October 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [asoc:for-5.5 122/122] sound/soc/codecs/tas2562.c:195:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer
by kbuild test robot 22 Oct '19

22 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 head: c173dba44c2d2ec2adaa4e607e9729b74236bd57 commit: c173dba44c2d2ec2adaa4e607e9729b74236bd57 [122/122] ASoC: tas2562: Introduce the TAS2562 amplifier reproduce: # apt-get install sparse # sparse version: v0.6.1-rc1-42-g38eda53-dirty git checkout c173dba44c2d2ec2adaa4e607e9729b74236bd57 make ARCH=x86_64 allmodconfig make C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/tas2562.c:195:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/soc/codecs/tas2562.c:202:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/soc/codecs/tas2562.c:209:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer >> sound/soc/codecs/tas2562.c:246:58: sparse: sparse: incorrect type in argument 2 (different base types) @@ expected int bitwidth @@ got restricted snd_pcm_formint bitwidth @@ >> sound/soc/codecs/tas2562.c:246:58: sparse: expected int bitwidth >> sound/soc/codecs/tas2562.c:246:58: sparse: got restricted snd_pcm_format_t sound/soc/codecs/tas2562.c:472:39: sparse: sparse: restricted snd_pcm_format_t degrades to integer vim +195 sound/soc/codecs/tas2562.c 189 190 static int tas2562_set_bitwidth(struct tas2562_data *tas2562, int bitwidth) 191 { 192 int ret; 193 194 switch (bitwidth) { > 195 case SNDRV_PCM_FORMAT_S16_LE: 196 snd_soc_component_update_bits(tas2562->component, 197 TAS2562_TDM_CFG2, 198 TAS2562_TDM_CFG2_RXWLEN_MASK, 199 TAS2562_TDM_CFG2_RXWLEN_16B); 200 tas2562->v_sense_slot = tas2562->i_sense_slot + 2; 201 break; 202 case SNDRV_PCM_FORMAT_S24_LE: 203 snd_soc_component_update_bits(tas2562->component, 204 TAS2562_TDM_CFG2, 205 TAS2562_TDM_CFG2_RXWLEN_MASK, 206 TAS2562_TDM_CFG2_RXWLEN_24B); 207 tas2562->v_sense_slot = tas2562->i_sense_slot + 4; 208 break; 209 case SNDRV_PCM_FORMAT_S32_LE: 210 snd_soc_component_update_bits(tas2562->component, 211 TAS2562_TDM_CFG2, 212 TAS2562_TDM_CFG2_RXWLEN_MASK, 213 TAS2562_TDM_CFG2_RXWLEN_32B); 214 tas2562->v_sense_slot = tas2562->i_sense_slot + 4; 215 break; 216 217 default: 218 dev_info(tas2562->dev, "Not supported params format\n"); 219 } 220 221 ret = snd_soc_component_update_bits(tas2562->component, 222 TAS2562_TDM_CFG5, 223 TAS2562_TDM_CFG5_VSNS_EN | TAS2562_TDM_CFG5_VSNS_SLOT_MASK, 224 TAS2562_TDM_CFG5_VSNS_EN | tas2562->v_sense_slot); 225 if (ret < 0) 226 return ret; 227 228 ret = snd_soc_component_update_bits(tas2562->component, 229 TAS2562_TDM_CFG6, 230 TAS2562_TDM_CFG6_ISNS_EN | TAS2562_TDM_CFG6_ISNS_SLOT_MASK, 231 TAS2562_TDM_CFG6_ISNS_EN | tas2562->i_sense_slot); 232 if (ret < 0) 233 return ret; 234 235 return 0; 236 } 237 238 static int tas2562_hw_params(struct snd_pcm_substream *substream, 239 struct snd_pcm_hw_params *params, 240 struct snd_soc_dai *dai) 241 { 242 struct snd_soc_component *component = dai->component; 243 struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); 244 int ret; 245 > 246 ret = tas2562_set_bitwidth(tas2562, params_format(params)); 247 if (ret) { 248 dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret); 249 return ret; 250 } 251 252 ret = tas2562_set_samplerate(tas2562, params_rate(params)); 253 if (ret) 254 dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret); 255 256 return ret; 257 } 258 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

5 4

[alsa-devel] [PATCH] ASoC: compress: fix unsigned integer overflow check
by Srinivas Kandagatla 21 Oct '19

21 Oct '19

From: Xiaojun Sang <xsang(a)codeaurora.org> Parameter fragments and fragment_size are type of u32. U32_MAX is the correct check. Signed-off-by: Xiaojun Sang <xsang(a)codeaurora.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/core/compress_offload.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 41905afada63..f34ce564d92c 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -528,7 +528,7 @@ static int snd_compress_check_input(struct snd_compr_params *params) { /* first let's check the buffer parameter's */ if (params->buffer.fragment_size == 0 || - params->buffer.fragments > INT_MAX / params->buffer.fragment_size || + params->buffer.fragments > U32_MAX / params->buffer.fragment_size || params->buffer.fragments == 0) return -EINVAL; -- 2.21.0

3 2

[alsa-devel] [PATCH 1/2] ASoC: msm8916-wcd-analog: Fix RX1 selection in RDAC2 MUX
by Stephan Gerhold 21 Oct '19

21 Oct '19

According to the PM8916 Hardware Register Description, CDC_D_CDC_CONN_HPHR_DAC_CTL has only a single bit (RX_SEL) to switch between RX1 (0) and RX2 (1). It is not possible to disable it entirely to achieve the "ZERO" state. However, at the moment the "RDAC2 MUX" mixer defines three possible values ("ZERO", "RX2" and "RX1"). Setting the mixer to "ZERO" actually configures it to RX1. Setting the mixer to "RX1" has (seemingly) no effect. Remove "ZERO" and replace it with "RX1" to fix this. Fixes: 585e881e5b9e ("ASoC: codecs: Add msm8916-wcd analog codec") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/codecs/msm8916-wcd-analog.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 667e9f73aba3..e3d311fb510e 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -306,7 +306,7 @@ struct pm8916_wcd_analog_priv { }; static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" }; -static const char *const rdac2_mux_text[] = { "ZERO", "RX2", "RX1" }; +static const char *const rdac2_mux_text[] = { "RX1", "RX2" }; static const char *const hph_text[] = { "ZERO", "Switch", }; static const struct soc_enum hph_enum = SOC_ENUM_SINGLE_VIRT( @@ -321,7 +321,7 @@ static const struct soc_enum adc2_enum = SOC_ENUM_SINGLE_VIRT( /* RDAC2 MUX */ static const struct soc_enum rdac2_mux_enum = SOC_ENUM_SINGLE( - CDC_D_CDC_CONN_HPHR_DAC_CTL, 0, 3, rdac2_mux_text); + CDC_D_CDC_CONN_HPHR_DAC_CTL, 0, 2, rdac2_mux_text); static const struct snd_kcontrol_new spkr_switch[] = { SOC_DAPM_SINGLE("Switch", CDC_A_SPKR_DAC_CTL, 7, 1, 0) -- 2.23.0

3 5

[alsa-devel] [PATCH v2] ASoC: samsung: i2s: Document clocks macros
by Marek Szyprowski 21 Oct '19

