[RFC PATCH 0/3] Separate BE DAI HW constraints from FE ones
HW constraints are needed to set limitations for HW parameters used to configure the DAIs. All DAIs on the same link must agree upon the HW parameters, so the parameters are affected by the DAIs' features and their limitations. In case of DPCM, the FE DAIs can be used to perform different kind of conversions, such as format or rate changing, bringing the audio stream to a configuration supported by all the DAIs of the BE's link. For this reason, the limitations of the BE DAIs are not always important for the HW parameters between user-space and FE, only for the paratemers between FE and BE DAI links. This brings us to this patch-set, which aims to separate the FE HW constraints from the BE HW constraints. This way, the FE can be used to perform more efficient HW conversions, on the initial audio stream parameters, to parameters supported by the BE DAIs. To achieve this, the first thing needed is to detect whether a HW constraint rule is enforced by a FE or a BE DAI. This means that snd_pcm_hw_rule_add() needs to be able to differentiate between the two type of DAIs. For this, the runtime pointer to struct snd_pcm_runtime is replaced with a pointer to struct snd_pcm_substream, to be able to reach substream->pcm->internal to differentiate between FE and BE DAIs. This change affects many sound drivers (and one gpu drm driver). All these changes are included in the first patch, to have a better overview of the implications created by this change. The second patch adds a new struct snd_pcm_hw_constraints under struct snd_soc_dpcm_runtime, which is used to store the HW constraint rules added by the BE DAIs. This structure is initialized with a subset of the runtime constraint rules which does not include the rules that affect the buffer or period size. snd_pcm_hw_rule_add() will add the BE rules to the new struct snd_pcm_hw_constraints. The third and last patch will apply the BE rule constraints, after the fixup callback. If the fixup HW parameters do not respect the BE constraint rules, the rules will exit with an error. The FE mask and interval constraints are copied to the BE ones, to satisfy the dai_link->dpcm_merged_* flags. The dai_link->dpcm_merged_* flags are used to know if the FE does format or sampling rate conversion.
I tested with ad1934 and wm8731 codecs as BEs, with a not-yet-mainlined ASRC as FE, that can also do format conversion. I realize that the change to snd_pcm_hw_rule_add() has a big impact, even though all the drivers use snd_pcm_hw_rule_add() with substream->runtime, so passing substream instead of runtime is not that risky.
Codrin Ciubotariu (3): pcm: use substream instead of runtime in snd_pcm_hw_rule_add() ASoC: soc-pcm: add hw_constraints for BE DAI links ASoC: soc-pcm: apply BE HW constraint rules
drivers/gpu/drm/vc4/vc4_hdmi.c | 2 +- include/sound/ac97_codec.h | 2 +- include/sound/pcm.h | 29 +++++-- include/sound/pcm_drm_eld.h | 2 +- include/sound/soc-dpcm.h | 1 + sound/arm/aaci.c | 4 +- sound/arm/pxa2xx-pcm-lib.c | 4 +- sound/core/pcm_drm_eld.c | 6 +- sound/core/pcm_lib.c | 66 ++++++++------- sound/core/pcm_native.c | 83 +++++++++++-------- sound/drivers/aloop.c | 8 +- sound/drivers/vx/vx_pcm.c | 8 +- sound/firewire/amdtp-am824.c | 8 +- sound/firewire/amdtp-am824.h | 2 +- sound/firewire/amdtp-stream.c | 12 +-- sound/firewire/amdtp-stream.h | 2 +- sound/firewire/bebob/bebob_pcm.c | 6 +- sound/firewire/dice/dice-pcm.c | 6 +- sound/firewire/digi00x/amdtp-dot.c | 6 +- sound/firewire/digi00x/digi00x-pcm.c | 6 +- sound/firewire/digi00x/digi00x.h | 2 +- sound/firewire/fireface/amdtp-ff.c | 6 +- sound/firewire/fireface/ff-pcm.c | 6 +- sound/firewire/fireface/ff.h | 2 +- sound/firewire/fireworks/fireworks_pcm.c | 6 +- sound/firewire/motu/amdtp-motu.c | 6 +- sound/firewire/motu/motu-pcm.c | 6 +- sound/firewire/motu/motu.h | 2 +- sound/firewire/oxfw/oxfw-pcm.c | 6 +- sound/firewire/tascam/amdtp-tascam.c | 6 +- sound/firewire/tascam/tascam-pcm.c | 2 +- sound/firewire/tascam/tascam.h | 2 +- sound/pci/ac97/ac97_pcm.c | 8 +- sound/pci/ali5451/ali5451.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/au88x0/au88x0_pcm.c | 6 +- sound/pci/azt3328.c | 2 +- sound/pci/bt87x.c | 8 +- sound/pci/ca0106/ca0106_main.c | 4 +- sound/pci/cmipci.c | 14 ++-- sound/pci/cs4281.c | 4 +- sound/pci/cs46xx/cs46xx_lib.c | 4 +- sound/pci/echoaudio/echoaudio.c | 24 +++--- sound/pci/emu10k1/emu10k1x.c | 4 +- sound/pci/emu10k1/emupcm.c | 10 +-- sound/pci/ens1370.c | 12 +-- sound/pci/es1938.c | 4 +- sound/pci/es1968.c | 2 +- sound/pci/fm801.c | 6 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_controller.c | 4 +- sound/pci/hda/patch_hdmi.c | 8 +- sound/pci/hda/patch_si3054.c | 2 +- sound/pci/ice1712/ice1712.c | 8 +- sound/pci/ice1712/ice1724.c | 30 +++---- sound/pci/intel8x0.c | 10 +-- sound/pci/intel8x0m.c | 2 +- sound/pci/lola/lola_pcm.c | 4 +- sound/pci/mixart/mixart.c | 8 +- sound/pci/nm256/nm256.c | 2 +- sound/pci/oxygen/oxygen_pcm.c | 8 +- sound/pci/pcxhr/pcxhr.c | 4 +- sound/pci/rme32.c | 14 ++-- sound/pci/rme96.c | 17 ++-- sound/pci/rme9652/hdsp.c | 24 +++--- sound/pci/rme9652/hdspm.c | 12 +-- sound/pci/rme9652/rme9652.c | 20 ++--- sound/pci/sonicvibes.c | 4 +- sound/pci/via82xx.c | 4 +- sound/pci/via82xx_modem.c | 2 +- sound/pci/ymfpci/ymfpci_main.c | 4 +- sound/soc/adi/axi-i2s.c | 2 +- sound/soc/adi/axi-spdif.c | 2 +- sound/soc/amd/acp-da7219-max98357a.c | 40 ++++----- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/bcm/bcm63xx-pcm-whistler.c | 4 +- sound/soc/bcm/cygnus-pcm.c | 5 +- sound/soc/bcm/cygnus-ssp.c | 2 +- sound/soc/codecs/ad193x.c | 2 +- sound/soc/codecs/adau1372.c | 2 +- sound/soc/codecs/ak4458.c | 2 +- sound/soc/codecs/ak4613.c | 6 +- sound/soc/codecs/ak5558.c | 2 +- sound/soc/codecs/cs35l33.c | 2 +- sound/soc/codecs/cs35l34.c | 2 +- sound/soc/codecs/cs35l35.c | 4 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs4234.c | 4 +- sound/soc/codecs/cs42l73.c | 2 +- sound/soc/codecs/cs43130.c | 4 +- sound/soc/codecs/cs53l30.c | 2 +- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/es8328.c | 2 +- sound/soc/codecs/hdmi-codec.c | 2 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/pcm512x.c | 6 +- sound/soc/codecs/sigmadsp.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/uda1334.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8524.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8741.c | 2 +- sound/soc/fsl/fsl_asrc.c | 4 +- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_sai.c | 4 +- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/fsl/fsl_xcvr.c | 9 +- sound/soc/fsl/imx-audmix.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 4 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 2 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 12 +-- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 12 +-- sound/soc/meson/aiu-encoder-i2s.c | 2 +- sound/soc/meson/aiu-fifo.c | 4 +- sound/soc/meson/axg-fifo.c | 4 +- sound/soc/sh/rcar/core.c | 6 +- sound/soc/soc-pcm.c | 76 ++++++++++++++++- sound/soc/sti/uniperif_player.c | 4 +- sound/soc/sti/uniperif_reader.c | 4 +- sound/soc/sunxi/sun4i-codec.c | 2 +- sound/soc/sunxi/sun8i-codec.c | 2 +- sound/soc/tegra/tegra_pcm.c | 2 +- sound/soc/ti/davinci-mcasp.c | 12 +-- sound/soc/ti/omap-mcbsp.c | 4 +- sound/soc/uniphier/aio-dma.c | 4 +- sound/soc/xilinx/xlnx_formatter_pcm.c | 2 +- sound/usb/hiface/pcm.c | 2 +- sound/usb/line6/capture.c | 2 +- sound/usb/line6/playback.c | 2 +- sound/usb/misc/ua101.c | 2 +- sound/usb/pcm.c | 24 +++--- 135 files changed, 535 insertions(+), 429 deletions(-)
Replace struct snd_pcm_runtime *runtime with struct snd_pcm_substream *substream in the header of snd_pcm_hw_rule_add(). This will allow us to use a different struct snd_pcm_hw_constraints later, for BEs. The drivers that call snd_pcm_hw_rule_add() and its wrappers are also updated.
Signed-off-by: Codrin Ciubotariu codrin.ciubotariu@microchip.com --- drivers/gpu/drm/vc4/vc4_hdmi.c | 2 +- include/sound/ac97_codec.h | 2 +- include/sound/pcm.h | 18 +++--- include/sound/pcm_drm_eld.h | 2 +- sound/arm/aaci.c | 4 +- sound/arm/pxa2xx-pcm-lib.c | 4 +- sound/core/pcm_drm_eld.c | 6 +- sound/core/pcm_lib.c | 58 ++++++++++--------- sound/core/pcm_native.c | 46 +++++++-------- sound/drivers/aloop.c | 8 +-- sound/drivers/vx/vx_pcm.c | 8 +-- sound/firewire/amdtp-am824.c | 8 +-- sound/firewire/amdtp-am824.h | 2 +- sound/firewire/amdtp-stream.c | 12 ++-- sound/firewire/amdtp-stream.h | 2 +- sound/firewire/bebob/bebob_pcm.c | 6 +- sound/firewire/dice/dice-pcm.c | 6 +- sound/firewire/digi00x/amdtp-dot.c | 6 +- sound/firewire/digi00x/digi00x-pcm.c | 6 +- sound/firewire/digi00x/digi00x.h | 2 +- sound/firewire/fireface/amdtp-ff.c | 6 +- sound/firewire/fireface/ff-pcm.c | 6 +- sound/firewire/fireface/ff.h | 2 +- sound/firewire/fireworks/fireworks_pcm.c | 6 +- sound/firewire/motu/amdtp-motu.c | 6 +- sound/firewire/motu/motu-pcm.c | 6 +- sound/firewire/motu/motu.h | 2 +- sound/firewire/oxfw/oxfw-pcm.c | 6 +- sound/firewire/tascam/amdtp-tascam.c | 6 +- sound/firewire/tascam/tascam-pcm.c | 2 +- sound/firewire/tascam/tascam.h | 2 +- sound/pci/ac97/ac97_pcm.c | 8 +-- sound/pci/ali5451/ali5451.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/au88x0/au88x0_pcm.c | 6 +- sound/pci/azt3328.c | 2 +- sound/pci/bt87x.c | 8 +-- sound/pci/ca0106/ca0106_main.c | 4 +- sound/pci/cmipci.c | 14 ++--- sound/pci/cs4281.c | 4 +- sound/pci/cs46xx/cs46xx_lib.c | 4 +- sound/pci/echoaudio/echoaudio.c | 24 ++++---- sound/pci/emu10k1/emu10k1x.c | 4 +- sound/pci/emu10k1/emupcm.c | 10 ++-- sound/pci/ens1370.c | 12 ++-- sound/pci/es1938.c | 4 +- sound/pci/es1968.c | 2 +- sound/pci/fm801.c | 6 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_controller.c | 4 +- sound/pci/hda/patch_hdmi.c | 8 +-- sound/pci/hda/patch_si3054.c | 2 +- sound/pci/ice1712/ice1712.c | 8 +-- sound/pci/ice1712/ice1724.c | 30 +++++----- sound/pci/intel8x0.c | 10 ++-- sound/pci/intel8x0m.c | 2 +- sound/pci/lola/lola_pcm.c | 4 +- sound/pci/mixart/mixart.c | 8 +-- sound/pci/nm256/nm256.c | 2 +- sound/pci/oxygen/oxygen_pcm.c | 8 +-- sound/pci/pcxhr/pcxhr.c | 4 +- sound/pci/rme32.c | 14 ++--- sound/pci/rme96.c | 17 +++--- sound/pci/rme9652/hdsp.c | 24 ++++---- sound/pci/rme9652/hdspm.c | 12 ++-- sound/pci/rme9652/rme9652.c | 20 +++---- sound/pci/sonicvibes.c | 4 +- sound/pci/via82xx.c | 4 +- sound/pci/via82xx_modem.c | 2 +- sound/pci/ymfpci/ymfpci_main.c | 4 +- sound/soc/adi/axi-i2s.c | 2 +- sound/soc/adi/axi-spdif.c | 2 +- sound/soc/amd/acp-da7219-max98357a.c | 40 ++++++------- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/bcm/bcm63xx-pcm-whistler.c | 4 +- sound/soc/bcm/cygnus-pcm.c | 5 +- sound/soc/bcm/cygnus-ssp.c | 2 +- sound/soc/codecs/ad193x.c | 2 +- sound/soc/codecs/adau1372.c | 2 +- sound/soc/codecs/ak4458.c | 2 +- sound/soc/codecs/ak4613.c | 6 +- sound/soc/codecs/ak5558.c | 2 +- sound/soc/codecs/cs35l33.c | 2 +- sound/soc/codecs/cs35l34.c | 2 +- sound/soc/codecs/cs35l35.c | 4 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs4234.c | 4 +- sound/soc/codecs/cs42l73.c | 2 +- sound/soc/codecs/cs43130.c | 4 +- sound/soc/codecs/cs53l30.c | 2 +- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/es8328.c | 2 +- sound/soc/codecs/hdmi-codec.c | 2 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/pcm512x.c | 6 +- sound/soc/codecs/sigmadsp.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/uda1334.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8524.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8741.c | 2 +- sound/soc/fsl/fsl_asrc.c | 4 +- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_sai.c | 4 +- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/fsl/fsl_xcvr.c | 9 ++- sound/soc/fsl/imx-audmix.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 4 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 2 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 12 ++-- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 12 ++-- sound/soc/meson/aiu-encoder-i2s.c | 2 +- sound/soc/meson/aiu-fifo.c | 4 +- sound/soc/meson/axg-fifo.c | 4 +- sound/soc/sh/rcar/core.c | 6 +- sound/soc/soc-pcm.c | 2 +- sound/soc/sti/uniperif_player.c | 4 +- sound/soc/sti/uniperif_reader.c | 4 +- sound/soc/sunxi/sun4i-codec.c | 2 +- sound/soc/sunxi/sun8i-codec.c | 2 +- sound/soc/tegra/tegra_pcm.c | 2 +- sound/soc/ti/davinci-mcasp.c | 12 ++-- sound/soc/ti/omap-mcbsp.c | 4 +- sound/soc/uniphier/aio-dma.c | 4 +- sound/soc/xilinx/xlnx_formatter_pcm.c | 2 +- sound/usb/hiface/pcm.c | 2 +- sound/usb/line6/capture.c | 2 +- sound/usb/line6/playback.c | 2 +- sound/usb/misc/ua101.c | 2 +- sound/usb/pcm.c | 24 ++++---- 134 files changed, 416 insertions(+), 417 deletions(-)
diff --git a/drivers/gpu/drm/vc4/vc4_hdmi.c b/drivers/gpu/drm/vc4/vc4_hdmi.c index 1fda574579af..c933427237cc 100644 --- a/drivers/gpu/drm/vc4/vc4_hdmi.c +++ b/drivers/gpu/drm/vc4/vc4_hdmi.c @@ -1084,7 +1084,7 @@ static int vc4_hdmi_audio_startup(struct snd_pcm_substream *substream, VC4_HDMI_RAM_PACKET_ENABLE)) return -ENODEV;
- ret = snd_pcm_hw_constraint_eld(substream->runtime, connector->eld); + ret = snd_pcm_hw_constraint_eld(substream, connector->eld); if (ret) return ret;
diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 49200ec26dc4..d98130cc23dd 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -404,7 +404,7 @@ int snd_ac97_pcm_assign(struct snd_ac97_bus *ac97, int snd_ac97_pcm_open(struct ac97_pcm *pcm, unsigned int rate, enum ac97_pcm_cfg cfg, unsigned short slots); int snd_ac97_pcm_close(struct ac97_pcm *pcm); -int snd_ac97_pcm_double_rate_rules(struct snd_pcm_runtime *runtime); +int snd_ac97_pcm_double_rate_rules(struct snd_pcm_substream *substream);
/* ad hoc AC97 device driver access */ extern struct bus_type ac97_bus_type; diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2e1200d17d0c..af7fce2b574d 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -984,36 +984,36 @@ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_par int snd_pcm_hw_constraint_minmax(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, unsigned int min, unsigned int max); int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var); -int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_list(struct snd_pcm_substream *substream, unsigned int cond, snd_pcm_hw_param_t var, const struct snd_pcm_hw_constraint_list *l); -int snd_pcm_hw_constraint_ranges(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_ranges(struct snd_pcm_substream *substream, unsigned int cond, snd_pcm_hw_param_t var, const struct snd_pcm_hw_constraint_ranges *r); -int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_ratnums(struct snd_pcm_substream *substream, unsigned int cond, snd_pcm_hw_param_t var, const struct snd_pcm_hw_constraint_ratnums *r); -int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_ratdens(struct snd_pcm_substream *substream, unsigned int cond, snd_pcm_hw_param_t var, const struct snd_pcm_hw_constraint_ratdens *r); -int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_msbits(struct snd_pcm_substream *substream, unsigned int cond, unsigned int width, unsigned int msbits); -int snd_pcm_hw_constraint_step(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_step(struct snd_pcm_substream *substream, unsigned int cond, snd_pcm_hw_param_t var, unsigned long step); -int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_pow2(struct snd_pcm_substream *substream, unsigned int cond, snd_pcm_hw_param_t var); -int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_rule_noresample(struct snd_pcm_substream *substream, unsigned int base_rate); -int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_rule_add(struct snd_pcm_substream *substream, unsigned int cond, int var, snd_pcm_hw_rule_func_t func, void *private, diff --git a/include/sound/pcm_drm_eld.h b/include/sound/pcm_drm_eld.h index 28a55a8beb28..d5590b12bfda 100644 --- a/include/sound/pcm_drm_eld.h +++ b/include/sound/pcm_drm_eld.h @@ -2,6 +2,6 @@ #ifndef __SOUND_PCM_DRM_ELD_H #define __SOUND_PCM_DRM_ELD_H
-int snd_pcm_hw_constraint_eld(struct snd_pcm_runtime *runtime, void *eld); +int snd_pcm_hw_constraint_eld(struct snd_pcm_substream *substream, void *eld);
#endif diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 0817ad21af74..b2174db24682 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -419,7 +419,7 @@ static int aaci_pcm_open(struct snd_pcm_substream *substream) runtime->hw.channels_max = 6;
/* Add rule describing channel dependency. */ - ret = snd_pcm_hw_rule_add(substream->runtime, 0, + ret = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, aaci_rule_channels, aaci, SNDRV_PCM_HW_PARAM_CHANNELS, -1); @@ -427,7 +427,7 @@ static int aaci_pcm_open(struct snd_pcm_substream *substream) return ret;
if (aacirun->pcm->r[1].slots) - snd_ac97_pcm_double_rate_rules(runtime); + snd_ac97_pcm_double_rate_rules(substream); }
/* diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index e81083e1bc68..2ea3146061ae 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -104,12 +104,12 @@ int pxa2xx_pcm_open(struct snd_pcm_substream *substream) * playback samples are lost if the DMA count is not a multiple * of the DMA burst size. Let's add a rule to enforce that. */ - ret = snd_pcm_hw_constraint_step(runtime, 0, + ret = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); if (ret) return ret;
- ret = snd_pcm_hw_constraint_step(runtime, 0, + ret = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32); if (ret) return ret; diff --git a/sound/core/pcm_drm_eld.c b/sound/core/pcm_drm_eld.c index 4b5faae5d16e..7988074b9c6f 100644 --- a/sound/core/pcm_drm_eld.c +++ b/sound/core/pcm_drm_eld.c @@ -77,17 +77,17 @@ static int eld_limit_channels(struct snd_pcm_hw_params *params, return snd_interval_refine(c, &t); }
-int snd_pcm_hw_constraint_eld(struct snd_pcm_runtime *runtime, void *eld) +int snd_pcm_hw_constraint_eld(struct snd_pcm_substream *substream, void *eld) { int ret;
- ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + ret = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, eld_limit_rates, eld, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (ret < 0) return ret;
- ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + ret = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, eld_limit_channels, eld, SNDRV_PCM_HW_PARAM_RATE, -1);
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index b7e3d8f44511..582144f99045 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1107,7 +1107,7 @@ static int snd_interval_step(struct snd_interval *i, unsigned int step)
/** * snd_pcm_hw_rule_add - add the hw-constraint rule - * @runtime: the pcm runtime instance + * @substream: the pcm substream * @cond: condition bits * @var: the variable to evaluate * @func: the evaluation function @@ -1116,15 +1116,17 @@ static int snd_interval_step(struct snd_interval *i, unsigned int step) * * Return: Zero if successful, or a negative error code on failure. */ -int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, +int snd_pcm_hw_rule_add(struct snd_pcm_substream *substream, unsigned int cond, int var, snd_pcm_hw_rule_func_t func, void *private, int dep, ...) { - struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints; + struct snd_pcm_hw_constraints *constrs = + &substream->runtime->hw_constraints; struct snd_pcm_hw_rule *c; unsigned int k; va_list args; + va_start(args, dep); if (constrs->rules_num >= constrs->rules_all) { struct snd_pcm_hw_rule *new; @@ -1258,21 +1260,21 @@ static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params,
/** * snd_pcm_hw_constraint_list - apply a list of constraints to a parameter - * @runtime: PCM runtime instance + * @substream: the pcm substream * @cond: condition bits * @var: hw_params variable to apply the list constraint * @l: list - * + * * Apply the list of constraints to an interval parameter. * * Return: Zero if successful, or a negative error code on failure. */ -int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_list(struct snd_pcm_substream *substream, unsigned int cond, snd_pcm_hw_param_t var, const struct snd_pcm_hw_constraint_list *l) { - return snd_pcm_hw_rule_add(runtime, cond, var, + return snd_pcm_hw_rule_add(substream, cond, var, snd_pcm_hw_rule_list, (void *)l, var, -1); } @@ -1289,7 +1291,7 @@ static int snd_pcm_hw_rule_ranges(struct snd_pcm_hw_params *params,
/** * snd_pcm_hw_constraint_ranges - apply list of range constraints to a parameter - * @runtime: PCM runtime instance + * @substream: the pcm substream * @cond: condition bits * @var: hw_params variable to apply the list of range constraints * @r: ranges @@ -1298,12 +1300,12 @@ static int snd_pcm_hw_rule_ranges(struct snd_pcm_hw_params *params, * * Return: Zero if successful, or a negative error code on failure. */ -int snd_pcm_hw_constraint_ranges(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_ranges(struct snd_pcm_substream *substream, unsigned int cond, snd_pcm_hw_param_t var, const struct snd_pcm_hw_constraint_ranges *r) { - return snd_pcm_hw_rule_add(runtime, cond, var, + return snd_pcm_hw_rule_add(substream, cond, var, snd_pcm_hw_rule_ranges, (void *)r, var, -1); } @@ -1326,19 +1328,19 @@ static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params,
