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[alsa-devel] [asoc:for-4.20 519/524] sound/soc//stm/stm32_sai_sub.c:379:2: error: unknown field 'recalc_rate' specified in initializer
by kbuild test robot 19 Oct '18

19 Oct '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 318e741ee13b5a72f3051d9bb6852b1f4d02d0bb commit: 8307b2afd386ccce369821daa2196068c47fe8cd [519/524] ASoC: stm32: sai: set sai as mclk clock provider config: openrisc-allyesconfig (attached as .config) compiler: or1k-linux-gcc (GCC) 6.0.0 20160327 (experimental) reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 8307b2afd386ccce369821daa2196068c47fe8cd # save the attached .config to linux build tree make.cross ARCH=openrisc All errors (new ones prefixed by >>): sound/soc//stm/stm32_sai_sub.c:260:16: error: field 'hw' has incomplete type struct clk_hw hw; ^~ In file included from include/linux/err.h:5:0, from include/linux/clk.h:15, from sound/soc//stm/stm32_sai_sub.c:19: sound/soc//stm/stm32_sai_sub.c: In function 'stm32_sai_mclk_round_rate': include/linux/kernel.h:997:32: error: dereferencing pointer to incomplete type 'struct clk_hw' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~ include/linux/compiler.h:335:18: note: in definition of macro '__compiletime_assert' int __cond = !(condition); \ ^~~~~~~~~ include/linux/compiler.h:358:2: note: in expansion of macro '_compiletime_assert' _compiletime_assert(condition, msg, __compiletime_assert_, __LINE__) ^~~~~~~~~~~~~~~~~~~ include/linux/build_bug.h:45:37: note: in expansion of macro 'compiletime_assert' #define BUILD_BUG_ON_MSG(cond, msg) compiletime_assert(!(cond), msg) ^~~~~~~~~~~~~~~~~~ include/linux/kernel.h:997:2: note: in expansion of macro 'BUILD_BUG_ON_MSG' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~~~~~~~~~~~ include/linux/kernel.h:997:20: note: in expansion of macro '__same_type' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c:265:27: note: in expansion of macro 'container_of' #define to_mclk_data(_hw) container_of(_hw, struct stm32_sai_mclk_data, hw) ^~~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c:313:37: note: in expansion of macro 'to_mclk_data' struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); ^~~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c: At top level: sound/soc//stm/stm32_sai_sub.c:376:21: error: variable 'mclk_ops' has initializer but incomplete type static const struct clk_ops mclk_ops = { ^~~~~~~ >> sound/soc//stm/stm32_sai_sub.c:377:2: error: unknown field 'enable' specified in initializer .enable = stm32_sai_mclk_enable, ^ sound/soc//stm/stm32_sai_sub.c:377:12: warning: excess elements in struct initializer .enable = stm32_sai_mclk_enable, ^~~~~~~~~~~~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c:377:12: note: (near initialization for 'mclk_ops') >> sound/soc//stm/stm32_sai_sub.c:378:2: error: unknown field 'disable' specified in initializer .disable = stm32_sai_mclk_disable, ^ sound/soc//stm/stm32_sai_sub.c:378:13: warning: excess elements in struct initializer .disable = stm32_sai_mclk_disable, ^~~~~~~~~~~~~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c:378:13: note: (near initialization for 'mclk_ops') >> sound/soc//stm/stm32_sai_sub.c:379:2: error: unknown field 'recalc_rate' specified in initializer .recalc_rate = stm32_sai_mclk_recalc_rate, ^ sound/soc//stm/stm32_sai_sub.c:379:17: warning: excess elements in struct initializer .recalc_rate = stm32_sai_mclk_recalc_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c:379:17: note: (near initialization for 'mclk_ops') >> sound/soc//stm/stm32_sai_sub.c:380:2: error: unknown field 'round_rate' specified in initializer .round_rate = stm32_sai_mclk_round_rate, ^ sound/soc//stm/stm32_sai_sub.c:380:16: warning: excess elements in struct initializer .round_rate = stm32_sai_mclk_round_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c:380:16: note: (near initialization for 'mclk_ops') >> sound/soc//stm/stm32_sai_sub.c:381:2: error: unknown field 'set_rate' specified in initializer .set_rate = stm32_sai_mclk_set_rate, ^ sound/soc//stm/stm32_sai_sub.c:381:14: warning: excess elements in struct initializer .set_rate = stm32_sai_mclk_set_rate, ^~~~~~~~~~~~~~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c:381:14: note: (near initialization for 'mclk_ops') sound/soc//stm/stm32_sai_sub.c: In function 'stm32_sai_add_mclk_provider': >> sound/soc//stm/stm32_sai_sub.c:389:22: error: implicit declaration of function '__clk_get_name' [-Werror=implicit-function-declaration] const char *pname = __clk_get_name(sai->sai_ck); ^~~~~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c:389:22: warning: initialization makes pointer from integer without a cast [-Wint-conversion] >> sound/soc//stm/stm32_sai_sub.c:414:18: error: implicit declaration of function 'CLK_HW_INIT' [-Werror=implicit-function-declaration] mclk->hw.init = CLK_HW_INIT(mclk_name, pname, &mclk_ops, 0); ^~~~~~~~~~~ >> sound/soc//stm/stm32_sai_sub.c:419:8: error: implicit declaration of function 'devm_clk_hw_register' [-Werror=implicit-function-declaration] ret = devm_clk_hw_register(&sai->pdev->dev, hw); ^~~~~~~~~~~~~~~~~~~~ >> sound/soc//stm/stm32_sai_sub.c:427:9: error: implicit declaration of function 'devm_of_clk_add_hw_provider' [-Werror=implicit-function-declaration] return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw); ^~~~~~~~~~~~~~~~~~~~~~~~~~~ >> sound/soc//stm/stm32_sai_sub.c:427:42: error: 'of_clk_hw_simple_get' undeclared (first use in this function) return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw); ^~~~~~~~~~~~~~~~~~~~ sound/soc//stm/stm32_sai_sub.c:427:42: note: each undeclared identifier is reported only once for each function it appears in sound/soc//stm/stm32_sai_sub.c: At top level: sound/soc//stm/stm32_sai_sub.c:376:29: error: storage size of 'mclk_ops' isn't known static const struct clk_ops mclk_ops = { ^~~~~~~~ sound/soc//stm/stm32_sai_sub.c: In function 'stm32_sai_add_mclk_provider': sound/soc//stm/stm32_sai_sub.c:428:1: warning: control reaches end of non-void function [-Wreturn-type] } ^ cc1: some warnings being treated as errors vim +/recalc_rate +379 sound/soc//stm/stm32_sai_sub.c 309 310 static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate, 311 unsigned long *prate) 312 { > 313 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 314 struct stm32_sai_sub_data *sai = mclk->sai_data; 315 int div; 316 317 div = stm32_sai_get_clk_div(sai, *prate, rate); 318 if (div < 0) 319 return div; 320 321 mclk->freq = *prate / div; 322 323 return mclk->freq; 324 } 325 326 static unsigned long stm32_sai_mclk_recalc_rate(struct clk_hw *hw, 327 unsigned long parent_rate) 328 { 329 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 330 331 return mclk->freq; 332 } 333 334 static int stm32_sai_mclk_set_rate(struct clk_hw *hw, unsigned long rate, 335 unsigned long parent_rate) 336 { 337 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 338 struct stm32_sai_sub_data *sai = mclk->sai_data; 339 unsigned int div; 340 int ret; 341 342 div = stm32_sai_get_clk_div(sai, parent_rate, rate); 343 if (div < 0) 344 return div; 345 346 ret = stm32_sai_set_clk_div(sai, div); 347 if (ret) 348 return ret; 349 350 mclk->freq = rate; 351 352 return 0; 353 } 354 355 static int stm32_sai_mclk_enable(struct clk_hw *hw) 356 { 357 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 358 struct stm32_sai_sub_data *sai = mclk->sai_data; 359 360 dev_dbg(&sai->pdev->dev, "Enable master clock\n"); 361 362 return regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, 363 SAI_XCR1_MCKEN, SAI_XCR1_MCKEN); 364 } 365 366 static void stm32_sai_mclk_disable(struct clk_hw *hw) 367 { 368 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 369 struct stm32_sai_sub_data *sai = mclk->sai_data; 370 371 dev_dbg(&sai->pdev->dev, "Disable master clock\n"); 372 373 regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_MCKEN, 0); 374 } 375 376 static const struct clk_ops mclk_ops = { > 377 .enable = stm32_sai_mclk_enable, > 378 .disable = stm32_sai_mclk_disable, > 379 .recalc_rate = stm32_sai_mclk_recalc_rate, > 380 .round_rate = stm32_sai_mclk_round_rate, > 381 .set_rate = stm32_sai_mclk_set_rate, 382 }; 383 384 static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai) 385 { 386 struct clk_hw *hw; 387 struct stm32_sai_mclk_data *mclk; 388 struct device *dev = &sai->pdev->dev; > 389 const char *pname = __clk_get_name(sai->sai_ck); 390 char *mclk_name, *p, *s = (char *)pname; 391 int ret, i = 0; 392 393 mclk = devm_kzalloc(dev, sizeof(mclk), GFP_KERNEL); 394 if (!mclk) 395 return -ENOMEM; 396 397 mclk_name = devm_kcalloc(dev, sizeof(char), 398 SAI_MCLK_NAME_LEN, GFP_KERNEL); 399 if (!mclk_name) 400 return -ENOMEM; 401 402 /* 403 * Forge mclk clock name from parent clock name and suffix. 404 * String after "_" char is stripped in parent name. 405 */ 406 p = mclk_name; 407 while (*s && *s != '_' && (i < (SAI_MCLK_NAME_LEN - 6))) { 408 *p++ = *s++; 409 i++; 410 } 411 STM_SAI_IS_SUB_A(sai) ? 412 strncat(p, "a_mclk", 6) : strncat(p, "b_mclk", 6); 413 > 414 mclk->hw.init = CLK_HW_INIT(mclk_name, pname, &mclk_ops, 0); 415 mclk->sai_data = sai; 416 hw = &mclk->hw; 417 418 dev_dbg(dev, "Register master clock %s\n", mclk_name); > 419 ret = devm_clk_hw_register(&sai->pdev->dev, hw); 420 if (ret) { 421 dev_err(dev, "mclk register returned %d\n", ret); 422 return ret; 423 } 424 sai->sai_mclk = hw->clk; 425 426 /* register mclk provider */ > 427 return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw); 428 } 429 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH v2 0/9] Add support for audiocodec in Allwinner A64
by Vasily Khoruzhick 19 Oct '18

19 Oct '18

This series adds Allwinner A64 audiocodec support into sun4i-i2s, sun8i-codec drivers, introduces new sun50i-codec-analog driver and enables sound on Pine64, SoPine boards and Pinebook. I2S for audiocodec in A64 is different from other 3 I2S modules but similar to one in A10, digital part of codec is compatible with A33 and analog controls part is completely different from other SoCs - it shares only few bits in few registers, so adding support for it into existing sun8i-codec-analog would mean duplicating all the widgets, controls and some routes and making it hard to read. Therefore it makes sense to introduce new driver. v2: - Use simple-amplifier for speaker amp on Pinebook - Rename sun50i-a64-i2s to sun50i-a64-codec-i2s to preserve compatible string for other 3 I2S modules in A64 in case if there's any incompatibility with H3 Marcus Cooper (1): ASoC: sun4i-i2s: Add compatibility with A64 codec I2S Vasily Khoruzhick (8): ASoC: sun8i-codec: Don't hardcode BCLK / LRCK ratio ASoC: sun8i-codec-analog: split regmap code into separate driver ASoC: dt-binding: Add bindings for Allwinner A64 codec's analog path controls ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64 arm64: dts: allwinner: a64: add nodes necessary for analog sound support arm64: dts: allwinner: a64: enable sound on Pine64 and SoPine arm64: dts: allwinner: a64: enable sound on Pinebook .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + .../bindings/sound/sun50i-codec-analog.txt | 12 + .../boot/dts/allwinner/sun50i-a64-pine64.dts | 30 ++ .../dts/allwinner/sun50i-a64-pinebook.dts | 42 ++ .../allwinner/sun50i-a64-sopine-baseboard.dts | 30 ++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 58 +++ sound/soc/sunxi/Kconfig | 17 +- sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-i2s.c | 21 + sound/soc/sunxi/sun50i-codec-analog.c | 444 ++++++++++++++++++ sound/soc/sunxi/sun8i-codec-analog.c | 79 +--- sound/soc/sunxi/sun8i-codec.c | 35 +- sound/soc/sunxi/sunxi-adda-pr-regmap.c | 102 ++++ sound/soc/sunxi/sunxi-adda-pr-regmap.h | 7 + 14 files changed, 798 insertions(+), 83 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt create mode 100644 sound/soc/sunxi/sun50i-codec-analog.c create mode 100644 sound/soc/sunxi/sunxi-adda-pr-regmap.c create mode 100644 sound/soc/sunxi/sunxi-adda-pr-regmap.h -- 2.19.0

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[alsa-devel] [PATCH 1/1] ASoC: cs42l51: fix mclk support
by Olivier Moysan 19 Oct '18

19 Oct '18

Fixes: 5e8d63a726f8 ("ASoC: cs42l51: add mclk support") The MCLK clock is made optional for cs42l51 codec. However, ASoC DAPM clock supply widget, expects the clock to be defined unconditionally. Register MCLK DAPM conditionally in codec driver, depending on clock presence in DT. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/codecs/cs42l51.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index eb40bff54cec..fd2bd74024c1 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -21,6 +21,7 @@ * - master mode *NOT* supported */ +#include <linux/clk.h> #include <linux/module.h> #include <linux/slab.h> #include <sound/core.h> @@ -41,6 +42,7 @@ enum master_slave_mode { struct cs42l51_private { unsigned int mclk; + struct clk *mclk_handle; unsigned int audio_mode; /* The mode (I2S or left-justified) */ enum master_slave_mode func; }; @@ -492,9 +494,13 @@ static int cs42l51_component_probe(struct snd_soc_component *component) { int ret, reg; struct snd_soc_dapm_context *dapm; + struct cs42l51_private *cs42l51; + cs42l51 = snd_soc_component_get_drvdata(component); dapm = snd_soc_component_get_dapm(component); - snd_soc_dapm_new_controls(dapm, cs42l51_dapm_mclk_widgets, 1); + + if (cs42l51->mclk_handle) + snd_soc_dapm_new_controls(dapm, cs42l51_dapm_mclk_widgets, 1); /* * DAC configuration @@ -548,6 +554,13 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap) dev_set_drvdata(dev, cs42l51); + cs42l51->mclk_handle = devm_clk_get(dev, "MCLK"); + if (IS_ERR(cs42l51->mclk_handle)) { + if (PTR_ERR(cs42l51->mclk_handle) != -ENOENT) + return PTR_ERR(cs42l51->mclk_handle); + cs42l51->mclk_handle = NULL; + } + /* Verify that we have a CS42L51 */ ret = regmap_read(regmap, CS42L51_CHIP_REV_ID, &val); if (ret < 0) { -- 2.7.4

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[alsa-devel] [asoc:for-4.20 519/523] sound/soc/stm/stm32_sai_sub.c:260:16: error: field 'hw' has incomplete type
by kbuild test robot 19 Oct '18

19 Oct '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 7ece50c9f790270e82b77589d34e54547c895a66 commit: 8307b2afd386ccce369821daa2196068c47fe8cd [519/523] ASoC: stm32: sai: set sai as mclk clock provider config: powerpc-allyesconfig (attached as .config) compiler: powerpc64-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 8307b2afd386ccce369821daa2196068c47fe8cd # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=powerpc All error/warnings (new ones prefixed by >>): >> sound/soc/stm/stm32_sai_sub.c:260:16: error: field 'hw' has incomplete type struct clk_hw hw; ^~ In file included from include/linux/err.h:5:0, from include/linux/clk.h:15, from sound/soc/stm/stm32_sai_sub.c:19: sound/soc/stm/stm32_sai_sub.c: In function 'stm32_sai_mclk_round_rate': include/linux/kernel.h:997:32: error: dereferencing pointer to incomplete type 'struct clk_hw' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~ include/linux/compiler.h:335:18: note: in definition of macro '__compiletime_assert' int __cond = !(condition); \ ^~~~~~~~~ include/linux/compiler.h:358:2: note: in expansion of macro '_compiletime_assert' _compiletime_assert(condition, msg, __compiletime_assert_, __LINE__) ^~~~~~~~~~~~~~~~~~~ include/linux/build_bug.h:45:37: note: in expansion of macro 'compiletime_assert' #define BUILD_BUG_ON_MSG(cond, msg) compiletime_assert(!(cond), msg) ^~~~~~~~~~~~~~~~~~ include/linux/kernel.h:997:2: note: in expansion of macro 'BUILD_BUG_ON_MSG' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~~~~~~~~~~~ include/linux/kernel.h:997:20: note: in expansion of macro '__same_type' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~~~~~~ >> sound/soc/stm/stm32_sai_sub.c:265:27: note: in expansion of macro 'container_of' #define to_mclk_data(_hw) container_of(_hw, struct stm32_sai_mclk_data, hw) ^~~~~~~~~~~~ >> sound/soc/stm/stm32_sai_sub.c:313:37: note: in expansion of macro 'to_mclk_data' struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); ^~~~~~~~~~~~ sound/soc/stm/stm32_sai_sub.c: At top level: >> sound/soc/stm/stm32_sai_sub.c:376:21: error: variable 'mclk_ops' has initializer but incomplete type static const struct clk_ops mclk_ops = { ^~~~~~~ >> sound/soc/stm/stm32_sai_sub.c:377:3: error: 'const struct clk_ops' has no member named 'enable' .enable = stm32_sai_mclk_enable, ^~~~~~ >> sound/soc/stm/stm32_sai_sub.c:377:12: warning: excess elements in struct initializer .enable = stm32_sai_mclk_enable, ^~~~~~~~~~~~~~~~~~~~~ sound/soc/stm/stm32_sai_sub.c:377:12: note: (near initialization for 'mclk_ops') >> sound/soc/stm/stm32_sai_sub.c:378:3: error: 'const struct clk_ops' has no member named 'disable' .disable = stm32_sai_mclk_disable, ^~~~~~~ sound/soc/stm/stm32_sai_sub.c:378:13: warning: excess elements in struct initializer .disable = stm32_sai_mclk_disable, ^~~~~~~~~~~~~~~~~~~~~~ sound/soc/stm/stm32_sai_sub.c:378:13: note: (near initialization for 'mclk_ops') >> sound/soc/stm/stm32_sai_sub.c:379:3: error: 'const struct clk_ops' has no member named 'recalc_rate' .recalc_rate = stm32_sai_mclk_recalc_rate, ^~~~~~~~~~~ sound/soc/stm/stm32_sai_sub.c:379:17: warning: excess elements in struct initializer .recalc_rate = stm32_sai_mclk_recalc_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/stm/stm32_sai_sub.c:379:17: note: (near initialization for 'mclk_ops') >> sound/soc/stm/stm32_sai_sub.c:380:3: error: 'const struct clk_ops' has no member named 'round_rate' .round_rate = stm32_sai_mclk_round_rate, ^~~~~~~~~~ sound/soc/stm/stm32_sai_sub.c:380:16: warning: excess elements in struct initializer .round_rate = stm32_sai_mclk_round_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/stm/stm32_sai_sub.c:380:16: note: (near initialization for 'mclk_ops') >> sound/soc/stm/stm32_sai_sub.c:381:3: error: 'const struct clk_ops' has no member named 'set_rate' .set_rate = stm32_sai_mclk_set_rate, ^~~~~~~~ sound/soc/stm/stm32_sai_sub.c:381:14: warning: excess elements in struct initializer .set_rate = stm32_sai_mclk_set_rate, ^~~~~~~~~~~~~~~~~~~~~~~ sound/soc/stm/stm32_sai_sub.c:381:14: note: (near initialization for 'mclk_ops') sound/soc/stm/stm32_sai_sub.c: In function 'stm32_sai_add_mclk_provider': >> sound/soc/stm/stm32_sai_sub.c:389:22: error: implicit declaration of function '__clk_get_name'; did you mean 'clk_get_rate'? [-Werror=implicit-function-declaration] const char *pname = __clk_get_name(sai->sai_ck); ^~~~~~~~~~~~~~ clk_get_rate >> sound/soc/stm/stm32_sai_sub.c:389:22: warning: initialization makes pointer from integer without a cast [-Wint-conversion] >> sound/soc/stm/stm32_sai_sub.c:414:18: error: implicit declaration of function 'CLK_HW_INIT'; did you mean 'KLIST_INIT'? [-Werror=implicit-function-declaration] mclk->hw.init = CLK_HW_INIT(mclk_name, pname, &mclk_ops, 0); ^~~~~~~~~~~ KLIST_INIT >> sound/soc/stm/stm32_sai_sub.c:419:8: error: implicit declaration of function 'devm_clk_hw_register'; did you mean 'device_unregister'? [-Werror=implicit-function-declaration] ret = devm_clk_hw_register(&sai->pdev->dev, hw); ^~~~~~~~~~~~~~~~~~~~ device_unregister >> sound/soc/stm/stm32_sai_sub.c:427:9: error: implicit declaration of function 'devm_of_clk_add_hw_provider'; did you mean 'of_clk_get_from_provider'? [-Werror=implicit-function-declaration] return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw); ^~~~~~~~~~~~~~~~~~~~~~~~~~~ of_clk_get_from_provider >> sound/soc/stm/stm32_sai_sub.c:427:42: error: 'of_clk_hw_simple_get' undeclared (first use in this function); did you mean 'ida_simple_get'? return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw); ^~~~~~~~~~~~~~~~~~~~ ida_simple_get sound/soc/stm/stm32_sai_sub.c:427:42: note: each undeclared identifier is reported only once for each function it appears in sound/soc/stm/stm32_sai_sub.c: At top level: >> sound/soc/stm/stm32_sai_sub.c:376:29: error: storage size of 'mclk_ops' isn't known static const struct clk_ops mclk_ops = { ^~~~~~~~ sound/soc/stm/stm32_sai_sub.c: In function 'stm32_sai_add_mclk_provider': >> sound/soc/stm/stm32_sai_sub.c:428:1: warning: control reaches end of non-void function [-Wreturn-type] } ^ cc1: some warnings being treated as errors vim +/hw +260 sound/soc/stm/stm32_sai_sub.c 258 259 struct stm32_sai_mclk_data { > 260 struct clk_hw hw; 261 unsigned long freq; 262 struct stm32_sai_sub_data *sai_data; 263 }; 264 > 265 #define to_mclk_data(_hw) container_of(_hw, struct stm32_sai_mclk_data, hw) 266 #define STM32_SAI_MAX_CLKS 1 267 268 static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, 269 unsigned long input_rate, 270 unsigned long output_rate) 271 { 272 int version = sai->pdata->conf->version; 273 int div; 274 275 div = DIV_ROUND_CLOSEST(input_rate, output_rate); 276 if (div > SAI_XCR1_MCKDIV_MAX(version)) { 277 dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); 278 return -EINVAL; 279 } 280 dev_dbg(&sai->pdev->dev, "SAI divider %d\n", div); 281 282 if (input_rate % div) 283 dev_dbg(&sai->pdev->dev, 284 "Rate not accurate. requested (%ld), actual (%ld)\n", 285 output_rate, input_rate / div); 286 287 return div; 288 } 289 290 static int stm32_sai_set_clk_div(struct stm32_sai_sub_data *sai, 291 unsigned int div) 292 { 293 int version = sai->pdata->conf->version; 294 int ret, cr1, mask; 295 296 if (div > SAI_XCR1_MCKDIV_MAX(version)) { 297 dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); 298 return -EINVAL; 299 } 300 301 mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version)); 302 cr1 = SAI_XCR1_MCKDIV_SET(div); 303 ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1); 304 if (ret < 0) 305 dev_err(&sai->pdev->dev, "Failed to update CR1 register\n"); 306 307 return ret; 308 } 309 310 static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate, 311 unsigned long *prate) 312 { > 313 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 314 struct stm32_sai_sub_data *sai = mclk->sai_data; 315 int div; 316 317 div = stm32_sai_get_clk_div(sai, *prate, rate); 318 if (div < 0) 319 return div; 320 321 mclk->freq = *prate / div; 322 323 return mclk->freq; 324 } 325 326 static unsigned long stm32_sai_mclk_recalc_rate(struct clk_hw *hw, 327 unsigned long parent_rate) 328 { 329 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 330 331 return mclk->freq; 332 } 333 334 static int stm32_sai_mclk_set_rate(struct clk_hw *hw, unsigned long rate, 335 unsigned long parent_rate) 336 { 337 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 338 struct stm32_sai_sub_data *sai = mclk->sai_data; 339 unsigned int div; 340 int ret; 341 342 div = stm32_sai_get_clk_div(sai, parent_rate, rate); 343 if (div < 0) 344 return div; 345 346 ret = stm32_sai_set_clk_div(sai, div); 347 if (ret) 348 return ret; 349 350 mclk->freq = rate; 351 352 return 0; 353 } 354 355 static int stm32_sai_mclk_enable(struct clk_hw *hw) 356 { 357 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 358 struct stm32_sai_sub_data *sai = mclk->sai_data; 359 360 dev_dbg(&sai->pdev->dev, "Enable master clock\n"); 361 362 return regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, 363 SAI_XCR1_MCKEN, SAI_XCR1_MCKEN); 364 } 365 366 static void stm32_sai_mclk_disable(struct clk_hw *hw) 367 { 368 struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); 369 struct stm32_sai_sub_data *sai = mclk->sai_data; 370 371 dev_dbg(&sai->pdev->dev, "Disable master clock\n"); 372 373 regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_MCKEN, 0); 374 } 375 > 376 static const struct clk_ops mclk_ops = { > 377 .enable = stm32_sai_mclk_enable, > 378 .disable = stm32_sai_mclk_disable, > 379 .recalc_rate = stm32_sai_mclk_recalc_rate, > 380 .round_rate = stm32_sai_mclk_round_rate, > 381 .set_rate = stm32_sai_mclk_set_rate, 382 }; 383 384 static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai) 385 { 386 struct clk_hw *hw; 387 struct stm32_sai_mclk_data *mclk; 388 struct device *dev = &sai->pdev->dev; > 389 const char *pname = __clk_get_name(sai->sai_ck); 390 char *mclk_name, *p, *s = (char *)pname; 391 int ret, i = 0; 392 393 mclk = devm_kzalloc(dev, sizeof(mclk), GFP_KERNEL); 394 if (!mclk) 395 return -ENOMEM; 396 397 mclk_name = devm_kcalloc(dev, sizeof(char), 398 SAI_MCLK_NAME_LEN, GFP_KERNEL); 399 if (!mclk_name) 400 return -ENOMEM; 401 402 /* 403 * Forge mclk clock name from parent clock name and suffix. 404 * String after "_" char is stripped in parent name. 405 */ 406 p = mclk_name; 407 while (*s && *s != '_' && (i < (SAI_MCLK_NAME_LEN - 6))) { 408 *p++ = *s++; 409 i++; 410 } 411 STM_SAI_IS_SUB_A(sai) ? 412 strncat(p, "a_mclk", 6) : strncat(p, "b_mclk", 6); 413 > 414 mclk->hw.init = CLK_HW_INIT(mclk_name, pname, &mclk_ops, 0); 415 mclk->sai_data = sai; 416 hw = &mclk->hw; 417 418 dev_dbg(dev, "Register master clock %s\n", mclk_name); > 419 ret = devm_clk_hw_register(&sai->pdev->dev, hw); 420 if (ret) { 421 dev_err(dev, "mclk register returned %d\n", ret); 422 return ret; 423 } 424 sai->sai_mclk = hw->clk; 425 426 /* register mclk provider */ > 427 return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw); > 428 } 429 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:for-4.20 519/523] sound/soc/stm/stm32_sai_sub.c:412:3: warning: 'strncat' specified bound 6 equals source length
by kbuild test robot 19 Oct '18

19 Oct '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 7ece50c9f790270e82b77589d34e54547c895a66 commit: 8307b2afd386ccce369821daa2196068c47fe8cd [519/523] ASoC: stm32: sai: set sai as mclk clock provider config: microblaze-allyesconfig (attached as .config) compiler: microblaze-linux-gcc (GCC) 8.1.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 8307b2afd386ccce369821daa2196068c47fe8cd # save the attached .config to linux build tree GCC_VERSION=8.1.0 make.cross ARCH=microblaze All warnings (new ones prefixed by >>): sound/soc/stm/stm32_sai_sub.c: In function 'stm32_sai_add_mclk_provider': >> sound/soc/stm/stm32_sai_sub.c:412:3: warning: 'strncat' specified bound 6 equals source length [-Wstringop-overflow=] strncat(p, "a_mclk", 6) : strncat(p, "b_mclk", 6); ^~~~~~~~~~~~~~~~~~~~~~~ sound/soc/stm/stm32_sai_sub.c:412:29: warning: 'strncat' specified bound 6 equals source length [-Wstringop-overflow=] strncat(p, "a_mclk", 6) : strncat(p, "b_mclk", 6); ^~~~~~~~~~~~~~~~~~~~~~~ vim +/strncat +412 sound/soc/stm/stm32_sai_sub.c 383 384 static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai) 385 { 386 struct clk_hw *hw; 387 struct stm32_sai_mclk_data *mclk; 388 struct device *dev = &sai->pdev->dev; 389 const char *pname = __clk_get_name(sai->sai_ck); 390 char *mclk_name, *p, *s = (char *)pname; 391 int ret, i = 0; 392 393 mclk = devm_kzalloc(dev, sizeof(mclk), GFP_KERNEL); 394 if (!mclk) 395 return -ENOMEM; 396 397 mclk_name = devm_kcalloc(dev, sizeof(char), 398 SAI_MCLK_NAME_LEN, GFP_KERNEL); 399 if (!mclk_name) 400 return -ENOMEM; 401 402 /* 403 * Forge mclk clock name from parent clock name and suffix. 404 * String after "_" char is stripped in parent name. 405 */ 406 p = mclk_name; 407 while (*s && *s != '_' && (i < (SAI_MCLK_NAME_LEN - 6))) { 408 *p++ = *s++; 409 i++; 410 } 411 STM_SAI_IS_SUB_A(sai) ? > 412 strncat(p, "a_mclk", 6) : strncat(p, "b_mclk", 6); 413 414 mclk->hw.init = CLK_HW_INIT(mclk_name, pname, &mclk_ops, 0); 415 mclk->sai_data = sai; 416 hw = &mclk->hw; 417 418 dev_dbg(dev, "Register master clock %s\n", mclk_name); 419 ret = devm_clk_hw_register(&sai->pdev->dev, hw); 420 if (ret) { 421 dev_err(dev, "mclk register returned %d\n", ret); 422 return ret; 423 } 424 sai->sai_mclk = hw->clk; 425 426 /* register mclk provider */ 427 return devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, hw); 428 } 429 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] Applied "ASoC: tegra_sgtl5000: fix device_node refcounting" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: tegra_sgtl5000: fix device_node refcounting has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a85227da2dcc291b762c8482a505bc7d0d2d4b07 Mon Sep 17 00:00:00 2001 From: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Date: Tue, 16 Oct 2018 12:47:29 +0200 Subject: [PATCH] ASoC: tegra_sgtl5000: fix device_node refcounting Similar to the following: commit 4321723648b0 ("ASoC: tegra_alc5632: fix device_node refcounting") commit 7c5dfd549617 ("ASoC: tegra: fix device_node refcounting") Signed-off-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Acked-by: Jon Hunter <jonathanh(a)nvidia.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/tegra/tegra_sgtl5000.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c index 45a4aa9d2a47..901457da25ec 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -149,14 +149,14 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing/invalid\n"); ret = -EINVAL; - goto err; + goto err_put_codec_of_node; } tegra_sgtl5000_dai.platform_of_node = tegra_sgtl5000_dai.cpu_of_node; ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) - goto err; + goto err_put_cpu_of_node; ret = snd_soc_register_card(card); if (ret) { @@ -169,6 +169,13 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&machine->util_data); +err_put_cpu_of_node: + of_node_put(tegra_sgtl5000_dai.cpu_of_node); + tegra_sgtl5000_dai.cpu_of_node = NULL; + tegra_sgtl5000_dai.platform_of_node = NULL; +err_put_codec_of_node: + of_node_put(tegra_sgtl5000_dai.codec_of_node); + tegra_sgtl5000_dai.codec_of_node = NULL; err: return ret; } @@ -183,6 +190,12 @@ static int tegra_sgtl5000_driver_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); + of_node_put(tegra_sgtl5000_dai.cpu_of_node); + tegra_sgtl5000_dai.cpu_of_node = NULL; + tegra_sgtl5000_dai.platform_of_node = NULL; + of_node_put(tegra_sgtl5000_dai.codec_of_node); + tegra_sgtl5000_dai.codec_of_node = NULL; + return ret; } -- 2.19.0.rc2

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[alsa-devel] [PATCH] ASoC: nau8822: new codec driver
by David Lin 19 Oct '18

19 Oct '18

Add driver for NAU88C22. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8822.txt | 16 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8822.c | 1136 +++++++++++++++++ sound/soc/codecs/nau8822.h | 204 +++ 5 files changed, 1363 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8822.txt create mode 100644 sound/soc/codecs/nau8822.c create mode 100644 sound/soc/codecs/nau8822.h diff --git a/Documentation/devicetree/bindings/sound/nau8822.txt b/Documentation/devicetree/bindings/sound/nau8822.txt new file mode 100644 index 000000000000..a471d162d4e5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8822.txt @@ -0,0 +1,16 @@ +NAU8822 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "nuvoton,nau8822" + + - reg : the I2C address of the device. + +Example: + +codec: nau8822@1a { + compatible = "nuvoton,nau8822"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index efb095dbcd71..57365dbc1581 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -109,6 +109,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MT6351 if MTK_PMIC_WRAP select SND_SOC_NAU8540 if I2C select SND_SOC_NAU8810 if I2C + select SND_SOC_NAU8822 if I2C select SND_SOC_NAU8824 if I2C select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC @@ -1299,6 +1300,10 @@ config SND_SOC_NAU8810 tristate "Nuvoton Technology Corporation NAU88C10 CODEC" depends on I2C +config SND_SOC_NAU8822 + tristate "Nuvoton Technology Corporation NAU88C22 CODEC" + depends on I2C + config SND_SOC_NAU8824 tristate "Nuvoton Technology Corporation NAU88L24 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7ae7c85e8219..181a7d222fd7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -106,6 +106,7 @@ snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o +snd-soc-nau8822-objs := nau8822.o snd-soc-nau8824-objs := nau8824.o snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi-codec.o @@ -366,6 +367,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o +obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o obj-$(CONFIG_SND_SOC_NAU8824) += snd-soc-nau8824.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c new file mode 100644 index 000000000000..622ce947f134 --- /dev/null +++ b/sound/soc/codecs/nau8822.c @@ -0,0 +1,1136 @@ +/* + * nau8822.c -- NAU8822 ALSA Soc Audio Codec driver + * + * Copyright 2017 Nuvoton Technology Corp. + * + * Author: David Lin <ctlin0(a)nuvoton.com> + * Co-author: John Hsu <kchsu0(a)nuvoton.com> + * Co-author: Seven Li <wtli(a)nuvoton.com> + * + * Based on WM8974.c + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <asm/div64.h> +#include "nau8822.h" + +#define NAU_PLL_FREQ_MAX 100000000 +#define NAU_PLL_FREQ_MIN 90000000 +#define NAU_PLL_REF_MAX 33000000 +#define NAU_PLL_REF_MIN 8000000 +#define NAU_PLL_OPTOP_MIN 6 + +static const int nau8822_mclk_scaler[] = { 10, 15, 20, 30, 40, 60, 80, 120 }; + +static const struct reg_default nau8822_reg_defaults[] = { + { NAU8822_REG_POWER_MANAGEMENT_1, 0x0000 }, + { NAU8822_REG_POWER_MANAGEMENT_2, 0x0000 }, + { NAU8822_REG_POWER_MANAGEMENT_3, 0x0000 }, + { NAU8822_REG_AUDIO_INTERFACE, 0x0050 }, + { NAU8822_REG_COMPANDING_CONTROL, 0x0000 }, + { NAU8822_REG_CLOCKING, 0x0140 }, + { NAU8822_REG_ADDITIONAL_CONTROL, 0x0000 }, + { NAU8822_REG_GPIO_CONTROL, 0x0000 }, + { NAU8822_REG_JACK_DETECT_CONTROL_1, 0x0000 }, + { NAU8822_REG_DAC_CONTROL, 0x0000 }, + { NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_JACK_DETECT_CONTROL_2, 0x0000 }, + { NAU8822_REG_ADC_CONTROL, 0x0100 }, + { NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0x00ff }, + { NAU8822_REG_EQ1, 0x012c }, + { NAU8822_REG_EQ2, 0x002c }, + { NAU8822_REG_EQ3, 0x002c }, + { NAU8822_REG_EQ4, 0x002c }, + { NAU8822_REG_EQ5, 0x002c }, + { NAU8822_REG_DAC_LIMITER_1, 0x0032 }, + { NAU8822_REG_DAC_LIMITER_2, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_1, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_2, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_3, 0x0000 }, + { NAU8822_REG_NOTCH_FILTER_4, 0x0000 }, + { NAU8822_REG_ALC_CONTROL_1, 0x0038 }, + { NAU8822_REG_ALC_CONTROL_2, 0x000b }, + { NAU8822_REG_ALC_CONTROL_3, 0x0032 }, + { NAU8822_REG_NOISE_GATE, 0x0010 }, + { NAU8822_REG_PLL_N, 0x0008 }, + { NAU8822_REG_PLL_K1, 0x000c }, + { NAU8822_REG_PLL_K2, 0x0093 }, + { NAU8822_REG_PLL_K3, 0x00e9 }, + { NAU8822_REG_3D_CONTROL, 0x0000 }, + { NAU8822_REG_RIGHT_SPEAKER_CONTROL, 0x0000 }, + { NAU8822_REG_INPUT_CONTROL, 0x0033 }, + { NAU8822_REG_LEFT_INP_PGA_CONTROL, 0x0010 }, + { NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0x0010 }, + { NAU8822_REG_LEFT_ADC_BOOST_CONTROL, 0x0100 }, + { NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0x0100 }, + { NAU8822_REG_OUTPUT_CONTROL, 0x0002 }, + { NAU8822_REG_LEFT_MIXER_CONTROL, 0x0001 }, + { NAU8822_REG_RIGHT_MIXER_CONTROL, 0x0001 }, + { NAU8822_REG_LHP_VOLUME, 0x0039 }, + { NAU8822_REG_RHP_VOLUME, 0x0039 }, + { NAU8822_REG_LSPKOUT_VOLUME, 0x0039 }, + { NAU8822_REG_RSPKOUT_VOLUME, 0x0039 }, + { NAU8822_REG_AUX2_MIXER, 0x0001 }, + { NAU8822_REG_AUX1_MIXER, 0x0001 }, + { NAU8822_REG_POWER_MANAGEMENT_4, 0x0000 }, + { NAU8822_REG_LEFT_TIME_SLOT, 0x0000 }, + { NAU8822_REG_MISC, 0x0020 }, + { NAU8822_REG_RIGHT_TIME_SLOT, 0x0000 }, + { NAU8822_REG_DEVICE_REVISION, 0x007f }, + { NAU8822_REG_DEVICE_ID, 0x001a }, + { NAU8822_REG_DAC_DITHER, 0x0114 }, + { NAU8822_REG_ALC_ENHANCE_1, 0x0000 }, + { NAU8822_REG_ALC_ENHANCE_2, 0x0000 }, + { NAU8822_REG_192KHZ_SAMPLING, 0x0008 }, + { NAU8822_REG_MISC_CONTROL, 0x0000 }, + { NAU8822_REG_INPUT_TIEOFF, 0x0000 }, + { NAU8822_REG_POWER_REDUCTION, 0x0000 }, + { NAU8822_REG_AGC_PEAK2PEAK, 0x0000 }, + { NAU8822_REG_AGC_PEAK_DETECT, 0x0000 }, + { NAU8822_REG_AUTOMUTE_CONTROL, 0x0000 }, + { NAU8822_REG_OUTPUT_TIEOFF, 0x0000 }, +}; + +static bool nau8822_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1: + case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5: + case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2: + case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4: + case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3: + case NAU8822_REG_3D_CONTROL: + case NAU8822_REG_RIGHT_SPEAKER_CONTROL: + case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL: + case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER: + case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID: + case NAU8822_REG_DAC_DITHER: + case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL: + case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF: + return true; + default: + return false; + } +} + +static bool nau8822_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8822_REG_RESET ... NAU8822_REG_JACK_DETECT_CONTROL_1: + case NAU8822_REG_DAC_CONTROL ... NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME: + case NAU8822_REG_EQ1 ... NAU8822_REG_EQ5: + case NAU8822_REG_DAC_LIMITER_1 ... NAU8822_REG_DAC_LIMITER_2: + case NAU8822_REG_NOTCH_FILTER_1 ... NAU8822_REG_NOTCH_FILTER_4: + case NAU8822_REG_ALC_CONTROL_1 ...NAU8822_REG_PLL_K3: + case NAU8822_REG_3D_CONTROL: + case NAU8822_REG_RIGHT_SPEAKER_CONTROL: + case NAU8822_REG_INPUT_CONTROL ... NAU8822_REG_LEFT_ADC_BOOST_CONTROL: + case NAU8822_REG_RIGHT_ADC_BOOST_CONTROL ... NAU8822_REG_AUX1_MIXER: + case NAU8822_REG_POWER_MANAGEMENT_4 ... NAU8822_REG_DEVICE_ID: + case NAU8822_REG_DAC_DITHER: + case NAU8822_REG_ALC_ENHANCE_1 ... NAU8822_REG_MISC_CONTROL: + case NAU8822_REG_INPUT_TIEOFF ... NAU8822_REG_OUTPUT_TIEOFF: + return true; + default: + return false; + } +} + +static bool nau8822_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8822_REG_RESET: + case NAU8822_REG_DEVICE_REVISION: + case NAU8822_REG_DEVICE_ID: + case NAU8822_REG_AGC_PEAK2PEAK: + case NAU8822_REG_AGC_PEAK_DETECT: + case NAU8822_REG_AUTOMUTE_CONTROL: + return true; + default: + return false; + } +} + +/* The EQ parameters get function is to get the 5 band equalizer control. + * The regmap raw read can't work here because regmap doesn't provide + * value format for value width of 9 bits. Therefore, the driver reads data + * from cache and makes value format according to the endianness of + * bytes type control element. + */ +static int nau8822_eq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + int i, reg; + u16 reg_val, *val; + + val = (u16 *)ucontrol->value.bytes.data; + reg = NAU8822_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + reg_val = snd_soc_component_read32(component, reg + i); + /* conversion of 16-bit integers between native CPU format + * and big endian format + */ + reg_val = cpu_to_be16(reg_val); + memcpy(val + i, &reg_val, sizeof(reg_val)); + } + + return 0; +} + +/* The EQ parameters put function is to make configuration of 5 band equalizer + * control. These configuration includes central frequency, equalizer gain, + * cut-off frequency, bandwidth control, and equalizer path. + * The regmap raw write can't work here because regmap doesn't provide + * register and value format for register with address 7 bits and value 9 bits. + * Therefore, the driver makes value format according to the endianness of + * bytes type control element and writes data to codec. + */ +static int nau8822_eq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + void *data; + u16 *val, value; + int i, reg, ret; + + data = kmemdup(ucontrol->value.bytes.data, + params->max, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + val = (u16 *)data; + reg = NAU8822_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + /* conversion of 16-bit integers between native CPU format + * and big endian format + */ + value = be16_to_cpu(*(val + i)); + ret = snd_soc_component_write(component, reg + i, value); + if (ret) { + dev_err(component->dev, + "EQ configuration fail, register: %x ret: %d\n", + reg + i, ret); + kfree(data); + return ret; + } + } + kfree(data); + + return 0; +} + +static const char * const nau8822_companding[] = { + "Off", "NC", "u-law", "A-law"}; + +static const struct soc_enum nau8822_companding_adc_enum = + SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_ADCCM_SFT, + ARRAY_SIZE(nau8822_companding), nau8822_companding); + +static const struct soc_enum nau8822_companding_dac_enum = + SOC_ENUM_SINGLE(NAU8822_REG_COMPANDING_CONTROL, NAU8822_DACCM_SFT, + ARRAY_SIZE(nau8822_companding), nau8822_companding); + +static const char * const nau8822_eqmode[] = {"Capture", "Playback"}; + +static const struct soc_enum nau8822_eqmode_enum = + SOC_ENUM_SINGLE(NAU8822_REG_EQ1, NAU8822_EQM_SFT, + ARRAY_SIZE(nau8822_eqmode), nau8822_eqmode); + +static const char * const nau8822_alc1[] = {"Off", "Right", "Left", "Both"}; +static const char * const nau8822_alc3[] = {"Normal", "Limiter"}; + +static const struct soc_enum nau8822_alc_enable_enum = + SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_1, NAU8822_ALCEN_SFT, + ARRAY_SIZE(nau8822_alc1), nau8822_alc1); + +static const struct soc_enum nau8822_alc_mode_enum = + SOC_ENUM_SINGLE(NAU8822_REG_ALC_CONTROL_3, NAU8822_ALCM_SFT, + ARRAY_SIZE(nau8822_alc3), nau8822_alc3); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(pga_boost_tlv, 0, 2000, 0); +static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1); +static const DECLARE_TLV_DB_SCALE(limiter_tlv, 0, 100, 0); + +static const struct snd_kcontrol_new nau8822_snd_controls[] = { + SOC_ENUM("ADC Companding", nau8822_companding_adc_enum), + SOC_ENUM("DAC Companding", nau8822_companding_dac_enum), + + SOC_ENUM("EQ Function", nau8822_eqmode_enum), + SND_SOC_BYTES_EXT("EQ Parameters", 10, + nau8822_eq_get, nau8822_eq_put), + + SOC_DOUBLE("DAC Inversion Switch", + NAU8822_REG_DAC_CONTROL, 0, 1, 1, 0), + SOC_DOUBLE_R_TLV("PCM Volume", + NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv), + + SOC_SINGLE("High Pass Filter Switch", + NAU8822_REG_ADC_CONTROL, 8, 1, 0), + SOC_SINGLE("High Pass Cut Off", + NAU8822_REG_ADC_CONTROL, 4, 7, 0), + + SOC_DOUBLE("ADC Inversion Switch", + NAU8822_REG_ADC_CONTROL, 0, 1, 1, 0), + SOC_DOUBLE_R_TLV("ADC Volume", + NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, 0, 255, 0, digital_tlv), + + SOC_SINGLE("DAC Limiter Switch", + NAU8822_REG_DAC_LIMITER_1, 8, 1, 0), + SOC_SINGLE("DAC Limiter Decay", + NAU8822_REG_DAC_LIMITER_1, 4, 15, 0), + SOC_SINGLE("DAC Limiter Attack", + NAU8822_REG_DAC_LIMITER_1, 0, 15, 0), + SOC_SINGLE("DAC Limiter Threshold", + NAU8822_REG_DAC_LIMITER_2, 4, 7, 0), + SOC_SINGLE_TLV("DAC Limiter Volume", + NAU8822_REG_DAC_LIMITER_2, 0, 12, 0, limiter_tlv), + + SOC_ENUM("ALC Mode", nau8822_alc_mode_enum), + SOC_ENUM("ALC Enable Switch", nau8822_alc_enable_enum), + SOC_SINGLE("ALC Min Gain", + NAU8822_REG_ALC_CONTROL_1, 0, 7, 0), + SOC_SINGLE("ALC Max Gain", + NAU8822_REG_ALC_CONTROL_1, 3, 7, 0), + SOC_SINGLE("ALC Hold", + NAU8822_REG_ALC_CONTROL_2, 4, 10, 0), + SOC_SINGLE("ALC Target", + NAU8822_REG_ALC_CONTROL_2, 0, 15, 0), + SOC_SINGLE("ALC Decay", + NAU8822_REG_ALC_CONTROL_3, 4, 10, 0), + SOC_SINGLE("ALC Attack", + NAU8822_REG_ALC_CONTROL_3, 0, 10, 0), + SOC_SINGLE("ALC Noise Gate Switch", + NAU8822_REG_NOISE_GATE, 3, 1, 0), + SOC_SINGLE("ALC Noise Gate Threshold", + NAU8822_REG_NOISE_GATE, 0, 7, 0), + + SOC_DOUBLE_R("PGA ZC Switch", + NAU8822_REG_LEFT_INP_PGA_CONTROL, + NAU8822_REG_RIGHT_INP_PGA_CONTROL, + 7, 1, 0), + SOC_DOUBLE_R_TLV("PGA Volume", + NAU8822_REG_LEFT_INP_PGA_CONTROL, + NAU8822_REG_RIGHT_INP_PGA_CONTROL, 0, 63, 0, inpga_tlv), + + SOC_DOUBLE_R("Headphone ZC Switch", + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, 7, 1, 0), + SOC_DOUBLE_R("Headphone Playback Switch", + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, 6, 1, 1), + SOC_DOUBLE_R_TLV("Headphone Volume", + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, 0, 63, 0, spk_tlv), + + SOC_DOUBLE_R("Speaker ZC Switch", + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, 7, 1, 0), + SOC_DOUBLE_R("Speaker Playback Switch", + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, 6, 1, 1), + SOC_DOUBLE_R_TLV("Speaker Volume", + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, 0, 63, 0, spk_tlv), + + SOC_DOUBLE_R("AUXOUT Playback Switch", + NAU8822_REG_AUX2_MIXER, + NAU8822_REG_AUX1_MIXER, 6, 1, 1), + + SOC_DOUBLE_R_TLV("PGA Boost Volume", + NAU8822_REG_LEFT_ADC_BOOST_CONTROL, + NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 8, 1, 0, pga_boost_tlv), + SOC_DOUBLE_R_TLV("L2/R2 Boost Volume", + NAU8822_REG_LEFT_ADC_BOOST_CONTROL, + NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 4, 7, 0, boost_tlv), + SOC_DOUBLE_R_TLV("Aux Boost Volume", + NAU8822_REG_LEFT_ADC_BOOST_CONTROL, + NAU8822_REG_RIGHT_ADC_BOOST_CONTROL, 0, 7, 0, boost_tlv), + + SOC_SINGLE("DAC 128x Oversampling Switch", + NAU8822_REG_DAC_CONTROL, 5, 1, 0), + SOC_SINGLE("ADC 128x Oversampling Switch", + NAU8822_REG_ADC_CONTROL, 5, 1, 0), +}; + +/* LMAIN and RMAIN Mixer */ +static const struct snd_kcontrol_new nau8822_left_out_mixer[] = { + SOC_DAPM_SINGLE("LINMIX Switch", + NAU8822_REG_LEFT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("LAUX Switch", + NAU8822_REG_LEFT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("LDAC Switch", + NAU8822_REG_LEFT_MIXER_CONTROL, 0, 1, 0), + SOC_DAPM_SINGLE("RDAC Switch", + NAU8822_REG_OUTPUT_CONTROL, 5, 1, 0), +}; + +static const struct snd_kcontrol_new nau8822_right_out_mixer[] = { + SOC_DAPM_SINGLE("RINMIX Switch", + NAU8822_REG_RIGHT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("RAUX Switch", + NAU8822_REG_RIGHT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("RDAC Switch", + NAU8822_REG_RIGHT_MIXER_CONTROL, 0, 1, 0), + SOC_DAPM_SINGLE("LDAC Switch", + NAU8822_REG_OUTPUT_CONTROL, 6, 1, 0), +}; + +/* AUX1 and AUX2 Mixer */ +static const struct snd_kcontrol_new nau8822_auxout1_mixer[] = { + SOC_DAPM_SINGLE("RDAC Switch", NAU8822_REG_AUX1_MIXER, 0, 1, 0), + SOC_DAPM_SINGLE("RMIX Switch", NAU8822_REG_AUX1_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("RINMIX Switch", NAU8822_REG_AUX1_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX1_MIXER, 3, 1, 0), + SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX1_MIXER, 4, 1, 0), +}; + +static const struct snd_kcontrol_new nau8822_auxout2_mixer[] = { + SOC_DAPM_SINGLE("LDAC Switch", NAU8822_REG_AUX2_MIXER, 0, 1, 0), + SOC_DAPM_SINGLE("LMIX Switch", NAU8822_REG_AUX2_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("LINMIX Switch", NAU8822_REG_AUX2_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("AUX1MIX Output Switch", + NAU8822_REG_AUX2_MIXER, 3, 1, 0), +}; + +/* Input PGA */ +static const struct snd_kcontrol_new nau8822_left_input_mixer[] = { + SOC_DAPM_SINGLE("L2 Switch", NAU8822_REG_INPUT_CONTROL, 2, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 0, 1, 0), +}; +static const struct snd_kcontrol_new nau8822_right_input_mixer[] = { + SOC_DAPM_SINGLE("R2 Switch", NAU8822_REG_INPUT_CONTROL, 6, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", NAU8822_REG_INPUT_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", NAU8822_REG_INPUT_CONTROL, 4, 1, 0), +}; + +/* Loopback Switch */ +static const struct snd_kcontrol_new nau8822_loopback = + SOC_DAPM_SINGLE("Switch", NAU8822_REG_COMPANDING_CONTROL, + NAU8822_ADDAP_SFT, 1, 0); + +static int check_mclk_select_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(source->dapm); + unsigned int value; + + value = snd_soc_component_read32(component, NAU8822_REG_CLOCKING); + + return (value & NAU8822_CLKM_MASK); +} + +static const struct snd_soc_dapm_widget nau8822_dapm_widgets[] = { + SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + NAU8822_REG_POWER_MANAGEMENT_3, 0, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + NAU8822_REG_POWER_MANAGEMENT_3, 1, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", + NAU8822_REG_POWER_MANAGEMENT_2, 0, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", + NAU8822_REG_POWER_MANAGEMENT_2, 1, 0), + + SOC_MIXER_ARRAY("Left Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_3, 2, 0, nau8822_left_out_mixer), + SOC_MIXER_ARRAY("Right Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_3, 3, 0, nau8822_right_out_mixer), + SOC_MIXER_ARRAY("AUX1 Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_1, 7, 0, nau8822_auxout1_mixer), + SOC_MIXER_ARRAY("AUX2 Output Mixer", + NAU8822_REG_POWER_MANAGEMENT_1, 6, 0, nau8822_auxout2_mixer), + + SOC_MIXER_ARRAY("Left Input Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, + 2, 0, nau8822_left_input_mixer), + SOC_MIXER_ARRAY("Right Input Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, + 3, 0, nau8822_right_input_mixer), + + SND_SOC_DAPM_PGA("Left Boost Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Boost Mixer", + NAU8822_REG_POWER_MANAGEMENT_2, 5, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Capture PGA", + NAU8822_REG_LEFT_INP_PGA_CONTROL, 6, 1, NULL, 0), + SND_SOC_DAPM_PGA("Right Capture PGA", + NAU8822_REG_RIGHT_INP_PGA_CONTROL, 6, 1, NULL, 0), + + SND_SOC_DAPM_PGA("Left Headphone Out", + NAU8822_REG_POWER_MANAGEMENT_2, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Headphone Out", + NAU8822_REG_POWER_MANAGEMENT_2, 8, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Speaker Out", + NAU8822_REG_POWER_MANAGEMENT_3, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Speaker Out", + NAU8822_REG_POWER_MANAGEMENT_3, 5, 0, NULL, 0), + + SND_SOC_DAPM_PGA("AUX1 Out", + NAU8822_REG_POWER_MANAGEMENT_3, 8, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX2 Out", + NAU8822_REG_POWER_MANAGEMENT_3, 7, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Mic Bias", + NAU8822_REG_POWER_MANAGEMENT_1, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", + NAU8822_REG_POWER_MANAGEMENT_1, 5, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("Digital Loopback", SND_SOC_NOPM, 0, 0, + &nau8822_loopback), + + SND_SOC_DAPM_INPUT("LMICN"), + SND_SOC_DAPM_INPUT("LMICP"), + SND_SOC_DAPM_INPUT("RMICN"), + SND_SOC_DAPM_INPUT("RMICP"), + SND_SOC_DAPM_INPUT("LAUX"), + SND_SOC_DAPM_INPUT("RAUX"), + SND_SOC_DAPM_INPUT("L2"), + SND_SOC_DAPM_INPUT("R2"), + SND_SOC_DAPM_OUTPUT("LHP"), + SND_SOC_DAPM_OUTPUT("RHP"), + SND_SOC_DAPM_OUTPUT("LSPK"), + SND_SOC_DAPM_OUTPUT("RSPK"), + SND_SOC_DAPM_OUTPUT("AUXOUT1"), + SND_SOC_DAPM_OUTPUT("AUXOUT2"), +}; + +static const struct snd_soc_dapm_route nau8822_dapm_routes[] = { + {"Right DAC", NULL, "PLL", check_mclk_select_pll}, + {"Left DAC", NULL, "PLL", check_mclk_select_pll}, + + /* LMAIN and RMAIN Mixer */ + {"Right Output Mixer", "LDAC Switch", "Left DAC"}, + {"Right Output Mixer", "RDAC Switch", "Right DAC"}, + {"Right Output Mixer", "RAUX Switch", "RAUX"}, + {"Right Output Mixer", "RINMIX Switch", "Right Boost Mixer"}, + + {"Left Output Mixer", "LDAC Switch", "Left DAC"}, + {"Left Output Mixer", "RDAC Switch", "Right DAC"}, + {"Left Output Mixer", "LAUX Switch", "LAUX"}, + {"Left Output Mixer", "LINMIX Switch", "Left Boost Mixer"}, + + /* AUX1 and AUX2 Mixer */ + {"AUX1 Output Mixer", "RDAC Switch", "Right DAC"}, + {"AUX1 Output Mixer", "RMIX Switch", "Right Output Mixer"}, + {"AUX1 Output Mixer", "RINMIX Switch", "Right Boost Mixer"}, + {"AUX1 Output Mixer", "LDAC Switch", "Left DAC"}, + {"AUX1 Output Mixer", "LMIX Switch", "Left Output Mixer"}, + + {"AUX2 Output Mixer", "LDAC Switch", "Left DAC"}, + {"AUX2 Output Mixer", "LMIX Switch", "Left Output Mixer"}, + {"AUX2 Output Mixer", "LINMIX Switch", "Left Boost Mixer"}, + {"AUX2 Output Mixer", "AUX1MIX Output Switch", "AUX1 Output Mixer"}, + + /* Outputs */ + {"Right Headphone Out", NULL, "Right Output Mixer"}, + {"RHP", NULL, "Right Headphone Out"}, + + {"Left Headphone Out", NULL, "Left Output Mixer"}, + {"LHP", NULL, "Left Headphone Out"}, + + {"Right Speaker Out", NULL, "Right Output Mixer"}, + {"RSPK", NULL, "Right Speaker Out"}, + + {"Left Speaker Out", NULL, "Left Output Mixer"}, + {"LSPK", NULL, "Left Speaker Out"}, + + {"AUX1 Out", NULL, "AUX1 Output Mixer"}, + {"AUX2 Out", NULL, "AUX2 Output Mixer"}, + {"AUXOUT1", NULL, "AUX1 Out"}, + {"AUXOUT2", NULL, "AUX2 Out"}, + + /* Boost Mixer */ + {"Right ADC", NULL, "PLL", check_mclk_select_pll}, + {"Left ADC", NULL, "PLL", check_mclk_select_pll}, + + {"Right ADC", NULL, "Right Boost Mixer"}, + + {"Right Boost Mixer", NULL, "RAUX"}, + {"Right Boost Mixer", NULL, "Right Capture PGA"}, + {"Right Boost Mixer", NULL, "R2"}, + + {"Left ADC", NULL, "Left Boost Mixer"}, + + {"Left Boost Mixer", NULL, "LAUX"}, + {"Left Boost Mixer", NULL, "Left Capture PGA"}, + {"Left Boost Mixer", NULL, "L2"}, + + /* Input PGA */ + {"Right Capture PGA", NULL, "Right Input Mixer"}, + {"Left Capture PGA", NULL, "Left Input Mixer"}, + + /* Enable Microphone Power */ + {"Right Capture PGA", NULL, "Mic Bias"}, + {"Left Capture PGA", NULL, "Mic Bias"}, + + {"Right Input Mixer", "R2 Switch", "R2"}, + {"Right Input Mixer", "MicN Switch", "RMICN"}, + {"Right Input Mixer", "MicP Switch", "RMICP"}, + + {"Left Input Mixer", "L2 Switch", "L2"}, + {"Left Input Mixer", "MicN Switch", "LMICN"}, + {"Left Input Mixer", "MicP Switch", "LMICP"}, + + /* Digital Loopback */ + {"Digital Loopback", "Switch", "Left ADC"}, + {"Digital Loopback", "Switch", "Right ADC"}, + {"Left DAC", NULL, "Digital Loopback"}, + {"Right DAC", NULL, "Digital Loopback"}, +}; + +static int nau8822_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + + nau8822->div_id = clk_id; + nau8822->sysclk = freq; + dev_dbg(component->dev, "master sysclk %dHz, source %s\n", freq, + clk_id == NAU8822_CLK_PLL ? "PLL" : "MCLK"); + + return 0; +} + +static int nau8822_calc_pll(unsigned int pll_in, unsigned int fs, + struct nau8822_pll *pll_param) +{ + u64 f2, f2_max, pll_ratio; + int i, scal_sel; + + if (pll_in > NAU_PLL_REF_MAX || pll_in < NAU_PLL_REF_MIN) + return -EINVAL; + f2_max = 0; + scal_sel = ARRAY_SIZE(nau8822_mclk_scaler); + + for (i = 0; i < scal_sel; i++) { + f2 = 256 * fs * 4 * nau8822_mclk_scaler[i] / 10; + if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX && + f2_max < f2) { + f2_max = f2; + scal_sel = i; + } + } + + if (ARRAY_SIZE(nau8822_mclk_scaler) == scal_sel) + return -EINVAL; + pll_param->mclk_scaler = scal_sel; + f2 = f2_max; + + /* Calculate the PLL 4-bit integer input and the PLL 24-bit fractional + * input; round up the 24+4bit. + */ + pll_ratio = div_u64(f2 << 28, pll_in); + pll_param->pre_factor = 0; + if (((pll_ratio >> 28) & 0xF) < NAU_PLL_OPTOP_MIN) { + pll_ratio <<= 1; + pll_param->pre_factor = 1; + } + pll_param->pll_int = (pll_ratio >> 28) & 0xF; + pll_param->pll_frac = ((pll_ratio & 0xFFFFFFF) >> 4); + + return 0; +} + +static int nau8822_config_clkdiv(struct snd_soc_dai *dai, int div, int rate) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + struct nau8822_pll *pll = &nau8822->pll; + int i, sclk, imclk; + + switch (nau8822->div_id) { + case NAU8822_CLK_MCLK: + /* Configure the master clock prescaler div to make system + * clock to approximate the internal master clock (IMCLK); + * and large or equal to IMCLK. + */ + div = 0; + imclk = rate * 256; + for (i = 1; i < ARRAY_SIZE(nau8822_mclk_scaler); i++) { + sclk = (nau8822->sysclk * 10) / nau8822_mclk_scaler[i]; + if (sclk < imclk) + break; + div = i; + } + dev_dbg(component->dev, "master clock prescaler %x for fs %d\n", + div, rate); + + /* master clock from MCLK and disable PLL */ + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK, + (div << NAU8822_MCLKSEL_SFT)); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, + NAU8822_CLKM_MCLK); + break; + + case NAU8822_CLK_PLL: + /* master clock from PLL and enable PLL */ + if (pll->mclk_scaler != div) { + dev_err(component->dev, + "master clock prescaler not meet PLL parameters\n"); + return -EINVAL; + } + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK, + (div << NAU8822_MCLKSEL_SFT)); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, + NAU8822_CLKM_PLL); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + struct nau8822_pll *pll_param = &nau8822->pll; + int ret, fs; + + fs = freq_out / 256; + + ret = nau8822_calc_pll(freq_in, fs, pll_param); + if (ret < 0) { + dev_err(component->dev, "Unsupported input clock %d\n", + freq_in); + return ret; + } + + dev_info(component->dev, + "pll_int=%x pll_frac=%x mclk_scaler=%x pre_factor=%x\n", + pll_param->pll_int, pll_param->pll_frac, + pll_param->mclk_scaler, pll_param->pre_factor); + + snd_soc_component_update_bits(component, + NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, + (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | + pll_param->pll_int); + snd_soc_component_write(component, + NAU8822_REG_PLL_K1, (pll_param->pll_frac >> NAU8822_PLLK1_SFT) & + NAU8822_PLLK1_MASK); + snd_soc_component_write(component, + NAU8822_REG_PLL_K2, (pll_param->pll_frac >> NAU8822_PLLK2_SFT) & + NAU8822_PLLK2_MASK); + snd_soc_component_write(component, + NAU8822_REG_PLL_K3, pll_param->pll_frac & NAU8822_PLLK3_MASK); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_MCLKSEL_MASK, + pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + + return 0; +} + +static int nau8822_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + u16 ctrl1_val = 0, ctrl2_val = 0; + + dev_dbg(component->dev, "%s\n", __func__); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl2_val &= ~1; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= 0x10; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= 0x8; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= 0x18; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl1_val |= 0x180; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= 0x100; + break; + case SND_SOC_DAIFMT_NB_IF: + ctrl1_val |= 0x80; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, + NAU8822_REG_AUDIO_INTERFACE, + NAU8822_AIFMT_MASK | NAU8822_LRP_MASK | NAU8822_BCLKP_MASK, + ctrl1_val); + snd_soc_component_update_bits(component, + NAU8822_REG_CLOCKING, NAU8822_CLKIOEN_MASK, ctrl2_val); + + return 0; +} + +static int nau8822_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + int val_len = 0, val_rate = 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len |= NAU8822_WLEN_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= NAU8822_WLEN_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + val_len |= NAU8822_WLEN_32; + break; + default: + return -EINVAL; + } + + switch (params_rate(params)) { + case 8000: + val_rate |= NAU8822_SMPLR_8K; + break; + case 11025: + val_rate |= NAU8822_SMPLR_12K; + break; + case 16000: + val_rate |= NAU8822_SMPLR_16K; + break; + case 22050: + val_rate |= NAU8822_SMPLR_24K; + break; + case 32000: + val_rate |= NAU8822_SMPLR_32K; + break; + case 44100: + case 48000: + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, + NAU8822_REG_AUDIO_INTERFACE, NAU8822_WLEN_MASK, val_len); + snd_soc_component_update_bits(component, + NAU8822_REG_ADDITIONAL_CONTROL, NAU8822_SMPLR_MASK, val_rate); + + /* If the master clock is from MCLK, provide the runtime FS for driver + * to get the master clock prescaler configuration. + */ + if (nau8822->div_id == NAU8822_CLK_MCLK) + nau8822_config_clkdiv(dai, 0, params_rate(params)); + + return 0; +} + +static int nau8822_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + + dev_dbg(component->dev, "%s: %d\n", __func__, mute); + + if (mute) + snd_soc_component_update_bits(component, + NAU8822_REG_DAC_CONTROL, 0x40, 0x40); + else + snd_soc_component_update_bits(component, + NAU8822_REG_DAC_CONTROL, 0x40, 0); + + return 0; +} + +static int nau8822_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_REFIMP_MASK, NAU8822_REFIMP_80K); + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_IOBUF_EN | NAU8822_ABIAS_EN, + NAU8822_IOBUF_EN | NAU8822_ABIAS_EN); + + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_REFIMP_MASK, NAU8822_REFIMP_3K); + mdelay(100); + } + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, + NAU8822_REFIMP_MASK, NAU8822_REFIMP_300K); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_component_write(component, + NAU8822_REG_POWER_MANAGEMENT_1, 0); + snd_soc_component_write(component, + NAU8822_REG_POWER_MANAGEMENT_2, 0); + snd_soc_component_write(component, + NAU8822_REG_POWER_MANAGEMENT_3, 0); + break; + } + + dev_dbg(component->dev, "%s: %d\n", __func__, level); + + return 0; +} + +#define NAU8822_RATES (SNDRV_PCM_RATE_8000_48000) + +#define NAU8822_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops nau8822_dai_ops = { + .hw_params = nau8822_hw_params, + .digital_mute = nau8822_mute, + .set_fmt = nau8822_set_dai_fmt, + .set_sysclk = nau8822_set_dai_sysclk, + .set_pll = nau8822_set_pll, +}; + +static struct snd_soc_dai_driver nau8822_dai = { + .name = "nau8822-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8822_RATES, + .formats = NAU8822_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8822_RATES, + .formats = NAU8822_FORMATS, + }, + .ops = &nau8822_dai_ops, + .symmetric_rates = 1, +}; + +static int nau8822_suspend(struct snd_soc_component *component) +{ + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + + snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); + + regcache_mark_dirty(nau8822->regmap); + + return 0; +} + +static int nau8822_resume(struct snd_soc_component *component) +{ + struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); + + regcache_sync(nau8822->regmap); + + snd_soc_component_force_bias_level(component, SND_SOC_BIAS_STANDBY); + + return 0; +} + +/* + * These registers contain an "update" bit - bit 8. This means, for example, + * that one can write new DAC digital volume for both channels, but only when + * the update bit is set, will also the volume be updated - simultaneously for + * both channels. + */ +static const int update_reg[] = { + NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME, + NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME, + NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME, + NAU8822_REG_LEFT_INP_PGA_CONTROL, + NAU8822_REG_RIGHT_INP_PGA_CONTROL, + NAU8822_REG_LHP_VOLUME, + NAU8822_REG_RHP_VOLUME, + NAU8822_REG_LSPKOUT_VOLUME, + NAU8822_REG_RSPKOUT_VOLUME, +}; + +static int nau8822_probe(struct snd_soc_component *component) +{ + int i; + + /* + * Set the update bit in all registers, that have one. This way all + * writes to those registers will also cause the update bit to be + * written. + */ + for (i = 0; i < ARRAY_SIZE(update_reg); i++) + snd_soc_component_update_bits(component, + update_reg[i], 0x100, 0x100); + + return 0; +} + +static const struct snd_soc_component_driver soc_component_dev_nau8822 = { + .probe = nau8822_probe, + .suspend = nau8822_suspend, + .resume = nau8822_resume, + .set_bias_level = nau8822_set_bias_level, + .controls = nau8822_snd_controls, + .num_controls = ARRAY_SIZE(nau8822_snd_controls), + .dapm_widgets = nau8822_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8822_dapm_widgets), + .dapm_routes = nau8822_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8822_dapm_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config nau8822_regmap_config = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = NAU8822_REG_MAX_REGISTER, + .volatile_reg = nau8822_volatile, + + .readable_reg = nau8822_readable_reg, + .writeable_reg = nau8822_writeable_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8822_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8822_reg_defaults), +}; + +static int nau8822_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8822 *nau8822 = dev_get_platdata(dev); + int ret; + + if (!nau8822) { + nau8822 = devm_kzalloc(dev, sizeof(*nau8822), GFP_KERNEL); + if (nau8822 == NULL) + return -ENOMEM; + } + i2c_set_clientdata(i2c, nau8822); + + nau8822->regmap = devm_regmap_init_i2c(i2c, &nau8822_regmap_config); + if (IS_ERR(nau8822->regmap)) { + ret = PTR_ERR(nau8822->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + nau8822->dev = dev; + + /* Reset the codec */ + ret = regmap_write(nau8822->regmap, NAU8822_REG_RESET, 0x00); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret); + return ret; + } + + ret = devm_snd_soc_register_component(dev, &soc_component_dev_nau8822, + &nau8822_dai, 1); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct i2c_device_id nau8822_i2c_id[] = { + { "nau8822", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8822_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id nau8822_of_match[] = { + { .compatible = "nuvoton,nau8822", }, + { } +}; +MODULE_DEVICE_TABLE(of, nau8822_of_match); +#endif + +static struct i2c_driver nau8822_i2c_driver = { + .driver = { + .name = "nau8822", + .of_match_table = of_match_ptr(nau8822_of_match), + }, + .probe = nau8822_i2c_probe, + .id_table = nau8822_i2c_id, +}; +module_i2c_driver(nau8822_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8822 codec driver"); +MODULE_AUTHOR("David Lin <ctlin0(a)nuvoton.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h new file mode 100644 index 000000000000..aa79c969cd44 --- /dev/null +++ b/sound/soc/codecs/nau8822.h @@ -0,0 +1,204 @@ +/* + * nau8822.h -- NAU8822 Soc Audio Codec driver + * + * Author: David Lin <ctlin0(a)nuvoton.com> + * Co-author: John Hsu <kchsu0(a)nuvoton.com> + * Co-author: Seven Li <wtli(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __NAU8822_H__ +#define __NAU8822_H__ + +#define NAU8822_REG_RESET 0x00 +#define NAU8822_REG_POWER_MANAGEMENT_1 0x01 +#define NAU8822_REG_POWER_MANAGEMENT_2 0x02 +#define NAU8822_REG_POWER_MANAGEMENT_3 0x03 +#define NAU8822_REG_AUDIO_INTERFACE 0x04 +#define NAU8822_REG_COMPANDING_CONTROL 0x05 +#define NAU8822_REG_CLOCKING 0x06 +#define NAU8822_REG_ADDITIONAL_CONTROL 0x07 +#define NAU8822_REG_GPIO_CONTROL 0x08 +#define NAU8822_REG_JACK_DETECT_CONTROL_1 0x09 +#define NAU8822_REG_DAC_CONTROL 0x0A +#define NAU8822_REG_LEFT_DAC_DIGITAL_VOLUME 0x0B +#define NAU8822_REG_RIGHT_DAC_DIGITAL_VOLUME 0x0C +#define NAU8822_REG_JACK_DETECT_CONTROL_2 0x0D +#define NAU8822_REG_ADC_CONTROL 0x0E +#define NAU8822_REG_LEFT_ADC_DIGITAL_VOLUME 0x0F +#define NAU8822_REG_RIGHT_ADC_DIGITAL_VOLUME 0x10 +#define NAU8822_REG_EQ1 0x12 +#define NAU8822_REG_EQ2 0x13 +#define NAU8822_REG_EQ3 0x14 +#define NAU8822_REG_EQ4 0x15 +#define NAU8822_REG_EQ5 0x16 +#define NAU8822_REG_DAC_LIMITER_1 0x18 +#define NAU8822_REG_DAC_LIMITER_2 0x19 +#define NAU8822_REG_NOTCH_FILTER_1 0x1B +#define NAU8822_REG_NOTCH_FILTER_2 0x1C +#define NAU8822_REG_NOTCH_FILTER_3 0x1D +#define NAU8822_REG_NOTCH_FILTER_4 0x1E +#define NAU8822_REG_ALC_CONTROL_1 0x20 +#define NAU8822_REG_ALC_CONTROL_2 0x21 +#define NAU8822_REG_ALC_CONTROL_3 0x22 +#define NAU8822_REG_NOISE_GATE 0x23 +#define NAU8822_REG_PLL_N 0x24 +#define NAU8822_REG_PLL_K1 0x25 +#define NAU8822_REG_PLL_K2 0x26 +#define NAU8822_REG_PLL_K3 0x27 +#define NAU8822_REG_3D_CONTROL 0x29 +#define NAU8822_REG_RIGHT_SPEAKER_CONTROL 0x2B +#define NAU8822_REG_INPUT_CONTROL 0x2C +#define NAU8822_REG_LEFT_INP_PGA_CONTROL 0x2D +#define NAU8822_REG_RIGHT_INP_PGA_CONTROL 0x2E +#define NAU8822_REG_LEFT_ADC_BOOST_CONTROL 0x2F +#define NAU8822_REG_RIGHT_ADC_BOOST_CONTROL 0x30 +#define NAU8822_REG_OUTPUT_CONTROL 0x31 +#define NAU8822_REG_LEFT_MIXER_CONTROL 0x32 +#define NAU8822_REG_RIGHT_MIXER_CONTROL 0x33 +#define NAU8822_REG_LHP_VOLUME 0x34 +#define NAU8822_REG_RHP_VOLUME 0x35 +#define NAU8822_REG_LSPKOUT_VOLUME 0x36 +#define NAU8822_REG_RSPKOUT_VOLUME 0x37 +#define NAU8822_REG_AUX2_MIXER 0x38 +#define NAU8822_REG_AUX1_MIXER 0x39 +#define NAU8822_REG_POWER_MANAGEMENT_4 0x3A +#define NAU8822_REG_LEFT_TIME_SLOT 0x3B +#define NAU8822_REG_MISC 0x3C +#define NAU8822_REG_RIGHT_TIME_SLOT 0x3D +#define NAU8822_REG_DEVICE_REVISION 0x3E +#define NAU8822_REG_DEVICE_ID 0x3F +#define NAU8822_REG_DAC_DITHER 0x41 +#define NAU8822_REG_ALC_ENHANCE_1 0x46 +#define NAU8822_REG_ALC_ENHANCE_2 0x47 +#define NAU8822_REG_192KHZ_SAMPLING 0x48 +#define NAU8822_REG_MISC_CONTROL 0x49 +#define NAU8822_REG_INPUT_TIEOFF 0x4A +#define NAU8822_REG_POWER_REDUCTION 0x4B +#define NAU8822_REG_AGC_PEAK2PEAK 0x4C +#define NAU8822_REG_AGC_PEAK_DETECT 0x4D +#define NAU8822_REG_AUTOMUTE_CONTROL 0x4E +#define NAU8822_REG_OUTPUT_TIEOFF 0x4F +#define NAU8822_REG_MAX_REGISTER NAU8822_REG_OUTPUT_TIEOFF + +/* NAU8822_REG_POWER_MANAGEMENT_1 (0x1) */ +#define NAU8822_REFIMP_MASK 0x3 +#define NAU8822_REFIMP_80K 0x1 +#define NAU8822_REFIMP_300K 0x2 +#define NAU8822_REFIMP_3K 0x3 +#define NAU8822_IOBUF_EN (0x1 << 2) +#define NAU8822_ABIAS_EN (0x1 << 3) + +/* NAU8822_REG_AUDIO_INTERFACE (0x4) */ +#define NAU8822_AIFMT_MASK (0x3 << 3) +#define NAU8822_WLEN_MASK (0x3 << 5) +#define NAU8822_WLEN_20 (0x1 << 5) +#define NAU8822_WLEN_24 (0x2 << 5) +#define NAU8822_WLEN_32 (0x3 << 5) +#define NAU8822_LRP_MASK (0x1 << 7) +#define NAU8822_BCLKP_MASK (0x1 << 8) + +/* NAU8822_REG_COMPANDING_CONTROL (0x5) */ +#define NAU8822_ADDAP_SFT 0 +#define NAU8822_ADCCM_SFT 1 +#define NAU8822_DACCM_SFT 3 + +/* NAU8822_REG_CLOCKING (0x6) */ +#define NAU8822_CLKIOEN_MASK 0x1 +#define NAU8822_MCLKSEL_SFT 5 +#define NAU8822_MCLKSEL_MASK (0x7 << 5) +#define NAU8822_BCLKSEL_SFT 2 +#define NAU8822_BCLKSEL_MASK (0x7 << 2) +#define NAU8822_CLKM_MASK (0x1 << 8) +#define NAU8822_CLKM_MCLK (0x0 << 8) +#define NAU8822_CLKM_PLL (0x1 << 8) + +/* NAU8822_REG_ADDITIONAL_CONTROL (0x08) */ +#define NAU8822_SMPLR_SFT 1 +#define NAU8822_SMPLR_MASK (0x7 << 1) +#define NAU8822_SMPLR_48K (0x0 << 1) +#define NAU8822_SMPLR_32K (0x1 << 1) +#define NAU8822_SMPLR_24K (0x2 << 1) +#define NAU8822_SMPLR_16K (0x3 << 1) +#define NAU8822_SMPLR_12K (0x4 << 1) +#define NAU8822_SMPLR_8K (0x5 << 1) + +/* NAU8822_REG_EQ1 (0x12) */ +#define NAU8822_EQ1GC_SFT 0 +#define NAU8822_EQ1CF_SFT 5 +#define NAU8822_EQM_SFT 8 + +/* NAU8822_REG_EQ2 (0x13) */ +#define NAU8822_EQ2GC_SFT 0 +#define NAU8822_EQ2CF_SFT 5 +#define NAU8822_EQ2BW_SFT 8 + +/* NAU8822_REG_EQ3 (0x14) */ +#define NAU8822_EQ3GC_SFT 0 +#define NAU8822_EQ3CF_SFT 5 +#define NAU8822_EQ3BW_SFT 8 + +/* NAU8822_REG_EQ4 (0x15) */ +#define NAU8822_EQ4GC_SFT 0 +#define NAU8822_EQ4CF_SFT 5 +#define NAU8822_EQ4BW_SFT 8 + +/* NAU8822_REG_EQ5 (0x16) */ +#define NAU8822_EQ5GC_SFT 0 +#define NAU8822_EQ5CF_SFT 5 + +/* NAU8822_REG_ALC_CONTROL_1 (0x20) */ +#define NAU8822_ALCMINGAIN_SFT 0 +#define NAU8822_ALCMXGAIN_SFT 3 +#define NAU8822_ALCEN_SFT 7 + +/* NAU8822_REG_ALC_CONTROL_2 (0x21) */ +#define NAU8822_ALCSL_SFT 0 +#define NAU8822_ALCHT_SFT 4 + +/* NAU8822_REG_ALC_CONTROL_3 (0x22) */ +#define NAU8822_ALCATK_SFT 0 +#define NAU8822_ALCDCY_SFT 4 +#define NAU8822_ALCM_SFT 8 + +/* NAU8822_REG_PLL_N (0x24) */ +#define NAU8822_PLLMCLK_DIV2 (0x1 << 4) +#define NAU8822_PLLN_MASK 0xF + +#define NAU8822_PLLK1_SFT 18 +#define NAU8822_PLLK1_MASK 0x3F + +/* NAU8822_REG_PLL_K2 (0x26) */ +#define NAU8822_PLLK2_SFT 9 +#define NAU8822_PLLK2_MASK 0x1FF + +/* NAU8822_REG_PLL_K3 (0x27) */ +#define NAU8822_PLLK3_MASK 0x1FF + +/* System Clock Source */ +enum { + NAU8822_CLK_MCLK, + NAU8822_CLK_PLL, +}; + +struct nau8822_pll { + int pre_factor; + int mclk_scaler; + int pll_frac; + int pll_int; +}; + +/* Codec Private Data */ +struct nau8822 { + struct device *dev; + struct regmap *regmap; + int mclk_idx; + struct nau8822_pll pll; + int sysclk; + int div_id; +}; + +#endif /* __NAU8822_H__ */ -- 2.18.0

