[alsa-devel] [PATCH 00/16] ASoC: add soc-dai.c
Hi Mark
I want to cleanup ALSA SoC. It will do nothing from "technical" point of view, but, will makes code readable.
I already have many cleanup patches in my local branch. This is first patch-set for it.
Kuninori Morimoto (16): ASoC: add soc-dai.c ASoC: soc-dai: mv soc_dai_hw_params() to soc-dai ASoC: soc-dai: add snd_soc_dai_hw_free() ASoC: soc-dai: add snd_soc_dai_startup() ASoC: soc-dai: add snd_soc_dai_shutdown() ASoC: soc-dai: add snd_soc_dai_prepare() ASoC: soc-dai: add snd_soc_dai_trigger() ASoC: soc-dai: add snd_soc_dai_bespoke_trigger() ASoC: soc-dai: add snd_soc_dai_delay() ASoC: soc-dai: add snd_soc_dai_suspend() ASoC: soc-dai: add snd_soc_dai_resume() ASoC: soc-dai: add snd_soc_dai_probe() ASoC: soc-dai: add snd_soc_dai_remove() ASoC: soc-dai: add snd_soc_dai_compress_new() ASoC: soc-dai: move snd_soc_dai_stream_valid() to soc-dai.c ASoC: soc-core: use existing snd_soc_dai_digital_mute()
include/sound/soc-dai.h | 25 +++ include/sound/soc.h | 4 - sound/soc/Makefile | 2 +- sound/soc/soc-compress.c | 9 +- sound/soc/soc-core.c | 316 ++++-------------------------------- sound/soc/soc-dai.c | 407 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/soc-dapm.c | 46 ++---- sound/soc/soc-pcm.c | 200 +++++++---------------- 8 files changed, 548 insertions(+), 461 deletions(-) create mode 100644 sound/soc/soc-dai.c
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ALSA SoC has many snd_soc_dai_xxx() function which is using dai->driver->ops->xxx. But, some of them are implemented as snd_soc_dai_xxx(), but others are directly using dai->driver->ops->xxx. Because of it, the code is not easy to read.
This patch creats new soc-dai.c and moves snd_soc_dai_xxx() functions into it. One exception is snd_soc_dai_is_dummy() which is based on soc-utils local variable. We need to keep it as-is there.
Others which is directly using dai->driver->ops->xxx will be implemented at soc-dai.c by incremental patches.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/Makefile | 2 +- sound/soc/soc-core.c | 243 ------------------------------------------------ sound/soc/soc-dai.c | 254 +++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 255 insertions(+), 244 deletions(-) create mode 100644 sound/soc/soc-dai.c
diff --git a/sound/soc/Makefile b/sound/soc/Makefile index d90ce8a..919c3c0 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ # SPDX-License-Identifier: GPL-2.0 -snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-utils.o +snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-utils.o soc-dai.o snd-soc-core-objs += soc-pcm.o soc-io.o soc-devres.o soc-ops.o snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fd6eaae..6e8c5c8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2395,26 +2395,6 @@ int snd_soc_add_dai_controls(struct snd_soc_dai *dai, EXPORT_SYMBOL_GPL(snd_soc_add_dai_controls);
/** - * 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_component_set_sysclk - configure COMPONENT system or master clock. * @component: COMPONENT * @clk_id: DAI specific clock ID @@ -2436,48 +2416,6 @@ int snd_soc_component_set_sysclk(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(snd_soc_component_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_component_set_pll - configure component PLL. * @component: COMPONENT @@ -2500,187 +2438,6 @@ int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id, } EXPORT_SYMBOL_GPL(snd_soc_component_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); - static int snd_soc_bind_card(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c new file mode 100644 index 0000000..a1009ea --- /dev/null +++ b/sound/soc/soc-dai.c @@ -0,0 +1,254 @@ +// 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);
The patch
ASoC: add soc-dai.c
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4
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 06f6e1d41427f394ad3f67ecf06efcd28a46932c Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Date: Mon, 22 Jul 2019 10:32:12 +0900 Subject: [PATCH] ASoC: add soc-dai.c
Current ALSA SoC has many snd_soc_dai_xxx() function which is using dai->driver->ops->xxx. But, some of them are implemented as snd_soc_dai_xxx(), but others are directly using dai->driver->ops->xxx. Because of it, the code is not easy to read.
This patch creats new soc-dai.c and moves snd_soc_dai_xxx() functions into it. One exception is snd_soc_dai_is_dummy() which is based on soc-utils local variable. We need to keep it as-is there.
Others which is directly using dai->driver->ops->xxx will be implemented at soc-dai.c by incremental patches.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871ryij1r6.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown broonie@kernel.org --- sound/soc/Makefile | 2 +- sound/soc/soc-core.c | 243 ----------------------------------------- sound/soc/soc-dai.c | 254 +++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 255 insertions(+), 244 deletions(-) create mode 100644 sound/soc/soc-dai.c
diff --git a/sound/soc/Makefile b/sound/soc/Makefile index d90ce8a32887..919c3c027c62 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ # SPDX-License-Identifier: GPL-2.0 -snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-utils.o +snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-utils.o soc-dai.o snd-soc-core-objs += soc-pcm.o soc-io.o soc-devres.o soc-ops.o snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fd6eaae6c0ed..6e8c5c8eeaec 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2394,26 +2394,6 @@ int snd_soc_add_dai_controls(struct snd_soc_dai *dai, } EXPORT_SYMBOL_GPL(snd_soc_add_dai_controls);
-/** - * 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_component_set_sysclk - configure COMPONENT system or master clock. * @component: COMPONENT @@ -2436,48 +2416,6 @@ int snd_soc_component_set_sysclk(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(snd_soc_component_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_component_set_pll - configure component PLL. * @component: COMPONENT @@ -2500,187 +2438,6 @@ int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id, } EXPORT_SYMBOL_GPL(snd_soc_component_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); - static int snd_soc_bind_card(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c new file mode 100644 index 000000000000..a1009ead40de --- /dev/null +++ b/sound/soc/soc-dai.c @@ -0,0 +1,254 @@ +// 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);
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Sometimes ALSA SoC naming is very random. Current soc_dai_hw_params() should use snd_soc_dai_xxx() style. And then, 1st parameter should be dai. Otherwise it is confusable. - soc_dai_hw_params(..., dai); + snd_soc_dai_hw_params(dai, ...);
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- include/sound/soc-dai.h | 4 ++++ include/sound/soc.h | 4 ---- sound/soc/soc-dai.c | 30 ++++++++++++++++++++++++++++++ sound/soc/soc-dapm.c | 4 ++-- sound/soc/soc-pcm.c | 35 +++-------------------------------- 5 files changed, 39 insertions(+), 38 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index f5d7004..3773262 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -145,6 +145,10 @@ int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai,
int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
+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_dai_ops { /* * DAI clocking configuration, all optional. diff --git a/include/sound/soc.h b/include/sound/soc.h index 4e80712..d770606 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -505,10 +505,6 @@ int snd_soc_params_to_bclk(struct snd_pcm_hw_params *parms); int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, const struct snd_pcm_hardware *hw);
-int soc_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai); - /* Jack reporting */ int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type, struct snd_soc_jack *jack, struct snd_soc_jack_pin *pins, diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index a1009ea..f883d27 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -252,3 +252,33 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, 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; +} diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f013b24..8fc6a01 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3839,7 +3839,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, } } source->active++; - ret = soc_dai_hw_params(&substream, params, source); + ret = snd_soc_dai_hw_params(source, &substream, params); if (ret < 0) goto out;
@@ -3861,7 +3861,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, } } sink->active++; - ret = soc_dai_hw_params(&substream, params, sink); + ret = snd_soc_dai_hw_params(sink, &substream, params); if (ret < 0) goto out;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 4878d22..420cc94 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -877,36 +877,6 @@ static void soc_pcm_codec_params_fixup(struct snd_pcm_hw_params *params, interval->max = channels; }
-int soc_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - 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; -} - static int soc_pcm_components_hw_free(struct snd_pcm_substream *substream, struct snd_soc_component *last) { @@ -989,7 +959,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, soc_pcm_codec_params_fixup(&codec_params, codec_dai->rx_mask);
- ret = soc_dai_hw_params(substream, &codec_params, codec_dai); + ret = snd_soc_dai_hw_params(codec_dai, substream, + &codec_params); if(ret < 0) goto codec_err;
@@ -1001,7 +972,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, snd_soc_dapm_update_dai(substream, &codec_params, codec_dai); }
- ret = soc_dai_hw_params(substream, params, cpu_dai); + ret = snd_soc_dai_hw_params(cpu_dai, substream, params); if (ret < 0) goto interface_err;
The patch
ASoC: soc-dai: mv soc_dai_hw_params() to soc-dai
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4