21 Oct '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Document clocks macros with their description from 'Documentation/devicetree/bindings/sound/samsung-i2s.txt' Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- v2: - Added proper commit message and description - Moved comments to the right side Best regards, Maciej Falkowski --- include/dt-bindings/sound/samsung-i2s.h | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/include/dt-bindings/sound/samsung-i2s.h b/include/dt-bindings/sound/samsung-i2s.h index 77545f14c379..250de0d6c734 100644 --- a/include/dt-bindings/sound/samsung-i2s.h +++ b/include/dt-bindings/sound/samsung-i2s.h @@ -2,8 +2,14 @@ #ifndef _DT_BINDINGS_SAMSUNG_I2S_H #define _DT_BINDINGS_SAMSUNG_I2S_H -#define CLK_I2S_CDCLK 0 -#define CLK_I2S_RCLK_SRC 1 -#define CLK_I2S_RCLK_PSR 2 +#define CLK_I2S_CDCLK 0 /* the CDCLK (CODECLKO) gate clock */ + +#define CLK_I2S_RCLK_SRC 1 /* the RCLKSRC mux clock (corresponding to + * RCLKSRC bit in IISMOD register) + */ + +#define CLK_I2S_RCLK_PSR 2 /* the RCLK prescaler divider clock + * (corresponding to the IISPSR register) + */ #endif /* _DT_BINDINGS_SAMSUNG_I2S_H */ -- 2.17.1

4 3

[alsa-devel] [PATCH v5 0/4] ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: support WoV
by Tzung-Bi Shih 21 Oct '19

21 Oct '19

This series makes mt6358, cros_ec_codec, and mt8183-mt6358-ts3a227-max98357 support WoV (wake on voice). ASoC: cros_ec_codec: support WoV - makes cros_ec_codec support WoV. ASoC: mediatek: mt6358: support WoV - sets necessary registers on mt6358 to support WoV. ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV - make machine driver mt8183-mt6358-ts3a227-max98357 support WoV if ec-codec is in DTS. Changes from v4: - remove the first 6 commits which have merged - fix a compile error by adding Kconfig dependency https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156992.h… Changes from v3: - fix a compile error by adding ifdef https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156750.h… - fix a coccinelle warning https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156754.h… Changes from v2: - rebase upon to "don't use snd_pcm_ops" series https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156170.h… - fix sparse errors https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156328.h… - use "reg" for SHM binding https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156657.h… Changes from v1: - fix a compile error and make kbuild bot happy https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156315.h… Tzung-Bi Shih (4): ASoC: cros_ec_codec: support WoV ASoC: mediatek: mt6358: support WoV ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV .../sound/mt8183-mt6358-ts3a227-max98357.txt | 3 + drivers/platform/chrome/cros_ec_trace.c | 1 + .../linux/platform_data/cros_ec_commands.h | 69 ++ sound/soc/codecs/Kconfig | 2 + sound/soc/codecs/cros_ec_codec.c | 700 +++++++++++++++++- sound/soc/codecs/mt6358.c | 105 +++ sound/soc/mediatek/Kconfig | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 70 +- 8 files changed, 946 insertions(+), 5 deletions(-) -- 2.23.0.866.gb869b98d4c-goog

3 12

[alsa-devel] [PATCH 0/6] ALSA: firewire: handle several IR/IT context in callback of the IT context
by Takashi Sakamoto 21 Oct '19

21 Oct '19

Hi, This patchset is a part of my continuous work to improve ALSA IEC 61883-1/6 packet streaming engine for clock-recovery, addressed in my previous message: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-August/153388.ht… For the clock-recovery, information in the sequence of tx packet from device should be used to generate the sequence of rx packet to the device. In current implementation of the engine, the packets are processed in different tasklet contexts for each IR/IT context. This is inconvenient to bypass information between these IR/IT contexts. In this patchset, the engine is improved to process all of IR/IT contexts in the same domain by a tasklet context for one of IT context. For convenience, the IT context is called as 'IRQ target'. As a result, 1394 OHCI controller is managed to generate hardware IRQ just for the IRQ target. All of rx/tx packets are processed in tasklet for the hardware IRQ. Takashi Sakamoto (6): ALSA: firewire-lib: add irq_target member into amdtp_domain struct ALSA: firewire-lib: replace pointer callback to flush isoc contexts in AMDTP domain ALSA: firewire-lib: replace ack callback to flush isoc contexts in AMDTP domain ALSA: firewire-lib: cancel flushing isoc context in the laste step to process context callback ALSA: firewire-lib: handle several AMDTP streams in callback handler of IRQ target ALSA: firewire-lib: postpone to start IR context sound/firewire/amdtp-stream.c | 329 +++++++++++++++----- sound/firewire/amdtp-stream.h | 17 +- sound/firewire/bebob/bebob_pcm.c | 18 +- sound/firewire/bebob/bebob_stream.c | 10 +- sound/firewire/dice/dice-pcm.c | 8 +- sound/firewire/dice/dice-stream.c | 2 +- sound/firewire/digi00x/digi00x-pcm.c | 8 +- sound/firewire/digi00x/digi00x-stream.c | 2 +- sound/firewire/fireface/ff-pcm.c | 8 +- sound/firewire/fireface/ff-stream.c | 10 +- sound/firewire/fireworks/fireworks_pcm.c | 10 +- sound/firewire/fireworks/fireworks_stream.c | 2 +- sound/firewire/motu/motu-pcm.c | 8 +- sound/firewire/motu/motu-stream.c | 2 +- sound/firewire/oxfw/oxfw-pcm.c | 8 +- sound/firewire/oxfw/oxfw-stream.c | 2 +- sound/firewire/tascam/tascam-pcm.c | 8 +- sound/firewire/tascam/tascam-stream.c | 2 +- 18 files changed, 325 insertions(+), 129 deletions(-) -- 2.20.1

2 9

[alsa-devel] [sound:for-next 37/38] include/linux/pci.h:2317:13: error: 'pci_pr3_present' defined but not used
by kbuild test robot 21 Oct '19

21 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: bacd861452d2be86a4df341b12e32db7dac8021e commit: 52525b7a3cf82adec5c6cf0ecbd23ff228badc94 [37/38] PCI: Add a helper to check Power Resource Requirements _PR3 existence config: sparc64-allmodconfig (attached as .config) compiler: sparc64-linux-gcc (GCC) 7.4.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 52525b7a3cf82adec5c6cf0ecbd23ff228badc94 # save the attached .config to linux build tree GCC_VERSION=7.4.0 make.cross ARCH=sparc64 If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): In file included from arch/sparc/kernel/iommu.c:20:0: >> include/linux/pci.h:2317:13: error: 'pci_pr3_present' defined but not used [-Werror=unused-function] static bool pci_pr3_present(struct pci_dev *pdev) { return false; } ^~~~~~~~~~~~~~~ cc1: all warnings being treated as errors vim +/pci_pr3_present +2317 include/linux/pci.h 2310 2311 void 2312 pci_msi_register_fwnode_provider(struct fwnode_handle *(*fn)(struct device *)); 2313 bool pci_pr3_present(struct pci_dev *pdev); 2314 #else 2315 static inline struct irq_domain * 2316 pci_host_bridge_acpi_msi_domain(struct pci_bus *bus) { return NULL; } > 2317 static bool pci_pr3_present(struct pci_dev *pdev) { return false; } 2318 #endif 2319 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 1