/** * snd_pcm_hw_constraint_ratnums - apply ratnums constraint to a parameter - * @runtime: PCM runtime instance + * @substream: the pcm substream * @cond: condition bits * @var: hw_params variable to apply the ratnums constraint * @r: struct snd_ratnums constriants * * Return: Zero if successful, or a negative error code on failure. */ -int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_ratnums(struct snd_pcm_substream *substream, unsigned int cond, snd_pcm_hw_param_t var, const struct snd_pcm_hw_constraint_ratnums *r) { - return snd_pcm_hw_rule_add(runtime, cond, var, + return snd_pcm_hw_rule_add(substream, cond, var, snd_pcm_hw_rule_ratnums, (void *)r, var, -1); } @@ -1360,19 +1362,19 @@ static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params,
/** * snd_pcm_hw_constraint_ratdens - apply ratdens constraint to a parameter - * @runtime: PCM runtime instance + * @substream: the pcm substream * @cond: condition bits * @var: hw_params variable to apply the ratdens constraint * @r: struct snd_ratdens constriants * * Return: Zero if successful, or a negative error code on failure. */ -int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_ratdens(struct snd_pcm_substream *substream, unsigned int cond, snd_pcm_hw_param_t var, const struct snd_pcm_hw_constraint_ratdens *r) { - return snd_pcm_hw_rule_add(runtime, cond, var, + return snd_pcm_hw_rule_add(substream, cond, var, snd_pcm_hw_rule_ratdens, (void *)r, var, -1); } @@ -1399,7 +1401,7 @@ static int snd_pcm_hw_rule_msbits(struct snd_pcm_hw_params *params,
/** * snd_pcm_hw_constraint_msbits - add a hw constraint msbits rule - * @runtime: PCM runtime instance + * @substream: the pcm substream * @cond: condition bits * @width: sample bits width * @msbits: msbits width @@ -1411,13 +1413,13 @@ static int snd_pcm_hw_rule_msbits(struct snd_pcm_hw_params *params, * * Return: Zero if successful, or a negative error code on failure. */ -int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_msbits(struct snd_pcm_substream *substream, unsigned int cond, unsigned int width, unsigned int msbits) { unsigned long l = (msbits << 16) | width; - return snd_pcm_hw_rule_add(runtime, cond, -1, + return snd_pcm_hw_rule_add(substream, cond, -1, snd_pcm_hw_rule_msbits, (void*) l, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); @@ -1433,19 +1435,19 @@ static int snd_pcm_hw_rule_step(struct snd_pcm_hw_params *params,
/** * snd_pcm_hw_constraint_step - add a hw constraint step rule - * @runtime: PCM runtime instance + * @substream: the pcm substream * @cond: condition bits * @var: hw_params variable to apply the step constraint * @step: step size * * Return: Zero if successful, or a negative error code on failure. */ -int snd_pcm_hw_constraint_step(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_step(struct snd_pcm_substream *substream, unsigned int cond, snd_pcm_hw_param_t var, unsigned long step) { - return snd_pcm_hw_rule_add(runtime, cond, var, + return snd_pcm_hw_rule_add(substream, cond, var, snd_pcm_hw_rule_step, (void *) step, var, -1); } @@ -1465,17 +1467,17 @@ static int snd_pcm_hw_rule_pow2(struct snd_pcm_hw_params *params, struct snd_pcm
/** * snd_pcm_hw_constraint_pow2 - add a hw constraint power-of-2 rule - * @runtime: PCM runtime instance + * @substream: the pcm substream * @cond: condition bits * @var: hw_params variable to apply the power-of-2 constraint * * Return: Zero if successful, or a negative error code on failure. */ -int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_pow2(struct snd_pcm_substream *substream, unsigned int cond, snd_pcm_hw_param_t var) { - return snd_pcm_hw_rule_add(runtime, cond, var, + return snd_pcm_hw_rule_add(substream, cond, var, snd_pcm_hw_rule_pow2, NULL, var, -1); } @@ -1493,15 +1495,15 @@ static int snd_pcm_hw_rule_noresample_func(struct snd_pcm_hw_params *params,
/** * snd_pcm_hw_rule_noresample - add a rule to allow disabling hw resampling - * @runtime: PCM runtime instance + * @substream: the pcm substream * @base_rate: the rate at which the hardware does not resample * * Return: Zero if successful, or a negative error code on failure. */ -int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_rule_noresample(struct snd_pcm_substream *substream, unsigned int base_rate) { - return snd_pcm_hw_rule_add(runtime, SNDRV_PCM_HW_PARAMS_NORESAMPLE, + return snd_pcm_hw_rule_add(substream, SNDRV_PCM_HW_PARAMS_NORESAMPLE, SNDRV_PCM_HW_PARAM_RATE, snd_pcm_hw_rule_noresample_func, (void *)(uintptr_t)base_rate, diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 17a85f4815d5..5feeef1b43f1 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2420,103 +2420,103 @@ static int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream) snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)); snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_FRAME_BITS));
- err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_FORMAT, snd_pcm_hw_rule_format, NULL, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, snd_pcm_hw_rule_sample_bits, NULL, - SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, snd_pcm_hw_rule_div, NULL, SNDRV_PCM_HW_PARAM_FRAME_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, snd_pcm_hw_rule_mul, NULL, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, snd_pcm_hw_rule_mulkdiv, (void*) 8, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, snd_pcm_hw_rule_mulkdiv, (void*) 8, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_pcm_hw_rule_div, NULL, SNDRV_PCM_HW_PARAM_FRAME_BITS, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, snd_pcm_hw_rule_mulkdiv, (void*) 1000000, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_PERIOD_TIME, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, snd_pcm_hw_rule_mulkdiv, (void*) 1000000, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_BUFFER_TIME, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIODS, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_PERIODS, snd_pcm_hw_rule_div, NULL, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, snd_pcm_hw_rule_div, NULL, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_PERIODS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, snd_pcm_hw_rule_mulkdiv, (void*) 8, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, snd_pcm_hw_rule_muldivk, (void*) 1000000, SNDRV_PCM_HW_PARAM_PERIOD_TIME, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, snd_pcm_hw_rule_mul, NULL, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_PERIODS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, snd_pcm_hw_rule_mulkdiv, (void*) 8, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, snd_pcm_hw_rule_muldivk, (void*) 1000000, SNDRV_PCM_HW_PARAM_BUFFER_TIME, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, snd_pcm_hw_rule_muldivk, (void*) 8, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, snd_pcm_hw_rule_muldivk, (void*) 8, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_TIME, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_TIME, snd_pcm_hw_rule_mulkdiv, (void*) 1000000, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_TIME, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_TIME, snd_pcm_hw_rule_mulkdiv, (void*) 1000000, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) @@ -2581,7 +2581,7 @@ static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) if (err < 0) return err;
- err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, snd_pcm_hw_rule_buffer_bytes_max, substream, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, -1); if (err < 0) @@ -2595,7 +2595,7 @@ static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) }
if (!(hw->rates & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))) { - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, snd_pcm_hw_rule_rate, hw, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 52637180af33..503845db7785 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -1226,19 +1226,19 @@ static int loopback_open(struct snd_pcm_substream *substream) /* use dynamic rules based on actual runtime->hw values */ /* note that the default rules created in the PCM midlevel code */ /* are cached -> they do not reflect the actual state */ - err = snd_pcm_hw_rule_add(runtime, 0, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_FORMAT, rule_format, dpcm, SNDRV_PCM_HW_PARAM_FORMAT, -1); if (err < 0) goto unlock; - err = snd_pcm_hw_rule_add(runtime, 0, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, rule_rate, dpcm, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) goto unlock; - err = snd_pcm_hw_rule_add(runtime, 0, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, rule_channels, dpcm, SNDRV_PCM_HW_PARAM_CHANNELS, -1); @@ -1250,7 +1250,7 @@ static int loopback_open(struct snd_pcm_substream *substream) * This rule only takes effect if a sound timer was chosen */ if (cable->snd_timer.instance) { - err = snd_pcm_hw_rule_add(runtime, 0, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, rule_period_bytes, dpcm, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, -1); diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index daffda99b4f7..12e9d53a5f48 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -541,8 +541,8 @@ static int vx_pcm_playback_open(struct snd_pcm_substream *subs) runtime->private_data = pipe;
/* align to 4 bytes (otherwise will be problematic when 24bit is used) */ - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4); - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 4); + snd_pcm_hw_constraint_step(subs, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4); + snd_pcm_hw_constraint_step(subs, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 4);
return 0; } @@ -924,8 +924,8 @@ static int vx_pcm_capture_open(struct snd_pcm_substream *subs) runtime->private_data = pipe;
/* align to 4 bytes (otherwise will be problematic when 24bit is used) */ - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4); - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 4); + snd_pcm_hw_constraint_step(subs, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4); + snd_pcm_hw_constraint_step(subs, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 4);
return 0; } diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c index fea92e148790..08380824166c 100644 --- a/sound/firewire/amdtp-am824.c +++ b/sound/firewire/amdtp-am824.c @@ -224,20 +224,20 @@ static void write_pcm_silence(struct amdtp_stream *s, /** * amdtp_am824_add_pcm_hw_constraints - add hw constraints for PCM substream * @s: the AMDTP stream for AM824 data block, must be initialized. - * @runtime: the PCM substream runtime + * @pcm: the PCM substream * */ int amdtp_am824_add_pcm_hw_constraints(struct amdtp_stream *s, - struct snd_pcm_runtime *runtime) + struct snd_pcm_substream *pcm) { int err;
- err = amdtp_stream_add_pcm_hw_constraints(s, runtime); + err = amdtp_stream_add_pcm_hw_constraints(s, pcm); if (err < 0) return err;
/* AM824 in IEC 61883-6 can deliver 24bit data. */ - return snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + return snd_pcm_hw_constraint_msbits(pcm, 0, 32, 24); } EXPORT_SYMBOL_GPL(amdtp_am824_add_pcm_hw_constraints);
diff --git a/sound/firewire/amdtp-am824.h b/sound/firewire/amdtp-am824.h index 06d280783581..13fea73639d6 100644 --- a/sound/firewire/amdtp-am824.h +++ b/sound/firewire/amdtp-am824.h @@ -39,7 +39,7 @@ void amdtp_am824_set_midi_position(struct amdtp_stream *s, unsigned int position);
int amdtp_am824_add_pcm_hw_constraints(struct amdtp_stream *s, - struct snd_pcm_runtime *runtime); + struct snd_pcm_substream *pcm);
void amdtp_am824_midi_trigger(struct amdtp_stream *s, unsigned int port, struct snd_rawmidi_substream *midi); diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 4e2f2bb7879f..eef5c3d90297 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -173,12 +173,12 @@ static int apply_constraint_to_size(struct snd_pcm_hw_params *params, /** * amdtp_stream_add_pcm_hw_constraints - add hw constraints for PCM substream * @s: the AMDTP stream, which must be initialized. - * @runtime: the PCM substream runtime + * @pcm: the PCM substream */ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, - struct snd_pcm_runtime *runtime) + struct snd_pcm_substream *pcm) { - struct snd_pcm_hardware *hw = &runtime->hw; + struct snd_pcm_hardware *hw = &pcm->runtime->hw; unsigned int ctx_header_size; unsigned int maximum_usec_per_period; int err; @@ -228,7 +228,7 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, // Due to the above protocol design, the minimum PCM frames per // interrupt should be double of the value of syt interval, thus it is // 250 usec. - err = snd_pcm_hw_constraint_minmax(runtime, + err = snd_pcm_hw_constraint_minmax(pcm->runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 250, maximum_usec_per_period); if (err < 0) @@ -244,13 +244,13 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, * depending on its sampling rate. For accurate period interrupt, it's * preferrable to align period/buffer sizes to current SYT_INTERVAL. */ - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + err = snd_pcm_hw_rule_add(pcm, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, apply_constraint_to_size, NULL, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) goto end; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + err = snd_pcm_hw_rule_add(pcm, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, apply_constraint_to_size, NULL, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_RATE, -1); diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index a3daa1f2c1c4..de2e9f0831d9 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -197,7 +197,7 @@ unsigned int amdtp_stream_get_max_payload(struct amdtp_stream *s); void amdtp_stream_update(struct amdtp_stream *s);
int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, - struct snd_pcm_runtime *runtime); + struct snd_pcm_substream *pcm);
void amdtp_stream_pcm_prepare(struct amdtp_stream *s); void amdtp_stream_pcm_abort(struct amdtp_stream *s); diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index f8d9a2041264..279264ce935c 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -112,19 +112,19 @@ pcm_init_hw_params(struct snd_bebob *bebob,
limit_channels_and_rates(&runtime->hw, formations);
- err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, formations, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) goto end;
- err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, hw_rule_rate, formations, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) goto end;
- err = amdtp_am824_add_pcm_hw_constraints(s, runtime); + err = amdtp_am824_add_pcm_hw_constraints(s, substream); end: return err; } diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index af8a90ee40f3..5e7e3ebd47b0 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -147,18 +147,18 @@ static int init_hw_info(struct snd_dice *dice, if (err < 0) return err;
- err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, dice_rate_constraint, substream, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, dice_channels_constraint, substream, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) return err;
- return amdtp_am824_add_pcm_hw_constraints(stream, runtime); + return amdtp_am824_add_pcm_hw_constraints(stream, substream); }
static int pcm_open(struct snd_pcm_substream *substream) diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c index d613642a2ce3..2188434b8642 100644 --- a/sound/firewire/digi00x/amdtp-dot.c +++ b/sound/firewire/digi00x/amdtp-dot.c @@ -320,16 +320,16 @@ static void read_midi_messages(struct amdtp_stream *s, __be32 *buffer, }
int amdtp_dot_add_pcm_hw_constraints(struct amdtp_stream *s, - struct snd_pcm_runtime *runtime) + struct snd_pcm_substream *pcm) { int err;
/* This protocol delivers 24 bit data in 32bit data channel. */ - err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + err = snd_pcm_hw_constraint_msbits(pcm, 0, 32, 24); if (err < 0) return err;
- return amdtp_stream_add_pcm_hw_constraints(s, runtime); + return amdtp_stream_add_pcm_hw_constraints(s, pcm); }
void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port, diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c index b7f6eda09f9f..a30eda437cb7 100644 --- a/sound/firewire/digi00x/digi00x-pcm.c +++ b/sound/firewire/digi00x/digi00x-pcm.c @@ -80,21 +80,21 @@ static int pcm_init_hw_params(struct snd_dg00x *dg00x, SNDRV_PCM_RATE_96000; snd_pcm_limit_hw_rates(runtime);
- err = snd_pcm_hw_rule_add(substream->runtime, 0, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, NULL, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) return err;
- err = snd_pcm_hw_rule_add(substream->runtime, 0, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, hw_rule_rate, NULL, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) return err;
- return amdtp_dot_add_pcm_hw_constraints(s, substream->runtime); + return amdtp_dot_add_pcm_hw_constraints(s, substream); }
static int pcm_open(struct snd_pcm_substream *substream) diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h index 129de8edd5ea..da81bff07f70 100644 --- a/sound/firewire/digi00x/digi00x.h +++ b/sound/firewire/digi00x/digi00x.h @@ -121,7 +121,7 @@ int amdtp_dot_set_parameters(struct amdtp_stream *s, unsigned int rate, unsigned int pcm_channels); void amdtp_dot_reset(struct amdtp_stream *s); int amdtp_dot_add_pcm_hw_constraints(struct amdtp_stream *s, - struct snd_pcm_runtime *runtime); + struct snd_pcm_substream *pcm); void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port, struct snd_rawmidi_substream *midi);
diff --git a/sound/firewire/fireface/amdtp-ff.c b/sound/firewire/fireface/amdtp-ff.c index 119c0076b17a..345d6bcdc610 100644 --- a/sound/firewire/fireface/amdtp-ff.c +++ b/sound/firewire/fireface/amdtp-ff.c @@ -101,15 +101,15 @@ static void write_pcm_silence(struct amdtp_stream *s, }
int amdtp_ff_add_pcm_hw_constraints(struct amdtp_stream *s, - struct snd_pcm_runtime *runtime) + struct snd_pcm_substream *pcm) { int err;
- err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + err = snd_pcm_hw_constraint_msbits(pcm, 0, 32, 24); if (err < 0) return err;
- return amdtp_stream_add_pcm_hw_constraints(s, runtime); + return amdtp_stream_add_pcm_hw_constraints(s, pcm); }
static unsigned int process_it_ctx_payloads(struct amdtp_stream *s, diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c index f978cc2fed7d..89c4e3d52470 100644 --- a/sound/firewire/fireface/ff-pcm.c +++ b/sound/firewire/fireface/ff-pcm.c @@ -121,19 +121,19 @@ static int pcm_init_hw_params(struct snd_ff *ff,
limit_channels_and_rates(&runtime->hw, pcm_channels);
- err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, (void *)pcm_channels, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) return err;
- err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, hw_rule_rate, (void *)pcm_channels, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) return err;
- return amdtp_ff_add_pcm_hw_constraints(s, runtime); + return amdtp_ff_add_pcm_hw_constraints(s, substream); }
static int pcm_open(struct snd_pcm_substream *substream) diff --git a/sound/firewire/fireface/ff.h b/sound/firewire/fireface/ff.h index 705e7df4f929..99a2c06e3bb6 100644 --- a/sound/firewire/fireface/ff.h +++ b/sound/firewire/fireface/ff.h @@ -138,7 +138,7 @@ void snd_ff_transaction_unregister(struct snd_ff *ff); int amdtp_ff_set_parameters(struct amdtp_stream *s, unsigned int rate, unsigned int pcm_channels); int amdtp_ff_add_pcm_hw_constraints(struct amdtp_stream *s, - struct snd_pcm_runtime *runtime); + struct snd_pcm_substream *substream); int amdtp_ff_init(struct amdtp_stream *s, struct fw_unit *unit, enum amdtp_stream_direction dir);
diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c index a0d5db1d8eb2..f6f611cb3a09 100644 --- a/sound/firewire/fireworks/fireworks_pcm.c +++ b/sound/firewire/fireworks/fireworks_pcm.c @@ -153,19 +153,19 @@ pcm_init_hw_params(struct snd_efw *efw,
limit_channels(&runtime->hw, pcm_channels);
- err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, pcm_channels, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) goto end;
- err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, hw_rule_rate, pcm_channels, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) goto end;
- err = amdtp_am824_add_pcm_hw_constraints(s, runtime); + err = amdtp_am824_add_pcm_hw_constraints(s, substream); end: return err; } diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c index edb31ac26868..17827a3948d0 100644 --- a/sound/firewire/motu/amdtp-motu.c +++ b/sound/firewire/motu/amdtp-motu.c @@ -210,16 +210,16 @@ static void write_pcm_silence(struct amdtp_stream *s, __be32 *buffer, }
int amdtp_motu_add_pcm_hw_constraints(struct amdtp_stream *s, - struct snd_pcm_runtime *runtime) + struct snd_pcm_substream *pcm) { int err;
/* TODO: how to set an constraint for exactly 24bit PCM sample? */ - err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + err = snd_pcm_hw_constraint_msbits(pcm, 0, 32, 24); if (err < 0) return err;
- return amdtp_stream_add_pcm_hw_constraints(s, runtime); + return amdtp_stream_add_pcm_hw_constraints(s, pcm); }
void amdtp_motu_midi_trigger(struct amdtp_stream *s, unsigned int port, diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c index 8e1437371263..b481097ec5f3 100644 --- a/sound/firewire/motu/motu-pcm.c +++ b/sound/firewire/motu/motu-pcm.c @@ -113,18 +113,18 @@ static int init_hw_info(struct snd_motu *motu,
limit_channels_and_rates(motu, runtime, formats);
- err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, motu_rate_constraint, formats, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, motu_channels_constraint, formats, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) return err;
