From ae1fbdff6dbcdfee9daee69fa1e7d26d1f31d1c7 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Fri, 30 Jun 2017 17:17:35 -0500 Subject: [PATCH 01/18] ASoC: imx-ssi: add check on platform_get_irq return value Check return value from call to platform_get_irq(), so in case of failure print error message and propagate the return value. Signed-off-by: Gustavo A. R. Silva Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/imx-ssi.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index b95132e2f9dc..06790615e04e 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -527,6 +527,10 @@ static int imx_ssi_probe(struct platform_device *pdev) } ssi->irq = platform_get_irq(pdev, 0); + if (ssi->irq < 0) { + dev_err(&pdev->dev, "Failed to get IRQ: %d\n", ssi->irq); + return ssi->irq; + } ssi->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(ssi->clk)) { From 9f8f5b5f6c515e0c9d9bc14996fa8b9414c5ce1a Mon Sep 17 00:00:00 2001 From: "oder_chiou@realtek.com" Date: Mon, 10 Jul 2017 11:14:57 +0800 Subject: [PATCH 02/18] ASoC: rt5663: Update the HW default values based on the shipping version The patch update the HW default values based on the shipping version. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5663.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index a33202affeb1..fa550e3c1332 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -466,7 +466,7 @@ static const struct reg_default rt5663_reg[] = { { 0x0006, 0x1000 }, { 0x000a, 0x0000 }, { 0x0010, 0x000f }, - { 0x0015, 0x42c1 }, + { 0x0015, 0x42f1 }, { 0x0016, 0x0000 }, { 0x0018, 0x000b }, { 0x0019, 0xafaf }, @@ -509,7 +509,7 @@ static const struct reg_default rt5663_reg[] = { { 0x008a, 0x0000 }, { 0x008b, 0x0000 }, { 0x008c, 0x0003 }, - { 0x008e, 0x0004 }, + { 0x008e, 0x0008 }, { 0x008f, 0x1000 }, { 0x0090, 0x0646 }, { 0x0091, 0x0e3e }, @@ -520,7 +520,7 @@ static const struct reg_default rt5663_reg[] = { { 0x0098, 0x0000 }, { 0x009a, 0x0000 }, { 0x009f, 0x0000 }, - { 0x00ae, 0x2000 }, + { 0x00ae, 0x6000 }, { 0x00af, 0x0000 }, { 0x00b6, 0x0000 }, { 0x00b7, 0x0000 }, @@ -538,7 +538,7 @@ static const struct reg_default rt5663_reg[] = { { 0x00d9, 0x08f9 }, { 0x00db, 0x0008 }, { 0x00dc, 0x00c0 }, - { 0x00dd, 0x6724 }, + { 0x00dd, 0x6729 }, { 0x00de, 0x3131 }, { 0x00df, 0x0008 }, { 0x00e0, 0x4000 }, @@ -578,7 +578,7 @@ static const struct reg_default rt5663_reg[] = { { 0x0116, 0x0000 }, { 0x0117, 0x0f00 }, { 0x0118, 0x0006 }, - { 0x0125, 0x2224 }, + { 0x0125, 0x2424 }, { 0x0126, 0x5550 }, { 0x0127, 0x0400 }, { 0x0128, 0x7711 }, @@ -596,8 +596,8 @@ static const struct reg_default rt5663_reg[] = { { 0x0145, 0x0002 }, { 0x0146, 0x0000 }, { 0x0160, 0x0e80 }, - { 0x0161, 0x0020 }, - { 0x0162, 0x0080 }, + { 0x0161, 0x0080 }, + { 0x0162, 0x0200 }, { 0x0163, 0x0800 }, { 0x0164, 0x0000 }, { 0x0165, 0x0000 }, @@ -676,8 +676,8 @@ static const struct reg_default rt5663_reg[] = { { 0x0251, 0x0000 }, { 0x0252, 0x028a }, { 0x02fa, 0x0000 }, - { 0x02fb, 0x0000 }, - { 0x02fc, 0x0000 }, + { 0x02fb, 0x00a4 }, + { 0x02fc, 0x0300 }, { 0x0300, 0x0000 }, { 0x03d0, 0x0000 }, { 0x03d1, 0x0000 }, From 5b43af6d25461d7de293e0704d3b4631dda9b1e8 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Fri, 30 Jun 2017 18:38:44 +0530 Subject: [PATCH 03/18] ASoC: Intel: Skylake: Fix default dma_buffer_size If the dma_buffer_size is not defined in topology, fix it to 2ms default value to make backward compatible. Fixes: f6e6ab1d16ec ("ASoC: Intel: Skylake: Fix dma buffer size calculation") Signed-off-by: Subhransu S. Prusty Acked-By: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-messages.