From 72cedf599fcebfd6cd2550274d7855838068d28c Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Tue, 21 Feb 2017 23:00:46 +0100 Subject: [PATCH 01/31] ASoC: mediatek: add I2C dependency for CS42XX8 We should not select drivers that depend on I2C when that is disabled, as it results in a build error: warning: (SND_SOC_MT2701_CS42448) selects SND_SOC_CS42XX8_I2C which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && I2C) sound/soc/codecs/cs42xx8-i2c.c:60:1: warning: data definition has no type or storage class module_i2c_driver(cs42xx8_i2c_driver); sound/soc/codecs/cs42xx8-i2c.c:60:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] Fixes: 1f458d53f76c ("ASoC: mediatek: Add mt2701-cs42448 driver and config option.") Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/mediatek/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 05cf809cf9e1..d7013bde6f45 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -13,7 +13,7 @@ config SND_SOC_MT2701 config SND_SOC_MT2701_CS42448 tristate "ASoc Audio driver for MT2701 with CS42448 codec" - depends on SND_SOC_MT2701 + depends on SND_SOC_MT2701 && I2C select SND_SOC_CS42XX8_I2C select SND_SOC_BT_SCO help From 4b30eebfc35c67771b5f58d9274d3e321b72d7a8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 3 Mar 2017 04:25:09 +0000 Subject: [PATCH 02/31] ASoC: rcar: avoid SSI_MODEx settings for SSI8 SSI8 is is sharing pin with SSI7, and nothing to do for SSI_MODEx. It is special pin and it needs special settings whole system, but we can't confirm it, because we never have SSI8 available board. This patch fixup SSI_MODEx settings error for SSI8 on connection test, but should be confirmed behavior on real board in the future. Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssiu.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 4e817c8a18c0..14fafdaf1395 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -64,7 +64,11 @@ static int rsnd_ssiu_init(struct rsnd_mod *mod, mask1 = (1 << 4) | (1 << 20); /* mask sync bit */ mask2 = (1 << 4); /* mask sync bit */ val1 = val2 = 0; - if (rsnd_ssi_is_pin_sharing(io)) { + if (id == 8) { + /* + * SSI8 pin is sharing with SSI7, nothing to do. + */ + } else if (rsnd_ssi_is_pin_sharing(io)) { int shift = -1; switch (id) { From 04c8f2bf9117de7b8e8bc0b90e8c4bff15f4f613 Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Wed, 1 Mar 2017 22:41:23 +0530 Subject: [PATCH 03/31] ASoC: hdac_hdmi: avoid reference to invalid variable of the pin list Using pin list array iterator outside the iteration of the list can point to dummy element, which can be invalid. So don't use pin variable outside the pin list iteration. This fixes the following coccinelle warning: sound/soc/codecs/hdac_hdmi.c:1419:5-8: ERROR: invalid reference to the index variable of the iterator Fixes: 2acd8309a3a4('ASoC: hdac_hdmi: Add support to handle MST capable pin') Reported-by: Julia Lawall Signed-off-by: Jeeja KP Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/codecs/hdac_hdmi.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 78fca8acd3ec..bb405698e102 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1534,21 +1534,20 @@ static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) pin->mst_capable = false; /* if not MST, default is port[0] */ hport = &pin->ports[0]; - goto out; } else { for (i = 0; i < pin->num_ports; i++) { pin->mst_capable = true; if (pin->ports[i].id == pipe) { hport = &pin->ports[i]; - goto out; + break; } } } + + if (hport) + hdac_hdmi_present_sense(pin, hport); } -out: - if (pin && hport) - hdac_hdmi_present_sense(pin, hport); } static struct i915_audio_component_audio_ops aops = { From 2fe42dd0f13812d38daaf05bcb1fd996afd0e87a Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Wed, 1 Mar 2017 22:41:24 +0530 Subject: [PATCH 04/31] ASoC: hdac_hdmi: don't update the iterator in pcm list remove Fix not to update the iterator element, instead use list_del to remove entry from the list. This fixes the following coccinelle and static checker warning: sound/soc/codecs/hdac_hdmi.c:1884:2-21:iterator with update on line 1885 sound/soc/codecs/hdac_hdmi.c:2011 hdac_hdmi_dev_remove() error: potential NULL dereference 'port'. Fixes: e0e5d3e5a53b('ASoC: hdac_hdmi: Add support for multiple ports to a PCM') Reported-by: Julia Lawall Reported-by: Dan Carpenter Signed-off-by: Jeeja KP Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/codecs/hdac_hdmi.