- return amdtp_motu_add_pcm_hw_constraints(stream, runtime); + return amdtp_motu_add_pcm_hw_constraints(stream, substream); }
static int pcm_open(struct snd_pcm_substream *substream) diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index 3d0236ee6716..c5f5552f5fc8 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -129,7 +129,7 @@ int amdtp_motu_set_parameters(struct amdtp_stream *s, unsigned int rate, unsigned int midi_ports, struct snd_motu_packet_format *formats); int amdtp_motu_add_pcm_hw_constraints(struct amdtp_stream *s, - struct snd_pcm_runtime *runtime); + struct snd_pcm_substream *pcm); void amdtp_motu_midi_trigger(struct amdtp_stream *s, unsigned int port, struct snd_rawmidi_substream *midi);
diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 2dfa7e179cb6..4a014593296b 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -126,19 +126,19 @@ static int init_hw_params(struct snd_oxfw *oxfw,
limit_channels_and_rates(&runtime->hw, formats);
- err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, formats, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) goto end;
- err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, hw_rule_rate, formats, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) goto end;
- err = amdtp_am824_add_pcm_hw_constraints(stream, runtime); + err = amdtp_am824_add_pcm_hw_constraints(stream, substream); end: return err; } diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c index f823a2ab3544..820d2daa800a 100644 --- a/sound/firewire/tascam/amdtp-tascam.c +++ b/sound/firewire/tascam/amdtp-tascam.c @@ -111,7 +111,7 @@ static void write_pcm_silence(struct amdtp_stream *s, __be32 *buffer, }
int amdtp_tscm_add_pcm_hw_constraints(struct amdtp_stream *s, - struct snd_pcm_runtime *runtime) + struct snd_pcm_substream *pcm) { int err;
@@ -119,11 +119,11 @@ int amdtp_tscm_add_pcm_hw_constraints(struct amdtp_stream *s, * Our implementation allows this protocol to deliver 24 bit sample in * 32bit data channel. */ - err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + err = snd_pcm_hw_constraint_msbits(pcm, 0, 32, 24); if (err < 0) return err;
- return amdtp_stream_add_pcm_hw_constraints(s, runtime); + return amdtp_stream_add_pcm_hw_constraints(s, pcm); }
static void read_status_messages(struct amdtp_stream *s, diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c index 36c1353f2494..a36cda1438d2 100644 --- a/sound/firewire/tascam/tascam-pcm.c +++ b/sound/firewire/tascam/tascam-pcm.c @@ -37,7 +37,7 @@ static int pcm_init_hw_params(struct snd_tscm *tscm, SNDRV_PCM_RATE_96000; snd_pcm_limit_hw_rates(runtime);
- return amdtp_tscm_add_pcm_hw_constraints(stream, runtime); + return amdtp_tscm_add_pcm_hw_constraints(stream, substream); }
static int pcm_open(struct snd_pcm_substream *substream) diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h index 78b7a08986a1..afbab9329d20 100644 --- a/sound/firewire/tascam/tascam.h +++ b/sound/firewire/tascam/tascam.h @@ -160,7 +160,7 @@ int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit, enum amdtp_stream_direction dir, unsigned int pcm_channels); int amdtp_tscm_set_parameters(struct amdtp_stream *s, unsigned int rate); int amdtp_tscm_add_pcm_hw_constraints(struct amdtp_stream *s, - struct snd_pcm_runtime *runtime); + struct snd_pcm_substream *pcm);
int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate); int snd_tscm_stream_get_clock(struct snd_tscm *tscm, diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c index 491de1a623cb..002d68b8952c 100644 --- a/sound/pci/ac97/ac97_pcm.c +++ b/sound/pci/ac97/ac97_pcm.c @@ -712,23 +712,23 @@ static int double_rate_hw_constraint_channels(struct snd_pcm_hw_params *params,
/** * snd_ac97_pcm_double_rate_rules - set double rate constraints - * @runtime: the runtime of the ac97 front playback pcm + * @substream: the substream of the ac97 front playback pcm * * Installs the hardware constraint rules to prevent using double rates and * more than two channels at the same time. * * Return: Zero if successful, or a negative error code on failure. */ -int snd_ac97_pcm_double_rate_rules(struct snd_pcm_runtime *runtime) +int snd_ac97_pcm_double_rate_rules(struct snd_pcm_substream *substream) { int err;
- err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, double_rate_hw_constraint_rate, NULL, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, double_rate_hw_constraint_channels, NULL, SNDRV_PCM_HW_PARAM_RATE, -1); return err; diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 0d66b92466d5..9d19b5ff0607 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -1569,7 +1569,7 @@ static int snd_ali_modem_open(struct snd_pcm_substream *substream, int rec,
if (err) return err; - return snd_pcm_hw_constraint_list(substream->runtime, 0, + return snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraint_rates); }
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 579425ccbb6a..32aca93254e7 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1082,7 +1082,7 @@ static int snd_atiixp_playback_open(struct snd_pcm_substream *substream) substream->runtime->hw.channels_max = chip->max_channels; if (chip->max_channels > 2) /* channels must be even */ - snd_pcm_hw_constraint_step(substream->runtime, 0, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); return 0; } diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 45e75afec7a0..cfc0edcd60e2 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -856,7 +856,7 @@ static int snd_atiixp_pcm_open(struct snd_pcm_substream *substream, dma->substream = substream; runtime->hw = snd_atiixp_pcm_hw; dma->ac97_pcm_type = pcm_type; - if ((err = snd_pcm_hw_constraint_list(runtime, 0, + if ((err = snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0) return err; diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index d019aa566de3..d072366ee67f 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -136,11 +136,11 @@ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream) return err; /* Avoid PAGE_SIZE boundary to fall inside of a period. */ if ((err = - snd_pcm_hw_constraint_pow2(runtime, 0, + snd_pcm_hw_constraint_pow2(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES)) < 0) return err;
- snd_pcm_hw_constraint_step(runtime, 0, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 64);
if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) { @@ -171,7 +171,7 @@ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream) VORTEX_IS_QUAD(vortex) && VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB) { runtime->hw.channels_max = 4; - snd_pcm_hw_constraint_list(runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_au8830_channels); } diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 51dcf1bc4c0c..40d2021ed46f 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2024,7 +2024,7 @@ snd_azf3328_pcm_open(struct snd_pcm_substream *substream, /* same parameters for all our codecs - at least we think so... */ runtime->hw = snd_azf3328_hardware;
- snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_azf3328_hw_constraints_rates); runtime->private_data = codec; return 0; diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 91512b345d19..e6fd99a9145d 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -379,7 +379,7 @@ static int snd_bt87x_set_digital_hw(struct snd_bt87x *chip, struct snd_pcm_runti return 0; }
-static int snd_bt87x_set_analog_hw(struct snd_bt87x *chip, struct snd_pcm_runtime *runtime) +static int snd_bt87x_set_analog_hw(struct snd_bt87x *chip, struct snd_pcm_substream *substream) { static const struct snd_ratnum analog_clock = { .num = ANALOG_CLOCK, @@ -393,8 +393,8 @@ static int snd_bt87x_set_analog_hw(struct snd_bt87x *chip, struct snd_pcm_runtim };
chip->reg_control &= ~(CTL_DA_IOM_DA | CTL_A_PWRDN); - runtime->hw = snd_bt87x_analog_hw; - return snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + substream->runtime->hw = snd_bt87x_analog_hw; + return snd_pcm_hw_constraint_ratnums(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &constraint_rates); }
@@ -410,7 +410,7 @@ static int snd_bt87x_pcm_open(struct snd_pcm_substream *substream) if (substream->pcm->device == DEVICE_DIGITAL) err = snd_bt87x_set_digital_hw(chip, runtime); else - err = snd_bt87x_set_analog_hw(chip, runtime); + err = snd_bt87x_set_analog_hw(chip, substream); if (err < 0) goto _error;
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index bee4710916c4..5563cfed12fa 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -577,7 +577,7 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; - if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) + if ((err = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) return err; snd_pcm_set_sync(substream);
@@ -671,7 +671,7 @@ static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substre if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; //snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_capture_period_sizes); - if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) + if ((err = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) return err; return 0; } diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 598446348da6..55a1e137456c 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -1647,7 +1647,7 @@ static int snd_cmipci_playback_open(struct snd_pcm_substream *substream) SNDRV_PCM_RATE_96000; runtime->hw.rate_max = 96000; } else if (cm->chip_version == 55) { - err = snd_pcm_hw_constraint_list(runtime, 0, + err = snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); if (err < 0) return err; @@ -1672,7 +1672,7 @@ static int snd_cmipci_capture_open(struct snd_pcm_substream *substream) runtime->hw.rate_min = 41000; runtime->hw.rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000; } else if (cm->chip_version == 55) { - err = snd_pcm_hw_constraint_list(runtime, 0, + err = snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); if (err < 0) return err; @@ -1697,11 +1697,11 @@ static int snd_cmipci_playback2_open(struct snd_pcm_substream *substream) if (cm->can_multi_ch) { runtime->hw.channels_max = cm->max_channels; if (cm->max_channels == 4) - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_channels_4); + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_channels_4); else if (cm->max_channels == 6) - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_channels_6); + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_channels_6); else if (cm->max_channels == 8) - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_channels_8); + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_channels_8); } } mutex_unlock(&cm->open_mutex); @@ -1710,7 +1710,7 @@ static int snd_cmipci_playback2_open(struct snd_pcm_substream *substream) SNDRV_PCM_RATE_96000; runtime->hw.rate_max = 96000; } else if (cm->chip_version == 55) { - err = snd_pcm_hw_constraint_list(runtime, 0, + err = snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); if (err < 0) return err; @@ -1733,7 +1733,7 @@ static int snd_cmipci_playback_spdif_open(struct snd_pcm_substream *substream) runtime->hw = snd_cmipci_playback_spdif; if (cm->chip_version >= 37) { runtime->hw.formats |= SNDRV_PCM_FMTBIT_S32_LE; - snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + snd_pcm_hw_constraint_msbits(substream, 0, 32, 24); } if (cm->can_96k) { runtime->hw.rates |= SNDRV_PCM_RATE_88200 | diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index bf3bb70ffaf9..8053ec410ff7 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -884,7 +884,7 @@ static int snd_cs4281_playback_open(struct snd_pcm_substream *substream) /* should be detected from the AC'97 layer, but it seems that although CS4297A rev B reports 18-bit ADC resolution, samples are 20-bit */ - snd_pcm_hw_constraint_msbits(runtime, 0, 32, 20); + snd_pcm_hw_constraint_msbits(substream, 0, 32, 20); return 0; }
@@ -903,7 +903,7 @@ static int snd_cs4281_capture_open(struct snd_pcm_substream *substream) /* should be detected from the AC'97 layer, but it seems that although CS4297A rev B reports 18-bit ADC resolution, samples are 20-bit */ - snd_pcm_hw_constraint_msbits(runtime, 0, 32, 20); + snd_pcm_hw_constraint_msbits(substream, 0, 32, 20); return 0; }
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 37f516e6a5c2..c7375c9b47b8 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -1511,7 +1511,7 @@ static int _cs46xx_playback_open_channel (struct snd_pcm_substream *substream,in cpcm->pcm_channel_id = pcm_channel_id;
- snd_pcm_hw_constraint_list(runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_sizes);
@@ -1594,7 +1594,7 @@ static int snd_cs46xx_capture_open(struct snd_pcm_substream *substream) chip->active_ctrl(chip, 1);
#ifdef CONFIG_SND_CS46XX_NEW_DSP - snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_sizes); #endif diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 9bd67ac33657..ae0d864aed5d 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -301,7 +301,7 @@ static int pcm_open(struct snd_pcm_substream *substream, snd_pcm_set_sync(substream);
/* Only mono and any even number of channels are allowed */ - if ((err = snd_pcm_hw_constraint_list(runtime, 0, + if ((err = snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &pipe->constr)) < 0) return err; @@ -314,16 +314,16 @@ static int pcm_open(struct snd_pcm_substream *substream, /* The hw accesses memory in chunks 32 frames long and they should be 32-bytes-aligned. It's not a requirement, but it seems that IRQs are generated with a resolution of 32 frames. Thus we need the following */ - if ((err = snd_pcm_hw_constraint_step(runtime, 0, + if ((err = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32)) < 0) return err; - if ((err = snd_pcm_hw_constraint_step(runtime, 0, + if ((err = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 32)) < 0) return err;
- if ((err = snd_pcm_hw_rule_add(substream->runtime, 0, + if ((err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, hw_rule_sample_rate, chip, SNDRV_PCM_HW_PARAM_RATE, -1)) < 0) @@ -361,12 +361,12 @@ static int pcm_analog_in_open(struct snd_pcm_substream *substream) if ((err = pcm_open(substream, num_analog_busses_in(chip) - substream->number)) < 0) return err; - if ((err = snd_pcm_hw_rule_add(substream->runtime, 0, + if ((err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_capture_channels_by_format, NULL, SNDRV_PCM_HW_PARAM_FORMAT, -1)) < 0) return err; - if ((err = snd_pcm_hw_rule_add(substream->runtime, 0, + if ((err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_FORMAT, hw_rule_capture_format_by_channels, NULL, SNDRV_PCM_HW_PARAM_CHANNELS, -1)) < 0) @@ -389,13 +389,13 @@ static int pcm_analog_out_open(struct snd_pcm_substream *substream) #endif if ((err = pcm_open(substream, max_channels - substream->number)) < 0) return err; - if ((err = snd_pcm_hw_rule_add(substream->runtime, 0, + if ((err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_playback_channels_by_format, NULL, SNDRV_PCM_HW_PARAM_FORMAT, -1)) < 0) return err; - if ((err = snd_pcm_hw_rule_add(substream->runtime, 0, + if ((err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_FORMAT, hw_rule_playback_format_by_channels, NULL, @@ -426,12 +426,12 @@ static int pcm_digital_in_open(struct snd_pcm_substream *substream) if (err < 0) goto din_exit;
- if ((err = snd_pcm_hw_rule_add(substream->runtime, 0, + if ((err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_capture_channels_by_format, NULL, SNDRV_PCM_HW_PARAM_FORMAT, -1)) < 0) goto din_exit; - if ((err = snd_pcm_hw_rule_add(substream->runtime, 0, + if ((err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_FORMAT, hw_rule_capture_format_by_channels, NULL, SNDRV_PCM_HW_PARAM_CHANNELS, -1)) < 0) @@ -463,13 +463,13 @@ static int pcm_digital_out_open(struct snd_pcm_substream *substream) if (err < 0) goto dout_exit;
- if ((err = snd_pcm_hw_rule_add(substream->runtime, 0, + if ((err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_playback_channels_by_format, NULL, SNDRV_PCM_HW_PARAM_FORMAT, -1)) < 0) goto dout_exit; - if ((err = snd_pcm_hw_rule_add(substream->runtime, 0, + if ((err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_FORMAT, hw_rule_playback_format_by_channels, NULL, SNDRV_PCM_HW_PARAM_CHANNELS, diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index d9a12cd01647..a223f3ae3c58 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -374,7 +374,7 @@ static int snd_emu10k1x_playback_open(struct snd_pcm_substream *substream) if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) { return err; } - if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) + if ((err = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) return err;
epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); @@ -552,7 +552,7 @@ static int snd_emu10k1x_pcm_open_capture(struct snd_pcm_substream *substream)
if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; - if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) + if ((err = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) return err;
epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index b2ddabb99438..8eea793adec3 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1136,7 +1136,7 @@ static int snd_emu10k1_playback_open(struct snd_pcm_substream *substream) sample_rate = 44100; else sample_rate = 48000; - err = snd_pcm_hw_rule_noresample(runtime, sample_rate); + err = snd_pcm_hw_rule_noresample(substream, sample_rate); if (err < 0) { kfree(epcm); return err; @@ -1185,8 +1185,8 @@ static int snd_emu10k1_capture_open(struct snd_pcm_substream *substream) runtime->hw = snd_emu10k1_capture; emu->capture_interrupt = snd_emu10k1_pcm_ac97adc_interrupt; emu->pcm_capture_substream = substream; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_capture_period_sizes); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_capture_rates); + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_capture_period_sizes); + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_capture_rates); return 0; }
@@ -1224,7 +1224,7 @@ static int snd_emu10k1_capture_mic_open(struct snd_pcm_substream *substream) runtime->hw.channels_min = 1; emu->capture_mic_interrupt = snd_emu10k1_pcm_ac97mic_interrupt; emu->pcm_capture_mic_substream = substream; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_capture_period_sizes); + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_capture_period_sizes); return 0; }
@@ -1332,7 +1332,7 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream) spin_unlock_irq(&emu->reg_lock); emu->capture_efx_interrupt = snd_emu10k1_pcm_efx_interrupt; emu->pcm_capture_efx_substream = substream; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_capture_period_sizes); + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_capture_period_sizes); return 0; }
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 3ccccdbc0029..4fe8fc2f77d3 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1105,10 +1105,10 @@ static int snd_ensoniq_playback1_open(struct snd_pcm_substream *substream) ensoniq->spdif_stream = ensoniq->spdif_default; spin_unlock_irq(&ensoniq->reg_lock); #ifdef CHIP1370 - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_es1370_hw_constraints_rates); #else - snd_pcm_hw_constraint_ratdens(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_ratdens(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_es1371_hw_constraints_dac_clock); #endif return 0; @@ -1128,10 +1128,10 @@ static int snd_ensoniq_playback2_open(struct snd_pcm_substream *substream) ensoniq->spdif_stream = ensoniq->spdif_default; spin_unlock_irq(&ensoniq->reg_lock); #ifdef CHIP1370 - snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_ratnums(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_es1370_hw_constraints_clock); #else - snd_pcm_hw_constraint_ratdens(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_ratdens(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_es1371_hw_constraints_dac_clock); #endif return 0; @@ -1147,10 +1147,10 @@ static int snd_ensoniq_capture_open(struct snd_pcm_substream *substream) runtime->hw = snd_ensoniq_capture; snd_pcm_set_sync(substream); #ifdef CHIP1370 - snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_ratnums(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_es1370_hw_constraints_clock); #else - snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_ratnums(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_es1371_hw_constraints_adc_clock); #endif return 0; diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index afc66347d162..f1e95e38388c 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -913,7 +913,7 @@ static int snd_es1938_capture_open(struct snd_pcm_substream *substream) return -EAGAIN; chip->capture_substream = substream; runtime->hw = snd_es1938_capture; - snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_ratnums(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_clocks); snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 0, 0xff00); return 0; @@ -938,7 +938,7 @@ static int snd_es1938_playback_open(struct snd_pcm_substream *substream) return -EINVAL; } runtime->hw = snd_es1938_playback; - snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_ratnums(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_clocks); snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 0, 0xff00); return 0; diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 5fa1861236f5..e6446bf66be9 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -1612,7 +1612,7 @@ static int snd_es1968_capture_open(struct snd_pcm_substream *substream) runtime->hw = snd_es1968_capture; runtime->hw.buffer_bytes_max = runtime->hw.period_bytes_max = calc_available_memory_size(chip) - 1024; /* keep MIXBUF size */ - snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES); + snd_pcm_hw_constraint_pow2(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES);