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index eca85827dbd2..fb2f1f603f3c 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -540,6 +540,14 @@ static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx, cpr_mconfig->gtw_cfg.dma_buffer_size = mconfig->dma_buffer_size * dma_io_buf; + /* fallback to 2ms default value */ + if (!cpr_mconfig->gtw_cfg.dma_buffer_size) { + if (mconfig->hw_conn_type == SKL_CONN_SOURCE) + cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * mconfig->obs; + else + cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * mconfig->ibs; + } + cpr_mconfig->cpr_feature_mask = 0; cpr_mconfig->gtw_cfg.config_length = 0; From ac1ca3ba9faae7e32f189edda14f6f147053d719 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Sun, 9 Jul 2017 21:30:32 +0200 Subject: [PATCH 04/18] ASoC: pxa: SND_PXA2XX_SOC should depend on HAS_DMA If NO_DMA=y: ERROR: "bad_dma_ops" [sound/soc/pxa/snd-soc-pxa2xx.ko] undefined! ERROR: "bad_dma_ops" [sound/arm/snd-pxa2xx-lib.ko] undefined! ERROR: "dma_common_mmap" [sound/arm/snd-pxa2xx-lib.ko] undefined! Add a dependency on HAS_DMA to fix this. Fixes: 73d7ee2e831f106c ("ASoC: pxa: add COMPILE_TEST on SND_PXA2XX_SOC") Signed-off-by: Geert Uytterhoeven Acked-by: Robert Jarzmik Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 960744e46edc..484ab3c2ad67 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,6 +1,7 @@ config SND_PXA2XX_SOC tristate "SoC Audio for the Intel PXA2xx chip" depends on ARCH_PXA || COMPILE_TEST + depends on HAS_DMA select SND_PXA2XX_LIB help Say Y or M if you want to add support for codecs attached to From b1cd2e34c69a2f3988786af451b6e17967c293a0 Mon Sep 17 00:00:00 2001 From: Banajit Goswami Date: Fri, 14 Jul 2017 23:15:05 -0700 Subject: [PATCH 05/18] ASoC: do not close shared backend dailink Multiple frontend dailinks may be connected to a backend dailink at the same time. When one of frontend dailinks is closed, the associated backend dailink should not be closed if it is connected to other active frontend dailinks. Change ensures that backend dailink is closed only after all connected frontend dailinks are closed. Signed-off-by: Gopikrishnaiah Anandan Signed-off-by: Banajit Goswami Signed-off-by: Patrick Lai Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcc5ece08668..93999b8a87d3 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -181,6 +181,10 @@ int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, dev_dbg(be->dev, "ASoC: BE %s event %d dir %d\n", be->dai_link->name, event, dir); + if ((event == SND_SOC_DAPM_STREAM_STOP) && + (be->dpcm[dir].users >= 1)) + continue; + snd_soc_dapm_stream_event(be, dir, event); } From c641e5b207ed7dfaa692820aeb5b6dde3de3e9b0 Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Wed, 12 Jul 2017 17:55:29 +0200 Subject: [PATCH 06/18] ASoC: fix pcm-creation regression This reverts commit 99b04f4c4051 ("ASoC: add Component level pcm_new/pcm_free"), which started calling the pcm_new callback for every component in a *card* when creating a new pcm, something which does not seem to make any sense. This specifically led to memory leaks in systems with more than one platform component and where DMA memory is allocated in the platform-driver callback. For example, when both mcasp devices are being used on an am335x board, DMA memory would be allocated twice for every DAI link during probe. When CONFIG_SND_VERBOSE_PROCFS was set this fortunately also led to warnings such as: WARNING: CPU: 0 PID: 565 at ../fs/proc/generic.c:346 proc_register+0x110/0x154 proc_dir_entry 'sub0/prealloc' already registered Since there seems to be no users of the new component callbacks, and the current implementation introduced a regression, let's revert the offending commit for now. Fixes: 99b04f4c4051 ("ASoC: add Component level pcm_new/pcm_free") Signed-off-by: Johan Hovold Reviewed-by: Linus Walleij Tested-by: Linus Walleij Signed-off-by: Mark Brown Cc: stable # 4.10 --- include/sound/soc.h | 6 ------ sound/soc/soc-core.c | 25 ------------------------- sound/soc/soc-pcm.c | 32 +++++++++----------------------- 3 files changed, 9 insertions(+), 54 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9c94b97c17f8..c4a8b1947566 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -795,10 +795,6 @@ struct snd_soc_component_driver { int (*suspend)(struct snd_soc_component *); int (*resume)(struct snd_soc_component *); - /* pcm creation and destruction */ - int (*pcm_new)(struct snd_soc_pcm_runtime *); - void (*pcm_free)(struct snd_pcm *); - /* DT */ int (*of_xlate_dai_name)(struct snd_soc_component *component, struct of_phandle_args *args, @@ -874,8 +870,6 @@ struct snd_soc_component { void (*remove)(struct snd_soc_component *); int (*suspend)(struct snd_soc_component *); int (*resume)(struct snd_soc_component *); - int (*pcm_new)(struct snd_soc_pcm_runtime *); - void (*pcm_free)(struct snd_pcm *); /* machine specific init */ int (*init)(struct snd_soc_component *component); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 921622a01944..c240e13ba057 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3171,8 +3171,6 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->remove = component->driver->remove; component->suspend = component->driver->suspend; component->resume = component->driver->resume; - component->pcm_new = component->driver->pcm_new; - component->pcm_free = component->driver->pcm_free; dapm = &component->dapm; dapm->dev = dev; @@ -3360,25 +3358,6 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component) platform->driver->remove(platform); } -static int snd_soc_platform_drv_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_platform *platform = rtd->platform; - - if (platform->driver->pcm_new) - return platform->driver->pcm_new(rtd); - else - return 0; -} - -static void snd_soc_platform_drv_pcm_free(struct snd_pcm *pcm) -{ - struct snd_soc_pcm_runtime *rtd = pcm->private_data; - struct snd_soc_platform *platform = rtd->platform; - - if (platform->driver->pcm_free) - platform->driver->pcm_free(pcm); -} - /** * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform @@ -3402,10 +3381,6 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->component.probe = snd_soc_platform_drv_probe; if (platform_drv->remove) platform->component.remove = snd_soc_platform_drv_remove; - if (platform_drv->pcm_new) - platform->component.pcm_new = snd_soc_platform_drv_pcm_new; - if (platform_drv->pcm_free) - platform->component.pcm_free = snd_soc_platform_drv_pcm_free; #ifdef CONFIG_DEBUG_FS platform->component.debugfs_prefix = "platform"; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcc5ece08668..553f7a76743c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2628,25 +2628,12 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) return ret; } -static void soc_pcm_free(struct snd_pcm *pcm) -{ - struct snd_soc_pcm_runtime *rtd = pcm->private_data; - struct snd_soc_component *component; - - list_for_each_entry(component, &rtd->card->component_dev_list, - card_list) { - if (component->pcm_free) - component->pcm_free(pcm); - } -} - /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_component *component; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; @@ -2756,18 +2743,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (capture) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &rtd->ops); - list_for_each_entry(component, &rtd->card->component_dev_list, card_list) { - if (component->pcm_new) { - ret = component->pcm_new(rtd); - if (ret < 0) { - dev_err(component->dev, - "ASoC: pcm constructor failed: %d\n", - ret); - return ret; - } + if (platform->driver->pcm_new) { + ret = platform->driver->pcm_new(rtd); + if (ret < 