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 aa6166c2ac28392d64f2d8b3acfb56c8fe657147 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Date: Mon, 22 Jul 2019 10:33:04 +0900 Subject: [PATCH] ASoC: soc-dai: mv soc_dai_hw_params() to soc-dai
Sometimes ALSA SoC naming is very random. Current soc_dai_hw_params() should use snd_soc_dai_xxx() style. And then, 1st parameter should be dai. Otherwise it is confusable. - soc_dai_hw_params(..., dai); + snd_soc_dai_hw_params(dai, ...);
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87zhl6hn5b.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown broonie@kernel.org --- include/sound/soc-dai.h | 4 ++++ include/sound/soc.h | 4 ---- sound/soc/soc-dai.c | 30 ++++++++++++++++++++++++++++++ sound/soc/soc-dapm.c | 4 ++-- sound/soc/soc-pcm.c | 35 +++-------------------------------- 5 files changed, 39 insertions(+), 38 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index f5d70041108f..3773262a1b77 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -145,6 +145,10 @@ int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai,
int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
+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_dai_ops { /* * DAI clocking configuration, all optional. diff --git a/include/sound/soc.h b/include/sound/soc.h index 4e8071269639..d770606732cd 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -505,10 +505,6 @@ int snd_soc_params_to_bclk(struct snd_pcm_hw_params *parms); int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, const struct snd_pcm_hardware *hw);
-int soc_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai); - /* Jack reporting */ int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type, struct snd_soc_jack *jack, struct snd_soc_jack_pin *pins, diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index a1009ead40de..f883d27d136f 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -252,3 +252,33 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, 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; +} diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f013b24c050a..8fc6a01f5d8b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3839,7 +3839,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, } } source->active++; - ret = soc_dai_hw_params(&substream, params, source); + ret = snd_soc_dai_hw_params(source, &substream, params); if (ret < 0) goto out;
@@ -3861,7 +3861,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, } } sink->active++; - ret = soc_dai_hw_params(&substream, params, sink); + ret = snd_soc_dai_hw_params(sink, &substream, params); if (ret < 0) goto out;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 4878d22ebd8c..420cc94e0a46 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -877,36 +877,6 @@ static void soc_pcm_codec_params_fixup(struct snd_pcm_hw_params *params, interval->max = channels; }
-int soc_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - 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; -} - static int soc_pcm_components_hw_free(struct snd_pcm_substream *substream, struct snd_soc_component *last) { @@ -989,7 +959,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, soc_pcm_codec_params_fixup(&codec_params, codec_dai->rx_mask);
- ret = soc_dai_hw_params(substream, &codec_params, codec_dai); + ret = snd_soc_dai_hw_params(codec_dai, substream, + &codec_params); if(ret < 0) goto codec_err;
@@ -1001,7 +972,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, snd_soc_dapm_update_dai(substream, &codec_params, codec_dai); }
- ret = soc_dai_hw_params(substream, params, cpu_dai); + ret = snd_soc_dai_hw_params(cpu_dai, substream, params); if (ret < 0) goto interface_err;
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_hw_free() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-dai.c | 7 +++++++ sound/soc/soc-dapm.c | 7 ++----- sound/soc/soc-pcm.c | 12 ++++-------- 4 files changed, 15 insertions(+), 13 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 3773262..5222b6a 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -148,6 +148,8 @@ int snd_soc_dai_is_dummy(struct snd_soc_dai *dai); int snd_soc_dai_hw_params(struct snd_soc_dai *dai, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params); +void snd_soc_dai_hw_free(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index f883d27..39a685e 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -282,3 +282,10 @@ int snd_soc_dai_hw_params(struct snd_soc_dai *dai,
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); +} diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8fc6a01..0783b05 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3898,9 +3898,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, snd_soc_dapm_widget_for_each_source_path(w, path) { source = path->source->priv;
- if (source->driver->ops->hw_free) - source->driver->ops->hw_free(&substream, - source); + snd_soc_dai_hw_free(source, &substream);
source->active--; if (source->driver->ops->shutdown) @@ -3912,8 +3910,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, snd_soc_dapm_widget_for_each_sink_path(w, path) { sink = path->sink->priv;
- if (sink->driver->ops->hw_free) - sink->driver->ops->hw_free(&substream, sink); + snd_soc_dai_hw_free(sink, &substream);
sink->active--; if (sink->driver->ops->shutdown) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 420cc94..58fc4e9 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1011,8 +1011,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, component_err: soc_pcm_components_hw_free(substream, component);
- if (cpu_dai->driver->ops->hw_free) - cpu_dai->driver->ops->hw_free(substream, cpu_dai); + snd_soc_dai_hw_free(cpu_dai, substream); cpu_dai->rate = 0;
interface_err: @@ -1023,8 +1022,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) continue;
- if (codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); + snd_soc_dai_hw_free(codec_dai, substream); codec_dai->rate = 0; }
@@ -1083,12 +1081,10 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) continue;
- if (codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); + snd_soc_dai_hw_free(codec_dai, substream); }
- if (cpu_dai->driver->ops->hw_free) - cpu_dai->driver->ops->hw_free(substream, cpu_dai); + snd_soc_dai_hw_free(cpu_dai, substream);
mutex_unlock(&rtd->pcm_mutex); return 0;
The patch
ASoC: soc-dai: add snd_soc_dai_hw_free()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4
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 846faaed9df7899e74311db3aec0a41a2f6bc345 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Date: Mon, 22 Jul 2019 10:33:19 +0900 Subject: [PATCH] ASoC: soc-dai: add snd_soc_dai_hw_free()
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_hw_free() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87y30qhn4w.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown broonie@kernel.org --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-dai.c | 7 +++++++ sound/soc/soc-dapm.c | 7 ++----- sound/soc/soc-pcm.c | 12 ++++-------- 4 files changed, 15 insertions(+), 13 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 3773262a1b77..5222b6a758f2 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -148,6 +148,8 @@ int snd_soc_dai_is_dummy(struct snd_soc_dai *dai); int snd_soc_dai_hw_params(struct snd_soc_dai *dai, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params); +void snd_soc_dai_hw_free(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index f883d27d136f..39a685e6acd5 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -282,3 +282,10 @@ int snd_soc_dai_hw_params(struct snd_soc_dai *dai,
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); +} diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8fc6a01f5d8b..0783b05133ad 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3898,9 +3898,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, snd_soc_dapm_widget_for_each_source_path(w, path) { source = path->source->priv;
- if (source->driver->ops->hw_free) - source->driver->ops->hw_free(&substream, - source); + snd_soc_dai_hw_free(source, &substream);
source->active--; if (source->driver->ops->shutdown) @@ -3912,8 +3910,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, snd_soc_dapm_widget_for_each_sink_path(w, path) { sink = path->sink->priv;
- if (sink->driver->ops->hw_free) - sink->driver->ops->hw_free(&substream, sink); + snd_soc_dai_hw_free(sink, &substream);
sink->active--; if (sink->driver->ops->shutdown) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 420cc94e0a46..58fc4e98ab59 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1011,8 +1011,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, component_err: soc_pcm_components_hw_free(substream, component);
- if (cpu_dai->driver->ops->hw_free) - cpu_dai->driver->ops->hw_free(substream, cpu_dai); + snd_soc_dai_hw_free(cpu_dai, substream); cpu_dai->rate = 0;
interface_err: @@ -1023,8 +1022,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) continue;
- if (codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); + snd_soc_dai_hw_free(codec_dai, substream); codec_dai->rate = 0; }
@@ -1083,12 +1081,10 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) continue;
- if (codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); + snd_soc_dai_hw_free(codec_dai, substream); }
- if (cpu_dai->driver->ops->hw_free) - cpu_dai->driver->ops->hw_free(substream, cpu_dai); + snd_soc_dai_hw_free(cpu_dai, substream);
mutex_unlock(&rtd->pcm_mutex); return 0;
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_startup() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-dai.c | 11 +++++++++++ sound/soc/soc-dapm.c | 28 ++++++++++------------------ sound/soc/soc-pcm.c | 27 +++++++++++---------------- 4 files changed, 34 insertions(+), 34 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 5222b6a..0d16c5b 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -150,6 +150,8 @@ int snd_soc_dai_hw_params(struct snd_soc_dai *dai, struct snd_pcm_hw_params *params); void snd_soc_dai_hw_free(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); +int snd_soc_dai_startup(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 39a685e..6e19663 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -289,3 +289,14 @@ void snd_soc_dai_hw_free(struct snd_soc_dai *dai, 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; +} diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0783b05..71bfd04 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3828,15 +3828,11 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, snd_soc_dapm_widget_for_each_source_path(w, path) { source = path->source->priv;