3 5

[alsa-devel] Applied "ASoC: sta32x: Add support for XTI clock" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: sta32x: Add support for XTI clock has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From fce9ec954a8af7e04cbf5b9daa8bec9c1df5cfe6 Mon Sep 17 00:00:00 2001 From: Daniel Mack <daniel(a)zonque.org> Date: Wed, 17 Oct 2018 13:37:03 +0200 Subject: [PATCH] ASoC: sta32x: Add support for XTI clock The STA32x chips feature an XTI clock input that needs to be stable before the reset signal is released. Therefore, the chip driver needs to get a handle to the clock. Instead of relying on other parts of the system to enable the clock, let the codec driver grab a handle itself. In order to keep existing boards working, clock support is made optional. Signed-off-by: Daniel Mack <daniel(a)zonque.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/st,sta32x.txt | 6 ++++ sound/soc/codecs/sta32x.c | 28 +++++++++++++++++++ 2 files changed, 34 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/st,sta32x.txt b/Documentation/devicetree/bindings/sound/st,sta32x.txt index ff4a685a4303..52265fb757c5 100644 --- a/Documentation/devicetree/bindings/sound/st,sta32x.txt +++ b/Documentation/devicetree/bindings/sound/st,sta32x.txt @@ -19,6 +19,10 @@ Required properties: Optional properties: + - clocks, clock-names: Clock specifier for XTI input clock. + If specified, the clock will be enabled when the codec is probed, + and disabled when it is removed. The 'clock-names' must be set to 'xti'. + - st,output-conf: number, Selects the output configuration: 0: 2-channel (full-bridge) power, 2-channel data-out 1: 2 (half-bridge). 1 (full-bridge) on-board power @@ -79,6 +83,8 @@ Example: codec: sta32x@38 { compatible = "st,sta32x"; reg = <0x1c>; + clocks = <&clock>; + clock-names = "xti"; reset-gpios = <&gpio1 19 0>; power-down-gpios = <&gpio1 16 0>; st,output-conf = /bits/ 8 <0x3>; // set output to 2-channel diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 22de1593443c..f753d2db0a5a 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -21,6 +21,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> +#include <linux/clk.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> @@ -142,6 +143,7 @@ static const char *sta32x_supply_names[] = { /* codec private data */ struct sta32x_priv { struct regmap *regmap; + struct clk *xti_clk; struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)]; struct snd_soc_component *component; struct sta32x_platform_data *pdata; @@ -879,6 +881,18 @@ static int sta32x_probe(struct snd_soc_component *component) struct sta32x_priv *sta32x = snd_soc_component_get_drvdata(component); struct sta32x_platform_data *pdata = sta32x->pdata; int i, ret = 0, thermal = 0; + + sta32x->component = component; + + if (sta32x->xti_clk) { + ret = clk_prepare_enable(sta32x->xti_clk); + if (ret != 0) { + dev_err(component->dev, + "Failed to enable clock: %d\n", ret); + return ret; + } + } + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); if (ret != 0) { @@ -981,6 +995,9 @@ static void sta32x_remove(struct snd_soc_component *component) sta32x_watchdog_stop(sta32x); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + if (sta32x->xti_clk) + clk_disable_unprepare(sta32x->xti_clk); } static const struct snd_soc_component_driver sta32x_component = { @@ -1097,6 +1114,17 @@ static int sta32x_i2c_probe(struct i2c_client *i2c, } #endif + /* Clock */ + sta32x->xti_clk = devm_clk_get(dev, "xti"); + if (IS_ERR(sta32x->xti_clk)) { + ret = PTR_ERR(sta32x->xti_clk); + + if (ret == -EPROBE_DEFER) + return ret; + + sta32x->xti_clk = NULL; + } + /* GPIOs */ sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_LOW); -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH] ASoC: pcm3168a: add hw constraint for capture channel
by Kuninori Morimoto 19 Oct '18

19 Oct '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> LEFT_J / I2S only can use TDM. commit 594680ea4a394 ("ASoC: pcm3168a: add hw constraint for channel") commit 3809688980205 ("ASoC: pcm3168a: add HW constraint for non RIGHT_J") added channel constraint for it, but, it was only for playback. This patch adds constraint for capture. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/pcm3168a.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 63aa025..52cc950 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -529,11 +529,17 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, break; case PCM3168A_FMT_LEFT_J: sample_min = 24; - channel_max = 8; + if (tx) + channel_max = 8; + else + channel_max = 6; break; case PCM3168A_FMT_I2S: sample_min = 24; - channel_max = 8; + if (tx) + channel_max = 8; + else + channel_max = 6; break; default: sample_min = 24; @@ -559,6 +565,7 @@ static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = { }; static const struct snd_soc_dai_ops pcm3168a_adc_dai_ops = { + .startup = pcm3168a_startup, .set_fmt = pcm3168a_set_dai_fmt_adc, .set_sysclk = pcm3168a_set_dai_sysclk, .hw_params = pcm3168a_hw_params -- 2.7.4

2 2

[alsa-devel] Applied "ASoC: rsnd: enable TDM settings for SSI parent" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: rsnd: enable TDM settings for SSI parent has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 6817d7593f3e3c8a9c11e7a07cb5646c70371f0a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 17 Oct 2018 01:55:37 +0000 Subject: [PATCH] ASoC: rsnd: enable TDM settings for SSI parent Some SSIs are sharing each pins (= WS/CLK pin for playback/capture). Then, SSI parent needs control WS/CLK setting for SSI slave. In such case, SSI parent needs TDM settings if SSI slave is working as TDM mode. But it is not cared in current driver. It can't capture TDM sound without this patch if SSIs were pin sharing. This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 with TDM sound. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/ssi.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 3adcc4f778f7..b42a0e0feab7 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -399,6 +399,17 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, if (rdai->sys_delay) cr_own |= DEL; + /* + * TDM Mode + * see + * rsnd_ssiu_init_gen2() + */ + wsr = ssi->wsr; + if (is_tdm) { + wsr |= WS_MODE; + cr_own |= CHNL_8; + } + /* * We shouldn't exchange SWSP after running. * This means, parent needs to care it. @@ -429,16 +440,6 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_mode = DIEN; /* PIO : enable Data interrupt */ } - /* - * TDM Extend Mode - * see - * rsnd_ssiu_init_gen2() - */ - wsr = ssi->wsr; - if (is_tdm) { - wsr |= WS_MODE; - cr_own |= CHNL_8; - } init_end: ssi->cr_own = cr_own; ssi->cr_mode = cr_mode; -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH] ASoC: rsnd: tidyup SSICR::SWSP for TDM
by Kuninori Morimoto 19 Oct '18

19 Oct '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> R-Car datasheet is indicating that WS output settings of SSICR::SWSP is inverted on TDM mode from non TDM mode settings. But, it is meaning that TDM should use 0 here. Without this patch, sound input/output 1ch will be 2ch, 2ch will be 3ch ..., be jumbled on I2S + TDM settings. This patch fixup it. This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 TDM sound. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index b42a0e0..fcb4df2 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -392,7 +392,7 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, if (rdai->bit_clk_inv) cr_own |= SCKP; - if (rdai->frm_clk_inv ^ is_tdm) + if (rdai->frm_clk_inv && !is_tdm) cr_own |= SWSP; if (rdai->data_alignment) cr_own |= SDTA; -- 2.7.4

2 2

[alsa-devel] [PATCH] ASoC: Intel: kbl_da7219_max98927: minor white space clean up
by Dan Carpenter 19 Oct '18

19 Oct '18

I just added a couple missing tabs. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 3ab96ee7bd3c..3fa1c3ca6d37 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -180,14 +180,14 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); if (ret < 0) { dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret); - return ret; + return ret; } } if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16); if (ret < 0) { dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret); - return ret; + return ret; } } }

2 2

[alsa-devel] [PATCH 0/4] ASoC: stm32: sai: add mclk clock provider
by Olivier Moysan 19 Oct '18

19 Oct '18

This patch set allows to use STM32 SAI interface as a master clock provider. The Cirrus cs42l51 codec patch gives the example of the source code required on codec side, to use SAI mclk provider. These patches allow to cover the following use cases: - SAI is master or slave and is master clock provider - SAI is master or slave and is not master clock provider (another sai provides the mclk clock) Implementation overview: - mclk configuration SAI is registered in clock framework: - either as a mclk provider (optional in DT). SAI generates mclk when requested by DAPM - or as a mclk consumer (optional in DT). cs42l51 codec is configured as a mclk consumer (optional in DT) - mclk power A DAPM is registered in codec to handle mclk enable/disable (DAPM route is set in soundcard node) - mclk rate: mclk rate is computed at runtime from stream rate and mclk-fs ratio provided in DT mclk rate is propagated through sysclk callback to the SAI which is mclk consumer. The mclk rate is set exclusively to avoid concurrent rate requests on SAI master. Regards Olivier Olivier Moysan (4): ASoC: dt-bindings: add mclk provider support to stm32 sai ASoC: dt-bindings: add mclk support to cs42l51 ASoC: stm32: sai: set sai as mclk clock provider ASoC: cs42l51: add mclk support .../devicetree/bindings/sound/cs42l51.txt | 17 ++ .../devicetree/bindings/sound/st,stm32-sai.txt | 7 + .../devicetree/bindings/trivial-devices.txt | 1 - sound/soc/codecs/cs42l51.c | 8 + sound/soc/stm/stm32_sai.h | 3 + sound/soc/stm/stm32_sai_sub.c | 275 ++++++++++++++++++--- 6 files changed, 274 insertions(+), 37 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cs42l51.txt -- 2.7.4

2 12

[alsa-devel] Applied "ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 2a2aefa41ce48ace8e1e963cb10c3f5ff43aa994 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald <rf(a)opensource.cirrus.com> Date: Fri, 19 Oct 2018 13:25:15 +0100 Subject: [PATCH] ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer The wm_adsp_buffer struct is the control header of a circular buffer used to transfer data from the firmware over the control interface to an ALSA compressed stream. The original names of the fields pointing to the data buffer were based on ADSP2V2 memory layout where they correspond to {XM, XM, YM}. But this circular buffer could be used on other types of DSP core that have different memory region types. Also the names and description of the size fields were not very clear. The field names and descriptions have been changed to be generic and not imply any particular memory types. This patch updates the wm_adsp driver to the new field names. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f61656070225..7ae10c632614 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -311,12 +311,12 @@ struct wm_adsp_alg_xm_struct { }; struct wm_adsp_buffer { - __be32 X_buf_base; /* XM base addr of first X area */ - __be32 X_buf_size; /* Size of 1st X area in words */ - __be32 X_buf_base2; /* XM base addr of 2nd X area */ - __be32 X_buf_brk; /* Total X size in words */ - __be32 Y_buf_base; /* YM base addr of Y area */ - __be32 wrap; /* Total size X and Y in words */ + __be32 buf1_base; /* Base addr of first buffer area */ + __be32 buf1_size; /* Size of buf1 area in DSP words */ + __be32 buf2_base; /* Base addr of 2nd buffer area */ + __be32 buf1_buf2_size; /* Size of buf1+buf2 in DSP words */ + __be32 buf3_base; /* Base addr of buf3 area */ + __be32 buf_total_size; /* Size of buf1+buf2+buf3 in DSP words */ __be32 high_water_mark; /* Point at which IRQ is asserted */ __be32 irq_count; /* bits 1-31 count IRQ assertions */ __be32 irq_ack; /* acked IRQ count, bit 0 enables IRQ */ @@ -393,18 +393,18 @@ struct wm_adsp_buffer_region_def { static const struct wm_adsp_buffer_region_def default_regions[] = { { .mem_type = WMFW_ADSP2_XM, - .base_offset = HOST_BUFFER_FIELD(X_buf_base), - .size_offset = HOST_BUFFER_FIELD(X_buf_size), + .base_offset = HOST_BUFFER_FIELD(buf1_base), + .size_offset = HOST_BUFFER_FIELD(buf1_size), }, { .mem_type = WMFW_ADSP2_XM, - .base_offset = HOST_BUFFER_FIELD(X_buf_base2), - .size_offset = HOST_BUFFER_FIELD(X_buf_brk), + .base_offset = HOST_BUFFER_FIELD(buf2_base), + .size_offset = HOST_BUFFER_FIELD(buf1_buf2_size), }, { .mem_type = WMFW_ADSP2_YM, - .base_offset = HOST_BUFFER_FIELD(Y_buf_base), - .size_offset = HOST_BUFFER_FIELD(wrap), + .base_offset = HOST_BUFFER_FIELD(buf3_base), + .size_offset = HOST_BUFFER_FIELD(buf_total_size), }, }; -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e3a360b8cdede74d25807fc405e5d8bfb025692f Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald <rf(a)opensource.cirrus.com> Date: Fri, 19 Oct 2018 13:25:16 +0100 Subject: [PATCH] ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate Increase the address value width in the debug log from 4 digits to 8 digits to allow for DSP cores with larger memory address ranges. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 7ae10c632614..a53dc174bbf0 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3345,7 +3345,7 @@ static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf) region->cumulative_size = offset; adsp_dbg(buf->dsp, - "region=%d type=%d base=%04x off=%04x size=%04x\n", + "region=%d type=%d base=%08x off=%08x size=%08x\n", i, region->mem_type, region->base_addr, region->offset, region->cumulative_size); } -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: tegra_sgtl5000: fix device_node refcounting" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: tegra_sgtl5000: fix device_node refcounting has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a85227da2dcc291b762c8482a505bc7d0d2d4b07 Mon Sep 17 00:00:00 2001 From: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Date: Tue, 16 Oct 2018 12:47:29 +0200 Subject: [PATCH] ASoC: tegra_sgtl5000: fix device_node refcounting Similar to the following: commit 4321723648b0 ("ASoC: tegra_alc5632: fix device_node refcounting") commit 7c5dfd549617 ("ASoC: tegra: fix device_node refcounting") Signed-off-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Acked-by: Jon Hunter <jonathanh(a)nvidia.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/tegra/tegra_sgtl5000.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c index 45a4aa9d2a47..901457da25ec 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -149,14 +149,14 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing/invalid\n"); ret = -EINVAL; - goto err; + goto err_put_codec_of_node; } tegra_sgtl5000_dai.platform_of_node = tegra_sgtl5000_dai.cpu_of_node; ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) - goto err; + goto err_put_cpu_of_node; ret = snd_soc_register_card(card); if (ret) { @@ -169,6 +169,13 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&machine->util_data); +err_put_cpu_of_node: + of_node_put(tegra_sgtl5000_dai.cpu_of_node); + tegra_sgtl5000_dai.cpu_of_node = NULL; + tegra_sgtl5000_dai.platform_of_node = NULL; +err_put_codec_of_node: + of_node_put(tegra_sgtl5000_dai.codec_of_node); + tegra_sgtl5000_dai.codec_of_node = NULL; err: return ret; } @@ -183,6 +190,12 @@ static int tegra_sgtl5000_driver_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); + of_node_put(tegra_sgtl5000_dai.cpu_of_node); + tegra_sgtl5000_dai.cpu_of_node = NULL; + tegra_sgtl5000_dai.platform_of_node = NULL; + of_node_put(tegra_sgtl5000_dai.codec_of_node); + tegra_sgtl5000_dai.codec_of_node = NULL; + return ret; } -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: sta32x: Add support for XTI clock" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: sta32x: Add support for XTI clock has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From fce9ec954a8af7e04cbf5b9daa8bec9c1df5cfe6 Mon Sep 17 00:00:00 2001 From: Daniel Mack <daniel(a)zonque.org> Date: Wed, 17 Oct 2018 13:37:03 +0200 Subject: [PATCH] ASoC: sta32x: Add support for XTI clock The STA32x chips feature an XTI clock input that needs to be stable before the reset signal is released. Therefore, the chip driver needs to get a handle to the clock. Instead of relying on other parts of the system to enable the clock, let the codec driver grab a handle itself. In order to keep existing boards working, clock support is made optional. Signed-off-by: Daniel Mack <daniel(a)zonque.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/st,sta32x.txt | 6 ++++ sound/soc/codecs/sta32x.c | 28 +++++++++++++++++++ 2 files changed, 34 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/st,sta32x.txt b/Documentation/devicetree/bindings/sound/st,sta32x.txt index ff4a685a4303..52265fb757c5 100644 --- a/Documentation/devicetree/bindings/sound/st,sta32x.txt +++ b/Documentation/devicetree/bindings/sound/st,sta32x.txt @@ -19,6 +19,10 @@ Required properties: Optional properties: + - clocks, clock-names: Clock specifier for XTI input clock. + If specified, the clock will be enabled when the codec is probed, + and disabled when it is removed. The 'clock-names' must be set to 'xti'. + - st,output-conf: number, Selects the output configuration: 0: 2-channel (full-bridge) power, 2-channel data-out 1: 2 (half-bridge). 1 (full-bridge) on-board power @@ -79,6 +83,8 @@ Example: codec: sta32x@38 { compatible = "st,sta32x"; reg = <0x1c>; + clocks = <&clock>; + clock-names = "xti"; reset-gpios = <&gpio1 19 0>; power-down-gpios = <&gpio1 16 0>; st,output-conf = /bits/ 8 <0x3>; // set output to 2-channel diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 22de1593443c..f753d2db0a5a 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -21,6 +21,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> +#include <linux/clk.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> @@ -142,6 +143,7 @@ static const char *sta32x_supply_names[] = { /* codec private data */ struct sta32x_priv { struct regmap *regmap; + struct clk *xti_clk; struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)]; struct snd_soc_component *component; struct sta32x_platform_data *pdata; @@ -879,6 +881,18 @@ static int sta32x_probe(struct snd_soc_component *component) struct sta32x_priv *sta32x = snd_soc_component_get_drvdata(component); struct sta32x_platform_data *pdata = sta32x->pdata; int i, ret = 0, thermal = 0; + + sta32x->component = component; + + if (sta32x->xti_clk) { + ret = clk_prepare_enable(sta32x->xti_clk); + if (ret != 0) { + dev_err(component->dev, + "Failed to enable clock: %d\n", ret); + return ret; + } + } + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); if (ret != 0) { @@ -981,6 +995,9 @@ static void sta32x_remove(struct snd_soc_component *component) sta32x_watchdog_stop(sta32x); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + if (sta32x->xti_clk) + clk_disable_unprepare(sta32x->xti_clk); } static const struct snd_soc_component_driver sta32x_component = { @@ -1097,6 +1114,17 @@ static int sta32x_i2c_probe(struct i2c_client *i2c, } #endif + /* Clock */ + sta32x->xti_clk = devm_clk_get(dev, "xti"); + if (IS_ERR(sta32x->xti_clk)) { + ret = PTR_ERR(sta32x->xti_clk); + + if (ret == -EPROBE_DEFER) + return ret; + + sta32x->xti_clk = NULL; + } + /* GPIOs */ sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_LOW); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: rsnd: enable TDM settings for SSI parent" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: rsnd: enable TDM settings for SSI parent has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 6817d7593f3e3c8a9c11e7a07cb5646c70371f0a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 17 Oct 2018 01:55:37 +0000 Subject: [PATCH] ASoC: rsnd: enable TDM settings for SSI parent Some SSIs are sharing each pins (= WS/CLK pin for playback/capture). Then, SSI parent needs control WS/CLK setting for SSI slave. In such case, SSI parent needs TDM settings if SSI slave is working as TDM mode. But it is not cared in current driver. It can't capture TDM sound without this patch if SSIs were pin sharing. This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 with TDM sound. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/ssi.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 3adcc4f778f7..b42a0e0feab7 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -399,6 +399,17 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, if (rdai->sys_delay) cr_own |= DEL; + /* + * TDM Mode + * see + * rsnd_ssiu_init_gen2() + */ + wsr = ssi->wsr; + if (is_tdm) { + wsr |= WS_MODE; + cr_own |= CHNL_8; + } + /* * We shouldn't exchange SWSP after running. * This means, parent needs to care it. @@ -429,16 +440,6 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_mode = DIEN; /* PIO : enable Data interrupt */ } - /* - * TDM Extend Mode - * see - * rsnd_ssiu_init_gen2() - */ - wsr = ssi->wsr; - if (is_tdm) { - wsr |= WS_MODE; - cr_own |= CHNL_8; - } init_end: ssi->cr_own = cr_own; ssi->cr_mode = cr_mode; -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer
by Richard Fitzgerald 19 Oct '18

19 Oct '18

The wm_adsp_buffer struct is the control header of a circular buffer used to transfer data from the firmware over the control interface to an ALSA compressed stream. The original names of the fields pointing to the data buffer were based on ADSP2V2 memory layout where they correspond to {XM, XM, YM}. But this circular buffer could be used on other types of DSP core that have different memory region types. Also the names and description of the size fields were not very clear. The field names and descriptions have been changed to be generic and not imply any particular memory types. This patch updates the wm_adsp driver to the new field names. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f61656070225..7ae10c632614 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -311,12 +311,12 @@ struct wm_adsp_alg_xm_struct { }; struct wm_adsp_buffer { - __be32 X_buf_base; /* XM base addr of first X area */ - __be32 X_buf_size; /* Size of 1st X area in words */ - __be32 X_buf_base2; /* XM base addr of 2nd X area */ - __be32 X_buf_brk; /* Total X size in words */ - __be32 Y_buf_base; /* YM base addr of Y area */ - __be32 wrap; /* Total size X and Y in words */ + __be32 buf1_base; /* Base addr of first buffer area */ + __be32 buf1_size; /* Size of buf1 area in DSP words */ + __be32 buf2_base; /* Base addr of 2nd buffer area */ + __be32 buf1_buf2_size; /* Size of buf1+buf2 in DSP words */ + __be32 buf3_base; /* Base addr of buf3 area */ + __be32 buf_total_size; /* Size of buf1+buf2+buf3 in DSP words */ __be32 high_water_mark; /* Point at which IRQ is asserted */ __be32 irq_count; /* bits 1-31 count IRQ assertions */ __be32 irq_ack; /* acked IRQ count, bit 0 enables IRQ */ @@ -393,18 +393,18 @@ struct wm_adsp_buffer_region_def { static const struct wm_adsp_buffer_region_def default_regions[] = { { .mem_type = WMFW_ADSP2_XM, - .base_offset = HOST_BUFFER_FIELD(X_buf_base), - .size_offset = HOST_BUFFER_FIELD(X_buf_size), + .base_offset = HOST_BUFFER_FIELD(buf1_base), + .size_offset = HOST_BUFFER_FIELD(buf1_size), }, { .mem_type = WMFW_ADSP2_XM, - .base_offset = HOST_BUFFER_FIELD(X_buf_base2), - .size_offset = HOST_BUFFER_FIELD(X_buf_brk), + .base_offset = HOST_BUFFER_FIELD(buf2_base), + .size_offset = HOST_BUFFER_FIELD(buf1_buf2_size), }, { .mem_type = WMFW_ADSP2_YM, - .base_offset = HOST_BUFFER_FIELD(Y_buf_base), - .size_offset = HOST_BUFFER_FIELD(wrap), + .base_offset = HOST_BUFFER_FIELD(buf3_base), + .size_offset = HOST_BUFFER_FIELD(buf_total_size), }, }; -- 2.11.0

1 1

[alsa-devel] AGL / Questions about the upcoming features of Alsa
by Thierry Bultel 19 Oct '18

19 Oct '18

Hi, My name is Thierry Bultel from Iot.bzh, and I am part of the responsible team for the audio framework (aka, 4A) in Automotive Grade Linux. For the ones interested in the audio architecture in AGL (and the ones that will read my list of questions below), these are useful links to follow: New AVIRT driver in replacement of SND_ALOOP: https://iot.bzh/download/public/2018/amm-dresden/ALSA_Virtual_Dynamic_Sound… 4A framework kickstart: https://iot.bzh/download/public/2018/amm-dresden/4a-kickstart-Loic-IoTbzh-D… In AGL, we have used the Alsa framework intensively. I wish I could meet the Alsa team next week in Edimburg but unfortunately I will not be available at that date. Other colleagues of mine will be there, however. Here are some points that I would like to be addressed at this meeting: - ioplug PCMs: There are known issues about potential (effective !) deadlocks when using ioplug PCMs. We have worked around this by setting LIBASOUND_THREAD_SAFE=0 Is there any pending work to avoid doing this and fix the bug ? - capture dispatching According to the experts, what would be the most effective way to implement a capture dispatcher ? The need is to have several applications listening to the same microphone. Is JACK the most suitable solution (at this time) ? (more precisely, this could behave as a 1to1 switch or 1toN, by duplicating the audio stream). A kernel solution based on AVIRT with routing could easily be done, too. - Documentation: Is there any pending rework of the documentation for having it a little bit more detailed ? For instance, having a page explaining the roles of period_size and buffer_size, would be great, since it impossible to perform sound playback correctly in an application, without understanding those notions. - Could something like a dynamic setting of the parameters of the rate converter plugin, become a reality one day ? - What is you overall mind about the AVIRT driver that we (with Fiberdyne) are developing ? - Is there a release features roadmap for Alsa (kernel & library) ? - What is your feeling about PipeWire, which could eventually replace PulseAudio (and seems to also invade the JACK land) ? Thanks ! Thierry Bultel

1 0

[alsa-devel] Audio miniconference schedule
by Mark Brown 18 Oct '18

18 Oct '18

As previously announced there's an audio miniconference happening on the 21st in Edinburgh, meeting around 9am and getting started once enough people have arrived (Cirrus people, please let me know if we need to start later, some people have said they'll be arriving late so we won't wait for everyone): https://blog.sirena.org.uk/2018/09/19/2018-linux-audio-miniconfernce (yes there is a typo in the URL). A number of people have proposed topics so I just went through them and pulled them into a proposed agenda, revised again after further feedback: - Introductions - Community governence issues (Takashi) - Virtualization (Liam) - AVB (Liam) - Testing (Takashi) - PCM improvements (Takashi) - Granularity - Timestamping - SoundWire (Liam) - Integrating the ALSA control core (Sakamoto-san) - Obsolete 'dimen' member in container for information to an element - Remove limitation on a container for value array to an element - User-defined element set left by finished process - Expand restriction to the number of user-defined control element set per sound card - Another approach to produce language bindings by GObject Introspection (Sakamoto-san) - ASoC rate domains/core refactoring (Charles) - ASoC/ALSA core merging (Takashi) - DSPs (Liam) - Sound Open Firmware (Liam) - Byte controls (Vinod/Patrick) - Calibration loading (Patrick) - Exposing graph at runtime (Patrick) - Batching commands (Patrick) - AGL (Stephane/Ronan) I hope I've not missed anything people suggested, the schedule is now looking pretty full but I'm hoping some of these topics can be relatively brief. I've dropped Y2038 from the expliit schedule, since Arnd Bergmann (who's been driving a lot of the general Y2038 work) will be in Edinburgh and may be free to join I'm suggesting that if he's free we cover that either first thing or just after lunch and otherwise Takashi and I will try to coordinate with him during the Maintainer's Summit on Monday (unless others have a great enthusiasm for the subject). What do people think? Any other topics? There's still time to sign up if you're interested in attending, you can sign up right up to the day of the event though registering early would help a lot with logistics.

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[alsa-devel] [PATCH] conf/ucm/Dell-WD15-Dock: Fix incorrect device names
by Takashi Iwai 18 Oct '18

18 Oct '18

The device name string for Dell WD15 (and its variants) dock is set as "WD15Dock", while the actual device name to be used is "Dock". Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1112292 Fixes: 8ebb40c96970 ("conf/ucm: Add a UCM profile for Dell WD15 Dock USB-audio") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/conf/ucm/Dell-WD15-Dock/HiFi.conf | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/src/conf/ucm/Dell-WD15-Dock/HiFi.conf b/src/conf/ucm/Dell-WD15-Dock/HiFi.conf index b07f34400086..1b6bd21237f1 100644 --- a/src/conf/ucm/Dell-WD15-Dock/HiFi.conf +++ b/src/conf/ucm/Dell-WD15-Dock/HiFi.conf @@ -3,7 +3,7 @@ SectionDevice."Headphone" { Value { PlaybackChannels "2" - PlaybackPCM "hw:WD15Dock,0" + PlaybackPCM "hw:Dock,0" } } @@ -12,7 +12,7 @@ SectionDevice."LineOut" { Value { PlaybackChannels "2" - PlaybackPCM "hw:WD15Dock,1" + PlaybackPCM "hw:Dock,1" } } @@ -21,6 +21,6 @@ SectionDevice."Mic" { Value { CaptureChannels "2" - CapturePCM "hw:WD15Dock,0" + CapturePCM "hw:Dock,0" } } -- 2.19.0

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[alsa-devel] [PATCH - Intervals 1/1] interval: Interpret (x x+1] correctly and return x+1
by twischer＠de.adit-jv.com 18 Oct '18

18 Oct '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without this change an interval of (x x+1] will be interpreted as an empty interval but the right value would be x+1. This leads to a failing snd_pcm_hw_params() call which returns -EINVAL. An example issue log is given in the following: snd_pcm_hw_params failed with err -22 (Invalid argument) ACCESS: MMAP_NONINTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 16 CHANNELS: 1 RATE: 16000 PERIOD_TIME: (15999 16000] PERIOD_SIZE: (255 256] PERIOD_BYTES: (510 512] PERIODS: [2 3) BUFFER_TIME: 32000 BUFFER_SIZE: 512 BUFFER_BYTES: 1024 In case of (x x+1) we have to interpret it anyway as a single value of x to compensate rounding issues. For example the period size will result in an interval of (352 353) when the period time is 16ms and the sample rate 22050 Hz (16ms * 22,05 kHz = 352,8 frames). But 352 has to be chosen to allow a buffer size of 705 (32ms * 22,05 kHz = 705,6 frames) which has to be >= 2x period size to avoid Xruns. The buffer size will not end up with an interval of (705 706) similar to the period size because snd_pcm_rate_hw_refine_cchange() calls snd_interval_floor() for the buffer size. Therefore this value will be interpreted as an integer interval instead of a real interval further on. This issue seems to exist since the change of 9bb985c38 ("pcm: snd_interval_refine_first/last: exclude value only if also excluded before") Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/interval_inline.h b/src/pcm/interval_inline.h index a68e292..5e59d72 100644 --- a/src/pcm/interval_inline.h +++ b/src/pcm/interval_inline.h @@ -51,12 +51,14 @@ INTERVAL_INLINE int snd_interval_single(const snd_interval_t *i) { assert(!snd_interval_empty(i)); return (i->min == i->max || - (i->min + 1 == i->max && i->openmax)); + (i->min + 1 == i->max && (i->openmin || i->openmax))); } INTERVAL_INLINE int snd_interval_value(const snd_interval_t *i) { assert(snd_interval_single(i)); + if (!i->openmax) + return i->max; return i->min; } -- 2.7.4

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[alsa-devel] [PATCH] ALSA: doc: Brush up the old writing-an-alsa-driver
by Takashi Iwai 18 Oct '18

18 Oct '18

Slightly brushing up and throw the old dust away from my ancient writing-an-alsa-driver document. The contents aren't changed so much but the obsoleted parts are dropped. Also, remove the date and the version number. It's useless. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is a patch included in my devres patchset. Since this is an independent fix for documentation, let's apply this at first. .../kernel-api/writing-an-alsa-driver.rst | 307 +++++++++--------- 1 file changed, 149 insertions(+), 158 deletions(-) diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index a0b268466cb1..b60a26807420 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -3,8 +3,6 @@ Writing an ALSA Driver ====================== :Author: Takashi Iwai <tiwai(a)suse.de> -:Date: Oct 15, 2007 -:Edition: 0.3.7 Preface ======= @@ -21,11 +19,6 @@ explain the general topic of linux kernel coding and doesn't cover low-level driver implementation details. It only describes the standard way to write a PCI sound driver on ALSA. -If you are already familiar with the older ALSA ver.0.5.x API, you can -check the drivers such as ``sound/pci/es1938.c`` or -``sound/pci/maestro3.c`` which have also almost the same code-base in -the ALSA 0.5.x tree, so you can compare the differences. - This document is still a draft version. Any feedback and corrections, please!! @@ -35,24 +28,7 @@ File Tree Structure General ------- -The ALSA drivers are provided in two ways. - -One is the trees provided as a tarball or via cvs from the ALSA's ftp -site, and another is the 2.6 (or later) Linux kernel tree. To -synchronize both, the ALSA driver tree is split into two different -trees: alsa-kernel and alsa-driver. The former contains purely the -source code for the Linux 2.6 (or later) tree. This tree is designed -only for compilation on 2.6 or later environment. The latter, -alsa-driver, contains many subtle files for compiling ALSA drivers -outside of the Linux kernel tree, wrapper functions for older 2.2 and -2.4 kernels, to adapt the latest kernel API, and additional drivers -which are still in development or in tests. The drivers in alsa-driver -tree will be moved to alsa-kernel (and eventually to the 2.6 kernel -tree) when they are finished and confirmed to work fine. - -The file tree structure of ALSA driver is depicted below. Both -alsa-kernel and alsa-driver have almost the same file structure, except -for “core” directory. It's named as “acore” in alsa-driver tree. +The file tree structure of ALSA driver is depicted below. :: @@ -61,14 +37,11 @@ for “core” directory. It's named as “acore” in alsa-driver tree. /oss /seq /oss - /instr - /ioctl32 /include /drivers /mpu401 /opl3 /i2c - /l3 /synth /emux /pci @@ -80,6 +53,7 @@ for “core” directory. It's named as “acore” in alsa-driver tree. /sparc /usb /pcmcia /(cards) + /soc /oss @@ -99,13 +73,6 @@ directory. The rawmidi OSS emulation is included in the ALSA rawmidi code since it's quite small. The sequencer code is stored in ``core/seq/oss`` directory (see `below <#core-seq-oss>`__). -core/ioctl32 -~~~~~~~~~~~~ - -This directory contains the 32bit-ioctl wrappers for 64bit architectures -such like x86-64, ppc64 and sparc64. For 32bit and alpha architectures, -these are not compiled. - core/seq ~~~~~~~~ @@ -119,11 +86,6 @@ core/seq/oss This contains the OSS sequencer emulation codes. -core/seq/instr -~~~~~~~~~~~~~~ - -This directory contains the modules for the sequencer instrument layer. - include directory ----------------- @@ -161,11 +123,6 @@ Although there is a standard i2c layer on Linux, ALSA has its own i2c code for some cards, because the soundcard needs only a simple operation and the standard i2c API is too complicated for such a purpose. -i2c/l3 -~~~~~~ - -This is a sub-directory for ARM L3 i2c. - synth directory --------------- @@ -209,11 +166,19 @@ The PCMCIA, especially PCCard drivers will go here. CardBus drivers will be in the pci directory, because their API is identical to that of standard PCI cards. +soc directory +------------- + +This directory contains the codes for ASoC (ALSA System on Chip) +layer including ASoC core, codec and machine drivers. + oss directory ------------- -The OSS/Lite source files are stored here in Linux 2.6 (or later) tree. -In the ALSA driver tarball, this directory is empty, of course :) +Here contains OSS/Lite codes. +All codes have been deprecated except for dmasound on m68k as of +writing this. + Basic Flow for PCI Drivers ========================== @@ -352,10 +317,8 @@ to details explained in the following section. /* (3) */ err = snd_mychip_create(card, pci, &chip); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto error; /* (4) */ strcpy(card->driver, "My Chip"); @@ -368,22 +331,23 @@ to details explained in the following section. /* (6) */ err = snd_card_register(card); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto error; /* (7) */ pci_set_drvdata(pci, card); dev++; return 0; + + error: + snd_card_free(card); + return err; } /* destructor -- see the "Destructor" sub-section */ static void snd_mychip_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } @@ -445,14 +409,26 @@ In this part, the PCI resources are allocated. struct mychip *chip; .... err = snd_mychip_create(card, pci, &chip); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto error; The details will be explained in the section `PCI Resource Management`_. +When something goes wrong, the probe function needs to deal with the +error. In this example, we have a single error handling path placed +at the end of the function. + +:: + + error: + snd_card_free(card); + return err; + +Since each component can be properly freed, the single +:c:func:`snd_card_free()` call should suffice in most cases. + + 4) Set the driver ID and name strings. ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ @@ -486,10 +462,8 @@ too. :: err = snd_card_register(card); - if (err < 0) { - snd_card_free(card); - return err; - } + if (err < 0) + goto error; Will be explained in the section `Management of Cards and Components`_, too. @@ -513,14 +487,13 @@ The destructor, remove callback, simply releases the card instance. Then the ALSA middle layer will release all the attached components automatically. -It would be typically like the following: +It would be typically just :c:func:`calling snd_card_free()`: :: static void snd_mychip_remove(struct pci_dev *pci) { snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); } @@ -546,7 +519,7 @@ in the source file. If the code is split into several files, the files without module options don't need them. In addition to these headers, you'll need ``<linux/interrupt.h>`` for -interrupt handling, and ``<asm/io.h>`` for I/O access. If you use the +interrupt handling, and ``<linux/io.h>`` for I/O access. If you use the :c:func:`mdelay()` or :c:func:`udelay()` functions, you'll need to include ``<linux/delay.h>`` too. @@ -720,6 +693,13 @@ function, which will call the real destructor. where :c:func:`snd_mychip_free()` is the real destructor. +The demerit of this method is the obviously more amount of codes. +The merit is, however, you can trigger the own callback at registering +and disconnecting the card via setting in snd_device_ops. +About the registering and disconnecting the card, see the subsections +below. + + Registration and Release ------------------------ @@ -905,10 +885,8 @@ Resource Allocation ------------------- The allocation of I/O ports and irqs is done via standard kernel -functions. Unlike ALSA ver.0.5.x., there are no helpers for that. And -these resources must be released in the destructor function (see below). -Also, on ALSA 0.9.x, you don't need to allocate (pseudo-)DMA for PCI -like in ALSA 0.5.x. +functions. These resources must be released in the destructor +function (see below). Now assume that the PCI device has an I/O port with 8 bytes and an interrupt. Then :c:type:`struct mychip <mychip>` will have the @@ -1064,7 +1042,8 @@ and the allocation would be like below: :: - if ((err = pci_request_regions(pci, "My Chip")) < 0) { + err = pci_request_regions(pci, "My Chip"); + if (err < 0) { kfree(chip); return err; } @@ -1086,6 +1065,21 @@ and the corresponding destructor would be: .... } +Of course, a modern way with :c:func:`pci_iomap()` will make things a +bit easier, too. + +:: + + err = pci_request_regions(pci, "My Chip"); + if (err < 0) { + kfree(chip); + return err; + } + chip->iobase_virt = pci_iomap(pci, 0, 0); + +which is paired with :c:func:`pci_iounmap()` at destructor. + + PCI Entries ----------- @@ -1154,13 +1148,6 @@ And at last, the module entries: Note that these module entries are tagged with ``__init`` and ``__exit`` prefixes. -Oh, one thing was forgotten. If you have no exported symbols, you need -to declare it in 2.2 or 2.4 kernels (it's not necessary in 2.6 kernels). - -:: - - EXPORT_NO_SYMBOLS; - That's all! PCM Interface @@ -2113,6 +2100,16 @@ non-contiguous buffers. The mmap calls this callback to get the page address. Some examples will be explained in the later section `Buffer and Memory Management`_, too. +mmap calllback +~~~~~~~~~~~~~~ + +This is another optional callback for controlling mmap behavior. +Once when defined, PCM core calls this callback when a page is +memory-mapped instead of dealing via the standard helper. +If you need special handling (due to some architecture or +device-specific issues), implement everything here as you like. + + PCM Interrupt Handler --------------------- @@ -2370,6 +2367,27 @@ to define the inverse rule: hw_rule_format_by_channels, NULL, SNDRV_PCM_HW_PARAM_CHANNELS, -1); +One typical usage of the hw constraints is to align the buffer size +with the period size. As default, ALSA PCM core doesn't enforce the +buffer size to be aligned with the period size. For example, it'd be +possible to have a combination like 256 period bytes with 999 buffer +bytes. + +Many device chips, however, require the buffer to be a multiple of +periods. In such a case, call +:c:func:`snd_pcm_hw_constraint_integer()` for +``SNDRV_PCM_HW_PARAM_PERIODS``. + +:: + + snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + +This assures that the number of periods is integer, hence the buffer +size is aligned with the period size. + +The hw constraint is a very much powerful mechanism to define the +preferred PCM configuration, and there are relevant helpers. I won't give more details here, rather I would like to say, “Luke, use the source.” @@ -3712,7 +3730,14 @@ example, for an intermediate buffer. Since the allocated pages are not contiguous, you need to set the ``page`` callback to obtain the physical address at every offset. -The implementation of ``page`` callback would be like this: +The easiest way to achieve it would be to use +:c:func:`snd_pcm_lib_alloc_vmalloc_buffer()` for allocating the buffer +via :c:func:`vmalloc()`, and set :c:func:`snd_pcm_sgbuf_ops_page()` to +the ``page`` callback. At release, you need to call +:c:func:`snd_pcm_lib_free_vmalloc_buffer()`. + +If you want to implementation the ``page`` manually, it would be like +this: :: @@ -3848,7 +3873,9 @@ Power Management If the chip is supposed to work with suspend/resume functions, you need to add power-management code to the driver. The additional code for -power-management should be ifdef-ed with ``CONFIG_PM``. +power-management should be ifdef-ed with ``CONFIG_PM``, or annotated +with __maybe_unused attribute; otherwise the compiler will complain +you. If the driver *fully* supports suspend/resume that is, the device can be properly resumed to its state when suspend was called, you can set the @@ -3879,18 +3906,16 @@ the case of PCI drivers, the callbacks look like below: :: - #ifdef CONFIG_PM - static int snd_my_suspend(struct pci_dev *pci, pm_message_t state) + static int __maybe_unused snd_my_suspend(struct device *dev) { .... /* do things for suspend */ return 0; } - static int snd_my_resume(struct pci_dev *pci) + static int __maybe_unused snd_my_resume(struct device *dev) { .... /* do things for suspend */ return 0; } - #endif The scheme of the real suspend job is as follows. @@ -3909,18 +3934,14 @@ The scheme of the real suspend job is as follows. 6. Stop the hardware if necessary. -7. Disable the PCI device by calling - :c:func:`pci_disable_device()`. Then, call - :c:func:`pci_save_state()` at last. - A typical code would be like: :: - static int mychip_suspend(struct pci_dev *pci, pm_message_t state) + static int __maybe_unused mychip_suspend(struct device *dev) { /* (1) */ - struct snd_card *card = pci_get_drvdata(pci); + struct snd_card *card = dev_get_drvdata(dev); struct mychip *chip = card->private_data; /* (2) */ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -3932,9 +3953,6 @@ A typical code would be like: snd_mychip_save_registers(chip); /* (6) */ snd_mychip_stop_hardware(chip); - /* (7) */ - pci_disable_device(pci); - pci_save_state(pci); return 0; } @@ -3943,44 +3961,35 @@ The scheme of the real resume job is as follows. 1. Retrieve the card and the chip data. -2. Set up PCI. First, call :c:func:`pci_restore_state()`. Then - enable the pci device again by calling - :c:func:`pci_enable_device()`. Call - :c:func:`pci_set_master()` if necessary, too. +2. Re-initialize the chip. -3. Re-initialize the chip. +3. Restore the saved registers if necessary. -4. Restore the saved registers if necessary. +4. Resume the mixer, e.g. calling :c:func:`snd_ac97_resume()`. -5. Resume the mixer, e.g. calling :c:func:`snd_ac97_resume()`. +5. Restart the hardware (if any). -6. Restart the hardware (if any). - -7. Call :c:func:`snd_power_change_state()` with +6. Call :c:func:`snd_power_change_state()` with ``SNDRV_CTL_POWER_D0`` to notify the processes. A typical code would be like: :: - static int mychip_resume(struct pci_dev *pci) + static int __maybe_unused mychip_resume(struct pci_dev *pci) { /* (1) */ - struct snd_card *card = pci_get_drvdata(pci); + struct snd_card *card = dev_get_drvdata(dev); struct mychip *chip = card->private_data; /* (2) */ - pci_restore_state(pci); - pci_enable_device(pci); - pci_set_master(pci); - /* (3) */ snd_mychip_reinit_chip(chip); - /* (4) */ + /* (3) */ snd_mychip_restore_registers(chip); - /* (5) */ + /* (4) */ snd_ac97_resume(chip->ac97); - /* (6) */ + /* (5) */ snd_mychip_restart_chip(chip); - /* (7) */ + /* (6) */ snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } @@ -4046,15 +4055,14 @@ And next, set suspend/resume callbacks to the pci_driver. :: + static SIMPLE_DEV_PM_OPS(snd_my_pm_ops, mychip_suspend, mychip_resume); + static struct pci_driver driver = { .name = KBUILD_MODNAME, .id_table = snd_my_ids, .probe = snd_my_probe, .remove = snd_my_remove, - #ifdef CONFIG_PM - .suspend = snd_my_suspend, - .resume = snd_my_resume, - #endif + .driver.pm = &snd_my_pm_ops, }; Module Parameters @@ -4078,7 +4086,7 @@ variables, instead. ``enable`` option is not always necessary in this case, but it would be better to have a dummy option for compatibility. The module parameters must be declared with the standard -``module_param()()``, ``module_param_array()()`` and +``module_param()``, ``module_param_array()`` and :c:func:`MODULE_PARM_DESC()` macros. The typical coding would be like below: @@ -4094,15 +4102,14 @@ The typical coding would be like below: module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); -Also, don't forget to define the module description, classes, license -and devices. Especially, the recent modprobe requires to define the +Also, don't forget to define the module description and the license. +Especially, the recent modprobe requires to define the module license as GPL, etc., otherwise the system is shown as “tainted”. :: - MODULE_DESCRIPTION("My Chip"); + MODULE_DESCRIPTION("Sound driver for My Chip"); MODULE_LICENSE("GPL"); - MODULE_SUPPORTED_DEVICE("{{Vendor,My Chip Name}}"); How To Put Your Driver Into ALSA Tree @@ -4117,21 +4124,17 @@ a question now: how to put my own driver into the ALSA driver tree? Here Suppose that you create a new PCI driver for the card “xyz”. The card module name would be snd-xyz. The new driver is usually put into the -alsa-driver tree, ``alsa-driver/pci`` directory in the case of PCI -cards. Then the driver is evaluated, audited and tested by developers -and users. After a certain time, the driver will go to the alsa-kernel -tree (to the corresponding directory, such as ``alsa-kernel/pci``) and -eventually will be integrated into the Linux 2.6 tree (the directory -would be ``linux/sound/pci``). +alsa-driver tree, ``sound/pci`` directory in the case of PCI +cards. In the following sections, the driver code is supposed to be put into -alsa-driver tree. The two cases are covered: a driver consisting of a +Linux kernel tree. The two cases are covered: a driver consisting of a single source file and one consisting of several source files. Driver with A Single Source File -------------------------------- -1. Modify alsa-driver/pci/Makefile +1. Modify sound/pci/Makefile Suppose you have a file xyz.c. Add the following two lines @@ -4160,52 +4163,43 @@ Driver with A Single Source File For the details of Kconfig script, refer to the kbuild documentation. -3. Run cvscompile script to re-generate the configure script and build - the whole stuff again. - Drivers with Several Source Files --------------------------------- Suppose that the driver snd-xyz have several source files. They are -located in the new subdirectory, pci/xyz. +located in the new subdirectory, sound/pci/xyz. -1. Add a new directory (``xyz``) in ``alsa-driver/pci/Makefile`` as - below +1. Add a new directory (``sound/pci/xyz``) in ``sound/pci/Makefile`` + as below :: - obj-$(CONFIG_SND) += xyz/ + obj-$(CONFIG_SND) += sound/pci/xyz/ -2. Under the directory ``xyz``, create a Makefile +2. Under the directory ``sound/pci/xyz``, create a Makefile :: - ifndef SND_TOPDIR - SND_TOPDIR=../.. - endif - - include $(SND_TOPDIR)/toplevel.config - include $(SND_TOPDIR)/Makefile.conf - snd-xyz-objs := xyz.o abc.o def.o - obj-$(CONFIG_SND_XYZ) += snd-xyz.o - include $(SND_TOPDIR)/Rules.make - 3. Create the Kconfig entry This procedure is as same as in the last section. -4. Run cvscompile script to re-generate the configure script and build - the whole stuff again. Useful Functions ================ :c:func:`snd_printk()` and friends ---------------------------------------- +---------------------------------- + +.. note:: This subsection describes a few helper functions for +decorating a bit more on the standard :c:func:`printk()` & co. +However, in general, the use of such helpers is no longer recommended. +If possible, try to stick with the standard functions like +:c:func:`dev_err()` or :c:func:`pr_err()`. ALSA provides a verbose version of the :c:func:`printk()` function. If a kernel config ``CONFIG_SND_VERBOSE_PRINTK`` is set, this function @@ -4221,13 +4215,10 @@ just like :c:func:`snd_printk()`. If the ALSA is compiled without the debugging flag, it's ignored. :c:func:`snd_printdd()` is compiled in only when -``CONFIG_SND_DEBUG_VERBOSE`` is set. Please note that -``CONFIG_SND_DEBUG_VERBOSE`` is not set as default even if you configure -the alsa-driver with ``--with-debug=full`` option. You need to give -explicitly ``--with-debug=detect`` option instead. +``CONFIG_SND_DEBUG_VERBOSE`` is set. :c:func:`snd_BUG()` ------------------------- +------------------- It shows the ``BUG?`` message and stack trace as well as :c:func:`snd_BUG_ON()` at the point. It's useful to show that a @@ -4236,7 +4227,7 @@ fatal error happens there. When no debug flag is set, this macro is ignored. :c:func:`snd_BUG_ON()` ----------------------------- +---------------------- :c:func:`snd_BUG_ON()` macro is similar with :c:func:`WARN_ON()` macro. For example, snd_BUG_ON(!pointer); or -- 2.19.0

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[alsa-devel] [PATCH v2] ASoC: pcm3168a: remove read-only status register from snd_kcontrol_new
by Kuninori Morimoto 18 Oct '18