0) { + dev_err(platform->dev, + "ASoC: pcm constructor failed: %d\n", + ret); + return ret; } } - pcm->private_free = soc_pcm_free; + + pcm->private_free = platform->driver->pcm_free; out: dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name, From 651e9268fb9b9944e063d731b09c0d2ad339bedb Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Wed, 12 Jul 2017 17:55:30 +0200 Subject: [PATCH 07/18] ASoC: ux500: Restore platform DAI assignments This reverts commit f1013cdeeeb9 ("ASoC: ux500: drop platform DAI assignments"), which seems to have been based on a misunderstanding and prevents the platform driver callbacks from being made (e.g. to preallocate DMA memory). The real culprit for the warnings about attempts to create duplicate procfs entries was commit 99b04f4c4051 ("ASoC: add Component level pcm_new/pcm_free" that broke PCM creation on systems that use more than one platform component. Fixes: f1013cdeeeb9 ("ASoC: ux500: drop platform DAI assignments") Signed-off-by: Johan Hovold Reviewed-by: Linus Walleij Tested-by: Linus Walleij Signed-off-by: Mark Brown Cc: stable # 4.11 --- sound/soc/ux500/mop500.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index b50f68a439ce..ba9fc099cf67 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -33,6 +33,7 @@ static struct snd_soc_dai_link mop500_dai_links[] = { .stream_name = "ab8500_0", .cpu_dai_name = "ux500-msp-i2s.1", .codec_dai_name = "ab8500-codec-dai.0", + .platform_name = "ux500-msp-i2s.1", .codec_name = "ab8500-codec.0", .init = mop500_ab8500_machine_init, .ops = mop500_ab8500_ops, @@ -42,6 +43,7 @@ static struct snd_soc_dai_link mop500_dai_links[] = { .stream_name = "ab8500_1", .cpu_dai_name = "ux500-msp-i2s.3", .codec_dai_name = "ab8500-codec-dai.1", + .platform_name = "ux500-msp-i2s.3", .codec_name = "ab8500-codec.0", .init = NULL, .ops = mop500_ab8500_ops, @@ -85,6 +87,8 @@ static int mop500_of_probe(struct platform_device *pdev, for (i = 0; i < 2; i++) { mop500_dai_links[i].cpu_of_node = msp_np[i]; mop500_dai_links[i].cpu_dai_name = NULL; + mop500_dai_links[i].platform_of_node = msp_np[i]; + mop500_dai_links[i].platform_name = NULL; mop500_dai_links[i].codec_of_node = codec_np; mop500_dai_links[i].codec_name = NULL; } From b50e2842b25fc14299ccf98dc9467b6304082bcb Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 20 Jul 2017 13:07:49 +0800 Subject: [PATCH 08/18] ASoC: rt5665: fix GPIO6 pin function define The GPIO6 pin function select value was wrong. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5665.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5665.h b/sound/soc/codecs/rt5665.h index 1db5c6a62a8e..d95249c4c47b 100644 --- a/sound/soc/codecs/rt5665.h +++ b/sound/soc/codecs/rt5665.h @@ -1692,8 +1692,8 @@ #define RT5665_GP6_PIN_MASK (0x3 << 5) #define RT5665_GP6_PIN_SFT 5 #define RT5665_GP6_PIN_GPIO6 (0x0 << 5) -#define RT5665_GP6_PIN_BCLK3 (0x0 << 5) -#define RT5665_GP6_PIN_PDM_SCL (0x1 << 5) +#define RT5665_GP6_PIN_BCLK3 (0x1 << 5) +#define RT5665_GP6_PIN_PDM_SCL (0x2 << 5) #define RT5665_GP7_PIN_MASK (0x3 << 3) #define RT5665_GP7_PIN_SFT 3 #define RT5665_GP7_PIN_GPIO7 (0x0 << 3) From 364e93ca5dd6f4d266c3a5ff169961d2caac19fb Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 20 Jul 2017 14:36:17 -0300 Subject: [PATCH 09/18] ASoC: sgtl5000: Use snd_soc_kcontrol_codec() Since commit a72952672005 ("ASoC: sgtl5000: add avc support") the following kernel crash happens after running a 'reboot' command: ALSA: Storing mixer settings... [ 20.031604] Unable to handle kernel paging request at virtual address fffffffe [ 20.039268] pgd = de2a0000 [ 20.041999] [fffffffe] *pgd=8fffd861, *pte=00000000, *ppte=00000000 [ 20.048387] Internal error: Oops: 80000007 [#1] SMP ARM The function that takes a kcontrol parameter