- if (source->driver->ops->startup) { - ret = source->driver->ops->startup(&substream, - source); - if (ret < 0) { - dev_err(source->dev, - "ASoC: startup() failed: %d\n", - ret); - goto out; - } + ret = snd_soc_dai_startup(source, &substream); + if (ret < 0) { + dev_err(source->dev, + "ASoC: startup() failed: %d\n", ret); + goto out; } source->active++; ret = snd_soc_dai_hw_params(source, &substream, params); @@ -3850,15 +3846,11 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, snd_soc_dapm_widget_for_each_sink_path(w, path) { sink = path->sink->priv;
- if (sink->driver->ops->startup) { - ret = sink->driver->ops->startup(&substream, - sink); - if (ret < 0) { - dev_err(sink->dev, - "ASoC: startup() failed: %d\n", - ret); - goto out; - } + ret = snd_soc_dai_startup(sink, &substream); + if (ret < 0) { + dev_err(sink->dev, + "ASoC: startup() failed: %d\n", ret); + goto out; } sink->active++; ret = snd_soc_dai_hw_params(sink, &substream, params); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 58fc4e9..9c8713a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -535,13 +535,11 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
/* startup the audio subsystem */ - if (cpu_dai->driver->ops->startup) { - ret = cpu_dai->driver->ops->startup(substream, cpu_dai); - if (ret < 0) { - dev_err(cpu_dai->dev, "ASoC: can't open interface" - " %s: %d\n", cpu_dai->name, ret); - goto out; - } + ret = snd_soc_dai_startup(cpu_dai, substream); + if (ret < 0) { + dev_err(cpu_dai->dev, "ASoC: can't open interface %s: %d\n", + cpu_dai->name, ret); + goto out; }
ret = soc_pcm_components_open(substream, &component); @@ -549,15 +547,12 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto component_err;
for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->ops->startup) { - ret = codec_dai->driver->ops->startup(substream, - codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, - "ASoC: can't open codec %s: %d\n", - codec_dai->name, ret); - goto codec_dai_err; - } + ret = snd_soc_dai_startup(codec_dai, substream); + if (ret < 0) { + dev_err(codec_dai->dev, + "ASoC: can't open codec %s: %d\n", + codec_dai->name, ret); + goto codec_dai_err; }
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
The patch
ASoC: soc-dai: add snd_soc_dai_startup()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4
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 5a52a04531486e2ab069b7882432c8b266db36e6 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Date: Mon, 22 Jul 2019 10:33:32 +0900 Subject: [PATCH] ASoC: soc-dai: add snd_soc_dai_startup()
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_startup() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87wogahn4i.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown broonie@kernel.org --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-dai.c | 11 +++++++++++ sound/soc/soc-dapm.c | 28 ++++++++++------------------ sound/soc/soc-pcm.c | 27 +++++++++++---------------- 4 files changed, 34 insertions(+), 34 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 5222b6a758f2..0d16c5bb20bb 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -150,6 +150,8 @@ int snd_soc_dai_hw_params(struct snd_soc_dai *dai, struct snd_pcm_hw_params *params); void snd_soc_dai_hw_free(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); +int snd_soc_dai_startup(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 39a685e6acd5..6e196636e42f 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -289,3 +289,14 @@ void snd_soc_dai_hw_free(struct snd_soc_dai *dai, 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; +} diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0783b05133ad..71bfd049480a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3828,15 +3828,11 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, snd_soc_dapm_widget_for_each_source_path(w, path) { source = path->source->priv;
- if (source->driver->ops->startup) { - ret = source->driver->ops->startup(&substream, - source); - if (ret < 0) { - dev_err(source->dev, - "ASoC: startup() failed: %d\n", - ret); - goto out; - } + ret = snd_soc_dai_startup(source, &substream); + if (ret < 0) { + dev_err(source->dev, + "ASoC: startup() failed: %d\n", ret); + goto out; } source->active++; ret = snd_soc_dai_hw_params(source, &substream, params); @@ -3850,15 +3846,11 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, snd_soc_dapm_widget_for_each_sink_path(w, path) { sink = path->sink->priv;
- if (sink->driver->ops->startup) { - ret = sink->driver->ops->startup(&substream, - sink); - if (ret < 0) { - dev_err(sink->dev, - "ASoC: startup() failed: %d\n", - ret); - goto out; - } + ret = snd_soc_dai_startup(sink, &substream); + if (ret < 0) { + dev_err(sink->dev, + "ASoC: startup() failed: %d\n", ret); + goto out; } sink->active++; ret = snd_soc_dai_hw_params(sink, &substream, params); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 58fc4e98ab59..9c8713a3eef1 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -535,13 +535,11 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
/* startup the audio subsystem */ - if (cpu_dai->driver->ops->startup) { - ret = cpu_dai->driver->ops->startup(substream, cpu_dai); - if (ret < 0) { - dev_err(cpu_dai->dev, "ASoC: can't open interface" - " %s: %d\n", cpu_dai->name, ret); - goto out; - } + ret = snd_soc_dai_startup(cpu_dai, substream); + if (ret < 0) { + dev_err(cpu_dai->dev, "ASoC: can't open interface %s: %d\n", + cpu_dai->name, ret); + goto out; }
ret = soc_pcm_components_open(substream, &component); @@ -549,15 +547,12 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto component_err;
for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->ops->startup) { - ret = codec_dai->driver->ops->startup(substream, - codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, - "ASoC: can't open codec %s: %d\n", - codec_dai->name, ret); - goto codec_dai_err; - } + ret = snd_soc_dai_startup(codec_dai, substream); + if (ret < 0) { + dev_err(codec_dai->dev, + "ASoC: can't open codec %s: %d\n", + codec_dai->name, ret); + goto codec_dai_err; }
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_shutdown() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-dai.c | 7 +++++++ sound/soc/soc-dapm.c | 7 ++----- sound/soc/soc-pcm.c | 18 ++++++------------ 4 files changed, 17 insertions(+), 17 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 0d16c5b..32545d4 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -152,6 +152,8 @@ void snd_soc_dai_hw_free(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); int snd_soc_dai_startup(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); +void snd_soc_dai_shutdown(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 6e19663..67ff6cc 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -300,3 +300,10 @@ int snd_soc_dai_startup(struct snd_soc_dai *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); +} diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 71bfd04..1d04612 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3893,9 +3893,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, snd_soc_dai_hw_free(source, &substream);
source->active--; - if (source->driver->ops->shutdown) - source->driver->ops->shutdown(&substream, - source); + snd_soc_dai_shutdown(source, &substream); }
substream.stream = SNDRV_PCM_STREAM_PLAYBACK; @@ -3905,8 +3903,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, snd_soc_dai_hw_free(sink, &substream);
sink->active--; - if (sink->driver->ops->shutdown) - sink->driver->ops->shutdown(&substream, sink); + snd_soc_dai_shutdown(sink, &substream); } break;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 9c8713a..ed5ae23 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -641,16 +641,13 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) i = rtd->num_codecs;
codec_dai_err: - for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) { - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); - } + for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) + snd_soc_dai_shutdown(codec_dai, substream);
component_err: soc_pcm_components_close(substream, component);
- if (cpu_dai->driver->ops->shutdown) - cpu_dai->driver->ops->shutdown(substream, cpu_dai); + snd_soc_dai_shutdown(cpu_dai, substream); out: mutex_unlock(&rtd->pcm_mutex);
@@ -728,13 +725,10 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
snd_soc_dai_digital_mute(cpu_dai, 1, substream->stream);
- if (cpu_dai->driver->ops->shutdown) - cpu_dai->driver->ops->shutdown(substream, cpu_dai); + snd_soc_dai_shutdown(cpu_dai, substream);
- for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); - } + for_each_rtd_codec_dai(rtd, i, codec_dai) + snd_soc_dai_shutdown(codec_dai, substream);
if (rtd->dai_link->ops->shutdown) rtd->dai_link->ops->shutdown(substream);
The patch
ASoC: soc-dai: add snd_soc_dai_shutdown()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4
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 330fcb5135e0588b1ea3b0bbab587d1317c1cf7b Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Date: Mon, 22 Jul 2019 10:33:39 +0900 Subject: [PATCH] ASoC: soc-dai: add snd_soc_dai_shutdown()