[alsa-devel] [GIT PULL] ASoC fixes for 5.4
by Mark Brown 21 Oct '19

21 Oct '19

The following changes since commit 147162f575152db80000fb3de26358264768ee9f: ASoC: ti: fix SND_SOC_DM365_VOICE_CODEC dependencies (2019-09-20 12:06:48 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.4-rc4 for you to fetch changes up to 95a32c98055f664f9b3f34c41e153d4dcedd0eff: ASoC: SOF: control: return true when kcontrol values change (2019-10-18 18:09:22 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.4 A collection of fixes that have arrived since the merge window. There are a small number of core fixes here but they are smaller ones around error handling. ---------------------------------------------------------------- Bard Liao (1): ASoC: intel: sof_rt5682: use separate route map for dmic Chuhong Yuan (1): ASoC: Intel: sof-rt5682: add a check for devm_clk_get Dan Carpenter (2): ASoC: soc-component: fix a couple missing error assignments ASoC: topology: Fix a signedness bug in soc_tplg_dapm_widget_create() Daniel Baluta (1): ASoC: simple_card_utils.h: Fix potential multiple redefinition error Dragos Tarcatu (1): ASoC: SOF: control: return true when kcontrol values change Hans de Goede (1): ASoC: core: Fix pcm code debugfs error Jaska Uimonen (3): ASoC: rt5682: add NULL handler to set_jack function ASoC: intel: sof_rt5682: add remove function to disable jack ASoC: intel: bytcr_rt5651: add null check to support_button_press Junya Monden (1): ASoC: rsnd: Reinitialize bit clock inversion flag for every format setting Kai Vehmanen (2): ASoC: SOF: pcm: fix resource leak in hw_free ASoC: SOF: Intel: hda: fix warnings during FW load Keyon Jie (1): ASoC: SOF: topology: fix parse fail issue for byte/bool tuple types Li Xu (1): ASoC: wm_adsp: Fix theoretical NULL pointer for alg_region Liam Girdwood (1): ASoC: SOF: Intel: initialise and verify FW crash dump data. Olivier Moysan (1): ASoC: stm32: sai: fix sysclk management on shutdown Pan Xiuli (1): ASoC: SOF: pcm: harden PCM STOP sequence Pierre-Louis Bossart (1): ASoC: SOF: loader: fix kernel oops on firmware boot failure Ranjani Sridharan (1): ASoC: SOF: Intel: hda: Disable DMI L1 entry during capture Robin Murphy (1): ASoc: rockchip: i2s: Fix RPM imbalance Sathyanarayana Nujella (1): ASoC: max98373: check for device node before parsing Srinivas Kandagatla (1): ASoC: msm8916-wcd-digital: add missing MIX2 path for RX1/2 Stuart Henderson (1): ASoC: wm_adsp: Don't generate kcontrols without READ flags Sylwester Nawrocki (2): ASoC: samsung: arndale: Add missing OF node dereferencing ASoC: wm8994: Do not register inapplicable controls for WM1811 include/sound/simple_card_utils.h | 8 ++--- sound/soc/codecs/max98373.c | 20 +++++++----- sound/soc/codecs/msm8916-wcd-digital.c | 22 +++++++++++++ sound/soc/codecs/rt5651.c | 3 ++ sound/soc/codecs/rt5682.c | 12 +++++-- sound/soc/codecs/wm8994.c | 43 ++++++++++++++---------- sound/soc/codecs/wm_adsp.c | 10 ++++-- sound/soc/intel/boards/sof_rt5682.c | 60 +++++++++++++++++++++++++++++++--- sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/samsung/arndale_rt5631.c | 34 ++++++++++++++++--- sound/soc/sh/rcar/core.c | 1 + sound/soc/soc-pcm.c | 17 +++++++--- sound/soc/soc-topology.c | 2 +- sound/soc/sof/control.c | 26 ++++++++++----- sound/soc/sof/intel/Kconfig | 10 ++++++ sound/soc/sof/intel/bdw.c | 7 ++++ sound/soc/sof/intel/byt.c | 6 ++++ sound/soc/sof/intel/hda-ctrl.c | 12 +++---- sound/soc/sof/intel/hda-loader.c | 1 + sound/soc/sof/intel/hda-stream.c | 45 +++++++++++++++++++++---- sound/soc/sof/intel/hda.c | 7 ++++ sound/soc/sof/intel/hda.h | 5 ++- sound/soc/sof/loader.c | 4 +-- sound/soc/sof/pcm.c | 35 +++++++++++++------- sound/soc/sof/topology.c | 4 ++- sound/soc/stm/stm32_sai_sub.c | 21 ++++++------ 26 files changed, 320 insertions(+), 97 deletions(-)

2 1

[alsa-devel] ALC711 support
by Kailang 21 Oct '19

21 Oct '19

Hi Takashi, Attach patch was support ALC711 HDA mode. BR, Kailang

2 1

[alsa-devel] Question about "957ce0c6b8a1 ASoC: soc-pcm: check symmetry after hw_params"
by S.j. Wang 21 Oct '19

21 Oct '19

Hi Jiada, Mark After this commit "957ce0c6b8a1 ASoC: soc-pcm: check symmetry after hw_params" The soc_pcm_params_symmetry can't take effect, because its position is after the change of cpu_dai->rate. The code of soc_pcm_params_symmetry is: static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { ... if (symmetry && cpu_dai->rate && cpu_dai->rate != rate) { dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n", cpu_dai->rate, rate); return -EINVAL; } ... } This code soc_pcm_hw_params of is: static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { ... /* store the parameters for each DAIs */ cpu_dai->rate = params_rate(params); cpu_dai->channels = params_channels(params); cpu_dai->sample_bits = snd_pcm_format_physical_width(params_format(params)); snd_soc_dapm_update_dai(substream, params, cpu_dai); ... ret = soc_pcm_params_symmetry(substream, params); if (ret) goto component_err; ... } Is this an issue? Best regards Wang shengjiu

2 1

[alsa-devel] [PATCH v2] dt-bindings: sound: Convert Samsung Exynos Odroid XU3/XU4 audio complex to dt-schema
by Marek Szyprowski 21 Oct '19