and returns the codec that registered the control is snd_soc_kcontrol_codec(), so use the correct function to fix the problem. Fixes: a72952672005 ("ASoC: sgtl5000: add avc support") Signed-off-by: Fabio Estevam Tested-by: Richard Leitner Reviewed-by: Richard Leitner Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 8f6814c1eb6b..80f6d1da7095 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -409,7 +409,7 @@ static int dac_put_volsw(struct snd_kcontrol *kcontrol, static int avc_get_threshold(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); int db, i; u16 reg = snd_soc_read(codec, SGTL5000_DAP_AVC_THRESHOLD); @@ -442,7 +442,7 @@ static int avc_get_threshold(struct snd_kcontrol *kcontrol, static int avc_put_threshold(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); int db; u16 reg; From 9d154e42a338a4142e7a656d662ebf98c4ceb26b Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 21 Jul 2017 18:29:20 +0200 Subject: [PATCH 10/18] ASoC: samsung: odroid: Fix EPLL frequency values To prevent incorrect setting of the EPLL the clock frequency values are changed to exact values as possible to obtain on the EPLL output with given PLL coefficients. This patch is required after recent change of the EPLL rate table by patch "clk: samsung: exynos5420: The EPLL rate table corrections". Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/odroid.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 0c0b00e40646..0834319ead42 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -42,17 +42,17 @@ static int odroid_card_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 32000: case 64000: - pll_freq = 131072000U; + pll_freq = 131072006U; break; case 44100: case 88200: case 176400: - pll_freq = 180633600U; + pll_freq = 180633609U; break; case 48000: case 96000: case 192000: - pll_freq = 196608000U; + pll_freq = 196608001U; break; default: return -EINVAL; From 295c5ba4c0886b7d55e229218b077fe3510b0ccd Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 24 Jul 2017 06:47:59 +0000 Subject: [PATCH 11/18] ASoC: sh: hac: add missing "int ret" commit b047e1cce8 ("ASoC: ac97: Support multi-platform AC'97") modified hac_soc_platform_probe(), but "int ret" was missed. This patch adds missing "int ret", otherwise, we will get linux/sound/soc/sh/hac.c: In function 'hac_soc_platform_probe': linux/sound/soc/sh/hac.c:318: error: 'ret' undeclared (first use in this function) linux/sound/soc/sh/hac.c:318: error: (Each undeclared identifier is reported only once linux/sound/soc/sh/hac.c:318: error: for each function it appears in.) Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/hac.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index 84c51037a7d0..624aaf569fef 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -315,6 +315,8 @@ static const struct snd_soc_component_driver sh4_hac_component = { static int hac_soc_platform_probe(struct platform_device *pdev) { + int ret; + ret = snd_soc_set_ac97_ops(&hac_ac97_ops); if (ret != 0) return ret; From b76e3f933327f9fd9df9a65a2d239e6e350cbee2 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 25 Jul 2017 12:08:12 +0530 Subject: [PATCH 12/18] ASoC: Intel: Skylake: Fix missing sentinels in sst_acpi_mach Couple of instances of sst_acpi_mach were having missing sentinels so add them up Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 334917ee41cf..9e3f8c04dd32 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -941,6 +941,7 @@ static struct sst_acpi_mach sst_bxtp_devdata[] = { .machine_quirk = sst_acpi_codec_list, .quirk_data = &bxt_codecs, }, + {} }; static struct sst_acpi_mach sst_kbl_devdata[] = { @@ -991,6 +992,7 @@ static struct sst_acpi_mach sst_glk_devdata[] = { .drv_name = "glk_alc298s_i2s", .fw_filename = "intel/dsp_fw_glk.bin", }, + {} }; /* PCI IDs */ From deab4563ad6a7f4668024455fa61b87f1d25ff73 Mon Sep 17 00:00:00 2001 From: Damien Riegel Date: Tue, 25 Jul 2017 13:51:24 -0400 Subject: [PATCH 13/18] ASoC: codecs: msm8916-analog: fix DIG_CLK_CTL_RXD3_CLK_EN define The wrong bit is assigned to DIG_CLK_CTL_RXD3_CLK_EN, change it for the correct one. Signed-off-by: Damien Riegel Acked-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/codecs/msm8916-wcd-analog.