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_shutdown() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87v9vuhn4b.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown broonie@kernel.org --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-dai.c | 7 +++++++ sound/soc/soc-dapm.c | 7 ++----- sound/soc/soc-pcm.c | 18 ++++++------------ 4 files changed, 17 insertions(+), 17 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 0d16c5bb20bb..32545d457b3d 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -152,6 +152,8 @@ void snd_soc_dai_hw_free(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); int snd_soc_dai_startup(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); +void snd_soc_dai_shutdown(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 6e196636e42f..67ff6cc1fe02 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -300,3 +300,10 @@ int snd_soc_dai_startup(struct snd_soc_dai *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); +} diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 71bfd049480a..1d04612601ad 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3893,9 +3893,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, snd_soc_dai_hw_free(source, &substream);
source->active--; - if (source->driver->ops->shutdown) - source->driver->ops->shutdown(&substream, - source); + snd_soc_dai_shutdown(source, &substream); }
substream.stream = SNDRV_PCM_STREAM_PLAYBACK; @@ -3905,8 +3903,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, snd_soc_dai_hw_free(sink, &substream);
sink->active--; - if (sink->driver->ops->shutdown) - sink->driver->ops->shutdown(&substream, sink); + snd_soc_dai_shutdown(sink, &substream); } break;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 9c8713a3eef1..ed5ae23c7104 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -641,16 +641,13 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) i = rtd->num_codecs;
codec_dai_err: - for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) { - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); - } + for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) + snd_soc_dai_shutdown(codec_dai, substream);
component_err: soc_pcm_components_close(substream, component);
- if (cpu_dai->driver->ops->shutdown) - cpu_dai->driver->ops->shutdown(substream, cpu_dai); + snd_soc_dai_shutdown(cpu_dai, substream); out: mutex_unlock(&rtd->pcm_mutex);
@@ -728,13 +725,10 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
snd_soc_dai_digital_mute(cpu_dai, 1, substream->stream);
- if (cpu_dai->driver->ops->shutdown) - cpu_dai->driver->ops->shutdown(substream, cpu_dai); + snd_soc_dai_shutdown(cpu_dai, substream);
- for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); - } + for_each_rtd_codec_dai(rtd, i, codec_dai) + snd_soc_dai_shutdown(codec_dai, substream);
if (rtd->dai_link->ops->shutdown) rtd->dai_link->ops->shutdown(substream);
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_prepare() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-dai.c | 11 +++++++++++ sound/soc/soc-pcm.c | 27 +++++++++++---------------- 3 files changed, 24 insertions(+), 16 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 32545d4..c7dff6a 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -154,6 +154,8 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); void snd_soc_dai_shutdown(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); +int snd_soc_dai_prepare(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 67ff6cc..cb81088 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -307,3 +307,14 @@ void snd_soc_dai_shutdown(struct snd_soc_dai *dai, 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; +} diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ed5ae23..d7611af 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -814,27 +814,22 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) }
for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->ops->prepare) { - ret = codec_dai->driver->ops->prepare(substream, - codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, - "ASoC: codec DAI prepare error: %d\n", - ret); - goto out; - } - } - } - - if (cpu_dai->driver->ops->prepare) { - ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); + ret = snd_soc_dai_prepare(codec_dai, substream); if (ret < 0) { - dev_err(cpu_dai->dev, - "ASoC: cpu DAI prepare error: %d\n", ret); + dev_err(codec_dai->dev, + "ASoC: codec DAI prepare error: %d\n", + ret); goto out; } }
+ ret = snd_soc_dai_prepare(cpu_dai, substream); + if (ret < 0) { + dev_err(cpu_dai->dev, + "ASoC: cpu DAI prepare error: %d\n", ret); + goto out; + } + /* cancel any delayed stream shutdown that is pending */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && rtd->pop_wait) {
The patch
ASoC: soc-dai: add snd_soc_dai_prepare()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4
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 4beb8e109d30d339d44308a767dd6f5614492f3e Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Date: Mon, 22 Jul 2019 10:33:45 +0900 Subject: [PATCH] ASoC: soc-dai: add snd_soc_dai_prepare()
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_prepare() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87tvbehn46.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown broonie@kernel.org --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-dai.c | 11 +++++++++++ sound/soc/soc-pcm.c | 27 +++++++++++---------------- 3 files changed, 24 insertions(+), 16 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 32545d457b3d..c7dff6a0b5b9 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -154,6 +154,8 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); void snd_soc_dai_shutdown(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); +int snd_soc_dai_prepare(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 67ff6cc1fe02..cb810888c563 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -307,3 +307,14 @@ void snd_soc_dai_shutdown(struct snd_soc_dai *dai, 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; +} diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ed5ae23c7104..d7611af90dce 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -814,27 +814,22 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) }
for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->ops->prepare) { - ret = codec_dai->driver->ops->prepare(substream, - codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, - "ASoC: codec DAI prepare error: %d\n", - ret); - goto out; - } - } - } - - if (cpu_dai->driver->ops->prepare) { - ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); + ret = snd_soc_dai_prepare(codec_dai, substream); if (ret < 0) { - dev_err(cpu_dai->dev, - "ASoC: cpu DAI prepare error: %d\n", ret); + dev_err(codec_dai->dev, + "ASoC: codec DAI prepare error: %d\n", + ret); goto out; } }
+ ret = snd_soc_dai_prepare(cpu_dai, substream); + if (ret < 0) { + dev_err(cpu_dai->dev, + "ASoC: cpu DAI prepare error: %d\n", ret); + goto out; + } + /* cancel any delayed stream shutdown that is pending */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && rtd->pop_wait) {
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_trigger() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-dai.c | 12 ++++++++++++ sound/soc/soc-pcm.c | 17 ++++++----------- 3 files changed, 20 insertions(+), 11 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index c7dff6a..72b8e76 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -156,6 +156,8 @@ void snd_soc_dai_shutdown(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); int snd_soc_dai_prepare(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); +int snd_soc_dai_trigger(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream, int cmd);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index cb81088..18c447e 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -318,3 +318,15 @@ int snd_soc_dai_prepare(struct snd_soc_dai *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; +} diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d7611af..a628b08 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1084,12 +1084,9 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) int i, ret;
for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->ops->trigger) { - ret = codec_dai->driver->ops->trigger(substream, - cmd, codec_dai); - if (ret < 0) - return ret; - } + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; }
for_each_rtdcom(rtd, rtdcom) { @@ -1104,11 +1101,9 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; }
- if (cpu_dai->driver->ops->trigger) { - ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); - if (ret < 0) - return ret; - } + snd_soc_dai_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret;
if (rtd->dai_link->ops->trigger) { ret = rtd->dai_link->ops->trigger(substream, cmd);
The patch
ASoC: soc-dai: add snd_soc_dai_trigger()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4
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 95aef35533844f35544851b0cdc1fc154b603307 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Date: Mon, 22 Jul 2019 10:33:51 +0900 Subject: [PATCH] ASoC: soc-dai: add snd_soc_dai_trigger()
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_trigger() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87sgqyhn40.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown broonie@kernel.org --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-dai.c | 12 ++++++++++++ sound/soc/soc-pcm.c | 17 ++++++----------- 3 files changed, 20 insertions(+), 11 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index c7dff6a0b5b9..72b8e76f1cc4 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -156,6 +156,8 @@ void snd_soc_dai_shutdown(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); int snd_soc_dai_prepare(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); +int snd_soc_dai_trigger(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream, int cmd);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index cb810888c563..18c447e169f6 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -318,3 +318,15 @@ int snd_soc_dai_prepare(struct snd_soc_dai *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; +} diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d7611af90dce..a628b08f966e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1084,12 +1084,9 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) int i, ret;