21 Oct '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec to newer dt-schema format. 'clocks' property is unneeded in the bindings and is left undefined in 'properties'. 'samsung,audio-widgets' and 'samsung,audio-routing' are optional from driver perspective and they are set as unrequired. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> [mszyprow: reordered non-standard properties] Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- .../bindings/sound/samsung,odroid.txt | 54 ----------- .../bindings/sound/samsung,odroid.yaml | 91 +++++++++++++++++++ 2 files changed, 91 insertions(+), 54 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung,odroid.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,odroid.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.txt b/Documentation/devicetree/bindings/sound/samsung,odroid.txt deleted file mode 100644 index e9da2200e173..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.txt +++ /dev/null @@ -1,54 +0,0 @@ -Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec - -Required properties: - - - compatible - "hardkernel,odroid-xu3-audio" - for Odroid XU3 board, - "hardkernel,odroid-xu4-audio" - for Odroid XU4 board (deprecated), - "samsung,odroid-xu3-audio" - for Odroid XU3 board (deprecated), - "samsung,odroid-xu4-audio" - for Odroid XU4 board (deprecated) - - model - the user-visible name of this sound complex - - clocks - should contain entries matching clock names in the clock-names - property - - samsung,audio-widgets - this property specifies off-codec audio elements - like headphones or speakers, for details see widgets.txt - - samsung,audio-routing - a list of the connections between audio - components; each entry is a pair of strings, the first being the - connection's sink, the second being the connection's source; - valid names for sources and sinks are the MAX98090's pins (as - documented in its binding), and the jacks on the board - - For Odroid X2: - "Headphone Jack", "Mic Jack", "DMIC" - - For Odroid U3, XU3: - "Headphone Jack", "Speakers" - - For Odroid XU4: - no entries - -Required sub-nodes: - - - 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S - controller - - 'codec' subnode with a 'sound-dai' property containing list of phandles - to the CODEC nodes, first entry must be corresponding to the MAX98090 - CODEC and the second entry must be the phandle of the HDMI IP block node - -Example: - -sound { - compatible = "hardkernel,odroid-xu3-audio"; - model = "Odroid-XU3"; - samsung,audio-routing = - "Headphone Jack", "HPL", - "Headphone Jack", "HPR", - "IN1", "Mic Jack", - "Mic Jack", "MICBIAS"; - - cpu { - sound-dai = <&i2s0 0>; - }; - codec { - sound-dai = <&hdmi>, <&max98090>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml new file mode 100644 index 000000000000..c6b244352d05 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml @@ -0,0 +1,91 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,odroid.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + oneOf: + - const: hardkernel,odroid-xu3-audio + + - const: hardkernel,odroid-xu4-audio + deprecated: true + + - const: samsung,odroid-xu3-audio + deprecated: true + + - const: samsung,odroid-xu4-audio + deprecated: true + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + cpu: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandles to the I2S controllers + + codec: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + List of phandles to the CODEC nodes, + first entry must be corresponding to the MAX98090 CODEC and + the second entry must be the phandle of the HDMI IP block node. + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio + components; each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source; + valid names for sources and sinks are the MAX98090's pins (as + documented in its binding), and the jacks on the board. + For Odroid X2: "Headphone Jack", "Mic Jack", "DMIC" + For Odroid U3, XU3: "Headphone Jack", "Speakers" + For Odroid XU4: no entries + + samsung,audio-widgets: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + This property specifies off-codec audio elements + like headphones or speakers, for details see widgets.txt + +required: + - compatible + - model + - cpu + - codec + +examples: + - | + sound { + compatible = "hardkernel,odroid-xu3-audio"; + model = "Odroid-XU3"; + samsung,audio-routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "IN1", "Mic Jack", + "Mic Jack", "MICBIAS"; + + cpu { + sound-dai = <&i2s0 0>; + }; + + codec { + sound-dai = <&hdmi>, <&max98090>; + }; + }; + -- 2.17.1

4 3

[alsa-devel] [PATCH 0/6] soundwire: intel/cadence: simplify PDI handling
by Pierre-Louis Bossart 21 Oct '19

21 Oct '19

These patches were originally submitted as '[RFC PATCH 00/11] soundwire: intel: simplify DAI/PDI handling'. There were no comments received. This series only provides the PDI changes, which makes it simpler to review. The DAI changes will be provided with the complete series for ASoC/SOF integration, which is a larger change. Bard Liao (3): soundwire: intel: fix intel_register_dai PDI offsets and numbers soundwire: intel: remove playback/capture stream_name soundwire: cadence_master: improve PDI allocation Pierre-Louis Bossart (3): soundwire: remove DAI_ID_RANGE definitions soundwire: cadence/intel: simplify PDI/port mapping soundwire: intel: don't filter out PDI0/1 drivers/soundwire/cadence_master.c | 158 +++++++---------------------- drivers/soundwire/cadence_master.h | 34 ++----- drivers/soundwire/intel.c | 155 ++++++---------------------- include/linux/soundwire/sdw.h | 3 - 4 files changed, 73 insertions(+), 277 deletions(-) -- 2.20.1

2 8

[alsa-devel] Fwd: [PATCH 29/46] Input: touchscreen: use wrapper for pxa2xx ac97 registers
by Arnd Bergmann 18 Oct '19

18 Oct '19

On Fri, Oct 18, 2019 at 8:48 PM Dmitry Torokhov <dmitry.torokhov(a)gmail.com> wrote: > > On Fri, Oct 18, 2019 at 05:41:44PM +0200, Arnd Bergmann wrote: > > To avoid a dependency on the pxa platform header files with > > hardcoded registers, change the driver to call a wrapper > > in the pxa2xx-ac97-lib that encapsulates all the other > > ac97 stuff. > > Not supper happy about adding module dependencies. Can we include > mach/regs-ac97.h from include/sound/pxa2xx-lib.h and use static inlines? > Someone needs to include mach/regs-ac97.h in the end... > > Or there is something later in the series that needs it? One of the goals of the series is to completely remove all mach/*.h headers and place them somewhere else, ideally inaccessible to device drivers. In case of mach/regs-ac97.h, the later patch "ASoC: pxa: ac97: use normal MMIO accessors" passes the physical register base address as a platform device resource that gets ioremapped in the ac97 driver, rather than hardcoding the virtual address in a global header. I agree that the exported function is still ugly, but I hope it's enough of an improvement over the previous state that we can do it anyway. If you have any better ideas though, I can give that a try, too. One possibility would be a higher-level interface exported on top of 'struct snd_ac97', but I could not figure out how to do this. Arnd

2 1

[alsa-devel] [PATCH v4 00/10] ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: support WoV
by Tzung-Bi Shih 18 Oct '19

18 Oct '19

This series makes mt6358, cros_ec_codec, and mt8183-mt6358-ts3a227-max98357 support WoV (wake on voice). The first 3 commits are some cleanups and refactors. It looks like breaking the existing interface. But please be noticed that, the cros_ec_codec has not used by any real device yet. The refactor is very necessary to keep the style consistent and for easier to further extend and maintain. platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX The 4th commit extends the feature offered from EC codec. platform/chrome: cros_ec: add common commands for EC codec The 5th commit changes the behavior of setting and getting DMIC gains. ASoC: cros_ec_codec: read max DMIC gain from EC codec The 6th and 7th commit make cros_ec_codec support WoV. ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV The 8th commit sets necessary registers on mt6358 to support WoV. ASoC: mediatek: mt6358: support WoV The last 2 commit make machine driver mt8183-mt6358-ts3a227-max98357 support WoV if ec-codec is in DTS. ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV Changes from v3: - fix a compile error by adding ifdef https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156750.h… - fix a coccinelle warning https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156754.h… Changes from v2: - rebase upon to "don't use snd_pcm_ops" series https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156170.h… - fix sparse errors https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156328.h… - use "reg" for SHM binding https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156657.h… Changes from v1: - fix a compile error and make kbuild bot happy https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156315.h… Tzung-Bi Shih (10): platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX platform/chrome: cros_ec: add common commands for EC codec ASoC: cros_ec_codec: read max DMIC gain from EC codec ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV ASoC: mediatek: mt6358: support WoV ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV .../bindings/sound/google,cros-ec-codec.txt | 24 +- .../sound/mt8183-mt6358-ts3a227-max98357.txt | 3 + drivers/platform/chrome/cros_ec_trace.c | 5 +- .../linux/platform_data/cros_ec_commands.h | 285 ++++- sound/soc/codecs/cros_ec_codec.c | 1128 +++++++++++++---- sound/soc/codecs/mt6358.c | 105 ++ sound/soc/mediatek/Kconfig | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 70 +- 8 files changed, 1296 insertions(+), 325 deletions(-) -- 2.23.0.700.g56cf767bdb-goog