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index bb405698e102..fd272a40485b 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1997,7 +1997,7 @@ static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev) struct hdac_hdmi_pin *pin, *pin_next; struct hdac_hdmi_cvt *cvt, *cvt_next; struct hdac_hdmi_pcm *pcm, *pcm_next; - struct hdac_hdmi_port *port; + struct hdac_hdmi_port *port, *port_next; int i; snd_soc_unregister_codec(&edev->hdac.dev); @@ -2007,8 +2007,9 @@ static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev) if (list_empty(&pcm->port_list)) continue; - list_for_each_entry(port, &pcm->port_list, head) - port = NULL; + list_for_each_entry_safe(port, port_next, + &pcm->port_list, head) + list_del(&port->head); list_del(&pcm->head); kfree(pcm); From 67430a39ca7a6af28aade5acb92d43ee257c1014 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 6 Mar 2017 16:54:33 +0000 Subject: [PATCH 05/31] ASoC: wm_adsp: Return an error on write to a disabled volatile control Volatile controls should only be accessed when the firmware is active, currently however writes to these controls will succeed, but the data will be lost, if the firmware is powered down. Update this behaviour such that an error is returned the same as it is for reads. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d151224ffcca..6313b3da967b 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -899,7 +899,10 @@ static int wm_coeff_put(struct snd_kcontrol *kctl, mutex_lock(&ctl->dsp->pwr_lock); - memcpy(ctl->cache, p, ctl->len); + if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) + ret = -EPERM; + else + memcpy(ctl->cache, p, ctl->len); ctl->set = 1; if (ctl->enabled && ctl->dsp->running) @@ -926,6 +929,8 @@ static int wm_coeff_tlv_put(struct snd_kcontrol *kctl, ctl->set = 1; if (ctl->enabled && ctl->dsp->running) ret = wm_coeff_write_control(ctl, ctl->cache, size); + else if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) + ret = -EPERM; } mutex_unlock(&ctl->dsp->pwr_lock); From 7b4af793a7a4f8e04175eb6600ba9c8ba855ad20 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 6 Mar 2017 16:54:34 +0000 Subject: [PATCH 06/31] ASoC: wm_adsp: Acknowledge controls should also check the DSP is running We should not be writing acknowledge controls until the firmware is running, as in the case of preloaded firmwares the DSP memory may be unaccessible to whilst in the preloaded state. This means a write to the control during this time could be lost. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- 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 6313b3da967b..bbdb72f73df1 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -952,7 +952,7 @@ static int wm_coeff_put_acked(struct snd_kcontrol *kctl, mutex_lock(&ctl->dsp->pwr_lock); - if (ctl->enabled) + if (ctl->enabled && ctl->dsp->running) ret = wm_coeff_write_acked_control(ctl, val); else ret = -EPERM; From d1a6fe41d3c4ff0d26f0b186d774493555ca5282 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 24 Feb 2017 11:48:41 +0900 Subject: [PATCH 07/31] ASoC: Intel: Skylake: fix invalid memory access due to wrong reference of pointer In 'skl_tplg_set_module_init_data()', a pointer to 'params' member of 'struct skl_algo_data' is calculated, then casted to (u32 *) and assigned to a member of