for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->ops->trigger) { - ret = codec_dai->driver->ops->trigger(substream, - cmd, codec_dai); - if (ret < 0) - return ret; - } + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; }
for_each_rtdcom(rtd, rtdcom) { @@ -1104,11 +1101,9 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; }
- if (cpu_dai->driver->ops->trigger) { - ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); - if (ret < 0) - return ret; - } + snd_soc_dai_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret;
if (rtd->dai_link->ops->trigger) { ret = rtd->dai_link->ops->trigger(substream, cmd);
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_bespoke_trigger() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-dai.c | 12 ++++++++++++ sound/soc/soc-pcm.c | 16 ++++++---------- 3 files changed, 20 insertions(+), 10 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 72b8e76..6a5566d 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -158,6 +158,8 @@ int snd_soc_dai_prepare(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); int snd_soc_dai_trigger(struct snd_soc_dai *dai, struct snd_pcm_substream *substream, int cmd); +int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream, int cmd);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 18c447e..6f466cf 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -330,3 +330,15 @@ int snd_soc_dai_trigger(struct snd_soc_dai *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; +} diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a628b08..a10627f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1123,19 +1123,15 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int i, ret;
for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->ops->bespoke_trigger) { - ret = codec_dai->driver->ops->bespoke_trigger(substream, - cmd, codec_dai); - if (ret < 0) - return ret; - } - } - - if (cpu_dai->driver->ops->bespoke_trigger) { - ret = cpu_dai->driver->ops->bespoke_trigger(substream, cmd, cpu_dai); + ret = snd_soc_dai_bespoke_trigger(codec_dai, substream, cmd); if (ret < 0) return ret; } + + snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret; + return 0; } /*
The patch
ASoC: soc-dai: add snd_soc_dai_bespoke_trigger()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4
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 5c0769af4caf8fbdad2e9c0051ab0081b8e22b0a Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Date: Mon, 22 Jul 2019 10:33:56 +0900 Subject: [PATCH] ASoC: soc-dai: add snd_soc_dai_bespoke_trigger()
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_bespoke_trigger() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87r26ihn3u.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown broonie@kernel.org --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-dai.c | 12 ++++++++++++ sound/soc/soc-pcm.c | 16 ++++++---------- 3 files changed, 20 insertions(+), 10 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 72b8e76f1cc4..6a5566d459ad 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -158,6 +158,8 @@ int snd_soc_dai_prepare(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); int snd_soc_dai_trigger(struct snd_soc_dai *dai, struct snd_pcm_substream *substream, int cmd); +int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream, int cmd);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 18c447e169f6..6f466cfcbeef 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -330,3 +330,15 @@ int snd_soc_dai_trigger(struct snd_soc_dai *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; +} diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a628b08f966e..a10627f1ceff 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1123,19 +1123,15 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int i, ret;
for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->ops->bespoke_trigger) { - ret = codec_dai->driver->ops->bespoke_trigger(substream, - cmd, codec_dai); - if (ret < 0) - return ret; - } - } - - if (cpu_dai->driver->ops->bespoke_trigger) { - ret = cpu_dai->driver->ops->bespoke_trigger(substream, cmd, cpu_dai); + ret = snd_soc_dai_bespoke_trigger(codec_dai, substream, cmd); if (ret < 0) return ret; } + + snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret; + return 0; } /*
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_bespoke_trigger() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-dai.c | 12 ++++++++++++ sound/soc/soc-pcm.c | 16 ++++++---------- 3 files changed, 20 insertions(+), 10 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 72b8e76..6a5566d 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -158,6 +158,8 @@ int snd_soc_dai_prepare(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); int snd_soc_dai_trigger(struct snd_soc_dai *dai, struct snd_pcm_substream *substream, int cmd); +int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream, int cmd);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 18c447e..6f466cf 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -330,3 +330,15 @@ int snd_soc_dai_trigger(struct snd_soc_dai *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; +} diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a628b08..a10627f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1123,19 +1123,15 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int i, ret;
for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->ops->bespoke_trigger) { - ret = codec_dai->driver->ops->bespoke_trigger(substream, - cmd, codec_dai); - if (ret < 0) - return ret; - } - } - - if (cpu_dai->driver->ops->bespoke_trigger) { - ret = cpu_dai->driver->ops->bespoke_trigger(substream, cmd, cpu_dai); + ret = snd_soc_dai_bespoke_trigger(codec_dai, substream, cmd); if (ret < 0) return ret; } + + snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret; + return 0; } /*
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_delay() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-dai.c | 11 +++++++++++ sound/soc/soc-pcm.c | 9 +++------ 3 files changed, 16 insertions(+), 6 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 6a5566d..7cfed30 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -160,6 +160,8 @@ int snd_soc_dai_trigger(struct snd_soc_dai *dai, struct snd_pcm_substream *substream, int cmd); int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai, struct snd_pcm_substream *substream, int cmd); +snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 6f466cf..5b5b979 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -342,3 +342,14 @@ int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *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; +} diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a10627f..f313772 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1169,14 +1169,11 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) /* base delay if assigned in pointer callback */ delay = runtime->delay;
- if (cpu_dai->driver->ops->delay) - delay += cpu_dai->driver->ops->delay(substream, cpu_dai); + delay += snd_soc_dai_delay(cpu_dai, substream);
for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->ops->delay) - codec_delay = max(codec_delay, - codec_dai->driver->ops->delay(substream, - codec_dai)); + codec_delay = max(codec_delay, + snd_soc_dai_delay(codec_dai, substream)); } delay += codec_delay;
The patch
ASoC: soc-dai: add snd_soc_dai_delay()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4
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 1dea80d4b2bd3b53c58f008ca2bcd73182583711 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Date: Mon, 22 Jul 2019 10:34:09 +0900 Subject: [PATCH] ASoC: soc-dai: add snd_soc_dai_delay()
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_delay() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87o91mhn3i.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown broonie@kernel.org --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-dai.c | 11 +++++++++++ sound/soc/soc-pcm.c | 9 +++------ 3 files changed, 16 insertions(+), 6 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 6a5566d459ad..7cfed3034511 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -160,6 +160,8 @@ int snd_soc_dai_trigger(struct snd_soc_dai *dai, struct snd_pcm_substream *substream, int cmd); int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai, struct snd_pcm_substream *substream, int cmd); +snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 6f466cfcbeef..5b5b979cd1f3 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -342,3 +342,14 @@ int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *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; +} diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a10627f1ceff..f3137723301c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1169,14 +1169,11 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) /* base delay if assigned in pointer callback */ delay = runtime->delay;
- if (cpu_dai->driver->ops->delay) - delay += cpu_dai->driver->ops->delay(substream, cpu_dai); + delay += snd_soc_dai_delay(cpu_dai, substream);
for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->ops->delay) - codec_delay = max(codec_delay, - codec_dai->driver->ops->delay(substream, - codec_dai)); + codec_delay = max(codec_delay, + snd_soc_dai_delay(codec_dai, substream)); } delay += codec_delay;
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ALSA SoC is directly using dai->driver->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_suspend() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- include/sound/soc-dai.h | 1 + sound/soc/soc-core.c | 8 ++++---- sound/soc/soc-dai.c | 6 ++++++ 3 files changed, 11 insertions(+), 4 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 7cfed30..6c5604a 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -162,6 +162,7 @@ int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai, struct snd_pcm_substream *substream, int cmd); snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); +void snd_soc_dai_suspend(struct snd_soc_dai *dai);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6e8c5c8..7493afb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -511,8 +511,8 @@ int snd_soc_suspend(struct device *dev) if (rtd->dai_link->ignore_suspend) continue;
