sound/soc/codecs/tas5720.c | 103 |
1 files changed, 95 insertions, 8 deletions
diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index ae3d032ac35a..6bd0e5d5347f 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -152,6 +152,7 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai, int slots, int slot_width) { struct snd_soc_component *component = dai->component; + struct tas5720_data *tas5720 = snd_soc_component_get_drvdata(component); unsigned int first_slot; int ret; @@ -185,6 +186,20 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai, if (ret < 0) goto error_snd_soc_component_update_bits; + /* Configure TDM slot width. This is only applicable to TAS5722. */ + switch (tas5720->devtype) { + case TAS5722: + ret = snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG, + TAS5722_TDM_SLOT_16B, + slot_width == 16 ? + TAS5722_TDM_SLOT_16B : 0); + if (ret < 0) + goto error_snd_soc_component_update_bits; + break; + default: + break; + } + return 0; error_snd_soc_component_update_bits: @@ -485,15 +500,56 @@ static const DECLARE_TLV_DB_RANGE(dac_analog_tlv, ); /* - * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB steps. Note that - * setting the gain below -100 dB (register value <0x7) is effectively a MUTE - * as per device datasheet. + * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB or 0.25 dB steps + * depending on the device. Note that setting the gain below -100 dB + * (register value <0x7) is effectively a MUTE as per device datasheet. + * + * Note that for the TAS5722 the digital volume controls are actually split + * over two registers, so we need custom getters/setters for access. */ -static DECLARE_TLV_DB_SCALE(dac_tlv, -10350, 50, 0); +static DECLARE_TLV_DB_SCALE(tas5720_dac_tlv, -10350, 50, 0); +static DECLARE_TLV_DB_SCALE(tas5722_dac_tlv, -10350, 25, 0); + +static int tas5722_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + unsigned int val; + + snd_soc_component_read(component, TAS5720_VOLUME_CTRL_REG, &val); + ucontrol->value.integer.value[0] = val << 1; + + snd_soc_component_read(component, TAS5722_DIGITAL_CTRL2_REG, &val); + ucontrol->value.integer.value[0] |= val & TAS5722_VOL_CONTROL_LSB; + + return 0; +} + +static int tas5722_volume_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + unsigned int sel = ucontrol->value.integer.value[0]; + + snd_soc_component_write(component, TAS5720_VOLUME_CTRL_REG, sel >> 1); + snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG, + TAS5722_VOL_CONTROL_LSB, sel); + + return 0; +} static const struct snd_kcontrol_new tas5720_snd_controls[] = { SOC_SINGLE_TLV("Speaker Driver Playback Volume", - TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, dac_tlv), + TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, tas5720_dac_tlv), + SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG, + TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv), +}; + +static const struct snd_kcontrol_new tas5722_snd_controls[] = { + SOC_SINGLE_EXT_TLV("Speaker Driver Playback Volume", + 0, 0, 511, 0, + tas5722_volume_get, tas5722_volume_set, + tas5722_dac_tlv), SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG, TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv), }; @@ -527,6 +583,23 @@ static const struct snd_soc_component_driver soc_component_dev_tas5720 = { .non_legacy_dai_naming = 1, }; +static const struct snd_soc_component_driver soc_component_dev_tas5722 = { + .probe = tas5720_codec_probe, + .remove = tas5720_codec_remove, + .suspend = tas5720_suspend, + .resume = tas5720_resume, + .controls = tas5722_snd_controls, + .num_controls = ARRAY_SIZE(tas5722_snd_controls), + .dapm_widgets = tas5720_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets), + .dapm_routes = tas5720_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + /* PCM rates supported by the TAS5720 driver */ #define TAS5720_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) @@ -613,9 +686,23 @@ static int tas5720_probe(struct i2c_client *client, dev_set_drvdata(dev, data); - ret = devm_snd_soc_register_component(&client->dev, - &soc_component_dev_tas5720, - tas5720_dai, ARRAY_SIZE(tas5720_dai)); + switch (id->driver_data) { + case TAS5720: + ret = devm_snd_soc_register_component(&client->dev, + &soc_component_dev_tas5720, + tas5720_dai, + ARRAY_SIZE(tas5720_dai)); + break; + case TAS5722: + ret = devm_snd_soc_register_component(&client->dev, + &soc_component_dev_tas5722, + tas5720_dai, + ARRAY_SIZE(tas5720_dai)); + break; + default: + dev_err(dev, "unexpected private driver data\n"); + return -EINVAL; + } if (ret < 0) { dev_err(dev, "failed to register component: %d\n", ret); return ret; |