configuration data. The configuration data is passed to the other functions and used to process intel IPC. In this processing, the value of member is used to get message data, however this can bring invalid memory access in 'skl_set_module_params()' as a result of calculation of a pointer for actual message data. (sound/soc/intel/skylake/skl-topology.c) skl_tplg_init_pipe_modules() ->skl_tplg_set_module_init_data() (has this bug) ->skl_tplg_set_module_params() (sound/soc/intel/skylake/skl-messages.c) ->skl_set_module_params() ((char *)param) + data_offset This commit fixes the bug. Fixes: abb740033b56 ("ASoC: Intel: Skylake: Add support to configure module params") Signed-off-by: Takashi Sakamoto Acked-by: Vinod Koul Signed-off-by: Mark Brown Cc: # v4.5+ --- sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index ed58b5b3555a..2dbfb1b24ef4 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -512,7 +512,7 @@ static int skl_tplg_set_module_init_data(struct snd_soc_dapm_widget *w) if (bc->set_params != SKL_PARAM_INIT) continue; - mconfig->formats_config.caps = (u32 *)&bc->params; + mconfig->formats_config.caps = (u32 *)bc->params; mconfig->formats_config.caps_size = bc->size; break; From cd3ac9affc43b44f49d7af70d275f0bd426ba643 Mon Sep 17 00:00:00 2001 From: Songjun Wu Date: Fri, 24 Feb 2017 15:10:43 +0800 Subject: [PATCH 08/31] ASoC: atmel-classd: fix audio clock rate Fix the audio clock rate according to the datasheet. Reported-by: Dushara Jayasinghe Signed-off-by: Songjun Wu Acked-by: Nicolas Ferre Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/atmel/atmel-classd.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index 89ac5f5a93eb..7ae46c2647d4 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -349,7 +349,7 @@ static int atmel_classd_codec_dai_digital_mute(struct snd_soc_dai *codec_dai, } #define CLASSD_ACLK_RATE_11M2896_MPY_8 (112896 * 100 * 8) -#define CLASSD_ACLK_RATE_12M288_MPY_8 (12228 * 1000 * 8) +#define CLASSD_ACLK_RATE_12M288_MPY_8 (12288 * 1000 * 8) static struct { int rate; From f1994a9c0930de4b2244816e62120cad08283cdc Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 8 Mar 2017 19:03:10 +0800 Subject: [PATCH 09/31] ASoC: rt5665: fix getting wrong work handler container We got rt5665 private data from wrong work. It will result in kernel panic. Signed-off-by: Bard Liao Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/rt5665.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 324461e985b3..fe2cf1ed8237 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -1241,7 +1241,7 @@ static irqreturn_t rt5665_irq(int irq, void *data) static void rt5665_jd_check_handler(struct work_struct *work) { struct rt5665_priv *rt5665 = container_of(work, struct rt5665_priv, - calibrate_work.work); + jd_check_work.work); if (snd_soc_read(rt5665->codec, RT5665_AJD1_CTRL) & 0x0010) { /* jack out */ From 593dd5d9fb66c0345cef5fc8caf4199bb6d4e093 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 8 Mar 2017 19:05:29 +0800 Subject: [PATCH 10/31] ASoC: rt5665: increase LDO level Too low LDO level will cause a few functions unstable. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5665.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index fe2cf1ed8237..a4c07c2ee414 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4798,7 +4798,7 @@ static int rt5665_i2c_probe(struct i2c_client *i2c, /* Enhance performance*/ regmap_update_bits(rt5665->regmap, RT5665_PWR_ANLG_1, RT5665_HP_DRIVER_MASK | RT5665_LDO1_DVO_MASK, - RT5665_HP_DRIVER_5X | RT5665_LDO1_DVO_09); + RT5665_HP_DRIVER_5X | RT5665_LDO1_DVO_12); INIT_DELAYED_WORK(&rt5665->jack_detect_work, rt5665_jack_detect_handler); From 8f365313beb20742b68cb29a7d1ca6b27c036d81 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 8 Mar 2017 19:05:31 +0800 Subject: [PATCH 11/31] ASoC: rt5665: Vref3 is necessary for Mono Amp Vref3 is necessary for Mono Amp. So add it to dapm routes Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5665.