- if (cpu_dai->driver->suspend && !cpu_dai->driver->bus_control) - cpu_dai->driver->suspend(cpu_dai); + if (!cpu_dai->driver->bus_control) + snd_soc_dai_suspend(cpu_dai); }
/* close any waiting streams */ @@ -584,8 +584,8 @@ int snd_soc_suspend(struct device *dev) if (rtd->dai_link->ignore_suspend) continue;
- if (cpu_dai->driver->suspend && cpu_dai->driver->bus_control) - cpu_dai->driver->suspend(cpu_dai); + if (cpu_dai->driver->bus_control) + snd_soc_dai_suspend(cpu_dai);
/* deactivate pins to sleep state */ pinctrl_pm_select_sleep_state(cpu_dai->dev); diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 5b5b979..3373598 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -353,3 +353,9 @@ snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai,
return delay; } + +void snd_soc_dai_suspend(struct snd_soc_dai *dai) +{ + if (dai->driver->suspend) + dai->driver->suspend(dai); +}
The patch
ASoC: soc-dai: add snd_soc_dai_suspend()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4
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 e0f2262292d0c8160cfd9a8c40425107fb65ab29 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Date: Mon, 22 Jul 2019 10:34:29 +0900 Subject: [PATCH] ASoC: soc-dai: add snd_soc_dai_suspend()
Current ALSA SoC is directly using dai->driver->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_suspend() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87muh6hn2x.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown broonie@kernel.org --- include/sound/soc-dai.h | 1 + sound/soc/soc-core.c | 8 ++++---- sound/soc/soc-dai.c | 6 ++++++ 3 files changed, 11 insertions(+), 4 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 7cfed3034511..6c5604a7dbc2 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -162,6 +162,7 @@ int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai, struct snd_pcm_substream *substream, int cmd); snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); +void snd_soc_dai_suspend(struct snd_soc_dai *dai);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6e8c5c8eeaec..7493afb2371c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -511,8 +511,8 @@ int snd_soc_suspend(struct device *dev) if (rtd->dai_link->ignore_suspend) continue;
- if (cpu_dai->driver->suspend && !cpu_dai->driver->bus_control) - cpu_dai->driver->suspend(cpu_dai); + if (!cpu_dai->driver->bus_control) + snd_soc_dai_suspend(cpu_dai); }
/* close any waiting streams */ @@ -584,8 +584,8 @@ int snd_soc_suspend(struct device *dev) if (rtd->dai_link->ignore_suspend) continue;
- if (cpu_dai->driver->suspend && cpu_dai->driver->bus_control) - cpu_dai->driver->suspend(cpu_dai); + if (cpu_dai->driver->bus_control) + snd_soc_dai_suspend(cpu_dai);
/* deactivate pins to sleep state */ pinctrl_pm_select_sleep_state(cpu_dai->dev); diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 5b5b979cd1f3..3373598e0682 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -353,3 +353,9 @@ snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai,
return delay; } + +void snd_soc_dai_suspend(struct snd_soc_dai *dai) +{ + if (dai->driver->suspend) + dai->driver->suspend(dai); +}
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ALSA SoC is directly using dai->driver->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_resume() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- include/sound/soc-dai.h | 1 + sound/soc/soc-core.c | 8 ++++---- sound/soc/soc-dai.c | 6 ++++++ 3 files changed, 11 insertions(+), 4 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 6c5604a..ed78e34 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -163,6 +163,7 @@ int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai, snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); void snd_soc_dai_suspend(struct snd_soc_dai *dai); +void snd_soc_dai_resume(struct snd_soc_dai *dai);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7493afb..5c02f90 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -631,8 +631,8 @@ static void soc_resume_deferred(struct work_struct *work) if (rtd->dai_link->ignore_suspend) continue;
- if (cpu_dai->driver->resume && cpu_dai->driver->bus_control) - cpu_dai->driver->resume(cpu_dai); + if (cpu_dai->driver->bus_control) + snd_soc_dai_resume(cpu_dai); }
for_each_card_components(card, component) { @@ -678,8 +678,8 @@ static void soc_resume_deferred(struct work_struct *work) if (rtd->dai_link->ignore_suspend) continue;
- if (cpu_dai->driver->resume && !cpu_dai->driver->bus_control) - cpu_dai->driver->resume(cpu_dai); + if (!cpu_dai->driver->bus_control) + snd_soc_dai_resume(cpu_dai); }
if (card->resume_post) diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 3373598..ddb6f21 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -359,3 +359,9 @@ 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); +}
The patch
ASoC: soc-dai: add snd_soc_dai_resume()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4
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 24b09d051164680f0a1d1910efe21ce36ad5c1ca Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Date: Mon, 22 Jul 2019 10:34:43 +0900 Subject: [PATCH] ASoC: soc-dai: add snd_soc_dai_resume()
Current ALSA SoC is directly using dai->driver->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_resume() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87lfwqhn2j.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown broonie@kernel.org --- include/sound/soc-dai.h | 1 + sound/soc/soc-core.c | 8 ++++---- sound/soc/soc-dai.c | 6 ++++++ 3 files changed, 11 insertions(+), 4 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 6c5604a7dbc2..ed78e34a814e 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -163,6 +163,7 @@ int snd_soc_dai_bespoke_trigger(struct snd_soc_dai *dai, snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); void snd_soc_dai_suspend(struct snd_soc_dai *dai); +void snd_soc_dai_resume(struct snd_soc_dai *dai);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7493afb2371c..5c02f90cea69 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -631,8 +631,8 @@ static void soc_resume_deferred(struct work_struct *work) if (rtd->dai_link->ignore_suspend) continue;
- if (cpu_dai->driver->resume && cpu_dai->driver->bus_control) - cpu_dai->driver->resume(cpu_dai); + if (cpu_dai->driver->bus_control) + snd_soc_dai_resume(cpu_dai); }
for_each_card_components(card, component) { @@ -678,8 +678,8 @@ static void soc_resume_deferred(struct work_struct *work) if (rtd->dai_link->ignore_suspend) continue;
- if (cpu_dai->driver->resume && !cpu_dai->driver->bus_control) - cpu_dai->driver->resume(cpu_dai); + if (!cpu_dai->driver->bus_control) + snd_soc_dai_resume(cpu_dai); }
if (card->resume_post) diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 3373598e0682..ddb6f217c0ed 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -359,3 +359,9 @@ 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); +}
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ALSA SoC is directly using dai->driver->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_probe() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- include/sound/soc-dai.h | 1 + sound/soc/soc-core.c | 15 +++++++-------- sound/soc/soc-dai.c | 7 +++++++ 3 files changed, 15 insertions(+), 8 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ed78e34..da8d8b8 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -164,6 +164,7 @@ snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); void snd_soc_dai_suspend(struct snd_soc_dai *dai); void snd_soc_dai_resume(struct snd_soc_dai *dai); +int snd_soc_dai_probe(struct snd_soc_dai *dai);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5c02f90..3e73468 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1434,18 +1434,17 @@ static int soc_probe_link_components(struct snd_soc_card *card,
static int soc_probe_dai(struct snd_soc_dai *dai, int order) { + int ret; + if (dai->probed || dai->driver->probe_order != order) return 0;
- if (dai->driver->probe) { - int ret = dai->driver->probe(dai); - - if (ret < 0) { - dev_err(dai->dev, "ASoC: failed to probe DAI %s: %d\n", - dai->name, ret); - return ret; - } + ret = snd_soc_dai_probe(dai); + if (ret < 0) { + dev_err(dai->dev, "ASoC: failed to probe DAI %s: %d\n", + dai->name, ret); + return ret; }
dai->probed = 1; diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index ddb6f21..55c1fac 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -365,3 +365,10 @@ 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; +}
The patch
ASoC: soc-dai: add snd_soc_dai_probe()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4
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 cfd9b5fbfe1e8763018aea2600aa0d6ff015ebfc Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Date: Mon, 22 Jul 2019 10:34:56 +0900 Subject: [PATCH] ASoC: soc-dai: add snd_soc_dai_probe()
Current ALSA SoC is directly using dai->driver->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_probe() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87k1cahn26.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown broonie@kernel.org --- include/sound/soc-dai.h | 1 + sound/soc/soc-core.c | 15 +++++++-------- sound/soc/soc-dai.c | 7 +++++++ 3 files changed, 15 insertions(+), 8 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ed78e34a814e..da8d8b889089 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -164,6 +164,7 @@ snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); void snd_soc_dai_suspend(struct snd_soc_dai *dai); void snd_soc_dai_resume(struct snd_soc_dai *dai); +int snd_soc_dai_probe(struct snd_soc_dai *dai);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5c02f90cea69..3e73468225f9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1434,18 +1434,17 @@ static int soc_probe_link_components(struct snd_soc_card *card,
static int soc_probe_dai(struct snd_soc_dai *dai, int order) { + int ret; + if (dai->probed || dai->driver->probe_order != order) return 0;