18 Oct '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> "DACn Zero Flag" and "ADCn Overflow Flag" are read-only status register. There is no detail explanation for these register on datasheet, but, judging from its behavior, it seems these register needs clock to read. Otherwise, I2C returns error. Therefor, amixer command will be failed under non working timing. These information are needed for driver, not for user. So, let's remove these from snd_kcontrol_new. Without this patch, user often fail amixer command. This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 TDM sound. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - removed directly, instead of using #if 0 - tidyup git-log sound/soc/codecs/pcm3168a.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 52cc950..e1c3a18 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -133,10 +133,6 @@ static const struct snd_kcontrol_new pcm3168a_snd_controls[] = { SOC_DOUBLE("DAC2 Invert Switch", PCM3168A_DAC_INV, 2, 3, 1, 0), SOC_DOUBLE("DAC3 Invert Switch", PCM3168A_DAC_INV, 4, 5, 1, 0), SOC_DOUBLE("DAC4 Invert Switch", PCM3168A_DAC_INV, 6, 7, 1, 0), - SOC_DOUBLE_STS("DAC1 Zero Flag", PCM3168A_DAC_ZERO, 0, 1, 1, 0), - SOC_DOUBLE_STS("DAC2 Zero Flag", PCM3168A_DAC_ZERO, 2, 3, 1, 0), - SOC_DOUBLE_STS("DAC3 Zero Flag", PCM3168A_DAC_ZERO, 4, 5, 1, 0), - SOC_DOUBLE_STS("DAC4 Zero Flag", PCM3168A_DAC_ZERO, 6, 7, 1, 0), SOC_ENUM("DAC Volume Control Type", pcm3168a_dac_volume_type), SOC_ENUM("DAC Volume Rate Multiplier", pcm3168a_dac_att_mult), SOC_ENUM("DAC De-Emphasis", pcm3168a_dac_demp), @@ -176,9 +172,6 @@ static const struct snd_kcontrol_new pcm3168a_snd_controls[] = { SOC_DOUBLE("ADC1 Mute Switch", PCM3168A_ADC_MUTE, 0, 1, 1, 0), SOC_DOUBLE("ADC2 Mute Switch", PCM3168A_ADC_MUTE, 2, 3, 1, 0), SOC_DOUBLE("ADC3 Mute Switch", PCM3168A_ADC_MUTE, 4, 5, 1, 0), - SOC_DOUBLE_STS("ADC1 Overflow Flag", PCM3168A_ADC_OV, 0, 1, 1, 0), - SOC_DOUBLE_STS("ADC2 Overflow Flag", PCM3168A_ADC_OV, 2, 3, 1, 0), - SOC_DOUBLE_STS("ADC3 Overflow Flag", PCM3168A_ADC_OV, 4, 5, 1, 0), SOC_ENUM("ADC Volume Control Type", pcm3168a_adc_volume_type), SOC_ENUM("ADC Volume Rate Multiplier", pcm3168a_adc_att_mult), SOC_ENUM("ADC Overflow Flag Polarity", pcm3168a_adc_ov_pol), -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: pcm3168a: remove read-only status register from snd_kcontrol_new
by Kuninori Morimoto 18 Oct '18

18 Oct '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> "DACn Zero Flag" and "ADCn Overflow Flag" are read-only status register. There is no detail explanation of these register on datasheet, but, judging from it behavior, it seems these register needs clock to reading. Therefor, amixer command will be failed under non working timing. Let's ignore these from snd_kcontrol_new so far, but keep it under #if 0. Without this patch, amixer command offten failed and can't sound settings. This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 TDM sound. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/pcm3168a.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 52cc950..29eaf9d 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -133,10 +133,13 @@ static const struct snd_kcontrol_new pcm3168a_snd_controls[] = { SOC_DOUBLE("DAC2 Invert Switch", PCM3168A_DAC_INV, 2, 3, 1, 0), SOC_DOUBLE("DAC3 Invert Switch", PCM3168A_DAC_INV, 4, 5, 1, 0), SOC_DOUBLE("DAC4 Invert Switch", PCM3168A_DAC_INV, 6, 7, 1, 0), +#if 0 + /* read-only status register needs clocks */ SOC_DOUBLE_STS("DAC1 Zero Flag", PCM3168A_DAC_ZERO, 0, 1, 1, 0), SOC_DOUBLE_STS("DAC2 Zero Flag", PCM3168A_DAC_ZERO, 2, 3, 1, 0), SOC_DOUBLE_STS("DAC3 Zero Flag", PCM3168A_DAC_ZERO, 4, 5, 1, 0), SOC_DOUBLE_STS("DAC4 Zero Flag", PCM3168A_DAC_ZERO, 6, 7, 1, 0), +#endif SOC_ENUM("DAC Volume Control Type", pcm3168a_dac_volume_type), SOC_ENUM("DAC Volume Rate Multiplier", pcm3168a_dac_att_mult), SOC_ENUM("DAC De-Emphasis", pcm3168a_dac_demp), @@ -176,9 +179,12 @@ static const struct snd_kcontrol_new pcm3168a_snd_controls[] = { SOC_DOUBLE("ADC1 Mute Switch", PCM3168A_ADC_MUTE, 0, 1, 1, 0), SOC_DOUBLE("ADC2 Mute Switch", PCM3168A_ADC_MUTE, 2, 3, 1, 0), SOC_DOUBLE("ADC3 Mute Switch", PCM3168A_ADC_MUTE, 4, 5, 1, 0), +#if 0 + /* read-only status register needs clocks */ SOC_DOUBLE_STS("ADC1 Overflow Flag", PCM3168A_ADC_OV, 0, 1, 1, 0), SOC_DOUBLE_STS("ADC2 Overflow Flag", PCM3168A_ADC_OV, 2, 3, 1, 0), SOC_DOUBLE_STS("ADC3 Overflow Flag", PCM3168A_ADC_OV, 4, 5, 1, 0), +#endif SOC_ENUM("ADC Volume Control Type", pcm3168a_adc_volume_type), SOC_ENUM("ADC Volume Rate Multiplier", pcm3168a_adc_att_mult), SOC_ENUM("ADC Overflow Flag Polarity", pcm3168a_adc_ov_pol), -- 2.7.4

2 2

[alsa-devel] [PATCH v2 0/9] ARM: dts: tegra: last minute fixes
by Marcel Ziswiler 17 Oct '18

17 Oct '18

This series comes with some last minutes fixes and further clean-up. Changes in v2: - Dropped "[PATCH v1 3/8] ARM: tegra: apalis/colibri_t30: fix hdmi regulator" as suggested by Russell et. al. - Added 2 new patches improving/fixing audio on Apalis TK1. Changes in v1: - Remove simple-panel compatible as suggested by Rob. Marcel Ziswiler (9): ARM: tegra: fix simple-panel compatibles ARM: tegra: apalis-tk1/colibri_t20/t30: eval/iris: fix regulator gpio enable ARM: tegra: colibri_t20: reorder pmic properties ARM: tegra: apalis-tk1: further regulator clean-up ARM: tegra: apalis_t30/tk1: annotate power I2C being on-module ARM: tegra: colibri_t30: further regulator clean-up ARM: tegra: apalis_t30: further regulator clean-up ASoC: tegra_sgtl5000: fix device_node refcounting ASoC: tegra_sgtl5000: fix platform name vs. of_node assignement arch/arm/boot/dts/tegra114-dalmore.dts | 3 +- arch/arm/boot/dts/tegra124-apalis-eval.dts | 4 +- arch/arm/boot/dts/tegra124-apalis-v1.2-eval.dts | 4 +- arch/arm/boot/dts/tegra124-apalis-v1.2.dtsi | 17 +++--- arch/arm/boot/dts/tegra124-apalis.dtsi | 17 +++--- arch/arm/boot/dts/tegra124-venice2.dts | 2 +- arch/arm/boot/dts/tegra20-colibri-eval-v3.dts | 4 +- arch/arm/boot/dts/tegra20-colibri-iris.dts | 4 +- arch/arm/boot/dts/tegra20-colibri.dtsi | 4 +- arch/arm/boot/dts/tegra20-harmony.dts | 2 +- arch/arm/boot/dts/tegra20-medcom-wide.dts | 2 +- arch/arm/boot/dts/tegra20-paz00.dts | 2 +- arch/arm/boot/dts/tegra20-seaboard.dts | 2 +- arch/arm/boot/dts/tegra20-ventana.dts | 2 +- arch/arm/boot/dts/tegra30-apalis-eval.dts | 2 +- arch/arm/boot/dts/tegra30-apalis-v1.1-eval.dts | 4 +- arch/arm/boot/dts/tegra30-apalis-v1.1.dtsi | 72 +++++++++++-------------- arch/arm/boot/dts/tegra30-apalis.dtsi | 70 ++++++++++-------------- arch/arm/boot/dts/tegra30-cardhu.dtsi | 2 +- arch/arm/boot/dts/tegra30-colibri-eval-v3.dts | 4 +- arch/arm/boot/dts/tegra30-colibri.dtsi | 45 ++++++++-------- sound/soc/tegra/tegra_sgtl5000.c | 22 +++++++- 22 files changed, 143 insertions(+), 147 deletions(-) -- 2.14.4

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Re: [alsa-devel] [PATCH 2/3] MIPS: Ci20: Enable SND_JZ4740_SOC driver
by Mathieu Malaterre 17 Oct '18

17 Oct '18

On Wed, Jun 6, 2018 at 9:38 PM Mathieu Malaterre <malat(a)debian.org> wrote: > > Update the Ci20's defconfig to enable the JZ4780's SND driver. > > Signed-off-by: Mathieu Malaterre <malat(a)debian.org> > --- > arch/mips/configs/ci20_defconfig | 4 ++++ > 1 file changed, 4 insertions(+) > > diff --git a/arch/mips/configs/ci20_defconfig b/arch/mips/configs/ci20_defconfig > index e1c14f6af824..0c08c7675b42 100644 > --- a/arch/mips/configs/ci20_defconfig > +++ b/arch/mips/configs/ci20_defconfig > @@ -104,6 +104,10 @@ CONFIG_REGULATOR=y > CONFIG_REGULATOR_DEBUG=y > CONFIG_REGULATOR_FIXED_VOLTAGE=y > # CONFIG_VGA_CONSOLE is not set > +CONFIG_SOUND=y > +CONFIG_SND=y > +CONFIG_SND_SOC=y > +CONFIG_SND_JZ4740_SOC=y technically I am also missing: CONFIG_SND_JZ4740_SOC_I2S=y > # CONFIG_HID is not set > # CONFIG_USB_SUPPORT is not set > CONFIG_MMC=y > -- > 2.11.0 >

1 0

Re: [alsa-devel] [PATCH 1/3] MIPS: jz4780: Allow access to jz4740-i2s
by Paul Burton 17 Oct '18

17 Oct '18

Hi Matthieu, On Wed, Jun 06, 2018 at 09:38:08PM +0200, Mathieu Malaterre wrote: > diff --git a/sound/soc/jz4740/Kconfig b/sound/soc/jz4740/Kconfig > index 1a354a6b6e87..35d82d96e781 100644 > --- a/sound/soc/jz4740/Kconfig > +++ b/sound/soc/jz4740/Kconfig > @@ -1,20 +1,20 @@ > config SND_JZ4740_SOC > - tristate "SoC Audio for Ingenic JZ4740 SoC" > - depends on MACH_JZ4740 || COMPILE_TEST > + tristate "SoC Audio for Ingenic JZ4740/JZ4780 SoC" > + depends on MACH_JZ4740 || MACH_JZ4780 || COMPILE_TEST Perhaps this could be MACH_INGENIC, or even just MIPS? Thanks, Paul

2 4

[alsa-devel] [PATCH] ASoC: sta32x: Add support for XTI clock
by Daniel Mack 17 Oct '18

17 Oct '18

The STA32x chips feature an XTI clock input that needs to be stable before the reset signal is released. Therefore, the chip driver needs to get a handle to the clock. Instead of relying on other parts of the system to enable the clock, let the codec driver grab a handle itself. In order to keep existing boards working, clock support is made optional. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- .../devicetree/bindings/sound/st,sta32x.txt | 6 +++++ sound/soc/codecs/sta32x.c | 25 +++++++++++++++++++ 2 files changed, 31 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/st,sta32x.txt b/Documentation/devicetree/bindings/sound/st,sta32x.txt index ff4a685a4303..52265fb757c5 100644 --- a/Documentation/devicetree/bindings/sound/st,sta32x.txt +++ b/Documentation/devicetree/bindings/sound/st,sta32x.txt @@ -19,6 +19,10 @@ Required properties: Optional properties: + - clocks, clock-names: Clock specifier for XTI input clock. + If specified, the clock will be enabled when the codec is probed, + and disabled when it is removed. The 'clock-names' must be set to 'xti'. + - st,output-conf: number, Selects the output configuration: 0: 2-channel (full-bridge) power, 2-channel data-out 1: 2 (half-bridge). 1 (full-bridge) on-board power @@ -79,6 +83,8 @@ Example: codec: sta32x@38 { compatible = "st,sta32x"; reg = <0x1c>; + clocks = <&clock>; + clock-names = "xti"; reset-gpios = <&gpio1 19 0>; power-down-gpios = <&gpio1 16 0>; st,output-conf = /bits/ 8 <0x3>; // set output to 2-channel diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index a878c8d92e68..f753d2db0a5a 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -21,6 +21,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> +#include <linux/clk.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> @@ -142,6 +143,7 @@ static const char *sta32x_supply_names[] = { /* codec private data */ struct sta32x_priv { struct regmap *regmap; + struct clk *xti_clk; struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)]; struct snd_soc_component *component; struct sta32x_platform_data *pdata; @@ -882,6 +884,15 @@ static int sta32x_probe(struct snd_soc_component *component) sta32x->component = component; + if (sta32x->xti_clk) { + ret = clk_prepare_enable(sta32x->xti_clk); + if (ret != 0) { + dev_err(component->dev, + "Failed to enable clock: %d\n", ret); + return ret; + } + } + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); if (ret != 0) { @@ -984,6 +995,9 @@ static void sta32x_remove(struct snd_soc_component *component) sta32x_watchdog_stop(sta32x); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + if (sta32x->xti_clk) + clk_disable_unprepare(sta32x->xti_clk); } static const struct snd_soc_component_driver sta32x_component = { @@ -1100,6 +1114,17 @@ static int sta32x_i2c_probe(struct i2c_client *i2c, } #endif + /* Clock */ + sta32x->xti_clk = devm_clk_get(dev, "xti"); + if (IS_ERR(sta32x->xti_clk)) { + ret = PTR_ERR(sta32x->xti_clk); + + if (ret == -EPROBE_DEFER) + return ret; + + sta32x->xti_clk = NULL; + } + /* GPIOs */ sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_LOW); -- 2.17.2

2 1

[alsa-devel] Applied "ASoC: sun4i-i2s: Add compatibility with A64 codec I2S" to the asoc tree
by Mark Brown 17 Oct '18

17 Oct '18

The patch ASoC: sun4i-i2s: Add compatibility with A64 codec I2S has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 66ecce3325383c8304063e7d5a30f4374ef5a33e Mon Sep 17 00:00:00 2001 From: Marcus Cooper <codekipper(a)gmail.com> Date: Wed, 17 Oct 2018 00:38:05 -0700 Subject: [PATCH] ASoC: sun4i-i2s: Add compatibility with A64 codec I2S The I2S block used for the audio codec in the A64 differs from other 3 I2S modules in A64 and isn't compatible with H3. But it is very similar to what is found in A10(sun4i). However, its TX FIFO is located at a different address. Signed-off-by: Marcus Cooper <codekipper(a)gmail.com> Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Acked-by: Maxime Ripard <maxime.ripard(a)bootlin.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/sun4i-i2s.txt | 2 ++ sound/soc/sunxi/sun4i-i2s.c | 21 +++++++++++++++++++ 2 files changed, 23 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt index b9d50d6cdef3..61e71c1729e0 100644 --- a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt +++ b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt @@ -10,6 +10,7 @@ Required properties: - "allwinner,sun6i-a31-i2s" - "allwinner,sun8i-a83t-i2s" - "allwinner,sun8i-h3-i2s" + - "allwinner,sun50i-a64-codec-i2s" - reg: physical base address of the controller and length of memory mapped region. - interrupts: should contain the I2S interrupt. @@ -26,6 +27,7 @@ Required properties for the following compatibles: - "allwinner,sun6i-a31-i2s" - "allwinner,sun8i-a83t-i2s" - "allwinner,sun8i-h3-i2s" + - "allwinner,sun50i-a64-codec-i2s" - resets: phandle to the reset line for this codec Example: diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index a4aa931ebfae..c63d226e2436 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -961,6 +961,23 @@ static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = { .field_rxchansel = REG_FIELD(SUN8I_I2S_RX_CHAN_SEL_REG, 0, 2), }; +static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = { + .has_reset = true, + .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, + .sun4i_i2s_regmap = &sun4i_i2s_regmap_config, + .has_slave_select_bit = true, + .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7), + .field_fmt_wss = REG_FIELD(SUN4I_I2S_FMT0_REG, 2, 3), + .field_fmt_sr = REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 5), + .field_fmt_bclk = REG_FIELD(SUN4I_I2S_FMT0_REG, 6, 6), + .field_fmt_lrclk = REG_FIELD(SUN4I_I2S_FMT0_REG, 7, 7), + .field_fmt_mode = REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 1), + .field_txchanmap = REG_FIELD(SUN4I_I2S_TX_CHAN_MAP_REG, 0, 31), + .field_rxchanmap = REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31), + .field_txchansel = REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2), + .field_rxchansel = REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2), +}; + static int sun4i_i2s_init_regmap_fields(struct device *dev, struct sun4i_i2s *i2s) { @@ -1169,6 +1186,10 @@ static const struct of_device_id sun4i_i2s_match[] = { .compatible = "allwinner,sun8i-h3-i2s", .data = &sun8i_h3_i2s_quirks, }, + { + .compatible = "allwinner,sun50i-a64-codec-i2s", + .data = &sun50i_a64_codec_i2s_quirks, + }, {} }; MODULE_DEVICE_TABLE(of, sun4i_i2s_match); -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH v5 0/9] Add support for audiocodec in Allwinner A64
by Vasily Khoruzhick 17 Oct '18

17 Oct '18

This series adds Allwinner A64 audiocodec support into sun4i-i2s, sun8i-codec drivers, introduces new sun50i-codec-analog driver and enables sound on Pine64, SoPine boards and Pinebook. I2S for audiocodec in A64 is different from other 3 I2S modules but similar to one in A10, digital part of codec is compatible with A33 and analog controls part is completely different from other SoCs - it shares only few bits in few registers, so adding support for it into existing sun8i-codec-analog would mean duplicating all the widgets, controls and some routes and making it hard to read. Therefore it makes sense to introduce new driver. v2: - Use simple-amplifier for speaker amp on Pinebook - Rename sun50i-a64-i2s to sun50i-a64-codec-i2s to preserve compatible string for other 3 I2S modules in A64 in case if there's any incompatibility with H3 v3: - renamed sunxi-adda-pr-regmap to sun8i-adda-pr-regmap - use ilog2() to calculate reg value for LRCK div instead of using a table v4: - dts: don't use 'Mic' and 'Headset Mic' widgets from sun8i-codec, define our board-level widgets instead. v5: - collect all the tags - drop the change to sun8i_codec_clk_div Marcus Cooper (1): ASoC: sun4i-i2s: Add compatibility with A64 codec I2S Vasily Khoruzhick (8): ASoC: sun8i-codec: Don't hardcode BCLK / LRCK ratio ASoC: sun8i-codec-analog: split regmap code into separate driver ASoC: dt-binding: Add bindings for Allwinner A64 codec's analog path controls ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64 arm64: dts: allwinner: a64: add nodes necessary for analog sound support arm64: dts: allwinner: a64: enable sound on Pine64 and SoPine arm64: dts: allwinner: a64: enable sound on Pinebook .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + .../bindings/sound/sun50i-codec-analog.txt | 12 + .../boot/dts/allwinner/sun50i-a64-pine64.dts | 29 ++ .../dts/allwinner/sun50i-a64-pinebook.dts | 42 ++ .../allwinner/sun50i-a64-sopine-baseboard.dts | 29 ++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 58 +++ sound/soc/sunxi/Kconfig | 17 +- sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-i2s.c | 21 + sound/soc/sunxi/sun50i-codec-analog.c | 444 ++++++++++++++++++ sound/soc/sunxi/sun8i-adda-pr-regmap.c | 102 ++++ sound/soc/sunxi/sun8i-adda-pr-regmap.h | 7 + sound/soc/sunxi/sun8i-codec-analog.c | 79 +--- sound/soc/sunxi/sun8i-codec.c | 22 +- 14 files changed, 785 insertions(+), 81 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt create mode 100644 sound/soc/sunxi/sun50i-codec-analog.c create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.c create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.h -- 2.19.0

3 11

[alsa-devel] Applied "ASoC: sun8i-codec-analog: split regmap code into separate driver" to the asoc tree
by Mark Brown 17 Oct '18

17 Oct '18

The patch ASoC: sun8i-codec-analog: split regmap code into separate driver has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 55b407f6468c481edc4a26b21c9f567ae57f01a4 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick <anarsoul(a)gmail.com> Date: Wed, 17 Oct 2018 00:38:07 -0700 Subject: [PATCH] ASoC: sun8i-codec-analog: split regmap code into separate driver It will be reused by sun50i-codec-analog later. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Acked-by: Maxime Ripard <maxime.ripard(a)bootlin.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sunxi/Kconfig | 7 +- sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun8i-adda-pr-regmap.c | 102 +++++++++++++++++++++++++ sound/soc/sunxi/sun8i-adda-pr-regmap.h | 7 ++ sound/soc/sunxi/sun8i-codec-analog.c | 79 +------------------ 5 files changed, 119 insertions(+), 77 deletions(-) create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.c create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.h diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 22408bc2d6ec..83b770cdfdaa 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -23,7 +23,7 @@ config SND_SUN8I_CODEC config SND_SUN8I_CODEC_ANALOG tristate "Allwinner sun8i Codec Analog Controls Support" depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST - select REGMAP + select SND_SUN8I_ADDA_PR_REGMAP help Say Y or M if you want to add support for the analog controls for the codec embedded in newer Allwinner SoCs. @@ -45,4 +45,9 @@ config SND_SUN4I_SPDIF help Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. + +config SND_SUN8I_ADDA_PR_REGMAP + tristate + select REGMAP + endmenu diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index 4a9ef67386ca..74b99d55cfca 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -4,3 +4,4 @@ obj-$(CONFIG_SND_SUN4I_I2S) += sun4i-i2s.o obj-$(CONFIG_SND_SUN4I_SPDIF) += sun4i-spdif.o obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o +obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o diff --git a/sound/soc/sunxi/sun8i-adda-pr-regmap.c b/sound/soc/sunxi/sun8i-adda-pr-regmap.c new file mode 100644 index 000000000000..e68ce9d2884d --- /dev/null +++ b/sound/soc/sunxi/sun8i-adda-pr-regmap.c @@ -0,0 +1,102 @@ +// SPDX-License-Identifier: GPL-2.0+ +/* + * This driver provides regmap to access to analog part of audio codec + * found on Allwinner A23, A31s, A33, H3 and A64 Socs + * + * Copyright 2016 Chen-Yu Tsai <wens(a)csie.org> + * Copyright (C) 2018 Vasily Khoruzhick <anarsoul(a)gmail.com> + */ + +#include <linux/io.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/regmap.h> + +#include "sun8i-adda-pr-regmap.h" + +/* Analog control register access bits */ +#define ADDA_PR 0x0 /* PRCM base + 0x1c0 */ +#define ADDA_PR_RESET BIT(28) +#define ADDA_PR_WRITE BIT(24) +#define ADDA_PR_ADDR_SHIFT 16 +#define ADDA_PR_ADDR_MASK GENMASK(4, 0) +#define ADDA_PR_DATA_IN_SHIFT 8 +#define ADDA_PR_DATA_IN_MASK GENMASK(7, 0) +#define ADDA_PR_DATA_OUT_SHIFT 0 +#define ADDA_PR_DATA_OUT_MASK GENMASK(7, 0) + +/* regmap access bits */ +static int adda_reg_read(void *context, unsigned int reg, unsigned int *val) +{ + void __iomem *base = (void __iomem *)context; + u32 tmp; + + /* De-assert reset */ + writel(readl(base) | ADDA_PR_RESET, base); + + /* Clear write bit */ + writel(readl(base) & ~ADDA_PR_WRITE, base); + + /* Set register address */ + tmp = readl(base); + tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT); + tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT; + writel(tmp, base); + + /* Read back value */ + *val = readl(base) & ADDA_PR_DATA_OUT_MASK; + + return 0; +} + +static int adda_reg_write(void *context, unsigned int reg, unsigned int val) +{ + void __iomem *base = (void __iomem *)context; + u32 tmp; + + /* De-assert reset */ + writel(readl(base) | ADDA_PR_RESET, base); + + /* Set register address */ + tmp = readl(base); + tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT); + tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT; + writel(tmp, base); + + /* Set data to write */ + tmp = readl(base); + tmp &= ~(ADDA_PR_DATA_IN_MASK << ADDA_PR_DATA_IN_SHIFT); + tmp |= (val & ADDA_PR_DATA_IN_MASK) << ADDA_PR_DATA_IN_SHIFT; + writel(tmp, base); + + /* Set write bit to signal a write */ + writel(readl(base) | ADDA_PR_WRITE, base); + + /* Clear write bit */ + writel(readl(base) & ~ADDA_PR_WRITE, base); + + return 0; +} + +static const struct regmap_config adda_pr_regmap_cfg = { + .name = "adda-pr", + .reg_bits = 5, + .reg_stride = 1, + .val_bits = 8, + .reg_read = adda_reg_read, + .reg_write = adda_reg_write, + .fast_io = true, + .max_register = 31, +}; + +struct regmap *sun8i_adda_pr_regmap_init(struct device *dev, + void __iomem *base) +{ + return devm_regmap_init(dev, NULL, base, &adda_pr_regmap_cfg); +} +EXPORT_SYMBOL_GPL(sun8i_adda_pr_regmap_init); + +MODULE_DESCRIPTION("Allwinner analog audio codec regmap driver"); +MODULE_AUTHOR("Vasily Khoruzhick <anarsoul(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sunxi-adda-pr"); diff --git a/sound/soc/sunxi/sun8i-adda-pr-regmap.h b/sound/soc/sunxi/sun8i-adda-pr-regmap.h new file mode 100644 index 000000000000..a5ae95dfebc1 --- /dev/null +++ b/sound/soc/sunxi/sun8i-adda-pr-regmap.h @@ -0,0 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0+ */ +/* + * Copyright (C) 2018 Vasily Khoruzhick <anarsoul(a)gmail.com> + */ + +struct regmap *sun8i_adda_pr_regmap_init(struct device *dev, + void __iomem *base); diff --git a/sound/soc/sunxi/sun8i-codec-analog.c b/sound/soc/sunxi/sun8i-codec-analog.c index 485e79f292c4..916a46bbc1c8 100644 --- a/sound/soc/sunxi/sun8i-codec-analog.c +++ b/sound/soc/sunxi/sun8i-codec-analog.c @@ -27,6 +27,8 @@ #include <sound/soc-dapm.h> #include <sound/tlv.h> +#include "sun8i-adda-pr-regmap.h" + /* Codec analog control register offsets and bit fields */ #define SUN8I_ADDA_HP_VOLC 0x00 #define SUN8I_ADDA_HP_VOLC_PA_CLK_GATE 7 @@ -120,81 +122,6 @@ #define SUN8I_ADDA_ADC_AP_EN_ADCLEN 6 #define SUN8I_ADDA_ADC_AP_EN_ADCG 0 -/* Analog control register access bits */ -#define ADDA_PR 0x0 /* PRCM base + 0x1c0 */ -#define ADDA_PR_RESET BIT(28) -#define ADDA_PR_WRITE BIT(24) -#define ADDA_PR_ADDR_SHIFT 16 -#define ADDA_PR_ADDR_MASK GENMASK(4, 0) -#define ADDA_PR_DATA_IN_SHIFT 8 -#define ADDA_PR_DATA_IN_MASK GENMASK(7, 0) -#define ADDA_PR_DATA_OUT_SHIFT 0 -#define ADDA_PR_DATA_OUT_MASK GENMASK(7, 0) - -/* regmap access bits */ -static int adda_reg_read(void *context, unsigned int reg, unsigned int *val) -{ - void __iomem *base = (void __iomem *)context; - u32 tmp; - - /* De-assert reset */ - writel(readl(base) | ADDA_PR_RESET, base); - - /* Clear write bit */ - writel(readl(base) & ~ADDA_PR_WRITE, base); - - /* Set register address */ - tmp = readl(base); - tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT); - tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT; - writel(tmp, base); - - /* Read back value */ - *val = readl(base) & ADDA_PR_DATA_OUT_MASK; - - return 0; -} - -static int adda_reg_write(void *context, unsigned int reg, unsigned int val) -{ - void __iomem *base = (void __iomem *)context; - u32 tmp; - - /* De-assert reset */ - writel(readl(base) | ADDA_PR_RESET, base); - - /* Set register address */ - tmp = readl(base); - tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT); - tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT; - writel(tmp, base); - - /* Set data to write */ - tmp = readl(base); - tmp &= ~(ADDA_PR_DATA_IN_MASK << ADDA_PR_DATA_IN_SHIFT); - tmp |= (val & ADDA_PR_DATA_IN_MASK) << ADDA_PR_DATA_IN_SHIFT; - writel(tmp, base); - - /* Set write bit to signal a write */ - writel(readl(base) | ADDA_PR_WRITE, base); - - /* Clear write bit */ - writel(readl(base) & ~ADDA_PR_WRITE, base); - - return 0; -} - -static const struct regmap_config adda_pr_regmap_cfg = { - .name = "adda-pr", - .reg_bits = 5, - .reg_stride = 1, - .val_bits = 8, - .reg_read = adda_reg_read, - .reg_write = adda_reg_write, - .fast_io = true, - .max_register = 24, -}; - /* mixer controls */ static const struct snd_kcontrol_new sun8i_codec_mixer_controls[] = { SOC_DAPM_DOUBLE_R("DAC Playback Switch", @@ -912,7 +839,7 @@ static int sun8i_codec_analog_probe(struct platform_device *pdev) return PTR_ERR(base); } - regmap = devm_regmap_init(&pdev->dev, NULL, base, &adda_pr_regmap_cfg); + regmap = sun8i_adda_pr_regmap_init(&pdev->dev, base); if (IS_ERR(regmap)) { dev_err(&pdev->dev, "Failed to create regmap\n"); return PTR_ERR(regmap); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: dt-binding: Add bindings for Allwinner A64 codec's analog path controls" to the asoc tree
by Mark Brown 17 Oct '18

17 Oct '18

The patch ASoC: dt-binding: Add bindings for Allwinner A64 codec's analog path controls has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From af2c06c41970eb0110a03696fd7f8f07854719db Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick <anarsoul(a)gmail.com> Date: Wed, 17 Oct 2018 00:38:08 -0700 Subject: [PATCH] ASoC: dt-binding: Add bindings for Allwinner A64 codec's analog path controls The internal codec on Allwinner A64 is split into 2 parts. The analog path controls are routed through an embedded custom register bus accessed through the PRCM block just as on A23/A33/H3. Add a binding for this hardware. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Acked-by: Maxime Ripard <maxime.ripard(a)bootlin.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/sun50i-codec-analog.txt | 12 ++++++++++++ 1 file changed, 12 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt diff --git a/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt b/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt new file mode 100644 index 000000000000..4f8ad0e04d20 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt @@ -0,0 +1,12 @@ +* Allwinner A64 Codec Analog Controls + +Required properties: +- compatible: must be one of the following compatibles: + - "allwinner,sun50i-a64-codec-analog" +- reg: must contain the registers location and length + +Example: + codec_analog: codec-analog@1f015c0 { + compatible = "allwinner,sun50i-a64-codec-analog"; + reg = <0x01f015c0 0x4>; + }; -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls" to the asoc tree
by Mark Brown 17 Oct '18

17 Oct '18

The patch ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 42371f327df0b8e9d479b929c5cd301846dd0f70 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick <anarsoul(a)gmail.com> Date: Wed, 17 Oct 2018 00:38:09 -0700 Subject: [PATCH] ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls The internal codec on A64 is split into 2 parts. The analog path controls are routed through an embedded custom register bus accessed through the PRCM block. Add an ASoC component driver for it. This should be tied to the codec audio card as an auxiliary device. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Acked-by: Maxime Ripard <maxime.ripard(a)bootlin.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-codec-analog.c | 444 ++++++++++++++++++++++++++ 3 files changed, 453 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-codec-analog.c diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 83b770cdfdaa..8a055ca1819a 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -28,6 +28,14 @@ config SND_SUN8I_CODEC_ANALOG Say Y or M if you want to add support for the analog controls for the codec embedded in newer Allwinner SoCs. +config SND_SUN50I_CODEC_ANALOG + tristate "Allwinner sun50i Codec Analog Controls Support" + depends on (ARM64 && ARCH_SUNXI) || COMPILE_TEST + select SND_SUNXI_ADDA_PR_REGMAP + help + Say Y or M if you want to add support for the analog controls for + the codec embedded in Allwinner A64 SoC. + config SND_SUN4I_I2S tristate "Allwinner A10 I2S Support" select SND_SOC_GENERIC_DMAENGINE_PCM diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index 74b99d55cfca..a86be340a076 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -3,5 +3,6 @@ obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o obj-$(CONFIG_SND_SUN4I_I2S) += sun4i-i2s.o obj-$(CONFIG_SND_SUN4I_SPDIF) += sun4i-spdif.o obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o +obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c new file mode 100644 index 000000000000..8f5f999df631 --- /dev/null +++ b/sound/soc/sunxi/sun50i-codec-analog.c @@ -0,0 +1,444 @@ +// SPDX-License-Identifier: GPL-2.0+ +/* + * This driver supports the analog controls for the internal codec + * found in Allwinner's A64 SoC. + * + * Copyright (C) 2016 Chen-Yu Tsai <wens(a)csie.org> + * Copyright (C) 2017 Marcus Cooper <codekipper(a)gmail.com> + * Copyright (C) 2018 Vasily Khoruzhick <anarsoul(a)gmail.com> + * + * Based on sun8i-codec-analog.c + * + */ + +#include <linux/io.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "sun8i-adda-pr-regmap.h" + +/* Codec analog control register offsets and bit fields */ +#define SUN50I_ADDA_HP_CTRL 0x00 +#define SUN50I_ADDA_HP_CTRL_PA_CLK_GATE 7 +#define SUN50I_ADDA_HP_CTRL_HPPA_EN 6 +#define SUN50I_ADDA_HP_CTRL_HPVOL 0 + +#define SUN50I_ADDA_OL_MIX_CTRL 0x01 +#define SUN50I_ADDA_OL_MIX_CTRL_MIC1 6 +#define SUN50I_ADDA_OL_MIX_CTRL_MIC2 5 +#define SUN50I_ADDA_OL_MIX_CTRL_PHONE 4 +#define SUN50I_ADDA_OL_MIX_CTRL_PHONEN 3 +#define SUN50I_ADDA_OL_MIX_CTRL_LINEINL 2 +#define SUN50I_ADDA_OL_MIX_CTRL_DACL 1 +#define SUN50I_ADDA_OL_MIX_CTRL_DACR 0 + +#define SUN50I_ADDA_OR_MIX_CTRL 0x02 +#define SUN50I_ADDA_OR_MIX_CTRL_MIC1 6 +#define SUN50I_ADDA_OR_MIX_CTRL_MIC2 5 +#define SUN50I_ADDA_OR_MIX_CTRL_PHONE 4 +#define SUN50I_ADDA_OR_MIX_CTRL_PHONEP 3 +#define SUN50I_ADDA_OR_MIX_CTRL_LINEINR 2 +#define SUN50I_ADDA_OR_MIX_CTRL_DACR 1 +#define SUN50I_ADDA_OR_MIX_CTRL_DACL 0 + +#define SUN50I_ADDA_LINEOUT_CTRL0 0x05 +#define SUN50I_ADDA_LINEOUT_CTRL0_LEN 7 +#define SUN50I_ADDA_LINEOUT_CTRL0_REN 6 +#define SUN50I_ADDA_LINEOUT_CTRL0_LSRC_SEL 5 +#define SUN50I_ADDA_LINEOUT_CTRL0_RSRC_SEL 4 + +#define SUN50I_ADDA_LINEOUT_CTRL1 0x06 +#define SUN50I_ADDA_LINEOUT_CTRL1_VOL 0 + +#define SUN50I_ADDA_MIC1_CTRL 0x07 +#define SUN50I_ADDA_MIC1_CTRL_MIC1G 4 +#define SUN50I_ADDA_MIC1_CTRL_MIC1AMPEN 3 +#define SUN50I_ADDA_MIC1_CTRL_MIC1BOOST 0 + +#define SUN50I_ADDA_MIC2_CTRL 0x08 +#define SUN50I_ADDA_MIC2_CTRL_MIC2G 4 +#define SUN50I_ADDA_MIC2_CTRL_MIC2AMPEN 3 +#define SUN50I_ADDA_MIC2_CTRL_MIC2BOOST 0 + +#define SUN50I_ADDA_LINEIN_CTRL 0x09 +#define SUN50I_ADDA_LINEIN_CTRL_LINEING 0 + +#define SUN50I_ADDA_MIX_DAC_CTRL 0x0a +#define SUN50I_ADDA_MIX_DAC_CTRL_DACAREN 7 +#define SUN50I_ADDA_MIX_DAC_CTRL_DACALEN 6 +#define SUN50I_ADDA_MIX_DAC_CTRL_RMIXEN 5 +#define SUN50I_ADDA_MIX_DAC_CTRL_LMIXEN 4 +#define SUN50I_ADDA_MIX_DAC_CTRL_RHPPAMUTE 3 +#define SUN50I_ADDA_MIX_DAC_CTRL_LHPPAMUTE 2 +#define SUN50I_ADDA_MIX_DAC_CTRL_RHPIS 1 +#define SUN50I_ADDA_MIX_DAC_CTRL_LHPIS 0 + +#define SUN50I_ADDA_L_ADCMIX_SRC 0x0b +#define SUN50I_ADDA_L_ADCMIX_SRC_MIC1 6 +#define SUN50I_ADDA_L_ADCMIX_SRC_MIC2 5 +#define SUN50I_ADDA_L_ADCMIX_SRC_PHONE 4 +#define SUN50I_ADDA_L_ADCMIX_SRC_PHONEN 3 +#define SUN50I_ADDA_L_ADCMIX_SRC_LINEINL 2 +#define SUN50I_ADDA_L_ADCMIX_SRC_OMIXRL 1 +#define SUN50I_ADDA_L_ADCMIX_SRC_OMIXRR 0 + +#define SUN50I_ADDA_R_ADCMIX_SRC 0x0c +#define SUN50I_ADDA_R_ADCMIX_SRC_MIC1 6 +#define SUN50I_ADDA_R_ADCMIX_SRC_MIC2 5 +#define SUN50I_ADDA_R_ADCMIX_SRC_PHONE 4 +#define SUN50I_ADDA_R_ADCMIX_SRC_PHONEP 3 +#define SUN50I_ADDA_R_ADCMIX_SRC_LINEINR 2 +#define SUN50I_ADDA_R_ADCMIX_SRC_OMIXR 1 +#define SUN50I_ADDA_R_ADCMIX_SRC_OMIXL 0 + +#define SUN50I_ADDA_ADC_CTRL 0x0d +#define SUN50I_ADDA_ADC_CTRL_ADCREN 7 +#define SUN50I_ADDA_ADC_CTRL_ADCLEN 6 +#define SUN50I_ADDA_ADC_CTRL_ADCG 0 + +#define SUN50I_ADDA_HS_MBIAS_CTRL 0x0e +#define SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN 7 + +#define SUN50I_ADDA_JACK_MIC_CTRL 0x1d +#define SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN 5 + +/* mixer controls */ +static const struct snd_kcontrol_new sun50i_a64_codec_mixer_controls[] = { + SOC_DAPM_DOUBLE_R("DAC Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_DACL, 1, 0), + SOC_DAPM_DOUBLE_R("DAC Reversed Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_DACR, 1, 0), + SOC_DAPM_DOUBLE_R("Line In Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_LINEINL, 1, 0), + SOC_DAPM_DOUBLE_R("Mic1 Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_MIC1, 1, 0), + SOC_DAPM_DOUBLE_R("Mic2 Playback Switch", + SUN50I_ADDA_OL_MIX_CTRL, + SUN50I_ADDA_OR_MIX_CTRL, + SUN50I_ADDA_OL_MIX_CTRL_MIC2, 1, 0), +}; + +/* ADC mixer controls */ +static const struct snd_kcontrol_new sun50i_codec_adc_mixer_controls[] = { + SOC_DAPM_DOUBLE_R("Mixer Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_OMIXRL, 1, 0), + SOC_DAPM_DOUBLE_R("Mixer Reversed Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_OMIXRR, 1, 0), + SOC_DAPM_DOUBLE_R("Line In Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_LINEINL, 1, 0), + SOC_DAPM_DOUBLE_R("Mic1 Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_MIC1, 1, 0), + SOC_DAPM_DOUBLE_R("Mic2 Capture Switch", + SUN50I_ADDA_L_ADCMIX_SRC, + SUN50I_ADDA_R_ADCMIX_SRC, + SUN50I_ADDA_L_ADCMIX_SRC_MIC2, 1, 0), +}; + +static const DECLARE_TLV_DB_SCALE(sun50i_codec_out_mixer_pregain_scale, + -450, 150, 0); +static const DECLARE_TLV_DB_RANGE(sun50i_codec_mic_gain_scale, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0), +); + +static const DECLARE_TLV_DB_SCALE(sun50i_codec_hp_vol_scale, -6300, 100, 1); + +static const DECLARE_TLV_DB_RANGE(sun50i_codec_lineout_vol_scale, + 0, 1, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + 2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0), +); + + +/* volume / mute controls */ +static const struct snd_kcontrol_new sun50i_a64_codec_controls[] = { + SOC_SINGLE_TLV("Headphone Playback Volume", + SUN50I_ADDA_HP_CTRL, + SUN50I_ADDA_HP_CTRL_HPVOL, 0x3f, 0, + sun50i_codec_hp_vol_scale), + + SOC_DOUBLE("Headphone Playback Switch", + SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_LHPPAMUTE, + SUN50I_ADDA_MIX_DAC_CTRL_RHPPAMUTE, 1, 0), + + /* Mixer pre-gain */ + SOC_SINGLE_TLV("Mic1 Playback Volume", SUN50I_ADDA_MIC1_CTRL, + SUN50I_ADDA_MIC1_CTRL_MIC1G, + 0x7, 0, sun50i_codec_out_mixer_pregain_scale), + + /* Microphone Amp boost gain */ + SOC_SINGLE_TLV("Mic1 Boost Volume", SUN50I_ADDA_MIC1_CTRL, + SUN50I_ADDA_MIC1_CTRL_MIC1BOOST, 0x7, 0, + sun50i_codec_mic_gain_scale), + + /* Mixer pre-gain */ + SOC_SINGLE_TLV("Mic2 Playback Volume", + SUN50I_ADDA_MIC2_CTRL, SUN50I_ADDA_MIC2_CTRL_MIC2G, + 0x7, 0, sun50i_codec_out_mixer_pregain_scale), + + /* Microphone Amp boost gain */ + SOC_SINGLE_TLV("Mic2 Boost Volume", SUN50I_ADDA_MIC2_CTRL, + SUN50I_ADDA_MIC2_CTRL_MIC2BOOST, 0x7, 0, + sun50i_codec_mic_gain_scale), + + /* ADC */ + SOC_SINGLE_TLV("ADC Gain Capture Volume", SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCG, 0x7, 0, + sun50i_codec_out_mixer_pregain_scale), + + /* Mixer pre-gain */ + SOC_SINGLE_TLV("Line In Playback Volume", SUN50I_ADDA_LINEIN_CTRL, + SUN50I_ADDA_LINEIN_CTRL_LINEING, + 0x7, 0, sun50i_codec_out_mixer_pregain_scale), + + SOC_SINGLE_TLV("Line Out Playback Volume", + SUN50I_ADDA_LINEOUT_CTRL1, + SUN50I_ADDA_LINEOUT_CTRL1_VOL, 0x1f, 0, + sun50i_codec_lineout_vol_scale), + + SOC_DOUBLE("Line Out Playback Switch", + SUN50I_ADDA_LINEOUT_CTRL0, + SUN50I_ADDA_LINEOUT_CTRL0_LEN, + SUN50I_ADDA_LINEOUT_CTRL0_REN, 1, 0), + +}; + +static const char * const sun50i_codec_hp_src_enum_text[] = { + "DAC", "Mixer", +}; + +static SOC_ENUM_DOUBLE_DECL(sun50i_codec_hp_src_enum, + SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_LHPIS, + SUN50I_ADDA_MIX_DAC_CTRL_RHPIS, + sun50i_codec_hp_src_enum_text); + +static const struct snd_kcontrol_new sun50i_codec_hp_src[] = { + SOC_DAPM_ENUM("Headphone Source Playback Route", + sun50i_codec_hp_src_enum), +}; + +static const char * const sun50i_codec_lineout_src_enum_text[] = { + "Stereo", "Mono Differential", +}; + +static SOC_ENUM_DOUBLE_DECL(sun50i_codec_lineout_src_enum, + SUN50I_ADDA_LINEOUT_CTRL0, + SUN50I_ADDA_LINEOUT_CTRL0_LSRC_SEL, + SUN50I_ADDA_LINEOUT_CTRL0_RSRC_SEL, + sun50i_codec_lineout_src_enum_text); + +static const struct snd_kcontrol_new sun50i_codec_lineout_src[] = { + SOC_DAPM_ENUM("Line Out Source Playback Route", + sun50i_codec_lineout_src_enum), +}; + +static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = { + /* DAC */ + SND_SOC_DAPM_DAC("Left DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_DACALEN, 0), + SND_SOC_DAPM_DAC("Right DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_DACAREN, 0), + /* ADC */ + SND_SOC_DAPM_ADC("Left ADC", NULL, SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCLEN, 0), + SND_SOC_DAPM_ADC("Right ADC", NULL, SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCREN, 0), + /* + * Due to this component and the codec belonging to separate DAPM + * contexts, we need to manually link the above widgets to their + * stream widgets at the card level. + */ + + SND_SOC_DAPM_MUX("Headphone Source Playback Route", + SND_SOC_NOPM, 0, 0, sun50i_codec_hp_src), + SND_SOC_DAPM_OUT_DRV("Headphone Amp", SUN50I_ADDA_HP_CTRL, + SUN50I_ADDA_HP_CTRL_HPPA_EN, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("HP"), + + SND_SOC_DAPM_MUX("Line Out Source Playback Route", + SND_SOC_NOPM, 0, 0, sun50i_codec_lineout_src), + SND_SOC_DAPM_OUTPUT("LINEOUT"), + + /* Microphone inputs */ + SND_SOC_DAPM_INPUT("MIC1"), + + /* Microphone Bias */ + SND_SOC_DAPM_SUPPLY("MBIAS", SUN50I_ADDA_HS_MBIAS_CTRL, + SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN, + 0, NULL, 0), + + /* Mic input path */ + SND_SOC_DAPM_PGA("Mic1 Amplifier", SUN50I_ADDA_MIC1_CTRL, + SUN50I_ADDA_MIC1_CTRL_MIC1AMPEN, 0, NULL, 0), + + /* Microphone input */ + SND_SOC_DAPM_INPUT("MIC2"), + + /* Microphone Bias */ + SND_SOC_DAPM_SUPPLY("HBIAS", SUN50I_ADDA_JACK_MIC_CTRL, + SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN, + 0, NULL, 0), + + /* Mic input path */ + SND_SOC_DAPM_PGA("Mic2 Amplifier", SUN50I_ADDA_MIC2_CTRL, + SUN50I_ADDA_MIC2_CTRL_MIC2AMPEN, 0, NULL, 0), + + /* Line input */ + SND_SOC_DAPM_INPUT("LINEIN"), + + /* Mixers */ + SND_SOC_DAPM_MIXER("Left Mixer", SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_LMIXEN, 0, + sun50i_a64_codec_mixer_controls, + ARRAY_SIZE(sun50i_a64_codec_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Mixer", SUN50I_ADDA_MIX_DAC_CTRL, + SUN50I_ADDA_MIX_DAC_CTRL_RMIXEN, 0, + sun50i_a64_codec_mixer_controls, + ARRAY_SIZE(sun50i_a64_codec_mixer_controls)), + SND_SOC_DAPM_MIXER("Left ADC Mixer", SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCLEN, 0, + sun50i_codec_adc_mixer_controls, + ARRAY_SIZE(sun50i_codec_adc_mixer_controls)), + SND_SOC_DAPM_MIXER("Right ADC Mixer", SUN50I_ADDA_ADC_CTRL, + SUN50I_ADDA_ADC_CTRL_ADCREN, 0, + sun50i_codec_adc_mixer_controls, + ARRAY_SIZE(sun50i_codec_adc_mixer_controls)), +}; + +static const struct snd_soc_dapm_route sun50i_a64_codec_routes[] = { + /* Left Mixer Routes */ + { "Left Mixer", "DAC Playback Switch", "Left DAC" }, + { "Left Mixer", "DAC Reversed Playback Switch", "Right DAC" }, + { "Left Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" }, + + /* Right Mixer Routes */ + { "Right Mixer", "DAC Playback Switch", "Right DAC" }, + { "Right Mixer", "DAC Reversed Playback Switch", "Left DAC" }, + { "Right Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" }, + + /* Left ADC Mixer Routes */ + { "Left ADC Mixer", "Mixer Capture Switch", "Left Mixer" }, + { "Left ADC Mixer", "Mixer Reversed Capture Switch", "Right Mixer" }, + { "Left ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" }, + + /* Right ADC Mixer Routes */ + { "Right ADC Mixer", "Mixer Capture Switch", "Right Mixer" }, + { "Right ADC Mixer", "Mixer Reversed Capture Switch", "Left Mixer" }, + { "Right ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" }, + + /* ADC Routes */ + { "Left ADC", NULL, "Left ADC Mixer" }, + { "Right ADC", NULL, "Right ADC Mixer" }, + + /* Headphone Routes */ + { "Headphone Source Playback Route", "DAC", "Left DAC" }, + { "Headphone Source Playback Route", "DAC", "Right DAC" }, + { "Headphone Source Playback Route", "Mixer", "Left Mixer" }, + { "Headphone Source Playback Route", "Mixer", "Right Mixer" }, + { "Headphone Amp", NULL, "Headphone Source Playback Route" }, + { "HP", NULL, "Headphone Amp" }, + + /* Microphone Routes */ + { "Mic1 Amplifier", NULL, "MIC1"}, + + /* Microphone Routes */ + { "Mic2 Amplifier", NULL, "MIC2"}, + { "Left Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" }, + { "Right Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" }, + { "Left ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" }, + { "Right ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" }, + + /* Line-in Routes */ + { "Left Mixer", "Line In Playback Switch", "LINEIN" }, + { "Right Mixer", "Line In Playback Switch", "LINEIN" }, + { "Left ADC Mixer", "Line In Capture Switch", "LINEIN" }, + { "Right ADC Mixer", "Line In Capture Switch", "LINEIN" }, + + /* Line-out Routes */ + { "Line Out Source Playback Route", "Stereo", "Left Mixer" }, + { "Line Out Source Playback Route", "Stereo", "Right Mixer" }, + { "Line Out Source Playback Route", "Mono Differential", "Left Mixer" }, + { "Line Out Source Playback Route", "Mono Differential", + "Right Mixer" }, + { "LINEOUT", NULL, "Line Out Source Playback Route" }, +}; + +static const struct snd_soc_component_driver sun50i_codec_analog_cmpnt_drv = { + .controls = sun50i_a64_codec_controls, + .num_controls = ARRAY_SIZE(sun50i_a64_codec_controls), + .dapm_widgets = sun50i_a64_codec_widgets, + .num_dapm_widgets = ARRAY_SIZE(sun50i_a64_codec_widgets), + .dapm_routes = sun50i_a64_codec_routes, + .num_dapm_routes = ARRAY_SIZE(sun50i_a64_codec_routes), +}; + +static const struct of_device_id sun50i_codec_analog_of_match[] = { + { + .compatible = "allwinner,sun50i-a64-codec-analog", + }, + {} +}; +MODULE_DEVICE_TABLE(of, sun50i_codec_analog_of_match); + +static int sun50i_codec_analog_probe(struct platform_device *pdev) +{ + struct resource *res; + struct regmap *regmap; + void __iomem *base; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) { + dev_err(&pdev->dev, "Failed to map the registers\n"); + return PTR_ERR(base); + } + + regmap = sun8i_adda_pr_regmap_init(&pdev->dev, base); + if (IS_ERR(regmap)) { + dev_err(&pdev->dev, "Failed to create regmap\n"); + return PTR_ERR(regmap); + } + + return devm_snd_soc_register_component(&pdev->dev, + &sun50i_codec_analog_cmpnt_drv, + NULL, 0); +} + +static struct platform_driver sun50i_codec_analog_driver = { + .driver = { + .name = "sun50i-codec-analog", + .of_match_table = sun50i_codec_analog_of_match, + }, + .probe = sun50i_codec_analog_probe, +}; +module_platform_driver(sun50i_codec_analog_driver); + +MODULE_DESCRIPTION("Allwinner internal codec analog controls driver for A64"); +MODULE_AUTHOR("Vasily Khoruzhick <anarsoul(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-codec-analog"); -- 2.19.0.rc2

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[alsa-devel] Applied "ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64" to the asoc tree
by Mark Brown 17 Oct '18

17 Oct '18

The patch ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64 has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7e95aac96b554193760aaeb64e263da58127bf27 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick <anarsoul(a)gmail.com> Date: Wed, 17 Oct 2018 00:38:10 -0700 Subject: [PATCH] ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64 Allwinner A64 uses the same digital codec part as in A33, so we need to build this driver on ARM64 as well. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> Acked-by: Maxime Ripard <maxime.ripard(a)bootlin.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sunxi/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 8a055ca1819a..66aad0d3f9c7 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -12,7 +12,7 @@ config SND_SUN4I_CODEC config SND_SUN8I_CODEC tristate "Allwinner SUN8I audio codec" depends on OF - depends on MACH_SUN8I || COMPILE_TEST + depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST select REGMAP_MMIO help This option enables the digital part of the internal audio codec for -- 2.19.0.rc2

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[alsa-devel] [PATCH v4 0/9] Add support for audiocodec in Allwinner A64
by Vasily Khoruzhick 17 Oct '18

17 Oct '18

This series adds Allwinner A64 audiocodec support into sun4i-i2s, sun8i-codec drivers, introduces new sun50i-codec-analog driver and enables sound on Pine64, SoPine boards and Pinebook. I2S for audiocodec in A64 is different from other 3 I2S modules but similar to one in A10, digital part of codec is compatible with A33 and analog controls part is completely different from other SoCs - it shares only few bits in few registers, so adding support for it into existing sun8i-codec-analog would mean duplicating all the widgets, controls and some routes and making it hard to read. Therefore it makes sense to introduce new driver. v2: - Use simple-amplifier for speaker amp on Pinebook - Rename sun50i-a64-i2s to sun50i-a64-codec-i2s to preserve compatible string for other 3 I2S modules in A64 in case if there's any incompatibility with H3 v3: - renamed sunxi-adda-pr-regmap to sun8i-adda-pr-regmap - use ilog2() to calculate reg value for LRCK div instead of using a table v4: - dts: don't use 'Mic' and 'Headset Mic' widgets from sun8i-codec, define our board-level widgets instead. Marcus Cooper (1): ASoC: sun4i-i2s: Add compatibility with A64 codec I2S Vasily Khoruzhick (8): ASoC: sun8i-codec: Don't hardcode BCLK / LRCK ratio ASoC: sun8i-codec-analog: split regmap code into separate driver ASoC: dt-binding: Add bindings for Allwinner A64 codec's analog path controls ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64 arm64: dts: allwinner: a64: add nodes necessary for analog sound support arm64: dts: allwinner: a64: enable sound on Pine64 and SoPine arm64: dts: allwinner: a64: enable sound on Pinebook .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + .../bindings/sound/sun50i-codec-analog.txt | 12 + .../boot/dts/allwinner/sun50i-a64-pine64.dts | 29 ++ .../dts/allwinner/sun50i-a64-pinebook.dts | 42 ++ .../allwinner/sun50i-a64-sopine-baseboard.dts | 29 ++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 58 +++ sound/soc/sunxi/Kconfig | 17 +- sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-i2s.c | 21 + sound/soc/sunxi/sun50i-codec-analog.c | 444 ++++++++++++++++++ sound/soc/sunxi/sun8i-adda-pr-regmap.c | 102 ++++ sound/soc/sunxi/sun8i-adda-pr-regmap.h | 7 + sound/soc/sunxi/sun8i-codec-analog.c | 79 +--- sound/soc/sunxi/sun8i-codec.c | 26 +- 14 files changed, 787 insertions(+), 83 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt create mode 100644 sound/soc/sunxi/sun50i-codec-analog.c create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.c create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.h -- 2.19.0

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[alsa-devel] [PATCH] ASoC: rsnd: enable TDM settings for SSI parent
by Kuninori Morimoto 17 Oct '18

17 Oct '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Some SSIs are sharing each pins (= WS/CLK pin for playback/capture). Then, SSI parent needs control WS/CLK setting for SSI slave. In such case, SSI parent needs TDM settings if SSI slave is working as TDM mode. But it is not cared in current driver. It can't capture TDM sound without this patch if SSIs were pin sharing. This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 with TDM sound. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 3adcc4f..b42a0e0 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -400,6 +400,17 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_own |= DEL; /* + * TDM Mode + * see + * rsnd_ssiu_init_gen2() + */ + wsr = ssi->wsr; + if (is_tdm) { + wsr |= WS_MODE; + cr_own |= CHNL_8; + } + + /* * We shouldn't exchange SWSP after running. * This means, parent needs to care it. */ @@ -429,16 +440,6 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_mode = DIEN; /* PIO : enable Data interrupt */ } - /* - * TDM Extend Mode - * see - * rsnd_ssiu_init_gen2() - */ - wsr = ssi->wsr; - if (is_tdm) { - wsr |= WS_MODE; - cr_own |= CHNL_8; - } init_end: ssi->cr_own = cr_own; ssi->cr_mode = cr_mode; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: wm8904: fix spelling mistake "Inveting" -> "Inverting"
by Colin King 16 Oct '18

16 Oct '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in sound control text Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/wm8904.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 2a3e5fbd04e4..ef83dae6e5ae 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -870,7 +870,7 @@ static SOC_ENUM_SINGLE_DECL(lin_inv_enum, WM8904_ANALOGUE_LEFT_INPUT_1, 4, lin_text); static const struct snd_kcontrol_new lin_inv_mux = - SOC_DAPM_ENUM("Left Capture Inveting Mux", lin_inv_enum); + SOC_DAPM_ENUM("Left Capture Inverting Mux", lin_inv_enum); static const char *rin_text[] = { "IN1R", "IN2R", "IN3R" @@ -886,7 +886,7 @@ static SOC_ENUM_SINGLE_DECL(rin_inv_enum, WM8904_ANALOGUE_RIGHT_INPUT_1, 4, rin_text); static const struct snd_kcontrol_new rin_inv_mux = - SOC_DAPM_ENUM("Right Capture Inveting Mux", rin_inv_enum); + SOC_DAPM_ENUM("Right Capture Inverting Mux", rin_inv_enum); static const char *aif_text[] = { "Left", "Right" -- 2.17.1

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[alsa-devel] [ANNOUNCE] 2018 Linux Audio miniconference
by Mark Brown 16 Oct '18

16 Oct '18

Hi, As in previous years we're trying to organize an audio miniconference so we can get together and talk through issues, especially design decisons, face to face. This year's event will be held on Sunday October 21st in Edinburgh, the day before ELC Europe starts there. We're still finalizing the venue (we expect to be able to confirm next week) but it'll be outside the ELC venue so there will be no need to purchase an ELC ticket to attend. As with previous years let's pull together an agenda through a mailing list discussion - if people could reply to this mail with any topics they'd like to discuss we can take it from there. Of course if we can sort things out more quickly via the mailing list that's even better! If you're planning to attend please fill out the form here: https://docs.google.com/spreadsheets/d/11p2pSNxS7sJysg1JwErWi0T2Dq7XKRtAGng… Thanks, Mark

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[alsa-devel] [PATCH] ALSA: hda: Add 2 more models to the power_save blacklist
by Hans de Goede 16 Oct '18

16 Oct '18

Power-saving is causing plops on audio start/stop on Dell Precision T3600 laptops and Intel DZ77BH boards, add these to the power_save blacklist. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Note both these models were reported through: https://bugzilla.redhat.com/show_bug.cgi?id=1525104 as usual alsa-info.sh output for both is available as attachment there. --- sound/pci/hda/hda_intel.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index aa4c672dbaf7..454ef6feee3a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2257,8 +2257,12 @@ static struct snd_pci_quirk power_save_blacklist[] = { /* https://bugzilla.redhat.com/show_bug.cgi?id=1581607 */ SND_PCI_QUIRK(0x1558, 0x3501, "Clevo W35xSS_370SS", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1028, 0x0497, "Dell Precision T3600", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ /* Note the P55A-UD3 and Z87-D3HP share the subsys id for the HDA dev */ SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P55A-UD3 / Z87-D3HP", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x8086, 0x2040, "Intel DZ77BH-55K", 0), /* https://bugzilla.kernel.org/show_bug.cgi?id=199607 */ SND_PCI_QUIRK(0x8086, 0x2057, "Intel NUC5i7RYB", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1520902 */ -- 2.19.0

2 1

[alsa-devel] [PATCH v3 0/9] Add support for audiocodec in Allwinner A64
by Vasily Khoruzhick 15 Oct '18

15 Oct '18

This series adds Allwinner A64 audiocodec support into sun4i-i2s, sun8i-codec drivers, introduces new sun50i-codec-analog driver and enables sound on Pine64, SoPine boards and Pinebook. I2S for audiocodec in A64 is different from other 3 I2S modules but similar to one in A10, digital part of codec is compatible with A33 and analog controls part is completely different from other SoCs - it shares only few bits in few registers, so adding support for it into existing sun8i-codec-analog would mean duplicating all the widgets, controls and some routes and making it hard to read. Therefore it makes sense to introduce new driver. v2: - Use simple-amplifier for speaker amp on Pinebook - Rename sun50i-a64-i2s to sun50i-a64-codec-i2s to preserve compatible string for other 3 I2S modules in A64 in case if there's any incompatibility with H3 v3: - renamed sunxi-adda-pr-regmap to sun8i-adda-pr-regmap - use ilog2() to calculate reg value for LRCK div instead of using a table Marcus Cooper (1): ASoC: sun4i-i2s: Add compatibility with A64 codec I2S Vasily Khoruzhick (8): ASoC: sun8i-codec: Don't hardcode BCLK / LRCK ratio ASoC: sun8i-codec-analog: split regmap code into separate driver ASoC: dt-binding: Add bindings for Allwinner A64 codec's analog path controls ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64 arm64: dts: allwinner: a64: add nodes necessary for analog sound support arm64: dts: allwinner: a64: enable sound on Pine64 and SoPine arm64: dts: allwinner: a64: enable sound on Pinebook .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + .../bindings/sound/sun50i-codec-analog.txt | 12 + .../boot/dts/allwinner/sun50i-a64-pine64.dts | 30 ++ .../dts/allwinner/sun50i-a64-pinebook.dts | 42 ++ .../allwinner/sun50i-a64-sopine-baseboard.dts | 30 ++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 58 +++ sound/soc/sunxi/Kconfig | 17 +- sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-i2s.c | 21 + sound/soc/sunxi/sun50i-codec-analog.c | 444 ++++++++++++++++++ sound/soc/sunxi/sun8i-adda-pr-regmap.c | 102 ++++ sound/soc/sunxi/sun8i-adda-pr-regmap.h | 7 + sound/soc/sunxi/sun8i-codec-analog.c | 79 +--- sound/soc/sunxi/sun8i-codec.c | 26 +- 14 files changed, 789 insertions(+), 83 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt create mode 100644 sound/soc/sunxi/sun50i-codec-analog.c create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.c create mode 100644 sound/soc/sunxi/sun8i-adda-pr-regmap.h -- 2.19.0

1 9

[alsa-devel] [PATCH] ASoC: sun4i-i2s: don't try to start up or shut down DAI if it's active
by Vasily Khoruzhick 15 Oct '18

15 Oct '18

Otherwise we may end up with shutting down I2S if shutdown() was called for capture substream, but playback is still running. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- sound/soc/sunxi/sun4i-i2s.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index a9ea329dc046..787b67c4f845 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -649,6 +649,9 @@ static int sun4i_i2s_startup(struct snd_pcm_substream *substream, { struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); + if (dai->active) + return 0; + /* Enable the whole hardware block */ regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG, SUN4I_I2S_CTRL_GL_EN, SUN4I_I2S_CTRL_GL_EN); @@ -667,6 +670,9 @@ static void sun4i_i2s_shutdown(struct snd_pcm_substream *substream, { struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); + if (dai->active) + return; + clk_disable_unprepare(i2s->mod_clk); /* Disable our output lines */ -- 2.19.0

3 5

[alsa-devel] [PATCH] ASoC: AMD: Add SND_JACK_LINEOUT jack type
by Akshu Agrawal 15 Oct '18

15 Oct '18

Some 3 pole connectors report impedance greater than threshold of 1000Ohm. Thus, da7219 reports them as LINEOUT. Adding the SND_JACK_LINEOUT type so that we don't fail to detect any 3 pole jack type. Also, changing SND_JACK_HEADPHONE | SND_JACK_MICROPHONE -> SND_JACK_HEADSET Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 0d97d00698a0..eced4820b0bc 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -75,7 +75,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) da7219_dai_clk = clk_get(component->dev, "da7219-dai-clks"); ret = snd_soc_card_jack_new(card, "Headset Jack", - SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | + SND_JACK_HEADSET | SND_JACK_LINEOUT | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, &cz_jack, NULL, 0); -- 2.16.4

2 1

[alsa-devel] [PATCH] ASoC: Intel: Boards: Add KBL Dialog Maxim I2S machine driver
by mac.chiang＠intel.com 15 Oct '18