spin_lock_irq(&chip->substream_lock); list_add(&es->list, &chip->substream_list); diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 6279eb156e36..e3128a3448a1 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -651,11 +651,11 @@ static int snd_fm801_playback_open(struct snd_pcm_substream *substream)
chip->playback_substream = substream; runtime->hw = snd_fm801_playback; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); if (chip->multichannel) { runtime->hw.channels_max = 6; - snd_pcm_hw_constraint_list(runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_channels); } @@ -672,7 +672,7 @@ static int snd_fm801_capture_open(struct snd_pcm_substream *substream)
chip->capture_substream = substream; runtime->hw = snd_fm801_capture; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2026f1ccaf5a..d8995b3b1e3d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3735,7 +3735,7 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec, } mutex_unlock(&codec->spdif_mutex); } - return snd_pcm_hw_constraint_step(substream->runtime, 0, + return snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); } EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_open); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index ca2f2ecd1488..8fc7fd6ce1d8 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -624,9 +624,9 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) option needs to be disabled */ buff_step = 4;
- snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, buff_step); - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, buff_step); snd_hda_power_up(apcm->codec); if (hinfo->ops.open) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 45ae845e82df..561492e98547 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1202,7 +1202,7 @@ static int hdmi_pcm_open_no_pin(struct hda_pcm_stream *hinfo, runtime->hw.formats = hinfo->formats; runtime->hw.rates = hinfo->rates;
- snd_pcm_hw_constraint_step(substream->runtime, 0, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); return 0; } @@ -1298,7 +1298,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, runtime->hw.formats = hinfo->formats; runtime->hw.rates = hinfo->rates;
- snd_pcm_hw_constraint_step(substream->runtime, 0, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); unlock: mutex_unlock(&spec->pcm_lock); @@ -3244,11 +3244,11 @@ static int simple_playback_pcm_open(struct hda_pcm_stream *hinfo, }
if (hw_constraints_channels != NULL) { - snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_constraints_channels); } else { - snd_pcm_hw_constraint_step(substream->runtime, 0, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); }
diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 763eae80a148..df70ae48cae0 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -162,7 +162,7 @@ static int si3054_pcm_open(struct hda_pcm_stream *hinfo, .mask = 0, }; substream->runtime->hw.period_bytes_min = 80; - return snd_pcm_hw_constraint_list(substream->runtime, 0, + return snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); }
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index d54cd5143e9f..fc3f90c644f4 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -1130,8 +1130,8 @@ static int snd_ice1712_playback_pro_open(struct snd_pcm_substream *substream) ice->playback_pro_substream = substream; runtime->hw = snd_ice1712_playback_pro; snd_pcm_set_sync(substream); - snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); + snd_pcm_hw_constraint_msbits(substream, 0, 32, 24); + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); if (is_pro_rate_locked(ice)) { runtime->hw.rate_min = PRO_RATE_DEFAULT; runtime->hw.rate_max = PRO_RATE_DEFAULT; @@ -1151,8 +1151,8 @@ static int snd_ice1712_capture_pro_open(struct snd_pcm_substream *substream) ice->capture_pro_substream = substream; runtime->hw = snd_ice1712_capture_pro; snd_pcm_set_sync(substream); - snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); + snd_pcm_hw_constraint_msbits(substream, 0, 32, 24); + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); if (is_pro_rate_locked(ice)) { runtime->hw.rate_min = PRO_RATE_DEFAULT; runtime->hw.rate_max = PRO_RATE_DEFAULT; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index ef2367d86148..5f31cd58e4bc 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -972,7 +972,7 @@ static int set_rate_constraints(struct snd_ice1712 *ice, runtime->hw.rate_min = ice->hw_rates->list[0]; runtime->hw.rate_max = ice->hw_rates->list[ice->hw_rates->count - 1]; runtime->hw.rates = SNDRV_PCM_RATE_KNOT; - return snd_pcm_hw_constraint_list(runtime, 0, + return snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, ice->hw_rates); } @@ -1011,7 +1011,7 @@ static int snd_vt1724_playback_pro_open(struct snd_pcm_substream *substream) ice->playback_pro_substream = substream; runtime->hw = snd_vt1724_playback_pro; snd_pcm_set_sync(substream); - snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + snd_pcm_hw_constraint_msbits(substream, 0, 32, 24); set_rate_constraints(ice, substream); mutex_lock(&ice->open_mutex); /* calculate the currently available channels */ @@ -1023,11 +1023,11 @@ static int snd_vt1724_playback_pro_open(struct snd_pcm_substream *substream) chs = (chs + 1) * 2; runtime->hw.channels_max = chs; if (chs > 2) /* channels must be even */ - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); mutex_unlock(&ice->open_mutex); - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, VT1724_BUFFER_ALIGN); - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); constrain_rate_if_locked(substream); if (ice->pro_open) @@ -1044,11 +1044,11 @@ static int snd_vt1724_capture_pro_open(struct snd_pcm_substream *substream) ice->capture_pro_substream = substream; runtime->hw = snd_vt1724_2ch_stereo; snd_pcm_set_sync(substream); - snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + snd_pcm_hw_constraint_msbits(substream, 0, 32, 24); set_rate_constraints(ice, substream); - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, VT1724_BUFFER_ALIGN); - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); constrain_rate_if_locked(substream); if (ice->pro_open) @@ -1192,10 +1192,10 @@ static int snd_vt1724_playback_spdif_open(struct snd_pcm_substream *substream) } else runtime->hw = snd_vt1724_spdif; snd_pcm_set_sync(substream); - snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + snd_pcm_hw_constraint_msbits(substream, 0, 32, 24); + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, VT1724_BUFFER_ALIGN); - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); constrain_rate_if_locked(substream); if (ice->spdif.ops.open) @@ -1229,10 +1229,10 @@ static int snd_vt1724_capture_spdif_open(struct snd_pcm_substream *substream) } else runtime->hw = snd_vt1724_spdif; snd_pcm_set_sync(substream); - snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + snd_pcm_hw_constraint_msbits(substream, 0, 32, 24); + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, VT1724_BUFFER_ALIGN); - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); constrain_rate_if_locked(substream); if (ice->spdif.ops.open) @@ -1378,7 +1378,7 @@ static int snd_vt1724_playback_indep_open(struct snd_pcm_substream *substream) ice->playback_con_substream_ds[substream->number] = substream; runtime->hw = snd_vt1724_2ch_stereo; snd_pcm_set_sync(substream); - snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + snd_pcm_hw_constraint_msbits(substream, 0, 32, 24); set_rate_constraints(ice, substream); return 0; } diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 35903d1a1cbd..6b926a526be6 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1116,24 +1116,24 @@ static int snd_intel8x0_playback_open(struct snd_pcm_substream *substream)
if (chip->multi8) { runtime->hw.channels_max = 8; - snd_pcm_hw_constraint_list(runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_channels8); } else if (chip->multi6) { runtime->hw.channels_max = 6; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_channels6); } else if (chip->multi4) { runtime->hw.channels_max = 4; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_channels4); } if (chip->dra) { - snd_ac97_pcm_double_rate_rules(runtime); + snd_ac97_pcm_double_rate_rules(substream); } if (chip->smp20bit) { runtime->hw.formats |= SNDRV_PCM_FMTBIT_S32_LE; - snd_pcm_hw_constraint_msbits(runtime, 0, 32, 20); + snd_pcm_hw_constraint_msbits(substream, 0, 32, 20); } return 0; } diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 13ef838b26c1..59df4e29d91c 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -605,7 +605,7 @@ static int snd_intel8x0m_pcm_open(struct snd_pcm_substream *substream, struct ic
ichdev->substream = substream; runtime->hw = snd_intel8x0m_stream; - err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + err = snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); if ( err < 0 ) return err; diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c index 684faaf40f31..1c3b352f007e 100644 --- a/sound/pci/lola/lola_pcm.c +++ b/sound/pci/lola/lola_pcm.c @@ -235,9 +235,9 @@ static int lola_pcm_open(struct snd_pcm_substream *substream) chip->ref_count_rate++; snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); /* period size = multiple of chip->granularity (8, 16 or 32 frames)*/ - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, chip->granularity); - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, chip->granularity); mutex_unlock(&chip->open_mutex); return 0; diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index a0bbb386dc25..96d856f2032c 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -763,8 +763,8 @@ static int snd_mixart_playback_open(struct snd_pcm_substream *subs)
runtime->private_data = stream;
- snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 64); + snd_pcm_hw_constraint_step(subs, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); + snd_pcm_hw_constraint_step(subs, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 64);
/* if a sample rate is already used, another stream cannot change */ if(mgr->ref_count_rate++) { @@ -844,8 +844,8 @@ static int snd_mixart_capture_open(struct snd_pcm_substream *subs)
runtime->private_data = stream;
- snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 64); + snd_pcm_hw_constraint_step(subs, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); + snd_pcm_hw_constraint_step(subs, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 64);
/* if a sample rate is already used, another stream cannot change */ if(mgr->ref_count_rate++) { diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 6cb689aa28c2..6a0263a09bdb 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -850,7 +850,7 @@ static void snd_nm256_setup_stream(struct nm256 *chip, struct nm256_stream *s, runtime->private_data = s; s->substream = substream;
- snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); }
diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index b2a3fcfe31d4..09dfb805c7de 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -148,21 +148,21 @@ static int oxygen_open(struct snd_pcm_substream *substream, } if (chip->model.pcm_hardware_filter) chip->model.pcm_hardware_filter(channel, &runtime->hw); - err = snd_pcm_hw_constraint_step(runtime, 0, + err = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); if (err < 0) return err; - err = snd_pcm_hw_constraint_step(runtime, 0, + err = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32); if (err < 0) return err; if (runtime->hw.formats & SNDRV_PCM_FMTBIT_S32_LE) { - err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + err = snd_pcm_hw_constraint_msbits(substream, 0, 32, 24); if (err < 0) return err; } if (runtime->hw.channels_max > 2) { - err = snd_pcm_hw_constraint_step(runtime, 0, + err = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); if (err < 0) diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 751f9744b089..9418ea63c661 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1056,9 +1056,9 @@ static int pcxhr_open(struct snd_pcm_substream *subs) runtime->private_data = stream;
/* better get a divisor of granularity values (96 or 192) */ - snd_pcm_hw_constraint_step(runtime, 0, + snd_pcm_hw_constraint_step(subs, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 32); - snd_pcm_hw_constraint_step(runtime, 0, + snd_pcm_hw_constraint_step(subs, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32); snd_pcm_set_sync(subs);
diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index 54f3e39f97f5..527e64bbbef3 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -816,13 +816,13 @@ static const struct snd_pcm_hw_constraint_list hw_constraints_period_bytes = { .mask = 0 };
-static void snd_rme32_set_buffer_constraint(struct rme32 *rme32, struct snd_pcm_runtime *runtime) +static void snd_rme32_set_buffer_constraint(struct rme32 *rme32, struct snd_pcm_substream *substream) { if (! rme32->fullduplex_mode) { - snd_pcm_hw_constraint_single(runtime, + snd_pcm_hw_constraint_single(substream->runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, RME32_BUFFER_SIZE); - snd_pcm_hw_constraint_list(runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_bytes); } @@ -862,7 +862,7 @@ static int snd_rme32_playback_spdif_open(struct snd_pcm_substream *substream) runtime->hw.rate_max = rate; }
- snd_rme32_set_buffer_constraint(rme32, runtime); + snd_rme32_set_buffer_constraint(rme32, substream);
rme32->wcreg_spdif_stream = rme32->wcreg_spdif; rme32->spdif_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; @@ -904,7 +904,7 @@ static int snd_rme32_capture_spdif_open(struct snd_pcm_substream *substream) runtime->hw.rate_max = rate; }
- snd_rme32_set_buffer_constraint(rme32, runtime); + snd_rme32_set_buffer_constraint(rme32, substream);
return 0; } @@ -940,7 +940,7 @@ snd_rme32_playback_adat_open(struct snd_pcm_substream *substream) runtime->hw.rate_max = rate; }
- snd_rme32_set_buffer_constraint(rme32, runtime); + snd_rme32_set_buffer_constraint(rme32, substream); return 0; }
@@ -974,7 +974,7 @@ snd_rme32_capture_adat_open(struct snd_pcm_substream *substream) rme32->capture_substream = substream; spin_unlock_irq(&rme32->lock);
- snd_rme32_set_buffer_constraint(rme32, runtime); + snd_rme32_set_buffer_constraint(rme32, substream); return 0; }
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 66082e9f526d..61bb8220e740 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -1154,19 +1154,20 @@ static const struct snd_pcm_hw_constraint_list hw_constraints_period_bytes = {
static void rme96_set_buffer_size_constraint(struct rme96 *rme96, - struct snd_pcm_runtime *runtime) + struct snd_pcm_substream *substream) { unsigned int size;
- snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, RME96_BUFFER_SIZE); if ((size = rme96->playback_periodsize) != 0 || (size = rme96->capture_periodsize) != 0) - snd_pcm_hw_constraint_single(runtime, + snd_pcm_hw_constraint_single(substream->runtime, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, size); else - snd_pcm_hw_constraint_list(runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_bytes); } @@ -1199,7 +1200,7 @@ snd_rme96_playback_spdif_open(struct snd_pcm_substream *substream) runtime->hw.rate_min = rate; runtime->hw.rate_max = rate; } - rme96_set_buffer_size_constraint(rme96, runtime); + rme96_set_buffer_size_constraint(rme96, substream);
rme96->wcreg_spdif_stream = rme96->wcreg_spdif; rme96->spdif_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; @@ -1236,7 +1237,7 @@ snd_rme96_capture_spdif_open(struct snd_pcm_substream *substream) rme96->capture_substream = substream; spin_unlock_irq(&rme96->lock); - rme96_set_buffer_size_constraint(rme96, runtime); + rme96_set_buffer_size_constraint(rme96, substream); return 0; }
@@ -1268,7 +1269,7 @@ snd_rme96_playback_adat_open(struct snd_pcm_substream *substream) runtime->hw.rate_min = rate; runtime->hw.rate_max = rate; } - rme96_set_buffer_size_constraint(rme96, runtime); + rme96_set_buffer_size_constraint(rme96, substream); return 0; }
@@ -1303,7 +1304,7 @@ snd_rme96_capture_adat_open(struct snd_pcm_substream *substream) rme96->capture_substream = substream; spin_unlock_irq(&rme96->lock);
- rme96_set_buffer_size_constraint(rme96, runtime); + rme96_set_buffer_size_constraint(rme96, substream); return 0; }
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 4cf879c42dc4..7677bcdc9142 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -4510,27 +4510,27 @@ static int snd_hdsp_playback_open(struct snd_pcm_substream *substream)
spin_unlock_irq(&hdsp->lock);
- snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hdsp_hw_constraints_period_sizes); + snd_pcm_hw_constraint_msbits(substream, 0, 32, 24); + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hdsp_hw_constraints_period_sizes); if (hdsp->clock_source_locked) { runtime->hw.rate_min = runtime->hw.rate_max = hdsp->system_sample_rate; } else if (hdsp->io_type == H9632) { runtime->hw.rate_max = 192000; runtime->hw.rates = SNDRV_PCM_RATE_KNOT; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hdsp_hw_constraints_9632_sample_rates); + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &hdsp_hw_constraints_9632_sample_rates); } if (hdsp->io_type == H9632) { runtime->hw.channels_min = hdsp->qs_out_channels; runtime->hw.channels_max = hdsp->ss_out_channels; }
- snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_hdsp_hw_rule_out_channels, hdsp, SNDRV_PCM_HW_PARAM_CHANNELS, -1); - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_hdsp_hw_rule_out_channels_rate, hdsp, SNDRV_PCM_HW_PARAM_RATE, -1); - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, snd_hdsp_hw_rule_rate_out_channels, hdsp, SNDRV_PCM_HW_PARAM_CHANNELS, -1);
@@ -4587,22 +4587,22 @@ static int snd_hdsp_capture_open(struct snd_pcm_substream *substream)
spin_unlock_irq(&hdsp->lock);
- snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hdsp_hw_constraints_period_sizes); + snd_pcm_hw_constraint_msbits(substream, 0, 32, 24); + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hdsp_hw_constraints_period_sizes); if (hdsp->io_type == H9632) { runtime->hw.channels_min = hdsp->qs_in_channels; runtime->hw.channels_max = hdsp->ss_in_channels; runtime->hw.rate_max = 192000; runtime->hw.rates = SNDRV_PCM_RATE_KNOT; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hdsp_hw_constraints_9632_sample_rates); + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &hdsp_hw_constraints_9632_sample_rates); } - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_hdsp_hw_rule_in_channels, hdsp, SNDRV_PCM_HW_PARAM_CHANNELS, -1); - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_hdsp_hw_rule_in_channels_rate, hdsp, SNDRV_PCM_HW_PARAM_RATE, -1); - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, snd_hdsp_hw_rule_rate_in_channels, hdsp, SNDRV_PCM_HW_PARAM_CHANNELS, -1); return 0; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 8d900c132f0f..e6ed2c1ee598 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6071,8 +6071,8 @@ static int snd_hdspm_open(struct snd_pcm_substream *substream)
spin_unlock_irq(&hdspm->lock);
- snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); - snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + snd_pcm_hw_constraint_msbits(substream, 0, 32, 24); + snd_pcm_hw_constraint_pow2(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
switch (hdspm->io_type) { case AIO: @@ -6097,22 +6097,22 @@ static int snd_hdspm_open(struct snd_pcm_substream *substream)
if (AES32 == hdspm->io_type) { runtime->hw.rates |= SNDRV_PCM_RATE_KNOT; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &hdspm_hw_constraints_aes32_sample_rates); } else { - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, (playback ? snd_hdspm_hw_rule_rate_out_channels : snd_hdspm_hw_rule_rate_in_channels), hdspm, SNDRV_PCM_HW_PARAM_CHANNELS, -1); }
- snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, (playback ? snd_hdspm_hw_rule_out_channels : snd_hdspm_hw_rule_in_channels), hdspm, SNDRV_PCM_HW_PARAM_CHANNELS, -1);
- snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, (playback ? snd_hdspm_hw_rule_out_channels_rate : snd_hdspm_hw_rule_in_channels_rate), hdspm, SNDRV_PCM_HW_PARAM_RATE, -1); diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 4df992e846f2..0f88e9381325 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -2287,15 +2287,15 @@ static int snd_rme9652_playback_open(struct snd_pcm_substream *substream)
spin_unlock_irq(&rme9652->lock);
- snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_period_sizes); - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_constraint_msbits(substream, 0, 32, 24); + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_period_sizes); + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_rme9652_hw_rule_channels, rme9652, SNDRV_PCM_HW_PARAM_CHANNELS, -1); - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_rme9652_hw_rule_channels_rate, rme9652, SNDRV_PCM_HW_PARAM_RATE, -1); - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, snd_rme9652_hw_rule_rate_channels, rme9652, SNDRV_PCM_HW_PARAM_CHANNELS, -1);