- if (dai->driver->probe) { - int ret = dai->driver->probe(dai); - - if (ret < 0) { - dev_err(dai->dev, "ASoC: failed to probe DAI %s: %d\n", - dai->name, ret); - return ret; - } + ret = snd_soc_dai_probe(dai); + if (ret < 0) { + dev_err(dai->dev, "ASoC: failed to probe DAI %s: %d\n", + dai->name, ret); + return ret; }
dai->probed = 1; diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index ddb6f217c0ed..55c1fac99613 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -365,3 +365,10 @@ 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; +}
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ALSA SoC is directly using dai->driver->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_remvoe() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- include/sound/soc-dai.h | 1 + sound/soc/soc-core.c | 13 ++++++------- sound/soc/soc-dai.c | 7 +++++++ 3 files changed, 14 insertions(+), 7 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index da8d8b8..2a11f17 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -165,6 +165,7 @@ snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai, void snd_soc_dai_suspend(struct snd_soc_dai *dai); void snd_soc_dai_resume(struct snd_soc_dai *dai); int snd_soc_dai_probe(struct snd_soc_dai *dai); +int snd_soc_dai_remove(struct snd_soc_dai *dai);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3e73468..727fd34 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -992,13 +992,12 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order) dai->driver->remove_order != order) return;
- if (dai->driver->remove) { - err = dai->driver->remove(dai); - if (err < 0) - dev_err(dai->dev, - "ASoC: failed to remove %s: %d\n", - dai->name, err); - } + err = snd_soc_dai_remove(dai); + if (err < 0) + dev_err(dai->dev, + "ASoC: failed to remove %s: %d\n", + dai->name, err); + dai->probed = 0; }
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 55c1fac..384765c 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -372,3 +372,10 @@ int snd_soc_dai_probe(struct snd_soc_dai *dai) 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; +}
The patch
ASoC: soc-dai: add snd_soc_dai_remove()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4
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 dcdab5820edd6123911dbd767ee1e389008b6a83 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Date: Mon, 22 Jul 2019 10:35:05 +0900 Subject: [PATCH] ASoC: soc-dai: add snd_soc_dai_remove()
Current ALSA SoC is directly using dai->driver->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_remvoe() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87imruhn1x.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown broonie@kernel.org --- include/sound/soc-dai.h | 1 + sound/soc/soc-core.c | 13 ++++++------- sound/soc/soc-dai.c | 7 +++++++ 3 files changed, 14 insertions(+), 7 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index da8d8b889089..2a11f177ce01 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -165,6 +165,7 @@ snd_pcm_sframes_t snd_soc_dai_delay(struct snd_soc_dai *dai, void snd_soc_dai_suspend(struct snd_soc_dai *dai); void snd_soc_dai_resume(struct snd_soc_dai *dai); int snd_soc_dai_probe(struct snd_soc_dai *dai); +int snd_soc_dai_remove(struct snd_soc_dai *dai);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3e73468225f9..727fd342b3fb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -992,13 +992,12 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order) dai->driver->remove_order != order) return;
- if (dai->driver->remove) { - err = dai->driver->remove(dai); - if (err < 0) - dev_err(dai->dev, - "ASoC: failed to remove %s: %d\n", - dai->name, err); - } + err = snd_soc_dai_remove(dai); + if (err < 0) + dev_err(dai->dev, + "ASoC: failed to remove %s: %d\n", + dai->name, err); + dai->probed = 0; }
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 55c1fac99613..384765c747da 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -372,3 +372,10 @@ int snd_soc_dai_probe(struct snd_soc_dai *dai) 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; +}
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
Current ALSA SoC is directly using dai->driver->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_compress_new() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-core.c | 15 ++++++++------- sound/soc/soc-dai.c | 8 ++++++++ 3 files changed, 18 insertions(+), 7 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2a11f17..0f8b095 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -166,6 +166,8 @@ void snd_soc_dai_suspend(struct snd_soc_dai *dai); void snd_soc_dai_resume(struct snd_soc_dai *dai); int snd_soc_dai_probe(struct snd_soc_dai *dai); int snd_soc_dai_remove(struct snd_soc_dai *dai); +int snd_soc_dai_compress_new(struct snd_soc_dai *dai, + struct snd_soc_pcm_runtime *rtd, int num);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 727fd34..458b090 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1545,15 +1545,16 @@ static int soc_probe_link_dais(struct snd_soc_card *card, num = rtd->dai_link->id; }
- if (cpu_dai->driver->compress_new) { - /* create compress_device" */ - ret = cpu_dai->driver->compress_new(rtd, num); - if (ret < 0) { + /* create compress_device if possible */ + ret = snd_soc_dai_compress_new(cpu_dai, rtd, num); + if (ret != -ENOTSUPP) { + if (ret < 0) dev_err(card->dev, "ASoC: can't create compress %s\n", dai_link->stream_name); - return ret; - } - } else if (!dai_link->params) { + return ret; + } + + if (!dai_link->params) { /* create the pcm */ ret = soc_new_pcm(rtd, num); if (ret < 0) { diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 384765c..e6f161b 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -379,3 +379,11 @@ int snd_soc_dai_remove(struct snd_soc_dai *dai) 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; +}
The patch
ASoC: soc-dai: add snd_soc_dai_compress_new()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4
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 b423c4202135f7794e0a9c55a884f5933d8e7156 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Date: Mon, 22 Jul 2019 10:35:29 +0900 Subject: [PATCH] ASoC: soc-dai: add snd_soc_dai_compress_new()
Current ALSA SoC is directly using dai->driver->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_compress_new() and use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87h87ehn1a.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown broonie@kernel.org --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-core.c | 15 ++++++++------- sound/soc/soc-dai.c | 8 ++++++++ 3 files changed, 18 insertions(+), 7 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2a11f177ce01..0f8b09520020 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -166,6 +166,8 @@ void snd_soc_dai_suspend(struct snd_soc_dai *dai); void snd_soc_dai_resume(struct snd_soc_dai *dai); int snd_soc_dai_probe(struct snd_soc_dai *dai); int snd_soc_dai_remove(struct snd_soc_dai *dai); +int snd_soc_dai_compress_new(struct snd_soc_dai *dai, + struct snd_soc_pcm_runtime *rtd, int num);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 727fd342b3fb..458b090f026a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1545,15 +1545,16 @@ static int soc_probe_link_dais(struct snd_soc_card *card, num = rtd->dai_link->id; }
- if (cpu_dai->driver->compress_new) { - /* create compress_device" */ - ret = cpu_dai->driver->compress_new(rtd, num); - if (ret < 0) { + /* create compress_device if possible */ + ret = snd_soc_dai_compress_new(cpu_dai, rtd, num); + if (ret != -ENOTSUPP) { + if (ret < 0) dev_err(card->dev, "ASoC: can't create compress %s\n", dai_link->stream_name); - return ret; - } - } else if (!dai_link->params) { + return ret; + } + + if (!dai_link->params) { /* create the pcm */ ret = soc_new_pcm(rtd, num); if (ret < 0) { diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 384765c747da..e6f161b9f975 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -379,3 +379,11 @@ int snd_soc_dai_remove(struct snd_soc_dai *dai) 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; +}
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
snd_soc_dai_stream_valid() is function to check stream validity. But, some code is using it, some code are checking stream->channels_min directly. Doing samethings by different method is confusable. This patch uses same funcntion for same purpose.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- include/sound/soc-dai.h | 1 + sound/soc/soc-compress.c | 9 ++++----- sound/soc/soc-dai.c | 18 ++++++++++++++++++ sound/soc/soc-pcm.c | 39 ++++++++++----------------------------- 4 files changed, 33 insertions(+), 34 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 0f8b095..dc48fe0 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -168,6 +168,7 @@ int snd_soc_dai_probe(struct snd_soc_dai *dai); int snd_soc_dai_remove(struct snd_soc_dai *dai); int snd_soc_dai_compress_new(struct snd_soc_dai *dai, struct snd_soc_pcm_runtime *rtd, int num); +bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index ddef4ff..2892110 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -872,14 +872,13 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) }
/* check client and interface hw capabilities */ - if (codec_dai->driver->playback.channels_min) + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) playback = 1; - if (codec_dai->driver->capture.channels_min) + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) capture = 1;
- capture = capture && cpu_dai->driver->capture.channels_min; - playback = playback && cpu_dai->driver->playback.channels_min; - /* * Compress devices are unidirectional so only one of the directions * should be set, check for that (xor) diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index e6f161b..1c7f638 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -387,3 +387,21 @@ int snd_soc_dai_compress_new(struct snd_soc_dai *dai, 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; +} diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index f313772..fabeac1 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -29,24 +29,6 @@