15 Oct '18

From: Mac Chiang <mac.chiang(a)intel.com> This patch adds Kabylake I2S machine driver with: DA7219 audio codec(SSP1) and MAXIM98927(SSP0) speaker amplifier. Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/Kconfig | 13 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/kbl_da7219_max98927.c | 983 +++++++++++++++++++++++++++ 3 files changed, 998 insertions(+) create mode 100644 sound/soc/intel/boards/kbl_da7219_max98927.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 88e4b42..73ca135 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -280,6 +280,19 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH create an alsa sound card for DA7219 + MAX98357A I2S audio codec. Say Y if you have such a device. +config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH + tristate "KBL with DA7219 and MAX98927 in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI + select SND_SOC_DA7219 + select SND_SOC_MAX98927 + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC Onboard Codec I2S machine driver. This will + create an alsa sound card for DA7219 + MAX98927 I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH tristate "SKL/KBL/BXT/APL with HDA Codecs" select SND_SOC_HDAC_HDMI diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 6e88373..5381e27 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -17,6 +17,7 @@ snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o +snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o snd-soc-kbl_rt5663_max98927-objs := kbl_rt5663_max98927.o snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o snd-soc-skl_rt286-objs := skl_rt286.o @@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH) += snd-soc-kbl_da7219_max98357a.o +obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH) += snd-soc-kbl_da7219_max98927.o obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH) += snd-soc-kbl_rt5663_max98927.o obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt5663_rt5514_max98927.o obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c new file mode 100644 index 0000000..3ab96ee --- /dev/null +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -0,0 +1,983 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2018 Intel Corporation. + +/* + * Intel Kabylake I2S Machine Driver with MAX98927 & DA7219 Codecs + * + * Modified from: + * Intel Kabylake I2S Machine driver supporting MAX98927 and + * RT5663 codecs + */ + +#include <linux/input.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../../codecs/da7219.h" +#include "../../codecs/hdac_hdmi.h" +#include "../skylake/skl.h" +#include "../../codecs/da7219-aad.h" + +#define KBL_DIALOG_CODEC_DAI "da7219-hifi" +#define MAX98927_CODEC_DAI "max98927-aif1" +#define MAXIM_DEV0_NAME "i2c-MX98927:00" +#define MAXIM_DEV1_NAME "i2c-MX98927:01" +#define DUAL_CHANNEL 2 +#define QUAD_CHANNEL 4 +#define NAME_SIZE 32 + +static struct snd_soc_card *kabylake_audio_card; +static struct snd_soc_jack kabylake_hdmi[3]; + +struct kbl_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct kbl_codec_private { + struct snd_soc_jack kabylake_headset; + struct list_head hdmi_pcm_list; +}; + +enum { + KBL_DPCM_AUDIO_PB = 0, + KBL_DPCM_AUDIO_CP, + KBL_DPCM_AUDIO_ECHO_REF_CP, + KBL_DPCM_AUDIO_REF_CP, + KBL_DPCM_AUDIO_DMIC_CP, + KBL_DPCM_AUDIO_HDMI1_PB, + KBL_DPCM_AUDIO_HDMI2_PB, + KBL_DPCM_AUDIO_HDMI3_PB, + KBL_DPCM_AUDIO_HS_PB, +}; + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret = 0; + + codec_dai = snd_soc_card_get_codec_dai(card, KBL_DIALOG_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); + return -EIO; + } + + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24576000, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(card->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, + DA7219_SYSCLK_MCLK, 0, 0); + if (ret) + dev_err(card->dev, "failed to stop PLL: %d\n", ret); + } else if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_PLL_SRM, + 0, DA7219_PLL_FREQ_OUT_98304); + if (ret) + dev_err(card->dev, "failed to start PLL: %d\n", ret); + } + + return ret; +} + +static const struct snd_kcontrol_new kabylake_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const struct snd_soc_dapm_widget kabylake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SPK("DP", NULL), + SND_SOC_DAPM_SPK("HDMI", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route kabylake_map[] = { + /* speaker */ + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, + + /* other jacks */ + { "DMic", NULL, "SoC DMIC" }, + + { "HDMI", NULL, "hif5 Output" }, + { "DP", NULL, "hif6 Output" }, + + /* CODEC BE connections */ + { "Left HiFi Playback", NULL, "ssp0 Tx" }, + { "Right HiFi Playback", NULL, "ssp0 Tx" }, + { "ssp0 Tx", NULL, "spk_out" }, + + /* IV feedback path */ + { "codec0_fb_in", NULL, "ssp0 Rx"}, + { "ssp0 Rx", NULL, "Left HiFi Capture" }, + { "ssp0 Rx", NULL, "Right HiFi Capture" }, + + /* AEC capture path */ + { "echo_ref_out", NULL, "ssp0 Rx" }, + + /* DMIC */ + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "DMIC AIF" }, + + { "hifi1", NULL, "iDisp1 Tx" }, + { "iDisp1 Tx", NULL, "iDisp1_out" }, + { "hifi2", NULL, "iDisp2 Tx" }, + { "iDisp2 Tx", NULL, "iDisp2_out" }, + { "hifi3", NULL, "iDisp3 Tx"}, + { "iDisp3 Tx", NULL, "iDisp3_out"}, +}; + +static const struct snd_soc_dapm_route kabylake_ssp1_map[] = { + { "Headphone Jack", NULL, "HPL" }, + { "Headphone Jack", NULL, "HPR" }, + + /* other jacks */ + { "MIC", NULL, "Headset Mic" }, + + /* CODEC BE connections */ + { "Playback", NULL, "ssp1 Tx" }, + { "ssp1 Tx", NULL, "codec1_out" }, + + { "hs_in", NULL, "ssp1 Rx" }, + { "ssp1 Rx", NULL, "Capture" }, + + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, +}; + +static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *runtime = substream->private_data; + int ret = 0, j; + + for (j = 0; j < runtime->num_codecs; j++) { + struct snd_soc_dai *codec_dai = runtime->codec_dais[j]; + + if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); + if (ret < 0) { + dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret); + return ret; + } + } + if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16); + if (ret < 0) { + dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret); + return ret; + } + } + } + + return 0; +} + +static struct snd_soc_ops kabylake_ssp0_ops = { + .hw_params = kabylake_ssp0_hw_params, +}; + +static int kabylake_ssp_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_dpcm *dpcm = container_of( + params, struct snd_soc_dpcm, hw_params); + struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; + struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + + /* + * The ADSP will convert the FE rate to 48k, stereo, 24 bit + */ + if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || + !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + snd_mask_none(fmt); + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + } + + /* + * The speaker on the SSP0 supports S16_LE and not S24_LE. + * thus changing the mask here + */ + if (!strcmp(be_dai_link->name, "SSP0-Codec")) + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S16_LE); + + return 0; +} + +static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_jack *jack; + struct snd_soc_card *card = rtd->card; + int ret; + + + ret = snd_soc_dapm_add_routes(&card->dapm, + kabylake_ssp1_map, + ARRAY_SIZE(kabylake_ssp1_map)); + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(kabylake_audio_card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &ctx->kabylake_headset, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &ctx->kabylake_headset; + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + + da7219_aad_jack_det(component, &ctx->kabylake_headset); + + ret = snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); + if (ret) + dev_err(rtd->dev, "SoC DMIC - Ignore suspend failed %d\n", ret); + + return ret; +} + +static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) +{ + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct kbl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = device; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI1_PB); +} + +static int kabylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI2_PB); +} + +static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) +{ + return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI3_PB); +} + +static int kabylake_da7219_fe_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dapm_context *dapm; + struct snd_soc_component *component = rtd->cpu_dai->component; + + dapm = snd_soc_component_get_dapm(component); + snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); + + return 0; +} + +static const unsigned int rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static unsigned int channels_quad[] = { + QUAD_CHANNEL, +}; + +static struct snd_pcm_hw_constraint_list constraints_channels_quad = { + .count = ARRAY_SIZE(channels_quad), + .list = channels_quad, + .mask = 0, +}; + +static int kbl_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + /* + * On this platform for PCM device we support, + * 48Khz + * stereo + * 16 bit audio + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 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_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + + return 0; +} + +static const struct snd_soc_ops kabylake_da7219_fe_ops = { + .startup = kbl_fe_startup, +}; + +static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* + * set BE channel constraint as user FE channels + */ + + if (params_channels(params) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; + + return 0; +} + +static int kabylake_dmic_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels_quad); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); +} + +static struct snd_soc_ops kabylake_dmic_ops = { + .startup = kabylake_dmic_startup, +}; + +static const unsigned int rates_16000[] = { + 16000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_16000 = { + .count = ARRAY_SIZE(rates_16000), + .list = rates_16000, +}; + +static const unsigned int ch_mono[] = { + 1, +}; +static const struct snd_pcm_hw_constraint_list constraints_refcap = { + .count = ARRAY_SIZE(ch_mono), + .list = ch_mono, +}; + +static int kabylake_refcap_startup(struct snd_pcm_substream *substream) +{ + substream->runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_refcap); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_16000); +} + + +static struct snd_soc_ops skylaye_refcap_ops = { + .startup = kabylake_refcap_startup, +}; + +static struct snd_soc_codec_conf max98927_codec_conf[] = { + + { + .dev_name = MAXIM_DEV0_NAME, + .name_prefix = "Right", + }, + + { + .dev_name = MAXIM_DEV1_NAME, + .name_prefix = "Left", + }, +}; + +static struct snd_soc_dai_link_component ssp0_codec_components[] = { + { /* Left */ + .name = MAXIM_DEV0_NAME, + .dai_name = MAX98927_CODEC_DAI, + }, + + { /* For Right */ + .name = MAXIM_DEV1_NAME, + .dai_name = MAX98927_CODEC_DAI, + }, + +}; + +/* kabylake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link kabylake_dais[] = { + /* Front End DAI links */ + [KBL_DPCM_AUDIO_PB] = { + .name = "Kbl Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .init = kabylake_da7219_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .ops = &kabylake_da7219_fe_ops, + }, + [KBL_DPCM_AUDIO_CP] = { + .name = "Kbl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + .ops = &kabylake_da7219_fe_ops, + }, + [KBL_DPCM_AUDIO_ECHO_REF_CP] = { + .name = "Kbl Audio Echo Reference cap", + .stream_name = "Echoreference Capture", + .cpu_dai_name = "Echoref Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .capture_only = 1, + .nonatomic = 1, + }, + [KBL_DPCM_AUDIO_REF_CP] = { + .name = "Kbl Audio Reference cap", + .stream_name = "Wake on Voice", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &skylaye_refcap_ops, + }, + [KBL_DPCM_AUDIO_DMIC_CP] = { + .name = "Kbl Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &kabylake_dmic_ops, + }, + [KBL_DPCM_AUDIO_HDMI1_PB] = { + .name = "Kbl HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI2_PB] = { + .name = "Kbl HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI3_PB] = { + .name = "Kbl HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HS_PB] = { + .name = "Kbl Audio Headset Playback", + .stream_name = "Headset Audio", + .cpu_dai_name = "System Pin2", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .nonatomic = 1, + .dynamic = 1, + .init = kabylake_da7219_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .ops = &kabylake_da7219_fe_ops, + + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .id = 0, + .cpu_dai_name = "SSP0 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codecs = ssp0_codec_components, + .num_codecs = ARRAY_SIZE(ssp0_codec_components), + .dai_fmt = SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_ssp_fixup, + .ops = &kabylake_ssp0_ops, + }, + { + /* SSP1 - Codec */ + .name = "SSP1-Codec", + .id = 1, + .cpu_dai_name = "SSP1 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codec_name = "i2c-DLGS7219:00", + .codec_dai_name = KBL_DIALOG_CODEC_DAI, + .init = kabylake_da7219_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_ssp_fixup, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .id = 2, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = kabylake_dmic_fixup, + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "iDisp1", + .id = 3, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = kabylake_hdmi1_init, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 4, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi2_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 5, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi3_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +/* kabylake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link kabylake_max98927_dais[] = { + /* Front End DAI links */ + [KBL_DPCM_AUDIO_PB] = { + .name = "Kbl Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .init = kabylake_da7219_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .ops = &kabylake_da7219_fe_ops, + }, + [KBL_DPCM_AUDIO_CP] = { + .name = "Kbl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + .ops = &kabylake_da7219_fe_ops, + }, + [KBL_DPCM_AUDIO_ECHO_REF_CP] = { + .name = "Kbl Audio Echo Reference cap", + .stream_name = "Echoreference Capture", + .cpu_dai_name = "Echoref Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .capture_only = 1, + .nonatomic = 1, + }, + [KBL_DPCM_AUDIO_REF_CP] = { + .name = "Kbl Audio Reference cap", + .stream_name = "Wake on Voice", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &skylaye_refcap_ops, + }, + [KBL_DPCM_AUDIO_DMIC_CP] = { + .name = "Kbl Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &kabylake_dmic_ops, + }, + [KBL_DPCM_AUDIO_HDMI1_PB] = { + .name = "Kbl HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI2_PB] = { + .name = "Kbl HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [KBL_DPCM_AUDIO_HDMI3_PB] = { + .name = "Kbl HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .id = 0, + .cpu_dai_name = "SSP0 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codecs = ssp0_codec_components, + .num_codecs = ARRAY_SIZE(ssp0_codec_components), + .dai_fmt = SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = kabylake_ssp_fixup, + .ops = &kabylake_ssp0_ops, + }, + { + .name = "dmic01", + .id = 1, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = kabylake_dmic_fixup, + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "iDisp1", + .id = 2, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:1f.3", + .dpcm_playback = 1, + .init = kabylake_hdmi1_init, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 3, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi2_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 4, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:1f.3", + .init = kabylake_hdmi3_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +static int kabylake_card_late_probe(struct snd_soc_card *card) +{ + struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); + struct kbl_hdmi_pcm *pcm; + struct snd_soc_component *component = NULL; + int err, i = 0; + char jack_name[NAME_SIZE]; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &kabylake_hdmi[i], + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &kabylake_hdmi[i]); + if (err < 0) + return err; + + i++; + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); + + return 0; +} + +/* kabylake audio machine driver for SPT + DA7219 */ +static struct snd_soc_card kbl_audio_card_da7219_m98927 = { + .name = "kblda7219m98927", + .owner = THIS_MODULE, + .dai_link = kabylake_dais, + .num_links = ARRAY_SIZE(kabylake_dais), + .controls = kabylake_controls, + .num_controls = ARRAY_SIZE(kabylake_controls), + .dapm_widgets = kabylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets), + .dapm_routes = kabylake_map, + .num_dapm_routes = ARRAY_SIZE(kabylake_map), + .codec_conf = max98927_codec_conf, + .num_configs = ARRAY_SIZE(max98927_codec_conf), + .fully_routed = true, + .late_probe = kabylake_card_late_probe, +}; + +/* kabylake audio machine driver for Maxim98927 */ +static struct snd_soc_card kbl_audio_card_max98927 = { + .name = "kblmax98927", + .owner = THIS_MODULE, + .dai_link = kabylake_max98927_dais, + .num_links = ARRAY_SIZE(kabylake_max98927_dais), + .controls = kabylake_controls, + .num_controls = ARRAY_SIZE(kabylake_controls), + .dapm_widgets = kabylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets), + .dapm_routes = kabylake_map, + .num_dapm_routes = ARRAY_SIZE(kabylake_map), + .codec_conf = max98927_codec_conf, + .num_configs = ARRAY_SIZE(max98927_codec_conf), + .fully_routed = true, + .late_probe = kabylake_card_late_probe, +}; + +static int kabylake_audio_probe(struct platform_device *pdev) +{ + struct kbl_codec_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + kabylake_audio_card = + (struct snd_soc_card *)pdev->id_entry->driver_data; + + kabylake_audio_card->dev = &pdev->dev; + snd_soc_card_set_drvdata(kabylake_audio_card, ctx); + + return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card); +} + +static const struct platform_device_id kbl_board_ids[] = { + { + .name = "kbl_da7219_max98927", + .driver_data = + (kernel_ulong_t)&kbl_audio_card_da7219_m98927, + }, + { + .name = "kbl_max98927", + .driver_data = + (kernel_ulong_t)&kbl_audio_card_max98927, + }, + { } +}; + +static struct platform_driver kabylake_audio = { + .probe = kabylake_audio_probe, + .driver = { + .name = "kbl_da7219_max98927", + .pm = &snd_soc_pm_ops, + }, + .id_table = kbl_board_ids, +}; + +module_platform_driver(kabylake_audio) + +/* Module information */ +MODULE_DESCRIPTION("Audio KabyLake Machine driver for MAX98927 & DA7219"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:kbl_da7219_max98927"); +MODULE_ALIAS("platform:kbl_max98927"); -- 2.7.4

2 2

[alsa-devel] [PATCH] ASoC: Intel: common: Add Kabylake Dialog+Maxim machine driver entry
by mac.chiang＠intel.com 15 Oct '18

15 Oct '18

From: Mac Chiang <mac.chiang(a)intel.com> This patch adds da7219_max98927 machine driver entry into machine table Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index 0ee173c..a317b77 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -32,6 +32,11 @@ static struct snd_soc_acpi_codecs kbl_7219_98357_codecs = { .codecs = {"MX98357A"} }; +static struct snd_soc_acpi_codecs kbl_7219_98927_codecs = { + .num_codecs = 1, + .codecs = {"MX98927"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { { .id = "INT343A", @@ -83,6 +88,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { .quirk_data = &kbl_7219_98357_codecs, .pdata = &skl_dmic_data, }, + { + .id = "DLGS7219", + .drv_name = "kbl_da7219_max98927", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_7219_98927_codecs, + .pdata = &skl_dmic_data + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines); -- 2.7.4

2 1

[alsa-devel] [alsa-utils][PATCH 0/9] alsactl: friendly to pluggable devices
by Takashi Sakamoto 14 Oct '18

14 Oct '18

Hi, A bug was reported that monitor mode of alsactl consumes much CPU time after disconnection of any sound card[1]. This patchset improves the mode to handle connection/disconnection of sound card. Observed control nodes are maintained by list structure instead of array. Linux specific epoll(7) is used to dispatch events. Linux specific inotify(7) is used to detect connection of new sound card. Linux specific signalfd(2) is used to catch Unix signals for termination. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-September/140580.… Takashi Sakamoto (9): alsactl: split event loop code to a function alsactl: add an iterator of registered instances of sound card alsactl: use epoll(7) instead of poll(2) alsactl: use link list to maintain source of events alsactl: use a list of source for event dispatcher instead of an array of source alsactl: obsolete array for maintenance of handlers alsactl: handle disconnection of sound card alsactl: handle detection of new sound card alsactl: use signalfd to catch UNIX signal alsactl/monitor.c | 422 ++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 375 insertions(+), 47 deletions(-) -- 2.19.1

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[alsa-devel] [PATCH 00/10] Add support for audiocodec in Allwinner A64
by Vasily Khoruzhick 13 Oct '18

13 Oct '18

This series adds Allwinner A64 audiocodec support into sun4i-i2s, sun8i-codec drivers, introduces new sun50i-codec-analog driver and enables sound on Pine64, SoPine boards and Pinebook. I2S for audiocodec in A64 is different from other 3 I2S modules but similar to one in A10, digital part of codec is compatible with A33 and analog controls part is completely different from other SoCs - it shares only few bits in few registers, so adding support for it into existing sun8i-codec-analog would mean duplicating all the widgets, controls and some routes and making it hard to read. Therefore it makes sense to introduce new driver. Marcus Cooper (1): ASoC: sun4i-i2s: Add compatibility with A64 codec I2S Vasily Khoruzhick (9): ASoC: sun8i-codec: Don't hardcode BCLK / LRCK ratio ASoC: sun8i-codec: add support for speaker amp GPIO ASoC: sun8i-codec-analog: split regmap code into separate driver ASoC: sunxi: Add bindings for A64 codec's analog path controls ASoC: sunxi: Add support for Allwinner A64 codec's analog path controls ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64 arm64: dts: allwinner: a64: add nodes necessary for analog sound support arm64: dts: allwinner: a64: enable sound on Pine64 and SoPine arm64: dts: allwinner: a64: enable sound on Pinebook .../devicetree/bindings/sound/sun4i-i2s.txt | 2 + .../bindings/sound/sun50i-codec-analog.txt | 14 + .../boot/dts/allwinner/sun50i-a64-pine64.dts | 30 ++ .../dts/allwinner/sun50i-a64-pinebook.dts | 33 ++ .../allwinner/sun50i-a64-sopine-baseboard.dts | 30 ++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 58 +++ sound/soc/sunxi/Kconfig | 17 +- sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-i2s.c | 21 + sound/soc/sunxi/sun50i-codec-analog.c | 451 ++++++++++++++++++ sound/soc/sunxi/sun8i-codec-analog.c | 79 +-- sound/soc/sunxi/sun8i-codec.c | 61 ++- sound/soc/sunxi/sunxi-adda-pr-regmap.c | 102 ++++ sound/soc/sunxi/sunxi-adda-pr-regmap.h | 7 + 14 files changed, 824 insertions(+), 83 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt create mode 100644 sound/soc/sunxi/sun50i-codec-analog.c create mode 100644 sound/soc/sunxi/sunxi-adda-pr-regmap.c create mode 100644 sound/soc/sunxi/sunxi-adda-pr-regmap.h -- 2.19.0

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[alsa-devel] [PATCH v2 0/5] MAX98088/9 of features
by Marco Felsch 12 Oct '18

12 Oct '18

Hi, this the v2 with the integrated review from the v1 [1]. I picked up the max98088 patches from Andreas series [2] and ported them to the new kernel. Regards, Marco [1] http://mailman.alsa-project.org/pipermail/alsa-devel/ \ 2018-September/140501.html [2] http://lists.infradead.org/pipermail/linux-arm-kernel/ \ 2015-February/324440.html Changes: v2: - make master clock optional - be more verbose in commit messages Andreas Färber (3): ASoC: dt-bindings: add max98088 audio codec ASoC: max98088: add OF support ASoC: max98088: Add master clock handling Marco Felsch (2): ASoC: dt-bindings: max98088: add external clock binding ASoC: max98988: make it selectable .../bindings/sound/maxim,max98088.txt | 23 ++++++++++++ sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/max98088.c | 36 +++++++++++++++++++ 3 files changed, 60 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98088.txt -- 2.19.0

3 7

[alsa-devel] [PATCH] ASoC: sta32x: set ->component pointer in private struct
by Daniel Mack 12 Oct '18

12 Oct '18

The ESD watchdog code in sta32x_watchdog() dereferences the pointer which is never assigned. This is a regression from a1be4cead9b950 ("ASoC: sta32x: Convert to direct regmap API usage.") which went unnoticed since nobody seems to use that ESD workaround. Fixes: a1be4cead9b950 ("ASoC: sta32x: Convert to direct regmap API usage.") Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/codecs/sta32x.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 22de1593443c..a878c8d92e68 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -879,6 +879,9 @@ static int sta32x_probe(struct snd_soc_component *component) struct sta32x_priv *sta32x = snd_soc_component_get_drvdata(component); struct sta32x_platform_data *pdata = sta32x->pdata; int i, ret = 0, thermal = 0; + + sta32x->component = component; + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); if (ret != 0) { -- 2.17.1

2 1

[alsa-devel] [PATCH 0/3] ASoC: pcm3168a: add TDM support
by Kuninori Morimoto 12 Oct '18

12 Oct '18

Hi Mark, Jaroslav These patches adds missing TDM support, and necessary patches for it. Kuninori Morimoto (3): ASoC: pcm3168a: add HW constraint for non RIGHT_J ASoC: pcm3168a: add hw constraint for channel ASoC: pcm3168a: add I2S/Left_J TDM support sound/soc/codecs/pcm3168a.c | 75 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 75 insertions(+) -- 2.7.4 Best regards --- Kuninori Morimoto

2 6

[alsa-devel] [bug report] ALSA: usb: initial USB Audio Device Class 3.0 support
by Dan Carpenter 12 Oct '18

12 Oct '18

Hello Ruslan Bilovol, The patch 9a2fe9b801f5: "ALSA: usb: initial USB Audio Device Class 3.0 support" from Mar 21, 2018, leads to the following static checker warning: sound/usb/stream.c:971 snd_usb_get_audioformat_uac3() warn: potentially allocating too little. 7 vs 1 sound/usb/stream.c 943 /* 944 * Get number of channels and channel map through 945 * High Capability Cluster Descriptor 946 * 947 * First step: get High Capability header and 948 * read size of Cluster Descriptor 949 */ 950 err = snd_usb_ctl_msg(chip->dev, 951 usb_rcvctrlpipe(chip->dev, 0), 952 UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR, 953 USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, 954 cluster_id, 955 snd_usb_ctrl_intf(chip), 956 &hc_header, sizeof(hc_header)); ^^^^^^^^^^ It looks like this comes from the USB (untrusted). 957 if (err < 0) 958 return ERR_PTR(err); 959 else if (err != sizeof(hc_header)) { 960 dev_err(&dev->dev, 961 "%u:%d : can't get High Capability descriptor\n", 962 iface_no, altno); 963 return ERR_PTR(-EIO); 964 } 965 966 /* 967 * Second step: allocate needed amount of memory 968 * and request Cluster Descriptor 969 */ 970 wLength = le16_to_cpu(hc_header.wLength); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ My private build of Smatch complains that all le16_to_cpu() data probably comes from untrusted sources. 971 cluster = kzalloc(wLength, GFP_KERNEL); ^^^^^^^ Maybe we're not allocating enough bytes for the cluster struct (8 bytes). 972 if (!cluster) 973 return ERR_PTR(-ENOMEM); 974 err = snd_usb_ctl_msg(chip->dev, 975 usb_rcvctrlpipe(chip->dev, 0), 976 UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR, 977 USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, 978 cluster_id, 979 snd_usb_ctrl_intf(chip), 980 cluster, wLength); 981 if (err < 0) { 982 kfree(cluster); 983 return ERR_PTR(err); 984 } else if (err != wLength) { 985 dev_err(&dev->dev, 986 "%u:%d : can't get Cluster Descriptor\n", 987 iface_no, altno); 988 kfree(cluster); 989 return ERR_PTR(-EIO); 990 } 991 992 num_channels = cluster->bNrChannels; ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 993 chmap = convert_chmap_v3(cluster); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ This code assumes that cluster is large enough without checking. 994 kfree(cluster); 995 regards, dan carpenter

1 1

[alsa-devel] [PATCH v2] pinctrl: madera: Fix uninitialized variable bug in madera_mux_set_mux
by Gustavo A. R. Silva 12 Oct '18

12 Oct '18

There is a potential execution path in which variable *ret* is checked in an IF statement, and then its value is used to report an error at line 659 without being properly initialized previously: 659 if (ret) 660 dev_err(priv->dev, "Failed to write to 0x%x (%d)\n", reg, ret); Fix this by initializing variable *ret* to 0 in order to avoid unpredictable or unintended results. Addresses-Coverity-ID: 1471969 ("Uninitialized scalar variable") Fixes: 218d72a77b0b ("pinctrl: madera: Add driver for Cirrus Logic Madera codecs") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- Changes in v2: - Initialize variable ret to 0 instead of -1. Thanks to Charles Keepax for pointing this out. - Update subject and commit log. drivers/pinctrl/cirrus/pinctrl-madera-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/pinctrl/cirrus/pinctrl-madera-core.c b/drivers/pinctrl/cirrus/pinctrl-madera-core.c index 0266302..a5dda83 100644 --- a/drivers/pinctrl/cirrus/pinctrl-madera-core.c +++ b/drivers/pinctrl/cirrus/pinctrl-madera-core.c @@ -608,7 +608,7 @@ static int madera_mux_set_mux(struct pinctrl_dev *pctldev, unsigned int n_chip_groups = priv->chip->n_pin_groups; const char *func_name = madera_mux_funcs[selector].name; unsigned int reg; - int i, ret; + int i, ret = 0; dev_dbg(priv->dev, "%s selecting %u (%s) for group %u (%s)\n", __func__, selector, func_name, group, -- 2.7.4

3 3

[alsa-devel] [PATCH] ALSA: firewire-motu: add missing entries to Kconfig
by Takashi Sakamoto 12 Oct '18

12 Oct '18

MOTU Traveler and Audio Express are supported as well. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 529d9f405fa9..8a146b039276 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -147,7 +147,9 @@ config SND_FIREWIRE_MOTU help Say Y here to enable support for FireWire devices which MOTU produced: * 828mk2 + * Traveler * 828mk3 + * Audio Express To compile this driver as a module, choose M here: the module will be called snd-firewire-motu. -- 2.19.1

2 3

[alsa-devel] [PATCH] ALSA: isight: fix leak of reference to firewire unit in error path of .probe callback
by Takashi Sakamoto 12 Oct '18

12 Oct '18

In some error paths, reference count of firewire unit is not decreased. This commit fixes the bug. Fixes: 5b14ec25a79b('ALSA: firewire: release reference count of firewire unit in .remove callback of bus driver') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/isight.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index de4decfb74d5..9ebe510ea26b 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -638,7 +638,7 @@ static int isight_probe(struct fw_unit *unit, if (!isight->audio_base) { dev_err(&unit->device, "audio unit base not found\n"); err = -ENXIO; - goto err_unit; + goto error; } fw_iso_resources_init(&isight->resources, unit); @@ -667,12 +667,12 @@ static int isight_probe(struct fw_unit *unit, dev_set_drvdata(&unit->device, isight); return 0; - -err_unit: - fw_unit_put(isight->unit); - mutex_destroy(&isight->mutex); error: snd_card_free(card); + + mutex_destroy(&isight->mutex); + fw_unit_put(isight->unit); + return err; } -- 2.19.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Add mic quirk for the Lenovo G50-30 (17aa:3905)
by Jeremy Cline 12 Oct '18

12 Oct '18

The Lenovo G50-30, like other G50 models, has a Conexant codec that requires a quirk for its inverted stereo dmic. Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1249364 Reported-by: Alexander Ploumistos <alex.ploumistos(a)gmail.com> Tested-by: Alexander Ploumistos <alex.ploumistos(a)gmail.com> Cc: stable(a)vger.kernel.org Signed-off-by: Jeremy Cline <jcline(a)redhat.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5592557fe50e..950e02e71766 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -943,6 +943,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD), + SND_PCI_QUIRK(0x17aa, 0x3905, "Lenovo G50-30", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), -- 2.19.1

2 1

[alsa-devel] [PATCH linux-next v3 1/1] ASoC: rsnd: ssi: Request dedicated dma channels for busif0 to 7
by jiada_wang＠mentor.com 12 Oct '18

12 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> Currently ssi driver only requests dma channel for SSI_0, with naming of 'rxu/txu', which is used to transfer data to/from busif0. But for GEN2/GEN3, there are also busif1 ~ busif7, which need to be used when SSI works in TDM Split/Ex-Split mode. This patch adds support to firstly try to request busif indexed dma channel with naming "rxu[busif]/txu[busif]", if these dma channel don't exist, then try to request 'rxu/txu' dma channel to keep compatibility with platform hasn't had device-tree updated. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- v3: - Drop device-tree changes - Drop automatic busif selection v2: - Instead of configure BUSIF by kctl interface "SSI Out/In BUSIF", use rsnd_ssi_select_busif() to automatically select BUSIF. - Keep 'rxu/txu' dma request, to keep compatibility with GEN2 series - Re-order patch-set to satisfy git bisect v1: initial version --- sound/soc/sh/rcar/ssi.c | 37 ++++++++++++++++++++++++++++++++----- 1 file changed, 32 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 3adcc4f778f7..7e21b29249d7 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -76,6 +76,8 @@ #define SSI_NAME "ssi" +#define SSI_DMA_NAME_SIZE 10 + struct rsnd_ssi { struct rsnd_mod mod; @@ -938,12 +940,37 @@ static struct dma_chan *rsnd_ssi_dma_req(struct rsnd_dai_stream *io, { struct rsnd_priv *priv = rsnd_mod_to_priv(mod); int is_play = rsnd_io_is_play(io); - char *name; + char name[SSI_DMA_NAME_SIZE]; - if (rsnd_ssi_use_busif(io)) - name = is_play ? "rxu" : "txu"; - else - name = is_play ? "rx" : "tx"; + if (rsnd_ssi_use_busif(io)) { + int busif = rsnd_ssi_get_busif(io); + struct dma_chan *chan = NULL; + + /* try to request rxu/txu channel with busif index first */ + if (is_play) + snprintf(name, SSI_DMA_NAME_SIZE, "rxu%d", busif); + else + snprintf(name, SSI_DMA_NAME_SIZE, "txu%d", busif); + + chan = rsnd_dma_request_channel(rsnd_ssi_of_node(priv), + mod, name); + if (!IS_ERR(chan)) + return chan; + + /* + * If dma channel with busif index doesn't exist + * request rxu/txu dma channel + */ + if (is_play) + snprintf(name, SSI_DMA_NAME_SIZE, "rxu"); + else + snprintf(name, SSI_DMA_NAME_SIZE, "txu"); + } else { + if (is_play) + snprintf(name, SSI_DMA_NAME_SIZE, "rx"); + else + snprintf(name, SSI_DMA_NAME_SIZE, "tx"); + } return rsnd_dma_request_channel(rsnd_ssi_of_node(priv), mod, name); -- 2.17.0

2 1

[alsa-devel] [PATCH] ALSA: cs4236: fix return check from call to pnp_port_start
by Colin King 11 Oct '18

11 Oct '18

From: Colin Ian King <colin.king(a)canonical.com> Currently the return from pnp_port_start is being checked for validity by checking if it is greater or equal to zero, however, the return is unsigned and hence this check is always true. An 0 return from pnp_port_start is actually an error, so fix the check by checking for validity when it is non-zero. Detected by CoverityScan, CID#1468334 ("Unsigned compared to 0") Fixes: a9824c868a2c ("[ALSA] Add CS4232 PnP BIOS support") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/isa/cs423x/cs4236.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 70559e59d18f..2211c39d2c18 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -292,7 +292,7 @@ static int snd_cs423x_pnp_init_mpu(int dev, struct pnp_dev *pdev) mpu_irq[dev] = SNDRV_AUTO_IRQ; } else { mpu_port[dev] = pnp_port_start(pdev, 0); - if (mpu_irq[dev] >= 0 && + if (!mpu_irq[dev] && pnp_irq_valid(pdev, 0) && pnp_irq(pdev, 0) >= 0) { mpu_irq[dev] = pnp_irq(pdev, 0); } else { -- 2.17.1

3 2

[alsa-devel] [PATCH 1/4] ASoC: max98373: Sort Digital Volume in reverse order
by Ryan Lee 11 Oct '18

11 Oct '18

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes : Sort 'max98373_digital_tlv' in reverse order. Fixed wrong use case of 'TLV_DB_SCALE_ITEM' - TLV_DB_SCALE_ITEM(min, step, mute) Max value was set in min field and negative value was set in step field before. sound/soc/codecs/max98373.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 1093f76..bb5ef52 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -454,7 +454,7 @@ SND_SOC_DAPM_SIGGEN("IMON"), SND_SOC_DAPM_SIGGEN("FBMON"), }; -static DECLARE_TLV_DB_SCALE(max98373_digital_tlv, 0, -50, 0); +static DECLARE_TLV_DB_SCALE(max98373_digital_tlv, -6350, 50, 1); static const DECLARE_TLV_DB_RANGE(max98373_spk_tlv, 0, 8, TLV_DB_SCALE_ITEM(0, 50, 0), 9, 10, TLV_DB_SCALE_ITEM(500, 100, 0), @@ -604,7 +604,7 @@ SOC_SINGLE("Dither Switch", MAX98373_R203F_AMP_DSP_CFG, SOC_SINGLE("DC Blocker Switch", MAX98373_R203F_AMP_DSP_CFG, MAX98373_AMP_DSP_CFG_DCBLK_SHIFT, 1, 0), SOC_SINGLE_TLV("Digital Volume", MAX98373_R203D_AMP_DIG_VOL_CTRL, - 0, 0x7F, 0, max98373_digital_tlv), + 0, 0x7F, 1, max98373_digital_tlv), SOC_SINGLE_TLV("Speaker Volume", MAX98373_R203E_AMP_PATH_GAIN, MAX98373_SPK_DIGI_GAIN_SHIFT, 10, 0, max98373_spk_tlv), SOC_SINGLE_TLV("FS Max Volume", MAX98373_R203E_AMP_PATH_GAIN, -- 2.7.4

2 7

[alsa-devel] Oops when unloading SOF driver modules
by Guennadi Liakhovetski 11 Oct '18

11 Oct '18

Hi, I'm tracking down a kernel Oops, occurring when unloading driver modules. A complete Oops is below. It triggers because cpu_dai->driver is NULL after remove_dai() in soc-topology.c freed it. On my Debian system this only happens when I unload modules directly after loading. If some time passes after loading the drivers, pulseaudio completes its probing and releases the hardware, then remove_dai() isn't called before snd_pcm_do_stop() but after, then everything is running correctly. I'm not sure what the best way to fix this would be. Any ideas? Thanks Guennadi Oct 1 10:23:32 UP-APL01 kernel: [ 142.978059] BUG: unable to handle kernel NULL pointer dereference at 0000000000000088 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978066] PGD 0 P4D 0 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978075] Oops: 0000 [#1] SMP NOPTI Oct 1 10:23:32 UP-APL01 kernel: [ 142.978083] CPU: 3 PID: 1240 Comm: alsa-sink-Port0 Tainted: G W E 4.18.0-upboard+ #1 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978087] Hardware name: AAEON UP-APL01/UP-APL01, BIOS UPA1AM36 04/10/2018 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978114] RIP: 0010:soc_pcm_trigger+0x16d/0x1d0 [snd_soc_core] Oct 1 10:23:32 UP-APL01 kernel: [ 142.978117] Code: 31 c0 48 85 c9 74 07 4c 8b 81 88 00 00 00 ba 6c 04 00 00 48 c7 c6 20 14 84 c0 48 c7 c7 72 34 84 c0 e8 55 a3 4a df 49 8b 45 18 <48> 8b 80 88 00 00 00 48 8b 80 88 00 00 00 48 85 c0 74 16 4c 89 ea Oct 1 10:23:32 UP-APL01 kernel: [ 142.978204] RSP: 0018:ffffb6b781243da8 EFLAGS: 00010286 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978207] RAX: 0000000000000000 RBX: ffff9146ae2a0800 RCX: 0000000000000000 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978210] RDX: 0000000000000000 RSI: ffff9146bfd96738 RDI: ffff9146bfd96738 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978212] RBP: ffff9146afdbbc00 R08: 00000000000006fd R09: 0000000000000004 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978215] R10: 00000000000001c3 R11: 0000000000000001 R12: 0000000000000000 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978217] R13: ffff9146b1b92f80 R14: ffff9146ae2a0eb0 R15: ffff9146ae2a0ea8 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978220] FS: 00007f3db6b24700(0000) GS:ffff9146bfd80000(0000) knlGS:0000000000000000 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978223] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978225] CR2: 0000000000000088 CR3: 0000000177ce6000 CR4: 00000000003406e0 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978228] Call Trace: Oct 1 10:23:32 UP-APL01 kernel: [ 142.978247] dpcm_fe_dai_do_trigger+0xee/0x220 [snd_soc_core] Oct 1 10:23:32 UP-APL01 kernel: [ 142.978258] snd_pcm_do_stop+0x65/0x80 [snd_pcm] Oct 1 10:23:32 UP-APL01 kernel: [ 142.978267] snd_pcm_action_single+0x36/0x80 [snd_pcm] Oct 1 10:23:32 UP-APL01 kernel: [ 142.978275] snd_pcm_drop+0x6f/0xa0 [snd_pcm] Oct 1 10:23:32 UP-APL01 kernel: [ 142.978283] snd_pcm_release_substream+0x39/0xd0 [snd_pcm] Oct 1 10:23:32 UP-APL01 kernel: [ 142.978291] snd_pcm_release+0x5d/0xb0 [snd_pcm] Oct 1 10:23:32 UP-APL01 kernel: [ 142.978298] __fput+0xd0/0x1e0 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978305] task_work_run+0x8a/0xb0 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978311] exit_to_usermode_loop+0xe6/0xf0 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978315] do_syscall_64+0xf3/0x110 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978321] entry_SYSCALL_64_after_hwframe+0x44/0xa9 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978325] RIP: 0033:0x7f3dc0988ab7 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978327] Code: 10 e9 67 ff ff ff 0f 1f 44 00 00 48 8b 15 c9 f3 2c 00 f7 d8 64 89 02 48 c7 c0 ff ff ff ff e9 6b ff ff ff b8 0b 00 00 00 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 8b 0d a1 f3 2c 00 f7 d8 64 89 01 48 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978374] RSP: 002b:00007f3db6b23c38 EFLAGS: 00000202 ORIG_RAX: 000000000000000b Oct 1 10:23:32 UP-APL01 kernel: [ 142.978377] RAX: 0000000000000000 RBX: 0000000000000000 RCX: 00007f3dc0988ab7 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978380] RDX: 0000000000000010 RSI: 0000000000001000 RDI: 00007f3dc284a000 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978382] RBP: 000056431cb73ac0 R08: 00007f3db763a578 R09: 00007f3db6b23b50 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978384] R10: 00000000fffffff8 R11: 0000000000000202 R12: 000056431cb73ac0 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978387] R13: 000056431cb73ac0 R14: 0000000000000000 R15: 000056431caad4b0 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978390] Modules linked in: sof_pci_dev(E-) snd_sof_intel_hda_common(E) sof_acpi_dev(E) snd_sof_intel_bdw(E) snd_sof_intel_byt(E) snd_sof_nocodec(E) snd_sof_intel_hsw(E) snd_sof(E) snd_sof_xtensa_dsp(E) snd_soc_acpi_intel_match(E) snd_soc_acpi(E) snd_soc_da7213(E) snd_soc_rt5640(E) snd_soc_rt5651(E) snd_soc_rt5645(E) snd_soc_rt5670(E) snd_soc_rl6231(E) nls_ascii(E) nls_cp437(E) vfat(E) fat(E) squashfs(E) loop(E) spi_pxa2xx_platform(E) intel_rapl(E) i2c_designware_platform(E) i2c_designware_core(E) x86_pkg_temp_thermal(E) coretemp(E) kvm_intel(E) kvm(E) irqbypass(E) crct10dif_pclmul(E) crc32_pclmul(E) ghash_clmulni_intel(E) snd_hda_core(E) efi_pstore(E) pcbc(E) snd_soc_core(E) snd_compress(E) snd_pcm(E) snd_seq_midi(E) snd_seq_midi_event(E) snd_rawmidi(E) aesni_intel(E) aes_x86_64(E) crypto_simd(E) Oct 1 10:23:32 UP-APL01 kernel: [ 142.978445] cryptd(E) glue_helper(E) intel_cstate(E) intel_rapl_perf(E) pwm_lpss_pci(E) efivars(E) lpc_ich(E) pwm_lpss(E) snd_seq(E) idma64(E) virt_dma(E) intel_lpss_pci(E) intel_lpss(E) mfd_core(E) snd_seq_device(E) snd_timer(E) tpm_crb(E) snd(E) tpm_tis(E) tpm_tis_core(E) tpm(E) evdev(E) rng_core(E) mei_me(E) mei(E) soundcore(E) pcc_cpufreq(E) sch_fq_codel(E) parport_pc(E) ppdev(E) lp(E) parport(E) efivarfs(E) ip_tables(E) x_tables(E) autofs4(E) ext4(E) crc32c_generic(E) crc16(E) mbcache(E) jbd2(E) fscrypto(E) hid_generic(E) usbhid(E) hid(E) mmc_block(E) dwc3(E) sdhci_pci(E) xhci_pci(E) cqhci(E) sdhci(E) crc32c_intel(E) i915(E) mmc_core(E) ahci(E) i2c_algo_bit(E) xhci_hcd(E) libahci(E) drm_kms_helper(E) dwc3_pci(E) usbcore(E) libata(E) r8169(E) mii(E) drm(E) scsi_mod(E) video(E) button(E) [last unloaded: snd_sof] Oct 1 10:23:32 UP-APL01 kernel: [ 142.978528] CR2: 0000000000000088 Oct 1 10:23:32 UP-APL01 kernel: [ 142.978532] ---[ end trace f60338bc9477e8ed ]---

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[alsa-devel] Trying to write control utility for YAMAHA Cavit DP-U50/AP-U70/RP-U200.
by Tahiro Hashizume 11 Oct '18

11 Oct '18

Hi The devices of my interest is in fact 18yrs old, but just can't stop using them for being too useful. (I own DP-U50 and RP-U200). For these devices: A.)Source selecting as well as alteration of the labels (for each input) to be displayed on the VDF are possible. B.)Internal DSP is highly configurable c.)Tuner presets are editable (RP-U200 only) through the Win32 utility (DP-U50.exe/AP-U70.exe/RP-U200.exe). Most of sound-related details should be at http://www.alsa-project.org/db/?f=9aad4f38c69efd1c6ae90de239063efeb5a61dce. Basically I am stuck with almost no experience in this, but am trying to write some code to implement the Linux-alternative of these utilities to at least enable the source selecting from Linux via USB (mostly intended to be used with shairplay-sync). Thought examples of device-specific utilities as found in alsa-tools may help but I just don't see connections between USB-dump and control signals that may be accepted by those listed in amixer. Side-notes(Findings so far): The device appears as a composite with several Audio interfaces and a HID-Class interface. Knowing this from early days, I first decided to see hexdump outputs on "/dev/hidraw0" and it indicated what looked like the representation of current status of the device while turning knobs and switching inputs. My guess is, the device takes special commands via the control and echos the current status via HID which is visible as "/dev/hidraw0". ("/dev/input/event0" is also present but doesn't really spit anything.) At this point, thought this was a bi-directional communication via HID-Class (by which I was wrong, as I found out later). Thinking collecting several packet captures under controlled conditions would help understand the mechanism, moved onto capturing WindowMessages (of RP-U200.exe) with Spy++ from which I was able to implement the automated operation of RP-U200.exe via Python. See: https://www.youtube.com/watch?v=-Il9vKtnr1A Figured out 1.)messages of "WM_USER + 1" are from "hid_io.dll" to RP-U200.exe 2.)messages of "WM_USER + 201" are the actual inputs to RP-U200.exe About files attached: a.)RP-U200.hexdump.txt shows outputs observed on "/dev/hidraw0" as I turned the volume knob from min to max and down to min. b.)"util_launch.pcapng.json" shows the interactions between host (Win10 x64) and RP-U200 as RP-U200.exe was being launched (Power:Off->Off). c.)"util_power_on.pcapng.json" shows the interactions between host (Win10 x64) and RP-U200 starting shortly before I pressed the Power button on RP-U200.exe, ending with the output relays closed for speaker outputs (Power:Off->On). Any comments should keep me encouraged. Thanks in advance. Tahiro.H

1 0

[alsa-devel] [asoc:for-4.20 333/490] sound/pci//hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'?
by kbuild test robot 11 Oct '18

11 Oct '18

Hi Rakesh, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 8036dbc490d16dc3d998246e14c9507ec8272ae2 commit: 6bae5ea9498926440ffc883f3dbceb0adc65e492 [333/490] ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers config: sh-allmodconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 6bae5ea9498926440ffc883f3dbceb0adc65e492 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): sound/pci//hda/patch_ca0132.c: In function 'patch_ca0132': >> sound/pci//hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'? [-Werror=implicit-function-declaration] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^~~~~~~~~ pcim_iomap sound/pci//hda/patch_ca0132.c:7650:18: warning: assignment makes pointer from integer without a cast [-Wint-conversion] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^ cc1: some warnings being treated as errors vim +7650 sound/pci//hda/patch_ca0132.c d5c016b56 Gabriele Martino 2015-05-18 7581 95c6e9cb7 Ian Minett 2011-06-15 7582 static int patch_ca0132(struct hda_codec *codec) 95c6e9cb7 Ian Minett 2011-06-15 7583 { 95c6e9cb7 Ian Minett 2011-06-15 7584 struct ca0132_spec *spec; a73d511c4 Ian Minett 2012-12-20 7585 int err; d5c016b56 Gabriele Martino 2015-05-18 7586 const struct snd_pci_quirk *quirk; 95c6e9cb7 Ian Minett 2011-06-15 7587 4e76a8833 Takashi Iwai 2014-02-25 7588 codec_dbg(codec, "patch_ca0132\n"); 95c6e9cb7 Ian Minett 2011-06-15 7589 95c6e9cb7 Ian Minett 2011-06-15 7590 spec = kzalloc(sizeof(*spec), GFP_KERNEL); 95c6e9cb7 Ian Minett 2011-06-15 7591 if (!spec) 95c6e9cb7 Ian Minett 2011-06-15 7592 return -ENOMEM; 95c6e9cb7 Ian Minett 2011-06-15 7593 codec->spec = spec; 993884f6a Chih-Chung Chang 2013-03-25 7594 spec->codec = codec; 95c6e9cb7 Ian Minett 2011-06-15 7595 225068ab2 Takashi Iwai 2015-05-29 7596 codec->patch_ops = ca0132_patch_ops; 225068ab2 Takashi Iwai 2015-05-29 7597 codec->pcm_format_first = 1; 225068ab2 Takashi Iwai 2015-05-29 7598 codec->no_sticky_stream = 1; 225068ab2 Takashi Iwai 2015-05-29 7599 d5c016b56 Gabriele Martino 2015-05-18 7600 /* Detect codec quirk */ d5c016b56 Gabriele Martino 2015-05-18 7601 quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks); d5c016b56 Gabriele Martino 2015-05-18 7602 if (quirk) d5c016b56 Gabriele Martino 2015-05-18 7603 spec->quirk = quirk->value; d5c016b56 Gabriele Martino 2015-05-18 7604 else d5c016b56 Gabriele Martino 2015-05-18 7605 spec->quirk = QUIRK_NONE; d5c016b56 Gabriele Martino 2015-05-18 7606 e24aa0a4c Takashi Iwai 2014-08-10 7607 spec->dsp_state = DSP_DOWNLOAD_INIT; a7e76271b Ian Minett 2012-12-20 7608 spec->num_mixers = 1; 017310fbe Connor McAdams 2018-05-08 7609 017310fbe Connor McAdams 2018-05-08 7610 /* Set which mixers each quirk uses. */ 017310fbe Connor McAdams 2018-05-08 7611 switch (spec->quirk) { 017310fbe Connor McAdams 2018-05-08 7612 case QUIRK_SBZ: e25e34450 Connor McAdams 2018-08-08 7613 spec->mixers[0] = desktop_mixer; 017310fbe Connor McAdams 2018-05-08 7614 snd_hda_codec_set_name(codec, "Sound Blaster Z"); 017310fbe Connor McAdams 2018-05-08 7615 break; e25e34450 Connor McAdams 2018-08-08 7616 case QUIRK_R3D: e25e34450 Connor McAdams 2018-08-08 7617 spec->mixers[0] = desktop_mixer; e25e34450 Connor McAdams 2018-08-08 7618 snd_hda_codec_set_name(codec, "Recon3D"); e25e34450 Connor McAdams 2018-08-08 7619 break; 017310fbe Connor McAdams 2018-05-08 7620 case QUIRK_R3DI: 017310fbe Connor McAdams 2018-05-08 7621 spec->mixers[0] = r3di_mixer; 017310fbe Connor McAdams 2018-05-08 7622 snd_hda_codec_set_name(codec, "Recon3Di"); 017310fbe Connor McAdams 2018-05-08 7623 break; 017310fbe Connor McAdams 2018-05-08 7624 default: a7e76271b Ian Minett 2012-12-20 7625 spec->mixers[0] = ca0132_mixer; 017310fbe Connor McAdams 2018-05-08 7626 break; 017310fbe Connor McAdams 2018-05-08 7627 } a7e76271b Ian Minett 2012-12-20 7628 08eca6b1f Connor McAdams 2018-08-08 7629 /* Setup whether or not to use alt functions/controls/pci_mmio */ 009b8f979 Connor McAdams 2018-05-08 7630 switch (spec->quirk) { 009b8f979 Connor McAdams 2018-05-08 7631 case QUIRK_SBZ: e42c7c731 Connor McAdams 2018-08-08 7632 case QUIRK_R3D: 08eca6b1f Connor McAdams 2018-08-08 7633 spec->use_alt_controls = true; 08eca6b1f Connor McAdams 2018-08-08 7634 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7635 spec->use_pci_mmio = true; 08eca6b1f Connor McAdams 2018-08-08 7636 break; 009b8f979 Connor McAdams 2018-05-08 7637 case QUIRK_R3DI: 47cdf76e4 Connor McAdams 2018-05-08 7638 spec->use_alt_controls = true; 009b8f979 Connor McAdams 2018-05-08 7639 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7640 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7641 break; 009b8f979 Connor McAdams 2018-05-08 7642 default: 47cdf76e4 Connor McAdams 2018-05-08 7643 spec->use_alt_controls = false; 009b8f979 Connor McAdams 2018-05-08 7644 spec->use_alt_functions = false; 08eca6b1f Connor McAdams 2018-08-08 7645 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7646 break; 009b8f979 Connor McAdams 2018-05-08 7647 } 009b8f979 Connor McAdams 2018-05-08 7648 08eca6b1f Connor McAdams 2018-08-08 7649 if (spec->use_pci_mmio) { 08eca6b1f Connor McAdams 2018-08-08 @7650 spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); 08eca6b1f Connor McAdams 2018-08-08 7651 if (spec->mem_base == NULL) { 08eca6b1f Connor McAdams 2018-08-08 7652 codec_warn(codec, "pci_iomap failed! Setting quirk to QUIRK_NONE."); 08eca6b1f Connor McAdams 2018-08-08 7653 spec->quirk = QUIRK_NONE; 08eca6b1f Connor McAdams 2018-08-08 7654 } 08eca6b1f Connor McAdams 2018-08-08 7655 } 08eca6b1f Connor McAdams 2018-08-08 7656 5aaca44d8 Ian Minett 2012-12-20 7657 spec->base_init_verbs = ca0132_base_init_verbs; 5aaca44d8 Ian Minett 2012-12-20 7658 spec->base_exit_verbs = ca0132_base_exit_verbs; 5aaca44d8 Ian Minett 2012-12-20 7659 993884f6a Chih-Chung Chang 2013-03-25 7660 INIT_DELAYED_WORK(&spec->unsol_hp_work, ca0132_unsol_hp_delayed); 993884f6a Chih-Chung Chang 2013-03-25 7661 95c6e9cb7 Ian Minett 2011-06-15 7662 ca0132_init_chip(codec); 95c6e9cb7 Ian Minett 2011-06-15 7663 95c6e9cb7 Ian Minett 2011-06-15 7664 ca0132_config(codec); 95c6e9cb7 Ian Minett 2011-06-15 7665 d5c016b56 Gabriele Martino 2015-05-18 7666 err = ca0132_prepare_verbs(codec); d5c016b56 Gabriele Martino 2015-05-18 7667 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7668 goto error; d5c016b56 Gabriele Martino 2015-05-18 7669 a73d511c4 Ian Minett 2012-12-20 7670 err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); a73d511c4 Ian Minett 2012-12-20 7671 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7672 goto error; a73d511c4 Ian Minett 2012-12-20 7673 95c6e9cb7 Ian Minett 2011-06-15 7674 return 0; cc91ceaf3 Takashi Iwai 2017-09-04 7675 cc91ceaf3 Takashi Iwai 2017-09-04 7676 error: cc91ceaf3 Takashi Iwai 2017-09-04 7677 ca0132_free(codec); cc91ceaf3 Takashi Iwai 2017-09-04 7678 return err; 95c6e9cb7 Ian Minett 2011-06-15 7679 } 95c6e9cb7 Ian Minett 2011-06-15 7680 :::::: The code at line 7650 was first introduced by commit :::::: 08eca6b1f1468a4021bac7b3929fd3eb491e2629 ALSA: hda/ca0132 - Add bool variable to enable/disable pci region2 mmio :::::: TO: Connor McAdams <conmanx360(a)gmail.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] Fixing EAPD for STAC9202 on an embedded platform.
by Andy Davison 11 Oct '18

11 Oct '18

Hi everyone, I’m hoping that somebody here might be able to spot where I’m going wrong; I'm working with a picture-frame-like device manufactured by the late OpenPeak Inc. known as the OpenFrame (or O2 Joggler in the UK). It has an internal speaker and a 3.5mm line out socket; none of the other connections to its STAC9202 codec chip are exposed to the end user. I've been porting Ubuntu Bionic to the device and everything is done, except for the audio driver. Back in kernel 3.18 it was possible to use the following kernel patch and userland ALSA patch to have these two outputs work correctly, with the insertion of a plug into the line out properly muting the internal speaker. https://github.com/andydvsn/OpenFrame-Kernel/blob/master/patches/03-stac920… https://github.com/andydvsn/OpenFrame-Ubuntu/blob/master/overlay-trusty/lib… In kernel 4.14 the same trick no longer works, so I've rewritten the patch to more properly add support for this variant of STAC9202 to ALSA. It now looks like this: https://gist.github.com/andydvsn/7bffa8de0ed691b14508485d75501ffc This seems to line up the pin configurations appropriately, but when a plug is inserted to the line out, although the audio to the internal speaker is muted, the speaker emits a crackling sound whenever the device is processing. It seems that it is not being properly powered down and is amplifying general circuit noise. Disconnecting the plug returns sound to the internal speaker and the crackling ceases. This is reliable across multiple devices. Here is the alsa-info output from both the crackling kernel 4.14 and the working-just-fine kernel 3.18. https://gist.github.com/andydvsn/ccf55e6f7961dd405fc39d41a6ce9ef6 I don't have sufficient knowledge of ALSA drivers and patches and have been working purely from the configurations used by others over the years, so I would really appreciate somebody looking at the alsa-info output and the new patch to determine where I should go from here. I believe this is all to do with EAPD control, but what options I have for controlling that aren't known to me right now. Ideally, I want to end up with the audio muting and switching correctly when the jack is inserted. A bonus would be to lose the unused (or at least inaccessible) audio input and output options in alsamixer and gain individual control of the internal speaker and line out levels (which appear to be paired together at the moment). All the best, A. -- Andrew Davison

2 11

[alsa-devel] [PATCH] pinctrl: madera: Fix uninitialized variable issue in madera_mux_set_mux
by Gustavo A. R. Silva 10 Oct '18

10 Oct '18

There is a potential execution path in which variable *ret* is checked in an IF statement, and then its value is used to report an error at line 659 without being properly initialized previously: 659 if (ret) 660 dev_err(priv->dev, "Failed to write to 0x%x (%d)\n", reg, ret); Fix this by initializing variable *ret* to -1. Addresses-Coverity-ID: 1471969 ("Uninitialized scalar variable") Fixes: 218d72a77b0b ("pinctrl: madera: Add driver for Cirrus Logic Madera codecs") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- drivers/pinctrl/cirrus/pinctrl-madera-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/pinctrl/cirrus/pinctrl-madera-core.c b/drivers/pinctrl/cirrus/pinctrl-madera-core.c index 0266302..a99ace8 100644 --- a/drivers/pinctrl/cirrus/pinctrl-madera-core.c +++ b/drivers/pinctrl/cirrus/pinctrl-madera-core.c @@ -608,7 +608,7 @@ static int madera_mux_set_mux(struct pinctrl_dev *pctldev, unsigned int n_chip_groups = priv->chip->n_pin_groups; const char *func_name = madera_mux_funcs[selector].name; unsigned int reg; - int i, ret; + int i, ret = -1; dev_dbg(priv->dev, "%s selecting %u (%s) for group %u (%s)\n", __func__, selector, func_name, group, -- 2.7.4

2 2

[alsa-devel] [PATCH] ASoC: max98988: add I2C dependency
by Arnd Bergmann 10 Oct '18

10 Oct '18

max98988 only builds with I2C support enabled, otherwise we get a build error: sound/soc/codecs/max98088.c:1789:1: error: data definition has no type or storage class [-Werror] module_i2c_driver(max98088_i2c_driver); ^~~~~~~~~~~~~~~~~ sound/soc/codecs/max98088.c:1789:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] sound/soc/codecs/max98088.c:1789:1: error: parameter names (without types) in function declaration [-Werror] sound/soc/codecs/max98088.c:1780:26: error: 'max98088_i2c_driver' defined but not used [-Werror=unused-variable] Fixes: 24ae67c58250 ("ASoC: max98988: make it selectable") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3c6bd6019b92..774d38310875 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -641,6 +641,7 @@ config SND_SOC_LM49453 config SND_SOC_MAX98088 tristate "Maxim MAX98088/9 Low-Power, Stereo Audio Codec" + depends on I2C config SND_SOC_MAX98090 tristate -- 2.18.0

3 2

[alsa-devel] [PATCH] ASoC: rsnd: use 32bit TDM width as default
by Kuninori Morimoto 10 Oct '18

10 Oct '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit fb2815f44a9e ("ASoC: rsnd: add support for 16/24 bit slot widths") added TDM width check, and return error if it was not 16/24/32 bit. But it is too strict. This patch uses 32bit same as default. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 40d7dc4..f930f51 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -744,8 +744,8 @@ static int rsnd_soc_set_dai_tdm_slot(struct snd_soc_dai *dai, case 32: break; default: - dev_err(dev, "unsupported slot width value: %d\n", slot_width); - return -EINVAL; + /* use default */ + slot_width = 32; } switch (slots) { -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: audio-graph-card: enable mclk-fs on codec node
by Kuninori Morimoto 10 Oct '18

10 Oct '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current audio-graph-card is supporting mclk-fs on CPU node side only. But having Codec node also is good idea. It will be just ignored if not defined. "rcpu_ep" is same as "cpu_ep", This patch tidyup it, too. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index fb6635f..25c819e 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -182,7 +182,8 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, if (ret < 0) goto dai_link_of_err; - of_property_read_u32(rcpu_ep, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(cpu_ep, "mclk-fs", &dai_props->mclk_fs); + of_property_read_u32(codec_ep, "mclk-fs", &dai_props->mclk_fs); ret = asoc_simple_card_parse_graph_cpu(cpu_ep, dai_link); if (ret < 0) -- 2.7.4

2 1

[alsa-devel] [PATCH 0/4] ALSA: firewire: block .remove callback of bus
by Takashi Sakamoto 10 Oct '18

10 Oct '18

Hi, In a discussion for devres support[1], I realize difference of unbind behaviour of drivers in ALSA firewire stack and in the others. For consistency behaviour inner the same subsystem for users, it's better to imitate the behaviour. Additionally, blocking .remove function simplifies codes to releasing device. This commit uses 'snd_card_free()' instead of 'snd_card_free_when_closed()' in .remove function and performs refactoring for release codes. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-September/140431.… Takashi Sakamoto (4): ALSA: firewire: block .remove callback of bus driver till all of ALSA character devices are released ALSA: firewire: release reference count of firewire unit in .remove callback of bus driver ALSA: bebob/fireworks: simplify handling of local device entry table ALSA: firewire: simplify cleanup process when failing to register sound card sound/firewire/bebob/bebob.c | 43 ++++++--------------- sound/firewire/dice/dice.c | 35 ++++------------- sound/firewire/digi00x/digi00x.c | 28 +++++--------- sound/firewire/fireface/ff.c | 28 +++++--------- sound/firewire/fireworks/fireworks.c | 56 ++++++++-------------------- sound/firewire/isight.c | 8 ++-- sound/firewire/motu/motu.c | 39 +++++-------------- sound/firewire/oxfw/oxfw.c | 39 +++++-------------- sound/firewire/tascam/tascam.c | 32 +++++----------- 9 files changed, 90 insertions(+), 218 deletions(-) -- 2.19.0

2 8

[alsa-devel] [PATCH] ALSA: hda/realtek - Fix the problem of the front MIC on the Lenovo M715
by Hui Wang 10 Oct '18

10 Oct '18

The front MIC on the Lenovo M715 can't record sound, after applying the ALC294_FIXUP_LENOVO_MIC_LOCATION, the problem is fixed. So add the pin configuration of this machine to the pin quirk table. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6f3c8e888c2a..2871c54f62ab 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6840,6 +6840,12 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1a, 0x02a11040}, {0x1b, 0x01014020}, {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0235, 0x17aa, "Lenovo", ALC294_FIXUP_LENOVO_MIC_LOCATION, + {0x14, 0x90170110}, + {0x19, 0x02a11030}, + {0x1a, 0x02a11040}, + {0x1b, 0x01011020}, + {0x21, 0x0221101f}), SND_HDA_PIN_QUIRK(0x10ec0235, 0x17aa, "Lenovo", ALC294_FIXUP_LENOVO_MIC_LOCATION, {0x14, 0x90170110}, {0x19, 0x02a11020}, -- 2.17.1

2 1

[alsa-devel] [PATCH linux-next v2 9/9] ASoC: rsnd: add busif property to dai stream
by jiada_wang＠mentor.com 10 Oct '18