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index a4c07c2ee414..7a0244e5e321 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -3912,6 +3912,7 @@ static const struct snd_soc_dapm_route rt5665_dapm_routes[] = { {"Mono MIX", "MONOVOL Switch", "MONOVOL"}, {"Mono Amp", NULL, "Mono MIX"}, {"Mono Amp", NULL, "Vref2"}, + {"Mono Amp", NULL, "Vref3"}, {"Mono Amp", NULL, "CLKDET SYS"}, {"Mono Amp", NULL, "CLKDET MONO"}, {"Mono Playback", "Switch", "Mono Amp"}, From 09b50c3703cc354100fe36202ef5e52ee128b904 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 8 Mar 2017 19:05:32 +0800 Subject: [PATCH 12/31] ASoC: rt5665: CLKDET is also a power of ASRC We need to power on CLKDET to use ASRC function. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5665.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 7a0244e5e321..61137160c116 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -3178,6 +3178,9 @@ static const struct snd_soc_dapm_route rt5665_dapm_routes[] = { {"DAC Mono Right Filter", NULL, "DAC Mono R ASRC", is_using_asrc}, {"DAC Stereo1 Filter", NULL, "DAC STO1 ASRC", is_using_asrc}, {"DAC Stereo2 Filter", NULL, "DAC STO2 ASRC", is_using_asrc}, + {"I2S1 ASRC", NULL, "CLKDET"}, + {"I2S2 ASRC", NULL, "CLKDET"}, + {"I2S3 ASRC", NULL, "CLKDET"}, /*Vref*/ {"Mic Det Power", NULL, "Vref2"}, From d6c098a1db468b7fd4635e831f276851dfd8852c Mon Sep 17 00:00:00 2001 From: Brian Norris Date: Wed, 8 Mar 2017 15:18:54 -0800 Subject: [PATCH 13/31] ASoC: don't dereference NULL pcm_{new,free} Not all platform drivers have pcm_{new,free} callbacks. Seen with a "snd-soc-dummy" codec from sound/soc/rockchip/rk3399_gru_sound.c. Fixes: 99b04f4c4051 ("ASoC: add Component level pcm_new/pcm_free") Signed-off-by: Brian Norris Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6dca408faae3..2722bb0c5573 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3326,7 +3326,10 @@ static int snd_soc_platform_drv_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_platform *platform = rtd->platform; - return platform->driver->pcm_new(rtd); + 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) @@ -3334,7 +3337,8 @@ 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; - platform->driver->pcm_free(pcm); + if (platform->driver->pcm_free) + platform->driver->pcm_free(pcm); } /** From a1c2ff53726907aff5feb37e4cfd45c1ff626431 Mon Sep 17 00:00:00 2001 From: Hiroyuki Yokoyama Date: Wed, 1 Mar 2017 03:51:00 +0000 Subject: [PATCH 14/31] ASoC: rsnd: fix sound route path when using SRC6/SRC9 This patch fixes the problem that the missing value of the route path setting table and incorrect values are set in the CMD_ROUTE_SELECT register. Signed-off-by: Hiroyuki Yokoyama [Kuninori: shared data on MIX and non-MIX case] Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/cmd.c | 36 ++++++++++++++++++++---------------- 1 file changed, 20 insertions(+), 16 deletions(-) diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index abb5eaac854a..7d92a24b7cfa 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -31,23 +31,24 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, struct rsnd_mod *mix = rsnd_io_to_mod_mix(io); struct device *dev = rsnd_priv_to_dev(priv); u32 data; + u32 path[] = { + [1] = 1 << 0, + [5] = 1 << 8, + [6] = 1 << 12, + [9] = 1 << 15, + }; if (!mix && !dvc) return 0; + if (ARRAY_SIZE(path) < rsnd_mod_id(mod) + 1) + return -ENXIO; + if (mix) { struct rsnd_dai *rdai; struct rsnd_mod *src; struct rsnd_dai_stream *tio; int i; - u32 path[] = { - [0] = 0, - [1] = 1 << 0, - [2] = 0, - [3] = 0, - [4] = 0, - [5] = 1 << 8 - }; /* * it is assuming that integrater is well understanding about @@ -70,16 +71,19 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, } else { struct rsnd_mod *src = rsnd_io_to_mod_src(io); - u32 path[] = { - [0] = 0x30000, - [1] = 0x30001, - [2] = 0x40000, - [3] = 0x10000, - [4] = 0x20000, - [5] = 0x40100 + u8 cmd_case[] = { + [0] = 0x3, + [1] = 0x3, + [2] = 0x4, + [3] = 0x1, + [4] = 0x2, + [5] = 0x4, + [6] = 0x1, + [9] = 0x2, }; - data = path[rsnd_mod_id(src)]; + data = path[rsnd_mod_id(src)] | + cmd_case[rsnd_mod_id(src)] << 16; } dev_dbg(dev, "ctu/mix path = 0x%08x", data); From 62a10498afb27370ec6018e9d802b74850fd8d9a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 14 Mar 2017 09:34:49 +0900 Subject: [PATCH 15/31] ASoC: rcar: clear DE bit only in PDMACHCR when it stops R-Car datasheet indicates "Clear DE in PDMACHCR" for transfer stop, but current code clears all bits in PDMACHCR. Because of this, DE bit might never been cleared, and it causes CMD overflow. This patch fixes this issue. Signed-off-by: Kuninori Morimoto Tested-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dma.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 1f405c833867..c2e199b4fcf4 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -454,6 +454,20 @@ static u32 rsnd_dmapp_read(struct rsnd_dma *dma, u32 reg) return ioread32(rsnd_dmapp_addr(dmac, dma, reg)); } +static void rsnd_dmapp_bset(struct rsnd_dma *dma, u32 data, u32 mask, u32 reg) +{ + struct rsnd_mod *mod = rsnd_mod_get(dma); + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); + volatile void __iomem *addr = rsnd_dmapp_addr(dmac, dma, reg); + u32 val = ioread32(addr); + + val &= ~mask; + val |= (data & mask); + + iowrite32(val, addr); +} + static int rsnd_dmapp_stop(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) @@ -461,10 +475,10 @@ static int rsnd_dmapp_stop(struct rsnd_mod *mod, struct rsnd_dma *dma = rsnd_mod_to_dma(mod); int i; - rsnd_dmapp_write(dma, 0, PDMACHCR); + rsnd_dmapp_bset(dma, 0, PDMACHCR_DE, PDMACHCR); for (i = 0; i < 1024; i++) { - if (0 == rsnd_dmapp_read(dma, PDMACHCR)) + if (0 == (rsnd_dmapp_read(dma, PDMACHCR) & PDMACHCR_DE)) return 0; udelay(1); } From 9986943ef5d61a9bea3c86000d91d3b789f0060e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 16 Mar 2017 04:22:09 +0000 Subject: [PATCH 16/31] ASoC: rcar: dma: remove unnecessary "volatile" commit 2a3af642eb20("ASoC: rcar: clear DE bit only in PDMACHCR...") added rsnd_dmapp_bset(), but it used copy-paste. Thus, it had unnecessary "volatile", and had below warning on x86. This patch fix it. sound/soc/sh/rcar/dma.c: In function 'rsnd_dmapp_bset': >> sound/soc/sh/rcar/dma.c:463:21: warning: passing argument 1 of \ 'ioread32' discards 'volatile' qualifier from pointer target \ type [-Wdiscarded-qualifiers] u32 val = ioread32(addr); ^~~~ In file included from arch/x86/include/asm/io.h:203:0, from arch/x86/include/asm/realmode.h:5, from arch/x86/include/asm/acpi.h:33, from arch/x86/include/asm/fixmap.h:19, from arch/x86/include/asm/apic.h:10, from arch/x86/include/asm/smp.h:12, from include/linux/smp.h:59, from include/linux/topology.h:33, from include/linux/gfp.h:8, from include/linux/idr.h:16, from include/linux/kernfs.h:14, from include/linux/sysfs.h:15, from include/linux/kobject.h:21, from include/linux/of.h:21, from include/linux/of_dma.h:16, from sound/soc/sh/rcar/dma.c:12: include/asm-generic/iomap.h:31:21: note: expected 'void *' \ but argument is of type 'volatile void *' extern unsigned int ioread32(void __iomem *); ^~~~~~~~ Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index c2e199b4fcf4..241cb3b08a07 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -459,7 +459,7 @@ static void rsnd_dmapp_bset(struct rsnd_dma *dma, u32 data, u32 mask, u32 reg) struct rsnd_mod *mod = rsnd_mod_get(dma); struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); - volatile void __iomem *addr = rsnd_dmapp_addr(dmac, dma, reg); + void __iomem *addr = rsnd_dmapp_addr(dmac, dma, reg); u32 val = ioread32(addr); val &= ~mask; From 763811987d5088d0461164f80e9d16869847a040 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 16 Mar 2017 13:58:40 +0800 Subject: [PATCH 17/31] ASoC: rt5665: fix define of RT5665_HP_DRIVER_5X It is (0x3 << 2), not (0x2 << 2). Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5665.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5665.h b/sound/soc/codecs/rt5665.h index 12f7080a0d3c..a30f5e6d0628 100644 --- a/sound/soc/codecs/rt5665.h +++ b/sound/soc/codecs/rt5665.h @@ -1106,7 +1106,7 @@ #define RT5665_HP_DRIVER_MASK (0x3 << 2) #define RT5665_HP_DRIVER_1X (0x0 << 2) #define RT5665_HP_DRIVER_3X (0x1 << 2) -#define RT5665_HP_DRIVER_5X (0x2 << 2) +#define RT5665_HP_DRIVER_5X (0x3 << 2) #define RT5665_LDO1_DVO_MASK (0x3) #define RT5665_LDO1_DVO_09 (0x0) #define RT5665_LDO1_DVO_10 (0x1) From 83749abaafed348cdac6a63b5f76aa9b1b42409b Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 20 Mar 2017 10:20:53 +0800 Subject: [PATCH 18/31] ASoC: rt5665: fix wrong shift rt5665_if2_1_adc_in_enum The shift is RT5665_IF2_1_ADC_IN_SFT not RT5665_IF3_ADC_IN_SFT. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5665.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 61137160c116..476135ec5726 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -2252,7 +2252,7 @@ static const char * const rt5665_if2_1_adc_in_src[] = { static const SOC_ENUM_SINGLE_DECL( rt5665_if2_1_adc_in_enum, RT5665_DIG_INF2_DATA, - RT5665_IF3_ADC_IN_SFT, rt5665_if2_1_adc_in_src); + RT5665_IF2_1_ADC_IN_SFT, rt5665_if2_1_adc_in_src); static const struct snd_kcontrol_new rt5665_if2_1_adc_in_mux = SOC_DAPM_ENUM("IF2_1 ADC IN Source", rt5665_if2_1_adc_in_enum); From a82f16188a32a3c889916c582ea2d9188e3c2734 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Myl=C3=A8ne=20Josserand?= Date: Sat, 18 Mar 2017 08:55:05 +0100 Subject: [PATCH 19/31] ASoC: sun8i-codec: Remove analog "HP" widget MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The "HP" widget is already present and take part to the analog part (sun8i-codec-analog). Remove it from the digital part as it is unnecessary. Signed-off-by: Mylène Josserand Acked-by: Chen-Yu Tsai Signed-off-by: Mark Brown --- sound/soc/sunxi/sun8i-codec.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index b92bdc8361af..d60f6fbd36a2 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -321,8 +321,6 @@ static const struct snd_soc_dapm_widget sun8i_codec_dapm_widgets[] = { SUN8I_MOD_RST_CTL_AIF1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("RST DAC", SUN8I_MOD_RST_CTL, SUN8I_MOD_RST_CTL_DAC, 0, NULL, 0), - - SND_SOC_DAPM_OUTPUT("HP"), }; static const struct snd_soc_dapm_route sun8i_codec_dapm_routes[] = { @@ -344,10 +342,6 @@ static const struct snd_soc_dapm_route sun8i_codec_dapm_routes[] = { /* DAC Mixer Routes */ { "Left DAC Mixer", "LSlot 0", "Digital Left DAC"}, { "Right DAC Mixer", "RSlot 0", "Digital Right DAC"}, - - /* End of route : HP out */ - { "HP", NULL, "Left DAC Mixer" }, - { "HP", NULL, "Right DAC Mixer" }, }; static struct snd_soc_dai_ops sun8i_codec_dai_ops = { From 649d55436137b397accb6a9d1b6975598c693bcd Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Myl=C3=A8ne=20Josserand?= Date: Sat, 18 Mar 2017 08:55:06 +0100 Subject: [PATCH 20/31] ASoC: sun8i-codec: Update mixer to use SOC_DAPM_DOUBLE MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Update the driver to use the new SOC_DAPM_DOUBLE definition on the digital DAC mixer. Update the names accordingly as, when they are shared, the controls are not prefixed with the widget's name anymore. Signed-off-by: Mylène Josserand Acked-by: Chen-Yu Tsai Signed-off-by: Mark Brown --- sound/soc/sunxi/sun8i-codec.c | 45 ++++++++++++++++------------------- 1 file changed, 21 insertions(+), 24 deletions(-) diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index d60f6fbd36a2..107fa8213600 