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index a78802920c3c..5710fd440bcd 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -36,7 +36,7 @@ #define CDC_D_CDC_DIG_CLK_CTL (0xf04A) #define DIG_CLK_CTL_RXD1_CLK_EN BIT(0) #define DIG_CLK_CTL_RXD2_CLK_EN BIT(1) -#define DIG_CLK_CTL_RXD3_CLK_EN BIT(3) +#define DIG_CLK_CTL_RXD3_CLK_EN BIT(2) #define DIG_CLK_CTL_TXD_CLK_EN BIT(4) #define DIG_CLK_CTL_NCP_CLK_EN_MASK BIT(6) #define DIG_CLK_CTL_NCP_CLK_EN BIT(6) From 349d63c33a34e39b205cf116d2406e096a029f8b Mon Sep 17 00:00:00 2001 From: Harsha Priya N Date: Thu, 27 Jul 2017 17:41:25 -0700 Subject: [PATCH 14/18] ASoC: Intel: Enabling ASRC for RT5663 codec on kabylake platform This patch fixes the cracking noise in rt5663 headphones for kabylake platform by calling rt5663_sel_asrc_clk_src() for RT5663_AD_STEREO_FILTER to set ASRC. The ASRC function is for asynchronous MCLK and LRCK. For RT5663 ASRC should be enabled to support special i2s clock format like Intel's 100fs. ASRC function will track i2s clock and generate a corresponding system clock for codec. Calling this function helps select the clock source for both RT5663_AD_STEREO_FILTER and RT5663_DA_STEREO_FILTER filters which fixes the crackling sound. Signed-off-by: Harsha Priya Acked-By: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 3fe4a0807095..cfde894d250f 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -319,7 +319,9 @@ static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream, int ret; /* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */ - rt5663_sel_asrc_clk_src(codec_dai->codec, RT5663_DA_STEREO_FILTER, 1); + rt5663_sel_asrc_clk_src(codec_dai->codec, + RT5663_DA_STEREO_FILTER | RT5663_AD_STEREO_FILTER, + RT5663_CLK_SEL_I2S1_ASRC); ret = snd_soc_dai_set_sysclk(codec_dai, RT5663_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); From 6512dd4dcf640412637ece8a412e83c3a0046d2f Mon Sep 17 00:00:00 2001 From: Harsha Priya N Date: Thu, 27 Jul 2017 17:41:26 -0700 Subject: [PATCH 15/18] ASoC: Intel: Use MCLK instead of BLCK as the sysclock for RT5514 codec on kabylake platform This patch fixes the pop noise in dmic recording using rt5514 on kabylake platform. This patch enables the rt5514 to use MCLK instead of BLCK as the sysclock which fixes the pop noise. Signed-off-by: Harsha Priya Acked-By: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index cfde894d250f..cfd89ca6a18d 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -351,19 +351,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, return ret; } - ret = snd_soc_dai_set_pll(codec_dai, 0, - RT5514_PLL1_S_BCLK, RT5514_AIF1_BCLK_FREQ, - RT5514_AIF1_SYSCLK_FREQ); - if (ret < 0) { - dev_err(rtd->dev, "set bclk err: %d\n", ret); - return ret; - } - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5514_SCLK_S_PLL1, RT5514_AIF1_SYSCLK_FREQ, - SND_SOC_CLOCK_IN); + RT5514_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); if (ret < 0) { - dev_err(rtd->dev, "set sclk err: %d\n", ret); + dev_err(rtd->dev, "set sysclk err: %d\n", ret); return ret; } } From 7e5824c93412c7fd6503e7769f8b6eb7199cd3b8 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 1 Aug 2017 10:30:53 +0800 Subject: [PATCH 16/18] ASoC: rt5665: fix wrong register for bclk ratio control The register of setting back ratio should be RT5665_ADDA_CLK_2 instead of RT5665_ADDA_CLK_1. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5665.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 370ed54d1e15..e597c893536d 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4368,12 +4368,12 @@ static int rt5665_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) switch (dai->id) { case RT5665_AIF2_1: case RT5665_AIF2_2: - snd_soc_update_bits(codec, RT5665_ADDA_CLK_1, + snd_soc_update_bits(codec, RT5665_ADDA_CLK_2, RT5665_I2S_BCLK_MS2_MASK, RT5665_I2S_BCLK_MS2_64); break; case RT5665_AIF3: - snd_soc_update_bits(codec, RT5665_ADDA_CLK_1, + snd_soc_update_bits(codec, RT5665_ADDA_CLK_2, RT5665_I2S_BCLK_MS3_MASK, RT5665_I2S_BCLK_MS3_64); break; From c0a480d1acf7dc184f9f3e7cf724483b0d28dc2e Mon Sep 17 00:00:00 2001 From: Tony Lindgren Date: Fri, 28 Jul 2017 01:23:15 -0700 Subject: [PATCH 17/18] device property: Fix usecount for of_graph_get_port_parent() Fix inconsistent use of of_graph_get_port_parent() where asoc_simple_card_parse_graph_dai() does of_node_get() before calling it while other callers do not. We can fix this by not trashing the node passed to of_graph_get_port_parent(). Let's also make sure the callers have correct refcounts and remove related incorrect of_node_put() calls for of_for_each_phandle as that's done by of_phandle_iterator_next() except when we break out of the loop early. Let's fix both issues with a single patch to avoid kobject refcounts getting messed up more if two patches are merged separately. Otherwise strange issues can happen caused by memory corruption caused by too many kobject_del() calls such as: BUG: sleeping function called from invalid context at kernel/locking/mutex.c:747 ... (___might_sleep) (__mutex_lock) (mutex_lock_nested) (kernfs_remove) (kobject_del) (kobject_put) (of_get_next_parent) (of_graph_get_port_parent) (asoc_simple_card_parse_graph_dai [snd_soc_simple_card_utils]) (asoc_graph_card_probe [snd_soc_audio_graph_card]) Fixes: 0ef472a973eb ("of_graph: add of_graph_get_port_parent()") Fixes: 2692c1c63c29 ("ASoC: add audio-graph-card support") Fixes: 1689333f8311 ("ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai()") Signed-off-by: Tony Lindgren Reviewed-by: Rob Herring Tested-by: Antonio Borneo Tested-by: Kuninori Morimoto Signed-off-by: Mark Brown --- drivers/of/property.c | 17 +++++++++++++++-- sound/soc/generic/audio-graph-card.c | 10 +++++----- sound/soc/generic/audio-graph-scu-card.c | 15 +++++++++------ sound/soc/generic/simple-card-utils.c | 8 +++----- sound/soc/soc-core.c | 2 ++ 5 files changed, 34 insertions(+), 18 deletions(-) diff --git a/drivers/of/property.c b/drivers/of/property.c index eda50b4be934..067f9fab7b77 100644 --- a/drivers/of/property.c +++ b/drivers/of/property.c @@ -708,6 +708,15 @@ struct device_node *of_graph_get_port_parent(struct device_node *node) { unsigned int depth; + if (!node) + return NULL; + + /* + * Preserve usecount for passed in node as of_get_next_parent() + * will do of_node_put() on it. + */ + of_node_get(node); + /* Walk 3 levels up only if there is 'ports' node. */ for (depth = 3; depth && node; depth--) { node = of_get_next_parent(node); @@ -728,12 +737,16 @@ EXPORT_SYMBOL(of_graph_get_port_parent); struct device_node *of_graph_get_remote_port_parent( const struct device_node *node) { - struct device_node *np; + struct device_node *np, *pp; /* Get remote endpoint node. */ np = of_graph_get_remote_endpoint(node); - return of_graph_get_port_parent(np); + pp = of_graph_get_port_parent(np); + + of_node_put(np); + + return pp; } EXPORT_SYMBOL(of_graph_get_remote_port_parent); diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 105ec3a6e30d..de2550c7a96b 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -224,9 +224,11 