10 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> in GEN3 SSI may use different BUSIF for data transfer, this patch adds busif property to each dai stream, to indicate the BUSIF used by playback/capture stream. Also adds rsnd_ssi_select_busif() to automatically select BUSIF (currently only BUSIF0 is selected) Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/core.c | 3 +++ sound/soc/sh/rcar/dma.c | 31 +++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 3 +++ sound/soc/sh/rcar/ssi.c | 30 +++++++++++++++++++++++++++++- 4 files changed, 66 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 40d7dc4f7839..5e3e6e65bcdf 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1158,6 +1158,9 @@ static int rsnd_hw_params(struct snd_pcm_substream *substream, struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); int ret; + int chan = rsnd_runtime_channel_for_ssi_with_params(io, hw_params); + + rsnd_ssi_select_busif(io, chan); ret = rsnd_dai_call(hw_params, io, substream, hw_params); if (ret) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 6d1947515dc8..bf87939f79bb 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -792,6 +792,37 @@ int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, return rsnd_dai_connect(*dma_mod, io, (*dma_mod)->type); } +void rsnd_dma_addr_update(struct rsnd_dai_stream *io) +{ + struct rsnd_mod *ssi_mod = rsnd_io_to_mod_ssi(io); + struct rsnd_mod *mod_from = NULL; + struct rsnd_mod *mod_to = NULL; + struct rsnd_priv *priv = rsnd_io_to_priv(io); + struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); + struct device *dev = rsnd_priv_to_dev(priv); + int is_play = rsnd_io_is_play(io); + struct rsnd_dma *dma = rsnd_mod_to_dma(io->dma); + + if (!dmac) + return; + + rsnd_dma_of_path(ssi_mod, io, is_play, &mod_from, &mod_to); + + dma->src_addr = rsnd_dma_addr(io, mod_from, is_play, 1); + dma->dst_addr = rsnd_dma_addr(io, mod_to, is_play, 0); + + if (mod_from && mod_to) { + struct rsnd_dmapp *dmapp = rsnd_dma_to_dmapp(dma); + + dmapp->chcr = rsnd_dmapp_get_chcr(io, mod_from, mod_to) | + PDMACHCR_DE; + } + + dev_dbg(dev, "%s[%d] %pad -> %pad\n", + rsnd_mod_name(ssi_mod), rsnd_mod_id(ssi_mod), + &dma->src_addr, &dma->dst_addr); +} + int rsnd_dma_probe(struct rsnd_priv *priv) { struct platform_device *pdev = rsnd_priv_to_pdev(priv); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 4464d1d0a042..7ce10267c87b 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -227,12 +227,14 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, u32 rsnd_get_adinr_bit(struct rsnd_mod *mod, struct rsnd_dai_stream *io); u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io); u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod); +void rsnd_ssi_select_busif(struct rsnd_dai_stream *io, int chan); /* * R-Car DMA */ int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, struct rsnd_mod **dma_mod); +void rsnd_dma_addr_update(struct rsnd_dai_stream *io); int rsnd_dma_probe(struct rsnd_priv *priv); struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, struct rsnd_mod *mod, char *name); @@ -457,6 +459,7 @@ struct rsnd_dai_stream { struct rsnd_dai *rdai; struct device *dmac_dev; /* for IPMMU */ u32 parent_ssi_status; + u32 busif; }; #define rsnd_io_to_mod(io, i) ((i) < RSND_MOD_MAX ? (io)->mod[(i)] : NULL) #define rsnd_io_to_mod_ssi(io) rsnd_io_to_mod((io), RSND_MOD_SSI) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 992aeac09e76..d515ee192276 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -154,7 +154,35 @@ int rsnd_ssi_use_busif(struct rsnd_dai_stream *io) int rsnd_ssi_get_busif(struct rsnd_dai_stream *io) { - return 0; /* BUSIF0 only for now */ + struct rsnd_priv *priv = rsnd_io_to_priv(io); + + if (!rsnd_is_gen3(priv)) + return 0; + + if (!rsnd_ssi_use_busif(io)) + return 0; + + return io->busif; +} + +void rsnd_ssi_select_busif(struct rsnd_dai_stream *io, int chan) +{ + struct rsnd_priv *priv = rsnd_io_to_priv(io); + u32 busif = 0; /* BUSIF0 only for now */ + + if (!rsnd_is_gen3(priv)) + return; + + if (!rsnd_ssi_use_busif(io)) + return; + + /* FIXME: calcuate BUSIF based on SSI mode in future */ + if (io->busif == busif) + return; + + io->busif = busif; + + rsnd_dma_addr_update(io); } static void rsnd_ssi_status_clear(struct rsnd_mod *mod) -- 2.17.0

3 9

[alsa-devel] [PATCH] ALSA: intel8x0: fix a redundant check bug
by Wenwen Wang 09 Oct '18

09 Oct '18

In snd_intel8x0_codec_semaphore(), the parameter 'codec' is firstly checked to see whether it is greater than 2. If yes, an error code EIO is returned. Otherwise, 'chip->in_sdin_init' is further checked. If 'chip->in_sdin_init' is not zero, 'codec' is updated immediately with 'chip->codec_isr_bits'. That means, the parameter 'codec' is not used in this branch. Actually, it is only used in the else branch when 'chip->in_sdin_init' is zero. Thus, the check on the parameter 'codec' is redundant if 'chip->in_sdin_init' is not zero. This can cause potential incorrect execution in this function. Suppose the parameter 'codec' is 3, which is greater than 2, and 'chip->in_sdin_init' is not zero. The current version of this function will return EIO after the first check because 'codec' is greater than 2. However, since 'codec' will be updated in the following execution when 'chip->in_sdin_init' is not zero, this check will be meaningless and the execution should continue, instead of returning the error code EIO. This patch avoids the above issue by moving the check on the parameter 'codec' to the else branch of the if statement that checks 'chip->in_sdin_init'. Signed-off-by: Wenwen Wang <wang6495(a)umn.edu> --- sound/pci/intel8x0.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 5ee468d..f619e58 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -515,13 +515,13 @@ static int snd_intel8x0_codec_semaphore(struct intel8x0 *chip, unsigned int code { int time; - if (codec > 2) - return -EIO; if (chip->in_sdin_init) { /* we don't know the ready bit assignment at the moment */ /* so we check any */ codec = chip->codec_isr_bits; } else { + if (codec > 2) + return -EIO; codec = chip->codec_bit[chip->ac97_sdin[codec]]; } -- 2.7.4

2 1

[alsa-devel] Remap pin 0x16 of ASUS ROG G751 to Headphone
by Håvard 09 Oct '18

09 Oct '18

Hi! On ASUS ROG G751 laptops the heaphone port is mapped to nothing, causing it to not function at all. I propose that we add a remap of pin 0x16 to Headphone in order to prevent this issue, and fix compability with the new mode=asus-g751. I don't know how to write patches, but this is what needs adding in sound/pci/hda/patch_realtek.c: [ALC668_FIXUP_ASUS_G751_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x16, 0x0421101f }, /* HP */ {} }, }, [ALC668_FIXUP_ASUS_G751] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { { 0x20, AC_VERB_SET_COEF_INDEX, 0xc3 }, { 0x20, AC_VERB_SET_PROC_COEF, 0x4000 }, {} }, .chained = true, .chain_id = ALC668_FIXUP_ASUS_G751_PINS }, And declaring the function earlier. -Håvard

2 2

[alsa-devel] [PATCH 0/4] Various cleanup + Mic Fix
by Connor McAdams 09 Oct '18

09 Oct '18

This patch set fixes the microphone inconsistency issue, which means the microphone now works all the time on all of the cards I've tested (ZxR, Z, AE-5), along with the input effects. It also includes changes suggested by Takashi Sakamoto, I believe I did what he asked properly, but if I messed it up I'm sure you guys will let me know. This should finish up most of the ca0132 work, with all inputs and outputs working on the desktop cards. Connor McAdams (4): ALSA: hda/ca0132 - Fix microphone inconsistency issues ALSA: hda/ca0132 - Clean up patch_ca0132() ALSA: hda/ca0132 - Add error checking in ca0132_build_controls() ALSA: hda/ca0132 - Fix input effect controls for desktop cards sound/pci/hda/patch_ca0132.c | 75 +++++++++++++++++++++++++++++++------------- 1 file changed, 54 insertions(+), 21 deletions(-) -- 2.7.4

3 6

[alsa-devel] [asoc:for-4.20 333/487] sound/pci//hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'?
by kbuild test robot 09 Oct '18

09 Oct '18

Hi Rakesh, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 24ae67c5825004bcbce90e7c89fed63f25d96260 commit: 6bae5ea9498926440ffc883f3dbceb0adc65e492 [333/487] ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers config: sh-allmodconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 6bae5ea9498926440ffc883f3dbceb0adc65e492 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): sound/pci//hda/patch_ca0132.c: In function 'patch_ca0132': >> sound/pci//hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'? [-Werror=implicit-function-declaration] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^~~~~~~~~ pcim_iomap sound/pci//hda/patch_ca0132.c:7650:18: warning: assignment makes pointer from integer without a cast [-Wint-conversion] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^ cc1: some warnings being treated as errors vim +7650 sound/pci//hda/patch_ca0132.c d5c016b56 Gabriele Martino 2015-05-18 7581 95c6e9cb7 Ian Minett 2011-06-15 7582 static int patch_ca0132(struct hda_codec *codec) 95c6e9cb7 Ian Minett 2011-06-15 7583 { 95c6e9cb7 Ian Minett 2011-06-15 7584 struct ca0132_spec *spec; a73d511c4 Ian Minett 2012-12-20 7585 int err; d5c016b56 Gabriele Martino 2015-05-18 7586 const struct snd_pci_quirk *quirk; 95c6e9cb7 Ian Minett 2011-06-15 7587 4e76a8833 Takashi Iwai 2014-02-25 7588 codec_dbg(codec, "patch_ca0132\n"); 95c6e9cb7 Ian Minett 2011-06-15 7589 95c6e9cb7 Ian Minett 2011-06-15 7590 spec = kzalloc(sizeof(*spec), GFP_KERNEL); 95c6e9cb7 Ian Minett 2011-06-15 7591 if (!spec) 95c6e9cb7 Ian Minett 2011-06-15 7592 return -ENOMEM; 95c6e9cb7 Ian Minett 2011-06-15 7593 codec->spec = spec; 993884f6a Chih-Chung Chang 2013-03-25 7594 spec->codec = codec; 95c6e9cb7 Ian Minett 2011-06-15 7595 225068ab2 Takashi Iwai 2015-05-29 7596 codec->patch_ops = ca0132_patch_ops; 225068ab2 Takashi Iwai 2015-05-29 7597 codec->pcm_format_first = 1; 225068ab2 Takashi Iwai 2015-05-29 7598 codec->no_sticky_stream = 1; 225068ab2 Takashi Iwai 2015-05-29 7599 d5c016b56 Gabriele Martino 2015-05-18 7600 /* Detect codec quirk */ d5c016b56 Gabriele Martino 2015-05-18 7601 quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks); d5c016b56 Gabriele Martino 2015-05-18 7602 if (quirk) d5c016b56 Gabriele Martino 2015-05-18 7603 spec->quirk = quirk->value; d5c016b56 Gabriele Martino 2015-05-18 7604 else d5c016b56 Gabriele Martino 2015-05-18 7605 spec->quirk = QUIRK_NONE; d5c016b56 Gabriele Martino 2015-05-18 7606 e24aa0a4c Takashi Iwai 2014-08-10 7607 spec->dsp_state = DSP_DOWNLOAD_INIT; a7e76271b Ian Minett 2012-12-20 7608 spec->num_mixers = 1; 017310fbe Connor McAdams 2018-05-08 7609 017310fbe Connor McAdams 2018-05-08 7610 /* Set which mixers each quirk uses. */ 017310fbe Connor McAdams 2018-05-08 7611 switch (spec->quirk) { 017310fbe Connor McAdams 2018-05-08 7612 case QUIRK_SBZ: e25e34450 Connor McAdams 2018-08-08 7613 spec->mixers[0] = desktop_mixer; 017310fbe Connor McAdams 2018-05-08 7614 snd_hda_codec_set_name(codec, "Sound Blaster Z"); 017310fbe Connor McAdams 2018-05-08 7615 break; e25e34450 Connor McAdams 2018-08-08 7616 case QUIRK_R3D: e25e34450 Connor McAdams 2018-08-08 7617 spec->mixers[0] = desktop_mixer; e25e34450 Connor McAdams 2018-08-08 7618 snd_hda_codec_set_name(codec, "Recon3D"); e25e34450 Connor McAdams 2018-08-08 7619 break; 017310fbe Connor McAdams 2018-05-08 7620 case QUIRK_R3DI: 017310fbe Connor McAdams 2018-05-08 7621 spec->mixers[0] = r3di_mixer; 017310fbe Connor McAdams 2018-05-08 7622 snd_hda_codec_set_name(codec, "Recon3Di"); 017310fbe Connor McAdams 2018-05-08 7623 break; 017310fbe Connor McAdams 2018-05-08 7624 default: a7e76271b Ian Minett 2012-12-20 7625 spec->mixers[0] = ca0132_mixer; 017310fbe Connor McAdams 2018-05-08 7626 break; 017310fbe Connor McAdams 2018-05-08 7627 } a7e76271b Ian Minett 2012-12-20 7628 08eca6b1f Connor McAdams 2018-08-08 7629 /* Setup whether or not to use alt functions/controls/pci_mmio */ 009b8f979 Connor McAdams 2018-05-08 7630 switch (spec->quirk) { 009b8f979 Connor McAdams 2018-05-08 7631 case QUIRK_SBZ: e42c7c731 Connor McAdams 2018-08-08 7632 case QUIRK_R3D: 08eca6b1f Connor McAdams 2018-08-08 7633 spec->use_alt_controls = true; 08eca6b1f Connor McAdams 2018-08-08 7634 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7635 spec->use_pci_mmio = true; 08eca6b1f Connor McAdams 2018-08-08 7636 break; 009b8f979 Connor McAdams 2018-05-08 7637 case QUIRK_R3DI: 47cdf76e4 Connor McAdams 2018-05-08 7638 spec->use_alt_controls = true; 009b8f979 Connor McAdams 2018-05-08 7639 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7640 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7641 break; 009b8f979 Connor McAdams 2018-05-08 7642 default: 47cdf76e4 Connor McAdams 2018-05-08 7643 spec->use_alt_controls = false; 009b8f979 Connor McAdams 2018-05-08 7644 spec->use_alt_functions = false; 08eca6b1f Connor McAdams 2018-08-08 7645 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7646 break; 009b8f979 Connor McAdams 2018-05-08 7647 } 009b8f979 Connor McAdams 2018-05-08 7648 08eca6b1f Connor McAdams 2018-08-08 7649 if (spec->use_pci_mmio) { 08eca6b1f Connor McAdams 2018-08-08 @7650 spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); 08eca6b1f Connor McAdams 2018-08-08 7651 if (spec->mem_base == NULL) { 08eca6b1f Connor McAdams 2018-08-08 7652 codec_warn(codec, "pci_iomap failed! Setting quirk to QUIRK_NONE."); 08eca6b1f Connor McAdams 2018-08-08 7653 spec->quirk = QUIRK_NONE; 08eca6b1f Connor McAdams 2018-08-08 7654 } 08eca6b1f Connor McAdams 2018-08-08 7655 } 08eca6b1f Connor McAdams 2018-08-08 7656 5aaca44d8 Ian Minett 2012-12-20 7657 spec->base_init_verbs = ca0132_base_init_verbs; 5aaca44d8 Ian Minett 2012-12-20 7658 spec->base_exit_verbs = ca0132_base_exit_verbs; 5aaca44d8 Ian Minett 2012-12-20 7659 993884f6a Chih-Chung Chang 2013-03-25 7660 INIT_DELAYED_WORK(&spec->unsol_hp_work, ca0132_unsol_hp_delayed); 993884f6a Chih-Chung Chang 2013-03-25 7661 95c6e9cb7 Ian Minett 2011-06-15 7662 ca0132_init_chip(codec); 95c6e9cb7 Ian Minett 2011-06-15 7663 95c6e9cb7 Ian Minett 2011-06-15 7664 ca0132_config(codec); 95c6e9cb7 Ian Minett 2011-06-15 7665 d5c016b56 Gabriele Martino 2015-05-18 7666 err = ca0132_prepare_verbs(codec); d5c016b56 Gabriele Martino 2015-05-18 7667 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7668 goto error; d5c016b56 Gabriele Martino 2015-05-18 7669 a73d511c4 Ian Minett 2012-12-20 7670 err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); a73d511c4 Ian Minett 2012-12-20 7671 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7672 goto error; a73d511c4 Ian Minett 2012-12-20 7673 95c6e9cb7 Ian Minett 2011-06-15 7674 return 0; cc91ceaf3 Takashi Iwai 2017-09-04 7675 cc91ceaf3 Takashi Iwai 2017-09-04 7676 error: cc91ceaf3 Takashi Iwai 2017-09-04 7677 ca0132_free(codec); cc91ceaf3 Takashi Iwai 2017-09-04 7678 return err; 95c6e9cb7 Ian Minett 2011-06-15 7679 } 95c6e9cb7 Ian Minett 2011-06-15 7680 :::::: The code at line 7650 was first introduced by commit :::::: 08eca6b1f1468a4021bac7b3929fd3eb491e2629 ALSA: hda/ca0132 - Add bool variable to enable/disable pci region2 mmio :::::: TO: Connor McAdams <conmanx360(a)gmail.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH][RFC][alsa-utils 0/9] alsactl: monitor mode friendly for
by Takashi Sakamoto 09 Oct '18

09 Oct '18

Hi, A command of 'alsactl' has 'monitor' mode to print events emitted by observed control nodes. This mode is not friendly for plug-and-play devices. A bug is reported that the mode consumes 100% of CPU time when detecting disconnection of observed sound card[1]. A process of alsactl doesn't release ALSA control character device at disconnection. This is quite bad in a view of system. Furthermore, this mode cannot handle control nodes for newly-detected sound card. This patchset is my attempt to improve the mode to handle disconnection of observed sound card and new detection of sound card. A rough summary of this patchset: - maintains observed control nodes by link-list - uses epoll(7) instead of poll(2) to avoid following the list every time of poll loop, - control node for disconnected sound card is not observed anymore and dropped from the list. - the list is rescan and observation is updated when detecting new sound card. Linux-specific inotify(7) is used for this purpose. I'm happy if receiving test reports and indication of new bugs. [1] [alsa-devel] alsactl monitor spins at 100% cpu after unplugging USB headphones http://mailman.alsa-project.org/pipermail/alsa-devel/2018-September/140580.… Takashi Sakamoto (9): alsactl: install signal handler alsactl: split event loop code to a function alsactl: add an iterator of registered instances of sound card alsactl: use epoll(7) instead of poll(2) alsactl: use link list to maintain source of events alsactl: use a list of source for event dispatcher instead of an array of source alsactl: obsolete array for maintenance of handlers alsactl: handle disconnection of sound card alsactl: handle detection of new sound card alsactl/monitor.c | 403 ++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 356 insertions(+), 47 deletions(-) -- 2.19.0

3 15

[alsa-devel] Applied "ASoC: max98988: make it selectable" to the asoc tree
by Mark Brown 08 Oct '18

08 Oct '18

The patch ASoC: max98988: make it selectable has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 24ae67c5825004bcbce90e7c89fed63f25d96260 Mon Sep 17 00:00:00 2001 From: Marco Felsch <m.felsch(a)pengutronix.de> Date: Fri, 5 Oct 2018 09:58:12 +0200 Subject: [PATCH] ASoC: max98988: make it selectable Currently the driver will build only if SND_SOC_ALL_CODECS is set. Adding a Kconfig menu description to build the driver standalone. Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9989d35e0fc6..3c6bd6019b92 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -640,7 +640,7 @@ config SND_SOC_LM49453 tristate config SND_SOC_MAX98088 - tristate + tristate "Maxim MAX98088/9 Low-Power, Stereo Audio Codec" config SND_SOC_MAX98090 tristate -- 2.19.0

1 0

[alsa-devel] [PATCH 00/11] Add ZxR support + bugfixes
by Connor McAdams 08 Oct '18

08 Oct '18

This patch series adds support for the Sound Blaster ZxR, as well as a few bug fixes. This should be the last ca0132 based Creative card that needed support to be added. Also, I did check to make sure each patch compiles properly this time, but you can check yourself just to be sure. :) Connor McAdams (11): ALSA: hda/ca0132 - Fix AE-5 control type ALSA: hda/ca0132 - Fix surround sound with output effects ALSA: hda/ca0132 - Add ZxR quirks + new quirk check function ALSA: hda/ca0132 - Add ZxR pincfg ALSA: hda/ca0132 - Add DBpro hda_codec_ops ALSA: hda/ca0132 - Add ZxR init commands ALSA: hda/ca0132 - Add ZxR DSP post-download commands ALSA: hda/ca0132 - Add ZxR input/output select commands ALSA: hda/ca0132 - Remove input select enum for ZxR ALSA: hda/ca0132 - Add ZxR 600 ohm gain control ALSA: hda/ca0132 - Add ZxR exit commands sound/pci/hda/patch_ca0132.c | 369 ++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 345 insertions(+), 24 deletions(-) -- 2.7.4

3 17

[alsa-devel] [PATCH 1/2] ASoC: Add device tree documentation file for wm8782 stereo DAC
by Daniel Mack 08 Oct '18

08 Oct '18

Add a device tree documentation file for the wm8782 stereo DAC to describe the regulator handles. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- .../devicetree/bindings/sound/wm8782.txt | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm8782.txt diff --git a/Documentation/devicetree/bindings/sound/wm8782.txt b/Documentation/devicetree/bindings/sound/wm8782.txt new file mode 100644 index 000000000000..256cdec6ec4d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8782.txt @@ -0,0 +1,17 @@ +WM8782 stereo ADC + +This device does not have any control interface or reset pins. + +Required properties: + + - compatible : "wlf,wm8782" + - Vdda-supply : phandle to a regulator for the analog power supply (2.7V - 5.5V) + - Vdd-supply : phandle to a regulator for the digital power supply (2.7V - 3.6V) + +Example: + +wm8782: stereo-adc { + compatible = "wlf,wm8782"; + Vdda-supply = <&vdda_supply>; + Vdd-supply = <&vdd_supply>; +}; -- 2.17.1

3 6

Re: [alsa-devel] Microphone detected, but no output for all ASUS G751xx with ALC668 chipset
by Connor McAdams 08 Oct '18

08 Oct '18

I might not get around to writing the fix myself, but, here's the COEF's from within Windows. I've marked the ones missing inside the main init function. So, if anyone else wants to add these in, feel free. On Wed, Sep 12, 2018 at 1:16 PM, Håvard <hovardslill(a)gmail.com> wrote: > That was without running the program (sorry) > > This is output when running program: > > Before there was no "Current verb...." > They came when I opened up windows microphone settings. (something I didn't > do before) > > -Håvard > > Den ons. 12. sep. 2018 kl. 19:11 skrev Håvard <hovardslill(a)gmail.com>: >> >> I think this is right! >> >> I first booted up, then tested speakers (That I could not hear) and then >> tested microphone (Which worked perfectly) in windows settings. >> >> (My first email was dissaproved as it was too large) >> >> Den ons. 12. sep. 2018 kl. 19:05 skrev Håvard <hovardslill(a)gmail.com>: >>> >>> I think this is right! >>> >>> I first booted up, then tested speakers (That I could not hear) and then >>> tested microphone (Which worked perfectly) in windows settings. >>> >>> -Håvard >>> >>> Den ons. 12. sep. 2018 kl. 18:49 skrev Connor McAdams >>> <conmanx360(a)gmail.com>: >>>> >>>> Also, sorry to keep sending more replies, make sure the only trace you >>>> have on is vfio_region_write, and not the read one. The read spams up >>>> the console. >>>> >>>> On Wed, Sep 12, 2018 at 12:43 PM, Connor McAdams <conmanx360(a)gmail.com> >>>> wrote: >>>> > Reason being, it may not have identified the CORB buffer properly or >>>> > something. That's very odd to have no data. Never seen that before. >>>> > >>>> > On Wed, Sep 12, 2018 at 12:42 PM, Connor McAdams >>>> > <conmanx360(a)gmail.com> wrote: >>>> >> Could you try again and make a copy of your terminal output? >>>> >> >>>> >> On Wed, Sep 12, 2018 at 12:41 PM, Håvard <hovardslill(a)gmail.com> >>>> >> wrote: >>>> >>> Hi again. >>>> >>> Sorry for not noticing all files were empty, did it not run for long >>>> >>> enough? >>>> >>> >>>> >>> -Håvard >>>> >>> >>>> >>> Den ons. 12. sep. 2018 kl. 18:39 skrev Håvard >>>> >>> <hovardslill(a)gmail.com>: >>>> >>>> >>>> >>>> Sorry my bad again it was working perfectly, just missed a step. >>>> >>>> >>>> >>>> I'll add the dumps as an attatchment. I had both external headset >>>> >>>> and >>>> >>>> microphone plugged in and let the VM run in the background for >>>> >>>> ~30min while >>>> >>>> doing other stuff. I did not play music or test my microphone in >>>> >>>> the VM >>>> >>>> either, If i did anything wrong, please tell me! >>>> >>>> >>>> >>>> It stopped at 0x104f4 >>>> >>>> >>>> >>>> -Håvard >>>> >>>> >>>> >>>> Den ons. 12. sep. 2018 kl. 00:31 skrev Connor McAdams >>>> >>>> <conmanx360(a)gmail.com>: >>>> >>>>> >>>> >>>>> Hm.... that's odd. They should show up in the folder you ran the >>>> >>>>> command from. Does your console show any "DumpMem entered..." or >>>> >>>>> something like that? You may have a permissions error. I've had >>>> >>>>> some >>>> >>>>> people report that as an issue before. >>>> >>>>> >>>> >>>>> On Tue, Sep 11, 2018 at 5:08 PM, Håvard <hovardslill(a)gmail.com> >>>> >>>>> wrote: >>>> >>>>> > Hi! >>>> >>>>> > >>>> >>>>> > Took a while, but I think I got it working. Did not see any >>>> >>>>> > "frame[xx]" >>>> >>>>> > files though. What dumps do you want? >>>> >>>>> > >>>> >>>>> > -Håvard >>>> >>>>> > >>>> >>>>> > Den tir. 11. sep. 2018 kl. 21:27 skrev Håvard >>>> >>>>> > <hovardslill(a)gmail.com>: >>>> >>>>> >> >>>> >>>>> >> Sorry about that, I forgot to enable a kernel config... >>>> >>>>> >> >>>> >>>>> >> -Håvard >>>> >>>>> >> >>>> >>>>> >> Den tir. 11. sep. 2018 kl. 21:20 skrev Connor McAdams >>>> >>>>> >> <conmanx360(a)gmail.com>: >>>> >>>>> >>> >>>> >>>>> >>> When it's bound as a stub, it shouldn't show up in alsamixer >>>> >>>>> >>> controls. >>>> >>>>> >>> You can check what module is loaded by doing lspci -v . It >>>> >>>>> >>> should say >>>> >>>>> >>> kernel driver in use: pci-stub. Also, did you run the >>>> >>>>> >>> vfio-bind >>>> >>>>> >>> script >>>> >>>>> >>> before trying it? >>>> >>>>> >>> >>>> >>>>> >>> On Tue, Sep 11, 2018 at 3:18 PM, Håvard >>>> >>>>> >>> <hovardslill(a)gmail.com> >>>> >>>>> >>> wrote: >>>> >>>>> >>> > Thank you for the offer! >>>> >>>>> >>> > >>>> >>>>> >>> > I run gentoo, but your github guide is very useful. I can't >>>> >>>>> >>> > seem to >>>> >>>>> >>> > bind my >>>> >>>>> >>> > audio driver to a pci-stud. Is the Sound card still supposed >>>> >>>>> >>> > to >>>> >>>>> >>> > function as >>>> >>>>> >>> > normal when I (think I) have bound it to a stud. >>>> >>>>> >>> > >>>> >>>>> >>> > -Håvard >>>> >>>>> >>> > >>>> >>>>> >>> > Den tir. 11. sep. 2018 kl. 20:26 skrev Connor McAdams >>>> >>>>> >>> > <conmanx360(a)gmail.com>: >>>> >>>>> >>> >> >>>> >>>>> >>> >> One thing you could try, is using the program I used to >>>> >>>>> >>> >> reverse >>>> >>>>> >>> >> engineer the Sound Blaster Z series of cards, QemuHDADump. >>>> >>>>> >>> >> If your >>>> >>>>> >>> >> processor supports pci-passthrough with a virtual machine, >>>> >>>>> >>> >> you >>>> >>>>> >>> >> could >>>> >>>>> >>> >> run a Windows virtual machine with the sound card in it and >>>> >>>>> >>> >> capture >>>> >>>>> >>> >> the commands. I'd be willing to look through the dumps to >>>> >>>>> >>> >> see if >>>> >>>>> >>> >> there >>>> >>>>> >>> >> are any special verbs or anything. >>>> >>>>> >>> >> >>>> >>>>> >>> >> You can find the program here: >>>> >>>>> >>> >> https://github.com/Conmanx360/QemuHDADump >>>> >>>>> >>> >> >>>> >>>>> >>> >> Let me know. >>>> >>>>> >>> >> >>>> >>>>> >>> >> On Tue, Sep 11, 2018 at 2:15 PM, Håvard >>>> >>>>> >>> >> <hovardslill(a)gmail.com> >>>> >>>>> >>> >> wrote: >>>> >>>>> >>> >> > Sorry, my gmail didn't update so I wrote my response >>>> >>>>> >>> >> > before I >>>> >>>>> >>> >> > read >>>> >>>>> >>> >> > your >>>> >>>>> >>> >> > last one. >>>> >>>>> >>> >> > >>>> >>>>> >>> >> > It is a separate mic port as the G751JT doesn't have any >>>> >>>>> >>> >> > headset >>>> >>>>> >>> >> > multijacks. >>>> >>>>> >>> >> > >>>> >>>>> >>> >> > I'll try and dig around in the kernel! Thanks for the >>>> >>>>> >>> >> > tip! >>>> >>>>> >>> >> > >>>> >>>>> >>> >> > -Håvard >>>> >>>>> >>> >> > >>>> >>>>> >>> >> > Den tir. 11. sep. 2018 kl. 20:09 skrev Håvard >>>> >>>>> >>> >> > <hovardslill(a)gmail.com>: >>>> >>>>> >>> >> > >>>> >>>>> >>> >> >> Thank you for taking your time and trying to help! :) >>>> >>>>> >>> >> >> >>>> >>>>> >>> >> >> Do you know where I can ask around for more help on the >>>> >>>>> >>> >> >> issue? >>>> >>>>> >>> >> >> I >>>> >>>>> >>> >> >> don't >>>> >>>>> >>> >> >> want to give up yet. >>>> >>>>> >>> >> >> >>>> >>>>> >>> >> >> -Håvard >>>> >>>>> >>> >> >> >>>> >>>>> >>> >> >> Den tir. 11. sep. 2018 kl. 20:02 skrev Takashi Iwai >>>> >>>>> >>> >> >> <tiwai(a)suse.de>: >>>> >>>>> >>> >> >> >>>> >>>>> >>> >> >>> On Tue, 11 Sep 2018 19:14:37 +0200, >>>> >>>>> >>> >> >>> Håvard wrote: >>>> >>>>> >>> >> >>> > >>>> >>>>> >>> >> >>> > Yes, microphone gets detected instantly and it >>>> >>>>> >>> >> >>> > automatically >>>> >>>>> >>> >> >>> > changes >>>> >>>>> >>> >> >>> > to >>>> >>>>> >>> >> >>> it >>>> >>>>> >>> >> >>> > in pavucontrol. >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> Then it likely requires some additional initialization >>>> >>>>> >>> >> >>> outside >>>> >>>>> >>> >> >>> HD-audio. It's hard to know, as it's pretty much >>>> >>>>> >>> >> >>> vendor-specific. >>>> >>>>> >>> >> >>> You can dig down the Windows, but I have no idea about >>>> >>>>> >>> >> >>> Windows >>>> >>>>> >>> >> >>> implementation, so can't give any hints, unfortunately. >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> Takashi >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> > >>>> >>>>> >>> >> >>> > Den tir. 11. sep. 2018 kl. 18:52 skrev Takashi Iwai >>>> >>>>> >>> >> >>> > <tiwai(a)suse.de>: >>>> >>>>> >>> >> >>> > >>>> >>>>> >>> >> >>> > > On Tue, 11 Sep 2018 18:40:23 +0200, >>>> >>>>> >>> >> >>> > > Håvard wrote: >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > Thank you for replying! >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > Enabling loopback in alsamixer: >>>> >>>>> >>> >> >>> > > > http://i.imgur.com/lNo6e7T.png >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > And unmuting more and more things in "Mic >>>> >>>>> >>> >> >>> > > > Playback >>>> >>>>> >>> >> >>> > > > Volume" >>>> >>>>> >>> >> >>> > > > in >>>> >>>>> >>> >> >>> > > hdaanalyzer: >>>> >>>>> >>> >> >>> > > > http://i.imgur.com/H0HiOhy.png >>>> >>>>> >>> >> >>> > > > made white noise come from the headset. However >>>> >>>>> >>> >> >>> > > > it did >>>> >>>>> >>> >> >>> > > > not >>>> >>>>> >>> >> >>> > > > change or >>>> >>>>> >>> >> >>> > > react >>>> >>>>> >>> >> >>> > > > at all when I talked or even muted the microphone >>>> >>>>> >>> >> >>> > > > physically. >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > I couldn't find "Mic Playback Switch" anywhere in >>>> >>>>> >>> >> >>> > > > either >>>> >>>>> >>> >> >>> > > > alsamixer >>>> >>>>> >>> >> >>> or >>>> >>>>> >>> >> >>> > > > hdaanalyzer. >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > It's a mixer mute switch. >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > > The microphone works perfectly fine under >>>> >>>>> >>> >> >>> > > > Windows, so I >>>> >>>>> >>> >> >>> > > > don't >>>> >>>>> >>> >> >>> > > > think >>>> >>>>> >>> >> >>> it is >>>> >>>>> >>> >> >>> > > > the mic pin. >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > But the fact above indicates the possibility of the >>>> >>>>> >>> >> >>> > > wrong >>>> >>>>> >>> >> >>> > > pin, >>>> >>>>> >>> >> >>> > > too. >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > Does the jack detection of the ext mic pin work? >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > Takashi >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > -Håvard >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > Den man. 10. sep. 2018 kl. 22:39 skrev Takashi >>>> >>>>> >>> >> >>> > > > Iwai >>>> >>>>> >>> >> >>> > > > <tiwai(a)suse.de >>>> >>>>> >>> >> >>> >: >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > > > On Thu, 06 Sep 2018 20:44:30 +0200, >>>> >>>>> >>> >> >>> > > > > Håvard wrote: >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > Additional relevant info: >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > A similar issue was also discussed three >>>> >>>>> >>> >> >>> > > > > > years ago >>>> >>>>> >>> >> >>> > > > > > on >>>> >>>>> >>> >> >>> > > > > > Sun >>>> >>>>> >>> >> >>> > > > > > Jun >>>> >>>>> >>> >> >>> > > 17:15:54 >>>> >>>>> >>> >> >>> > > > > CEST >>>> >>>>> >>> >> >>> > > > > > 2015 and was about his surround sound setup, >>>> >>>>> >>> >> >>> > > > > > but did >>>> >>>>> >>> >> >>> > > > > > not >>>> >>>>> >>> >> >>> > > > > > touch >>>> >>>>> >>> >> >>> on the >>>> >>>>> >>> >> >>> > > > > > external microphone problem: >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093317.html >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > alsa-info.sh: >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >>> http://www.alsa-project.org/db/?f=1d8616ba5977308e03db6c3a86e36e9e9b38d6f0 >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > Graph of ALC668 chipset from hdaanalyzer: >>>> >>>>> >>> >> >>> > > > > > http://i.imgur.com/c08DNJW.png >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > setting alsa-mode[1-8] does nothing to help >>>> >>>>> >>> >> >>> > > > > > the >>>> >>>>> >>> >> >>> > > > > > issue. >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > There are several of similar bug reports >>>> >>>>> >>> >> >>> > > > > > around the >>>> >>>>> >>> >> >>> > > > > > web >>>> >>>>> >>> >> >>> experiencing >>>> >>>>> >>> >> >>> > > > > > similar issues, and on different distros. >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > Microphone works perfectly in windows >>>> >>>>> >>> >> >>> > > > > > >>>> >>>>> >>> >> >>> > > > > > I have a ASUS ROG G751JT, but this problem >>>> >>>>> >>> >> >>> > > > > > seems to >>>> >>>>> >>> >> >>> > > > > > happen >>>> >>>>> >>> >> >>> > > > > > with >>>> >>>>> >>> >> >>> all >>>> >>>>> >>> >> >>> > > > > laptops >>>> >>>>> >>> >> >>> > > > > > under the G751 name. >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > > > When you enable the loopback volume and switch, >>>> >>>>> >>> >> >>> > > > > and >>>> >>>>> >>> >> >>> > > > > unmute/adjust >>>> >>>>> >>> >> >>> "Mic >>>> >>>>> >>> >> >>> > > > > Playback Volume", and "Mic Playback Switch", do >>>> >>>>> >>> >> >>> > > > > you >>>> >>>>> >>> >> >>> > > > > hear >>>> >>>>> >>> >> >>> > > > > the >>>> >>>>> >>> >> >>> > > > > input >>>> >>>>> >>> >> >>> > > > > from the ext mic? It's a route directly from >>>> >>>>> >>> >> >>> > > > > NID 0x18 >>>> >>>>> >>> >> >>> > > > > to >>>> >>>>> >>> >> >>> > > > > the >>>> >>>>> >>> >> >>> mixer >>>> >>>>> >>> >> >>> > > > > NID 0x0b, then output mixer NID 0x0c, then >>>> >>>>> >>> >> >>> > > > > outputs. >>>> >>>>> >>> >> >>> > > > > So >>>> >>>>> >>> >> >>> > > > > this >>>> >>>>> >>> >> >>> > > > > can >>>> >>>>> >>> >> >>> be >>>> >>>>> >>> >> >>> > > > > used to verify the hardware routing. >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > > > If you don't hear via this route, it means that >>>> >>>>> >>> >> >>> > > > > the >>>> >>>>> >>> >> >>> > > > > input >>>> >>>>> >>> >> >>> > > > > from >>>> >>>>> >>> >> >>> the ext >>>> >>>>> >>> >> >>> > > > > mic pin itself is broken, and it implies that >>>> >>>>> >>> >> >>> > > > > something >>>> >>>>> >>> >> >>> > > > > outside >>>> >>>>> >>> >> >>> > > > > HD-audio codec. >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > > > Takashi >>>> >>>>> >>> >> >>> > > > > >>>> >>>>> >>> >> >>> > > > [2 <text/html; UTF-8 (quoted-printable)>] >>>> >>>>> >>> >> >>> > > > >>>> >>>>> >>> >> >>> > > >>>> >>>>> >>> >> >>> > [2 <text/html; UTF-8 (quoted-printable)>] >>>> >>>>> >>> >> >>> > >>>> >>>>> >>> >> >>> >>>> >>>>> >>> >> >> >>>> >>>>> >>> >> > _______________________________________________ >>>> >>>>> >>> >> > Alsa-devel mailing list >>>> >>>>> >>> >> > Alsa-devel(a)alsa-project.org >>>> >>>>> >>> >> > >>>> >>>>> >>> >> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

3 26

[alsa-devel] [PATCH 1/2] ALSA: hda - Add quirk for ASUS G751 laptop
by Takashi Iwai 07 Oct '18

07 Oct '18

ASUS G751 requires the extra COEF initialization to make it microphone working properly. Reported-and-tested-by: Håvard <hovardslill(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 69283633ae91..389e8e996589 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7738,6 +7738,7 @@ enum { ALC662_FIXUP_ASUS_Nx50, ALC668_FIXUP_ASUS_Nx51_HEADSET_MODE, ALC668_FIXUP_ASUS_Nx51, + ALC668_FIXUP_ASUS_G751, ALC891_FIXUP_HEADSET_MODE, ALC891_FIXUP_DELL_MIC_NO_PRESENCE, ALC662_FIXUP_ACER_VERITON, @@ -8007,6 +8008,14 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC668_FIXUP_ASUS_Nx51_HEADSET_MODE, }, + [ALC668_FIXUP_ASUS_G751] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0xc3 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x4000 }, + {} + }, + }, [ALC891_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode, @@ -8080,6 +8089,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50), SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A), SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_ASUS_Nx50), + SND_PCI_QUIRK(0x1043, 0x12ff, "ASUS G751", ALC668_FIXUP_ASUS_G751), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51), -- 2.18.0

1 1

Re: [alsa-devel] WARNING in port_delete
by syzbot 06 Oct '18

06 Oct '18

syzbot has found a reproducer for the following crash on: HEAD commit: 091a1eaa0e30 Merge branch 'akpm' git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=16623e59400000 kernel config: https://syzkaller.appspot.com/x/.config?x=c0af03fe452b65fb dashboard link: https://syzkaller.appspot.com/bug?extid=e4c8abb920efa77bace9 compiler: gcc (GCC) 8.0.1 20180413 (experimental) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=102e2c91400000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+e4c8abb920efa77bace9(a)syzkaller.appspotmail.com WARNING: CPU: 1 PID: 23693 at sound/core/seq/seq_ports.c:275 port_delete+0x177/0x1b0 sound/core/seq/seq_ports.c:275 Kernel panic - not syncing: panic_on_warn set ... CPU: 1 PID: 23693 Comm: syz-executor0 Not tainted 4.19.0-rc6+ #47 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x1c4/0x2b4 lib/dump_stack.c:113 panic+0x238/0x4e7 kernel/panic.c:184 __warn.cold.8+0x163/0x1ba kernel/panic.c:536 report_bug+0x254/0x2d0 lib/bug.c:186 fixup_bug arch/x86/kernel/traps.c:178 [inline] do_error_trap+0x1fc/0x4d0 arch/x86/kernel/traps.c:296 kobject: 'loop1' (00000000baf940ef): kobject_uevent_env kobject: 'loop1' (00000000baf940ef): fill_kobj_path: path = '/devices/virtual/block/loop1' do_invalid_op+0x1b/0x20 arch/x86/kernel/traps.c:316 invalid_op+0x14/0x20 arch/x86/entry/entry_64.S:993 RIP: 0010:port_delete+0x177/0x1b0 sound/core/seq/seq_ports.c:275 Code: e7 99 fb 45 85 e4 75 14 e8 66 e6 99 fb 48 89 df e8 ee 29 dd fb 31 c0 5b 41 5c 5d c3 e8 52 e6 99 fb 0f 0b eb e3 e8 49 e6 99 fb <0f> 0b eb 9b e8 a0 55 dd fb eb bd e8 99 55 dd fb e9 77 ff ff ff e8 RSP: 0018:ffff8801a7477890 EFLAGS: 00010293 RAX: ffff8801a84e05c0 RBX: ffff8801cf0be4c0 RCX: ffffffff85e4eee1 RDX: 0000000000000000 RSI: ffffffff85e4ef47 RDI: 0000000000000005 RBP: ffff8801a74778a0 R08: ffff8801a84e05c0 R09: 00000000deede79a R10: 00000000da8ba726 R11: 0000000000000000 R12: 0000000000000001 R13: ffff8801cf0be510 R14: 1ffff10034e8ef20 R15: ffff8801cb447b00 snd_seq_delete_port+0x3cb/0x4a0 sound/core/seq/seq_ports.c:303 snd_seq_ioctl_delete_port+0xba/0x190 sound/core/seq/seq_clientmgr.c:1324 snd_seq_kernel_client_ctl+0x15a/0x190 sound/core/seq/seq_clientmgr.c:2353 snd_seq_event_port_detach+0xfc/0x170 sound/core/seq/seq_ports.c:705 delete_port+0x7d/0xc0 sound/core/seq/oss/seq_oss_init.c:354 snd_seq_oss_release+0xf2/0x130 sound/core/seq/oss/seq_oss_init.c:433 odev_release+0x52/0x70 sound/core/seq/oss/seq_oss.c:153 __fput+0x385/0xa30 fs/file_table.c:278 ____fput+0x15/0x20 fs/file_table.c:309 task_work_run+0x1e8/0x2a0 kernel/task_work.c:113 tracehook_notify_resume include/linux/tracehook.h:193 [inline] exit_to_usermode_loop+0x318/0x380 arch/x86/entry/common.c:166 prepare_exit_to_usermode arch/x86/entry/common.c:197 [inline] syscall_return_slowpath arch/x86/entry/common.c:268 [inline] do_syscall_64+0x6be/0x820 arch/x86/entry/common.c:293 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x457579 Code: 1d b4 fb ff c3 66 2e 0f 1f 84 00 00 00 00 00 66 90 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 0f 83 eb b3 fb ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:00007f85fa204c78 EFLAGS: 00000246 ORIG_RAX: 0000000000000003 RAX: 0000000000000000 RBX: 0000000000000001 RCX: 0000000000457579 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 0000000000000006 RBP: 000000000072bfa0 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007f85fa2056d4 R13: 00000000004ef912 R14: 00000000004cc460 R15: 00000000ffffffff Kernel Offset: disabled Rebooting in 86400 seconds..

1 0

[alsa-devel] [PATCH] ALSA: hda/sigmatel - Disable automute for Elo VuPoint
by Michael Pobega 05 Oct '18

05 Oct '18

The Elo VuPoint 15MX has two headphone jacks of which neither work by default. Disabling automute allows ALSA to work normally with the speakers & left headphone jack. Future pin configuration changes may be required in the future to get the right headphone jack working in tandem. Signed-off-by: Michael Pobega <mpobega(a)neverware.com> --- sound/pci/hda/patch_sigmatel.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 046705b4691a..d8168aa2cef3 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -77,6 +77,7 @@ enum { STAC_DELL_M6_BOTH, STAC_DELL_EQ, STAC_ALIENWARE_M17X, + STAC_ELO_VUPOINT_15MX, STAC_92HD89XX_HP_FRONT_JACK, STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK, STAC_92HD73XX_ASUS_MOBO, @@ -1879,6 +1880,18 @@ static void stac92hd73xx_fixup_no_jd(struct hda_codec *codec, codec->no_jack_detect = 1; } + +static void stac92hd73xx_disable_automute(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct sigmatel_spec *spec = codec->spec; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + spec->gen.suppress_auto_mute = 1; +} + static const struct hda_fixup stac92hd73xx_fixups[] = { [STAC_92HD73XX_REF] = { .type = HDA_FIXUP_FUNC, @@ -1904,6 +1917,10 @@ static const struct hda_fixup stac92hd73xx_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = stac92hd73xx_fixup_alienware_m17x, }, + [STAC_ELO_VUPOINT_15MX] = { + .type = HDA_FIXUP_FUNC, + .v.func = stac92hd73xx_disable_automute, + }, [STAC_92HD73XX_INTEL] = { .type = HDA_FIXUP_PINS, .v.pins = intel_dg45id_pin_configs, @@ -1942,6 +1959,7 @@ static const struct hda_model_fixup stac92hd73xx_models[] = { { .id = STAC_DELL_M6_BOTH, .name = "dell-m6" }, { .id = STAC_DELL_EQ, .name = "dell-eq" }, { .id = STAC_ALIENWARE_M17X, .name = "alienware" }, + { .id = STAC_ELO_VUPOINT_15MX, .name = "elo-vupoint-15mx" }, { .id = STAC_92HD73XX_ASUS_MOBO, .name = "asus-mobo" }, {} }; @@ -1991,6 +2009,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = { "Alienware M17x", STAC_ALIENWARE_M17X), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490, "Alienware M17x R3", STAC_DELL_EQ), + SND_PCI_QUIRK(0x1059, 0x1011, + "ELO VuPoint 15MX", STAC_ELO_VUPOINT_15MX), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1927, "HP Z1 G2", STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x2b17, -- 2.17.1

2 2

[alsa-devel] Applied "ASoC: adau1761: Use the standard fall-through annotation" to the asoc tree
by Mark Brown 05 Oct '18

05 Oct '18

The patch ASoC: adau1761: Use the standard fall-through annotation has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7e29317928bd79a03a9c35816afa709988b5d31b Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Thu, 4 Oct 2018 20:30:02 +0200 Subject: [PATCH] ASoC: adau1761: Use the standard fall-through annotation As a preparatory patch for the upcoming -Wimplicit-fallthrough compiler checks, replace with the standard "fall through" annotation at the right place. It has to be put right before the next label. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau1761.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index be136e981653..bef3e9e74c26 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -518,7 +518,8 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_component *component) ARRAY_SIZE(adau1761_jack_detect_controls)); if (ret) return ret; - case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: /* fallthrough */ + /* fall through */ + case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: ret = snd_soc_dapm_add_routes(dapm, adau1761_no_dmic_routes, ARRAY_SIZE(adau1761_no_dmic_routes)); if (ret) -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH 0/5] ASoC: Fix fall-through annotations
by Takashi Iwai 05 Oct '18

05 Oct '18

Hi, this is a patch set to add / fix the "fall through" annotations to be compliant with gcc's -Wimplicit-fallthrough option. Takashi === Takashi Iwai (5): ASoC: adau1761: Use the standard fall-through annotation ASoC: pcm186x: Use the standard fall-through annotation ASoC: rt274: Add fall-through annotations ASoC: intel: skylake: Add fall-through annotation ASoC: topology: Use the standard fall-through annotations sound/soc/codecs/adau1761.c | 3 ++- sound/soc/codecs/pcm186x.c | 3 ++- sound/soc/codecs/rt274.c | 2 ++ sound/soc/intel/skylake/skl-pcm.c | 1 + sound/soc/soc-topology.c | 4 ++-- 5 files changed, 9 insertions(+), 4 deletions(-) -- 2.18.0

3 11

[alsa-devel] Applied "ASoC: intel: skylake: Add fall-through annotation" to the asoc tree
by Mark Brown 05 Oct '18

05 Oct '18

The patch ASoC: intel: skylake: Add fall-through annotation has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e4bfd61571f5db5c69a7a49de401543cc7d6c87c Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Thu, 4 Oct 2018 20:30:05 +0200 Subject: [PATCH] ASoC: intel: skylake: Add fall-through annotation As a preparatory patch for the upcoming -Wimplicit-fallthrough compiler checks, add the "fall through" annotation in Intel SST skylake driver. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 00b7a91b18c9..557f80c0bfe5 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -495,6 +495,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, stream->lpib); snd_hdac_ext_stream_set_lpib(stream, stream->lpib); } + /* fall through */ case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH] ASoC: dt-bindings: add dt bindings for adau1977 audio codec
by Alexandru Ardelean 05 Oct '18

05 Oct '18

Add device-tree bindings documentation file for ADAU1977 audio codec. This describes device-tree fields that are already supported by the driver. The driver supports both I2C AND and SPI, and this doc covers both aspects of the DT configuration. Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> --- .../bindings/sound/adi,adau1977.txt | 54 +++++++++++++++++++ 1 file changed, 54 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau1977.txt diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.txt b/Documentation/devicetree/bindings/sound/adi,adau1977.txt new file mode 100644 index 000000000000..e79aeef73f28 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau1977.txt @@ -0,0 +1,54 @@ +Analog Devices ADAU1977/ADAU1978/ADAU1979 + +Datasheets: +http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1977.pdf +http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1978.pdf +http://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1979.pdf + +This driver supports both the I2C and SPI bus. + +Required properties: + - compatible: Should contain one of the following: + "adi,adau1977" + "adi,adau1978" + "adi,adau1979" + + - AVDD-supply: analog power supply for the device, please consult + Documentation/devicetree/bindings/regulator/regulator.txt + +Optional properties: + - reset-gpio: the reset pin for the chip, for more details consult + Documentation/devicetree/bindings/gpio/gpio.txt + + - DVDD-supply: supply voltage for the digital core, please consult + Documentation/devicetree/bindings/regulator/regulator.txt + +For required properties on SPI, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Required properties on I2C: + + - reg: The i2c address. Value depends on the state of ADDR0 + and ADDR1, as wired in hardware. + +Examples: + + adau1977_spi: adau1977@0 { + compatible = "adi,adau1977"; + spi-max-frequency = <600000>; + + AVDD-supply = <&regulator>; + DVDD-supply = <&regulator_digital>; + + reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>; + }; + + adau1977_i2c: adau1977@11 { + compatible = "adi,adau1977"; + reg = <0x11>; + + AVDD-supply = <&regulator>; + DVDD-supply = <&regulator_digital>; + + reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>; + }; -- 2.17.1

2 1

[alsa-devel] [PATCH v5] ALSA: usb-audio: Add custom mixer status quirks for RME CC devices
by Jussi Laako 05 Oct '18

05 Oct '18

Adds several vendor specific mixer quirks for RME's Class Compliant USB devices. These provide extra status information from the device otherwise not available. These include AES/SPDIF rate and status information, current system sampling rate and measured frequency. This information is especially useful in cases where device's clock is slaved to external clock source. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/mixer_quirks.c | 381 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 381 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index cbfb48bdea51..85ae0ff2382a 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -29,6 +29,7 @@ #include <linux/hid.h> #include <linux/init.h> +#include <linux/math64.h> #include <linux/slab.h> #include <linux/usb.h> #include <linux/usb/audio.h> @@ -1817,6 +1818,380 @@ static int dell_dock_mixer_init(struct usb_mixer_interface *mixer) return 0; } +/* RME Class Compliant device quirks */ + +#define SND_RME_GET_STATUS1 23 +#define SND_RME_GET_CURRENT_FREQ 17 +#define SND_RME_CLK_SYSTEM_SHIFT 16 +#define SND_RME_CLK_SYSTEM_MASK 0x1f +#define SND_RME_CLK_AES_SHIFT 8 +#define SND_RME_CLK_SPDIF_SHIFT 12 +#define SND_RME_CLK_AES_SPDIF_MASK 0xf +#define SND_RME_CLK_SYNC_SHIFT 6 +#define SND_RME_CLK_SYNC_MASK 0x3 +#define SND_RME_CLK_FREQMUL_SHIFT 18 +#define SND_RME_CLK_FREQMUL_MASK 0x7 +#define SND_RME_CLK_SYSTEM(x) \ + ((x >> SND_RME_CLK_SYSTEM_SHIFT) & SND_RME_CLK_SYSTEM_MASK) +#define SND_RME_CLK_AES(x) \ + ((x >> SND_RME_CLK_AES_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK) +#define SND_RME_CLK_SPDIF(x) \ + ((x >> SND_RME_CLK_SPDIF_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK) +#define SND_RME_CLK_SYNC(x) \ + ((x >> SND_RME_CLK_SYNC_SHIFT) & SND_RME_CLK_SYNC_MASK) +#define SND_RME_CLK_FREQMUL(x) \ + ((x >> SND_RME_CLK_FREQMUL_SHIFT) & SND_RME_CLK_FREQMUL_MASK) +#define SND_RME_CLK_AES_LOCK 0x1 +#define SND_RME_CLK_AES_SYNC 0x4 +#define SND_RME_CLK_SPDIF_LOCK 0x2 +#define SND_RME_CLK_SPDIF_SYNC 0x8 +#define SND_RME_SPDIF_IF_SHIFT 4 +#define SND_RME_SPDIF_FORMAT_SHIFT 5 +#define SND_RME_BINARY_MASK 0x1 +#define SND_RME_SPDIF_IF(x) \ + ((x >> SND_RME_SPDIF_IF_SHIFT) & SND_RME_BINARY_MASK) +#define SND_RME_SPDIF_FORMAT(x) \ + ((x >> SND_RME_SPDIF_FORMAT_SHIFT) & SND_RME_BINARY_MASK) + +static const u32 snd_rme_rate_table[] = { + 32000, 44100, 48000, 50000, + 64000, 88200, 96000, 100000, + 128000, 176400, 192000, 200000, + 256000, 352800, 384000, 400000, + 512000, 705600, 768000, 800000 +}; +/* maximum number of items for AES and S/PDIF rates for above table */ +#define SND_RME_RATE_IDX_AES_SPDIF_NUM 12 + +enum snd_rme_domain { + SND_RME_DOMAIN_SYSTEM, + SND_RME_DOMAIN_AES, + SND_RME_DOMAIN_SPDIF +}; + +enum snd_rme_clock_status { + SND_RME_CLOCK_NOLOCK, + SND_RME_CLOCK_LOCK, + SND_RME_CLOCK_SYNC +}; + +static int snd_rme_read_value(struct snd_usb_audio *chip, + unsigned int item, + u32 *value) +{ + struct usb_device *dev = chip->dev; + int err; + + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + item, + USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0, 0, + value, sizeof(*value)); + if (err < 0) + dev_err(&dev->dev, + "unable to issue vendor read request %d (ret = %d)", + item, err); + return err; +} + +static int snd_rme_get_status1(struct snd_kcontrol *kcontrol, + u32 *status1) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct snd_usb_audio *chip = list->mixer->chip; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, status1); + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_rme_rate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + u32 rate = 0; + int idx; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_SYSTEM: + idx = SND_RME_CLK_SYSTEM(status1); + if (idx < ARRAY_SIZE(snd_rme_rate_table)) + rate = snd_rme_rate_table[idx]; + break; + case SND_RME_DOMAIN_AES: + idx = SND_RME_CLK_AES(status1); + if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM) + rate = snd_rme_rate_table[idx]; + break; + case SND_RME_DOMAIN_SPDIF: + idx = SND_RME_CLK_SPDIF(status1); + if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM) + rate = snd_rme_rate_table[idx]; + break; + default: + return -EINVAL; + } + ucontrol->value.integer.value[0] = rate; + return 0; +} + +static int snd_rme_sync_state_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int idx = SND_RME_CLOCK_NOLOCK; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_AES: /* AES */ + if (status1 & SND_RME_CLK_AES_SYNC) + idx = SND_RME_CLOCK_SYNC; + else if (status1 & SND_RME_CLK_AES_LOCK) + idx = SND_RME_CLOCK_LOCK; + break; + case SND_RME_DOMAIN_SPDIF: /* SPDIF */ + if (status1 & SND_RME_CLK_SPDIF_SYNC) + idx = SND_RME_CLOCK_SYNC; + else if (status1 & SND_RME_CLK_SPDIF_LOCK) + idx = SND_RME_CLOCK_LOCK; + break; + default: + return -EINVAL; + } + ucontrol->value.enumerated.item[0] = idx; + return 0; +} + +static int snd_rme_spdif_if_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_IF(status1); + return 0; +} + +static int snd_rme_spdif_format_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_FORMAT(status1); + return 0; +} + +static int snd_rme_sync_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_CLK_SYNC(status1); + return 0; +} + +static int snd_rme_current_freq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct snd_usb_audio *chip = list->mixer->chip; + u32 status1; + const u64 num = 104857600000000ULL; + u32 den; + unsigned int freq; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, &status1); + if (err < 0) + goto end; + err = snd_rme_read_value(chip, SND_RME_GET_CURRENT_FREQ, &den); + if (err < 0) + goto end; + freq = (den == 0) ? 0 : div64_u64(num, den); + freq <<= SND_RME_CLK_FREQMUL(status1); + ucontrol->value.integer.value[0] = freq; + +end: + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_rme_rate_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_SYSTEM: + uinfo->value.integer.min = 32000; + uinfo->value.integer.max = 800000; + break; + case SND_RME_DOMAIN_AES: + case SND_RME_DOMAIN_SPDIF: + default: + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 200000; + } + uinfo->value.integer.step = 0; + return 0; +} + +static int snd_rme_sync_state_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const sync_states[] = { + "No Lock", "Lock", "Sync" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(sync_states), sync_states); +} + +static int snd_rme_spdif_if_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const spdif_if[] = { + "Coaxial", "Optical" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(spdif_if), spdif_if); +} + +static int snd_rme_spdif_format_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const optical_type[] = { + "Consumer", "Professional" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(optical_type), optical_type); +} + +static int snd_rme_sync_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const sync_sources[] = { + "Internal", "AES", "SPDIF", "Internal" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(sync_sources), sync_sources); +} + +static struct snd_kcontrol_new snd_rme_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "AES Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_AES + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "AES Sync", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_state_info, + .get = snd_rme_sync_state_get, + .private_value = SND_RME_DOMAIN_AES + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_SPDIF + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Sync", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_state_info, + .get = snd_rme_sync_state_get, + .private_value = SND_RME_DOMAIN_SPDIF + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Interface", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_spdif_if_info, + .get = snd_rme_spdif_if_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Format", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_spdif_format_info, + .get = snd_rme_spdif_format_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Sync Source", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_source_info, + .get = snd_rme_sync_source_get + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "System Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_SYSTEM + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Current Frequency", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_current_freq_get + } +}; + +static int snd_rme_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(snd_rme_controls); ++i) { + err = add_single_ctl_with_resume(mixer, 0, + NULL, + &snd_rme_controls[i], + NULL); + if (err < 0) + return err; + } + + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1904,6 +2279,12 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x0bda, 0x4014): /* Dell WD15 dock */ err = dell_dock_mixer_init(mixer); break; + + case USB_ID(0x2a39, 0x3fd2): /* RME ADI-2 Pro */ + case USB_ID(0x2a39, 0x3fd3): /* RME ADI-2 DAC */ + case USB_ID(0x2a39, 0x3fd4): /* RME */ + err = snd_rme_controls_create(mixer); + break; } return err; -- 2.17.1

2 1

[alsa-devel] [PATCH v2 0/5] ASoC: qdsp6: add compress offload support
by Srinivas Kandagatla 05 Oct '18

05 Oct '18

This patchset adds support to very basic MP3 playback compress offload support via QDSP ASM module. Tested this patchset on DB410c with APQ8016 and DB820c with APQ8096 using tinycompress library. Adding other codec support should be trivial w.r.t qdsp6 side, however there are other dependencies like compress UAPI header changes and FastRPC which are being worked in parallel. Once ready will post them! thanks, srini Changes since v1 (https://patchwork.kernel.org/cover/10585805/) - added support to snd_compr_set_runtime_buffer() - make use of generic copy function - few trivial cleanups suggested by Vinod Koul Srinivas Kandagatla (5): ALSA: soc-compress: add support to snd_compr_set_runtime_buffer() ALSA: compress: make use of runtime buffer for copy ASoC: q6asm-dai: dt-bindings: Add support to compress dais ASoC: qdsp6: q6asm: add support to MP3 format ASoC: qdsp6: q6asm-dai: Add support to compress offload .../devicetree/bindings/sound/qcom,q6asm.txt | 27 ++ include/sound/compress_driver.h | 19 + sound/core/compress_offload.c | 18 +- sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/qdsp6/q6asm-dai.c | 377 +++++++++++++++++- sound/soc/qcom/qdsp6/q6asm.c | 5 + 6 files changed, 443 insertions(+), 4 deletions(-) -- 2.19.0

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[alsa-devel] [PATCH v4] ALSA: usb-audio: Add custom mixer status quirks for RME CC devices
by Jussi Laako 05 Oct '18

05 Oct '18

Adds several vendor specific mixer quirks for RME's Class Compliant USB devices. These provide extra status information from the device otherwise not available. These include AES/SPDIF rate and status information, current system sampling rate and measured frequency. This information is especially useful in cases where device's clock is slaved to external clock source. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/mixer_quirks.c | 380 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 380 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index cbfb48bdea51..6988dc21c31e 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -29,6 +29,7 @@ #include <linux/hid.h> #include <linux/init.h> +#include <linux/math64.h> #include <linux/slab.h> #include <linux/usb.h> #include <linux/usb/audio.h> @@ -1817,6 +1818,379 @@ static int dell_dock_mixer_init(struct usb_mixer_interface *mixer) return 0; } +/* RME Class Compliant device quirks */ + +#define SND_RME_GET_STATUS1 23 +#define SND_RME_GET_CURRENT_FREQ 17 +#define SND_RME_CLK_SYSTEM_SHIFT 16 +#define SND_RME_CLK_SYSTEM_MASK 0x1f +#define SND_RME_CLK_AES_SHIFT 8 +#define SND_RME_CLK_SPDIF_SHIFT 12 +#define SND_RME_CLK_AES_SPDIF_MASK 0xf +#define SND_RME_CLK_SYNC_SHIFT 6 +#define SND_RME_CLK_SYNC_MASK 0x3 +#define SND_RME_CLK_FREQMUL_SHIFT 18 +#define SND_RME_CLK_FREQMUL_MASK 0x7 +#define SND_RME_CLK_SYSTEM(x) \ + ((x >> SND_RME_CLK_SYSTEM_SHIFT) & SND_RME_CLK_SYSTEM_MASK) +#define SND_RME_CLK_AES(x) \ + ((x >> SND_RME_CLK_AES_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK) +#define SND_RME_CLK_SPDIF(x) \ + ((x >> SND_RME_CLK_SPDIF_SHIFT) & SND_RME_CLK_AES_SPDIF_MASK) +#define SND_RME_CLK_SYNC(x) \ + ((x >> SND_RME_CLK_SYNC_SHIFT) & SND_RME_CLK_SYNC_MASK) +#define SND_RME_CLK_FREQMUL(x) \ + ((x >> SND_RME_CLK_FREQMUL_SHIFT) & SND_RME_CLK_FREQMUL_MASK) +#define SND_RME_CLK_AES_LOCK 0x1 +#define SND_RME_CLK_AES_SYNC 0x4 +#define SND_RME_CLK_SPDIF_LOCK 0x2 +#define SND_RME_CLK_SPDIF_SYNC 0x8 +#define SND_RME_SPDIF_IF_SHIFT 4 +#define SND_RME_SPDIF_FORMAT_SHIFT 5 +#define SND_RME_BINARY_MASK 0x1 +#define SND_RME_SPDIF_IF(x) \ + ((x >> SND_RME_SPDIF_IF_SHIFT) & SND_RME_BINARY_MASK) +#define SND_RME_SPDIF_FORMAT(x) \ + ((x >> SND_RME_SPDIF_FORMAT_SHIFT) & SND_RME_BINARY_MASK) + +static const u32 snd_rme_rate_table[] = { + 32000, 44100, 48000, 50000, + 64000, 88200, 96000, 100000, + 128000, 176400, 192000, 200000, + 256000, 352800, 384000, 400000, + 512000, 705600, 768000, 800000 +}; +#define SND_RME_RATE_IDX_AES_SPDIF_NUM 12 + +enum snd_rme_domain { + SND_RME_DOMAIN_SYSTEM, + SND_RME_DOMAIN_AES, + SND_RME_DOMAIN_SPDIF +}; + +enum snd_rme_clock_status { + SND_RME_CLOCK_NOLOCK, + SND_RME_CLOCK_LOCK, + SND_RME_CLOCK_SYNC +}; + +static int snd_rme_read_value(struct snd_usb_audio *chip, + unsigned int item, + u32 *value) +{ + struct usb_device *dev = chip->dev; + int err; + + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + item, + USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0, 0, + value, sizeof(*value)); + if (err < 0) + dev_err(&dev->dev, + "unable to issue vendor read request %d (ret = %d)", + item, err); + return err; +} + +static int snd_rme_get_status1(struct snd_kcontrol *kcontrol, + u32 *status1) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct snd_usb_audio *chip = list->mixer->chip; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, status1); + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_rme_rate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + u32 rate = 0; + int idx; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_SYSTEM: + idx = SND_RME_CLK_SYSTEM(status1); + if (idx < ARRAY_SIZE(snd_rme_rate_table)) + rate = snd_rme_rate_table[idx]; + break; + case SND_RME_DOMAIN_AES: + idx = SND_RME_CLK_AES(status1); + if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM) + rate = snd_rme_rate_table[idx]; + break; + case SND_RME_DOMAIN_SPDIF: + idx = SND_RME_CLK_SPDIF(status1); + if (idx < SND_RME_RATE_IDX_AES_SPDIF_NUM) + rate = snd_rme_rate_table[idx]; + break; + default: + return -EINVAL; + } + ucontrol->value.integer.value[0] = rate; + return 0; +} + +static int snd_rme_sync_state_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int idx = SND_RME_CLOCK_NOLOCK; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_AES: /* AES */ + if (status1 & SND_RME_CLK_AES_SYNC) + idx = SND_RME_CLOCK_SYNC; + else if (status1 & SND_RME_CLK_AES_LOCK) + idx = SND_RME_CLOCK_LOCK; + break; + case SND_RME_DOMAIN_SPDIF: /* SPDIF */ + if (status1 & SND_RME_CLK_SPDIF_SYNC) + idx = SND_RME_CLOCK_SYNC; + else if (status1 & SND_RME_CLK_SPDIF_LOCK) + idx = SND_RME_CLOCK_LOCK; + break; + default: + return -EINVAL; + } + ucontrol->value.enumerated.item[0] = idx; + return 0; +} + +static int snd_rme_spdif_if_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_IF(status1); + return 0; +} + +static int snd_rme_spdif_format_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_SPDIF_FORMAT(status1); + return 0; +} + +static int snd_rme_sync_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status1; + int err; + + err = snd_rme_get_status1(kcontrol, &status1); + if (err < 0) + return err; + ucontrol->value.enumerated.item[0] = SND_RME_CLK_SYNC(status1); + return 0; +} + +static int snd_rme_current_freq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct snd_usb_audio *chip = list->mixer->chip; + u32 status1; + const u64 num = 104857600000000ULL; + u32 den; + unsigned int freq; + int err; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + err = snd_rme_read_value(chip, SND_RME_GET_STATUS1, &status1); + if (err < 0) + goto end; + err = snd_rme_read_value(chip, SND_RME_GET_CURRENT_FREQ, &den); + if (err < 0) + goto end; + freq = (den == 0) ? 0 : div64_u64(num, den); + freq <<= SND_RME_CLK_FREQMUL(status1); + ucontrol->value.integer.value[0] = freq; + +end: + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_rme_rate_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + switch (kcontrol->private_value) { + case SND_RME_DOMAIN_SYSTEM: + uinfo->value.integer.min = 32000; + uinfo->value.integer.max = 800000; + break; + case SND_RME_DOMAIN_AES: + case SND_RME_DOMAIN_SPDIF: + default: + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 200000; + } + uinfo->value.integer.step = 0; + return 0; +} + +static int snd_rme_sync_state_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const sync_states[] = { + "No Lock", "Lock", "Sync" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(sync_states), sync_states); +} + +static int snd_rme_spdif_if_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const spdif_if[] = { + "Coaxial", "Optical" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(spdif_if), spdif_if); +} + +static int snd_rme_spdif_format_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const optical_type[] = { + "Consumer", "Professional" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(optical_type), optical_type); +} + +static int snd_rme_sync_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const sync_sources[] = { + "Internal", "AES", "SPDIF", "Internal" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(sync_sources), sync_sources); +} + +static struct snd_kcontrol_new snd_rme_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "AES Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_AES + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "AES Sync", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_state_info, + .get = snd_rme_sync_state_get, + .private_value = SND_RME_DOMAIN_AES + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_SPDIF + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Sync", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_state_info, + .get = snd_rme_sync_state_get, + .private_value = SND_RME_DOMAIN_SPDIF + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Interface", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_spdif_if_info, + .get = snd_rme_spdif_if_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Format", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_spdif_format_info, + .get = snd_rme_spdif_format_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Sync Source", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_source_info, + .get = snd_rme_sync_source_get + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "System Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_rate_get, + .private_value = SND_RME_DOMAIN_SYSTEM + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Current Frequency", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_current_freq_get + } +}; + +static int snd_rme_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(snd_rme_controls); ++i) { + err = add_single_ctl_with_resume(mixer, 0, + NULL, + &snd_rme_controls[i], + NULL); + if (err < 0) + return err; + } + + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1904,6 +2278,12 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x0bda, 0x4014): /* Dell WD15 dock */ err = dell_dock_mixer_init(mixer); break; + + case USB_ID(0x2a39, 0x3fd2): /* RME ADI-2 Pro */ + case USB_ID(0x2a39, 0x3fd3): /* RME ADI-2 DAC */ + case USB_ID(0x2a39, 0x3fd4): /* RME */ + err = snd_rme_controls_create(mixer); + break; } return err; -- 2.17.1

1 0

[alsa-devel] [PATCH v3 0/1] ALSA: usb-audio: Add custom mixer status quirks for RME CC devices
by Jussi Laako 05 Oct '18

05 Oct '18

Here are the requested updates. I have kept the locking separate from the helper, because in frequency calculation result depends on both values. I can move locking to the helper too if necessary, but I thought it would be logically cleaner this way...