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -259,25 +259,20 @@ static int sun8i_codec_hw_params(struct snd_pcm_substream *substream, return 0; } -static const struct snd_kcontrol_new sun8i_output_left_mixer_controls[] = { - SOC_DAPM_SINGLE("LSlot 0", SUN8I_DAC_MXR_SRC, - SUN8I_DAC_MXR_SRC_DACL_MXR_SRC_AIF1DA0L, 1, 0), - SOC_DAPM_SINGLE("LSlot 1", SUN8I_DAC_MXR_SRC, - SUN8I_DAC_MXR_SRC_DACL_MXR_SRC_AIF1DA1L, 1, 0), - SOC_DAPM_SINGLE("DACL", SUN8I_DAC_MXR_SRC, - SUN8I_DAC_MXR_SRC_DACL_MXR_SRC_AIF2DACL, 1, 0), - SOC_DAPM_SINGLE("ADCL", SUN8I_DAC_MXR_SRC, - SUN8I_DAC_MXR_SRC_DACL_MXR_SRC_ADCL, 1, 0), -}; - -static const struct snd_kcontrol_new sun8i_output_right_mixer_controls[] = { - SOC_DAPM_SINGLE("RSlot 0", SUN8I_DAC_MXR_SRC, +static const struct snd_kcontrol_new sun8i_dac_mixer_controls[] = { + SOC_DAPM_DOUBLE("AIF1 Slot 0 Digital DAC Playback Switch", + SUN8I_DAC_MXR_SRC, + SUN8I_DAC_MXR_SRC_DACL_MXR_SRC_AIF1DA0L, SUN8I_DAC_MXR_SRC_DACR_MXR_SRC_AIF1DA0R, 1, 0), - SOC_DAPM_SINGLE("RSlot 1", SUN8I_DAC_MXR_SRC, + SOC_DAPM_DOUBLE("AIF1 Slot 1 Digital DAC Playback Switch", + SUN8I_DAC_MXR_SRC, + SUN8I_DAC_MXR_SRC_DACL_MXR_SRC_AIF1DA1L, SUN8I_DAC_MXR_SRC_DACR_MXR_SRC_AIF1DA1R, 1, 0), - SOC_DAPM_SINGLE("DACR", SUN8I_DAC_MXR_SRC, + SOC_DAPM_DOUBLE("AIF2 Digital DAC Playback Switch", SUN8I_DAC_MXR_SRC, + SUN8I_DAC_MXR_SRC_DACL_MXR_SRC_AIF2DACL, SUN8I_DAC_MXR_SRC_DACR_MXR_SRC_AIF2DACR, 1, 0), - SOC_DAPM_SINGLE("ADCR", SUN8I_DAC_MXR_SRC, + SOC_DAPM_DOUBLE("ADC Digital DAC Playback Switch", SUN8I_DAC_MXR_SRC, + SUN8I_DAC_MXR_SRC_DACL_MXR_SRC_ADCL, SUN8I_DAC_MXR_SRC_DACR_MXR_SRC_ADCR, 1, 0), }; @@ -293,12 +288,12 @@ static const struct snd_soc_dapm_widget sun8i_codec_dapm_widgets[] = { SUN8I_AIF1_DACDAT_CTRL_AIF1_DA0R_ENA, 0), /* DAC Mixers */ - SND_SOC_DAPM_MIXER("Left DAC Mixer", SND_SOC_NOPM, 0, 0, - sun8i_output_left_mixer_controls, - ARRAY_SIZE(sun8i_output_left_mixer_controls)), - SND_SOC_DAPM_MIXER("Right DAC Mixer", SND_SOC_NOPM, 0, 0, - sun8i_output_right_mixer_controls, - ARRAY_SIZE(sun8i_output_right_mixer_controls)), + SND_SOC_DAPM_MIXER("Left Digital DAC Mixer", SND_SOC_NOPM, 0, 0, + sun8i_dac_mixer_controls, + ARRAY_SIZE(sun8i_dac_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Digital DAC Mixer", SND_SOC_NOPM, 0, 0, + sun8i_dac_mixer_controls, + ARRAY_SIZE(sun8i_dac_mixer_controls)), /* Clocks */ SND_SOC_DAPM_SUPPLY("MODCLK AFI1", SUN8I_MOD_CLK_ENA, @@ -340,8 +335,10 @@ static const struct snd_soc_dapm_route sun8i_codec_dapm_routes[] = { { "Digital Right DAC", NULL, "DAC" }, /* DAC Mixer Routes */ - { "Left DAC Mixer", "LSlot 0", "Digital Left DAC"}, - { "Right DAC Mixer", "RSlot 0", "Digital Right DAC"}, + { "Left Digital DAC Mixer", "AIF1 Slot 0 Digital DAC Playback Switch", + "Digital Left DAC"}, + { "Right Digital DAC Mixer", "AIF1 Slot 0 Digital DAC Playback Switch ", + "Digital Right DAC"}, }; static struct snd_soc_dai_ops sun8i_codec_dai_ops = { From 79e26de81448fad80f6c15ae1e3c9e7fca2740cd Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Myl=C3=A8ne=20Josserand?= Date: Sat, 18 Mar 2017 08:55:07 +0100 Subject: [PATCH 21/31] ASoC: sun8i-codec: Fix space on audio-routing widget MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit An unwanted space is present in an audio widget's name on the dapm routing. It causes an error on the recognition of this widget (error: ("no dapm match for AIF1 Slot 0 Right"). Remove the space fixes it. Signed-off-by: Mylène Josserand Acked-by: Chen-Yu Tsai Signed-off-by: Mark Brown --- sound/soc/sunxi/sun8i-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index 107fa8213600..adb13fbd2006 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -337,7 +337,7 @@ static const struct snd_soc_dapm_route sun8i_codec_dapm_routes[] = { /* DAC Mixer Routes */ { "Left Digital DAC Mixer", "AIF1 Slot 0 Digital DAC Playback Switch", "Digital Left DAC"}, - { "Right Digital DAC Mixer", "AIF1 Slot 0 Digital DAC Playback Switch ", + { "Right Digital DAC Mixer", "AIF1 Slot 0 Digital DAC Playback Switch", "Digital Right DAC"}, }; From d1792285ca63e17f8a7eb42efa48834c261a2d8f Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Myl=C3=A8ne=20Josserand?