@@ -2347,15 +2347,15 @@ static int snd_rme9652_capture_open(struct snd_pcm_substream *substream)
spin_unlock_irq(&rme9652->lock);
- snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_period_sizes); - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_constraint_msbits(substream, 0, 32, 24); + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_period_sizes); + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_rme9652_hw_rule_channels, rme9652, SNDRV_PCM_HW_PARAM_CHANNELS, -1); - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_rme9652_hw_rule_channels_rate, rme9652, SNDRV_PCM_HW_PARAM_RATE, -1); - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, snd_rme9652_hw_rule_rate_channels, rme9652, SNDRV_PCM_HW_PARAM_CHANNELS, -1); return 0; diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 7de10997775f..3fc098edde47 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -796,7 +796,7 @@ static int snd_sonicvibes_playback_open(struct snd_pcm_substream *substream) sonic->mode |= SV_MODE_PLAY; sonic->playback_substream = substream; runtime->hw = snd_sonicvibes_playback; - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, snd_sonicvibes_hw_constraint_dac_rate, NULL, SNDRV_PCM_HW_PARAM_RATE, -1); + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, snd_sonicvibes_hw_constraint_dac_rate, NULL, SNDRV_PCM_HW_PARAM_RATE, -1); return 0; }
@@ -808,7 +808,7 @@ static int snd_sonicvibes_capture_open(struct snd_pcm_substream *substream) sonic->mode |= SV_MODE_CAPTURE; sonic->capture_substream = substream; runtime->hw = snd_sonicvibes_capture; - snd_pcm_hw_constraint_ratdens(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_ratdens(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_sonicvibes_hw_constraints_adc_clock); return 0; } diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index fd1f2f9cfbc3..143cfaf17449 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1201,7 +1201,7 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev, return err;
if (use_src) { - err = snd_pcm_hw_rule_noresample(runtime, 48000); + err = snd_pcm_hw_rule_noresample(substream, 48000); if (err < 0) return err; } @@ -1279,7 +1279,7 @@ static int snd_via8233_multi_open(struct snd_pcm_substream *substream) return err; substream->runtime->hw.channels_max = 6; if (chip->revision == VIA_REV_8233A) - snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_channels); return 0; diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 30253306f67c..2c8d9902bf75 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -738,7 +738,7 @@ static int snd_via82xx_modem_pcm_open(struct via82xx_modem *chip, struct viadev
runtime->hw = snd_via82xx_hw; - if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + if ((err = snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0) return err;
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index cacc6a9d14c8..ea5e7aeba44f 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -885,7 +885,7 @@ static int snd_ymfpci_playback_open_1(struct snd_pcm_substream *substream) 5334, UINT_MAX); if (err < 0) return err; - err = snd_pcm_hw_rule_noresample(runtime, 48000); + err = snd_pcm_hw_rule_noresample(substream, 48000); if (err < 0) return err;
@@ -1010,7 +1010,7 @@ static int snd_ymfpci_capture_open(struct snd_pcm_substream *substream, 5334, UINT_MAX); if (err < 0) return err; - err = snd_pcm_hw_rule_noresample(runtime, 48000); + err = snd_pcm_hw_rule_noresample(substream, 48000); if (err < 0) return err;
diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index aa082131fb90..c39f6963fb4e 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -117,7 +117,7 @@ static int axi_i2s_startup(struct snd_pcm_substream *substream,
regmap_write(i2s->regmap, AXI_I2S_REG_RESET, mask);
- ret = snd_pcm_hw_constraint_ratnums(substream->runtime, 0, + ret = snd_pcm_hw_constraint_ratnums(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &i2s->rate_constraints); if (ret) diff --git a/sound/soc/adi/axi-spdif.c b/sound/soc/adi/axi-spdif.c index 9b3d81c41c8c..710adaaae3dd 100644 --- a/sound/soc/adi/axi-spdif.c +++ b/sound/soc/adi/axi-spdif.c @@ -120,7 +120,7 @@ static int axi_spdif_startup(struct snd_pcm_substream *substream, struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai); int ret;
- ret = snd_pcm_hw_constraint_ratnums(substream->runtime, 0, + ret = snd_pcm_hw_constraint_ratnums(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &spdif->rate_constraints); if (ret) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index e65e007fc604..2d50fa3a039d 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -255,9 +255,9 @@ static int cz_da7219_play_startup(struct snd_pcm_substream *substream) */
runtime->hw.channels_max = DUAL_CHANNEL; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
machine->play_i2s_instance = I2S_SP_INSTANCE; @@ -276,9 +276,9 @@ static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) */
runtime->hw.channels_max = DUAL_CHANNEL; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
machine->cap_i2s_instance = I2S_SP_INSTANCE; @@ -298,9 +298,9 @@ static int cz_max_startup(struct snd_pcm_substream *substream) */
runtime->hw.channels_max = DUAL_CHANNEL; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
machine->play_i2s_instance = I2S_BT_INSTANCE; @@ -319,9 +319,9 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) */
runtime->hw.channels_max = DUAL_CHANNEL; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
machine->cap_i2s_instance = I2S_BT_INSTANCE; @@ -340,9 +340,9 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) */
runtime->hw.channels_max = DUAL_CHANNEL; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
machine->cap_i2s_instance = I2S_SP_INSTANCE; @@ -367,9 +367,9 @@ static int cz_rt5682_play_startup(struct snd_pcm_substream *substream) */
runtime->hw.channels_max = DUAL_CHANNEL; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
machine->play_i2s_instance = I2S_SP_INSTANCE; @@ -388,9 +388,9 @@ static int cz_rt5682_cap_startup(struct snd_pcm_substream *substream) */
runtime->hw.channels_max = DUAL_CHANNEL; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
machine->cap_i2s_instance = I2S_SP_INSTANCE; @@ -410,9 +410,9 @@ static int cz_rt5682_max_startup(struct snd_pcm_substream *substream) */
runtime->hw.channels_max = DUAL_CHANNEL; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
machine->play_i2s_instance = I2S_BT_INSTANCE; @@ -431,9 +431,9 @@ static int cz_rt5682_dmic0_startup(struct snd_pcm_substream *substream) */
runtime->hw.channels_max = DUAL_CHANNEL; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
machine->cap_i2s_instance = I2S_BT_INSTANCE; @@ -452,9 +452,9 @@ static int cz_rt5682_dmic1_startup(struct snd_pcm_substream *substream) */
runtime->hw.channels_max = DUAL_CHANNEL; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
machine->cap_i2s_instance = I2S_SP_INSTANCE; diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 6a63e8797a0b..bd117faa7dc6 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -295,7 +295,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, dir_mask = SSC_DIR_MASK_CAPTURE; }
- ret = snd_pcm_hw_rule_add(substream->runtime, 0, + ret = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, atmel_ssc_hw_rule_rate, ssc_p, diff --git a/sound/soc/bcm/bcm63xx-pcm-whistler.c b/sound/soc/bcm/bcm63xx-pcm-whistler.c index 7ec8559d53a2..eb05bdf0f640 100644 --- a/sound/soc/bcm/bcm63xx-pcm-whistler.c +++ b/sound/soc/bcm/bcm63xx-pcm-whistler.c @@ -211,12 +211,12 @@ static int bcm63xx_pcm_open(struct snd_soc_component *component, struct bcm63xx_runtime_data *prtd;
runtime->hw = bcm63xx_pcm_hardware; - ret = snd_pcm_hw_constraint_step(runtime, 0, + ret = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); if (ret) goto out;
- ret = snd_pcm_hw_constraint_step(runtime, 0, + ret = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32); if (ret) goto out; diff --git a/sound/soc/bcm/cygnus-pcm.c b/sound/soc/bcm/cygnus-pcm.c index 56b71b965624..8117be6b98b4 100644 --- a/sound/soc/bcm/cygnus-pcm.c +++ b/sound/soc/bcm/cygnus-pcm.c @@ -582,7 +582,6 @@ static int cygnus_pcm_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); - struct snd_pcm_runtime *runtime = substream->runtime; struct cygnus_aio_port *aio; int ret;
@@ -594,12 +593,12 @@ static int cygnus_pcm_open(struct snd_soc_component *component,
snd_soc_set_runtime_hwparams(substream, &cygnus_pcm_hw);
- ret = snd_pcm_hw_constraint_step(runtime, 0, + ret = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, PERIOD_BYTES_MIN); if (ret < 0) return ret;
- ret = snd_pcm_hw_constraint_step(runtime, 0, + ret = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, PERIOD_BYTES_MIN); if (ret < 0) return ret; diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index 6e634b448293..db425c72891f 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -754,7 +754,7 @@ static int cygnus_ssp_startup(struct snd_pcm_substream *substream, substream->runtime->hw.rate_min = CYGNUS_RATE_MIN; substream->runtime->hw.rate_max = CYGNUS_RATE_MAX;
- snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &cygnus_rate_constraint); return 0; } diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index f37ab7eda615..efb46ca4dc6d 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -363,7 +363,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, static int ad193x_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - return snd_pcm_hw_constraint_list(substream->runtime, 0, + return snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, &constr); } diff --git a/sound/soc/codecs/adau1372.c b/sound/soc/codecs/adau1372.c index 6811a8b3866d..54ec744b95e6 100644 --- a/sound/soc/codecs/adau1372.c +++ b/sound/soc/codecs/adau1372.c @@ -755,7 +755,7 @@ static int adau1372_startup(struct snd_pcm_substream *substream, struct snd_soc_ { struct adau1372 *adau1372 = snd_soc_dai_get_drvdata(dai);
- snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &adau1372->rate_constraints);
return 0; diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 85a1d00894a9..574a2ce1b900 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -555,7 +555,7 @@ static int ak4458_startup(struct snd_pcm_substream *substream, { int ret;
- ret = snd_pcm_hw_constraint_list(substream->runtime, 0, + ret = snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &ak4458_rate_constraints);
diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index fe208cfdd3ba..ee67c7aa8284 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -255,7 +255,7 @@ static void ak4613_dai_shutdown(struct snd_pcm_substream *substream, }
static void ak4613_hw_constraints(struct ak4613_priv *priv, - struct snd_pcm_runtime *runtime) + struct snd_pcm_substream *substream) { static const unsigned int ak4613_rates[] = { 32000, @@ -294,7 +294,7 @@ static void ak4613_hw_constraints(struct ak4613_priv *priv, constraint->count = i + 1; }
- snd_pcm_hw_constraint_list(runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, constraint); }
@@ -306,7 +306,7 @@ static int ak4613_dai_startup(struct snd_pcm_substream *substream,
priv->cnt++;
- ak4613_hw_constraints(priv, substream->runtime); + ak4613_hw_constraints(priv, substream);
return 0; } diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index 85bdd0534180..a7630218d11a 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -242,7 +242,7 @@ static const struct snd_pcm_hw_constraint_list ak5558_rate_constraints = { static int ak5558_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - return snd_pcm_hw_constraint_list(substream->runtime, 0, + return snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &ak5558_rate_constraints); } diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index 7ad7b733af9b..45449746e5c4 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -520,7 +520,7 @@ static const struct snd_pcm_hw_constraint_list cs35l33_constraints = { static int cs35l33_pcm_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &cs35l33_constraints); return 0; diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c index 110ee2d06358..fe90dabf97d8 100644 --- a/sound/soc/codecs/cs35l34.c +++ b/sound/soc/codecs/cs35l34.c @@ -577,7 +577,7 @@ static int cs35l34_pcm_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) {
- snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &cs35l34_constraints); return 0; } diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index 55d529aa0011..e06bafcbd068 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -609,7 +609,7 @@ static int cs35l35_pcm_startup(struct snd_pcm_substream *substream, if (!substream->runtime) return 0;
- snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &cs35l35_constraints);
regmap_update_bits(cs35l35->regmap, CS35L35_AMP_INP_DRV_CTL, @@ -637,7 +637,7 @@ static int cs35l35_pdm_startup(struct snd_pcm_substream *substream, if (!substream->runtime) return 0;
- snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &cs35l35_pdm_constraints);
diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index 4451ca9f4916..2e8f06e0193d 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -963,7 +963,7 @@ static const struct snd_pcm_hw_constraint_list cs35l36_constraints = { static int cs35l36_pcm_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &cs35l36_constraints);
return 0; diff --git a/sound/soc/codecs/cs4234.c b/sound/soc/codecs/cs4234.c index 20126cc675b1..d95bf52cc2b6 100644 --- a/sound/soc/codecs/cs4234.c +++ b/sound/soc/codecs/cs4234.c @@ -505,7 +505,7 @@ static int cs4234_dai_startup(struct snd_pcm_substream *sub, struct snd_soc_dai for (i = 0; i < cs4234->rate_constraint.nrats; i++) cs4234->rate_dividers[i].num = cs4234->mclk_rate / CS4234_MCLK_SCALE;
- ret = snd_pcm_hw_constraint_ratnums(sub->runtime, 0, + ret = snd_pcm_hw_constraint_ratnums(sub, 0, SNDRV_PCM_HW_PARAM_RATE, &cs4234->rate_constraint); if (ret < 0) @@ -515,7 +515,7 @@ static int cs4234_dai_startup(struct snd_pcm_substream *sub, struct snd_soc_dai * MCLK/rate may be a valid ratio but out-of-spec (e.g. 24576000/64000) * so this rule limits the range of sample rate for given MCLK. */ - return snd_pcm_hw_rule_add(sub->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + return snd_pcm_hw_rule_add(sub, 0, SNDRV_PCM_HW_PARAM_RATE, cs4234_dai_rule_rate, cs4234, -1); }
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index c3f974ec78e5..7d9109884e86 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1144,7 +1144,7 @@ static const struct snd_pcm_hw_constraint_list constraints_12_24 = { static int cs42l73_pcm_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_12_24); return 0; diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 80bc7c10ed75..add2808ded06 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1444,7 +1444,7 @@ static const struct snd_pcm_hw_constraint_list cs43130_asp_constraints = { static int cs43130_pcm_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - return snd_pcm_hw_constraint_list(substream->runtime, 0, + return snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &cs43130_asp_constraints); } @@ -1461,7 +1461,7 @@ static const struct snd_pcm_hw_constraint_list cs43130_dop_constraints = { static int cs43130_dop_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - return snd_pcm_hw_constraint_list(substream->runtime, 0, + return snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &cs43130_dop_constraints); } diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index 3d67cbf9eaaa..3868394e969b 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -751,7 +751,7 @@ static const struct snd_pcm_hw_constraint_list src_constraints = { static int cs53l30_pcm_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &src_constraints);
return 0; diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 067757d1d70a..caf197915060 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -450,7 +450,7 @@ static int es8316_pcm_startup(struct snd_pcm_substream *substream, struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component);
if (es8316->sysclk_constraints.list) - snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &es8316->sysclk_constraints);
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 9632afc2d4d6..500614a6b979 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -463,7 +463,7 @@ static int es8328_startup(struct snd_pcm_substream *substream, struct es8328_priv *es8328 = snd_soc_component_get_drvdata(component);
if (es8328->master && es8328->sysclk_constraints) - snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, es8328->sysclk_constraints);
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 1567ba196ab9..1ac1f55bbff4 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -411,7 +411,7 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, if (ret) goto err;
- ret = snd_pcm_hw_constraint_eld(substream->runtime, hcp->eld); + ret = snd_pcm_hw_constraint_eld(substream, hcp->eld); if (ret) goto err;
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index bc30a1dc7530..41effa498eef 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1916,7 +1916,7 @@ static int max98090_dai_startup(struct snd_pcm_substream *substream, /* Remove 24-bit format support if it is not in right justified mode. */ if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_RIGHT_J) { substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; - snd_pcm_hw_constraint_msbits(substream->runtime, 0, 16, 16); + snd_pcm_hw_constraint_msbits(substream, 0, 16, 16); } return 0; } diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 09b2d730e9fd..51aebac49595 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -314,7 +314,7 @@ static int max9867_startup(struct snd_pcm_substream *substream, snd_soc_component_get_drvdata(dai->component);
if (max9867->constraints) - snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, max9867->constraints);
return 0; diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 4dc844f3c1fc..d1168759e60e 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -592,7 +592,7 @@ static int pcm512x_dai_startup_master(struct snd_pcm_substream *substream, }
if (pcm512x->pll_out) - return snd_pcm_hw_rule_add(substream->runtime, 0, + return snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, pcm512x_hw_rule_rate, pcm512x, @@ -613,7 +613,7 @@ static int pcm512x_dai_startup_master(struct snd_pcm_substream *substream, rats_no_pll->den_max = 128; rats_no_pll->den_step = 1;
- return snd_pcm_hw_constraint_ratnums(substream->runtime, 0, + return snd_pcm_hw_constraint_ratnums(substream, 0, SNDRV_PCM_HW_PARAM_RATE, constraints_no_pll); } @@ -639,7 +639,7 @@ static int pcm512x_dai_startup_slave(struct snd_pcm_substream *substream, PCM512x_SREF, PCM512x_SREF_BCK); }
- return snd_pcm_hw_constraint_list(substream->runtime, 0, + return snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_slave); } diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 76c77dc8ecf7..64df463c091a 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -804,7 +804,7 @@ int sigmadsp_restrict_params(struct sigmadsp *sigmadsp, if (sigmadsp->rate_constraints.count == 0) return 0;
- return snd_pcm_hw_constraint_list(substream->runtime, 0, + return snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &sigmadsp->rate_constraints); } EXPORT_SYMBOL_GPL(sigmadsp_restrict_params); diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 7964e922b07f..827c47487b50 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -330,7 +330,7 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, struct ssm2602_priv *ssm2602 = snd_soc_component_get_drvdata(component);
if (ssm2602->sysclk_constraints) { - snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, ssm2602->sysclk_constraints); } diff --git a/sound/soc/codecs/uda1334.c b/sound/soc/codecs/uda1334.c index 21ab8c5487ba..0a00f6bd58f3 100644 --- a/sound/soc/codecs/uda1334.c +++ b/sound/soc/codecs/uda1334.c @@ -103,7 +103,7 @@ static int uda1334_startup(struct snd_pcm_substream *substream, return -EINVAL; }
- snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &uda1334->rate_constraint);
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index c8b50aac6c18..f32a40a0737d 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -133,7 +133,7 @@ static int wm8523_startup(struct snd_pcm_substream *substream, return -EINVAL; }
- snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &wm8523->rate_constraint);
diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c index 81f858f6bd67..95d6030fa9a5 100644 --- a/sound/soc/codecs/wm8524.c +++ b/sound/soc/codecs/wm8524.c @@ -71,7 +71,7 @@ static int wm8524_startup(struct snd_pcm_substream *substream, return -EINVAL; }
- snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &wm8524->rate_constraint);
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index dcee7b2bd3d7..c1a8bc1f8048 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -531,7 +531,7 @@ static int wm8731_startup(struct snd_pcm_substream *substream, struct wm8731_priv *wm8731 = snd_soc_component_get_drvdata(dai->component);
if (wm8731->constraints) - snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, wm8731->constraints);
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 0e3994326936..800f073417fa 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -179,7 +179,7 @@ static int wm8741_startup(struct snd_pcm_substream *substream, struct wm8741_priv *wm8741 = snd_soc_component_get_drvdata(component);