2 2

[alsa-devel] [PATCH] ALSA: usb-audio: update quirk for B&W PX to remove microphone
by Nicolas Huaman 05 Oct '18

05 Oct '18

A quirk in snd-usb-audio was added to automate setting sample rate to 4800k and remove the previously exposed nonfunctional microphone for the Bowers & Wilkins PX: commit 240a8af929c7c57dcde28682725b29cf8474e8e5 https://lore.kernel.org/patchwork/patch/919689/ However the headphones where updated shortly after that to remove the unintentional microphone functionality. I guess because of this the headphones now crash when connecting them via USB while the quirk is active. Dmesg: snd-usb-audio: probe of 2-3:1.0 failed with error -22 usb 2-3: 2:1: cannot get min/max values for control 2 (id 2) This patch removes the microfone and allows the headphones to connect and work out of the box. It is based on the current mainline kernel and successfully applied an tested on my machine (4.18.10.arch1-1). Signed-off-by: Nicolas Huaman <nicolas(a)herochao.de> --- sound/usb/quirks-table.h | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 08aa78007020..849953e5775c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3346,19 +3346,14 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = 0, .type = QUIRK_AUDIO_STANDARD_MIXER, }, - /* Capture */ - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE, - }, /* Playback */ { - .ifnum = 2, + .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, - .iface = 2, + .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_FILL_MAX | -- 2.19.0

3 3

[alsa-devel] [PATCH v2 0/1] ALSA: usb-audio: Add custom mixer status quirks for RME CC devices
by Jussi Laako 04 Oct '18

04 Oct '18

This patch has been updated with following changes and should be now good for merging: - Replace 64-bit division with suitable function to make it available on all architectures - Use specific VID:PID pairs instead of just VID to avoid confusions Last item was changed after discussion with RME.

2 4

[alsa-devel] [PATCH 0/5] ALSA: Fix fall-through annotations
by Takashi Iwai 04 Oct '18

04 Oct '18

Hi, here is some trivial patches to add / fix the fall-through annotations. The patchset for ASoC will follow later. Takashi === Takashi Iwai (5): ALSA: memalloc: Add fall-through annotation ALSA: caiaq: Add fall-through annotation ALSA: seq: oss: Use the standard fall-through annotation ALSA: opti92xx-ad1848: Use the standard fall-through annotation ALSA: au88xx: Add fall-through annotations sound/core/memalloc.c | 1 + sound/core/seq/oss/seq_oss_timer.c | 2 +- sound/isa/opti9xx/opti92x-ad1848.c | 6 ++++-- sound/pci/au88x0/au88x0_core.c | 6 ++++++ sound/usb/caiaq/device.c | 1 + 5 files changed, 13 insertions(+), 3 deletions(-) -- 2.18.0

1 5

[alsa-devel] [PATCH] ASoC: ts3a227e: fix a typo in documentation
by Tzung-Bi Shih 04 Oct '18

04 Oct '18

There is no "MACBIAS". It should be "MICBIAS". Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Documentation/devicetree/bindings/sound/ts3a227e.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt index 3ed8359144d3..21ab45bc7e8f 100644 --- a/Documentation/devicetree/bindings/sound/ts3a227e.txt +++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt @@ -14,7 +14,7 @@ Required properties: Optional properies: - ti,micbias: Intended MICBIAS voltage (datasheet section 9.6.7). - Select 0/1/2/3/4/5/6/7 to specify MACBIAS voltage + Select 0/1/2/3/4/5/6/7 to specify MICBIAS voltage 2.1V/2.2V/2.3V/2.4V/2.5V/2.6V/2.7V/2.8V Default value is "1" (2.2V). -- 2.19.0.605.g01d371f741-goog

2 2

[alsa-devel] [PATCH] ASoC: intel: skylake: Add missing break in skl_tplg_get_token()
by Takashi Iwai 04 Oct '18

04 Oct '18

skl_tplg_get_token() misses a break in the big switch() block for SKL_TKN_U8_CORE_ID entry. Spotted nicely by -Wimplicit-fallthrough compiler option. Fixes: 6277e83292a2 ("ASoC: Intel: Skylake: Parse vendor tokens to build module data") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/skylake/skl-topology.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 2620d77729c5..f99c600f86e4 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2461,6 +2461,7 @@ static int skl_tplg_get_token(struct device *dev, case SKL_TKN_U8_CORE_ID: mconfig->core_id = tkn_elem->value; + break; case SKL_TKN_U8_MOD_TYPE: mconfig->m_type = tkn_elem->value; -- 2.18.0

2 1

[alsa-devel] [PATCH] ASoC: qdsp6: q6asm-dai: checking NULL vs IS_ERR()
by Dan Carpenter 04 Oct '18

04 Oct '18

The q6asm_audio_client_alloc() doesn't return NULL, it returns error pointers. Fixes: 2a9e92d371db ("ASoC: qdsp6: q6asm: Add q6asm dai driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index c3806d7037fc..a16c71c03058 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -318,10 +318,11 @@ static int q6asm_dai_open(struct snd_pcm_substream *substream) prtd->audio_client = q6asm_audio_client_alloc(dev, (q6asm_cb)event_handler, prtd, stream_id, LEGACY_PCM_MODE); - if (!prtd->audio_client) { + if (IS_ERR(prtd->audio_client)) { pr_info("%s: Could not allocate memory\n", __func__); + ret = PTR_ERR(prtd->audio_client); kfree(prtd); - return -ENOMEM; + return ret; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)

2 1

[alsa-devel] [PATCH] ASoC: pxa-ssp: enable and disable extclk if given
by Daniel Mack 04 Oct '18

04 Oct '18

If a "extclk" clock is given, enable and disable it when appropriate. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/pxa/pxa-ssp.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 69033e1a84e6..adcf8ba9d287 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -103,6 +103,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, pxa_ssp_disable(ssp); } + if (priv->extclk) + clk_prepare_enable(priv->extclk); + dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL); if (!dma) return -ENOMEM; @@ -125,6 +128,9 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(ssp->clk); } + if (priv->extclk) + clk_disable_unprepare(priv->extclk); + kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); snd_soc_dai_set_dma_data(cpu_dai, substream, NULL); } -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: add fault detect recovery property to DT bindings
by Daniel Mack 04 Oct '18

04 Oct '18

The driver already has support for setting the FDRB bit in the CONFA register through platform data, but there was no property to set it in the device-tree bindings. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- Documentation/devicetree/bindings/sound/st,sta32x.txt | 3 +++ sound/soc/codecs/sta32x.c | 2 ++ 2 files changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/st,sta32x.txt b/Documentation/devicetree/bindings/sound/st,sta32x.txt index 255de3ae5b2f..ff4a685a4303 100644 --- a/Documentation/devicetree/bindings/sound/st,sta32x.txt +++ b/Documentation/devicetree/bindings/sound/st,sta32x.txt @@ -39,6 +39,9 @@ Optional properties: - st,thermal-warning-recover: If present, thermal warning recovery is enabled. + - st,fault-detect-recovery: + If present, fault detect recovery is enabled. + - st,thermal-warning-adjustment: If present, thermal warning adjustment is enabled. diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index d5035f2f2b2b..22de1593443c 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -1038,6 +1038,8 @@ static int sta32x_probe_dt(struct device *dev, struct sta32x_priv *sta32x) of_property_read_u8(np, "st,ch3-output-mapping", &pdata->ch3_output_mapping); + if (of_get_property(np, "st,fault-detect-recovery", NULL)) + pdata->fault_detect_recovery = 1; if (of_get_property(np, "st,thermal-warning-recovery", NULL)) pdata->thermal_warning_recovery = 1; if (of_get_property(np, "st,thermal-warning-adjustment", NULL)) -- 2.17.1

2 1

[alsa-devel] [BUG] sound: pci: trident: a possible data race
by Jia-Ju Bai 04 Oct '18

04 Oct '18

CPU0: snd_trident_hw_free snd_trident_free_voice line 3870: spin_lock_irqsave() line 3881: voice->substream = NULL; [WRITE] CPU1: snd_trident_interrupt line 3798: snd_pcm_period_elapsed(voice->substream); [READ] As for voice->substream, the WRITE operation in CPU0 is performed with holding a spinlock, but the READ operation in CPU1 is performed without holding this spinlock, so there may exist a data race. Best wishes, Jia-Ju Bai

2 4

[alsa-devel] Problem and Patch for B&W PX Bluetooth Headphones
by Nicolas Herochao 04 Oct '18

04 Oct '18

Hello, I'm writing this mail because of a Crash when connecting the B&W PX headphones via usb. Half a year ago this used to work with Linux after setting the default sample rate of pulseaudio to 48000k. A Launchpad Bug seems to describe this exact issue (detailed description below): https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1773697. It pointed me to the right direction, and today I have successfully compiled and tested a small patch that fixes the errors while also setting the sample rate to 48000k, making these headphones plug&play on any newer firmware. They sound awesome via USB! I would like to submit this patch, but it would be my first contribution and it could potentially break the B&W PX on older firmware. I can't test it since I see no obvious ways to downgrade the firmware. The automatic detection of these headphones always worked except for the sample-rate and perhaps the worst result is that old firmware will show a non-functional microphone, like on Windows. So I decided to first present the patch on this mailing list and ask for feedback and help. If anyone wants to pick the patch up, I'd appreciate that as well. My questions: * Can a quirk somehow handle multiple firmware versions of a usb device? * If not: Will this quirk allow old firmware to work out of the box but show a non-functional microphone? (And is that okay?) Thanks for your time and any feedback you can provide! Sincerely, Nicolas PS: more detail on the cause of the issue: ------ A quirk in snd-usb-audio was added to automate setting sample rate to 4800k and remove the previously exposed nonfunctional microphone: commit 240a8af929c7c57dcde28682725b29cf8474e8e5 https://lore.kernel.org/patchwork/patch/919689/ However the headphones where updated shortly after that to remove the unintentional microphone functionality. I guess because of this the headphones now crash when connecting them via USB while the quirk is active with the following dmesg errors: snd-usb-audio: probe of 2-3:1.0 failed with error -22 usb 2-3: 2:1: cannot get min/max values for control 2 (id 2) --------------- Finally, here is the patch (based on the current mainline kernel and generated according to all Guidelines I could find) which fixes the quirk by removing the microphone part: ---------------- From 748587283db85e219902abf4ac012864005943c8 Mon Sep 17 00:00:00 2001 From: Nicolas Huaman <nicolas(a)herochao.de> Date: Fri, 28 Sep 2018 18:05:15 +0200 Subject: [PATCH 1/1] SOUND: update quirk for B&W PX to remove microphone Signed-off-by: Nicolas Huaman <nicolas(a)herochao.de> --- sound/usb/quirks-table.h | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 08aa78007020..849953e5775c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3346,19 +3346,14 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = 0, .type = QUIRK_AUDIO_STANDARD_MIXER, }, - /* Capture */ - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE, - }, /* Playback */ { - .ifnum = 2, + .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, - .iface = 2, + .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_FILL_MAX | -- 2.19.0

2 1

[alsa-devel] [PATCH 0/5] ALSA: use managed-resource to maintain allocated buffers
by Takashi Sakamoto 04 Oct '18

04 Oct '18

Hi, Linux Device Driver subsystem has a feature of 'devres', which is resource allocator/deallocator synchronously to reference counter of kobject corresponding to device structure. This feature avoid memory leak from programming mistakes. This patchset replaces calls of allocator/deallocator with devres helper functions. There're two types of target device structure; firewire unit device and sound card device. All drivers in ALSA firewire stack use reference counting to serialize timing to release the type of two device, therefore no worries of dereferencing to released memory objects. This patchset is for v4.20 and based on my series of fix patches in 'sound-4.19-rc5' branch. But current 'for-next' branch doesn't include them. So this patchset is rebased to upstream 'master' branch (HEAD=067284726963). I'm OK to take them pending till the '-rc5' is merged. Takashi Sakamoto (5): ALSA: firewire: use managed-resource of fw unit device for private data ALSA: bebob: use managed-resource to maintain data specific to M-Audio FW-1814/ProjectMix I/O ALSA: fireworks: use managed-resource to maintain response buffer ALSA: oxfw: use managed-resource to maintain model-specific data ALSA: oxfw: use managed-resource to maintain cache of stream formats sound/firewire/bebob/bebob.c | 17 ++++++----------- sound/firewire/bebob/bebob_maudio.c | 5 +++-- sound/firewire/dice/dice.c | 8 +++----- sound/firewire/digi00x/digi00x.c | 9 ++++----- sound/firewire/fireface/ff.c | 10 +++------- sound/firewire/fireworks/fireworks.c | 15 +++++---------- sound/firewire/motu/motu.c | 10 ++++------ sound/firewire/oxfw/oxfw-scs1x.c | 5 +++-- sound/firewire/oxfw/oxfw-spkr.c | 5 +++-- sound/firewire/oxfw/oxfw-stream.c | 13 ++++++++----- sound/firewire/oxfw/oxfw.c | 28 ++++------------------------ sound/firewire/tascam/tascam.c | 10 +++------- 12 files changed, 49 insertions(+), 86 deletions(-) -- 2.17.1

2 6

[alsa-devel] [PATCH] ALSA: hda/realtek - Cannot adjust speaker's volume on Dell XPS 27 7760
by Kai-Heng Feng 04 Oct '18

04 Oct '18

The issue is the same as commit dd9aa335c880 ("ALSA: hda/realtek - Can't adjust speaker's volume on a Dell AIO"), the output requires to connect to a node with Amp-out capability. Applying the same fixup ALC298_FIXUP_SPK_VOLUME can fix the issue. BugLink: https://bugs.launchpad.net/bugs/1775068 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6f3c8e888c2a..69283633ae91 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6409,6 +6409,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3), SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), -- 2.17.1

2 1

[alsa-devel] [PATCH 0/7] Add devres support to card resources
by Takashi Iwai 04 Oct '18

04 Oct '18

Hi, here is the patchset containing the changes to support a device-managed resource management for ALSA card and memory pages. The actual code changes are identical with the previous RFC series, but a few documentation updates are included in this official set. Takashi === Takashi Iwai (7): ALSA: core: Add device-managed page allocator helper ALSA: core: Add managed card creation ALSA: intel8x0: Allocate resources with device-managed APIs ALSA: atiixp: Allocate resources with device-managed APIs ALSA: hda: Allocate resources with device-managed APIs ALSA: doc: Brush up the old writing-an-alsa-driver ALSA: doc: Add device-managed resource section .../kernel-api/writing-an-alsa-driver.rst | 340 ++++++++++-------- include/sound/core.h | 5 + include/sound/memalloc.h | 4 + sound/core/init.c | 95 ++++- sound/core/memalloc.c | 88 ++++- sound/pci/atiixp.c | 104 ++---- sound/pci/atiixp_modem.c | 104 ++---- sound/pci/hda/hda_controller.h | 1 - sound/pci/hda/hda_intel.c | 43 +-- sound/pci/intel8x0.c | 143 +++----- sound/pci/intel8x0m.c | 143 +++----- 11 files changed, 514 insertions(+), 556 deletions(-) -- 2.18.0

2 19

[alsa-devel] [PATCH linux-next v2 0/9] ASoC: rsnd: support to use different BUSIF for GEN3
by jiada_wang＠mentor.com 04 Oct '18

04 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> Rcar audio DMA/SSIU already support to use different BUSIF to transfer data, but BUSIF is fixed at 0. This patchset adds busif property to dai stream, GEN3 series can automatically select BUSIF to transfer data, based on channel and SSI mode (currently only BUSIF0 is selected). Device tree files are updated to use extended audio dmac register, and add dma request number for busif0 ~ busif7 for GEN3. --- v2: - Instead of configure BUSIF by kctl interface "SSI Out/In BUSIF", use rsnd_ssi_select_busif() to automatically select BUSIF. - Keep 'rxu/txu' dma request, to keep compatibility with GEN2 series - Re-order patch-set to satisfy git bisect v1: initial version Jiada Wang (9): arm64: r8a7795: add dma request number for busif0 ~ busif7 arm64: r8a7796: add dma request number for busif0 ~ busif7 arm64: r8a774a1: add dma request number for busif0 ~ busif7 arm64: r8a7795: change to use extended audio dmac register arm64: r8a7796: change to use extended audio dmac register arm64: r8a774a1: change to use extended audio dmac register arm64: r8a77965: change to use extended audio dmac register ASoC: rsnd: ssi: Request dedicated dma channels for busif0 to 7 ASoC: rsnd: add busif property to dai stream arch/arm64/boot/dts/renesas/r8a774a1.dtsi | 166 +++++++++++++++++++--- arch/arm64/boot/dts/renesas/r8a7795.dtsi | 166 +++++++++++++++++++--- arch/arm64/boot/dts/renesas/r8a7796.dtsi | 166 +++++++++++++++++++--- arch/arm64/boot/dts/renesas/r8a77965.dtsi | 2 +- sound/soc/sh/rcar/core.c | 3 + sound/soc/sh/rcar/dma.c | 31 ++++ sound/soc/sh/rcar/rsnd.h | 3 + sound/soc/sh/rcar/ssi.c | 58 +++++++- 8 files changed, 525 insertions(+), 70 deletions(-) -- 2.17.0

3 6

[alsa-devel] [PATCH linux-next v2 8/9] ASoC: rsnd: ssi: Request dedicated dma channels for busif0 to 7
by jiada_wang＠mentor.com 04 Oct '18

04 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> Currently ssi driver only request dma channel for SSI_0, which is used to transfer data to/from busif0. But in GEN3 busif1 to busif7 also maybe used, dedicated dma channels are requested for data transfer through these busif. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/ssi.c | 28 +++++++++++++++++++++++----- 1 file changed, 23 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 3adcc4f778f7..992aeac09e76 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -76,6 +76,8 @@ #define SSI_NAME "ssi" +#define SSI_DMA_NAME_SIZE 10 + struct rsnd_ssi { struct rsnd_mod mod; @@ -938,12 +940,28 @@ static struct dma_chan *rsnd_ssi_dma_req(struct rsnd_dai_stream *io, { struct rsnd_priv *priv = rsnd_mod_to_priv(mod); int is_play = rsnd_io_is_play(io); - char *name; + char name[SSI_DMA_NAME_SIZE]; - if (rsnd_ssi_use_busif(io)) - name = is_play ? "rxu" : "txu"; - else - name = is_play ? "rx" : "tx"; + if (rsnd_ssi_use_busif(io)) { + if (is_play) + snprintf(name, SSI_DMA_NAME_SIZE, "rxu"); + else + snprintf(name, SSI_DMA_NAME_SIZE, "txu"); + + if (rsnd_is_gen3(priv)) { + /* + * Gen3 supports to use all available BUSIF + */ + int busif = rsnd_ssi_get_busif(io); + + snprintf(name, SSI_DMA_NAME_SIZE, "%s%d", name, busif); + } + } else { + if (is_play) + snprintf(name, SSI_DMA_NAME_SIZE, "rx"); + else + snprintf(name, SSI_DMA_NAME_SIZE, "tx"); + } return rsnd_dma_request_channel(rsnd_ssi_of_node(priv), mod, name); -- 2.17.0

3 2

[alsa-devel] [PATCH] ALSA: intel8x0: Fix fall-through annotations
by Gustavo A. R. Silva 03 Oct '18

03 Oct '18

Replace "fallthru" with a proper "fall through" annotation. This fix is part of the ongoing efforts to enabling -Wimplicit-fallthrough Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/pci/intel8x0.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 9517f9b..ffddcdf 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -825,7 +825,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: ichdev->suspended = 0; - /* fallthru */ + /* fall through */ case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: val = ICH_IOCE | ICH_STARTBM; @@ -833,7 +833,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd break; case SNDRV_PCM_TRIGGER_SUSPEND: ichdev->suspended = 1; - /* fallthru */ + /* fall through */ case SNDRV_PCM_TRIGGER_STOP: val = 0; break; @@ -867,7 +867,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: ichdev->suspended = 0; - /* fallthru */ + /* fall through */ case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -884,7 +884,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd break; case SNDRV_PCM_TRIGGER_SUSPEND: ichdev->suspended = 1; - /* fallthru */ + /* fall through */ case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* pause */ -- 2.7.4

2 5

[alsa-devel] [PATCH linux-next v2 7/9] arm64: r8a77965: change to use extended audio dmac register
by jiada_wang＠mentor.com 03 Oct '18

03 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> Basic audio dmac register only supports busif from 0 to 3, in order to use busif4 ~ busif7, extended audio dmac register need to be used. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- arch/arm64/boot/dts/renesas/r8a77965.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/renesas/r8a77965.dtsi b/arch/arm64/boot/dts/renesas/r8a77965.dtsi index fd00010a5ed7..6c0d5053a81c 100644 --- a/arch/arm64/boot/dts/renesas/r8a77965.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a77965.dtsi @@ -1405,7 +1405,7 @@ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + <0 0xec760000 0 0x200>; /* Audio DMAC peri peri*/ /* placeholder */ rcar_sound,dvc { -- 2.17.0

1 0

[alsa-devel] [PATCH linux-next v2 6/9] arm64: r8a774a1: change to use extended audio dmac register
by jiada_wang＠mentor.com 03 Oct '18

03 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> Basic audio dmac register only supports busif from 0 to 3, in order to use busif4 ~ busif7, extended audio dmac register need to be used. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- arch/arm64/boot/dts/renesas/r8a774a1.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/renesas/r8a774a1.dtsi b/arch/arm64/boot/dts/renesas/r8a774a1.dtsi index ad2c9c0148ae..c04ca03f2987 100644 --- a/arch/arm64/boot/dts/renesas/r8a774a1.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a774a1.dtsi @@ -1120,7 +1120,7 @@ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + <0 0xec760000 0 0x200>; /* Audio DMAC peri peri*/ reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; clocks = <&cpg CPG_MOD 1005>, -- 2.17.0

1 0

[alsa-devel] [PATCH linux-next v2 5/9] arm64: r8a7796: change to use extended audio dmac register
by jiada_wang＠mentor.com 03 Oct '18

03 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> Basic audio dmac register only supports busif from 0 to 3, in order to use busif4 ~ busif7, extended audio dmac register need to be used. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- arch/arm64/boot/dts/renesas/r8a7796.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/renesas/r8a7796.dtsi b/arch/arm64/boot/dts/renesas/r8a7796.dtsi index 53b361c5b16d..4526f62ee1f0 100644 --- a/arch/arm64/boot/dts/renesas/r8a7796.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a7796.dtsi @@ -1670,7 +1670,7 @@ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + <0 0xec760000 0 0x200>; /* Audio DMAC peri peri*/ reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; clocks = <&cpg CPG_MOD 1005>, -- 2.17.0

1 0

[alsa-devel] snd_pcm_poll_descriptors() error return
by Alan Horstmann 02 Oct '18

02 Oct '18

Hi, A user of the Portaudio library is reporting that from Alsa version 1.1.4 the function snd_pcm_poll_descriptors() can return -32 (Broken Pipe), and this fails an assert within Portaudio. Earlier 1.1.3 does not seem to show this issue. For more details, see https://app.assembla.com/spaces/portaudio/support/tickets/268. The Alsa docs don't mention any possible negative error code from this function - has the behaviour changed, or is it likely something else is triggering the difference? Regards Alan

3 3

[alsa-devel] Thread spinning in kernel snd_pcm_link()/snd_pcm_unlink()
by Rob Duncan 02 Oct '18

02 Oct '18

I'm trying to address a bug where we end up with a thread spinning and consuming an entire cpu. The issue seems to be this code in sound/core/pcm_native.c: /* Writer in rwsem may block readers even during its waiting in queue, * and this may lead to a deadlock when the code path takes read sem * twice (e.g. one in snd_pcm_action_nonatomic() and another in * snd_pcm_stream_lock()). As a (suboptimal) workaround, let writer to * spin until it gets the lock. */ static inline void down_write_nonblock(struct rw_semaphore *lock) { while (!down_write_trylock(lock)) cond_resched(); } The original commit for this is 67ec1072b053c15564e6090ab30127895dc77a89 What we're suspecting is that a normal thread (SCHED_OTHER) has a reader lock and a real-time thread using SCHED_RR or SCHED_FIFO is trying to take the writer lock. If both threads are pinned to the same CPU for some reason then the reader thread will never get scheduled (because the real-time writer thread is still runnable), and we will never make progress. Does this sound right? What can we do to fix this? Thanks, Rob.

2 2

[alsa-devel] [PATCH alsa-lib] ioplug: Fix comment in struct snd_pcm_ioplug
by Andre Guedes 02 Oct '18

02 Oct '18

This patch fixes the comment about 'pcm' field in 'struct snd_pcm_ioplug' which wrongly refers to snd_pcm_extplug_create(). Signed-off-by: Andre Guedes <andre.guedes(a)intel.com> --- include/pcm_ioplug.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/pcm_ioplug.h b/include/pcm_ioplug.h index b16fc8b8..81ac8613 100644 --- a/include/pcm_ioplug.h +++ b/include/pcm_ioplug.h @@ -105,7 +105,7 @@ struct snd_pcm_ioplug { */ void *private_data; /** - * PCM handle filled by #snd_pcm_extplug_create() + * PCM handle filled by #snd_pcm_ioplug_create() */ snd_pcm_t *pcm; -- 2.14.4

3 2

[alsa-devel] [PATCH - Multiple plugins 0/4] Fix leaks on multiple plugins
by Andre Guedes 02 Oct '18

02 Oct '18

Hi all, While I was working on AAF plugin RFC v2 series [1], I spotted several leaks in some plugins when handling *_hw_constraint() error cases. This series fixes these issues. Regards, Andre [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-September/140290.… Andre Guedes (4): a52: Fix leaks when a52_set_hw_constraint() fails jack: Fix leaks when jack_set_hw_constraint() fails oss: Fix leaks when oss_hw_constraint() fails usb_stream: Fix leaks when us_set_hw_constraint() fails a52/pcm_a52.c | 2 +- jack/pcm_jack.c | 1 + oss/pcm_oss.c | 2 +- usb_stream/pcm_usb_stream.c | 1 + 4 files changed, 4 insertions(+), 2 deletions(-) -- 2.14.4

3 6

[alsa-devel] [PATCH] ALSA: sb8: Fix fall-through annotations
by Gustavo A. R. Silva 02 Oct '18

02 Oct '18

Replace "fallthru" with a proper "fall through" annotation. This fix is part of the ongoing efforts to enabling -Wimplicit-fallthrough Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/isa/sb/sb8_main.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index 4817977..8288fae 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -130,13 +130,13 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; break; } - /* fallthru */ + /* fall through */ case SB_HW_201: if (rate > 23000) { chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; break; } - /* fallthru */ + /* fall through */ case SB_HW_20: chip->playback_format = SB_DSP_LO_OUTPUT_AUTO; break; @@ -287,7 +287,7 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) chip->capture_format = SB_DSP_HI_INPUT_AUTO; break; } - /* fallthru */ + /* fall through */ case SB_HW_20: chip->capture_format = SB_DSP_LO_INPUT_AUTO; break; @@ -387,7 +387,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) case SB_MODE_PLAYBACK_16: /* ok.. playback is active */ if (chip->hardware != SB_HW_JAZZ16) break; - /* fallthru */ + /* fall through */ case SB_MODE_PLAYBACK_8: substream = chip->playback_substream; if (chip->playback_format == SB_DSP_OUTPUT) @@ -397,7 +397,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) case SB_MODE_CAPTURE_16: if (chip->hardware != SB_HW_JAZZ16) break; - /* fallthru */ + /* fall through */ case SB_MODE_CAPTURE_8: substream = chip->capture_substream; if (chip->capture_format == SB_DSP_INPUT) -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: firewire-lib: add PCM rules to obsolete PCM constraints based on LCM of SYT_INTERVAL
by Takashi Sakamoto 02 Oct '18

02 Oct '18

In blocking mode of IEC 61883-1/6, when one isochronous packet includes data for events, the data is for the same number of events as SYT_INTERVAL decided according to sampling transmission frequency (SFC). IEC 61883-1/6 engine of ALSA firewire stack applies constraints of period and buffer size of PCM intermediate buffer of PCM substream. At present, this constraint is designed to round the size up/down to 32 frames. This value comes from the least common multiple (LCM) of SYT_INTERVAL. Although this looks to work well, in lower sampling rate, applications are not allowed to set size of period quite near period time constraint (at present 5 msec per period). This commit adds PCM rules for period/buffer size and rate to obsoletes the constraints based on LCM. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 78 +++++++++++++++++++++++++++++++---- 1 file changed, 71 insertions(+), 7 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index cb9acfe60f6a..fcd965f1d69e 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -140,6 +140,59 @@ const unsigned int amdtp_rate_table[CIP_SFC_COUNT] = { }; EXPORT_SYMBOL(amdtp_rate_table); +static int apply_constraint_to_size(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *s = hw_param_interval(params, rule->var); + const struct snd_interval *r = + hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval t = { + .min = s->min, .max = s->max, .integer = 1, + }; + int i; + + for (i = 0; i < CIP_SFC_COUNT; ++i) { + unsigned int rate = amdtp_rate_table[i]; + unsigned int step = amdtp_syt_intervals[i]; + + if (!snd_interval_test(r, rate)) + continue; + + t.min = roundup(t.min, step); + t.max = rounddown(t.max, step); + } + + if (snd_interval_checkempty(&t)) + return -EINVAL; + + return snd_interval_refine(s, &t); +} + +static int apply_constraint_to_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *r = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + const struct snd_interval *s = hw_param_interval_c(params, rule->deps[0]); + struct snd_interval t = { + .min = UINT_MAX, .max = 0, .integer = 1, + }; + int i; + + for (i = 0; i < CIP_SFC_COUNT; ++i) { + unsigned int step = amdtp_syt_intervals[i]; + unsigned int rate = amdtp_rate_table[i]; + + if (s->min % step || s->max % step) + continue; + + t.min = min(t.min, rate); + t.max = max(t.max, rate); + } + + return snd_interval_refine(r, &t); +} + /** * amdtp_stream_add_pcm_hw_constraints - add hw constraints for PCM substream * @s: the AMDTP stream, which must be initialized. @@ -194,16 +247,27 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, * number equals to SYT_INTERVAL. So the number is 8, 16 or 32, * depending on its sampling rate. For accurate period interrupt, it's * preferrable to align period/buffer sizes to current SYT_INTERVAL. - * - * TODO: These constraints can be improved with proper rules. - * Currently apply LCM of SYT_INTERVALs. */ - err = snd_pcm_hw_constraint_step(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32); + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + apply_constraint_to_size, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + goto end; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + apply_constraint_to_rate, NULL, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); + if (err < 0) + goto end; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + apply_constraint_to_size, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + goto end; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + apply_constraint_to_rate, NULL, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1); if (err < 0) goto end; - err = snd_pcm_hw_constraint_step(runtime, 0, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 32); end: return err; } -- 2.17.1

2 1

[alsa-devel] [PATCH][alsa-lib] timer: fix wrong comment to refer to 'SNDRV_TIMER_PSFLG_*'
by Takashi Sakamoto 02 Oct '18

02 Oct '18

ALSA timer core has a comment referring to 'SNDRV_MIXER_PSFLG_*' in a definition of 'struct snd_timer_params' of UAPI header. I can see this in initial state of ALSA timer core, at least in 'alsa-driver-0.4.0.tar.gz'. This commit fixes the comment. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 6041cab2..4d35c84d 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -750,7 +750,7 @@ struct snd_timer_info { #define SNDRV_TIMER_PSFLG_EARLY_EVENT (1<<2) /* write early event to the poll queue */ struct snd_timer_params { - unsigned int flags; /* flags - SNDRV_MIXER_PSFLG_* */ + unsigned int flags; /* flags - SNDRV_TIMER_PSFLG_* */ unsigned int ticks; /* requested resolution in ticks */ unsigned int queue_size; /* total size of queue (32-1024) */ unsigned int reserved0; /* reserved, was: failure locations */ -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: timer: fix wrong comment to refer to 'SNDRV_TIMER_PSFLG_*'
by Takashi Sakamoto 02 Oct '18

02 Oct '18

ALSA timer core has a comment referring to 'SNDRV_MIXER_PSFLG_*' in a definition of 'struct snd_timer_params' of UAPI header. I can see this in initial state of ALSA timer core, at least in 'alsa-driver-0.4.0.tar.gz'. This commit fixes the comment. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/uapi/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index ed0a120d4f08..404d4b9ffe76 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -752,7 +752,7 @@ struct snd_timer_info { #define SNDRV_TIMER_PSFLG_EARLY_EVENT (1<<2) /* write early event to the poll queue */ struct snd_timer_params { - unsigned int flags; /* flags - SNDRV_MIXER_PSFLG_* */ + unsigned int flags; /* flags - SNDRV_TIMER_PSFLG_* */ unsigned int ticks; /* requested resolution in ticks */ unsigned int queue_size; /* total size of queue (32-1024) */ unsigned int reserved0; /* reserved, was: failure locations */ -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: hda: Fix mismatch for register mask and value in ext controller.
by Keyon Jie 02 Oct '18

02 Oct '18

E.g. for snd_hdac_ext_bus_link_power_up(), we should set mask to be AZX_MLCTL_SPA(it was 0), and AZX_MLCTL_SPA as value to power up it, here correct it and several similar mismatches. Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> --- sound/hda/ext/hdac_ext_controller.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 5bc4a1d587d4..60cb00fd0c69 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -48,9 +48,11 @@ void snd_hdac_ext_bus_ppcap_enable(struct hdac_bus *bus, bool enable) } if (enable) - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_GPROCEN); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_GPROCEN, AZX_PPCTL_GPROCEN); else - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_GPROCEN, 0); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_GPROCEN, 0); } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_enable); @@ -68,9 +70,11 @@ void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_bus *bus, bool enable) } if (enable) - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_PIE); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_PIE, AZX_PPCTL_PIE); else - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_PIE, 0); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_PIE, 0); } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_int_enable); @@ -194,7 +198,8 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) */ int snd_hdac_ext_bus_link_power_up(struct hdac_ext_link *link) { - snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA); + snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, + AZX_MLCTL_SPA, AZX_MLCTL_SPA); return check_hdac_link_power_active(link, true); } @@ -222,8 +227,8 @@ int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus) int ret; list_for_each_entry(hlink, &bus->hlink_list, list) { - snd_hdac_updatel(hlink->ml_addr, - AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA); + snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, + AZX_MLCTL_SPA, AZX_MLCTL_SPA); ret = check_hdac_link_power_active(hlink, true); if (ret < 0) return ret; @@ -243,7 +248,8 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus) int ret; list_for_each_entry(hlink, &bus->hlink_list, list) { - snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, AZX_MLCTL_SPA, 0); + snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, + AZX_MLCTL_SPA, 0); ret = check_hdac_link_power_active(hlink, false); if (ret < 0) return ret; -- 2.17.1

2 1

[alsa-devel] Re-reading alsa-lib configuration when opening sound device
by Pavel Hofman 02 Oct '18

02 Oct '18

Hi, I am playing with virtual balanced inputs https://www.diyaudio.com/forums/pc-based/326747-virtual-balanced-regular-so… . For proper functionality it needs a regular recalibration of the ttable multiplication coefficients of route plugin in .asoundrc. The audio stack is alsa -> wine -> Arta measurement software. When I stop measuring in arta (without exiting the software), the soundcard gets closed - confirmed in /proc/asound/.../hw_params. Now I change the ttable coeffs. After re-opening the device (continuing with measurement in the running arta instance) the changes in .asoundrc are ignored. I have to restart the arta application for the changes to take effect. When I run e.g. aplay inbetween, it accepts the changes right away. Upon continuing in arta, the original ttable coeffs are still loaded for this instance. I understand it is the way wine handles the device somewhere in https://github.com/wine-mirror/wine/blob/master/dlls/winealsa.drv/mmdevdrv.c . Please when is the alsa-lib config loaded by the library/application? Is there any way to force a config reload from outside? Would e.g. calling snd_config_update before snd_pcm_open on line https://github.com/wine-mirror/wine/blob/master/dlls/winealsa.drv/mmdevdrv.… help? If so, what overhead does it impose? Perhaps I could ask the wine guys to consider such change if it worked and required minimum overhead. I very much appreciate any help. Best regards, Pavel.

2 4

[alsa-devel] alsactl monitor spins at 100% cpu after unplugging USB headphones
by Thomas Gläßle 02 Oct '18

02 Oct '18

Hi, on both my desktop and laptop (archlinux), `alsactl monitor` beginns spinning at 100% cpu when I unplug my USB headset. This happens every time and can be reproduced as follows: - plug in USB headphones - start `alsactl monitor` - unplug headphones It happens only if the headphones are already plugged in when starting alsactl. The increased activity does not lead to spamming stdout or similar, alsactl continues to perform seemingly correct, i.e. reports when I change volume levels etc, but nothing apart from the ordinary. I'm sorry if this is not the right place to report this issue or if the problem is already, I couldn't find anything about it. Below is the output from alsa-info.sh: with headphones: http://www.alsa-project.org/db/?f=0dfc632e3adafb8bc78a182e0a225bcb13bea9fa after unplugging: http://www.alsa-project.org/db/?f=41dbf0735fdd7cd2416a6b4a602505715d25b44f Please let me know if you need further info. Best regards, Thomas

2 2

[alsa-devel] [PATCH linux-next 10/10] ASoC: rsnd: Add kctrl to configure dai's busif
by jiada_wang＠mentor.com 02 Oct '18

02 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> ssi0, ssi1, ssi2, ssi3, ssi4 and ssi9 have busif0 to busif7, currently only busif0 is used by all ssi. But tdm split and tdm ex-split mode, which require busif1 to busif7. This patch adds kctrl in ssi for dai-link, to make it possible to configure the busif used by each dai-link. Also adds rsnd_dma_addr_update() interface to make it possible to update dma data address when there is change in condition of dai connection. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/dma.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 2 ++ sound/soc/sh/rcar/ssi.c | 31 ++++++++++++++++++++++++++++++- 3 files changed, 64 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 6d1947515dc8..8c284fd73fc1 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -792,6 +792,38 @@ int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, return rsnd_dai_connect(*dma_mod, io, (*dma_mod)->type); } +void rsnd_dma_addr_update(struct rsnd_dai_stream *io, + struct rsnd_mod *mod, + struct rsnd_mod *dma_mod) +{ + struct rsnd_mod *mod_from = NULL; + struct rsnd_mod *mod_to = NULL; + struct rsnd_priv *priv = rsnd_io_to_priv(io); + struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); + struct device *dev = rsnd_priv_to_dev(priv); + int is_play = rsnd_io_is_play(io); + struct rsnd_dma *dma = rsnd_mod_to_dma(dma_mod); + + if (!dmac) + return; + + rsnd_dma_of_path(mod, io, is_play, &mod_from, &mod_to); + + dma->src_addr = rsnd_dma_addr(io, mod_from, is_play, 1); + dma->dst_addr = rsnd_dma_addr(io, mod_to, is_play, 0); + + if (mod_from && mod_to) { + struct rsnd_dmapp *dmapp = rsnd_dma_to_dmapp(dma); + + dmapp->chcr = rsnd_dmapp_get_chcr(io, mod_from, mod_to) | + PDMACHCR_DE; + } + + dev_dbg(dev, "%s[%d] %pad -> %pad\n", + rsnd_mod_name(mod), rsnd_mod_id(mod), + &dma->src_addr, &dma->dst_addr); +} + int rsnd_dma_probe(struct rsnd_priv *priv) { struct platform_device *pdev = rsnd_priv_to_pdev(priv); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 59b6d89d8edc..6c16d8403f7a 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -233,6 +233,8 @@ u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod); */ int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, struct rsnd_mod **dma_mod); +void rsnd_dma_addr_update(struct rsnd_dai_stream *io, + struct rsnd_mod *mod, struct rsnd_mod *dma_mod); int rsnd_dma_probe(struct rsnd_priv *priv); struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, struct rsnd_mod *mod, char *name); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 4ac4b5b75ae2..b469db238697 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -160,6 +160,15 @@ int rsnd_ssi_get_busif(struct rsnd_dai_stream *io) return io->busif->val; } +static void rsnd_ssi_set_busif(struct rsnd_dai_stream *io, + struct rsnd_mod *mod) +{ + if (!rsnd_ssi_use_busif(io)) + return; + + rsnd_dma_addr_update(io, mod, io->dma); +} + static void rsnd_ssi_status_clear(struct rsnd_mod *mod) { rsnd_mod_write(mod, SSISR, 0); @@ -718,6 +727,10 @@ static int rsnd_ssi_pcm_new(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd) { + int is_play = rsnd_io_is_play(io); + int id = rsnd_mod_id(mod); + int ret = 0; + /* * rsnd_rdai_is_clk_master() will be enabled after set_fmt, * and, pcm_new will be called after it. @@ -725,7 +738,23 @@ static int rsnd_ssi_pcm_new(struct rsnd_mod *mod, */ rsnd_ssi_parent_attach(mod, io); - return 0; + if (!rsnd_ssi_use_busif(io)) + return 0; + + if (rsnd_ssi_is_multi_slave(mod, io)) + return 0; + + /* SSI5 to SSI8 only have one BUSIF */ + if (id <= 4 || id == 9) + ret = rsnd_kctrl_new_s(mod, io, rtd, + is_play ? + "SSI Out BUSIF" : + "SSI In BUSIF", + rsnd_kctrl_accept_anytime, + rsnd_ssi_set_busif, + io->busif, 7); + + return ret; } static int rsnd_ssi_common_probe(struct rsnd_mod *mod, -- 2.17.0

3 2

[alsa-devel] [PATCH linux-next 09/10] ASoC: rsnd: add busif property to dai stream
by jiada_wang＠mentor.com 02 Oct '18

02 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> SSI may use different busif for data transfer, this patch adds busif property to each dai stream, to indicate the busif used by playback/capture stream. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/core.c | 16 +++++++++++++++- sound/soc/sh/rcar/rsnd.h | 3 +++ sound/soc/sh/rcar/ssi.c | 5 ++++- 3 files changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 40d7dc4f7839..aee3132c72b9 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -560,6 +560,14 @@ struct rsnd_dai *rsnd_rdai_get(struct rsnd_priv *priv, int id) return priv->rdai + id; } +struct rsnd_kctrl_cfg_s *rsnd_busif_get(struct rsnd_priv *priv, int id) +{ + if ((id < 0) || (id >= 2 * rsnd_rdai_nr(priv))) + return NULL; + + return priv->busif + id; +} + static struct snd_soc_dai_driver *rsnd_daidrv_get(struct rsnd_priv *priv, int id) { @@ -1045,6 +1053,9 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, io_playback = &rdai->playback; io_capture = &rdai->capture; + io_playback->busif = rsnd_busif_get(priv, dai_i); + io_capture->busif = rsnd_busif_get(priv, rsnd_rdai_nr(priv) + dai_i); + snprintf(rdai->name, RSND_DAI_NAME_SIZE, "rsnd-dai.%d", dai_i); rdai->priv = priv; @@ -1108,6 +1119,7 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) struct snd_soc_dai_driver *rdrv; struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_dai *rdai; + struct rsnd_kctrl_cfg_s *busif; int nr; int is_graph; int dai_i; @@ -1123,12 +1135,14 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) rdrv = devm_kcalloc(dev, nr, sizeof(*rdrv), GFP_KERNEL); rdai = devm_kcalloc(dev, nr, sizeof(*rdai), GFP_KERNEL); - if (!rdrv || !rdai) + busif = devm_kcalloc(dev, 2 * nr, sizeof(*busif), GFP_KERNEL); + if (!rdrv || !rdai || !busif) return -ENOMEM; priv->rdai_nr = nr; priv->daidrv = rdrv; priv->rdai = rdai; + priv->busif = busif; /* * parse all dai diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 4464d1d0a042..59b6d89d8edc 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -456,6 +456,7 @@ struct rsnd_dai_stream { struct rsnd_mod *dma; struct rsnd_dai *rdai; struct device *dmac_dev; /* for IPMMU */ + struct rsnd_kctrl_cfg_s *busif; u32 parent_ssi_status; }; #define rsnd_io_to_mod(io, i) ((i) < RSND_MOD_MAX ? (io)->mod[(i)] : NULL) @@ -629,6 +630,8 @@ struct rsnd_priv { */ struct snd_soc_dai_driver *daidrv; struct rsnd_dai *rdai; + struct rsnd_kctrl_cfg_s *busif; + int rdai_nr; }; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 75a3d15084c2..4ac4b5b75ae2 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -154,7 +154,10 @@ int rsnd_ssi_use_busif(struct rsnd_dai_stream *io) int rsnd_ssi_get_busif(struct rsnd_dai_stream *io) { - return 0; /* BUSIF0 only for now */ + if (!rsnd_ssi_use_busif(io)) + return 0; + + return io->busif->val; } static void rsnd_ssi_status_clear(struct rsnd_mod *mod) -- 2.17.0

3 2

[alsa-devel] [PATCH linux-next 00/10] ASoC: rsnd: support to set different busif
by jiada_wang＠mentor.com 02 Oct '18

02 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> Rcar audio DMA/SSIU already support to use different BUSIF to transfer data, but busif is fixed at 0. This patch set adds kctrl interface "SSI Out/In BUSIF", which enables to configure busif for each dai-stream used in transfer. device tree files are updated to use extended audio dmac register, and add dma request number for busif1 ~ busif7 Jiada Wang (10): ASoC: rsnd: ssi: Request dedicated dma channels for busif1 to 7 arm64: r8a7795: add dma request number for busif1 ~ busif7 arm64: r8a7796: add dma request number for busif1 ~ busif7 arm64: r8a774a1: add dma request number for busif1 ~ busif7 arm64: r8a7795: change to use extended audio dmac register arm64: r8a7796: change to use extended audio dmac register arm64: r8a774a1: change to use extended audio dmac register arm64: r8a77965: change to use extended audio dmac register ASoC: rsnd: add busif property to dai stream ASoC: rsnd: Add kctrl to configure dai's busif arch/arm64/boot/dts/renesas/r8a774a1.dtsi | 139 ++++++++++++++++++---- arch/arm64/boot/dts/renesas/r8a7795.dtsi | 132 +++++++++++++++++--- arch/arm64/boot/dts/renesas/r8a7796.dtsi | 138 +++++++++++++++++---- arch/arm64/boot/dts/renesas/r8a77965.dtsi | 2 +- sound/soc/sh/rcar/core.c | 16 ++- sound/soc/sh/rcar/dma.c | 32 +++++ sound/soc/sh/rcar/rsnd.h | 5 + sound/soc/sh/rcar/ssi.c | 56 +++++++-- 8 files changed, 451 insertions(+), 69 deletions(-) -- 2.17.0

3 6

[alsa-devel] [PATCH] ASoC: Convert a few more users to using %pOFn instead of device_node.name
by Rob Herring 01 Oct '18

01 Oct '18

In preparation to remove the node name pointer from struct device_node, convert printf users to use the %pOFn format specifier. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Olivier Moysan <olivier.moysan(a)st.com> Cc: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Cc: Maxime Coquelin <mcoquelin.stm32(a)gmail.com> Cc: Alexandre Torgue <alexandre.torgue(a)st.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-stm32(a)st-md-mailman.stormreply.com Cc: linux-arm-kernel(a)lists.infradead.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/stm/stm32_sai.c | 8 ++++---- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index 192f4d7b37b6..bff7d71d0742 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -828,7 +828,7 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) /* request irq */ irq_id = platform_get_irq(pdev, 0); if (!irq_id) { - dev_err(dev, "%s no irq found\n", dev->of_node->name); + dev_err(dev, "%pOFn no irq found\n", dev->of_node); return -ENXIO; } ret = devm_request_irq(dev, irq_id, mt6797_afe_irq_handler, diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index c0b6697503fd..166aed28330d 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -1092,7 +1092,7 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev) irq_id = platform_get_irq(pdev, 0); if (irq_id <= 0) { - dev_err(afe->dev, "np %s no irq\n", afe->dev->of_node->name); + dev_err(afe->dev, "np %pOFn no irq\n", afe->dev->of_node); return irq_id < 0 ? irq_id : -ENXIO; } ret = devm_request_irq(afe->dev, irq_id, mt8173_afe_irq_handler, diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index d597eba61992..bcb35cae2a2c 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -74,14 +74,14 @@ static int stm32_sai_sync_conf_provider(struct stm32_sai_data *sai, int synco) return ret; } - dev_dbg(&sai->pdev->dev, "Set %s%s as synchro provider\n", - sai->pdev->dev.of_node->name, + dev_dbg(&sai->pdev->dev, "Set %pOFn%s as synchro provider\n", + sai->pdev->dev.of_node, synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); prev_synco = FIELD_GET(SAI_GCR_SYNCOUT_MASK, readl_relaxed(sai->base)); if (prev_synco != STM_SAI_SYNC_OUT_NONE && synco != prev_synco) { - dev_err(&sai->pdev->dev, "%s%s already set as sync provider\n", - sai->pdev->dev.of_node->name, + dev_err(&sai->pdev->dev, "%pOFn%s already set as sync provider\n", + sai->pdev->dev.of_node, prev_synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); clk_disable_unprepare(sai->pclk); return -EINVAL; -- 2.17.1

1 0

[alsa-devel] [PATCH] slimbus: ngd: QCOM_QMI_HELPERS has to be selected
by Niklas Cassel 01 Oct '18

01 Oct '18

QCOM_QMI_HELPERS is a hidden kconfig, so the proper usage is to select it, not depend upon it. Signed-off-by: Niklas Cassel <niklas.cassel(a)linaro.org> --- drivers/slimbus/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/slimbus/Kconfig b/drivers/slimbus/Kconfig index 9d73ad806698..9cca2c3559f2 100644 --- a/drivers/slimbus/Kconfig +++ b/drivers/slimbus/Kconfig @@ -22,8 +22,8 @@ config SLIM_QCOM_CTRL config SLIM_QCOM_NGD_CTRL tristate "Qualcomm SLIMbus Satellite Non-Generic Device Component" - depends on QCOM_QMI_HELPERS depends on HAS_IOMEM && DMA_ENGINE + select QCOM_QMI_HELPERS help Select driver if Qualcomm's SLIMbus Satellite Non-Generic Device Component is programmed using Linux kernel. -- 2.17.1

3 2

[alsa-devel] KASAN: use-after-free Read in snd_seq_ioctl_get_subscription
by syzbot 01 Oct '18

01 Oct '18

Hello, syzbot found the following crash on: HEAD commit: e704966c45e4 Merge branch 'for-linus' of git://git.kernel... git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=1277814e400000 kernel config: https://syzkaller.appspot.com/x/.config?x=a8212f992609a887 dashboard link: https://syzkaller.appspot.com/bug?extid=9437020c82413d00222d compiler: gcc (GCC) 8.0.1 20180413 (experimental) userspace arch: i386 Unfortunately, I don't have any reproducer for this crash yet. IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+9437020c82413d00222d(a)syzkaller.appspotmail.com ================================================================== BUG: KASAN: use-after-free in snd_seq_ioctl_get_subscription+0x241/0x250 sound/core/seq/seq_clientmgr.c:1913 Read of size 80 at addr ffff880198190300 by task syz-executor1/22402 CPU: 0 PID: 22402 Comm: syz-executor1 Not tainted 4.19.0-rc5+ #162 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x1c4/0x2b4 lib/dump_stack.c:113 print_address_description.cold.8+0x9/0x1ff mm/kasan/report.c:256 kasan_report_error mm/kasan/report.c:354 [inline] kasan_report.cold.9+0x242/0x309 mm/kasan/report.c:412 __asan_report_load_n_noabort+0xf/0x20 mm/kasan/report.c:443 snd_seq_ioctl_get_subscription+0x241/0x250 sound/core/seq/seq_clientmgr.c:1913 snd_seq_ioctl+0x253/0x440 sound/core/seq/seq_clientmgr.c:2138 snd_seq_ioctl_compat+0x16c/0x2d0 sound/core/seq/seq_compat.c:123 __do_compat_sys_ioctl fs/compat_ioctl.c:1419 [inline] __se_compat_sys_ioctl fs/compat_ioctl.c:1365 [inline] __ia32_compat_sys_ioctl+0x20e/0x630 fs/compat_ioctl.c:1365 do_syscall_32_irqs_on arch/x86/entry/common.c:326 [inline] do_fast_syscall_32+0x34d/0xfb2 arch/x86/entry/common.c:397 entry_SYSENTER_compat+0x70/0x7f arch/x86/entry/entry_64_compat.S:139 RIP: 0023:0xf7f2bca9 Code: 85 d2 74 02 89 0a 5b 5d c3 8b 04 24 c3 8b 0c 24 c3 8b 1c 24 c3 90 90 90 90 90 90 90 90 90 90 90 90 51 52 55 89 e5 0f 34 cd 80 <5d> 5a 59 c3 90 90 90 90 eb 0d 90 90 90 90 90 90 90 90 90 90 90 90 RSP: 002b:00000000f5f270cc EFLAGS: 00000296 ORIG_RAX: 0000000000000036 RAX: ffffffffffffffda RBX: 0000000000000004 RCX: 00000000c0505350 RDX: 0000000020000140 RSI: 0000000000000000 RDI: 0000000000000000 RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Allocated by task 22402: save_stack+0x43/0xd0 mm/kasan/kasan.c:448 set_track mm/kasan/kasan.c:460 [inline] kasan_kmalloc+0xc7/0xe0 mm/kasan/kasan.c:553 kmem_cache_alloc_trace+0x152/0x750 mm/slab.c:3620 kmalloc include/linux/slab.h:513 [inline] kzalloc include/linux/slab.h:707 [inline] snd_seq_port_connect+0xe4/0x730 sound/core/seq/seq_ports.c:571 snd_seq_ioctl_subscribe_port+0x1d8/0x310 sound/core/seq/seq_clientmgr.c:1458 snd_seq_ioctl+0x253/0x440 sound/core/seq/seq_clientmgr.c:2138 snd_seq_ioctl_compat+0x16c/0x2d0 sound/core/seq/seq_compat.c:123 __do_compat_sys_ioctl fs/compat_ioctl.c:1419 [inline] __se_compat_sys_ioctl fs/compat_ioctl.c:1365 [inline] __ia32_compat_sys_ioctl+0x20e/0x630 fs/compat_ioctl.c:1365 do_syscall_32_irqs_on arch/x86/entry/common.c:326 [inline] do_fast_syscall_32+0x34d/0xfb2 arch/x86/entry/common.c:397 entry_SYSENTER_compat+0x70/0x7f arch/x86/entry/entry_64_compat.S:139 Freed by task 22399: save_stack+0x43/0xd0 mm/kasan/kasan.c:448 set_track mm/kasan/kasan.c:460 [inline] __kasan_slab_free+0x102/0x150 mm/kasan/kasan.c:521 kasan_slab_free+0xe/0x10 mm/kasan/kasan.c:528 __cache_free mm/slab.c:3498 [inline] kfree+0xcf/0x230 mm/slab.c:3813 clear_subscriber_list+0x1ba/0x390 sound/core/seq/seq_ports.c:254 port_delete+0x7c/0x1b0 sound/core/seq/seq_ports.c:270 snd_seq_delete_all_ports+0x5fb/0x940 sound/core/seq/seq_ports.c:333 seq_free_client1.part.9+0xcb/0x270 sound/core/seq/seq_clientmgr.c:262 seq_free_client1 sound/core/seq/seq_clientmgr.c:256 [inline] seq_free_client+0x82/0x100 sound/core/seq/seq_clientmgr.c:285 snd_seq_release+0x51/0xe0 sound/core/seq/seq_clientmgr.c:366 __fput+0x385/0xa30 fs/file_table.c:278 ____fput+0x15/0x20 fs/file_table.c:309 task_work_run+0x1e8/0x2a0 kernel/task_work.c:113 tracehook_notify_resume include/linux/tracehook.h:193 [inline] exit_to_usermode_loop+0x318/0x380 arch/x86/entry/common.c:166 prepare_exit_to_usermode arch/x86/entry/common.c:197 [inline] syscall_return_slowpath arch/x86/entry/common.c:268 [inline] do_syscall_32_irqs_on arch/x86/entry/common.c:341 [inline] do_fast_syscall_32+0xcd5/0xfb2 arch/x86/entry/common.c:397 entry_SYSENTER_compat+0x70/0x7f arch/x86/entry/entry_64_compat.S:139 The buggy address belongs to the object at ffff880198190300 which belongs to the cache kmalloc-128 of size 128 The buggy address is located 0 bytes inside of 128-byte region [ffff880198190300, ffff880198190380) The buggy address belongs to the page: page:ffffea0006606400 count:1 mapcount:0 mapping:ffff8801da800640 index:0x0 flags: 0x2fffc0000000100(slab) raw: 02fffc0000000100 ffffea000713cc48 ffffea0006e7eb08 ffff8801da800640 raw: 0000000000000000 ffff880198190000 0000000100000015 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff880198190200: fc fc fc fc fc fc fc fc 00 00 00 00 00 00 00 00 ffff880198190280: 00 00 00 00 00 00 00 00 fc fc fc fc fc fc fc fc > ffff880198190300: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff880198190380: fc fc fc fc fc fc fc fc fb fb fb fb fb fb fb fb ffff880198190400: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#bug-status-tracking for how to communicate with syzbot.

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[alsa-devel] i2s\pcm\spdif\slimbus
by Muni Sekhar 29 Sep '18

29 Sep '18

Hi All, Can someone share the links to i2s\pcm\spdif\slimbus bus specifications. -- Thanks, Sekhar

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[alsa-devel] Fwd: Help with sound on Cherry Trail laptop with ES8316 sound chip
by Sundar Nagarajan 29 Sep '18

29 Sep '18

Hello, I have (separate) issues with two different Cherry Trail laptops that have the ES8316 chipset for sound. I already posted the following issues on Pierre Bossart's Intel UCM repository(https://github.com/plbossart/UCM): Issue 42: RDP Thinbook 1430 laptop (ES8316): Can't select between built-in speakers and headphone output (https://github.com/plbossart/UCM/issues/42) Pierre's input was as follows: The UCM file you are using gives the clue already: #Codec IC has only 1 analog out, switching SPK/HP is done externally, #probably by means of some analog switch controlled by gpio There is nothing we can do at the UCM level, this has to be handled at a lower level (i.e. the kernel) In addition, on a different laptop, I have a different problem with the ES3816 chipset. Issue 41: No sound on speakers on Chuwi Hi12 laptop - headphone sound works (es8316) (https://github.com/plbossart/UCM/issues/41) I reached out to Mark Brown, Daniel Drake and David Yang who were listed on the kernel.org commit for the ytcht_es8316.c driver (https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git/commit/…) and Mark suggested reaching out to you folks. What is the right place for me to post my issues to get your ideas and help? I have access to these machines, and I can compile kernels and perform testing, collect logs etc. Regards, Sundar ---------- Forwarded message --------- From: Mark Brown <broonie(a)kernel.org> Date: Thu, Sep 27, 2018 at 3:14 PM Subject: Re: Help with sound on Cherry Trail laptop with ES8316 sound chip To: Sundar Nagarajan <sun.nagarajan(a)gmail.com> Cc: Daniel Drake <drake(a)endlessm.com>, David Yang <yangxiaohua(a)everest-semi.com> On Wed, Sep 26, 2018 at 12:53:44PM -0700, Sundar Nagarajan wrote: > )I found your email from the kernel.org commit log for the > bytcht_es8316.c driver > (https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git/commit/…. > I have (separate) issues with two different Cherry Trail laptops that > have the ES8316 chipset. I posted the following issues on Pierre > Bossart's Intel UCM repository(https://github.com/plbossart/UCM): He's on sabbatical at the minute FWIW. > What is the right place for me to post my issues to get your ideas and help? You'd be best contacting the Intel maintainers about x86 specific issues: INTEL ASoC DRIVERS M: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> M: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> M: Jie Yang <yang.jie(a)linux.intel.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/intel/ That includes Pierre but also a few other people, hopefully at least one of them will be able to help or point you to someone who can. You should also copy in the alsa-devel list (alsa-devel(a)alsa-project.org) in case someone else can help and so people can google the discussion later, though that'll mean your e-mail address ends up in the archives so you might not be happy with that.