2 16

[alsa-devel] [PATCH] ASoC: stm32: sai: fix sysclk management on shutdown
by Olivier Moysan 18 Oct '19

18 Oct '19

The commit below, adds a call to sysclk callback on shutdown. This introduces a regression in stm32 SAI driver, as some clock services are called twice, leading to unbalanced calls. Move processing related to mclk from shutdown to sysclk callback. When requested frequency is 0, assume shutdown and release mclk. Fixes: 2458adb8f92a ("SoC: simple-card-utils: set 0Hz to sysclk when shutdown") Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index d7501f88aaa6..a4060813bc74 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -505,10 +505,20 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai, if (dir == SND_SOC_CLOCK_OUT && sai->sai_mclk) { ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, - (unsigned int)~SAI_XCR1_NODIV); + freq ? 0 : SAI_XCR1_NODIV); if (ret < 0) return ret; + /* Assume shutdown if requested frequency is 0Hz */ + if (!freq) { + /* Release mclk rate only if rate was actually set */ + if (sai->mclk_rate) { + clk_rate_exclusive_put(sai->sai_mclk); + sai->mclk_rate = 0; + } + return 0; + } + /* If master clock is used, set parent clock now */ ret = stm32_sai_set_parent_clock(sai, freq); if (ret) @@ -1093,15 +1103,6 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, STM_SAI_IMR_REGX, SAI_XIMR_MASK, 0); - regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, - SAI_XCR1_NODIV); - - /* Release mclk rate only if rate was actually set */ - if (sai->mclk_rate) { - clk_rate_exclusive_put(sai->sai_mclk); - sai->mclk_rate = 0; - } - clk_disable_unprepare(sai->sai_ck); spin_lock_irqsave(&sai->irq_lock, flags); -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: sof-rt5682: add a check for devm_clk_get
by Chuhong Yuan 18 Oct '19

18 Oct '19

sof_audio_probe misses a check for devm_clk_get and may cause problems. Add a check for it to fix the bug. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/intel/boards/sof_rt5682.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index a437567b8cee..6d15c7ff66bf 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -576,6 +576,15 @@ static int sof_audio_probe(struct platform_device *pdev) /* need to get main clock from pmc */ if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) { ctx->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (IS_ERR(ctx->mclk)) { + ret = PTR_ERR(ctx->mclk); + + dev_err(&pdev->dev, + "Failed to get MCLK from pmc_plt_clk_3: %d\n", + ret); + return ret; + } + ret = clk_prepare_enable(ctx->mclk); if (ret < 0) { dev_err(&pdev->dev, -- 2.20.1

3 2

[alsa-devel] [PATCH] ASoC: rsnd: Reinitialize bit clock inversion flag for every format setting
by Eugeniu Rosca 18 Oct '19

18 Oct '19

From: Junya Monden <jmonden(a)jp.adit-jv.com> Unlike other format-related DAI parameters, rdai->bit_clk_inv flag is not properly re-initialized when setting format for new stream processing. The inversion, if requested, is then applied not to default, but to a previous value, which leads to SCKP bit in SSICR register being set incorrectly. Fix this by re-setting the flag to its initial value, determined by format. Fixes: 1a7889ca8aba3 ("ASoC: rsnd: fixup SND_SOC_DAIFMT_xB_xF behavior") Cc: Andrew Gabbasov <andrew_gabbasov(a)mentor.com> Cc: Jiada Wang <jiada_wang(a)mentor.com> Cc: Timo Wischer <twischer(a)de.adit-jv.com> Cc: stable(a)vger.kernel.org # v3.17+ Signed-off-by: Junya Monden <jmonden(a)jp.adit-jv.com> Signed-off-by: Eugeniu Rosca <erosca(a)de.adit-jv.com> --- sound/soc/sh/rcar/core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index bda5b958d0dc..e9596c2096cd 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -761,6 +761,7 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } /* set format */ + rdai->bit_clk_inv = 0; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: rdai->sys_delay = 0; -- 2.23.0

3 2

[alsa-devel] [PATCH] ASoC: SOF: control: return true when kcontrol values change
by Pierre-Louis Bossart 18 Oct '19

18 Oct '19

From: Dragos Tarcatu <dragos_tarcatu(a)mentor.com> All the kcontrol put() functions are currently returning 0 when successful. This does not go well with alsamixer as it does not seem to get notified on SND_CTL_EVENT_MASK_VALUE callbacks when values change for (some of) the sof kcontrols. This patch fixes that by returning true for volume, switch and enum type kcontrols when values do change in put(). Signed-off-by: Dragos Tarcatu <dragos_tarcatu(a)mentor.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/control.c | 26 ++++++++++++++++++-------- 1 file changed, 18 insertions(+), 8 deletions(-) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index 2c4abd406c4f..7baf7f1507c3 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -91,13 +91,16 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = scontrol->sdev; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + bool change = false; + u32 value; /* update each channel */ for (i = 0; i < channels; i++) { - cdata->chanv[i].value = - mixer_to_ipc(ucontrol->value.integer.value[i], + value = mixer_to_ipc(ucontrol->value.integer.value[i], scontrol->volume_table, sm->max + 1); + change = change || (value != cdata->chanv[i].value); cdata->chanv[i].channel = i; + cdata->chanv[i].value = value; } /* notify DSP of mixer updates */ @@ -107,8 +110,7 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol, SOF_CTRL_TYPE_VALUE_CHAN_GET, SOF_CTRL_CMD_VOLUME, true); - - return 0; + return change; } int snd_sof_switch_get(struct snd_kcontrol *kcontrol, @@ -136,11 +138,15 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = scontrol->sdev; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + bool change = false; + u32 value; /* update each channel */ for (i = 0; i < channels; i++) { - cdata->chanv[i].value = ucontrol->value.integer.value[i]; + value = ucontrol->value.integer.value[i]; + change = change || (value != cdata->chanv[i].value); cdata->chanv[i].channel = i; + cdata->chanv[i].value = value; } if (scontrol->led_ctl.use_led) @@ -154,7 +160,7 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol, SOF_CTRL_CMD_SWITCH, true); - return 0; + return change; } int snd_sof_enum_get(struct snd_kcontrol *kcontrol, @@ -182,11 +188,15 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = scontrol->sdev; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + bool change = false; + u32 value; /* update each channel */ for (i = 0; i < channels; i++) { - cdata->chanv[i].value = ucontrol->value.enumerated.item[i]; + value = ucontrol->value.enumerated.item[i]; + change = change || (value != cdata->chanv[i].value); cdata->chanv[i].channel = i; + cdata->chanv[i].value = value; } /* notify DSP of enum updates */ @@ -197,7 +207,7 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol, SOF_CTRL_CMD_ENUM, true); - return 0; + return change; } int snd_sof_bytes_get(struct snd_kcontrol *kcontrol, -- 2.20.1

2 1

[alsa-devel] [PATCH v3 00/10] ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: support WoV
by Tzung-Bi Shih 18 Oct '19