#define DPCM_MAX_BE_USERS 8
-/* - * snd_soc_dai_stream_valid() - check if a DAI supports the given stream - * - * Returns true if the DAI supports the indicated stream type. - */ -static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) -{ - struct snd_soc_pcm_stream *codec_stream; - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - codec_stream = &dai->driver->playback; - else - codec_stream = &dai->driver->capture; - - /* If the codec specifies any channels at all, it supports the stream */ - return codec_stream->channels_min; -} - /** * snd_soc_runtime_activate() - Increment active count for PCM runtime components * @rtd: ASoC PCM runtime that is activated @@ -2688,8 +2670,8 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) new ? "new" : "old", fe->dai_link->name);
/* skip if FE doesn't have playback capability */ - if (!fe->cpu_dai->driver->playback.channels_min || - !fe->codec_dai->driver->playback.channels_min) + if (!snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK) || + !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_PLAYBACK)) goto capture;
/* skip if FE isn't currently playing */ @@ -2719,8 +2701,8 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
capture: /* skip if FE doesn't have capture capability */ - if (!fe->cpu_dai->driver->capture.channels_min || - !fe->codec_dai->driver->capture.channels_min) + if (!snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE) || + !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_CAPTURE)) return 0;
/* skip if FE isn't currently capturing */ @@ -3030,14 +3012,13 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture = rtd->dai_link->dpcm_capture; } else { for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->playback.channels_min) + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) playback = 1; - if (codec_dai->driver->capture.channels_min) + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) capture = 1; } - - capture = capture && cpu_dai->driver->capture.channels_min; - playback = playback && cpu_dai->driver->playback.channels_min; }
if (rtd->dai_link->playback_only) { @@ -3375,11 +3356,11 @@ static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM;
- if (fe->cpu_dai->driver->playback.channels_min) + if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_PLAYBACK, buf + offset, out_count - offset);
- if (fe->cpu_dai->driver->capture.channels_min) + if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_CAPTURE, buf + offset, out_count - offset);
The patch
ASoC: soc-dai: move snd_soc_dai_stream_valid() to soc-dai.c
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4
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 467fece8fbc6774a3a3bd0981e1a342fb5022706 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Date: Mon, 22 Jul 2019 10:36:16 +0900 Subject: [PATCH] ASoC: soc-dai: move snd_soc_dai_stream_valid() to soc-dai.c
snd_soc_dai_stream_valid() is function to check stream validity. But, some code is using it, some code are checking stream->channels_min directly. Doing samethings by different method is confusable. This patch uses same funcntion for same purpose.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87ftmyhmzz.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown broonie@kernel.org --- include/sound/soc-dai.h | 1 + sound/soc/soc-compress.c | 9 ++++----- sound/soc/soc-dai.c | 18 ++++++++++++++++++ sound/soc/soc-pcm.c | 39 ++++++++++----------------------------- 4 files changed, 33 insertions(+), 34 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 0f8b09520020..dc48fe081a20 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -168,6 +168,7 @@ int snd_soc_dai_probe(struct snd_soc_dai *dai); int snd_soc_dai_remove(struct snd_soc_dai *dai); int snd_soc_dai_compress_new(struct snd_soc_dai *dai, struct snd_soc_pcm_runtime *rtd, int num); +bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream);
struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index ddef4ff677ce..289211069a1e 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -872,14 +872,13 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) }
/* check client and interface hw capabilities */ - if (codec_dai->driver->playback.channels_min) + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) playback = 1; - if (codec_dai->driver->capture.channels_min) + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) capture = 1;
- capture = capture && cpu_dai->driver->capture.channels_min; - playback = playback && cpu_dai->driver->playback.channels_min; - /* * Compress devices are unidirectional so only one of the directions * should be set, check for that (xor) diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index e6f161b9f975..1c7f63871c1d 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -387,3 +387,21 @@ int snd_soc_dai_compress_new(struct snd_soc_dai *dai, 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; +} diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index f3137723301c..fabeac164a6c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -29,24 +29,6 @@
#define DPCM_MAX_BE_USERS 8
-/* - * snd_soc_dai_stream_valid() - check if a DAI supports the given stream - * - * Returns true if the DAI supports the indicated stream type. - */ -static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) -{ - struct snd_soc_pcm_stream *codec_stream; - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - codec_stream = &dai->driver->playback; - else - codec_stream = &dai->driver->capture; - - /* If the codec specifies any channels at all, it supports the stream */ - return codec_stream->channels_min; -} - /** * snd_soc_runtime_activate() - Increment active count for PCM runtime components * @rtd: ASoC PCM runtime that is activated @@ -2688,8 +2670,8 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) new ? "new" : "old", fe->dai_link->name);
/* skip if FE doesn't have playback capability */ - if (!fe->cpu_dai->driver->playback.channels_min || - !fe->codec_dai->driver->playback.channels_min) + if (!snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK) || + !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_PLAYBACK)) goto capture;
/* skip if FE isn't currently playing */ @@ -2719,8 +2701,8 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
capture: /* skip if FE doesn't have capture capability */ - if (!fe->cpu_dai->driver->capture.channels_min || - !fe->codec_dai->driver->capture.channels_min) + if (!snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE) || + !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_CAPTURE)) return 0;
/* skip if FE isn't currently capturing */ @@ -3030,14 +3012,13 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture = rtd->dai_link->dpcm_capture; } else { for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->playback.channels_min) + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) playback = 1; - if (codec_dai->driver->capture.channels_min) + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) capture = 1; } - - capture = capture && cpu_dai->driver->capture.channels_min; - playback = playback && cpu_dai->driver->playback.channels_min; }
if (rtd->dai_link->playback_only) { @@ -3375,11 +3356,11 @@ static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM;
- if (fe->cpu_dai->driver->playback.channels_min) + if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_PLAYBACK, buf + offset, out_count - offset);
- if (fe->cpu_dai->driver->capture.channels_min) + if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_CAPTURE, buf + offset, out_count - offset);
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com
ALSA SoC already has snd_soc_dai_digital_mute() for digital_mute, let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/soc-core.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 458b090..7ecfe64 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -487,10 +487,9 @@ int snd_soc_suspend(struct device *dev) continue;
for_each_rtd_codec_dai(rtd, i, dai) { - struct snd_soc_dai_driver *drv = dai->driver; - - if (drv->ops->digital_mute && dai->playback_active) - drv->ops->digital_mute(dai, 1); + if (dai->playback_active) + snd_soc_dai_digital_mute(dai, 1, + SNDRV_PCM_STREAM_PLAYBACK); } }
@@ -665,10 +664,9 @@ static void soc_resume_deferred(struct work_struct *work) continue;
for_each_rtd_codec_dai(rtd, i, dai) { - struct snd_soc_dai_driver *drv = dai->driver; - - if (drv->ops->digital_mute && dai->playback_active) - drv->ops->digital_mute(dai, 0); + if (dai->playback_active) + snd_soc_dai_digital_mute(dai, 0, + SNDRV_PCM_STREAM_PLAYBACK); } }
The patch
ASoC: soc-core: use existing snd_soc_dai_digital_mute()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4
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 88fdffa2eef45fc9f6009ee590741c44a57c49c7 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Date: Mon, 22 Jul 2019 10:36:27 +0900 Subject: [PATCH] ASoC: soc-core: use existing snd_soc_dai_digital_mute()
ALSA SoC already has snd_soc_dai_digital_mute() for digital_mute, let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87ef2ihmzo.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown broonie@kernel.org --- sound/soc/soc-core.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 458b090f026a..7ecfe641ca46 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -487,10 +487,9 @@ int snd_soc_suspend(struct device *dev) continue;
for_each_rtd_codec_dai(rtd, i, dai) { - struct snd_soc_dai_driver *drv = dai->driver; - - if (drv->ops->digital_mute && dai->playback_active) - drv->ops->digital_mute(dai, 1); + if (dai->playback_active) + snd_soc_dai_digital_mute(dai, 1, + SNDRV_PCM_STREAM_PLAYBACK); } }
@@ -665,10 +664,9 @@ static void soc_resume_deferred(struct work_struct *work) continue;
for_each_rtd_codec_dai(rtd, i, dai) { - struct snd_soc_dai_driver *drv = dai->driver; - - if (drv->ops->digital_mute && dai->playback_active) - drv->ops->digital_mute(dai, 0); + if (dai->playback_active) + snd_soc_dai_digital_mute(dai, 0, + SNDRV_PCM_STREAM_PLAYBACK); } }
participants (2)
-
Kuninori Morimoto -
Mark Brown