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[alsa-devel] [PATCH v2] ASoC: soc-utils: Rename dummy_dma_ops to snd_dummy_dma_ops
by Matthias Kaehlcke 28 Sep '18

28 Sep '18

The symbols 'dummy_dma_ops' is declared with different data types by sound/soc/soc-utils.c and arch/arm64/include/asm/dma-mapping.h. This leads to conflicts when soc-utils.c (indirectly) includes dma-mapping.h: sound/soc/soc-utils.c:282:33: error: conflicting types for 'dummy_dma_ops' static const struct snd_pcm_ops dummy_dma_ops = { ^ ... arch/arm64/include/asm/dma-mapping.h:27:33: note: previous declaration of 'dummy_dma_ops' was here extern const struct dma_map_ops dummy_dma_ops; ^ Rename the symbol in soc-utils.c to 'snd_dummy_dma_ops' to avoid the conflict. Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- Changes in v2: - use 'snd_dummy_dma_ops' instead of 'snd_dummy_ops' --- sound/soc/soc-utils.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index e0c93496c0cd..e3b9dd634c6d 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -273,13 +273,13 @@ static int dummy_dma_open(struct snd_pcm_substream *substream) return 0; } -static const struct snd_pcm_ops dummy_dma_ops = { +static const struct snd_pcm_ops snd_dummy_dma_ops = { .open = dummy_dma_open, .ioctl = snd_pcm_lib_ioctl, }; static const struct snd_soc_component_driver dummy_platform = { - .ops = &dummy_dma_ops, + .ops = &snd_dummy_dma_ops, }; static const struct snd_soc_component_driver dummy_codec = { -- 2.19.0.605.g01d371f741-goog

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[alsa-devel] [PATCH] ASoC: pxa: avoid AC97_BUS build warning
by Arnd Bergmann 28 Sep '18

28 Sep '18

Selecting AC97_BUS_NEW from SND_PXA2XX_SOC_AC97 leads to a Kconfig warning if any other driver selects AC97_BUS: WARNING: unmet direct dependencies detected for AC97_BUS_COMPAT Depends on [n]: SOUND [=y] && !UML && SND [=y] && AC97_BUS_NEW [=y] && !AC97_BUS [=y] Selected by [y]: - SND_SOC_WM9713 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && AC97_BUS_NEW [=y] I don't know if that combination is supposed to work. Assuming it is not, this adds a dependency on all users for PXA to avoids the combination. Fixes: 1c8bc7b3de5e ("ASoC: pxa: switch to new ac97 bus support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/pxa/Kconfig | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 29f577e6dfc0..943b44de1464 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -79,6 +79,7 @@ config SND_PXA2XX_SOC_TOSA tristate "SoC AC97 Audio support for Tosa" depends on SND_PXA2XX_SOC && MACH_TOSA depends on MFD_TC6393XB + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -88,6 +89,7 @@ config SND_PXA2XX_SOC_TOSA config SND_PXA2XX_SOC_E740 tristate "SoC AC97 Audio support for e740" depends on SND_PXA2XX_SOC && MACH_E740 + depends on !AC97_BUS select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -97,6 +99,7 @@ config SND_PXA2XX_SOC_E740 config SND_PXA2XX_SOC_E750 tristate "SoC AC97 Audio support for e750" depends on SND_PXA2XX_SOC && MACH_E750 + depends on !AC97_BUS select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -106,6 +109,7 @@ config SND_PXA2XX_SOC_E750 config SND_PXA2XX_SOC_E800 tristate "SoC AC97 Audio support for e800" depends on SND_PXA2XX_SOC && MACH_E800 + depends on !AC97_BUS select SND_SOC_WM9712 select SND_PXA2XX_SOC_AC97 help @@ -116,6 +120,7 @@ config SND_PXA2XX_SOC_EM_X270 tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300" depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \ MACH_CM_X300) + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -126,6 +131,7 @@ config SND_PXA2XX_SOC_PALM27X bool "SoC Audio support for Palm T|X, T5, E2 and LifeDrive" depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \ MACH_PALMT5 || MACH_PALMTE2) + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -155,6 +161,7 @@ config SND_SOC_TTC_DKB config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_PXA_SOC_SSP select SND_SOC_WM9713 @@ -194,6 +201,7 @@ config SND_PXA2XX_SOC_MAGICIAN config SND_PXA2XX_SOC_MIOA701 tristate "SoC Audio support for MIO A701" depends on SND_PXA2XX_SOC && MACH_MIOA701 + depends on !AC97_BUS select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9713 help -- 2.18.0

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[alsa-devel] [PATCH] ASoC: atmel: add SND_SOC_I2C_AND_SPI dependency
by Arnd Bergmann 28 Sep '18

28 Sep '18

Selecting SND_SOC_WM8731 is only allowed when all its dependencies are already there: WARNING: unmet direct dependencies detected for SND_SOC_WM8731 Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_I2C_AND_SPI [=m] Selected by [y]: - SND_SOC_MIKROE_PROTO [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_ATMEL_SOC [=y] && OF [=y] Selected by [m]: - SND_AT91_SOC_SAM9X5_WM8731 [=m] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_ATMEL_SOC [=y] && (ARCH_AT91 [=y] || COMPILE_TEST [=y]) && ATMEL_SSC [=y] && SND_SOC_I2C_AND_SPI [=m] - SND_SOC_ALL_CODECS [=m] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && COMPILE_TEST [=y] && SND_SOC_I2C_AND_SPI [=m] Fixes: a45f8853a5f9 ("ASoC: Add driver for PROTO Audio CODEC (with a WM8731)") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/atmel/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 81a0712d4f14..64f86f0b87e5 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -101,6 +101,7 @@ config SND_ATMEL_SOC_I2S config SND_SOC_MIKROE_PROTO tristate "Support for Mikroe-PROTO board" depends on OF + depends on SND_SOC_I2C_AND_SPI select SND_SOC_WM8731 help Say Y or M if you want to add support for MikroElektronika PROTO Audio -- 2.18.0

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[alsa-devel] [PATCH] ASoC: wm9712: fix unused variable warning
by Arnd Bergmann 28 Sep '18

28 Sep '18

The 'ret' variable is now only used in an #ifdef, and causes a warning if it is declared outside of that block: sound/soc/codecs/wm9712.c: In function 'wm9712_soc_probe': sound/soc/codecs/wm9712.c:641:6: error: unused variable 'ret' [-Werror=unused-variable] Fixes: 2ed1a8e0ce8d ("ASoC: wm9712: add ac97 new bus support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/wm9712.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index ade34c26ad2f..e873baa9e778 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -638,13 +638,14 @@ static int wm9712_soc_probe(struct snd_soc_component *component) { struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component); struct regmap *regmap; - int ret; if (wm9712->mfd_pdata) { wm9712->ac97 = wm9712->mfd_pdata->ac97; regmap = wm9712->mfd_pdata->regmap; } else { #ifdef CONFIG_SND_SOC_AC97_BUS + int ret; + wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID, WM9712_VENDOR_ID_MASK); if (IS_ERR(wm9712->ac97)) { -- 2.18.0

3 2

[alsa-devel] [PATCH] ASoC: rsnd: Add r8a7744 support
by Biju Das 28 Sep '18

28 Sep '18

Document RZ/G1N (R8A7744) SoC bindings. Signed-off-by: Biju Das <biju.das(a)bp.renesas.com> Reviewed-by: Chris Paterson <Chris.Paterson2(a)renesas.com> --- This patch is tested against linux-next next-20180927. --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index f4688c5..d92b705 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -343,6 +343,7 @@ Required properties: "renesas,rcar_sound-gen3" if generation3 (or RZ/G2) Examples with soctypes are: - "renesas,rcar_sound-r8a7743" (RZ/G1M) + - "renesas,rcar_sound-r8a7744" (RZ/G1N) - "renesas,rcar_sound-r8a7745" (RZ/G1E) - "renesas,rcar_sound-r8a774a1" (RZ/G2M) - "renesas,rcar_sound-r8a7778" (R-Car M1A) -- 2.7.4

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[alsa-devel] Applied "ASoC: qcom: qdsp6: remove duplicated include" to the asoc tree
by Mark Brown 28 Sep '18

28 Sep '18

The patch ASoC: qcom: qdsp6: remove duplicated include has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 06da26e5ce157bf2af09ef4a9f5dc30af4599fa0 Mon Sep 17 00:00:00 2001 From: YueHaibing <yuehaibing(a)huawei.com> Date: Wed, 26 Sep 2018 15:33:03 +0800 Subject: [PATCH] ASoC: qcom: qdsp6: remove duplicated include Remove duplicated includes linux/of_platform.h and linux/wait.h Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm.c | 1 - sound/soc/qcom/qdsp6/q6core.c | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 2b2c7233bb5f..e1cfa846a1dc 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -11,7 +11,6 @@ #include <linux/spinlock.h> #include <linux/kref.h> #include <linux/of.h> -#include <linux/of_platform.h> #include <uapi/sound/asound.h> #include <linux/delay.h> #include <linux/slab.h> diff --git a/sound/soc/qcom/qdsp6/q6core.c b/sound/soc/qcom/qdsp6/q6core.c index ca1be7305524..cdfc8ab6cfc0 100644 --- a/sound/soc/qcom/qdsp6/q6core.c +++ b/sound/soc/qcom/qdsp6/q6core.c @@ -10,7 +10,6 @@ #include <linux/of.h> #include <linux/of_platform.h> #include <linux/jiffies.h> -#include <linux/wait.h> #include <linux/soc/qcom/apr.h> #include "q6core.h" #include "q6dsp-errno.h" -- 2.19.0

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Re: [alsa-devel] [PATCH] ASoC: nau8822: new codec driver
by AC30 CTLin0 27 Sep '18

27 Sep '18

OK, I will resend a new patch. However, from my last patch file, I can see DT file where is located at "Documentation/devicetree/bindings/sound/nau8822.txt". And don't you see any DT file? On 9/26/2018 1:52 AM, Mark Brown wrote: > On Tue, Sep 25, 2018 at 09:56:13AM -0700, Mark Brown wrote: >> On Tue, Sep 25, 2018 at 10:50:19AM +0800, AC30 CTLin0 wrote: >>> May I learn the status of patch review due to our customer ask us about >>> upstream status? >> I don't know, I don't seem to have a copy of this patch in my review >> queue. Please resend: > Found a copy in the list archives, but still no idea where it went in my > review queue. My best guess is that it got eaten by spam filtering > somewhere but I'm not 100% sure. Sorry about that. :( The driver > looks pretty clean, a few small points: > > - The Mute Switch controls should just be called Switch so that > applications know how to handle them. > - PGA Boost +20dB should just be "PGA Boost Volume" with the +20dB > specified by having it be a SOC_DOUBLE_R_TLV(). > - The switch statements in hw_params() should have default: > cases that return errors so that if an incorrect parameter makes it > through we detect it. > - There's a DT compatible string but no DT binding file is supplied, > should just be a very simple one as there's no parameters. > > These are all small, hopefully should be quick to update. =========================================================================================== The privileged confidential information contained in this email is intended for use only by the addressees as indicated by the original sender of this email. If you are not the addressee indicated in this email or are not responsible for delivery of the email to such a person, please kindly reply to the sender indicating this fact and delete all copies of it from your computer and network server immediately. Your cooperation is highly appreciated. It is advised that any unauthorized use of confidential information of Nuvoton is strictly prohibited; and any information in this email irrelevant to the official business of Nuvoton shall be deemed as neither given nor endorsed by Nuvoton.

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[alsa-devel] [PATCH linux-next 08/10] arm64: r8a77965: change to use extended audio dmac register
by jiada_wang＠mentor.com 27 Sep '18

27 Sep '18

From: Jiada Wang <jiada_wang(a)mentor.com> Basic audio dmac register only supports busif from 0 to 3, in order to use busif4 ~ busif7, extended audio dmac register need to be used. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- arch/arm64/boot/dts/renesas/r8a77965.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/renesas/r8a77965.dtsi b/arch/arm64/boot/dts/renesas/r8a77965.dtsi index fd00010a5ed7..6c0d5053a81c 100644 --- a/arch/arm64/boot/dts/renesas/r8a77965.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a77965.dtsi @@ -1405,7 +1405,7 @@ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + <0 0xec760000 0 0x200>; /* Audio DMAC peri peri*/ /* placeholder */ rcar_sound,dvc { -- 2.17.0

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[alsa-devel] [PATCH linux-next 07/10] arm64: r8a774a1: change to use extended audio dmac register
by jiada_wang＠mentor.com 27 Sep '18

27 Sep '18

From: Jiada Wang <jiada_wang(a)mentor.com> Basic audio dmac register only supports busif from 0 to 3, in order to use busif4 ~ busif7, extended audio dmac register need to be used. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- arch/arm64/boot/dts/renesas/r8a774a1.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/renesas/r8a774a1.dtsi b/arch/arm64/boot/dts/renesas/r8a774a1.dtsi index c9aad3e3b1b0..c3e0f2151176 100644 --- a/arch/arm64/boot/dts/renesas/r8a774a1.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a774a1.dtsi @@ -1120,7 +1120,7 @@ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + <0 0xec760000 0 0x200>; /* Audio DMAC peri peri*/ reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; clocks = <&cpg CPG_MOD 1005>, -- 2.17.0

1 0

[alsa-devel] [PATCH linux-next 06/10] arm64: r8a7796: change to use extended audio dmac register
by jiada_wang＠mentor.com 27 Sep '18

27 Sep '18

From: Jiada Wang <jiada_wang(a)mentor.com> Basic audio dmac register only supports busif from 0 to 3, in order to use busif4 ~ busif7, extended audio dmac register need to be used. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- arch/arm64/boot/dts/renesas/r8a7796.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/renesas/r8a7796.dtsi b/arch/arm64/boot/dts/renesas/r8a7796.dtsi index e9b05d997c4e..827a704154cd 100644 --- a/arch/arm64/boot/dts/renesas/r8a7796.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a7796.dtsi @@ -1670,7 +1670,7 @@ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + <0 0xec760000 0 0x200>; /* Audio DMAC peri peri*/ reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; clocks = <&cpg CPG_MOD 1005>, -- 2.17.0

1 0

[alsa-devel] [PATCH linux-next 05/10] arm64: r8a7795: change to use extended audio dmac register
by jiada_wang＠mentor.com 27 Sep '18

27 Sep '18

From: Jiada Wang <jiada_wang(a)mentor.com> Basic audio dmac register only supports busif from 0 to 3, in order to use busif4 ~ busif7, extended audio dmac register need to be used. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- arch/arm64/boot/dts/renesas/r8a7795.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm64/boot/dts/renesas/r8a7795.dtsi b/arch/arm64/boot/dts/renesas/r8a7795.dtsi index 0a66bda7124f..32d191102a81 100644 --- a/arch/arm64/boot/dts/renesas/r8a7795.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a7795.dtsi @@ -1796,7 +1796,7 @@ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + <0 0xec760000 0 0x200>; /* Audio DMAC peri peri*/ reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; clocks = <&cpg CPG_MOD 1005>, -- 2.17.0

1 0

[alsa-devel] [PATCH 0/5] MAX98088/9 of features
by Marco Felsch 26 Sep '18

26 Sep '18

Hi, I picked up the max98088 patches from Andreas series [1] and ported them to the new kernel. Regards, Marco [1] http://lists.infradead.org/pipermail/linux-arm-kernel/ \ 2015-February/324440.html Andreas Färber (3): ASoC: dt-bindings: add max98088 audio codec ASoC: max98088: add OF support ASoC: max98088: Add master clock handling Marco Felsch (2): ASoC: dt-bindings: max98088: add external clock binding ASoC: max98988: make it selectable .../bindings/sound/maxim,max98088.txt | 20 ++++++++++ sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/max98088.c | 39 +++++++++++++++++++ 3 files changed, 60 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98088.txt -- 2.19.0

2 11

Re: [alsa-devel] linux-next: Tree for Sep 25 (sound/soc/codecs/wm8731)
by Randy Dunlap 25 Sep '18

25 Sep '18

On 9/24/18 10:39 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20180924: > on i386: WARNING: unmet direct dependencies detected for SND_SOC_WM8731 Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_I2C_AND_SPI [=m] Selected by [y]: - SND_SOC_MIKROE_PROTO [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_ATMEL_SOC [=y] && OF [=y] and then: ld: sound/soc/codecs/wm8731.o: in function `wm8731_i2c_probe': wm8731.c:(.text+0x609): undefined reference to `__devm_regmap_init_i2c' ld: sound/soc/codecs/wm8731.o: in function `wm8731_modinit': wm8731.c:(.init.text+0x8): undefined reference to `i2c_register_driver' ld: sound/soc/codecs/wm8731.o: in function `wm8731_exit': wm8731.c:(.exit.text+0x6): undefined reference to `i2c_del_driver' but CONFIG_I2C=m. Full randconfig file is attached. Note: I only see the kconfig warnings when I build for i386 on an x86_64 machine. If I build on an i386 system, I don't see those warnings, but I do see the linker errors. -- ~Randy

1 0

[alsa-devel] [PATCH v2] ASoC: cs4265: Add a MIC pre. route
by Matt Flax 25 Sep '18

25 Sep '18

The cs4265 driver is missing a microphone preamp enable. This patch enables/disables the microphone preamp when mic selection is made using the kcontrol. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- v2: Removed duplicate MIC lines. --- sound/soc/codecs/cs4265.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 275677de..4e55ac46 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -221,10 +221,11 @@ static const struct snd_soc_dapm_route cs4265_audio_map[] = { {"LINEOUTR", NULL, "DAC"}, {"SPDIFOUT", NULL, "SPDIF"}, + {"Pre-amp MIC", NULL, "MICL"}, + {"Pre-amp MIC", NULL, "MICR"}, + {"ADC Mux", "MIC", "Pre-amp MIC"}, {"ADC Mux", "LINEIN", "LINEINL"}, {"ADC Mux", "LINEIN", "LINEINR"}, - {"ADC Mux", "MIC", "MICL"}, - {"ADC Mux", "MIC", "MICR"}, {"ADC", NULL, "ADC Mux"}, {"DOUT", NULL, "ADC"}, {"DAI1 Capture", NULL, "DOUT"}, -- 2.17.1

3 2

[alsa-devel] [PATCH] ASoC: soc-utils: Rename dummy_dma_ops to snd_dummy_ops
by Matthias Kaehlcke 25 Sep '18

25 Sep '18

The symbols 'dummy_dma_ops' is declared with different data types by sound/soc/soc-utils.c and arch/arm64/include/asm/dma-mapping.h. This leads to conflicts when soc-utils.c (indirectly) includes dma-mapping.h: sound/soc/soc-utils.c:282:33: error: conflicting types for 'dummy_dma_ops' static const struct snd_pcm_ops dummy_dma_ops = { ^ ... arch/arm64/include/asm/dma-mapping.h:27:33: note: previous declaration of 'dummy_dma_ops' was here extern const struct dma_map_ops dummy_dma_ops; ^ Rename the symbol in soc-utils.c to 'snd_dummy_ops' to avoid the conflict. Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- sound/soc/soc-utils.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index e0c93496c0cd..ab44133bc0bf 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -273,13 +273,13 @@ static int dummy_dma_open(struct snd_pcm_substream *substream) return 0; } -static const struct snd_pcm_ops dummy_dma_ops = { +static const struct snd_pcm_ops snd_dummy_ops = { .open = dummy_dma_open, .ioctl = snd_pcm_lib_ioctl, }; static const struct snd_soc_component_driver dummy_platform = { - .ops = &dummy_dma_ops, + .ops = &snd_dummy_ops, }; static const struct snd_soc_component_driver dummy_codec = { -- 2.19.0.605.g01d371f741-goog

3 2

[alsa-devel] [PATCH - Fix typo in configure.ac 1/1] PATCH - Fix typo in configure.ac
by plevine457＠gmail.com 25 Sep '18

25 Sep '18

From: Peter Levine <plevine457(a)gmail.com> Signed-off-by: Peter Levine <plevine457(a)gmail.com> diff --git a/configure.ac b/configure.ac index 3e2d233..cc3740b 100644 --- a/configure.ac +++ b/configure.ac @@ -90,7 +90,7 @@ if test "$use_maemo_rm" = "yes"; then fi AC_ARG_ENABLE([libav], - AS_HELP_STRING([--disable-avlib], [Do not build plugins depending on libav/ffmpeg (a52,lavrate...)])) + AS_HELP_STRING([--disable-libav], [Do not build plugins depending on libav/ffmpeg (a52,lavrate...)])) if test "x$enable_libav" != "xno"; then PKG_CHECK_MODULES(LIBAV, [libavcodec libavutil libavresample], [HAVE_LIBAV=yes], [HAVE_LIBAV=no]) -- 2.19.0

2 1

Re: [alsa-devel] [PATCH 1/3] ARM: configs: omap2plus: Add tlv320aic23 as module
by Tony Lindgren 25 Sep '18

25 Sep '18

* Adam Ford <aford173(a)gmail.com> [180923 08:41]: > The AM3517 EVM has an expander board which has two tlv320aic23 > codecs. This enables the driver for these codecs as a module. Applying into omap-for-v4.20/defconfig thanks. Tony

1 0

Re: [alsa-devel] [PATCH 3/3] ARM: OMAP2PLUS: Remove references to AM3517-EVM since it's in DT
by Tony Lindgren 25 Sep '18

25 Sep '18

* Adam Ford <aford173(a)gmail.com> [180923 08:41]: > Since the audio codecs used on the AM3517-EVM are now available > in device tree, and the audio driver wasn't really being used > anyway, this patch removes the remaining AM3517 legacy board > options and sound soc drivers. Mark do you want to pick this one? If so: Acked-by: Tony Lindgren <tony(a)atomide.com> Maybe update the subject line according too. Regards, Tony

1 0

Re: [alsa-devel] [PATCH 2/3] ARM: DTS: AM3517-EVM: Add support for UI board and Audio
by Tony Lindgren 25 Sep '18

25 Sep '18

* Adam Ford <aford173(a)gmail.com> [180923 15:41]: > The full AM3517-EVM shipped with an add-on board which contained > two Audio codecs, a GPIO expander with a variety of buttons, and > some other features. This patch enables these portions of the UI > board because they don't directly conflict with existing features. Applying this one into omap-for-v4.20/dt thanks. Tony

1 0

[alsa-devel] [PATCH] ASoC: cs4265: Add a MIC pre. route
by Matt Flax 24 Sep '18

24 Sep '18

The cs4265 driver is missing a microphone preamp enable. This patch enables/disables the microphone preamp when mic selection is made using the kcontrol. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/codecs/cs4265.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 275677de..332d714a 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -230,6 +230,10 @@ static const struct snd_soc_dapm_route cs4265_audio_map[] = { {"DAI1 Capture", NULL, "DOUT"}, {"DAI2 Capture", NULL, "DOUT"}, + { "Pre-amp MIC", NULL, "MICL" }, + { "Pre-amp MIC", NULL, "MICR" }, + { "ADC Mux", "MIC", "Pre-amp MIC" }, + /* Loopback */ {"Loopback", "Switch", "ADC"}, {"DAC", NULL, "Loopback"}, -- 2.17.1

2 1

Re: [alsa-devel] [PATCH 3/3] ARM: OMAP2PLUS: Remove references to AM3517-EVM since it's in DT
by Jarkko Nikula 23 Sep '18

23 Sep '18

On 09/23/18 18:37, Adam Ford wrote: > Since the audio codecs used on the AM3517-EVM are now available > in device tree, and the audio driver wasn't really being used > anyway, this patch removes the remaining AM3517 legacy board > options and sound soc drivers. > > Signed-off-by: Adam Ford <aford173(a)gmail.com> > For sound/soc/omap/ changes: Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com>

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[alsa-devel] Pinpoint audio audio corruption (Layla3G)
by Mark Hills 23 Sep '18

23 Sep '18

I have a bug in ALSA somewhere; playback is corrupted. Nothing is pro-actively telling me of a problem (eg. underrun) I could use some assistance getting to the bottom of this. Listen to [1] for what it sounds like -- analogue recording from the output. Look at the waveform [2] in Baudlien and it seems both L+R channels geting zero'd for some buffer period of 2.5 or 5 milliseconds. This is Layla3G, snd-echo3g. This problem has been rare, but persisted for many years. I've finally got something reproducable to investigate. It began ~10 years ago where very occasionally recordings or playback would do this. This was a simple self-contained application in production use; written by me, using plughw:Layla3G,0,0 (no desktops, browsers, dmix etc.) Problems were frequent enough to be annoying, but infrequent enough that I just had to put it down to 'cosmic rays', with no useful bug report. These days I am on a desktop using dmix; and more recently with Google Chrome and found I can reliably 'trigger' it. To reset, I must close all applications using /dev/snd, and re-open. It seems to be: * whether or not Google Chrome is first to open the device * about 10-60 minutes after Chrome has opened the device * more prone to happening if other applications are using the dmix * not related to buffer sizes; I use larger sizes in the asoundrc it doesn't affect the likelihood, or the 'sound' when it does happen * I've seen two Layla's affected * I've had many kernels and PCs over the years There's a bug somewhere, and with nothing printed to dmesg, console etc., I'm at a loss here, imagining my problem could be anywhere between dmix, alsa-lib, snd-echo3g or even the firmware itself. Whatever the case, I've finally got some combination I can do in userland as a trigger. Below is: - .asoundrc describing today's setup (previously I used stereo with no route, still with problems) - output of the 'aplay' that produced the recording above; and - some other stats. [1] http://www.pogo.org.uk/~mark/tmp/layla-glitch.flac [2] http://www.pogo.org.uk/~mark/tmp/layla-glitch.png Many thanks -- Mark pcm.!default { type plug slave.pcm layla_multi } pcm.layla_multi { type route slave.pcm { type dmix ipc_key 1742 slave { pcm "hw:Layla3G,0,0" period_size 992 buffer_size 10912 channels 5 rate 48000 format S16_LE } } ttable { # Headphones 0.0 1 1.1 1 # Speakers 0.2 1 1.3 1 # Subwoofer; mono mix 0.4 0.5 1.4 0.5 } } $ uname -a Linux stax 4.18.5 #113 SMP PREEMPT Mon Sep 3 23:14:59 BST 2018 x86_64 Intel(R) Core(TM) i5-6500 CPU @ 3.20GHz GenuineIntel GNU/Linux $ head -99l /proc/asound/Layla3G/pcm0p/sub0/* ==> /proc/asound/Layla3G/pcm0p/sub0/hw_params <== access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 6 rate: 48000 (48000/1) period_size: 992 buffer_size: 10912 ==> /proc/asound/Layla3G/pcm0p/sub0/info <== card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: Analog PCM name: Layla3G subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 8 subdevices_avail: 7 ==> /proc/asound/Layla3G/pcm0p/sub0/prealloc <== 128 ==> /proc/asound/Layla3G/pcm0p/sub0/prealloc_max <== 256 ==> /proc/asound/Layla3G/pcm0p/sub0/status <== state: RUNNING owner_pid : 9250 trigger_time: 663739.627559736 tstamp : 680208.555428028 delay : -790500256 avail : 790511168 avail_max : 790511168 ----- hw_ptr : 790500256 appl_ptr : 0 ==> /proc/asound/Layla3G/pcm0p/sub0/sw_params <== tstamp_mode: ENABLE period_step: 1 avail_min: 1 start_threshold: 1 stop_threshold: 6142909891733356544 silence_threshold: 0 silence_size: 6142909891733356544 boundary: 6142909891733356544 $ aplay -vv all48000.wav Playing WAVE 'all48000.wav' : Float 32 bit Little Endian, Rate 48000 Hz, Stereo Plug PCM: Linear Integer <-> Linear Float conversion PCM (S16_LE) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : FLOAT_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 10912 period_size : 992 period_time : 20666 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 992 period_event : 0 start_threshold : 10912 stop_threshold : 10912 silence_threshold: 0 silence_size : 0 boundary : 6142909891733356544 Slave: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 1 2 <- none 3 <- none 4 <- none 5 <- none Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 10912 period_size : 992 period_time : 20666 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 992 period_event : 0 start_threshold : 10912 stop_threshold : 10912 silence_threshold: 0 silence_size : 0 boundary : 6142909891733356544 Slave: Route conversion PCM Transformation table: 0 <- 0 1 <- 1 2 <- 0 3 <- 1 4 <- 0*0.5 + 1*0.5 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 6 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 10912 period_size : 992 period_time : 20666 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 992 period_event : 0 start_threshold : 10912 stop_threshold : 10912 silence_threshold: 0 silence_size : 0 boundary : 6142909891733356544 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 6 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 10912 period_size : 992 period_time : 20666 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 992 period_event : 0 start_threshold : 10912 stop_threshold : 10912 silence_threshold: 0 silence_size : 0 boundary : 6142909891733356544 Hardware PCM card 0 'Layla3G' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 6 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 10912 period_size : 992 period_time : 20666 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 992 period_event : 0 start_threshold : 1 stop_threshold : 0 silence_threshold: 0 silence_size : 0 boundary : 6142909891733356544 appl_ptr : 0 hw_ptr : 807111136 ############################ + | 56%^C Aborted by signal Interrupt... aplay: pcm_write:2004: write error: Interrupted system call

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: ensure that cluster header size is enough
by Dan Carpenter 22 Sep '18

22 Sep '18

I think we need a check to make sure that "hc_header.wLength" is large enough for a struct struct uac3_cluster_header_descriptor. Fixes: 9a2fe9b801f5 ("ALSA: usb: initial USB Audio Device Class 3.0 support") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 67cf849aa16b..f7179ce39a19 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -968,6 +968,12 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, * and request Cluster Descriptor */ wLength = le16_to_cpu(hc_header.wLength); + if (wLength < sizeof(*cluster)) { + dev_err(&dev->dev, + "%u:%d : cluster header size too small %d\n", + iface_no, altno, wLength); + return ERR_PTR(-EIO); + } cluster = kzalloc(wLength, GFP_KERNEL); if (!cluster) return ERR_PTR(-ENOMEM);

2 1

Re: [alsa-devel] [PATCH] ALSA: usb-audio: Add custom mixer status quirks for RME CC devices
by Jussi Laako 22 Sep '18

22 Sep '18

Hi Takashi, Please hold applying the patch, I'm having final checks with RME regarding the VID or VID:PID based approach when to use this functionality. If it seems better to use strict VID:PID table instead, I will send an updated patch. - Jussi

1 0

[alsa-devel] [RFC] ALSA: usb-audio: Add custom mixer status quirks for RME CC devices
by Jussi Laako 22 Sep '18

22 Sep '18

This patch accessing vendor specific registers of RME Class Compliant devices works otherwise fine, but there is still one small problem. It is supposed to be a polled from the hardware, since the values indicate device status and are subject to change at all times. Behavior is correct if I keep running "amixer" all the time to retrieve values - sampling rate changes are visible and values get updated. However, when I poll the values from an application without reopening the mixer, they are being cached somewhere and the changes are not reflected. This problem doesn't exist for example with PCIe based RME HDSPe AIO and application gets updated values, but in this USB based device case they don't get updated. I have flagged all the mixer controls as read-only and volatile, but the volatile flag is not honored in this (assuming I've understood it correctly). Any guidance on how to fix the caching problem is welcome!

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[alsa-devel] [PATCH 0/2] ASoC: for_each_component_xx() macro
by Kuninori Morimoto 21 Sep '18

21 Sep '18

Hi Mark These are v2 of for_each_component_xx() macro. v1 -> v2 is renamed comment_all to comment. Kuninori Morimoto (2): ASoC: add for_each_component() macro ASoC: add for_each_component_dais() macro include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 35 ++++++++++++++++++++--------------- 2 files changed, 25 insertions(+), 15 deletions(-) -- 2.7.4

2 4

[alsa-devel] [PATCH] ASoC: rsnd: fixup SSI clock during suspend/resume modes
by Kuninori Morimoto 21 Sep '18

21 Sep '18

From: Dmytro Prokopchuk <dmytro.prokopchuk(a)globallogic.com> Prepare <-> Cleanup functions pair has balanced calls. But in case of suspend mode no call to rsnd_soc_dai_shutdown() function, so cleanup isn't called. OTOH during resume mode function rsnd_soc_dai_prepare() is called, but calling rsnd_ssi_prepare() is skipped (rsnd_status_update() returns zero, bacause was not cleanup before). We need to call rsnd_ssi_prepare(), because it enables SSI clocks by calling rsnd_ssi_master_clk_start(). This patch allows to call prepare/cleanup functions always. Signed-off-by: Dmytro Prokopchuk <dmytro.prokopchuk(a)globallogic.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> [kuninori: adjusted to upstream] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 7 +++---- sound/soc/sh/rcar/rsnd.h | 14 +++++++------- 2 files changed, 10 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 0bbc4b0..6d19475 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -134,10 +134,9 @@ static int rsnd_dmaen_prepare(struct rsnd_mod *mod, struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); struct device *dev = rsnd_priv_to_dev(priv); - if (dmaen->chan) { - dev_err(dev, "it already has dma channel\n"); - return -EIO; - } + /* maybe suspended */ + if (dmaen->chan) + return 0; /* * DMAEngine request uses mutex lock. diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index e857311..4464d1d 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -318,9 +318,8 @@ struct rsnd_mod { /* * status * - * 0xH0000CBA + * 0xH0000CB0 * - * A 0: prepare 1: cleanup * B 0: init 1: quit * C 0: start 1: stop * @@ -331,9 +330,8 @@ struct rsnd_mod { * H 0: hw_params * H 0: pointer * H 0: prepare + * H 0: cleanup */ -#define __rsnd_mod_shift_prepare 0 -#define __rsnd_mod_shift_cleanup 0 #define __rsnd_mod_shift_init 4 #define __rsnd_mod_shift_quit 4 #define __rsnd_mod_shift_start 8 @@ -345,11 +343,13 @@ struct rsnd_mod { #define __rsnd_mod_shift_fallback 28 /* always called */ #define __rsnd_mod_shift_hw_params 28 /* always called */ #define __rsnd_mod_shift_pointer 28 /* always called */ +#define __rsnd_mod_shift_prepare 28 /* always called */ +#define __rsnd_mod_shift_cleanup 28 /* always called */ #define __rsnd_mod_add_probe 0 #define __rsnd_mod_add_remove 0 -#define __rsnd_mod_add_prepare 1 -#define __rsnd_mod_add_cleanup -1 +#define __rsnd_mod_add_prepare 0 +#define __rsnd_mod_add_cleanup 0 #define __rsnd_mod_add_init 1 #define __rsnd_mod_add_quit -1 #define __rsnd_mod_add_start 1 @@ -363,7 +363,7 @@ struct rsnd_mod { #define __rsnd_mod_call_probe 0 #define __rsnd_mod_call_remove 0 #define __rsnd_mod_call_prepare 0 -#define __rsnd_mod_call_cleanup 1 +#define __rsnd_mod_call_cleanup 0 #define __rsnd_mod_call_init 0 #define __rsnd_mod_call_quit 1 #define __rsnd_mod_call_start 0 -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: tidyup for_each_card_prelinks() dai_link
by Kuninori Morimoto 21 Sep '18

21 Sep '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 7fe072b4df5d0 ("ASoC: add for_each_card_prelinks() macro") added new for_each_card_prelinks() macro, but it had typo. This patch fixup it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 8 ++++---- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 8 ++++---- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 6 +++--- 3 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c index 875f846..97f9f38 100644 --- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c +++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c @@ -311,9 +311,9 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->platform_name) + if (dai_link->platform_name) continue; - dai_links->platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } card->dev = dev; @@ -326,9 +326,9 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->codec_name) + if (dai_link->codec_name) continue; - dai_links->codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } codec_node_bt_mrg = of_parse_phandle(pdev->dev.of_node, diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c index c67f629..6bc1d3d 100644 --- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c +++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c @@ -107,9 +107,9 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->platform_name) + if (dai_link->platform_name) continue; - dai_links->platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } card->dev = &pdev->dev; @@ -122,9 +122,9 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->codec_name) + if (dai_link->codec_name) continue; - dai_links->codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } ret = snd_soc_of_parse_audio_routing(card, "audio-routing"); diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c index ff2e0ca..cc41eb5 100644 --- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c +++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c @@ -172,7 +172,7 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) for_each_card_prelinks(card, i, dai_link) { if (dai_link->platform_name) continue; - dai_links->platform_of_node = platform_node; + dai_link->platform_of_node = platform_node; } codec_node = of_parse_phandle(pdev->dev.of_node, @@ -183,9 +183,9 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { - if (dai_links->codec_name) + if (dai_link->codec_name) continue; - dai_links->codec_of_node = codec_node; + dai_link->codec_of_node = codec_node; } ret = devm_snd_soc_register_card(&pdev->dev, card); -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: qcom: qdsp6: remove duplicated include from q6adm.c" to the asoc tree
by Mark Brown 21 Sep '18

21 Sep '18

The patch ASoC: qcom: qdsp6: remove duplicated include from q6adm.c has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c78d42c7fbd6f2a27a665bcd1ab60c9df617f7c9 Mon Sep 17 00:00:00 2001 From: zhong jiang <zhongjiang(a)huawei.com> Date: Fri, 21 Sep 2018 18:24:58 +0800 Subject: [PATCH] ASoC: qcom: qdsp6: remove duplicated include from q6adm.c We include wait.h twice in q6adm.c. it is unnecessary. hence remove it. Further, order the include files as alphabet. Signed-off-by: zhong jiang <zhongjiang(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/qdsp6/q6adm.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 932c3ebfd252..da242515e146 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -2,25 +2,24 @@ // Copyright (c) 2011-2017, The Linux Foundation. All rights reserved. // Copyright (c) 2018, Linaro Limited -#include <linux/slab.h> -#include <linux/wait.h> -#include <linux/kernel.h> #include <linux/device.h> -#include <linux/module.h> -#include <linux/sched.h> #include <linux/jiffies.h> +#include <linux/kernel.h> +#include <linux/kref.h> +#include <linux/module.h> #include <linux/of.h> #include <linux/of_platform.h> -#include <linux/kref.h> -#include <linux/wait.h> -#include <linux/soc/qcom/apr.h> #include <linux/platform_device.h> +#include <linux/sched.h> +#include <linux/slab.h> +#include <linux/soc/qcom/apr.h> +#include <linux/wait.h> #include <sound/asound.h> #include "q6adm.h" #include "q6afe.h" #include "q6core.h" -#include "q6dsp-errno.h" #include "q6dsp-common.h" +#include "q6dsp-errno.h" #define ADM_CMD_DEVICE_OPEN_V5 0x00010326 #define ADM_CMDRSP_DEVICE_OPEN_V5 0x00010329 -- 2.19.0

1 0

[alsa-devel] [PATCH] ASoC: fix a use after free case
by Guennadi Liakhovetski 21 Sep '18

21 Sep '18

Unloading ASoC modules as used by the SOF driver leads to an object being used after it's been freed. Fix this be clearing a reference to it and making sure to check for its presence. This fixes issue #144. Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/soc/soc-core.c | 6 +++--- sound/soc/soc-topology.c | 5 +++++ 2 files changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9cfe10d8040c..dce9d53ff283 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -942,9 +942,9 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order) { int err; - if (dai && dai->probed && - dai->driver->remove_order == order) { - if (dai->driver->remove) { + if (dai && dai->probed) { + if (dai->driver && dai->driver->remove_order == order && + dai->driver->remove) { err = dai->driver->remove(dai); if (err < 0) dev_err(dai->dev, diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 66e77e020745..6d438ec8c0ec 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -502,6 +502,7 @@ static void remove_dai(struct snd_soc_component *comp, { struct snd_soc_dai_driver *dai_drv = container_of(dobj, struct snd_soc_dai_driver, dobj); + struct snd_soc_dai *dai; if (pass != SOC_TPLG_PASS_PCM_DAI) return; @@ -509,6 +510,10 @@ static void remove_dai(struct snd_soc_component *comp, if (dobj->ops && dobj->ops->dai_unload) dobj->ops->dai_unload(comp, dobj); + list_for_each_entry(dai, &comp->dai_list, list) + if (dai->driver == dai_drv) + dai->driver = NULL; + kfree(dai_drv->name); list_del(&dobj->list); kfree(dai_drv); -- 2.17.1

1 0

[alsa-devel] [PATCH] ASoC:topology:bug fix:check return value avoid oops.
by Guennadi Liakhovetski 21 Sep '18

21 Sep '18

From: Wu Zhigang <zhigang.wu(a)linux.intel.com> check the return value to free the kcontrols instance to avoid oops caused by the pointer dereference. Signed-off-by: Wu Zhigang <zhigang.wu(a)linux.intel.com> [guennadi.liakhovetski(a)linux.intel.com add handling of .autodisable=1 cases] Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/soc/soc-dapm.c | 99 ++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 90 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7e96793050c9..8dc31a7a42fd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2402,12 +2402,11 @@ static void dapm_free_path(struct snd_soc_dapm_path *path) kfree(path); } -void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) +static void snd_soc_dapm_free_widget_data(struct snd_soc_dapm_widget *w) { struct snd_soc_dapm_path *p, *next_p; enum snd_soc_dapm_direction dir; - list_del(&w->list); /* * remove source and sink paths associated to this widget. * While removing the path, remove reference to it from both @@ -2420,6 +2419,12 @@ void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) kfree(w->kcontrols); kfree_const(w->name); +} + +void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) +{ + list_del(&w->list); + snd_soc_dapm_free_widget_data(w); kfree(w); } @@ -3038,11 +3043,16 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); */ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) { - struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_widget *w, *last; unsigned int val; + int ret = 0; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); + /* + * widgets with the snd_soc_dapm_kcontrol ID and .num_controls = 0 can + * be appended to the list while scanning it, this is safe. + */ list_for_each_entry(w, &card->widgets, list) { if (w->new) @@ -3053,8 +3063,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) sizeof(struct snd_kcontrol *), GFP_KERNEL); if (!w->kcontrols) { - mutex_unlock(&card->dapm_mutex); - return -ENOMEM; + ret = -ENOMEM; + goto out_free; } } @@ -3062,23 +3072,29 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_switch: case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: - dapm_new_mixer(w); + ret = dapm_new_mixer(w); break; case snd_soc_dapm_mux: case snd_soc_dapm_demux: - dapm_new_mux(w); + ret = dapm_new_mux(w); break; case snd_soc_dapm_pga: case snd_soc_dapm_out_drv: - dapm_new_pga(w); + ret = dapm_new_pga(w); break; case snd_soc_dapm_dai_link: - dapm_new_dai_link(w); + ret = dapm_new_dai_link(w); break; default: break; } + if (ret < 0) { + kfree(w->kcontrols); + w->kcontrols = NULL; + goto out_free; + } + /* Read the initial power state from the device */ if (w->reg >= 0) { soc_dapm_read(w->dapm, w->reg, &val); @@ -3087,6 +3103,11 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) if (val == w->on_val) w->power = 1; } + } + + list_for_each_entry(w, &card->widgets, list) { + if (w->new) + continue; w->new = 1; @@ -3097,6 +3118,66 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) dapm_power_widgets(card, SND_SOC_DAPM_STREAM_NOP); mutex_unlock(&card->dapm_mutex); return 0; + +out_free: + last = w; + + /* + * If any new widgets have been created above for .autodisable = 1 + * controls, they are also on this list, but at its very end. We're + * processing an error case, so the loop above was interrupted before + * reaching dynamically added widgets, since the latter cannot fail - + * their .num_controls = 0, so allocation cannot fail, and their type is + * snd_soc_dapm_kcontrol, those cannot fail either. Therefore "last" + * points to a widget before the first dynamic one. + */ + list_for_each_entry(w, &card->widgets, list) { + unsigned int i; + + if (w->new) + continue; + + if (w == last) + break; + + switch (w->id) { + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: + for (i = 0; i < w->num_kcontrols; i++) { + struct snd_kcontrol *kcontrol = w->kcontrols[i]; + struct dapm_kcontrol_data *data = + kcontrol->private_data; + struct soc_mixer_control *mc = + (struct soc_mixer_control *) + kcontrol->private_value; + + if (mc->autodisable) + snd_soc_dapm_free_widget(data->widget); + } + break; + case snd_soc_dapm_demux: + case snd_soc_dapm_mux: + for (i = 0; i < w->num_kcontrols; i++) { + struct snd_kcontrol *kcontrol = w->kcontrols[i]; + struct dapm_kcontrol_data *data = + kcontrol->private_data; + struct soc_enum *e = (struct soc_enum *) + kcontrol->private_value; + + if (e->autodisable) + snd_soc_dapm_free_widget(data->widget); + } + break; + default: + break; + } + + snd_soc_dapm_free_widget_data(w); + } + + mutex_unlock(&card->dapm_mutex); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); -- 2.17.1

1 0

[alsa-devel] [PATCH v2] ALSA: xen-front: Refine indentations and constify snd_pcm_ops
by Nick Simonov 21 Sep '18

21 Sep '18

snd_pcm_ops are not supposed to change. So mark the non-const structs as const. Also, refine indentation to increase readability. Signed-off-by: Nick Simonov <nicksimonovv(a)gmail.com> --- Changes v2: - Fix typo in commit description, ncrease to increase. sound/xen/xen_snd_front_alsa.c | 46 +++++++++++++++++++++--------------------- 1 file changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c index 129180e..2cbd967 100644 --- a/sound/xen/xen_snd_front_alsa.c +++ b/sound/xen/xen_snd_front_alsa.c @@ -637,31 +637,31 @@ static int alsa_pb_fill_silence(struct snd_pcm_substream *substream, * to know when the buffer can be transferred to the backend. */ -static struct snd_pcm_ops snd_drv_alsa_playback_ops = { - .open = alsa_open, - .close = alsa_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = alsa_hw_params, - .hw_free = alsa_hw_free, - .prepare = alsa_prepare, - .trigger = alsa_trigger, - .pointer = alsa_pointer, - .copy_user = alsa_pb_copy_user, - .copy_kernel = alsa_pb_copy_kernel, - .fill_silence = alsa_pb_fill_silence, +static const struct snd_pcm_ops snd_drv_alsa_playback_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_pb_copy_user, + .copy_kernel = alsa_pb_copy_kernel, + .fill_silence = alsa_pb_fill_silence, }; -static struct snd_pcm_ops snd_drv_alsa_capture_ops = { - .open = alsa_open, - .close = alsa_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = alsa_hw_params, - .hw_free = alsa_hw_free, - .prepare = alsa_prepare, - .trigger = alsa_trigger, - .pointer = alsa_pointer, - .copy_user = alsa_cap_copy_user, - .copy_kernel = alsa_cap_copy_kernel, +static const struct snd_pcm_ops snd_drv_alsa_capture_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_cap_copy_user, + .copy_kernel = alsa_cap_copy_kernel, }; static int new_pcm_instance(struct xen_snd_front_card_info *card_info, -- 2.7.4

2 3

[alsa-devel] [PATCH 00/11] add add for_each_xx() macro
by Kuninori Morimoto 20 Sep '18

20 Sep '18

Hi Mark Current ALSA SoC is using many list_for_each_entry() and/or for() loop for each lists. But, these need to use same pattern, and it is not readable. Now, "codec list loop for rtd", and "codec list loop for dai_link" had been replaced to for_each_xxx() macro. These patches replace "xxx list loop for yyy" by using for_each_yyy_xxx() macro. # Maybe some of them are overkill replacement, and/or # using not good naming. I'm happy to get feedback 1) - 2) : fixup previous for_each_xxx() macro patch 3) - 11) : new for_each_xxx() macro Kuninori Morimoto (11): 1) ASoC: convert for_each_rtd_codec_dai() for missing part 2) ASoC: rename for_each_rtd_codec_dai_reverse to rollback 3) ASoC: add for_each_card_prelinks() macro 4) ASoC: add for_each_card_links() macro 5) ASoC: add for_each_card_rtds() macro 6) ASoC: add for_each_card_components() macro 7) ASoC: add for_each_component_all() macro 8) ASoC: add for_each_component_dais() macro 9) ASoC: add for_each_comp_order() macro 10) ASoC: add for_each_dpcm_fe() macro 11) ASoC: add for_each_dpcm_be() macro include/sound/soc-dpcm.h | 10 ++ include/sound/soc.h | 29 ++++- sound/soc/codecs/hdac_hdmi.c | 2 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/pcm030-audio-fabric.c | 5 +- sound/soc/generic/simple-card-utils.c | 6 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 4 +- sound/soc/intel/boards/broadwell.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 4 +- sound/soc/intel/boards/bytcr_rt5651.c | 4 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 4 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 5 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 13 +-- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 13 +-- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 13 +-- sound/soc/mediatek/mt8173/mt8173-max98090.c | 13 +-- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 7 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 7 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 7 +- sound/soc/meson/axg-card.c | 3 +- sound/soc/qcom/apq8096.c | 7 +- sound/soc/qcom/sdm845.c | 7 +- sound/soc/samsung/tm2_wm5110.c | 13 +-- sound/soc/sh/rcar/ctu.c | 2 +- sound/soc/sh/rcar/src.c | 2 +- sound/soc/soc-compress.c | 4 +- sound/soc/soc-core.c | 130 +++++++++++------------ sound/soc/soc-dapm.c | 2 +- sound/soc/soc-pcm.c | 101 +++++++++--------- 29 files changed, 230 insertions(+), 193 deletions(-) -- 2.7.4

2 21

[alsa-devel] Applied "ASoC: rt5663: Remove the boost volume in the beginning of playback" to the asoc tree
by Mark Brown 20 Sep '18

20 Sep '18

The patch ASoC: rt5663: Remove the boost volume in the beginning of playback has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From fc795bf7224efda8c35e505966c2757856064247 Mon Sep 17 00:00:00 2001 From: Oder Chiou <oder_chiou(a)realtek.com> Date: Wed, 19 Sep 2018 09:56:47 +0800 Subject: [PATCH] ASoC: rt5663: Remove the boost volume in the beginning of playback The patch removes the boost volume in the beginning of playback while the DAC volume set to lower. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5663.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 9bd24ad42240..2444fad7c2df 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -72,6 +72,7 @@ struct rt5663_priv { static const struct reg_sequence rt5663_patch_list[] = { { 0x002a, 0x8020 }, { 0x0086, 0x0028 }, + { 0x0100, 0xa020 }, { 0x0117, 0x0f28 }, { 0x02fb, 0x8089 }, }; @@ -580,7 +581,7 @@ static const struct reg_default rt5663_reg[] = { { 0x00fd, 0x0001 }, { 0x00fe, 0x10ec }, { 0x00ff, 0x6406 }, - { 0x0100, 0xa0a0 }, + { 0x0100, 0xa020 }, { 0x0108, 0x4444 }, { 0x0109, 0x4444 }, { 0x010a, 0xaaaa }, @@ -2337,6 +2338,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, 0x8000); snd_soc_component_update_bits(component, RT5663_DEPOP_1, 0x3000, 0x3000); + snd_soc_component_update_bits(component, + RT5663_DIG_VOL_ZCD, 0x00c0, 0x0080); } break; @@ -2351,6 +2354,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, RT5663_OVCD_HP_MASK, RT5663_OVCD_HP_EN); snd_soc_component_update_bits(component, RT5663_DACREF_LDO, 0x3e0e, 0); + snd_soc_component_update_bits(component, + RT5663_DIG_VOL_ZCD, 0x00c0, 0); } break; -- 2.19.0

1 0

[alsa-devel] [PATCH v2 00/12] Sound Open Firmware Core
by Liam Girdwood 20 Sep '18

20 Sep '18

Sound Open Firmware (SOF) is a host and DSP architecture agnostic audio DSP firmware. SOF is not tied to any specific host architecture or any specific physical IO communication type (it will work with on SoC DSPs, or DSP connected via SPI/I2C). SOF is also not coupled to any particular DSP architecture and has abstraction similar to Linux to allow porting to other DSP architectures. https://www.sofproject.org/ This patch series introduces the SOF core, later series will add host platform support for SOF. The SOF core manages all the core DSP services and ALSA/ASoC IO. The core is responsible for loading firmware, parsing topology, exposing PCMs and kcontrols, providing debug and trace mechanisms and performing IPC between host and DSP. The SOF core also has logic to allow reuse of existing machine drivers for other platforms/machines without any code modification. i.e. DAI links can be modified at runtime to bind with SOF and SOF topologies instead of existing hard coded DAI links and topology. Changes since V1 :- o Added PM source. o Updated header comments to C++ style o Used generic private data in snd_soc_pcm_runtime o remove sof_get_pages() o Added comments for compressed SG pages. o Check return values for runtime_pm calls. Liam Girdwood (12): ASoC: SOF: Add Sound Open Firmware driver core ASoC: SOF: Add Sound Open Firmware KControl support ASoC: SOF: Add driver debug support. ASoC: SOF: Add support for IPC IO between DSP and Host ASoC: SOF: Add PCM operations support ASoC: SOF: Add support for loading topologies ASoC: SOF: Add DSP firmware trace event support ASoC: SOF: Add DSP HW abstraction operations ASoC: SOF: Add firmware loader support ASoC: SOF: Add PM support ASoC: SOF: Add userspace ABI support ASoC: SOF: Add Build support for SOF core include/sound/soc.h | 3 + include/sound/sof.h | 82 ++ include/uapi/sound/sof-abi.h | 29 + include/uapi/sound/sof-eq.h | 102 ++ include/uapi/sound/sof-fw.h | 67 + include/uapi/sound/sof-ipc.h | 938 ++++++++++++ include/uapi/sound/sof-tone.h | 27 + include/uapi/sound/sof-topology.h | 94 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/sof/Kconfig | 36 + sound/soc/sof/Makefile | 6 + sound/soc/sof/control.c | 270 ++++ sound/soc/sof/core.c | 381 +++++ sound/soc/sof/debug.c | 157 ++ sound/soc/sof/ipc.c | 788 ++++++++++ sound/soc/sof/loader.c | 287 ++++ sound/soc/sof/ops.c | 210 +++ sound/soc/sof/ops.h | 298 ++++ sound/soc/sof/pcm.c | 733 ++++++++++ sound/soc/sof/pm.c | 345 +++++ sound/soc/sof/sof-priv.h | 519 +++++++ sound/soc/sof/topology.c | 2222 +++++++++++++++++++++++++++++ sound/soc/sof/trace.c | 293 ++++ 24 files changed, 7889 insertions(+) create mode 100644 include/sound/sof.h create mode 100644 include/uapi/sound/sof-abi.h create mode 100644 include/uapi/sound/sof-eq.h create mode 100644 include/uapi/sound/sof-fw.h create mode 100644 include/uapi/sound/sof-ipc.h create mode 100644 include/uapi/sound/sof-tone.h create mode 100644 include/uapi/sound/sof-topology.h create mode 100644 sound/soc/sof/Kconfig create mode 100644 sound/soc/sof/Makefile create mode 100644 sound/soc/sof/control.c create mode 100644 sound/soc/sof/core.c create mode 100644 sound/soc/sof/debug.c create mode 100644 sound/soc/sof/ipc.c create mode 100644 sound/soc/sof/loader.c create mode 100644 sound/soc/sof/ops.c create mode 100644 sound/soc/sof/ops.h create mode 100644 sound/soc/sof/pcm.c create mode 100644 sound/soc/sof/pm.c create mode 100644 sound/soc/sof/sof-priv.h create mode 100644 sound/soc/sof/topology.c create mode 100644 sound/soc/sof/trace.c -- 2.17.1

2 22

[alsa-devel] SoftVol in user space
by Riccardo Casagrande 20 Sep '18

20 Sep '18

Hello devs, I have to add a volume control to my custom card (It has an i.MX6 chip and 3 FDA2100 from TI as audio chip) because the FDAs have no volume controller. This below is the .asoundrc file I made. It works great if I manually made a speaker-test of one of devicefdaX sound card. Then I can see on alsamixer the volume control, but my need is to access and control softvol from user space. pcm.softvolfda0 { type softvol slave { pcm "hw:0,0" } control { name "volume0" card 0 device 0 } } pcm.softvolfda1 { type softvol slave { pcm "hw:0,1" } control { name "volume1" card 0 device 1 } } pcm.softvolfda2 { type softvol slave { pcm "hw:0,2" } control { name "volume2" card 0 device 2 } } pcm.devicefda0 { type plug slave.pcm "softvolfda0" } pcm.devicefda1 { type plug slave.pcm "softvolfda1" } pcm.devicefda2 { type plug slave.pcm "softvolfda2" } I saw that on alsalib there is this command int snd_pcm_softvol_open (snd_pcm_t ** pcmp, const char * name, snd_pcm_format_t sformat, int ctl_card, snd_ctl_elem_id_t * ctl_id, int cchannels, double min_dB, double max_dB, int resolution, snd_pcm_t * slave, int close_slave ) but I can't find any example of usage.This is the log of the error: mp && slave' failed. Aborted Any suggestion is welcome Riccardo Casagrande

1 0

[alsa-devel] [PATCH 00/15] Add support for Sound BlasterX AE-5
by Connor McAdams 20 Sep '18

20 Sep '18

This patch series adds support for the Sound BlasterX AE-5. I have personally tested it, and had others test it as well. All outputs work, front headphone, rear headphone, rear line-out, surround, digital. The only issue currently is that the microphone suffers from inconsistency like the Sound Blaster Z does, and Creative may be willing to help fix that, but if they're not, I have a few ideas. I also have the ZxR working currently. That will require some more complicated changes of the driver, so I have left it out and just focused on the AE-5. Connor McAdams (15): ALSA: hda/ca0132 - Define new verbs and control params ALSA: hda/ca0132 - Add quirk for Sound BlasterX AE-5 ALSA: hda/ca0132 - Add pincfg for AE-5 ALSA: hda/ca0132 - Set AE-5 bools and select mixer ALSA: hda/ca0132 - Change ca0132_mmio_init for AE-5 ALSA: hda/ca0132 - Add AE-5 pre-init and ca0113 functions ALSA: hda/ca0132 - Add AE-5 regular init setup ALSA: hda/ca0132 - Change firmware name and usage ALSA: hda/ca0132 - Merge post-dsp functions + cleanup ALSA: hda/ca0132 - Add DSP setup functions for AE-5 ALSA: hda/ca0132 - Clean up ca0132_alt_out_select ALSA: hda/ca0132 - Add output set commands for AE-5 ALSA: hda/ca0132 - Add input selection commands for AE-5 ALSA: hda/ca0132 - Add AE-5 specific controls ALSA: hda/ca0132 - Add AE-5 exit function sound/pci/hda/patch_ca0132.c | 1283 ++++++++++++++++++++++++++++++++++-------- 1 file changed, 1038 insertions(+), 245 deletions(-) -- 2.7.4

2 17

[alsa-devel] [PATCH - hwmixvolume v2 1/7] hwmixvolume: use four spaces instead of one tab for indent
by Emmanuel Gil Peyrot 20 Sep '18

20 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> This is the recommended coding style for all Python programs, as specified in PEP-0008[1]. [1] https://www.python.org/dev/peps/pep-0008/ Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index ef80bc8..038bcb3 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -26,285 +26,285 @@ EVENT_INFO = alsahcontrol.event_mask['INFO'] EVENT_REMOVE = alsahcontrol.event_mask_remove class Stream: - def __init__(self, element, parent): - self.element = element - self.element.set_callback(self) - self.parent = parent - self.label = None - self.scales = [] - self.adjustments = [] - self.callback(self.element, EVENT_INFO) - - def destroy(self): - self.deactivate() - - def callback(self, e, mask): - if mask == EVENT_REMOVE: - self.deactivate() - elif (mask & EVENT_INFO) != 0: - info = alsahcontrol.Info(self.element) - if info.is_inactive: - self.deactivate() - else: - self.activate() - elif (mask & EVENT_VALUE) != 0: - self.update_scales_from_ctl() - - def activate(self): - if self.label: - return - info = alsahcontrol.Info(self.element) - value = alsahcontrol.Value(self.element) - value.read() - values = value.get_tuple(TYPE_INTEGER, info.count) - self.label = gtk.Label(self.get_label(info)) - self.label.set_single_line_mode(True) - self.parent.scales_vbox.pack_start(self.label, expand=False) - for i in range(info.count): - adj = gtk.Adjustment(value=values[i], - lower=info.min, upper=info.max, - step_incr=1, - page_incr=(info.max-info.min+1)/8) - adj.connect('value-changed', self.update_ctl_from_scale, i) - scale = gtk.HScale(adj) - scale.set_draw_value(False) - self.parent.scales_vbox.pack_start(scale, expand=False) - self.scales.append(scale) - self.adjustments.append(adj) - self.parent.scales_vbox.show_all() - self.parent.update_msg_label() - - def deactivate(self): - if not self.label: - return - self.label.destroy() - for s in self.scales: - s.destroy() - self.label = None - self.scales = [] - self.adjustments = [] - self.parent.update_msg_label() - - def update_scales_from_ctl(self): - if not self.label: - return - count = len(self.adjustments) - value = alsahcontrol.Value(self.element) - value.read() - values = value.get_tuple(TYPE_INTEGER, count) - for i in range(count): - self.adjustments[i].set_value(values[i]) - - def update_ctl_from_scale(self, adj, index): - scale_value = adj.get_value() - value_to_set = int(round(adj.get_value())) - count = len(self.adjustments) - value = alsahcontrol.Value(self.element) - if self.parent.lock_check.get_active(): - values = [value_to_set for i in range(count)] - else: - value.read() - values = value.get_array(TYPE_INTEGER, count) - values[index] = value_to_set - value.set_array(TYPE_INTEGER, values) - value.write() - if value_to_set != scale_value: - adj.set_value(value_to_set) - - def get_label(self, info): - pid = self.get_pid(info) - if pid: - cmdline = self.get_pid_cmdline(pid) - if cmdline: - return cmdline - else: - return "PID %d" % pid - else: - name = info.name - if name[-7:] == " Volume": - name = name[:-7] - if name[-9:] == " Playback": - name = name[:-9] - return name - - def get_pid(self, info): - card = self.parent.current_card - device = info.device - subdevice = info.subdevice - if subdevice == 0: - subdevice = info.index - filename = "/proc/asound/card%d/pcm%dp/sub%d/status" % (card, device, subdevice) - try: - f = open(filename, "r") - except IOError: - return None - try: - for line in f.readlines(): - if line[:9] == "owner_pid": - return int(line.split(':')[1].strip()) - finally: - f.close() - return None - - def get_pid_cmdline(self, pid): - try: - f = open("/proc/%d/cmdline" % pid, "r") - except IOError: - return None - try: - cmdline = f.read() - finally: - f.close() - return cmdline.replace('\x00', ' ').strip() + def __init__(self, element, parent): + self.element = element + self.element.set_callback(self) + self.parent = parent + self.label = None + self.scales = [] + self.adjustments = [] + self.callback(self.element, EVENT_INFO) + + def destroy(self): + self.deactivate() + + def callback(self, e, mask): + if mask == EVENT_REMOVE: + self.deactivate() + elif (mask & EVENT_INFO) != 0: + info = alsahcontrol.Info(self.element) + if info.is_inactive: + self.deactivate() + else: + self.activate() + elif (mask & EVENT_VALUE) != 0: + self.update_scales_from_ctl() + + def activate(self): + if self.label: + return + info = alsahcontrol.Info(self.element) + value = alsahcontrol.Value(self.element) + value.read() + values = value.get_tuple(TYPE_INTEGER, info.count) + self.label = gtk.Label(self.get_label(info)) + self.label.set_single_line_mode(True) + self.parent.scales_vbox.pack_start(self.label, expand=False) + for i in range(info.count): + adj = gtk.Adjustment(value=values[i], + lower=info.min, upper=info.max, + step_incr=1, + page_incr=(info.max-info.min+1)/8) + adj.connect('value-changed', self.update_ctl_from_scale, i) + scale = gtk.HScale(adj) + scale.set_draw_value(False) + self.parent.scales_vbox.pack_start(scale, expand=False) + self.scales.append(scale) + self.adjustments.append(adj) + self.parent.scales_vbox.show_all() + self.parent.update_msg_label() + + def deactivate(self): + if not self.label: + return + self.label.destroy() + for s in self.scales: + s.destroy() + self.label = None + self.scales = [] + self.adjustments = [] + self.parent.update_msg_label() + + def update_scales_from_ctl(self): + if not self.label: + return + count = len(self.adjustments) + value = alsahcontrol.Value(self.element) + value.read() + values = value.get_tuple(TYPE_INTEGER, count) + for i in range(count): + self.adjustments[i].set_value(values[i]) + + def update_ctl_from_scale(self, adj, index): + scale_value = adj.get_value() + value_to_set = int(round(adj.get_value())) + count = len(self.adjustments) + value = alsahcontrol.Value(self.element) + if self.parent.lock_check.get_active(): + values = [value_to_set for i in range(count)] + else: + value.read() + values = value.get_array(TYPE_INTEGER, count) + values[index] = value_to_set + value.set_array(TYPE_INTEGER, values) + value.write() + if value_to_set != scale_value: + adj.set_value(value_to_set) + + def get_label(self, info): + pid = self.get_pid(info) + if pid: + cmdline = self.get_pid_cmdline(pid) + if cmdline: + return cmdline + else: + return "PID %d" % pid + else: + name = info.name + if name[-7:] == " Volume": + name = name[:-7] + if name[-9:] == " Playback": + name = name[:-9] + return name + + def get_pid(self, info): + card = self.parent.current_card + device = info.device + subdevice = info.subdevice + if subdevice == 0: + subdevice = info.index + filename = "/proc/asound/card%d/pcm%dp/sub%d/status" % (card, device, subdevice) + try: + f = open(filename, "r") + except IOError: + return None + try: + for line in f.readlines(): + if line[:9] == "owner_pid": + return int(line.split(':')[1].strip()) + finally: + f.close() + return None + + def get_pid_cmdline(self, pid): + try: + f = open("/proc/%d/cmdline" % pid, "r") + except IOError: + return None + try: + cmdline = f.read() + finally: + f.close() + return cmdline.replace('\x00', ' ').strip() class MixerWindow(gtk.Window): - card_numbers = alsacard.card_list() - current_card = -1 - hcontrol = None - scales_vbox = None - msg_label = None - streams = [] - hctl_sources = [] - - def __init__(self): - gtk.Window.__init__(self) - self.connect('destroy', lambda w: gtk.main_quit()) - self.set_title("Hardware Mixer Volumes") - - vbox = gtk.VBox() - self.add(vbox) - - hbox = gtk.HBox() - vbox.pack_start(hbox, expand=False) - - label = gtk.Label("_Sound Card: ") - label.set_use_underline(True) - hbox.pack_start(label, expand=False) - - combo = gtk.combo_box_new_text() - for i in self.card_numbers: - str = "%d: %s" % (i, alsacard.card_get_name(i)) - combo.append_text(str) - if len(self.card_numbers) > 0: - combo.set_active(0) - combo.connect('changed', lambda c: self.change_card(self.card_numbers[combo.get_active()])) - hbox.pack_start(combo) - label.set_mnemonic_widget(combo) - - self.lock_check = gtk.CheckButton(label="_Lock Channels") - self.lock_check.set_active(True) - vbox.pack_start(self.lock_check, expand=False) - - scrollwin = gtk.ScrolledWindow() - scrollwin.set_policy(hscrollbar_policy=gtk.POLICY_NEVER, vscrollbar_policy=gtk.POLICY_AUTOMATIC) - scrollwin.set_shadow_type(gtk.SHADOW_NONE) - vbox.pack_start(scrollwin) - - self.scales_vbox = gtk.VBox() - scrollwin.add_with_viewport(self.scales_vbox) - - label = gtk.Label() - label.set_single_line_mode(True) - line_height = label.size_request()[1] - label.destroy() - scale = gtk.HScale() - scale.set_draw_value(False) - line_height += scale.size_request()[1] - scale.destroy() - # always have space for at least four sliders - scrollwin.set_size_request(width=-1, height=line_height*4+4) - - # TODO: select the default card or the first card with stream controls - if len(self.card_numbers) > 0: - self.change_card(self.card_numbers[0]) - self.update_msg_label() - - self.show_all() - - def change_card(self, cardnum): - for s in self.hctl_sources: - gobject.source_remove(s) - self.hctl_sources = [] - - self.hcontrol = self.open_hcontrol_for_card(cardnum) - - for s in self.streams: - s.destroy() - self.streams = [] - - self.current_card = cardnum - - if not self.hcontrol: - self.update_msg_label() - return - - for id in self.hcontrol.list(): - if not self.is_stream_elem(id): - continue - elem = alsahcontrol.Element(self.hcontrol, id[0]) - info = alsahcontrol.Info(elem) - if not self.is_stream_info(info): - continue - stream = Stream(elem, self) - self.streams.append(stream) - - for fd,condition in self.hcontrol.poll_fds: - self.hctl_sources.append(gobject.io_add_watch(fd, condition, self.hctl_io_callback)) - - self.update_msg_label() - - self.scales_vbox.show_all() - - def update_msg_label(self): - needs_msg = len(self.scales_vbox.get_children()) < 2 - has_msg = self.msg_label - if has_msg and not needs_msg: - self.msg_label.destroy() - self.msg_label = None - elif needs_msg: - if len(self.streams) > 0: - msg = "There are no open streams." - else: - msg = "This card does not have stream controls." - if not has_msg: - self.msg_label = gtk.Label(msg) - self.scales_vbox.pack_start(self.msg_label) - self.scales_vbox.show_all() - elif self.msg_label.get_text() != msg: - self.msg_label.set_text(msg) - - def open_hcontrol_for_card(self, cardnum): - devname = "hw:CARD=" + str(cardnum) - try: - hc = alsahcontrol.HControl(name=devname, - mode=alsahcontrol.open_mode['NONBLOCK']) - except: - # TODO: alsa error msg - dlg = gtk.MessageDialog(self, - gtk.DIALOG_MODAL | gtk.DIALOG_DESTROY_WITH_PARENT, - gtk.MESSAGE_ERROR, gtk.BUTTONS_OK, - "Cannot open sound card control device.") - dlg.run() - dlg.destroy() - return None - return hc - - def is_stream_elem(self, id): - return ((id[1] == INTF_PCM and - id[4] in ("PCM Playback Volume", "EMU10K1 PCM Volume")) or - (id[1] == INTF_MIXER and - id[4] == "VIA DXS Playback Volume")) - - def is_stream_info(self, info): - return info.is_readable and info.is_writable and info.type == TYPE_INTEGER - - def hctl_io_callback(self, source, condition): - self.hcontrol.handle_events() - return True + card_numbers = alsacard.card_list() + current_card = -1 + hcontrol = None + scales_vbox = None + msg_label = None + streams = [] + hctl_sources = [] + + def __init__(self): + gtk.Window.__init__(self) + self.connect('destroy', lambda w: gtk.main_quit()) + self.set_title("Hardware Mixer Volumes") + + vbox = gtk.VBox() + self.add(vbox) + + hbox = gtk.HBox() + vbox.pack_start(hbox, expand=False) + + label = gtk.Label("_Sound Card: ") + label.set_use_underline(True) + hbox.pack_start(label, expand=False) + + combo = gtk.combo_box_new_text() + for i in self.card_numbers: + str = "%d: %s" % (i, alsacard.card_get_name(i)) + combo.append_text(str) + if len(self.card_numbers) > 0: + combo.set_active(0) + combo.connect('changed', lambda c: self.change_card(self.card_numbers[combo.get_active()])) + hbox.pack_start(combo) + label.set_mnemonic_widget(combo) + + self.lock_check = gtk.CheckButton(label="_Lock Channels") + self.lock_check.set_active(True) + vbox.pack_start(self.lock_check, expand=False) + + scrollwin = gtk.ScrolledWindow() + scrollwin.set_policy(hscrollbar_policy=gtk.POLICY_NEVER, vscrollbar_policy=gtk.POLICY_AUTOMATIC) + scrollwin.set_shadow_type(gtk.SHADOW_NONE) + vbox.pack_start(scrollwin) + + self.scales_vbox = gtk.VBox() + scrollwin.add_with_viewport(self.scales_vbox) + + label = gtk.Label() + label.set_single_line_mode(True) + line_height = label.size_request()[1] + label.destroy() + scale = gtk.HScale() + scale.set_draw_value(False) + line_height += scale.size_request()[1] + scale.destroy() + # always have space for at least four sliders + scrollwin.set_size_request(width=-1, height=line_height*4+4) + + # TODO: select the default card or the first card with stream controls + if len(self.card_numbers) > 0: + self.change_card(self.card_numbers[0]) + self.update_msg_label() + + self.show_all() + + def change_card(self, cardnum): + for s in self.hctl_sources: + gobject.source_remove(s) + self.hctl_sources = [] + + self.hcontrol = self.open_hcontrol_for_card(cardnum) + + for s in self.streams: + s.destroy() + self.streams = [] + + self.current_card = cardnum + + if not self.hcontrol: + self.update_msg_label() + return + + for id in self.hcontrol.list(): + if not self.is_stream_elem(id): + continue + elem = alsahcontrol.Element(self.hcontrol, id[0]) + info = alsahcontrol.Info(elem) + if not self.is_stream_info(info): + continue + stream = Stream(elem, self) + self.streams.append(stream) + + for fd,condition in self.hcontrol.poll_fds: + self.hctl_sources.append(gobject.io_add_watch(fd, condition, self.hctl_io_callback)) + + self.update_msg_label() + + self.scales_vbox.show_all() + + def update_msg_label(self): + needs_msg = len(self.scales_vbox.get_children()) < 2 + has_msg = self.msg_label + if has_msg and not needs_msg: + self.msg_label.destroy() + self.msg_label = None + elif needs_msg: + if len(self.streams) > 0: + msg = "There are no open streams." + else: + msg = "This card does not have stream controls." + if not has_msg: + self.msg_label = gtk.Label(msg) + self.scales_vbox.pack_start(self.msg_label) + self.scales_vbox.show_all() + elif self.msg_label.get_text() != msg: + self.msg_label.set_text(msg) + + def open_hcontrol_for_card(self, cardnum): + devname = "hw:CARD=" + str(cardnum) + try: + hc = alsahcontrol.HControl(name=devname, + mode=alsahcontrol.open_mode['NONBLOCK']) + except: + # TODO: alsa error msg + dlg = gtk.MessageDialog(self, + gtk.DIALOG_MODAL | gtk.DIALOG_DESTROY_WITH_PARENT, + gtk.MESSAGE_ERROR, gtk.BUTTONS_OK, + "Cannot open sound card control device.") + dlg.run() + dlg.destroy() + return None + return hc + + def is_stream_elem(self, id): + return ((id[1] == INTF_PCM and + id[4] in ("PCM Playback Volume", "EMU10K1 PCM Volume")) or + (id[1] == INTF_MIXER and + id[4] == "VIA DXS Playback Volume")) + + def is_stream_info(self, info): + return info.is_readable and info.is_writable and info.type == TYPE_INTEGER + + def hctl_io_callback(self, source, condition): + self.hcontrol.handle_events() + return True def main(): - MixerWindow() - gtk.main() + MixerWindow() + gtk.main() main() -- 2.19.0