18 Oct '19

This series makes mt6358, cros_ec_codec, and mt8183-mt6358-ts3a227-max98357 support WoV (wake on voice). The first 3 commits are some cleanups and refactors. It looks like breaking the existing interface. But please be noticed that, the cros_ec_codec has not used by any real device yet. The refactor is very necessary to keep the style consistent and for easier to further extend and maintain. platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX The 4th commit extends the feature offered from EC codec. platform/chrome: cros_ec: add common commands for EC codec The 5th commit changes the behavior of setting and getting DMIC gains. ASoC: cros_ec_codec: read max DMIC gain from EC codec The 6th and 7th commit make cros_ec_codec support WoV. ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV The 8th commit sets necessary registers on mt6358 to support WoV. ASoC: mediatek: mt6358: support WoV The last 2 commit make machine driver mt8183-mt6358-ts3a227-max98357 support WoV if ec-codec is in DTS. ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV Changes from v1: - fix a compile error and make kbuild bot happy https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156315.h… Changes from v2: - rebase upon to "don't use snd_pcm_ops" series https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156170.h… - fix sparse errors https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156328.h… - use "reg" for SHM binding https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156657.h… Tzung-Bi Shih (10): WIP: platform/chrome: cros_ec: remove unused EC feature WIP: ASoC: cros_ec_codec: refactor I2S RX WIP: ASoC: cros_ec_codec: extract DMIC EC command from I2S RX WIP: platform/chrome: cros_ec: add common commands for EC codec WIP: ASoC: cros_ec_codec: read max DMIC gain from EC codec WIP: ASoC: dt-bindings: cros_ec_codec: add SHM bindings WIP: ASoC: cros_ec_codec: support WoV WIP: ASoC: mediatek: mt6358: support WoV WIP: ASoC: dt-bindings: mt8183: add ec-codec WIP: ASoC: mediatek: mt8183: support WoV .../bindings/sound/google,cros-ec-codec.txt | 24 +- .../sound/mt8183-mt6358-ts3a227-max98357.txt | 3 + drivers/platform/chrome/cros_ec_trace.c | 5 +- .../linux/platform_data/cros_ec_commands.h | 285 ++++- sound/soc/codecs/cros_ec_codec.c | 1128 +++++++++++++---- sound/soc/codecs/mt6358.c | 105 ++ sound/soc/mediatek/Kconfig | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 70 +- 8 files changed, 1296 insertions(+), 325 deletions(-) -- 2.23.0.700.g56cf767bdb-goog

5 28

[alsa-devel] [PATCH 1/2] ASoc: Add Texas Instruments TAS2562 amplifier binding
by Dan Murphy 18 Oct '19

18 Oct '19

Add the DT binding for the TAS2562 amplifier. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2562.txt | 34 +++++++++++++++++++ 1 file changed, 34 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2562.txt diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt new file mode 100644 index 000000000000..658e1fb18a99 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2562.txt @@ -0,0 +1,34 @@ +Texas Instruments TAS2562 Smart PA + +The TAS2562 is a mono, digital input Class-D audio amplifier optimized for +efficiently driving high peak power into small loudspeakers. +Integrated speaker voltage and current sense provides for +real time monitoring of loudspeaker behavior. + +Required properties: + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - compatible: - Should contain "ti,tas2562". + - reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f. + - ti,imon-slot-no:- TDM TX current sense time slot. + +Optional properties: +- interrupt-parent: phandle to the interrupt controller which provides + the interrupt. +- interrupts: (GPIO) interrupt to which the chip is connected. +- shut-down: GPIO used to control the state of the device. + +Examples: +tas2562@4c { + #address-cells = <1>; + #size-cells = <0>; + compatible = "ti,tas2562"; + reg = <0x4c>; + + interrupt-parent = <&gpio1>; + interrupts = <14>; + + shut-down = <&gpio1 15 0>; + ti,imon-slot-no = <0>; +}; + -- 2.22.0.214.g8dca754b1e

3 4

[alsa-devel] [PATCH][next] ASoC: rt1011: fix spelling mistake "temperture" -> "temperature"
by Colin King 18 Oct '19

18 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_dbg message. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt1011.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index ad049cfddcb0..dcd397a83cb4 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -2373,7 +2373,7 @@ static int rt1011_parse_dp(struct rt1011_priv *rt1011, struct device *dev) device_property_read_u32(dev, "realtek,r0_calib", &rt1011->r0_calib); - dev_dbg(dev, "%s: r0_calib: 0x%x, temperture_calib: 0x%x", + dev_dbg(dev, "%s: r0_calib: 0x%x, temperature_calib: 0x%x", __func__, rt1011->r0_calib, rt1011->temperature_calib); return 0; -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic32x4: add a check for devm_clk_get
by Chuhong Yuan 18 Oct '19

18 Oct '19

aic32x4_set_dai_sysclk misses a check for devm_clk_get and may miss the failure. Add a check to fix it. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/codecs/tlv320aic32x4.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 68165de1c8de..b4e9a6c73f90 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -573,6 +573,9 @@ static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct clk *pll; pll = devm_clk_get(component->dev, "pll"); + if (IS_ERR(pll)) + return PTR_ERR(pll); + mclk = clk_get_parent(pll); return clk_set_rate(mclk, freq); -- 2.20.1

2 1

[alsa-devel] [PATCH v3] ASoC: dt-bindings: Convert Allwinner A10 codec to a schema
by Maxime Ripard 18 Oct '19

