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author | Linus Torvalds <torvalds@linux-foundation.org> | 2019-09-17 17:43:33 -0700 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2019-09-17 17:43:33 -0700 |
commit | 6ab8ad31601f29470eb895fd95e5c963e125aa1b (patch) | |
tree | 73327fe9fc2ee62e7815fa0a666fdf46aaab7322 /sound/soc/soc-dai.c | |
parent | ea982ba7f79141d86eb7a440fcba6796ed718b9b (diff) | |
parent | 9bf9bf5440b99edfba496388c90b52ebcd9df715 (diff) | |
download | linux-6ab8ad31601f29470eb895fd95e5c963e125aa1b.tar.bz2 |
Merge tag 'sound-5.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"As shown in diffstat and logs, it was again a busy development cycle
at this time, too. The most significant changes are still on-going
refactoring / modernization works for ASoC core and drivers, but there
are lots of other changes as well. Here we go, some highlights below:
ASoC:
- Quite a lot of cleanup / refactoring of ASoC core and APIs; most of
them are systematic, but also including cleanups and modernization
- A bulk of updates for some ASoC platforms, Freescale, sunxi and
Intel SST/SOF
- Initial support for Sound Open Firmware on i.MX8
- Removal of deprecated w90x900 and nuc900 drivers
- New support for Cirrus Logic CS47L15 and CS47L92, Freescale i.MX
7ULP and 8MQ, Meson G12A and NXP UDA1334
USB-audio:
- More validations of descriptor units for hardening against bugs
reported by fuzzers
- PCM device assignment workaround for a past call-order change
- Scarlett Gen2 mixer interface, a few more more quirks
HD-audio:
- Support for audio component with AMD/ATI and Nvidia HDMI codecs
- Clean up HD-audio core and remove indirect access ops for Intel SOF
- DMIC detection at probe; it would make systems automatically
falling back to SST/SOF driver on devices that need DMIC handling.
Needs a new Kconfig to set, and beware that it's still new and a
bit experimental
FireWire:
- Lots of code refactoring and cleanups"
* tag 'sound-5.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (521 commits)
ASoC: sdm845: remove unneeded semicolon
ASoC: fsl_sai: Implement set_bclk_ratio
ASoC: dmaengine: Replace strncpy() with strscpy_pad() for pcm->name
ASoC: wcd9335: remove redundant use of ret variable
ALSA: firewire-tascam: check intermediate state of clock status and retry
ALSA: firewire-tascam: handle error code when getting current source of clock
ASoC: hdmi-codec: Add an op to set callback function for plug event
ASoC: rt5677: keep analog power register at SND_SOC_BIAS_OFF
ASoC: rt5677: Remove magic number register writes
ASoC: soc-core: self contained soc_unbind_aux_dev()
ASoC: soc-core: add soc_unbind_aux_dev()
ASoC: soc-core: self contained soc_bind_aux_dev()
ASoC: soc-core: move soc_probe_link_dais() next to soc_remove_link_dais()
ASoC: soc-core: self contained soc_probe_link_dais()
ASoC: soc-core: add new soc_link_init()
ASoC: soc-core: move soc_probe_dai() next to soc_remove_dai()
ASoC: soc-core: self contained soc_remove_link_dais()
ASoC: soc-core: self contained soc_remove_link_components()
ASoC: soc-core: self contained soc_probe_link_components()
ASoC: rt1308: make array pd static const, makes object smaller
...