3 5

[alsa-devel] [PATCH] ALSA: xen-front: Refine indentations and constify snd_pcm_ops
by Nick Simonov 20 Sep '18

20 Sep '18

snd_pcm_ops are not supposed to change. So mark the non-const structs as const. Also, refine indentation to ncrease readability. Signed-off-by: Nick Simonov <nicksimonovv(a)gmail.com> --- sound/xen/xen_snd_front_alsa.c | 46 +++++++++++++++++++++--------------------- 1 file changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c index 129180e..2cbd967 100644 --- a/sound/xen/xen_snd_front_alsa.c +++ b/sound/xen/xen_snd_front_alsa.c @@ -637,31 +637,31 @@ static int alsa_pb_fill_silence(struct snd_pcm_substream *substream, * to know when the buffer can be transferred to the backend. */ -static struct snd_pcm_ops snd_drv_alsa_playback_ops = { - .open = alsa_open, - .close = alsa_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = alsa_hw_params, - .hw_free = alsa_hw_free, - .prepare = alsa_prepare, - .trigger = alsa_trigger, - .pointer = alsa_pointer, - .copy_user = alsa_pb_copy_user, - .copy_kernel = alsa_pb_copy_kernel, - .fill_silence = alsa_pb_fill_silence, +static const struct snd_pcm_ops snd_drv_alsa_playback_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_pb_copy_user, + .copy_kernel = alsa_pb_copy_kernel, + .fill_silence = alsa_pb_fill_silence, }; -static struct snd_pcm_ops snd_drv_alsa_capture_ops = { - .open = alsa_open, - .close = alsa_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = alsa_hw_params, - .hw_free = alsa_hw_free, - .prepare = alsa_prepare, - .trigger = alsa_trigger, - .pointer = alsa_pointer, - .copy_user = alsa_cap_copy_user, - .copy_kernel = alsa_cap_copy_kernel, +static const struct snd_pcm_ops snd_drv_alsa_capture_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_cap_copy_user, + .copy_kernel = alsa_cap_copy_kernel, }; static int new_pcm_instance(struct xen_snd_front_card_info *card_info, -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: q6afe: dt-bindings: Update input range for qcom, sd-lines
by Rohit kumar 19 Sep '18

19 Sep '18

Input to qcom,sd-lines should be between 0 and 3 instead of 1 to 4 as 0 corresponds to BIT(0) which is MI2S_SD0 line. Bit 1 to 3 corresponds to SD1 to SD3 lines respectively. Updated documentation for the same. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- Documentation/devicetree/bindings/sound/qcom,q6afe.txt | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt index a817940..d74888b 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt @@ -49,7 +49,7 @@ configuration of each dai. Must contain the following properties. Usage: required for mi2s interface Value type: <prop-encoded-array> Definition: Must be list of serial data lines used by this dai. - should be one or more of the 1-4 sd lines. + should be one or more of the 0-3 sd lines. - qcom,tdm-sync-mode: Usage: required for tdm interface @@ -137,42 +137,42 @@ q6afe@4 { prim-mi2s-rx@16 { reg = <16>; - qcom,sd-lines = <1 3>; + qcom,sd-lines = <0 2>; }; prim-mi2s-tx@17 { reg = <17>; - qcom,sd-lines = <2>; + qcom,sd-lines = <1>; }; sec-mi2s-rx@18 { reg = <18>; - qcom,sd-lines = <1 4>; + qcom,sd-lines = <0 3>; }; sec-mi2s-tx@19 { reg = <19>; - qcom,sd-lines = <2>; + qcom,sd-lines = <1>; }; tert-mi2s-rx@20 { reg = <20>; - qcom,sd-lines = <2 4>; + qcom,sd-lines = <1 3>; }; tert-mi2s-tx@21 { reg = <21>; - qcom,sd-lines = <1>; + qcom,sd-lines = <0>; }; quat-mi2s-rx@22 { reg = <22>; - qcom,sd-lines = <1>; + qcom,sd-lines = <0>; }; quat-mi2s-tx@23 { reg = <23>; - qcom,sd-lines = <2>; + qcom,sd-lines = <1>; }; }; }; -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 1

Re: [alsa-devel] [PATCH - hwmixvolume v2 0/7] Migrate to GTK+ 3.0 and allow python3
by Takashi Sakamoto 19 Sep '18

19 Sep '18

Hi, On Sep 18 2018 22:42, Emmanuel Gil Peyrot wrote: > From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> > > hwmixvolume was one of the last few programs on my system to depend on > PyGTK. This series ports it to GTK+ 3.x using gobject-introspection, > which allows it to run on Python 3.x too. > > My audio card does not have stream controls so I’ve been unable to test > whether the functionality is working properly, but from my reading of > the documentation it should be fine. > > Changes since v1: > - Reordered tab→space patch before everything else, to make it possible > to merge it as is without any other change. > - Merged fixup commits into the main switch patch, as they had no reason > to stand by themselves. > - Replaced iteration on file.readlines() with file.__iter__() directly > (thanks Sakamoto-san). > - Fix a typo in the README (thanks Sakamoto-san). > - Explicit more the reason for each patch (thanks Iwai-san), except for > trivial ones. > > Emmanuel Gil Peyrot (7): > hwmixvolume: use four spaces instead of one tab for indent > hwmixvolume: replace PyGTK with gobject-introspection > hwmixvolume: switch to GTK+ 3.0 and GLib 2.0 > hwmixvolume: use a with context to open files > hwmixvolume: switch the shebang to python > hwmixvolume: add my copyright > hwmixvolume: mention the new dependencies in the README > > hwmixvolume/README | 3 +- > hwmixvolume/hwmixvolume | 560 ++++++++++++++++++++-------------------- > 2 files changed, 283 insertions(+), 280 deletions(-) I reviewed all of the above patches. Reviewed-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Thanks Takashi Sakamoto

1 0

[alsa-devel] [PATCH - hwmixvolume v2 3/7] hwmixvolume: switch to GTK+ 3.0 and GLib 2.0
by Emmanuel Gil Peyrot 19 Sep '18

19 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> This replaces VBox and HBox with Grid (using Gtk.Orientation), HScale with Scale, creates labels with mnemonics, set hexpand and vexpand properly, use the correct enum container classes, use the correct getter for size request, and finally update to the correct GLib watch function. Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 64d232c..8e0b6b8 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -60,18 +60,18 @@ class Stream: value = alsahcontrol.Value(self.element) value.read() values = value.get_tuple(TYPE_INTEGER, info.count) - self.label = Gtk.Label(self.get_label(info)) + self.label = Gtk.Label.new(self.get_label(info)) self.label.set_single_line_mode(True) - self.parent.scales_vbox.pack_start(self.label, expand=False) + self.parent.scales_vbox.add(self.label) for i in range(info.count): adj = Gtk.Adjustment(value=values[i], lower=info.min, upper=info.max, step_incr=1, page_incr=(info.max-info.min+1)/8) adj.connect('value-changed', self.update_ctl_from_scale, i) - scale = Gtk.HScale(adj) + scale = Gtk.Scale(orientation=Gtk.Orientation.HORIZONTAL, adjustment=adj) scale.set_draw_value(False) - self.parent.scales_vbox.pack_start(scale, expand=False) + self.parent.scales_vbox.add(scale) self.scales.append(scale) self.adjustments.append(adj) self.parent.scales_vbox.show_all() @@ -174,45 +174,48 @@ class MixerWindow(Gtk.Window): self.connect('destroy', lambda w: Gtk.main_quit()) self.set_title("Hardware Mixer Volumes") - vbox = Gtk.VBox() + vbox = Gtk.Grid() + vbox.set_orientation(Gtk.Orientation.VERTICAL) self.add(vbox) - hbox = Gtk.HBox() - vbox.pack_start(hbox, expand=False) + hbox = Gtk.Grid() + vbox.add(hbox) - label = Gtk.Label("_Sound Card: ") - label.set_use_underline(True) - hbox.pack_start(label, expand=False) + label = Gtk.Label.new_with_mnemonic("_Sound Card: ") + hbox.add(label) - combo = Gtk.combo_box_new_text() + combo = Gtk.ComboBoxText() + combo.set_hexpand(True) for i in self.card_numbers: str = "%d: %s" % (i, alsacard.card_get_name(i)) combo.append_text(str) if len(self.card_numbers) > 0: combo.set_active(0) combo.connect('changed', lambda c: self.change_card(self.card_numbers[combo.get_active()])) - hbox.pack_start(combo) + hbox.add(combo) label.set_mnemonic_widget(combo) - self.lock_check = Gtk.CheckButton(label="_Lock Channels") + self.lock_check = Gtk.CheckButton.new_with_mnemonic(label="_Lock Channels") self.lock_check.set_active(True) - vbox.pack_start(self.lock_check, expand=False) + vbox.add(self.lock_check) scrollwin = Gtk.ScrolledWindow() - scrollwin.set_policy(hscrollbar_policy=Gtk.POLICY_NEVER, vscrollbar_policy=Gtk.POLICY_AUTOMATIC) - scrollwin.set_shadow_type(Gtk.SHADOW_NONE) - vbox.pack_start(scrollwin) + scrollwin.set_policy(hscrollbar_policy=Gtk.PolicyType.NEVER, vscrollbar_policy=Gtk.PolicyType.AUTOMATIC) + scrollwin.set_shadow_type(Gtk.ShadowType.NONE) + scrollwin.set_vexpand(True) + vbox.add(scrollwin) - self.scales_vbox = Gtk.VBox() - scrollwin.add_with_viewport(self.scales_vbox) + self.scales_vbox = Gtk.Grid() + self.scales_vbox.set_orientation(Gtk.Orientation.VERTICAL) + scrollwin.add(self.scales_vbox) label = Gtk.Label() label.set_single_line_mode(True) - line_height = label.size_request()[1] + line_height = max(label.get_size_request().height, 0) label.destroy() - scale = Gtk.HScale() + scale = Gtk.Scale(orientation=Gtk.Orientation.HORIZONTAL) scale.set_draw_value(False) - line_height += scale.size_request()[1] + line_height += max(scale.get_size_request().height, 0) scale.destroy() # always have space for at least four sliders scrollwin.set_size_request(width=-1, height=line_height*4+4) @@ -252,7 +255,7 @@ class MixerWindow(Gtk.Window): self.streams.append(stream) for fd,condition in self.hcontrol.poll_fds: - self.hctl_sources.append(GLib.io_add_watch(fd, condition, self.hctl_io_callback)) + self.hctl_sources.append(GLib.io_add_watch(fd, 0, GLib.IOCondition(condition), self.hctl_io_callback)) self.update_msg_label() @@ -270,8 +273,9 @@ class MixerWindow(Gtk.Window): else: msg = "This card does not have stream controls." if not has_msg: - self.msg_label = Gtk.Label(msg) - self.scales_vbox.pack_start(self.msg_label) + self.msg_label = Gtk.Label.new(msg) + self.msg_label.set_vexpand(True) + self.scales_vbox.add(self.msg_label) self.scales_vbox.show_all() elif self.msg_label.get_text() != msg: self.msg_label.set_text(msg) @@ -284,8 +288,8 @@ class MixerWindow(Gtk.Window): except: # TODO: alsa error msg dlg = Gtk.MessageDialog(self, - Gtk.DIALOG_MODAL | Gtk.DIALOG_DESTROY_WITH_PARENT, - Gtk.MESSAGE_ERROR, Gtk.BUTTONS_OK, + Gtk.DialogFlags.MODAL | Gtk.DialogFlags.DESTROY_WITH_PARENT, + Gtk.MessageType.ERROR, Gtk.ButtonsType.OK, "Cannot open sound card control device.") dlg.run() dlg.destroy() -- 2.19.0

2 3

[alsa-devel] [PATCH 2/2] sound: soc: wm8903: Cut gpio_base data
by Linus Walleij 19 Sep '18

19 Sep '18

This variable is only used inside the file, and we should never hard-code the GPIO base, so delete this. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- sound/soc/codecs/wm8903.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 23e43ff40ded..545512dc4c45 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -144,7 +144,6 @@ struct wm8903_platform_data { int micdet_delay; /* Delay after microphone detection (ms) */ - int gpio_base; u32 gpio_cfg[WM8903_NUM_GPIO]; /* Default register values for GPIO pin mux */ }; @@ -1876,17 +1875,12 @@ static const struct gpio_chip wm8903_template_chip = { static void wm8903_init_gpio(struct wm8903_priv *wm8903) { - struct wm8903_platform_data *pdata = wm8903->pdata; int ret; wm8903->gpio_chip = wm8903_template_chip; wm8903->gpio_chip.ngpio = WM8903_NUM_GPIO; wm8903->gpio_chip.parent = wm8903->dev; - - if (pdata->gpio_base) - wm8903->gpio_chip.base = pdata->gpio_base; - else - wm8903->gpio_chip.base = -1; + wm8903->gpio_chip.base = -1; ret = gpiochip_add_data(&wm8903->gpio_chip, wm8903); if (ret != 0) -- 2.17.1

2 4

[alsa-devel] [PATCH Resend] ASoC: rt5663: Remove the boost volume in the beginning of playback
by Oder Chiou 19 Sep '18

19 Sep '18

The patch removes the boost volume in the beginning of playback while the DAC volume set to lower. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 9bd24ad..2444fad 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -72,6 +72,7 @@ struct rt5663_priv { static const struct reg_sequence rt5663_patch_list[] = { { 0x002a, 0x8020 }, { 0x0086, 0x0028 }, + { 0x0100, 0xa020 }, { 0x0117, 0x0f28 }, { 0x02fb, 0x8089 }, }; @@ -580,7 +581,7 @@ static const struct reg_default rt5663_reg[] = { { 0x00fd, 0x0001 }, { 0x00fe, 0x10ec }, { 0x00ff, 0x6406 }, - { 0x0100, 0xa0a0 }, + { 0x0100, 0xa020 }, { 0x0108, 0x4444 }, { 0x0109, 0x4444 }, { 0x010a, 0xaaaa }, @@ -2337,6 +2338,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, 0x8000); snd_soc_component_update_bits(component, RT5663_DEPOP_1, 0x3000, 0x3000); + snd_soc_component_update_bits(component, + RT5663_DIG_VOL_ZCD, 0x00c0, 0x0080); } break; @@ -2351,6 +2354,8 @@ static int rt5663_hp_event(struct snd_soc_dapm_widget *w, RT5663_OVCD_HP_MASK, RT5663_OVCD_HP_EN); snd_soc_component_update_bits(component, RT5663_DACREF_LDO, 0x3e0e, 0); + snd_soc_component_update_bits(component, + RT5663_DIG_VOL_ZCD, 0x00c0, 0); } break; -- 2.7.4

1 0

[alsa-devel] [PATCH] sound: soc: rt5677-spi: Drop unused GPIO include
by Linus Walleij 18 Sep '18

18 Sep '18

This SPI driver does not use the legacy GPIO header so just delete it. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- sound/soc/codecs/rt5677-spi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index bd51f3655ee3..84501c2020c7 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -18,7 +18,6 @@ #include <linux/interrupt.h> #include <linux/irq.h> #include <linux/slab.h> -#include <linux/gpio.h> #include <linux/sched.h> #include <linux/uaccess.h> #include <linux/regulator/consumer.h> -- 2.17.1

2 2

[alsa-devel] [PATCH 1/2] sound: soc: wm8903: Pull in platform data
by Linus Walleij 18 Sep '18

18 Sep '18

No platform in the kernel is using platform data to populate the WM8903 so pull it into the source file. This is done to handle further simplifications. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- include/sound/wm8903.h | 24 ------------------------ sound/soc/codecs/wm8903.c | 24 ++++++++++++++++++++++++ 2 files changed, 24 insertions(+), 24 deletions(-) diff --git a/include/sound/wm8903.h b/include/sound/wm8903.h index b310c5a3a958..b2ef76d5d00d 100644 --- a/include/sound/wm8903.h +++ b/include/sound/wm8903.h @@ -11,12 +11,6 @@ #ifndef __LINUX_SND_WM8903_H #define __LINUX_SND_WM8903_H -/* - * Used to enable configuration of a GPIO to all zeros; a gpio_cfg value of - * zero in platform data means "don't touch this pin". - */ -#define WM8903_GPIO_CONFIG_ZERO 0x8000 - /* * R6 (0x06) - Mic Bias Control 0 */ @@ -245,22 +239,4 @@ #define WM8903_GP5_DB_SHIFT 0 /* GP5_DB */ #define WM8903_GP5_DB_WIDTH 1 /* GP5_DB */ -#define WM8903_NUM_GPIO 5 - -struct wm8903_platform_data { - bool irq_active_low; /* Set if IRQ active low, default high */ - - /* Default register value for R6 (Mic bias), used to configure - * microphone detection. In conjunction with gpio_cfg this - * can be used to route the microphone status signals out onto - * the GPIOs for use with snd_soc_jack_add_gpios(). - */ - u16 micdet_cfg; - - int micdet_delay; /* Delay after microphone detection (ms) */ - - int gpio_base; - u32 gpio_cfg[WM8903_NUM_GPIO]; /* Default register values for GPIO pin mux */ -}; - #endif diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 6cb3c153ba19..23e43ff40ded 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -124,6 +124,30 @@ static const char *wm8903_supply_names[WM8903_NUM_SUPPLIES] = { "DCVDD", }; +/* + * Used to enable configuration of a GPIO to all zeros; a gpio_cfg value of + * zero in platform data means "don't touch this pin". + */ +#define WM8903_GPIO_CONFIG_ZERO 0x8000 + +#define WM8903_NUM_GPIO 5 + +struct wm8903_platform_data { + bool irq_active_low; /* Set if IRQ active low, default high */ + + /* Default register value for R6 (Mic bias), used to configure + * microphone detection. In conjunction with gpio_cfg this + * can be used to route the microphone status signals out onto + * the GPIOs for use with snd_soc_jack_add_gpios(). + */ + u16 micdet_cfg; + + int micdet_delay; /* Delay after microphone detection (ms) */ + + int gpio_base; + u32 gpio_cfg[WM8903_NUM_GPIO]; /* Default register values for GPIO pin mux */ +}; + struct wm8903_priv { struct wm8903_platform_data *pdata; struct device *dev; -- 2.17.1

2 1

[alsa-devel] Applied "ASoC: remove redundant include" to the asoc tree
by Mark Brown 18 Sep '18

18 Sep '18

The patch ASoC: remove redundant include has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 29ca7d32d7f10737e8d165fcf40fe31d44b06bee Mon Sep 17 00:00:00 2001 From: zhong jiang <zhongjiang(a)huawei.com> Date: Tue, 18 Sep 2018 16:16:24 +0800 Subject: [PATCH] ASoC: remove redundant include module.h already contained moduleparam.h, so it is safe to remove the redundant include. The issue is detected with the help of Coccinelle. Signed-off-by: zhong jiang <zhongjiang(a)huawei.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5651.c | 1 - sound/soc/codecs/wm8904.c | 1 - sound/soc/codecs/wm8974.c | 1 - sound/soc/soc-dapm.c | 1 - 4 files changed, 4 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 985852fd9723..b613103d801b 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -10,7 +10,6 @@ */ #include <linux/module.h> -#include <linux/moduleparam.h> #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 1965635ec07c..2a3e5fbd04e4 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -13,7 +13,6 @@ #include <linux/clk.h> #include <linux/module.h> -#include <linux/moduleparam.h> #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 43edaf8cd276..593a11960888 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -11,7 +11,6 @@ */ #include <linux/module.h> -#include <linux/moduleparam.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 43983c69f6aa..ee6b9758ec15 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -18,7 +18,6 @@ // device reopen. #include <linux/module.h> -#include <linux/moduleparam.h> #include <linux/init.h> #include <linux/async.h> #include <linux/delay.h> -- 2.19.0

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[alsa-devel] [PATCH 2/5] ASoC: rt5682: Improve HP performance
by shumingf＠realtek.com 18 Sep '18

18 Sep '18

From: Shuming Fan <shumingf(a)realtek.com> We change the settings while HP power-up for better performance. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 32 +++++++++++++++++++++++++++++--- sound/soc/codecs/rt5682.h | 14 ++++++++++++++ 2 files changed, 43 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 31ec7c0..bf7d910 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1437,6 +1437,28 @@ static const struct snd_kcontrol_new hpor_switch = SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1, RT5682_R_MUTE_SFT, 1, 1); +static int rt5682_charge_pump_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_update_bits(component, + RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_HV); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(component, + RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_LV); + break; + default: + return 0; + } + + return 0; +} + static int rt5682_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1449,8 +1471,6 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, RT5682_HP_LOGIC_CTRL_2, 0x0012); snd_soc_component_write(component, RT5682_HP_CTRL_2, 0x6000); - snd_soc_component_update_bits(component, RT5682_STO_NG2_CTRL_1, - RT5682_NG2_EN_MASK, RT5682_NG2_EN); snd_soc_component_update_bits(component, RT5682_DEPOP_1, 0x60, 0x60); break; @@ -1723,7 +1743,8 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("HP Amp R", RT5682_PWR_ANLG_1, RT5682_PWR_HA_R_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Charge Pump", 1, RT5682_DEPOP_1, - RT5682_PUMP_EN_SFT, 0, NULL, 0), + RT5682_PUMP_EN_SFT, 0, rt5682_charge_pump_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5682_DEPOP_1, RT5682_CAPLESS_EN_SFT, 0, NULL, 0), @@ -1884,6 +1905,7 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"HP Amp", NULL, "Charge Pump"}, {"HP Amp", NULL, "CLKDET SYS"}, {"HP Amp", NULL, "CBJ Power"}, + {"HP Amp", NULL, "Vref1"}, {"HP Amp", NULL, "Vref2"}, {"HPOL Playback", "Switch", "HP Amp"}, {"HPOR Playback", "Switch", "HP Amp"}, @@ -2616,6 +2638,10 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, RT5682_GP4_PIN_MASK | RT5682_GP5_PIN_MASK, RT5682_GP4_PIN_ADCDAT1 | RT5682_GP5_PIN_DACDAT1); regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); + regmap_update_bits(rt5682->regmap, RT5682_BIAS_CUR_CTRL_8, + RT5682_HPA_CP_BIAS_CTRL_MASK, RT5682_HPA_CP_BIAS_3UA); + regmap_update_bits(rt5682->regmap, RT5682_CHARGE_PUMP_1, + RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ); INIT_DELAYED_WORK(&rt5682->jack_detect_work, rt5682_jack_detect_handler); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 8068140..d82a830 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -1214,6 +1214,20 @@ #define RT5682_JDH_NO_PLUG (0x1 << 4) #define RT5682_JDH_PLUG (0x0 << 4) +/* Bias current control 8 (0x0111) */ +#define RT5682_HPA_CP_BIAS_CTRL_MASK (0x3 << 2) +#define RT5682_HPA_CP_BIAS_2UA (0x0 << 2) +#define RT5682_HPA_CP_BIAS_3UA (0x1 << 2) +#define RT5682_HPA_CP_BIAS_4UA (0x2 << 2) +#define RT5682_HPA_CP_BIAS_6UA (0x3 << 2) + +/* Charge Pump Internal Register1 (0x0125) */ +#define RT5682_CP_CLK_HP_MASK (0x3 << 4) +#define RT5682_CP_CLK_HP_100KHZ (0x0 << 4) +#define RT5682_CP_CLK_HP_200KHZ (0x1 << 4) +#define RT5682_CP_CLK_HP_300KHZ (0x2 << 4) +#define RT5682_CP_CLK_HP_600KHZ (0x3 << 4) + /* Chopper and Clock control for DAC (0x013a)*/ #define RT5682_CKXEN_DAC1_MASK (0x1 << 13) #define RT5682_CKXEN_DAC1_SFT 13 -- 2.7.4

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[alsa-devel] [PATCH 3/5] ASoC: rt5682: Remove HP volume control
by shumingf＠realtek.com 18 Sep '18

18 Sep '18

From: Shuming Fan <shumingf(a)realtek.com> This patch removed Headphone Playback Volume control. Due to codec settings, we don't want the user to change HP analog gain. The user could use DAC1 Playback Volume control to change playback volume. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index bf7d910..2725eb7 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -749,7 +749,6 @@ static bool rt5682_readable_register(struct device *dev, unsigned int reg) } } -static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0); static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); @@ -1108,10 +1107,6 @@ static void rt5682_jack_detect_handler(struct work_struct *work) } static const struct snd_kcontrol_new rt5682_snd_controls[] = { - /* Headphone Output Volume */ - SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5682_HPL_GAIN, - RT5682_HPR_GAIN, RT5682_G_HP_SFT, 15, 1, hp_vol_tlv), - /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL, RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 86, 0, dac_vol_tlv), -- 2.7.4

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[alsa-devel] [PATCH 1/5] ASoC: rt5682: Update calibration function
by shumingf＠realtek.com 18 Sep '18

18 Sep '18

From: Shuming Fan <shumingf(a)realtek.com> The ADC/DAC path should open while calibration process. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 731c6a8..31ec7c0 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2451,17 +2451,22 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) mutex_lock(&rt5682->calibrate_mutex); rt5682_reset(rt5682->regmap); - regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2bf); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af); usleep_range(15000, 20000); - regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2bf); + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2af); regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0300); regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x8000); regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0100); + regmap_write(rt5682->regmap, RT5682_HP_IMP_SENS_CTRL_19, 0x3800); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000); + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005); + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c); + regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321); regmap_write(rt5682->regmap, RT5682_HP_LOGIC_CTRL_2, 0x0004); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_3, 0x06a1); + regmap_write(rt5682->regmap, RT5682_A_DAC1_MUX, 0x0311); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0xfc00); @@ -2478,8 +2483,12 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) pr_err("HP Calibration Failure\n"); /* restore settings */ + regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0x02af); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0080); regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x0000); regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); + regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000); + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); mutex_unlock(&rt5682->calibrate_mutex); -- 2.7.4

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[alsa-devel] [PATCH 4/5] ASoC: rt5682: Fix the boost volume at the begining of playback
by shumingf＠realtek.com 18 Sep '18

18 Sep '18

From: Shuming Fan <shumingf(a)realtek.com> This patch fixed the boost volume at the begining of playback while DAC volume set to lower level. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 2725eb7..1809966 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -68,6 +68,7 @@ struct rt5682_priv { static const struct reg_sequence patch_list[] = { {0x01c1, 0x1000}, + {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, }; static const struct reg_default rt5682_reg[] = { @@ -1468,6 +1469,8 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, RT5682_HP_CTRL_2, 0x6000); snd_soc_component_update_bits(component, RT5682_DEPOP_1, 0x60, 0x60); + snd_soc_component_update_bits(component, + RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0080); break; case SND_SOC_DAPM_POST_PMD: @@ -1475,6 +1478,8 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, RT5682_DEPOP_1, 0x60, 0x0); snd_soc_component_write(component, RT5682_HP_CTRL_2, 0x0000); + snd_soc_component_update_bits(component, + RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0000); break; default: -- 2.7.4

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[alsa-devel] [PATCH v2] ASoC: AMD: Fix capture unstable in beginning for some runs
by Akshu Agrawal 18 Sep '18

18 Sep '18

alsa_conformance_test -C hw:0,4 -p 1024 --debug would sometime show: TIME_DIFF(s) HW_LEVEL READ RATE 0.000095970 1024 1024 10670001.041992 0.042609555 1024 2048 24032.168372 0.021330364 1024 3072 48006.681930 0.021339559 1024 4096 47985.996337 The issue is that in dma pointer function we can have stale value of the register for current descriptor of channel. The register retains the number of the last descriptor that was transferred. Fix ensures that we report position, 0, till the one period worth of data is transferred. After one period of data, in handler of period completion interrupt we update the config and correct value of descriptor starts reflecting. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: fix overrun issue and missed case where bytescount = rtd->bytescount sound/soc/amd/acp-pcm-dma.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index b0e245c..0ac4b5b 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1035,16 +1035,22 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { period_bytes = frames_to_bytes(runtime, runtime->period_size); - dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr); - if (dscr == rtd->dma_dscr_idx_1) - pos = period_bytes; - else - pos = 0; bytescount = acp_get_byte_count(rtd); - if (bytescount > rtd->bytescount) + if (bytescount >= rtd->bytescount) bytescount -= rtd->bytescount; - delay = do_div(bytescount, period_bytes); - runtime->delay = bytes_to_frames(runtime, delay); + if (bytescount < period_bytes) { + pos = 0; + } else { + dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr); + if (dscr == rtd->dma_dscr_idx_1) + pos = period_bytes; + else + pos = 0; + } + if (bytescount > 0) { + delay = do_div(bytescount, period_bytes); + runtime->delay = bytes_to_frames(runtime, delay); + } } else { buffersize = frames_to_bytes(runtime, runtime->buffer_size); bytescount = acp_get_byte_count(rtd); -- 1.9.1

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[alsa-devel] [PATCH 5/5] ASoC: rt5682: Minor code modification
by shumingf＠realtek.com 18 Sep '18

18 Sep '18

From: Shuming Fan <shumingf(a)realtek.com> Minor code changes are: - improve the readability in patch list - add i2c remove function - regmap_register_patch changes to regmap_multi_reg_write Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 1809966..340f904 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -67,7 +67,7 @@ struct rt5682_priv { }; static const struct reg_sequence patch_list[] = { - {0x01c1, 0x1000}, + {RT5682_HP_IMP_SENS_CTRL_19, 0x1000}, {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, }; @@ -2584,7 +2584,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, rt5682_calibrate(rt5682); - ret = regmap_register_patch(rt5682->regmap, patch_list, + ret = regmap_multi_reg_write(rt5682->regmap, patch_list, ARRAY_SIZE(patch_list)); if (ret != 0) dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); @@ -2659,11 +2659,17 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, } - return devm_snd_soc_register_component(&i2c->dev, - &soc_component_dev_rt5682, + return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5682, rt5682_dai, ARRAY_SIZE(rt5682_dai)); } +static int rt5682_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_component(&i2c->dev); + + return 0; +} + static void rt5682_i2c_shutdown(struct i2c_client *client) { struct rt5682_priv *rt5682 = i2c_get_clientdata(client); @@ -2694,6 +2700,7 @@ static struct i2c_driver rt5682_i2c_driver = { .acpi_match_table = ACPI_PTR(rt5682_acpi_match), }, .probe = rt5682_i2c_probe, + .remove = rt5682_i2c_remove, .shutdown = rt5682_i2c_shutdown, .id_table = rt5682_i2c_id, }; -- 2.7.4

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[alsa-devel] [PATCH - hwmixvolume v2 7/7] hwmixvolume: mention the new dependencies in the README
by Emmanuel Gil Peyrot 18 Sep '18

18 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/README b/hwmixvolume/README index 9884ec2..56c7554 100644 --- a/hwmixvolume/README +++ b/hwmixvolume/README @@ -8,7 +8,8 @@ that use hardware mixing, i.e., those based on the following chips: * Yamaha DS-1 (YMF-724/740/744/754) (driver: snd-ymfpci) -This tool requires Python, pygtk, and alsa-pyton 1.0.22 or later. +This tool requires Python, PyGObject, GTK+ 3.0 or later, and alsa-python 1.0.22 +or later. It is recommended to use at least Linux kernel 2.6.32 or alsa-driver 1.0.22; otherwise, the name of the program that is using a stream cannot be shown. -- 2.19.0

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[alsa-devel] [PATCH - hwmixvolume v2 6/7] hwmixvolume: add my copyright
by Emmanuel Gil Peyrot 18 Sep '18

18 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 4804a9b..871c2c5 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -2,6 +2,7 @@ # hwmixvolume - ALSA hardware mixer volume control applet # Copyright (c) 2009-2010 Clemens Ladisch +# Copyright (c) 2018 Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> # # Permission to use, copy, modify, and/or distribute this software for any # purpose with or without fee is hereby granted, provided that the above -- 2.19.0

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[alsa-devel] [PATCH - hwmixvolume v2 5/7] hwmixvolume: switch the shebang to python
by Emmanuel Gil Peyrot 18 Sep '18

18 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> This signifies that this code is now compatible with both Python 2 and Python 3, as per PEP-0394[1]. [1] https://www.python.org/dev/peps/pep-0394/ Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 7f8ba8e..4804a9b 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -1,4 +1,4 @@ -#!/usr/bin/python2 +#!/usr/bin/env python # hwmixvolume - ALSA hardware mixer volume control applet # Copyright (c) 2009-2010 Clemens Ladisch -- 2.19.0

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[alsa-devel] [PATCH - hwmixvolume v2 4/7] hwmixvolume: use a with context to open files
by Emmanuel Gil Peyrot 18 Sep '18

18 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> This feature has been added in Python 2.5 and automatically closes an open file once the context exits. Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 8e0b6b8..7f8ba8e 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -138,26 +138,20 @@ class Stream: subdevice = info.index filename = "/proc/asound/card%d/pcm%dp/sub%d/status" % (card, device, subdevice) try: - f = open(filename, "r") + with open(filename, "r") as f: + for line in f: + if line[:9] == "owner_pid": + return int(line.split(':')[1].strip()) except IOError: return None - try: - for line in f.readlines(): - if line[:9] == "owner_pid": - return int(line.split(':')[1].strip()) - finally: - f.close() return None def get_pid_cmdline(self, pid): try: - f = open("/proc/%d/cmdline" % pid, "r") + with open("/proc/%d/cmdline" % pid, "r") as f: + cmdline = f.read() except IOError: return None - try: - cmdline = f.read() - finally: - f.close() return cmdline.replace('\x00', ' ').strip() class MixerWindow(Gtk.Window): -- 2.19.0

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[alsa-devel] [PATCH - hwmixvolume v2 2/7] hwmixvolume: replace PyGTK with gobject-introspection
by Emmanuel Gil Peyrot 18 Sep '18

18 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> This doesn’t work yet, we require Gtk 3.0 rather than 2.0 and the API changed quite a lot, so this is but a preparatory patch. This is done so that we can get rid of GTK+ 2 which has been EOL for many years already, and to add Python 3 support because Python 2 will very soon be EOL as well. Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 038bcb3..64d232c 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -15,7 +15,10 @@ # OTHER TORTIOUS ACTION, ARISING OUT OF OR IN CONNECTION WITH THE USE OR # PERFORMANCE OF THIS SOFTWARE. -import gobject, gtk +import gi +gi.require_version('GLib', '2.0') +gi.require_version('Gtk', '3.0') +from gi.repository import GLib, Gtk from pyalsa import alsacard, alsahcontrol INTF_PCM = alsahcontrol.interface_id['PCM'] @@ -57,16 +60,16 @@ class Stream: value = alsahcontrol.Value(self.element) value.read() values = value.get_tuple(TYPE_INTEGER, info.count) - self.label = gtk.Label(self.get_label(info)) + self.label = Gtk.Label(self.get_label(info)) self.label.set_single_line_mode(True) self.parent.scales_vbox.pack_start(self.label, expand=False) for i in range(info.count): - adj = gtk.Adjustment(value=values[i], + adj = Gtk.Adjustment(value=values[i], lower=info.min, upper=info.max, step_incr=1, page_incr=(info.max-info.min+1)/8) adj.connect('value-changed', self.update_ctl_from_scale, i) - scale = gtk.HScale(adj) + scale = Gtk.HScale(adj) scale.set_draw_value(False) self.parent.scales_vbox.pack_start(scale, expand=False) self.scales.append(scale) @@ -157,7 +160,7 @@ class Stream: f.close() return cmdline.replace('\x00', ' ').strip() -class MixerWindow(gtk.Window): +class MixerWindow(Gtk.Window): card_numbers = alsacard.card_list() current_card = -1 hcontrol = None @@ -167,21 +170,21 @@ class MixerWindow(gtk.Window): hctl_sources = [] def __init__(self): - gtk.Window.__init__(self) - self.connect('destroy', lambda w: gtk.main_quit()) + Gtk.Window.__init__(self) + self.connect('destroy', lambda w: Gtk.main_quit()) self.set_title("Hardware Mixer Volumes") - vbox = gtk.VBox() + vbox = Gtk.VBox() self.add(vbox) - hbox = gtk.HBox() + hbox = Gtk.HBox() vbox.pack_start(hbox, expand=False) - label = gtk.Label("_Sound Card: ") + label = Gtk.Label("_Sound Card: ") label.set_use_underline(True) hbox.pack_start(label, expand=False) - combo = gtk.combo_box_new_text() + combo = Gtk.combo_box_new_text() for i in self.card_numbers: str = "%d: %s" % (i, alsacard.card_get_name(i)) combo.append_text(str) @@ -191,23 +194,23 @@ class MixerWindow(gtk.Window): hbox.pack_start(combo) label.set_mnemonic_widget(combo) - self.lock_check = gtk.CheckButton(label="_Lock Channels") + self.lock_check = Gtk.CheckButton(label="_Lock Channels") self.lock_check.set_active(True) vbox.pack_start(self.lock_check, expand=False) - scrollwin = gtk.ScrolledWindow() - scrollwin.set_policy(hscrollbar_policy=gtk.POLICY_NEVER, vscrollbar_policy=gtk.POLICY_AUTOMATIC) - scrollwin.set_shadow_type(gtk.SHADOW_NONE) + scrollwin = Gtk.ScrolledWindow() + scrollwin.set_policy(hscrollbar_policy=Gtk.POLICY_NEVER, vscrollbar_policy=Gtk.POLICY_AUTOMATIC) + scrollwin.set_shadow_type(Gtk.SHADOW_NONE) vbox.pack_start(scrollwin) - self.scales_vbox = gtk.VBox() + self.scales_vbox = Gtk.VBox() scrollwin.add_with_viewport(self.scales_vbox) - label = gtk.Label() + label = Gtk.Label() label.set_single_line_mode(True) line_height = label.size_request()[1] label.destroy() - scale = gtk.HScale() + scale = Gtk.HScale() scale.set_draw_value(False) line_height += scale.size_request()[1] scale.destroy() @@ -223,7 +226,7 @@ class MixerWindow(gtk.Window): def change_card(self, cardnum): for s in self.hctl_sources: - gobject.source_remove(s) + GLib.source_remove(s) self.hctl_sources = [] self.hcontrol = self.open_hcontrol_for_card(cardnum) @@ -249,7 +252,7 @@ class MixerWindow(gtk.Window): self.streams.append(stream) for fd,condition in self.hcontrol.poll_fds: - self.hctl_sources.append(gobject.io_add_watch(fd, condition, self.hctl_io_callback)) + self.hctl_sources.append(GLib.io_add_watch(fd, condition, self.hctl_io_callback)) self.update_msg_label() @@ -267,7 +270,7 @@ class MixerWindow(gtk.Window): else: msg = "This card does not have stream controls." if not has_msg: - self.msg_label = gtk.Label(msg) + self.msg_label = Gtk.Label(msg) self.scales_vbox.pack_start(self.msg_label) self.scales_vbox.show_all() elif self.msg_label.get_text() != msg: @@ -280,9 +283,9 @@ class MixerWindow(gtk.Window): mode=alsahcontrol.open_mode['NONBLOCK']) except: # TODO: alsa error msg - dlg = gtk.MessageDialog(self, - gtk.DIALOG_MODAL | gtk.DIALOG_DESTROY_WITH_PARENT, - gtk.MESSAGE_ERROR, gtk.BUTTONS_OK, + dlg = Gtk.MessageDialog(self, + Gtk.DIALOG_MODAL | Gtk.DIALOG_DESTROY_WITH_PARENT, + Gtk.MESSAGE_ERROR, Gtk.BUTTONS_OK, "Cannot open sound card control device.") dlg.run() dlg.destroy() @@ -304,7 +307,7 @@ class MixerWindow(gtk.Window): def main(): MixerWindow() - gtk.main() + Gtk.main() main() -- 2.19.0

1 0

[alsa-devel] [PATCH] ASoC: es8328: Fix fall-through annotations
by Gustavo A. R. Silva 17 Sep '18

17 Sep '18

Replace "fallthru" with a proper "fall through" annotation. This fix is part of the ongoing efforts to enabling -Wimplicit-fallthrough Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/es8328.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 4b5827d..04a3aa7 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -566,14 +566,14 @@ static int es8328_set_sysclk(struct snd_soc_dai *codec_dai, break; case 22579200: mclkdiv2 = 1; - /* fallthru */ + /* fall through */ case 11289600: es8328->sysclk_constraints = &constraints_11289; es8328->mclk_ratios = ratios_11289; break; case 24576000: mclkdiv2 = 1; - /* fallthru */ + /* fall through */ case 12288000: es8328->sysclk_constraints = &constraints_12288; es8328->mclk_ratios = ratios_12288; -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: hisilicon: fix fall-through annotations
by Gustavo A. R. Silva 17 Sep '18

17 Sep '18

Replace "fallthru" with a proper "fall through" annotation. This fix is part of the ongoing efforts to enabling -Wimplicit-fallthrough Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/hisilicon/hi6210-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 53344a3..a69e5b11 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -269,13 +269,13 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_U16_LE: signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT; - /* fallthru */ + /* fall through */ case SNDRV_PCM_FORMAT_S16_LE: bits = HII2S_BITS_16; break; case SNDRV_PCM_FORMAT_U24_LE: signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT; - /* fallthru */ + /* fall through */ case SNDRV_PCM_FORMAT_S24_LE: bits = HII2S_BITS_24; break; -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: qdsp6: q6asm-dai: remove duplicated include from q6asm-dai.c" to the asoc tree
by Mark Brown 17 Sep '18

17 Sep '18

The patch ASoC: qdsp6: q6asm-dai: remove duplicated include from q6asm-dai.c has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 24b7a0aa1abec41b62c9a29a75e511d29f95033b Mon Sep 17 00:00:00 2001 From: YueHaibing <yuehaibing(a)huawei.com> Date: Fri, 14 Sep 2018 01:36:04 +0000 Subject: [PATCH] ASoC: qdsp6: q6asm-dai: remove duplicated include from q6asm-dai.c Remove duplicated include. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Acked-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm-dai.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index c75fab38905d..c3806d7037fc 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -8,7 +8,6 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <sound/soc.h> -#include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/pcm.h> #include <asm/dma.h> -- 2.19.0

1 0

[alsa-devel] [PATCH 0/2] ASoC: soc-core: adjust platform for new style
by Kuninori Morimoto 17 Sep '18

17 Sep '18

Hi Mark These are v2 of "ASoC: soc-core: use more generic method to find platform" but, renamed patch series naming. v1 patch set had bug under DPCM which can't probe sound card. The cause was "platform name" management and "component" finding loop. These patch set solved this issue. And then, patch set became reduced Now, ALSA SoC is using snd_soc_dai_link_component for platform instead of legacy style (= platform_name/platform_of_node/platform). OTOH, CPU/Codec are finding its DAI by using common snd_soc_find_dai() function which uses snd_soc_dai_link_component. Then, of course, we want to use same style for platform. These patch do it. Kuninori Morimoto (2): ASoC: soc-core: manage platform name under snd_soc_init_platform() ASoC: soc-core: add snd_soc_is_matching_component() sound/soc/soc-core.c | 73 +++++++++++++++++++++++++++------------------------- 1 file changed, 38 insertions(+), 35 deletions(-) -- 2.7.4

2 4

[alsa-devel] [PATCH] ASoC: rt5514-spi: Get the period_bytes in the copy work to make sure the value correctly
by Oder Chiou 17 Sep '18

17 Sep '18

The value of period_bytes will get the zero before the hw_params() is not run completely. Move the function snd_pcm_lib_period_bytes() to copy work, and make sure that is not zero. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514-spi.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 6478d10..4d46f45 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -91,6 +91,14 @@ static void rt5514_spi_copy_work(struct work_struct *work) runtime = rt5514_dsp->substream->runtime; period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); + if (!period_bytes) { + schedule_delayed_work(&rt5514_dsp->copy_work, 5); + goto done; + } + + if (rt5514_dsp->buf_size % period_bytes) + rt5514_dsp->buf_size = (rt5514_dsp->buf_size / period_bytes) * + period_bytes; if (rt5514_dsp->get_size >= rt5514_dsp->buf_size) { rt5514_spi_burst_read(RT5514_BUFFER_VOICE_WP, (u8 *)&buf, @@ -149,13 +157,11 @@ static void rt5514_spi_copy_work(struct work_struct *work) static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp) { - size_t period_bytes; u8 buf[8]; if (!rt5514_dsp->substream) return; - period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); rt5514_dsp->get_size = 0; /** @@ -183,10 +189,6 @@ static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp) rt5514_dsp->buf_size = rt5514_dsp->buf_limit - rt5514_dsp->buf_base; - if (rt5514_dsp->buf_size % period_bytes) - rt5514_dsp->buf_size = (rt5514_dsp->buf_size / period_bytes) * - period_bytes; - if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit && rt5514_dsp->buf_rp && rt5514_dsp->buf_size) schedule_delayed_work(&rt5514_dsp->copy_work, 0); -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: fireworks: fix memory leak of response buffer at error path
by Takashi Sakamoto 17 Sep '18

17 Sep '18

After allocating memory object for response buffer, ALSA fireworks driver has leak of the memory object at error path. This commit releases the object at the error path. Fixes: 7d3c1d5901aa('ALSA: fireworks: delayed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireworks/fireworks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 71a0613d3da0..f2d073365cf6 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -301,6 +301,8 @@ do_registration(struct work_struct *work) snd_efw_transaction_remove_instance(efw); snd_efw_stream_destroy_duplex(efw); snd_card_free(efw->card); + kfree(efw->resp_buf); + efw->resp_buf = NULL; dev_info(&efw->unit->device, "Sound card registration failed: %d\n", err); } -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: oxfw: fix memory leak of discovered stream formats at error path
by Takashi Sakamoto 17 Sep '18

17 Sep '18

After finishing discover of stream formats, ALSA OXFW driver has memory leak of allocated memory object at error path. This commit releases the memory object at the error path. Fixes: 6c29230e2a5f ('ALSA: oxfw: delayed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 1e5b2c802635..63b3e1ce986e 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -207,6 +207,7 @@ static int detect_quirks(struct snd_oxfw *oxfw) static void do_registration(struct work_struct *work) { struct snd_oxfw *oxfw = container_of(work, struct snd_oxfw, dwork.work); + int i; int err; if (oxfw->registered) @@ -269,6 +270,12 @@ static void do_registration(struct work_struct *work) snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); if (oxfw->has_output) snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream); + for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; ++i) { + kfree(oxfw->tx_stream_formats[i]); + oxfw->tx_stream_formats[i] = NULL; + kfree(oxfw->rx_stream_formats[i]); + oxfw->rx_stream_formats[i] = NULL; + } snd_card_free(oxfw->card); dev_info(&oxfw->unit->device, "Sound card registration failed: %d\n", err); -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: oxfw: fix memory leak for model-dependent data at error path
by Takashi Sakamoto 17 Sep '18

17 Sep '18

After allocating model-dependent data, ALSA OXFW driver has memory leak of the data at error path. This commit releases the data at the error path. Fixes: 6c29230e2a5f ('ALSA: oxfw: delayed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 63b3e1ce986e..9c8c527ebff9 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -277,6 +277,8 @@ static void do_registration(struct work_struct *work) oxfw->rx_stream_formats[i] = NULL; } snd_card_free(oxfw->card); + kfree(oxfw->spec); + oxfw->spec = NULL; dev_info(&oxfw->unit->device, "Sound card registration failed: %d\n", err); } -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: bebob: fix memory leak for M-Audio FW1814 and ProjectMix I/O at error path
by Takashi Sakamoto 17 Sep '18

17 Sep '18

After allocating model-dependent data for M-Audio FW1814 and ProjectMix I/O, ALSA bebob driver has memory leak at error path. This commit releases the allocated data at the error path. Fixes: 04a2c73c97eb('ALSA: bebob: delayed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob.c | 2 ++ sound/firewire/bebob/bebob_maudio.c | 4 ---- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 730ea91d9be8..93676354f87f 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -263,6 +263,8 @@ do_registration(struct work_struct *work) error: mutex_unlock(&devices_mutex); snd_bebob_stream_destroy_duplex(bebob); + kfree(bebob->maudio_special_quirk); + bebob->maudio_special_quirk = NULL; snd_card_free(bebob->card); dev_info(&bebob->unit->device, "Sound card registration failed: %d\n", err); diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c index bd55620c6a47..164f2126a676 100644 --- a/sound/firewire/bebob/bebob_maudio.c +++ b/sound/firewire/bebob/bebob_maudio.c @@ -290,10 +290,6 @@ snd_bebob_maudio_special_discover(struct snd_bebob *bebob, bool is1814) bebob->midi_output_ports = 2; } end: - if (err < 0) { - kfree(params); - bebob->maudio_special_quirk = NULL; - } mutex_unlock(&bebob->mutex); return err; } -- 2.17.1

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v4.19
by Mark Brown 17 Sep '18

17 Sep '18

The following changes since commit f2cf0ef7c0ce141bb38f315c34c56e6ef5667a27: ASoC: adav80x: mark expected switch fall-through (2018-08-09 14:41:50 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.19-rc4 for you to fetch changes up to 75383f8d39d4c0fb96083dd460b7b139fbdac492: sound: don't call skl_init_chip() to reset intel skl soc (2018-09-12 11:22:09 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.19 This is the usual set of small fixes scatterd around various drivers, plus one fix for DAPM and a UAPI build fix. There's not a huge amount that stands out here relative to anything else. ---------------------------------------------------------------- Akshu Agrawal (1): ASoC: AMD: Ensure reset bit is cleared before configuring Andrew F. Davis (1): ASoC: tas6424: Save last fault register even when clear Charles Keepax (1): ASoC: dapm: Fix NULL pointer deference on CODEC to CODEC DAIs Danny Smith (1): ASoC: sigmadsp: safeload should not have lower byte limit Dmitry V. Levin (1): ASoC: uapi: fix sound/skl-tplg-interface.h userspace compilation errors Hans de Goede (1): ASoC: Intel: bytcr_rt5640: Add quirks for 2 more devices Jiada Wang (1): ASoC: rsnd: fixup not to call clk_get/set under non-atomic Katsuhiro Suzuki (1): ASoC: uniphier: change status to orphan Kuninori Morimoto (2): ASoC: rsnd: adg: care clock-frequency size ASoC: rsnd: don't fallback to PIO mode when -EPROBE_DEFER Marcel Ziswiler (1): ASoC: wm9712: fix replace codec to component Oder Chiou (1): ASoC: rt5514: Fix the issue of the delay volume applied again Pierre-Louis Bossart (1): ASoC: wm8804: Add ACPI support Ryan Lee (2): ASoC: max98373: Added speaker FS gain cotnrol register to volatile. ASoC: max98373: Added 10ms sleep after amp software reset Shuming Fan (1): ASoC: rt5682: Change DAC/ADC volume scale Srinivas Kandagatla (1): ASoC: q6routing: initialize data correctly Sébastien Szymanski (1): ASoC: cs4265: fix MMTLR Data switch control Yong Zhi (1): ASoC: Intel: Skylake: Acquire irq after RIRB allocation Yu Zhao (3): Revert "ASoC: Intel: Skylake: Acquire irq after RIRB allocation" sound: enable interrupt after dma buffer initialization sound: don't call skl_init_chip() to reset intel skl soc MAINTAINERS | 3 +- include/sound/hdaudio.h | 1 + include/sound/soc-dapm.h | 1 + include/uapi/sound/skl-tplg-interface.h | 106 ++++++++++++++++---------------- sound/hda/hdac_controller.c | 15 +++-- sound/soc/amd/acp-pcm-dma.c | 21 +++++++ sound/soc/codecs/cs4265.c | 4 +- sound/soc/codecs/max98373.c | 3 + sound/soc/codecs/rt5514.c | 8 +-- sound/soc/codecs/rt5682.c | 8 +-- sound/soc/codecs/sigmadsp.c | 3 +- sound/soc/codecs/tas6424.c | 12 +++- sound/soc/codecs/wm8804-i2c.c | 15 ++++- sound/soc/codecs/wm9712.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 26 ++++++++ sound/soc/intel/skylake/skl.c | 2 +- sound/soc/qcom/qdsp6/q6routing.c | 4 +- sound/soc/sh/rcar/adg.c | 5 ++ sound/soc/sh/rcar/core.c | 21 ++++++- sound/soc/sh/rcar/dma.c | 4 ++ sound/soc/sh/rcar/rsnd.h | 7 +++ sound/soc/sh/rcar/ssi.c | 16 +++-- sound/soc/soc-core.c | 4 +- sound/soc/soc-dapm.c | 4 ++ 24 files changed, 208 insertions(+), 87 deletions(-)

2 1

[alsa-devel] Bug#908938: alsa-utils: amixer does not unmute pulseaudio
by Elimar Riesebieter 17 Sep '18

17 Sep '18

Control: forwarded -1 alsa-devel(a)alsa-project.org Hi alsa devs, please notice Debian Bug https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=908938 and let us know whether a fix is possible. Thanks in advance -- Elimar * Łukasz Stelmach <steelman(a)post.pl> [2018-09-16 11:23 +0200]: > Package: alsa-utils > Version: 1.1.3-1 > Severity: normal > > Dear Maintainer, > > I've got PulseAudio running, so amixer (and alsamixer) controls the > default PA output. I use the following command to control muting of > audio. > > amixer -q -c 0 sset Master toggle > > It mutes the audio but can't unmute. Neither can > > amixer -q -c 0 sset Master unmute > > Only alsamixer and pavucontrol can unmute. -- Do you smell something burning or is it me?

3 2

[alsa-devel] [PATCH] Documentation: soundwire: fix stream.rst markup warnings
by Randy Dunlap 17 Sep '18

17 Sep '18

From: Randy Dunlap <rdunlap(a)infradead.org> Fix kernel-doc markup warnings in soundwire/stream.rst: rc4/Documentation/driver-api/soundwire/stream.rst:177: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:203: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:248: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:277: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:304: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:328: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:352: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:364: WARNING: Explicit markup ends without a blank line; unexpected unindent. Fixes: 89634f99a83e ("Documentation: soundwire: Add more documentation") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Sanyog Kale <sanyog.r.kale(a)intel.com> Cc: Shreyas NC <shreyas.nc(a)intel.com> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: linux-doc(a)vger.kernel.org --- Documentation/driver-api/soundwire/stream.rst | 8 ++++++++ 1 file changed, 8 insertions(+) --- lnx-419-rc4.orig/Documentation/driver-api/soundwire/stream.rst +++ lnx-419-rc4/Documentation/driver-api/soundwire/stream.rst @@ -174,6 +174,7 @@ per stream. From ASoC DPCM framework, th .startup() operation. .. code-block:: c + int sdw_alloc_stream(char * stream_name); @@ -200,6 +201,7 @@ only be invoked once by respective Maste framework, this stream state is linked to .hw_params() operation. .. code-block:: c + int sdw_stream_add_master(struct sdw_bus * bus, struct sdw_stream_config * stream_config, struct sdw_ports_config * ports_config, @@ -245,6 +247,7 @@ stream. From ASoC DPCM framework, this s .prepare() operation. .. code-block:: c + int sdw_prepare_stream(struct sdw_stream_runtime * stream); @@ -274,6 +277,7 @@ stream. From ASoC DPCM framework, this s .trigger() start operation. .. code-block:: c + int sdw_enable_stream(struct sdw_stream_runtime * stream); SDW_STREAM_DISABLED @@ -301,6 +305,7 @@ per stream. From ASoC DPCM framework, th .trigger() stop operation. .. code-block:: c + int sdw_disable_stream(struct sdw_stream_runtime * stream); @@ -325,6 +330,7 @@ per stream. From ASoC DPCM framework, th .trigger() stop operation. .. code-block:: c + int sdw_deprepare_stream(struct sdw_stream_runtime * stream); @@ -349,6 +355,7 @@ all the Master(s) and Slave(s) associate framework, this stream state is linked to .hw_free() operation. .. code-block:: c + int sdw_stream_remove_master(struct sdw_bus * bus, struct sdw_stream_runtime * stream); int sdw_stream_remove_slave(struct sdw_slave * slave, @@ -361,6 +368,7 @@ stream assigned as part of ALLOCATED sta In .shutdown() the data structure maintaining stream state are freed up. .. code-block:: c + void sdw_release_stream(struct sdw_stream_runtime * stream); Not Supported

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[alsa-devel] [PATCH] ASoC: AMD: Fix capture unstable in beginning for some runs
by Akshu Agrawal 17 Sep '18

17 Sep '18

alsa_conformance_test -C hw:0,4 -p 1024 --debug would sometime show: TIME_DIFF(s) HW_LEVEL READ RATE 0.000095970 1024 1024 10670001.041992 0.042609555 1024 2048 24032.168372 0.021330364 1024 3072 48006.681930 0.021339559 1024 4096 47985.996337 The issue is that in dma pointer function we can have stale value of the register for current descriptor of channel. The register retains the number of the last descriptor that was transferred. Fix ensures that we report position, 0, till the one period worth of data is transferred. After one period of data, in handler of period completion interrupt we update the config and correct value of descriptor starts reflecting. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index c7e972b..f27e20c 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1050,6 +1050,8 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) bytescount -= rtd->bytescount; delay = do_div(bytescount, period_bytes); runtime->delay = bytes_to_frames(runtime, delay); + if (bytescount < period_bytes) + pos = 0; } else { buffersize = frames_to_bytes(runtime, runtime->buffer_size); bytescount = acp_get_byte_count(rtd); -- 1.9.1

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[alsa-devel] [PATCH v3 00/13] ASoC: Add support to WCD9335 Audio Codec
by Srinivas Kandagatla 17 Sep '18

17 Sep '18

Thankyou for reviewing v2 patchset, here is v3 addressing comments from v2. Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC. It is integrated in multiple Qualcomm SoCs like: MSM8996, MSM8976, and MSM8956 chipsets. WCD9335 had multiple functional blocks, like: Soundwire controller, interrupt mux, pin controller, Audio codec, MBHC, MAD(Mic activity Detection), Ultrasonic proximity and pen detection, Battery-voltage monitoring and Codec processing engine. Currently this patchset has been only tested with SLIMbus interface due to hardware avaiablity, but it can be easily made to work with both SLIMbus and I2C/I2S. This patchset adds very basic playback and capture support witout much fancy features. New features will be added once the basic support is in. This patchset is tested on top of linux-next on DB820c for both playback , capture paths and MBHC. Some parts of the code has been inherited from Qualcomm andriod kernel, so credits to various authors. WCD9335 can be interfaced via I2S/I2C or SLIMbus. Here is my test branch incase someone want to try this out: https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=wcd9335 Thanks, Srini Changes since v2 (https://lwn.net/Articles/761107/) - Updated dt bindings according to Robs Comments. - Added MBHC support with new bindings - Moved mfd irq in to single file as suggested by Lee Jones. - Addressed various minor comments on mfd driver by Lee Jones. - Rebased on top of 4.19-rc1 - renamed clsh as clsh-v2 as its v2 block Srinivas Kandagatla (13): ASoC: dt-bindings: update wcd9335 bindings. mfd: wcd9335: add support to wcd9335 core mfd: wcd9335: add wcd irq support ASoC: wcd9335: add support to wcd9335 codec ASoC: wcd9335: add CLASS-H Controller support ASoC: wcd9335: add basic controls ASoC: wcd9335: add playback dapm widgets ASoC: wcd9335: add capture dapm widgets ASoC: wcd9335: add audio routings ASoC: dt-bindings: Add WCD9335 MBHC specific properties ASoC: wcd9335: add mbhc support ASoC: apq8096: add slim support ASoC: apq8096: add headset JACK support .../bindings/sound/qcom,wcd9335.txt | 30 +- drivers/mfd/Kconfig | 14 + drivers/mfd/Makefile | 4 + drivers/mfd/wcd9335-core.c | 369 ++ include/linux/mfd/wcd9335/registers.h | 627 ++ include/linux/mfd/wcd9335/wcd9335.h | 65 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-clsh-v2.c | 577 ++ sound/soc/codecs/wcd-clsh-v2.h | 49 + sound/soc/codecs/wcd9335.c | 5234 +++++++++++++++++ sound/soc/qcom/apq8096.c | 122 +- 12 files changed, 7092 insertions(+), 6 deletions(-) create mode 100644 drivers/mfd/wcd9335-core.c create mode 100644 include/linux/mfd/wcd9335/registers.h create mode 100644 include/linux/mfd/wcd9335/wcd9335.h create mode 100644 sound/soc/codecs/wcd-clsh-v2.c create mode 100644 sound/soc/codecs/wcd-clsh-v2.h create mode 100644 sound/soc/codecs/wcd9335.c -- 2.18.0

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