18 Oct '19

The Allwinner SoCs have an embedded audio codec that is supported in Linux, with a matching Device Tree binding. Now that we have the DT validation in place, let's convert the device tree bindings for that controller over to a YAML schemas. Signed-off-by: Maxime Ripard <mripard(a)kernel.org> --- Changes from v2: - Change the audio-routing values to an enum, and enforce boundaries on the size - Add restrictions to the possible values of audio-routing Changes from v1: - Fix subject prefix --- .../sound/allwinner,sun4i-a10-codec.yaml | 267 ++++++++++++++++++ .../devicetree/bindings/sound/sun4i-codec.txt | 94 ------ 2 files changed, 267 insertions(+), 94 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml delete mode 100644 Documentation/devicetree/bindings/sound/sun4i-codec.txt diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml new file mode 100644 index 000000000000..b8f89c7258eb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml @@ -0,0 +1,267 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A10 Codec Device Tree Bindings + +maintainers: + - Chen-Yu Tsai <wens(a)csie.org> + - Maxime Ripard <maxime.ripard(a)bootlin.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + enum: + - allwinner,sun4i-a10-codec + - allwinner,sun6i-a31-codec + - allwinner,sun7i-a20-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: codec + + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + + resets: + maxItems: 1 + + allwinner,audio-routing: + description: |- + A list of the connections between audio components. Each entry + is a pair of strings, the first being the connection's sink, the + second being the connection's source. + allOf: + - $ref: /schemas/types.yaml#definitions/non-unique-string-array + - minItems: 2 + maxItems: 18 + items: + enum: + # Audio Pins on the SoC + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + + # Microphone Biases from the SoC + - HBIAS + - MBIAS + + # Board Connectors + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + allwinner,codec-analog-controls: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to the codec analog controls in the PRCM + + allwinner,pa-gpios: + description: GPIO to enable the external amplifier + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +allOf: + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + then: + if: + properties: + compatible: + const: allwinner,sun6i-a31-codec + + then: + required: + - resets + - allwinner,audio-routing + + else: + required: + - resets + - allwinner,audio-routing + - allwinner,codec-analog-controls + + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-a23-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - LINEIN + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-h3-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-v3s-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - MIC1 + - HBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + +additionalProperties: false + +examples: + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun7i-a20-codec"; + reg = <0x01c22c00 0x40>; + interrupts = <0 30 4>; + clocks = <&apb0_gates 0>, <&codec_clk>; + clock-names = "apb", "codec"; + dmas = <&dma 0 19>, <&dma 0 19>; + dma-names = "rx", "tx"; + }; + + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun6i-a31-codec"; + reg = <0x01c22c00 0x98>; + interrupts = <0 29 4>; + clocks = <&ccu 61>, <&ccu 135>; + clock-names = "apb", "codec"; + resets = <&ccu 42>; + dmas = <&dma 15>, <&dma 15>; + dma-names = "rx", "tx"; + allwinner,audio-routing = + "Headphone", "HP", + "Speaker", "LINEOUT", + "LINEIN", "Line In", + "MIC1", "MBIAS", + "MIC1", "Mic", + "MIC2", "HBIAS", + "MIC2", "Headset Mic"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt deleted file mode 100644 index 66579bbd3294..000000000000 --- a/Documentation/devicetree/bindings/sound/sun4i-codec.txt +++ /dev/null @@ -1,94 +0,0 @@ -* Allwinner A10 Codec - -Required properties: -- compatible: must be one of the following compatibles: - - "allwinner,sun4i-a10-codec" - - "allwinner,sun6i-a31-codec" - - "allwinner,sun7i-a20-codec" - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- reg: must contain the registers location and length -- interrupts: must contain the codec interrupt -- dmas: DMA channels for tx and rx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should include "tx" and "rx". -- clocks: a list of phandle + clock-specifer pairs, one for each entry - in clock-names. -- clock-names: should contain the following: - - "apb": the parent APB clock for this controller - - "codec": the parent module clock - -Optional properties: -- allwinner,pa-gpios: gpio to enable external amplifier - -Required properties for the following compatibles: - - "allwinner,sun6i-a31-codec" - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- resets: phandle to the reset control for this device -- allwinner,audio-routing: A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. Valid names include: - - Audio pins on the SoC: - "HP" - "HPCOM" - "LINEIN" (not on sun8i-v3s) - "LINEOUT" (not on sun8i-a23 or sun8i-v3s) - "MIC1" - "MIC2" (not on sun8i-v3s) - "MIC3" (sun6i-a31 only) - - Microphone biases from the SoC: - "HBIAS" - "MBIAS" (not on sun8i-v3s) - - Board connectors: - "Headphone" - "Headset Mic" - "Line In" - "Line Out" - "Mic" - "Speaker" - -Required properties for the following compatibles: - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- allwinner,codec-analog-controls: A phandle to the codec analog controls - block in the PRCM. - -Example: -codec: codec@1c22c00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun7i-a20-codec"; - reg = <0x01c22c00 0x40>; - interrupts = <0 30 4>; - clocks = <&apb0_gates 0>, <&codec_clk>; - clock-names = "apb", "codec"; - dmas = <&dma 0 19>, <&dma 0 19>; - dma-names = "rx", "tx"; -}; - -codec: codec@1c22c00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun6i-a31-codec"; - reg = <0x01c22c00 0x98>; - interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>; - clocks = <&ccu CLK_APB1_CODEC>, <&ccu CLK_CODEC>; - clock-names = "apb", "codec"; - resets = <&ccu RST_APB1_CODEC>; - dmas = <&dma 15>, <&dma 15>; - dma-names = "rx", "tx"; - allwinner,audio-routing = - "Headphone", "HP", - "Speaker", "LINEOUT", - "LINEIN", "Line In", - "MIC1", "MBIAS", - "MIC1", "Mic", - "MIC2", "HBIAS", - "MIC2", "Headset Mic"; -}; -- 2.23.0

3 2

[alsa-devel] [PATCH v6] dt-bindings: sound: Convert Samsung I2S controller to dt-schema
by Marek Szyprowski 18 Oct '19