Diffstat (limited to 'sound/soc/soc-dai.c')
-rw-r--r-- | sound/soc/soc-dai.c | 407 |
1 files changed, 407 insertions, 0 deletions
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c new file mode 100644 index 000000000000..1c7f63871c1d --- /dev/null +++ b/sound/soc/soc-dai.c @@ -0,0 +1,407 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// soc-dai.c +// +// Copyright (C) 2019 Renesas Electronics Corp. +// Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> +// + +#include <sound/soc.h> +#include <sound/soc-dai.h> + +/** + * snd_soc_dai_set_sysclk - configure DAI system or master clock. + * @dai: DAI + * @clk_id: DAI specific clock ID + * @freq: new clock frequency in Hz + * @dir: new clock direction - input/output. + * + * Configures the DAI master (MCLK) or system (SYSCLK) clocking. + */ +int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + if (dai->driver->ops->set_sysclk) + return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir); + + return snd_soc_component_set_sysclk(dai->component, clk_id, 0, + freq, dir); +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); + +/** + * snd_soc_dai_set_clkdiv - configure DAI clock dividers. + * @dai: DAI + * @div_id: DAI specific clock divider ID + * @div: new clock divisor. + * + * Configures the clock dividers. This is used to derive the best DAI bit and + * frame clocks from the system or master clock. It's best to set the DAI bit + * and frame clocks as low as possible to save system power. + */ +int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, + int div_id, int div) +{ + if (dai->driver->ops->set_clkdiv) + return dai->driver->ops->set_clkdiv(dai, div_id, div); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); + +/** + * snd_soc_dai_set_pll - configure DAI PLL. + * @dai: DAI + * @pll_id: DAI specific PLL ID + * @source: DAI specific source for the PLL + * @freq_in: PLL input clock frequency in Hz + * @freq_out: requested PLL output clock frequency in Hz + * + * Configures and enables PLL to generate output clock based on input clock. + */ +int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + if (dai->driver->ops->set_pll) + return dai->driver->ops->set_pll(dai, pll_id, source, + freq_in, freq_out); + + return snd_soc_component_set_pll(dai->component, pll_id, source, + freq_in, freq_out); +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); + +/** + * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. + * @dai: DAI + * @ratio: Ratio of BCLK to Sample rate. + * + * Configures the DAI for a preset BCLK to sample rate ratio. + */ +int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + if (dai->driver->ops->set_bclk_ratio) + return dai->driver->ops->set_bclk_ratio(dai, ratio); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio); + +/** + * snd_soc_dai_set_fmt - configure DAI hardware audio format. + * @dai: DAI + * @fmt: SND_SOC_DAIFMT_* format value. + * + * Configures the DAI hardware format and clocking. + */ +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + if (dai->driver->ops->set_fmt == NULL) + return -ENOTSUPP; + return dai->driver->ops->set_fmt(dai, fmt); +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); + +/** + * snd_soc_xlate_tdm_slot - generate tx/rx slot mask. + * @slots: Number of slots in use. + * @tx_mask: bitmask representing active TX slots. + * @rx_mask: bitmask representing active RX slots. + * + * Generates the TDM tx and rx slot default masks for DAI. + */ +static int snd_soc_xlate_tdm_slot_mask(unsigned int slots, + unsigned int *tx_mask, + unsigned int *rx_mask) +{ + if (*tx_mask || *rx_mask) + return 0; + + if (!slots) + return -EINVAL; + + *tx_mask = (1 << slots) - 1; + *rx_mask = (1 << slots) - 1; + + return 0; +} + +/** + * snd_soc_dai_set_tdm_slot() - Configures a DAI for TDM operation + * @dai: The DAI to configure + * @tx_mask: bitmask representing active TX slots. + * @rx_mask: bitmask representing active RX slots. + * @slots: Number of slots in use. + * @slot_width: Width in bits for each slot. + * + * This function configures the specified DAI for TDM operation. @slot contains + * the total number of slots of the TDM stream and @slot_with the width of each + * slot in bit clock cycles. @tx_mask and @rx_mask are bitmasks specifying the + * active slots of the TDM stream for the specified DAI, i.e. which slots the + * DAI should write to or read from. If a bit is set the corresponding slot is + * active, if a bit is cleared the corresponding slot is inactive. Bit 0 maps to + * the first slot, bit 1 to the second slot and so on. The first active slot + * maps to the first channel of the DAI, the second active slot to the second + * channel and so on. + * + * TDM mode can be disabled by passing 0 for @slots. In this case @tx_mask, + * @rx_mask and @slot_width will be ignored. + * + * Returns 0 on success, a negative error code otherwise. + */ +int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + if (dai->driver->ops->xlate_tdm_slot_mask) + dai->driver->ops->xlate_tdm_slot_mask(slots, + &tx_mask, &rx_mask); + else + snd_soc_xlate_tdm_slot_mask(slots, &tx_mask, &rx_mask); + + dai->tx_mask = tx_mask; + dai->rx_mask = rx_mask; + + if (dai->driver->ops->set_tdm_slot) + return dai->driver->ops->set_tdm_slot(dai, tx_mask, rx_mask, + slots, slot_width); + else + return -ENOTSUPP; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); + +/** + * snd_soc_dai_set_channel_map - configure DAI audio channel