if (wm8741->sysclk) - snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, wm8741->sysclk_constraints);
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 63d236ef5c4d..623b6521d62a 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -592,11 +592,11 @@ static int fsl_asrc_dai_startup(struct snd_pcm_substream *substream,
/* Odd channel number is not valid for older ASRC (channel_bits==3) */ if (asrc_priv->soc->channel_bits == 3) - snd_pcm_hw_constraint_step(substream->runtime, 0, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2);
- return snd_pcm_hw_constraint_list(substream->runtime, 0, + return snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &fsl_asrc_rate_constraints); }
diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index 600e0d670ca6..66e569e1d8b6 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -1400,7 +1400,7 @@ static const struct snd_pcm_hw_constraint_list easrc_rate_constraints = { static int fsl_easrc_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - return snd_pcm_hw_constraint_list(substream->runtime, 0, + return snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &easrc_rate_constraints); } diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 407a45e48eee..cbebd5e4b982 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -684,12 +684,12 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, * tx/rx maxburst */ if (sai->soc_data->use_edma) - snd_pcm_hw_constraint_step(substream->runtime, 0, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, tx ? sai->dma_params_tx.maxburst : sai->dma_params_rx.maxburst);
- ret = snd_pcm_hw_constraint_list(substream->runtime, 0, + ret = snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &fsl_sai_rate_constraints);
return ret; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2b57b60431bb..12beb4b20d77 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -644,7 +644,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * period. But SSI would still access fifo1 with an invalid data. */ if (ssi->use_dual_fifo) - snd_pcm_hw_constraint_step(substream->runtime, 0, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 2);
return 0; diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 6cb558165848..fe22ca177fbc 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -498,19 +498,18 @@ static int fsl_xcvr_prepare(struct snd_pcm_substream *substream, return 0; }
-static int fsl_xcvr_constr(const struct snd_pcm_substream *substream, +static int fsl_xcvr_constr(struct snd_pcm_substream *substream, const struct snd_pcm_hw_constraint_list *channels, const struct snd_pcm_hw_constraint_list *rates) { - struct snd_pcm_runtime *rt = substream->runtime; int ret;
- ret = snd_pcm_hw_constraint_list(rt, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - channels); + ret = snd_pcm_hw_constraint_list(substream, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, channels); if (ret < 0) return ret;
- ret = snd_pcm_hw_constraint_list(rt, 0, SNDRV_PCM_HW_PARAM_RATE, + ret = snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, rates); if (ret < 0) return ret; diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index cbdc0a2c09c5..3e66e59f37e9 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -52,7 +52,7 @@ static int imx_audmix_fe_startup(struct snd_pcm_substream *substream) int ret;
if (clk_rate % 24576000 == 0) { - ret = snd_pcm_hw_constraint_list(runtime, 0, + ret = snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &imx_audmix_rate_constraints); if (ret < 0) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index c2a5933bfcfc..1e8d3a1800af 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -117,13 +117,13 @@ static int kirkwood_dma_open(struct snd_soc_component *component, if (err < 0) return err;
- err = snd_pcm_hw_constraint_step(runtime, 0, + err = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, priv->burst); if (err < 0) return err;
- err = snd_pcm_hw_constraint_step(substream->runtime, 0, + err = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, priv->burst); if (err < 0) diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index 3cb2adf420bb..11a75a84e3d0 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -47,7 +47,7 @@ int mtk_afe_fe_startup(struct snd_pcm_substream *substream,
memif->substream = substream;
- snd_pcm_hw_constraint_step(substream->runtime, 0, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 16); /* enable agent */ mtk_regmap_update_bits(afe->regmap, memif->data->agent_disable_reg, diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c index 44a8d5cfb0aa..76e61d089657 100644 --- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c +++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c @@ -108,7 +108,7 @@ static int mt2701_cs42448_fe_ops_startup(struct snd_pcm_substream *substream) { int err;
- err = snd_pcm_hw_constraint_list(substream->runtime, 0, + err = snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &mt2701_cs42448_constraints_rates); if (err < 0) { diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index a4d26a6fc849..f9ce6f6f22ab 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -199,15 +199,15 @@ mt8183_da7219_max98357_startup(
struct snd_pcm_runtime *runtime = substream->runtime;
- snd_pcm_hw_constraint_list(runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); runtime->hw.channels_max = 2; - snd_pcm_hw_constraint_list(runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels);
runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; - snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + snd_pcm_hw_constraint_msbits(substream, 0, 16, 16);
return 0; } @@ -239,15 +239,15 @@ mt8183_da7219_max98357_bt_sco_startup(
struct snd_pcm_runtime *runtime = substream->runtime;
- snd_pcm_hw_constraint_list(runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); runtime->hw.channels_max = 1; - snd_pcm_hw_constraint_list(runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels);
runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; - snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + snd_pcm_hw_constraint_msbits(substream, 0, 16, 16);
return 0; } diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 94dcbd36c869..21d0264d75d3 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -139,15 +139,15 @@ mt8183_mt6358_startup(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
- snd_pcm_hw_constraint_list(runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); runtime->hw.channels_max = 2; - snd_pcm_hw_constraint_list(runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels);
runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; - snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + snd_pcm_hw_constraint_msbits(substream, 0, 16, 16);
return 0; } @@ -179,15 +179,15 @@ mt8183_mt6358_ts3a227_max98357_bt_sco_startup(
struct snd_pcm_runtime *runtime = substream->runtime;
- snd_pcm_hw_constraint_list(runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); runtime->hw.channels_max = 1; - snd_pcm_hw_constraint_list(runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels);
runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; - snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + snd_pcm_hw_constraint_msbits(substream, 0, 16, 16);
return 0; } diff --git a/sound/soc/meson/aiu-encoder-i2s.c b/sound/soc/meson/aiu-encoder-i2s.c index 932224552146..43c3cb695fa5 100644 --- a/sound/soc/meson/aiu-encoder-i2s.c +++ b/sound/soc/meson/aiu-encoder-i2s.c @@ -329,7 +329,7 @@ static int aiu_encoder_i2s_startup(struct snd_pcm_substream *substream, int ret;
/* Make sure the encoder gets either 2 or 8 channels */ - ret = snd_pcm_hw_constraint_list(substream->runtime, 0, + ret = snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_channel_constraints); if (ret) { diff --git a/sound/soc/meson/aiu-fifo.c b/sound/soc/meson/aiu-fifo.c index 4ad23267cace..04cfe68bf0de 100644 --- a/sound/soc/meson/aiu-fifo.c +++ b/sound/soc/meson/aiu-fifo.c @@ -140,13 +140,13 @@ int aiu_fifo_startup(struct snd_pcm_substream *substream, * Make sure the buffer and period size are multiple of the fifo burst * size */ - ret = snd_pcm_hw_constraint_step(substream->runtime, 0, + ret = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, fifo->fifo_block); if (ret) return ret;
- ret = snd_pcm_hw_constraint_step(substream->runtime, 0, + ret = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, fifo->fifo_block); if (ret) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index b2e867113226..90b00df0fd11 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -229,13 +229,13 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, * Make sure the buffer and period size are multiple of the FIFO * burst */ - ret = snd_pcm_hw_constraint_step(ss->runtime, 0, + ret = snd_pcm_hw_constraint_step(ss, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, AXG_FIFO_BURST); if (ret) return ret;
- ret = snd_pcm_hw_constraint_step(ss->runtime, 0, + ret = snd_pcm_hw_constraint_step(ss, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, AXG_FIFO_BURST); if (ret) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 8696a993c478..bb2cb135917e 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -992,7 +992,7 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream,
snd_soc_set_runtime_hwparams(substream, &rsnd_pcm_hardware);
- snd_pcm_hw_constraint_list(runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, constraint);
snd_pcm_hw_constraint_integer(runtime, @@ -1005,11 +1005,11 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, if (rsnd_rdai_is_clk_master(rdai)) { int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
- snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, rsnd_soc_hw_rule_rate, is_play ? &rdai->playback : &rdai->capture, SNDRV_PCM_HW_PARAM_CHANNELS, -1); - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, rsnd_soc_hw_rule_channels, is_play ? &rdai->playback : &rdai->capture, SNDRV_PCM_HW_PARAM_RATE, -1); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2df70ab851ea..65f8ea73bae7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -436,7 +436,7 @@ static void soc_pcm_set_msb(struct snd_pcm_substream *substream, int bits) if (!bits) return;
- ret = snd_pcm_hw_constraint_msbits(substream->runtime, 0, 0, bits); + ret = snd_pcm_hw_constraint_msbits(substream, 0, 0, bits); if (ret != 0) dev_warn(rtd->dev, "ASoC: Failed to set MSB %d: %d\n", bits, ret); diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index 2ed92c990b97..f7a1d5b8da9e 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -706,7 +706,7 @@ static int uni_player_startup(struct snd_pcm_substream *substream, return 0;
/* refine hw constraint in tdm mode */ - ret = snd_pcm_hw_rule_add(substream->runtime, 0, + ret = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, sti_uniperiph_fix_tdm_chan, player, SNDRV_PCM_HW_PARAM_CHANNELS, @@ -714,7 +714,7 @@ static int uni_player_startup(struct snd_pcm_substream *substream, if (ret < 0) return ret;
- return snd_pcm_hw_rule_add(substream->runtime, 0, + return snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_FORMAT, sti_uniperiph_fix_tdm_format, player, SNDRV_PCM_HW_PARAM_FORMAT, diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 136059331211..7d14d4c011e8 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -366,7 +366,7 @@ static int uni_reader_startup(struct snd_pcm_substream *substream, return 0;
/* refine hw constraint in tdm mode */ - ret = snd_pcm_hw_rule_add(substream->runtime, 0, + ret = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, sti_uniperiph_fix_tdm_chan, reader, SNDRV_PCM_HW_PARAM_CHANNELS, @@ -374,7 +374,7 @@ static int uni_reader_startup(struct snd_pcm_substream *substream, if (ret < 0) return ret;
- return snd_pcm_hw_rule_add(substream->runtime, 0, + return snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_FORMAT, sti_uniperiph_fix_tdm_format, reader, SNDRV_PCM_HW_PARAM_FORMAT, diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 6c13cc84b3fb..48436ab771e3 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -617,7 +617,7 @@ static int sun4i_codec_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card);
- snd_pcm_hw_constraint_list(substream->runtime, 0, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &sun4i_codec_constraints);
/* diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index 460924fc173f..c3abba92aabe 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -429,7 +429,7 @@ static int sun8i_codec_startup(struct snd_pcm_substream *substream, else return -EINVAL;
- return snd_pcm_hw_constraint_list(substream->runtime, 0, + return snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_RATE, list); }
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 573374b89b10..b49794e1d9f6 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -85,7 +85,7 @@ int tegra_pcm_open(struct snd_soc_component *component, snd_soc_set_runtime_hwparams(substream, &tegra_pcm_hardware);
/* Ensure period size is multiple of 8 */ - ret = snd_pcm_hw_constraint_step(substream->runtime, 0, + ret = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 0x8); if (ret) { dev_err(rtd->dev, "failed to set constraint %d\n", ret); diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index b94220306d1a..33549c94e6bd 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1512,7 +1512,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_CHANNELS, 0, max_channels);
- snd_pcm_hw_constraint_list(substream->runtime, + snd_pcm_hw_constraint_list(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &mcasp->chconstr[substream->stream]);
@@ -1521,7 +1521,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, * Only allow formats which require same amount of bits on the * bus as the currently running stream */ - ret = snd_pcm_hw_rule_add(substream->runtime, 0, + ret = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_FORMAT, davinci_mcasp_hw_rule_format_width, ruledata, @@ -1531,7 +1531,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, } else if (mcasp->slot_width) { /* Only allow formats require <= slot_width bits on the bus */ - ret = snd_pcm_hw_rule_add(substream->runtime, 0, + ret = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_FORMAT, davinci_mcasp_hw_rule_slot_width, ruledata, @@ -1545,14 +1545,14 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, * set constraints based on what we can provide. */ if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { - ret = snd_pcm_hw_rule_add(substream->runtime, 0, + ret = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, davinci_mcasp_hw_rule_rate, ruledata, SNDRV_PCM_HW_PARAM_FORMAT, -1); if (ret) return ret; - ret = snd_pcm_hw_rule_add(substream->runtime, 0, + ret = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_FORMAT, davinci_mcasp_hw_rule_format, ruledata, @@ -1561,7 +1561,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, return ret; }
- snd_pcm_hw_rule_add(substream->runtime, 0, + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, davinci_mcasp_hw_rule_min_periodsize, NULL, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 6025b30bbe77..b65c00ccfcdb 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -808,14 +808,14 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, * This applies only for the playback stream. */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_pcm_hw_rule_add(substream->runtime, 0, + snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, omap_mcbsp_hwrule_min_buffersize, mcbsp, SNDRV_PCM_HW_PARAM_CHANNELS, -1);
/* Make sure, that the period size is always even */ - snd_pcm_hw_constraint_step(substream->runtime, 0, + snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 2); }
diff --git a/sound/soc/uniphier/aio-dma.c b/sound/soc/uniphier/aio-dma.c index 3c1628a3a1ac..b659d09343d8 100644 --- a/sound/soc/uniphier/aio-dma.c +++ b/sound/soc/uniphier/aio-dma.c @@ -96,11 +96,9 @@ static irqreturn_t aiodma_irq(int irq, void *p) static int uniphier_aiodma_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - snd_soc_set_runtime_hwparams(substream, &uniphier_aiodma_hw);
- return snd_pcm_hw_constraint_step(runtime, 0, + return snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 256); }
diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index 1d59fb668c77..300f47984f5a 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -369,7 +369,7 @@ static int xlnx_formatter_pcm_open(struct snd_soc_component *component, runtime->private_data = stream_data;
/* Resize the period size divisible by 64 */ - err = snd_pcm_hw_constraint_step(runtime, 0, + err = snd_pcm_hw_constraint_step(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64); if (err) { dev_err(component->dev, diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 71f17f02f341..74997ff3e5c3 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -374,7 +374,7 @@ static int hiface_pcm_open(struct snd_pcm_substream *alsa_sub) alsa_rt->hw.rate_max = 384000;
/* explicit constraints needed as we added SNDRV_PCM_RATE_KNOT */ - ret = snd_pcm_hw_constraint_list(alsa_sub->runtime, 0, + ret = snd_pcm_hw_constraint_list(alsa_sub, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_extra_rates); if (ret < 0) { diff --git a/sound/usb/line6/capture.c b/sound/usb/line6/capture.c index 970c9bdce0b2..6161dd638729 100644 --- a/sound/usb/line6/capture.c +++ b/sound/usb/line6/capture.c @@ -222,7 +222,7 @@ static int snd_line6_capture_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_line6_pcm *line6pcm = snd_pcm_substream_chip(substream);
- err = snd_pcm_hw_constraint_ratdens(runtime, 0, + err = snd_pcm_hw_constraint_ratdens(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &line6pcm->properties->rates); if (err < 0) diff --git a/sound/usb/line6/playback.c b/sound/usb/line6/playback.c index 8233c61e23f1..1965395e9a2d 100644 --- a/sound/usb/line6/playback.c +++ b/sound/usb/line6/playback.c @@ -373,7 +373,7 @@ static int snd_line6_playback_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_line6_pcm *line6pcm = snd_pcm_substream_chip(substream);
- err = snd_pcm_hw_constraint_ratdens(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + err = snd_pcm_hw_constraint_ratdens(substream, 0, SNDRV_PCM_HW_PARAM_RATE, &line6pcm->properties->rates); if (err < 0) return err; diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 5834d1dc317e..a8a2f9011e52 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -641,7 +641,7 @@ static int set_stream_hw(struct ua101 *ua, struct snd_pcm_substream *substream, UINT_MAX); if (err < 0) return err; - err = snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24); + err = snd_pcm_hw_constraint_msbits(substream, 0, 32, 24); return err; }
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index e5311b6bb3f6..a213a5d590e2 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -933,9 +933,10 @@ static int hw_rule_periods_implicit_fb(struct snd_pcm_hw_params *params, * set up the runtime hardware information. */
-static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substream *subs) +static int setup_hw_info(struct snd_pcm_substream *substream, struct snd_usb_substream *subs) { - const struct audioformat *fp; + struct snd_pcm_runtime *runtime = substream->runtime; + struct audioformat *fp; unsigned int pt, ptmin; int param_period_time_if_needed = -1; int err; @@ -982,7 +983,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre if (err < 0) return err;
- err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, hw_rule_rate, subs, SNDRV_PCM_HW_PARAM_RATE, SNDRV_PCM_HW_PARAM_FORMAT, @@ -991,8 +992,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre -1); if (err < 0) return err; - - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, subs, SNDRV_PCM_HW_PARAM_CHANNELS, SNDRV_PCM_HW_PARAM_FORMAT, @@ -1001,7 +1001,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_FORMAT, hw_rule_format, subs, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_RATE, @@ -1011,7 +1011,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre if (err < 0) return err; if (param_period_time_if_needed >= 0) { - err = snd_pcm_hw_rule_add(runtime, 0, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_TIME, hw_rule_period_time, subs, SNDRV_PCM_HW_PARAM_FORMAT, @@ -1023,22 +1023,22 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre }
/* additional hw constraints for implicit fb */ - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_FORMAT, hw_rule_format_implicit_fb, subs, SNDRV_PCM_HW_PARAM_FORMAT, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_RATE, hw_rule_rate_implicit_fb, subs, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, hw_rule_period_size_implicit_fb, subs, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIODS, + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_PERIODS, hw_rule_periods_implicit_fb, subs, SNDRV_PCM_HW_PARAM_PERIODS, -1); if (err < 0) @@ -1065,7 +1065,7 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream) subs->dsd_dop.channel = 0; subs->dsd_dop.marker = 1;
- ret = setup_hw_info(runtime, subs); + ret = setup_hw_info(substream, subs); if (ret < 0) return ret; ret = snd_usb_autoresume(subs->stream->chip);
Add a new struct snd_pcm_hw_constraints under struct snd_soc_dpcm_runtime. The BE DAIs can use the new structure to add constraints that will not affect the FE of the PCM and will only apply to BE HW parameters.