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { ret = asoc_graph_card_dai_link_of(it.node, priv, idx++); - of_node_put(it.node); - if (ret < 0) + if (ret < 0) { + of_node_put(it.node); + return ret; + } } return asoc_simple_card_parse_card_name(card, NULL); @@ -239,10 +241,8 @@ static int asoc_graph_get_dais_count(struct device *dev) int count = 0; int rc; - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { + of_for_each_phandle(&it, rc, node, "dais", NULL, 0) count++; - of_node_put(it.node); - } return count; } diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index dcd2df37bc3b..758ac06f3a99 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -215,7 +215,6 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) codec_ep = of_graph_get_remote_endpoint(cpu_ep); rcpu_ep = of_graph_get_remote_endpoint(codec_ep); - of_node_put(cpu_port); of_node_put(cpu_ep); of_node_put(codec_ep); of_node_put(rcpu_ep); @@ -232,8 +231,10 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, NULL, &daifmt); - if (ret < 0) + if (ret < 0) { + of_node_put(cpu_port); goto parse_of_err; + } } dai_idx = 0; @@ -250,7 +251,6 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) codec_ep = of_graph_get_remote_endpoint(cpu_ep); codec_port = of_graph_get_port_parent(codec_ep); - of_node_put(cpu_port); of_node_put(cpu_ep); of_node_put(codec_ep); of_node_put(codec_port); @@ -266,13 +266,17 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) /* Back-End (= Codec) */ ret = asoc_graph_card_dai_link_of(codec_ep, priv, daifmt, dai_idx++, 0); - if (ret < 0) + if (ret < 0) { + of_node_put(cpu_port); goto parse_of_err; + } } else { /* Front-End (= CPU) */ ret = asoc_graph_card_dai_link_of(cpu_ep, priv, daifmt, dai_idx++, 1); - if (ret < 0) + if (ret < 0) { + of_node_put(cpu_port); goto parse_of_err; + } } } } @@ -306,7 +310,6 @@ static int asoc_graph_get_dais_count(struct device *dev) codec_ep = of_graph_get_remote_endpoint(cpu_ep); codec_port = of_graph_get_port_parent(codec_ep); - of_node_put(cpu_port); of_node_put(cpu_ep); of_node_put(codec_ep); of_node_put(codec_port); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 26d64fa40c9c..7d7ab4aee42e 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -263,6 +263,9 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep) id = i; i++; } + + of_node_put(node); + if (id < 0) return -ENODEV; @@ -282,11 +285,6 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep, if (!dai_name) return 0; - /* - * of_graph_get_port_parent() will call - * of_node_put(). So, call of_node_get() here - */ - of_node_get(ep); node = of_graph_get_port_parent(ep); /* Get dai->name */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 921622a01944..0cf8498fa36c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4113,6 +4113,8 @@ int snd_soc_get_dai_id(struct device_node *ep) } mutex_unlock(&client_mutex); + of_node_put(node); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_get_dai_id); From 3f3c371421e601fa93b6cb7fb52da9ad59ec90b4 Mon Sep 17 00:00:00 2001 From: "Sergei A. Trusov" Date: Wed, 2 Aug 2017 20:23:48 +1000 Subject: [PATCH 18/18] ALSA: hda - Fix speaker output from VAIO VPCL14M1R Sony VAIO VPCL14M1R needs the quirk to make the speaker working properly. Tested-by: Dmitriy Cc: Signed-off-by: Sergei A. Trusov Signed-off-by: Takashi Iwai --- 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 443a45eaec32..a91a9ef00c40 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2296,6 +2296,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x8691, "ASUS ROG Ranger VIII", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), + SND_PCI_QUIRK(0x104d, 0x9060, "Sony Vaio VPCL14M1R", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9044, "Sony VAIO AiO", ALC882_FIXUP_NO_PRIMARY_HP),