map + * @dai: DAI + * @tx_num: how many TX channels + * @tx_slot: pointer to an array which imply the TX slot number channel + * 0~num-1 uses + * @rx_num: how many RX channels + * @rx_slot: pointer to an array which imply the RX slot number channel + * 0~num-1 uses + * + * configure the relationship between channel number and TDM slot number. + */ +int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + if (dai->driver->ops->set_channel_map) + return dai->driver->ops->set_channel_map(dai, tx_num, tx_slot, + rx_num, rx_slot); + else + return -ENOTSUPP; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map); + +/** + * snd_soc_dai_get_channel_map - Get DAI audio channel map + * @dai: DAI + * @tx_num: how many TX channels + * @tx_slot: pointer to an array which imply the TX slot number channel + * 0~num-1 uses + * @rx_num: how many RX channels + * @rx_slot: pointer to an array which imply the RX slot number channel + * 0~num-1 uses + */ +int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai, + unsigned int *tx_num, unsigned int *tx_slot, + unsigned int *rx_num, unsigned int *rx_slot) +{ + if (dai->driver->ops->get_channel_map) + return dai->driver->ops->get_channel_map(dai, tx_num, tx_slot, + rx_num, rx_slot); + else + return -ENOTSUPP; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_get_channel_map); + +/** + * snd_soc_dai_set_tristate - configure DAI system or master clock. + * @dai: DAI + * @tristate: tristate enable + * + * Tristates the DAI so that others can use it. + */ +int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + if (dai->driver->ops->set_tristate) + return dai->driver->ops->set_tristate(dai, tristate); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); + +/** + * snd_soc_dai_digital_mute - configure DAI system or master clock. + * @dai: DAI + * @mute: mute enable + * @direction: stream to mute + * + * Mutes the DAI DAC. + */ +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + if (dai->driver->ops->mute_stream) + return dai->driver->ops->mute_stream(dai, mute, direction); + else if (direction == SNDRV_PCM_STREAM_PLAYBACK && + dai->driver->ops->digital_mute) + return dai->driver->ops->digital_mute(dai, mute); + else + return -ENOTSUPP; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); + +int snd_soc_dai_hw_params(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret; + + /* perform any topology hw_params fixups before DAI */ + if (rtd->dai_link->be_hw_params_fixup) { + ret = rtd->dai_link->be_hw_params_fixup(rtd, params); + if (ret < 0) { + dev_err(rtd->dev, + "ASoC: hw_params topology fixup failed %d\n", + ret); + return ret; + } + } + + if (dai->driver->ops->hw_params) { + ret = dai->driver->ops->hw_params(substream, params, dai); + if (ret < 0) { + dev_err(dai->dev, "ASoC: can't set %s hw params: %d\n", + dai->name, ret); + return ret; + } + } + + return 0; +} + +void snd_soc_dai_hw_free(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream) +{ + if (dai->driver->ops->hw_free) + dai->driver->ops->hw_free(substream, dai); +} + +int snd_soc_dai_startup(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream) +{ + int ret = 0; + + if (dai->driver->ops->startup) + ret = dai->driver->ops->startup(substream, dai); + + return ret; +} + +void snd_soc_dai_shutdown(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream) +{ + if (dai->driver->ops->shutdown) + dai->driver->ops->shutdown(substream, dai); +} + +int snd_soc_dai_prepare(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream) +{ + int ret = 0; + + if (dai->driver->ops->prepare) + ret = dai->driver->ops->prepare(substream, dai); + + return ret; +} + +int snd_soc_dai_trigger(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream, + int cmd) +{ + int ret = 0; + + if (dai->driver->ops->trigger) + ret = dai->driver->ops->trigger(substream, cmd, dai); + + return ret; +} + +int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream, + int cmd) +{ + int ret = 0; + + if (dai->driver->ops->bespoke_trigger) + ret = dai->driver->ops->bespoke_trigger(substream, cmd, dai); + + return ret; +} + +snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream) +{ + int delay = 0; + + if (dai->driver->ops->delay) + delay = dai->driver->ops->delay(substream, dai); + + return delay; +} + +void snd_soc_dai_suspend(struct snd_soc_dai *dai) +{ + if (dai->driver->suspend) + dai->driver->suspend(dai); +} + +void snd_soc_dai_resume(struct snd_soc_dai *dai) +{ + if (dai->driver->resume) + dai->driver->resume(dai); +} + +int snd_soc_dai_probe(struct snd_soc_dai *dai) +{ + if (dai->driver->probe) + return dai->driver->probe(dai); + return 0; +} + +int snd_soc_dai_remove(struct snd_soc_dai *dai) +{ + if (dai->driver->remove) + return dai->driver->remove(dai); + return 0; +} + +int snd_soc_dai_compress_new(struct snd_soc_dai *dai, + struct snd_soc_pcm_runtime *rtd, int num) +{ + if (dai->driver->compress_new) + return dai->driver->compress_new(rtd, num); + return -ENOTSUPP; +} + +/* + * snd_soc_dai_stream_valid() - check if a DAI supports the given stream + * + * Returns true if the DAI supports the indicated stream type. + */ +bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int dir) +{ + struct snd_soc_pcm_stream *stream; + + if (dir == SNDRV_PCM_STREAM_PLAYBACK) + stream = &dai->driver->playback; + else + stream = &dai->driver->capture; + + /* If the codec specifies any channels at all, it supports the stream */ + return stream->channels_min; +} |