Signed-off-by: Codrin Ciubotariu codrin.ciubotariu@microchip.com --- include/sound/pcm.h | 9 ++++++++ include/sound/soc-dpcm.h | 1 + sound/core/pcm_lib.c | 12 +++++++++-- sound/core/pcm_native.c | 20 ++++++++++-------- sound/soc/soc-pcm.c | 44 ++++++++++++++++++++++++++++++++++++++++ 5 files changed, 76 insertions(+), 10 deletions(-)
diff --git a/include/sound/pcm.h b/include/sound/pcm.h index af7fce2b574d..198d37d04d78 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -977,6 +977,15 @@ int snd_interval_ratnum(struct snd_interval *i, void _snd_pcm_hw_params_any(struct snd_pcm_hw_params *params); void _snd_pcm_hw_param_setempty(struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var);
+int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule); +int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule); +int snd_pcm_hw_rule_mul(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule); +int snd_pcm_hw_rule_div(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule); + int snd_pcm_hw_refine(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params);
int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index e296a3949b18..c5825876824a 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -95,6 +95,7 @@ struct snd_soc_dpcm_runtime { int users; struct snd_pcm_runtime *runtime; struct snd_pcm_hw_params hw_params; + struct snd_pcm_hw_constraints hw_constraints;
/* state and update */ enum snd_soc_dpcm_update runtime_update; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 582144f99045..125fafdf7517 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -16,6 +16,7 @@ #include <sound/info.h> #include <sound/pcm.h> #include <sound/pcm_params.h> +#include <sound/soc.h> #include <sound/timer.h>
#include "pcm_local.h" @@ -1121,12 +1122,19 @@ int snd_pcm_hw_rule_add(struct snd_pcm_substream *substream, unsigned int cond, snd_pcm_hw_rule_func_t func, void *private, int dep, ...) { - struct snd_pcm_hw_constraints *constrs = - &substream->runtime->hw_constraints; + struct snd_pcm_hw_constraints *constrs; struct snd_pcm_hw_rule *c; unsigned int k; va_list args;
+ if (substream->pcm->internal) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dpcm_runtime *dpcm = &rtd->dpcm[substream->stream]; + + constrs = &dpcm->hw_constraints; + } else { + constrs = &substream->runtime->hw_constraints; + } va_start(args, dep); if (constrs->rules_num >= constrs->rules_all) { struct snd_pcm_hw_rule *new; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 5feeef1b43f1..d6f14162bce5 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2278,23 +2278,25 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream) /* * hw configurator */ -static int snd_pcm_hw_rule_mul(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +int snd_pcm_hw_rule_mul(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) { struct snd_interval t; snd_interval_mul(hw_param_interval_c(params, rule->deps[0]), hw_param_interval_c(params, rule->deps[1]), &t); return snd_interval_refine(hw_param_interval(params, rule->var), &t); } +EXPORT_SYMBOL(snd_pcm_hw_rule_mul);
-static int snd_pcm_hw_rule_div(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +int snd_pcm_hw_rule_div(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) { struct snd_interval t; snd_interval_div(hw_param_interval_c(params, rule->deps[0]), hw_param_interval_c(params, rule->deps[1]), &t); return snd_interval_refine(hw_param_interval(params, rule->var), &t); } +EXPORT_SYMBOL(snd_pcm_hw_rule_div);
static int snd_pcm_hw_rule_muldivk(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) @@ -2316,8 +2318,8 @@ static int snd_pcm_hw_rule_mulkdiv(struct snd_pcm_hw_params *params, return snd_interval_refine(hw_param_interval(params, rule->var), &t); }
-static int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) { snd_pcm_format_t k; const struct snd_interval *i = @@ -2337,9 +2339,10 @@ static int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params, } return snd_mask_refine(mask, &m); } +EXPORT_SYMBOL(snd_pcm_hw_rule_format);
-static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) { struct snd_interval t; snd_pcm_format_t k; @@ -2363,6 +2366,7 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, t.integer = 1; return snd_interval_refine(hw_param_interval(params, rule->var), &t); } +EXPORT_SYMBOL(snd_pcm_hw_rule_sample_bits);
#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 #error "Change this table" diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 65f8ea73bae7..dae246918e0d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1471,6 +1471,43 @@ void dpcm_be_dai_stop(struct snd_soc_pcm_runtime *fe, int stream, } }
+static int dpcm_hw_constraints_init(struct snd_pcm_substream *substream) +{ + int err; + + if (!substream->pcm->internal) + return 0; + + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_FORMAT, + snd_pcm_hw_rule_format, NULL, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + snd_pcm_hw_rule_sample_bits, NULL, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + snd_pcm_hw_rule_div, NULL, + SNDRV_PCM_HW_PARAM_FRAME_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, + snd_pcm_hw_rule_mul, NULL, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_rule_div, NULL, + SNDRV_PCM_HW_PARAM_FRAME_BITS, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + -1); + if (err < 0) + return err; + return 0; +} + int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_pcm_runtime *be; @@ -1513,6 +1550,13 @@ int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) stream ? "capture" : "playback", be->dai_link->name);
be_substream->runtime = be->dpcm[stream].runtime; + + /* initialize the BE constraints */ + err = dpcm_hw_constraints_init(be_substream); + if (err < 0) { + dev_dbg(be->dev, "dpcm_hw_constraints_init failed\n"); + goto unwind; + } err = soc_pcm_open(be_substream); if (err < 0) { be->dpcm[stream].users--;
Apply the HW constraint rules added by the BE DAIs. The constraint rules are applied after the fixup is called for the HW parameters. This way, if the HW parameters do not correspond with the HW capabilities, the constraint rules will return an error. The mask and the interval constraints are the same as the ones used for FE. The DAI link dpcm_merged_* flags are used to check if the FE and BE must share the same HW parameters.
Signed-off-by: Codrin Ciubotariu codrin.ciubotariu@microchip.com --- include/sound/pcm.h | 2 ++ sound/core/pcm_native.c | 17 +++++++++++++---- sound/soc/soc-pcm.c | 30 ++++++++++++++++++++++++++++++ 3 files changed, 45 insertions(+), 4 deletions(-)
diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 198d37d04d78..b56a4435439a 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -987,6 +987,8 @@ int snd_pcm_hw_rule_div(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule);
int snd_pcm_hw_refine(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params); +int constrain_params_by_rules(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params);
int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, u_int64_t mask); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index d6f14162bce5..2ff6e182c7f5 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -22,6 +22,7 @@ #include <sound/pcm_params.h> #include <sound/timer.h> #include <sound/minors.h> +#include <sound/soc.h> #include <linux/uio.h> #include <linux/delay.h>
@@ -326,11 +327,10 @@ static int constrain_interval_params(struct snd_pcm_substream *substream, return 0; }
-static int constrain_params_by_rules(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +int constrain_params_by_rules(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) { - struct snd_pcm_hw_constraints *constrs = - &substream->runtime->hw_constraints; + struct snd_pcm_hw_constraints *constrs; unsigned int k; unsigned int *rstamps; unsigned int vstamps[SNDRV_PCM_HW_PARAM_LAST_INTERVAL + 1]; @@ -342,6 +342,14 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, bool again; int changed, err = 0;
+ if (substream->pcm->internal) { + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + + constrs = &rtd->dpcm[substream->stream].hw_constraints; + } else { + constrs = &substream->runtime->hw_constraints; + } + /* * Each application of rule has own sequence number. * @@ -446,6 +454,7 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, kfree(rstamps); return err; } +EXPORT_SYMBOL(constrain_params_by_rules);
static int fixup_unreferenced_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dae246918e0d..5bd71d48c0de 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1934,11 +1934,25 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) memcpy(&dpcm->hw_params, &fe->dpcm[stream].hw_params, sizeof(struct snd_pcm_hw_params));
+ /* copy FE mask and interval constraints */ + memcpy(&be->dpcm[stream].hw_constraints.masks, + &be_substream->runtime->hw_constraints.masks, + sizeof(be_substream->runtime->hw_constraints.masks)); + memcpy(&be->dpcm[stream].hw_constraints.intervals, + &be_substream->runtime->hw_constraints.intervals, + sizeof(be_substream->runtime->hw_constraints.intervals)); + /* perform any hw_params fixups */ ret = snd_soc_link_be_hw_params_fixup(be, &dpcm->hw_params); if (ret < 0) goto unwind;
+ /* apply constrain rules */ + dpcm->hw_params.rmask = ~0U; + ret = constrain_params_by_rules(be_substream, &dpcm->hw_params); + if (ret < 0) + goto unwind; + /* copy the fixed-up hw params for BE dai */ memcpy(&be->dpcm[stream].hw_params, &dpcm->hw_params, sizeof(struct snd_pcm_hw_params)); @@ -2002,6 +2016,22 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream,
memcpy(&fe->dpcm[stream].hw_params, params, sizeof(struct snd_pcm_hw_params)); + if (!fe->dai_link->dpcm_merged_format) { + snd_mask_any(hw_param_mask(&fe->dpcm[stream].hw_params, + SNDRV_PCM_HW_PARAM_FORMAT)); + snd_interval_any(hw_param_interval(&fe->dpcm[stream].hw_params, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS)); + snd_interval_any(hw_param_interval(&fe->dpcm[stream].hw_params, + SNDRV_PCM_HW_PARAM_FRAME_BITS)); + } + if (!fe->dai_link->dpcm_merged_chan) { + snd_interval_any(hw_param_interval(&fe->dpcm[stream].hw_params, + SNDRV_PCM_HW_PARAM_CHANNELS)); + } + if (!fe->dai_link->dpcm_merged_rate) { + snd_interval_any(hw_param_interval(&fe->dpcm[stream].hw_params, + SNDRV_PCM_HW_PARAM_RATE)); + } ret = dpcm_be_dai_hw_params(fe, stream); if (ret < 0) goto out;
Dne 23. 03. 21 v 12:43 Codrin Ciubotariu napsal(a):
To achieve this, the first thing needed is to detect whether a HW constraint rule is enforced by a FE or a BE DAI. This means that snd_pcm_hw_rule_add() needs to be able to differentiate between the two type of DAIs. For this, the runtime pointer to struct snd_pcm_runtime is replaced with a pointer to struct snd_pcm_substream, to be able to reach substream->pcm->internal to differentiate between FE and BE DAIs.
Think about other not-so-invasive solution. What about to use 'runtime->private_data' (struct snd_soc_pcm_runtime *) to determine FE / BE?
Jaroslav
On 23.03.2021 14:15, Jaroslav Kysela wrote:
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Dne 23. 03. 21 v 12:43 Codrin Ciubotariu napsal(a):
To achieve this, the first thing needed is to detect whether a HW constraint rule is enforced by a FE or a BE DAI. This means that snd_pcm_hw_rule_add() needs to be able to differentiate between the two type of DAIs. For this, the runtime pointer to struct snd_pcm_runtime is replaced with a pointer to struct snd_pcm_substream, to be able to reach substream->pcm->internal to differentiate between FE and BE DAIs.
Think about other not-so-invasive solution. What about to use 'runtime->private_data' (struct snd_soc_pcm_runtime *) to determine FE / BE?
I think struct snd_soc_pcm_runtime * is placed in substream->private_data, while runtime->private_data is used more by the platform drivers. runtime->trigger_master could be an idea, but it looks like it's initialized just before the trigger callback is called, way after the constraint rules are added and I am not sure it can be initialized earlier...
Best regards, Codrin
On 23.03.2021 16:18, Codrin.Ciubotariu@microchip.com wrote:
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On 23.03.2021 14:15, Jaroslav Kysela wrote:
EXTERNAL EMAIL: Do not click links or open attachments unless you know the content is safe
Dne 23. 03. 21 v 12:43 Codrin Ciubotariu napsal(a):
To achieve this, the first thing needed is to detect whether a HW constraint rule is enforced by a FE or a BE DAI. This means that snd_pcm_hw_rule_add() needs to be able to differentiate between the two type of DAIs. For this, the runtime pointer to struct snd_pcm_runtime is replaced with a pointer to struct snd_pcm_substream, to be able to reach substream->pcm->internal to differentiate between FE and BE DAIs.
Think about other not-so-invasive solution. What about to use 'runtime->private_data' (struct snd_soc_pcm_runtime *) to determine FE / BE?
I think struct snd_soc_pcm_runtime * is placed in substream->private_data, while runtime->private_data is used more by the platform drivers. runtime->trigger_master could be an idea, but it looks like it's initialized just before the trigger callback is called, way after the constraint rules are added and I am not sure it can be initialized earlier...
Best regards, Codrin
How about using a different API for ASoC only, since that's the place of DPCM. Only drivers that do not involve DSPs would have to to be changed to call the new snd_pcm_hw_rule_add() variant. Another solution would be to have a different snd_soc_pcm_runtime for BE interfaces (with a new hw_constraints member of course). What do you think?
Thanks! Codrin
On Wed, Apr 14, 2021 at 02:58:10PM +0000, Codrin.Ciubotariu@microchip.com wrote:
How about using a different API for ASoC only, since that's the place of DPCM. Only drivers that do not involve DSPs would have to to be changed to call the new snd_pcm_hw_rule_add() variant. Another solution would be to have a different snd_soc_pcm_runtime for BE interfaces (with a new hw_constraints member of course). What do you think?
I'm really not convinced we want to continue to pile stuff on top of DPCM, it is just fundamentally not up to modelling what modern systems are able to do - it's already making things more fragile than they should be and more special cases seems like it's going to end up making that worse. That said I've not seen the code but...
On 15.04.2021 19:17, Mark Brown wrote:
On Wed, Apr 14, 2021 at 02:58:10PM +0000, Codrin.Ciubotariu@microchip.com wrote:
How about using a different API for ASoC only, since that's the place of DPCM. Only drivers that do not involve DSPs would have to to be changed to call the new snd_pcm_hw_rule_add() variant. Another solution would be to have a different snd_soc_pcm_runtime for BE interfaces (with a new hw_constraints member of course). What do you think?
I'm really not convinced we want to continue to pile stuff on top of DPCM, it is just fundamentally not up to modelling what modern systems are able to do - it's already making things more fragile than they should be and more special cases seems like it's going to end up making that worse. That said I've not seen the code but...
Are there any plans for refactoring DPCM? any ideas ongoing? I also have some changes for PCM dmaengine, in the same 'style', similar to what I sent some time ago... I can adjust to different ideas, if there are any, but, for a start, can anyone confirm that the problem I am trying to fix is real?
Best regards, Codrin
On Thu, Apr 15, 2021 at 04:56:00PM +0000, Codrin.Ciubotariu@microchip.com wrote:
Are there any plans for refactoring DPCM? any ideas ongoing? I also have some changes for PCM dmaengine, in the same 'style', similar to what I sent some time ago... I can adjust to different ideas, if there are any, but, for a start, can anyone confirm that the problem I am trying to fix is real?
Lars-Peter's presentation from ELC in 2016 (!) is probably the clearest summary of the ideas:
https://elinux.org/images/e/e7/Audio_on_Linux.pdf https://youtu.be/6oQF2TzCYtQ
Essentially the idea is to represent everything, including the DSPs, as ASoC components and be able to represent the digital links between components in the DAPM graph in the same way we currently represent analogue links. This means we need a way to propagate digital configuration along the DAPM graph (or a parallel, cross linked digital one). Sadly I'm not really aware of anyone actively working on the actual conversion at the minute, Morimoto-san has done a lot of great preparatory work to make everything a component which makes the whole thing more tractable.
On 15.04.2021 20:25, Mark Brown wrote:
On Thu, Apr 15, 2021 at 04:56:00PM +0000, Codrin.Ciubotariu@microchip.com wrote:
Are there any plans for refactoring DPCM? any ideas ongoing? I also have some changes for PCM dmaengine, in the same 'style', similar to what I sent some time ago... I can adjust to different ideas, if there are any, but, for a start, can anyone confirm that the problem I am trying to fix is real?
Lars-Peter's presentation from ELC in 2016 (!) is probably the clearest summary of the ideas:
https://elinux.org/images/e/e7/Audio_on_Linux.pdf https://youtu.be/6oQF2TzCYtQEssentially the idea is to represent everything, including the DSPs, as ASoC components and be able to represent the digital links between components in the DAPM graph in the same way we currently represent analogue links. This means we need a way to propagate digital configuration along the DAPM graph (or a parallel, cross linked digital one). Sadly I'm not really aware of anyone actively working on the actual conversion at the minute, Morimoto-san has done a lot of great preparatory work to make everything a component which makes the whole thing more tractable.
Thank you for the links! So basically the machine driver disappears and all the components will be visible in user-space. If there is a list with the 'steps' or tasks to achieve this? I can try to pitch in.
Best regards, Codrin
On Fri, Apr 16, 2021 at 04:03:05PM +0000, Codrin.Ciubotariu@microchip.com wrote:
Thank you for the links! So basically the machine driver disappears and all the components will be visible in user-space.
Not entirely - you still need something to say how they're wired together but it'll be a *lot* simpler for anything that currently used DPCM.
If there is a list with the 'steps' or tasks to achieve this? I can try to pitch in.
Not really written down that I can think of. I think the next steps that I can think of right now are unfortunately bigger and harder ones, mainly working out a way to represent digital configuration as a graph that can be attached to/run in parallel with DAPM other people might have some better ideas though. Sorry, I appreciate that this isn't super helpful :/
On 4/16/21 11:31 AM, Mark Brown wrote:
On Fri, Apr 16, 2021 at 04:03:05PM +0000, Codrin.Ciubotariu@microchip.com wrote:
Thank you for the links! So basically the machine driver disappears and all the components will be visible in user-space.
Not entirely - you still need something to say how they're wired together but it'll be a *lot* simpler for anything that currently used DPCM.
If there is a list with the 'steps' or tasks to achieve this? I can try to pitch in.
Not really written down that I can think of. I think the next steps that I can think of right now are unfortunately bigger and harder ones, mainly working out a way to represent digital configuration as a graph that can be attached to/run in parallel with DAPM other people might have some better ideas though. Sorry, I appreciate that this isn't super helpful :/
I see a need for this in our future SoundWire/SDCA work. So far I was planning to model the entities as 'widgets' and lets DAPM propagate activation information for power management, however there are also bits of information in 'Clusters' (number of channels and spatial relationships) that could change dynamically and would be interesting to propagate across entities, so that when we get a stream of data on the bus we know what it is.
when we discussed the multi-configuration support for BT offload, it also became apparent that we don't fully control the sample rate changes between FE and BE, we only control the start and ends. I fully agree that the division between front- and back-ends is becoming limiting and DPCM is not only complicated but difficult to stretch further.
On Fri, Apr 16, 2021 at 11:47:01AM -0500, Pierre-Louis Bossart wrote:
On 4/16/21 11:31 AM, Mark Brown wrote:
Not really written down that I can think of. I think the next steps that I can think of right now are unfortunately bigger and harder ones, mainly working out a way to represent digital configuration as a graph that can be attached to/run in parallel with DAPM other people might have some better ideas though. Sorry, I appreciate that this isn't super helpful :/
I see a need for this in our future SoundWire/SDCA work. So far I was planning to model the entities as 'widgets' and lets DAPM propagate activation information for power management, however there are also bits of information in 'Clusters' (number of channels and spatial relationships) that could change dynamically and would be interesting to propagate across entities, so that when we get a stream of data on the bus we know what it is.