18 Oct '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung I2S controller to newer dt-schema format. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> [mszyprow: integrated fix for minor spelling issues] Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Reviewed-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- v6: - integrated fix for minor spelling issues - collected tags --- .../devicetree/bindings/sound/samsung-i2s.txt | 84 ----------- .../bindings/sound/samsung-i2s.yaml | 138 ++++++++++++++++++ 2 files changed, 138 insertions(+), 84 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt deleted file mode 100644 index a88cb00fa096..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ /dev/null @@ -1,84 +0,0 @@ -* Samsung I2S controller - -Required SoC Specific Properties: - -- compatible : should be one of the following. - - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. - - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with - secondary fifo, s/w reset control and internal mux for root clk src. - - samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for - playback, stereo channel capture, secondary fifo using internal - or external dma, s/w reset control, internal mux for root clk src - and 7.1 channel TDM support for playback. TDM (Time division multiplexing) - is to allow transfer of multiple channel audio data on single data line. - - samsung,exynos7-i2s: with all the available features of exynos5 i2s, - exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo - with only external dma and more no.of root clk sampling frequencies. - - samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports - stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with - slightly modified bit offsets. - -- reg: physical base address of the controller and length of memory mapped - region. -- dmas: list of DMA controller phandle and DMA request line ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. -- clocks: Handle to iis clock and RCLK source clk. -- clock-names: - i2s0 uses some base clocks from CMU and some are from audio subsystem internal - clock controller. The clock names for i2s0 should be "iis", "i2s_opclk0" and - "i2s_opclk1" as shown in the example below. - i2s1 and i2s2 uses clocks from CMU. The clock names for i2s1 and i2s2 should - be "iis" and "i2s_opclk0". - "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources of the root - clk. i2s0 has internal mux to select the source of root clk and i2s1 and i2s2 - doesn't have any such mux. -- #clock-cells: should be 1, this property must be present if the I2S device - is a clock provider in terms of the common clock bindings, described in - ../clock/clock-bindings.txt. -- clock-output-names (deprecated): from the common clock bindings, names of - the CDCLK I2S output clocks, suggested values are "i2s_cdclk0", "i2s_cdclk1", - "i2s_cdclk3" for the I2S0, I2S1, I2S2 devices respectively. - -There are following clocks available at the I2S device nodes: - CLK_I2S_CDCLK - the CDCLK (CODECLKO) gate clock, - CLK_I2S_RCLK_PSR - the RCLK prescaler divider clock (corresponding to the - IISPSR register), - CLK_I2S_RCLK_SRC - the RCLKSRC mux clock (corresponding to RCLKSRC bit in - IISMOD register). - -Refer to the SoC datasheet for availability of the above clocks. -The CLK_I2S_RCLK_PSR and CLK_I2S_RCLK_SRC clocks are usually only available -in the IIS Multi Audio Interface. - -Note: Old DTs may not have the #clock-cells property and then not use the I2S -node as a clock supplier. - -Optional SoC Specific Properties: - -- samsung,idma-addr: Internal DMA register base address of the audio - sub system(used in secondary sound source). -- pinctrl-0: Should specify pin control groups used for this controller. -- pinctrl-names: Should contain only one value - "default". -- #sound-dai-cells: should be 1. - - -Example: - -i2s0: i2s@3830000 { - compatible = "samsung,s5pv210-i2s"; - reg = <0x03830000 0x100>; - dmas = <&pdma0 10 - &pdma0 9 - &pdma0 8>; - dma-names = "tx", "rx", "tx-sec"; - clocks = <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_SCLK_I2S>; - clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; - #clock-cells = <1>; - samsung,idma-addr = <0x03000000>; - pinctrl-names = "default"; - pinctrl-0 = <&i2s0_bus>; - #sound-dai-cells = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.yaml b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml new file mode 100644 index 000000000000..53e3bad4178c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml @@ -0,0 +1,138 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung SoC I2S controller + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + description: | + samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. + + samsung,s5pv210-i2s: for 8/16/24bit multichannel (5.1) I2S with + secondary FIFO, s/w reset control and internal mux for root clock + source. + + samsung,exynos5420-i2s: for 8/16/24bit multichannel (5.1) I2S for + playback, stereo channel capture, secondary FIFO using internal + or external DMA, s/w reset control, internal mux for root clock + source and 7.1 channel TDM support for playback; TDM (Time division + multiplexing) is to allow transfer of multiple channel audio data on + single data line. + + samsung,exynos7-i2s: with all the available features of Exynos5 I2S. + Exynos7 I2S has 7.1 channel TDM support for capture, secondary FIFO + with only external DMA and more number of root clock sampling + frequencies. + + samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports + stereo channels. Exynos7 I2S1 upgraded to 5.1 multichannel with + slightly modified bit offsets. + enum: + - samsung,s3c6410-i2s + - samsung,s5pv210-i2s + - samsung,exynos5420-i2s + - samsung,exynos7-i2s + - samsung,exynos7-i2s1 + + reg: + maxItems: 1 + + dmas: + minItems: 2 + maxItems: 3 + + dma-names: + oneOf: + - items: + - const: tx + - const: rx + - items: + - const: tx + - const: rx + - const: tx-sec + + clocks: + minItems: 1 + maxItems: 3 + + clock-names: + oneOf: + - items: + - const: iis + - items: # for I2S0 + - const: iis + - const: i2s_opclk0 + - const: i2s_opclk1 + - items: # for I2S1 and I2S2 + - const: iis + - const: i2s_opclk0 + description: | + "iis" is the I2S bus clock and i2s_opclk0, i2s_opclk1 are sources + of the root clock. I2S0 has internal mux to select the source + of root clock and I2S1 and I2S2 doesn't have any such mux. + + "#clock-cells": + const: 1 + + clock-output-names: + deprecated: true + oneOf: + - items: # for I2S0 + - const: i2s_cdclk0 + - items: # for I2S1 + - const: i2s_cdclk1 + - items: # for I2S2 + - const: i2s_cdclk2 + description: Names of the CDCLK I2S output clocks. + + samsung,idma-addr: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Internal DMA register base address of the audio + subsystem (used in secondary sound source). + + pinctrl-0: + description: Should specify pin control groups used for this controller. + + pinctrl-names: + const: default + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - clock-names + +examples: + - | + #include <dt-bindings/clock/exynos-audss-clk.h> + + i2s0: i2s@3830000 { + compatible = "samsung,s5pv210-i2s"; + reg = <0x03830000 0x100>; + dmas = <&pdma0 10>, + <&pdma0 9>, + <&pdma0 8>; + dma-names = "tx", "rx", "tx-sec"; + clocks = <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_SCLK_I2S>; + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + samsung,idma-addr = <0x03000000>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + #sound-dai-cells = <1>; + }; -- 2.17.1

2 1

[alsa-devel] [PATCH 00/12] ALSA: firewire: configure interval of hw IRQ via ALSA PCM interface
by Takashi Sakamoto 18 Oct '19

18 Oct '19

Hi, ALSA IEC 61883-1/6 packet streaming engine controls OHCI 1394 controller to generate hardware IRQ for fixed number (=16) of isoc packets. Additionally, the size of packet buffer is fixed for triple of interval of hardware IRQ (=48). This commit allows ALSA PCM applications to configure these parameters by ALSA PCM interface. As a result, the size of PCM period is available in the range between 125 usec and 64 msec (or 32 msec). The hardware IRQ is scheduled for isoc packet which includes event across PCM period boundary. Takashi Sakamoto (12): ALSA: firewire-lib: use variable size of queue for isoc packets instead of fixed size ALSA: bebob: share PCM buffer size for both direction ALSA: fireworks: share PCM buffer size for both direction ALSA: oxfw: share PCM buffer size for both direction ALSA: dice: share PCM buffer size for both direction ALSA: firewire-digi00x: share PCM buffer size for both direction ALSA: firewire-tascam: share PCM buffer size for both direction ALSA: firewire-motu: share PCM buffer size for both direction ALSA: fireface: share PCM buffer size for both direction ALSA: firewire-lib: schedule hardware IRQ according to the size of PCM period ALSA: firewire-lib: tune the maximum available size of PCM period ALSA: firewire-lib: tune the minimum available size of PCM period sound/firewire/amdtp-stream.c | 154 ++++++++++++++------ sound/firewire/amdtp-stream.h | 9 +- sound/firewire/bebob/bebob.h | 3 +- sound/firewire/bebob/bebob_midi.c | 2 +- sound/firewire/bebob/bebob_pcm.c | 12 +- sound/firewire/bebob/bebob_stream.c | 5 +- sound/firewire/dice/dice-midi.c | 2 +- sound/firewire/dice/dice-pcm.c | 20 ++- sound/firewire/dice/dice-stream.c | 5 +- sound/firewire/dice/dice.h | 3 +- sound/firewire/digi00x/digi00x-midi.c | 2 +- sound/firewire/digi00x/digi00x-pcm.c | 12 +- sound/firewire/digi00x/digi00x-stream.c | 5 +- sound/firewire/digi00x/digi00x.h | 3 +- sound/firewire/fireface/ff-pcm.c | 13 +- sound/firewire/fireface/ff-stream.c | 5 +- sound/firewire/fireface/ff.h | 3 +- sound/firewire/fireworks/fireworks.h | 3 +- sound/firewire/fireworks/fireworks_midi.c | 2 +- sound/firewire/fireworks/fireworks_pcm.c | 12 +- sound/firewire/fireworks/fireworks_stream.c | 5 +- sound/firewire/motu/motu-midi.c | 2 +- sound/firewire/motu/motu-pcm.c | 12 +- sound/firewire/motu/motu-stream.c | 5 +- sound/firewire/motu/motu.h | 3 +- sound/firewire/oxfw/oxfw-midi.c | 4 +- sound/firewire/oxfw/oxfw-pcm.c | 17 ++- sound/firewire/oxfw/oxfw-stream.c | 5 +- sound/firewire/oxfw/oxfw.h | 3 +- sound/firewire/tascam/tascam-pcm.c | 12 +- sound/firewire/tascam/tascam-stream.c | 5 +- sound/firewire/tascam/tascam.h | 3 +- 32 files changed, 266 insertions(+), 85 deletions(-) -- 2.20.1

2 13