Yes, I was thinking along similar lines last time I looked at it - I was using the term digital domains. You'd need some impedence matching objects for things like SRCs, and the ability to flag which configuration matters within a domain (eg, a lot of things will covert to the maximum supported bit width for internal operation so bit width only matters on external interfaces) but I think for a first pass we can get away with forcing everything other than what DPCM has as front ends into static configurations.
On 4/16/21 1:55 PM, Mark Brown wrote:
On Fri, Apr 16, 2021 at 11:47:01AM -0500, Pierre-Louis Bossart wrote:
On 4/16/21 11:31 AM, Mark Brown wrote:
Not really written down that I can think of. I think the next steps that I can think of right now are unfortunately bigger and harder ones, mainly working out a way to represent digital configuration as a graph that can be attached to/run in parallel with DAPM other people might have some better ideas though. Sorry, I appreciate that this isn't super helpful :/
I see a need for this in our future SoundWire/SDCA work. So far I was planning to model the entities as 'widgets' and lets DAPM propagate activation information for power management, however there are also bits of information in 'Clusters' (number of channels and spatial relationships) that could change dynamically and would be interesting to propagate across entities, so that when we get a stream of data on the bus we know what it is.
Yes, I was thinking along similar lines last time I looked at it - I was using the term digital domains. You'd need some impedence matching objects for things like SRCs, and the ability to flag which configuration matters within a domain (eg, a lot of things will covert to the maximum supported bit width for internal operation so bit width only matters on external interfaces) but I think for a first pass we can get away with forcing everything other than what DPCM has as front ends into static configurations.
You lost me on the last sentence. did you mean "forcing everything into static configurations except for what DPCM has as front-ends"?
It may already be too late for static configurations, Intel, NXP and others have started to enable cases where the dailink configuration varies.
FWIW both the USB and SDCA class document are very careful with the notion of constraints and whether an entity is implemented in the analog or digital domains. There are 'clock sources' that may be used in specific terminals but no notion of explicit SRC in the graph to leave implementers a lot of freedom. There is a notion of 'Usage' that describes e.g. FullBand or WideBand without defining what the representation is. The bit width is also not described except where necessary (history buffers and external bus-facing interfaces). Like you said it's mostly the boundaries of the domains that matter.
On Fri, Apr 16, 2021 at 02:39:25PM -0500, Pierre-Louis Bossart wrote:
On 4/16/21 1:55 PM, Mark Brown wrote:
to the maximum supported bit width for internal operation so bit width only matters on external interfaces) but I think for a first pass we can get away with forcing everything other than what DPCM has as front ends into static configurations.
You lost me on the last sentence. did you mean "forcing everything into static configurations except for what DPCM has as front-ends"?
Yes...
It may already be too late for static configurations, Intel, NXP and others have started to enable cases where the dailink configuration varies.
Well, they won't be able to use any new stuff until someone implements support for dynamic configurations in the new stuff.
On 16.04.2021 19:31, Mark Brown wrote:
On Fri, Apr 16, 2021 at 04:03:05PM +0000, Codrin.Ciubotariu@microchip.com wrote:
Thank you for the links! So basically the machine driver disappears and all the components will be visible in user-space.
Not entirely - you still need something to say how they're wired together but it'll be a *lot* simpler for anything that currently used DPCM.
Right, device-tree/acpi wiring is not enough...
If there is a list with the 'steps' or tasks to achieve this? I can try to pitch in.
Not really written down that I can think of. I think the next steps that I can think of right now are unfortunately bigger and harder ones, mainly working out a way to represent digital configuration as a graph that can be attached to/run in parallel with DAPM other people might have some better ideas though. Sorry, I appreciate that this isn't super helpful :/
I think I have good picture, a more robust card->dai_link, not with just FEs and BEs ... I will try to monitor the alsa list more, see what other people are working on. Thank you for your time!
Best regards, Codrin
On 3/23/21 6:43 AM, Codrin Ciubotariu wrote:
HW constraints are needed to set limitations for HW parameters used to configure the DAIs. All DAIs on the same link must agree upon the HW parameters, so the parameters are affected by the DAIs' features and their limitations. In case of DPCM, the FE DAIs can be used to perform different kind of conversions, such as format or rate changing, bringing the audio stream to a configuration supported by all the DAIs of the BE's link. For this reason, the limitations of the BE DAIs are not always important for the HW parameters between user-space and FE, only for the paratemers between FE and BE DAI links. This brings us to this patch-set, which aims to separate the FE HW constraints from the BE HW constraints. This way, the FE can be used to perform more efficient HW conversions, on the initial audio stream parameters, to parameters supported by the BE DAIs. To achieve this, the first thing needed is to detect whether a HW constraint rule is enforced by a FE or a BE DAI. This means that snd_pcm_hw_rule_add() needs to be able to differentiate between the two type of DAIs. For this, the runtime pointer to struct snd_pcm_runtime is replaced with a pointer to struct snd_pcm_substream, to be able to reach substream->pcm->internal to differentiate between FE and BE DAIs. This change affects many sound drivers (and one gpu drm driver). All these changes are included in the first patch, to have a better overview of the implications created by this change. The second patch adds a new struct snd_pcm_hw_constraints under struct snd_soc_dpcm_runtime, which is used to store the HW constraint rules added by the BE DAIs. This structure is initialized with a subset of the runtime constraint rules which does not include the rules that affect the buffer or period size. snd_pcm_hw_rule_add() will add the BE rules to the new struct snd_pcm_hw_constraints. The third and last patch will apply the BE rule constraints, after the fixup callback. If the fixup HW parameters do not respect the BE constraint rules, the rules will exit with an error. The FE mask and interval constraints are copied to the BE ones, to satisfy the dai_link->dpcm_merged_* flags. The dai_link->dpcm_merged_* flags are used to know if the FE does format or sampling rate conversion.
I tested with ad1934 and wm8731 codecs as BEs, with a not-yet-mainlined ASRC as FE, that can also do format conversion. I realize that the change to snd_pcm_hw_rule_add() has a big impact, even though all the drivers use snd_pcm_hw_rule_add() with substream->runtime, so passing substream instead of runtime is not that risky.
can you use the BE hw_params_fixup instead?
That's what we use for SOF.
The FE hw_params are propagated to the BE, and then the BE can update the hw_params based on its own limitations and pass the result downstream, e.g. to a codec.
I'll copy below my understanding of the flow, which we discussed recently in the SOF team:
my understanding is that we start with the front-end hw_params as the basis for the back-end hw_params.
static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *fe = asoc_substream_to_rtd(substream); int ret, stream = substream->stream;
mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE);
memcpy(&fe->dpcm[stream].hw_params, params, sizeof(struct snd_pcm_hw_params)); ret = dpcm_be_dai_hw_params(fe, stream); <<< the BE is handled first. if (ret < 0) { dev_err(fe->dev,"ASoC: hw_params BE failed %d\n", ret); goto out; }
dev_dbg(fe->dev, "ASoC: hw_params FE %s rate %d chan %x fmt %d\n", fe->dai_link->name, params_rate(params), params_channels(params), params_format(params));
/* call hw_params on the frontend */ ret = soc_pcm_hw_params(substream, params);
then each BE will be configured
int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret;
for_each_dpcm_be(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream);
/* is this op for this BE ? */ if (!snd_soc_dpcm_be_can_update(fe, be, stream)) continue;
/* copy params for each dpcm */ memcpy(&dpcm->hw_params, &fe->dpcm[stream].hw_params, sizeof(struct snd_pcm_hw_params));
/* perform any hw_params fixups */ ret = snd_soc_link_be_hw_params_fixup(be, &dpcm->hw_params);
The fixup is the key, in SOF this is where we are going to look for information from the topology.
/* fixup the BE DAI link to match any values from topology */ int sof_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, SOF_AUDIO_PCM_DRV_NAME); struct snd_sof_dai *dai = snd_sof_find_dai(component, (char *)rtd->dai_link->name); struct snd_soc_dpcm *dpcm;
/* no topology exists for this BE, try a common configuration */ if (!dai) { dev_warn(component->dev, "warning: no topology found for BE DAI %s config\n", rtd->dai_link->name);
/* set 48k, stereo, 16bits by default */ rate->min = 48000; rate->max = 48000;
channels->min = 2; channels->max = 2;
snd_mask_none(fmt); snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
return 0; }
/* read format from topology */ snd_mask_none(fmt);
switch (dai->comp_dai.config.frame_fmt) { case SOF_IPC_FRAME_S16_LE: snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); break; case SOF_IPC_FRAME_S24_4LE: snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); break; case SOF_IPC_FRAME_S32_LE: snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S32_LE); break; default: dev_err(component->dev, "error: No available DAI format!\n"); return -EINVAL; }
/* read rate and channels from topology */ switch (dai->dai_config->type) { case SOF_DAI_INTEL_SSP: rate->min = dai->dai_config->ssp.fsync_rate; rate->max = dai->dai_config->ssp.fsync_rate; channels->min = dai->dai_config->ssp.tdm_slots; channels->max = dai->dai_config->ssp.tdm_slots;
On 23.03.2021 21:25, Pierre-Louis Bossart wrote:
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On 3/23/21 6:43 AM, Codrin Ciubotariu wrote:
HW constraints are needed to set limitations for HW parameters used to configure the DAIs. All DAIs on the same link must agree upon the HW parameters, so the parameters are affected by the DAIs' features and their limitations. In case of DPCM, the FE DAIs can be used to perform different kind of conversions, such as format or rate changing, bringing the audio stream to a configuration supported by all the DAIs of the BE's link. For this reason, the limitations of the BE DAIs are not always important for the HW parameters between user-space and FE, only for the paratemers between FE and BE DAI links. This brings us to this patch-set, which aims to separate the FE HW constraints from the BE HW constraints. This way, the FE can be used to perform more efficient HW conversions, on the initial audio stream parameters, to parameters supported by the BE DAIs. To achieve this, the first thing needed is to detect whether a HW constraint rule is enforced by a FE or a BE DAI. This means that snd_pcm_hw_rule_add() needs to be able to differentiate between the two type of DAIs. For this, the runtime pointer to struct snd_pcm_runtime is replaced with a pointer to struct snd_pcm_substream, to be able to reach substream->pcm->internal to differentiate between FE and BE DAIs. This change affects many sound drivers (and one gpu drm driver). All these changes are included in the first patch, to have a better overview of the implications created by this change. The second patch adds a new struct snd_pcm_hw_constraints under struct snd_soc_dpcm_runtime, which is used to store the HW constraint rules added by the BE DAIs. This structure is initialized with a subset of the runtime constraint rules which does not include the rules that affect the buffer or period size. snd_pcm_hw_rule_add() will add the BE rules to the new struct snd_pcm_hw_constraints. The third and last patch will apply the BE rule constraints, after the fixup callback. If the fixup HW parameters do not respect the BE constraint rules, the rules will exit with an error. The FE mask and interval constraints are copied to the BE ones, to satisfy the dai_link->dpcm_merged_* flags. The dai_link->dpcm_merged_* flags are used to know if the FE does format or sampling rate conversion.
I tested with ad1934 and wm8731 codecs as BEs, with a not-yet-mainlined ASRC as FE, that can also do format conversion. I realize that the change to snd_pcm_hw_rule_add() has a big impact, even though all the drivers use snd_pcm_hw_rule_add() with substream->runtime, so passing substream instead of runtime is not that risky.
can you use the BE hw_params_fixup instead?
That's what we use for SOF.
The FE hw_params are propagated to the BE, and then the BE can update the hw_params based on its own limitations and pass the result downstream, e.g. to a codec.
I'll copy below my understanding of the flow, which we discussed recently in the SOF team:
my understanding is that we start with the front-end hw_params as the basis for the back-end hw_params.
static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *fe = asoc_substream_to_rtd(substream); int ret, stream = substream->stream;
mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE);
memcpy(&fe->dpcm[stream].hw_params, params, sizeof(struct snd_pcm_hw_params)); ret = dpcm_be_dai_hw_params(fe, stream); <<< the BE is handled first. if (ret < 0) { dev_err(fe->dev,"ASoC: hw_params BE failed %d\n", ret); goto out; }
dev_dbg(fe->dev, "ASoC: hw_params FE %s rate %d chan %x fmt %d\n", fe->dai_link->name, params_rate(params), params_channels(params), params_format(params));
/* call hw_params on the frontend */ ret = soc_pcm_hw_params(substream, params);
then each BE will be configured
int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret;
for_each_dpcm_be(fe, stream, dpcm) {
struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream);
/* is this op for this BE ? */ if (!snd_soc_dpcm_be_can_update(fe, be, stream)) continue;
/* copy params for each dpcm */ memcpy(&dpcm->hw_params, &fe->dpcm[stream].hw_params, sizeof(struct snd_pcm_hw_params));
/* perform any hw_params fixups */ ret = snd_soc_link_be_hw_params_fixup(be, &dpcm->hw_params);
The fixup is the key, in SOF this is where we are going to look for information from the topology.
/* fixup the BE DAI link to match any values from topology */ int sof_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, SOF_AUDIO_PCM_DRV_NAME); struct snd_sof_dai *dai = snd_sof_find_dai(component, (char *)rtd->dai_link->name); struct snd_soc_dpcm *dpcm;
/* no topology exists for this BE, try a common configuration */ if (!dai) { dev_warn(component->dev, "warning: no topology found for BE DAI %s config\n", rtd->dai_link->name);
/* set 48k, stereo, 16bits by default */ rate->min = 48000; rate->max = 48000;
channels->min = 2; channels->max = 2;
snd_mask_none(fmt); snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
return 0; }
/* read format from topology */ snd_mask_none(fmt);
switch (dai->comp_dai.config.frame_fmt) { case SOF_IPC_FRAME_S16_LE: snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); break; case SOF_IPC_FRAME_S24_4LE: snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); break; case SOF_IPC_FRAME_S32_LE: snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S32_LE); break; default: dev_err(component->dev, "error: No available DAI format!\n"); return -EINVAL; }
/* read rate and channels from topology */ switch (dai->dai_config->type) { case SOF_DAI_INTEL_SSP: rate->min = dai->dai_config->ssp.fsync_rate; rate->max = dai->dai_config->ssp.fsync_rate; channels->min = dai->dai_config->ssp.tdm_slots; channels->max = dai->dai_config->ssp.tdm_slots;
I am using hw_params_fixup, but it's not enough. The first thing I do is to not add the BE HW constraint rules in runtime->hw_constraints, because this will potentially affect the FE HW params. If the FE does sampling rate conversion, for example, applying the sampling rate constrain rules from a BE codec on FE is useless. The second thing I do is to apply these BE HW constraint rules to the BE HW params. It's true that the BE HW params can be fine tuned via hw_params_fixup (topology, device-tree or even static parameters) and set in such a way that avoid the BE HW constraints, so we could ignore the constraint rules added by their drivers. It's not every elegant and running the BE HW constraint rules for the fixup BE HW params can be a sanity check. Also, I am thinking that if the FE does no conversion (be_hw_params_fixup missing) and more than one BEs are available, applying the HW constraint rules on the same set of BE HW params could rule out the incompatible BE DAI links and start the compatible ones only. I am not sure this is a real usecase.
Thank you for your insights!
Best regards, Codrin
I am using hw_params_fixup, but it's not enough. The first thing I do is to not add the BE HW constraint rules in runtime->hw_constraints, because this will potentially affect the FE HW params. If the FE does sampling rate conversion, for example, applying the sampling rate constrain rules from a BE codec on FE is useless. The second thing I do is to apply these BE HW constraint rules to the BE HW params. It's true that the BE HW params can be fine tuned via hw_params_fixup (topology, device-tree or even static parameters) and set in such a way that avoid the BE HW constraints, so we could ignore the constraint rules added by their drivers. It's not every elegant and running the BE HW constraint rules for the fixup BE HW params can be a sanity check. Also, I am thinking that if the FE does no conversion (be_hw_params_fixup missing) and more than one BEs are available, applying the HW constraint rules on the same set of BE HW params could rule out the incompatible BE DAI links and start the compatible ones only. I am not sure this is a real usecase.
Even after a second cup of coffee I was not able to follow why the hw_params_fixup was not enough? That paragraph is rather dense.
And to be frank I don't fully understand the problem statement above: "separate the FE HW constraints from the BE HW constraints". We have existing solutions with a DSP-based SRC adjusting FE rates to what is required by the BE dailink.
Maybe it would help to show examples of what you can do today and what you cannot, so that we are on the same wavelength on what the limitations are and how to remove them?
On 24.03.2021 17:28, Pierre-Louis Bossart wrote:
I am using hw_params_fixup, but it's not enough. The first thing I do is to not add the BE HW constraint rules in runtime->hw_constraints, because this will potentially affect the FE HW params. If the FE does sampling rate conversion, for example, applying the sampling rate constrain rules from a BE codec on FE is useless. The second thing I do is to apply these BE HW constraint rules to the BE HW params. It's true that the BE HW params can be fine tuned via hw_params_fixup (topology, device-tree or even static parameters) and set in such a way that avoid the BE HW constraints, so we could ignore the constraint rules added by their drivers. It's not every elegant and running the BE HW constraint rules for the fixup BE HW params can be a sanity check. Also, I am thinking that if the FE does no conversion (be_hw_params_fixup missing) and more than one BEs are available, applying the HW constraint rules on the same set of BE HW params could rule out the incompatible BE DAI links and start the compatible ones only. I am not sure this is a real usecase.
Even after a second cup of coffee I was not able to follow why the hw_params_fixup was not enough? That paragraph is rather dense.
Right, sorry about that. :)
And to be frank I don't fully understand the problem statement above: "separate the FE HW constraints from the BE HW constraints". We have existing solutions with a DSP-based SRC adjusting FE rates to what is required by the BE dailink.
Maybe it would help to show examples of what you can do today and what you cannot, so that we are on the same wavelength on what the limitations are and how to remove them?
For example, ad1934 driver sets a constraint to always have 32 sample bits [1] and this rule will be added in struct snd_pcm_runtime -> hw_constraints [2]. As you can see, this is added early and always. So this rule will affect the HW parameters used from the user-space [3] and, in our example, only audio streams that have formats of 4B containers will be used (S24_LE, S32_LE, etc). For playback, if the audio stream won't have this format, the stream will need to be converted in software, using plug ALSA plugin for example. This is OK for normal PCM:
HW params user <----------> CPU <---> AD1934 space ^ DAI | | | -------------------------| sample bits constraint rule (32b)
For DPCM, we have the same behavior (unfortunately). ad1934, as a BE codec, will add it's rule in the same place, which means that it will again affect the HW parameters set by user-space. So user-space will have to convert all audio streams to have a 32b sample size. If the FE is capable of format conversing, this software conversion is useless.
FE HW params BE HW params user <----------> FE <--------------> BE CPU <----> BE AD1934 space ^ DAI DAI | | | --------------------------------------------------| sample bits constraint rule (32b)
The format conversion will be done in software instead of hardware (FE).
I hope I was able to be more clear now. Thanks for taking time to understand this. I owe you a coffee. :)
Best regards, Codrin
[1] https://elixir.bootlin.com/linux/v5.12-rc4/source/sound/soc/codecs/ad193x.c#... [2] https://elixir.bootlin.com/linux/v5.12-rc4/source/sound/core/pcm_lib.c#L1141 [3] https://elixir.bootlin.com/linux/v5.12-rc4/source/sound/core/pcm_native.c#L6...
participants (5)
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Codrin Ciubotariu -
Codrin.Ciubotariu＠microchip.com
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Jaroslav Kysela -
Mark Brown -
Pierre-Louis Bossart