= Date: Sat, 18 Mar 2017 08:55:08 +0100 Subject: [PATCH 22/31] ASoC: sun8i-codec: Convert to use SND_SOC_DAPM_AIF_IN MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Update the driver to use SND_SOC_DAPM_AIF_IN instead of SND_SOC_DAPM_DAC. Rename the interface's widgets to be more precise on which slot the interface is connected. Signed-off-by: Mylène Josserand Acked-by: Chen-Yu Tsai Signed-off-by: Mark Brown --- sound/soc/sunxi/sun8i-codec.c | 20 +++++++++++--------- 1 file changed, 11 insertions(+), 9 deletions(-) diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index adb13fbd2006..7527ba29a5a0 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -281,11 +281,13 @@ static const struct snd_soc_dapm_widget sun8i_codec_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("DAC", SUN8I_DAC_DIG_CTRL, SUN8I_DAC_DIG_CTRL_ENDA, 0, NULL, 0), - /* Analog DAC */ - SND_SOC_DAPM_DAC("Digital Left DAC", "Playback", SUN8I_AIF1_DACDAT_CTRL, - SUN8I_AIF1_DACDAT_CTRL_AIF1_DA0L_ENA, 0), - SND_SOC_DAPM_DAC("Digital Right DAC", "Playback", SUN8I_AIF1_DACDAT_CTRL, - SUN8I_AIF1_DACDAT_CTRL_AIF1_DA0R_ENA, 0), + /* Analog DAC AIF */ + SND_SOC_DAPM_AIF_IN("AIF1 Slot 0 Left", "Playback", 0, + SUN8I_AIF1_DACDAT_CTRL, + SUN8I_AIF1_DACDAT_CTRL_AIF1_DA0L_ENA, 0), + SND_SOC_DAPM_AIF_IN("AIF1 Slot 0 Right", "Playback", 0, + SUN8I_AIF1_DACDAT_CTRL, + SUN8I_AIF1_DACDAT_CTRL_AIF1_DA0R_ENA, 0), /* DAC Mixers */ SND_SOC_DAPM_MIXER("Left Digital DAC Mixer", SND_SOC_NOPM, 0, 0, @@ -331,14 +333,14 @@ static const struct snd_soc_dapm_route sun8i_codec_dapm_routes[] = { { "DAC", NULL, "MODCLK DAC" }, /* DAC Routes */ - { "Digital Left DAC", NULL, "DAC" }, - { "Digital Right DAC", NULL, "DAC" }, + { "AIF1 Slot 0 Right", NULL, "DAC" }, + { "AIF1 Slot 0 Left", NULL, "DAC" }, /* DAC Mixer Routes */ { "Left Digital DAC Mixer", "AIF1 Slot 0 Digital DAC Playback Switch", - "Digital Left DAC"}, + "AIF1 Slot 0 Left"}, { "Right Digital DAC Mixer", "AIF1 Slot 0 Digital DAC Playback Switch", - "Digital Right DAC"}, + "AIF1 Slot 0 Right"}, }; static struct snd_soc_dai_ops sun8i_codec_dai_ops = { From eb3abaea7ea42619a48fa84e4b1ff48f1b18d863 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Myl=C3=A8ne=20Josserand?= Date: Sat, 18 Mar 2017 08:55:09 +0100 Subject: [PATCH 23/31] ARM: dts: sun8i: Update audio-routing with renamed widgets MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The digital AIF interfaces has been renamed in the sun8i audio codec driver so the audio-routing in the device tree must be renamed too. Signed-off-by: Mylène Josserand Acked-by: Chen-Yu Tsai Signed-off-by: Mark Brown --- arch/arm/boot/dts/sun8i-a33.dtsi | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/arch/arm/boot/dts/sun8i-a33.dtsi b/arch/arm/boot/dts/sun8i-a33.dtsi index 18c174fef84f..0467fb365bfc 100644 --- a/arch/arm/boot/dts/sun8i-a33.dtsi +++ b/arch/arm/boot/dts/sun8i-a33.dtsi @@ -113,8 +113,8 @@ sound: sound { simple-audio-card,mclk-fs = <512>; simple-audio-card,aux-devs = <&codec_analog>; simple-audio-card,routing = - "Left DAC", "Digital Left DAC", - "Right DAC", "Digital Right DAC"; + "Left DAC", "AIF1 Slot 0 Left", + "Right DAC", "AIF1 Slot 0 Right"; status = "disabled"; simple-audio-card,cpu { From c6736a94d0e527ddc0d1eb99dbc59886a9ecf471 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Mar 2017 13:26:02 +0100 Subject: [PATCH 24/31] ALSA: x86: Make CONFIG_SND_X86 bool CONFIG_SND_X86 is a menu config to filter only for x86-specific drivers in its sub-menu, and this doesn't have to be tristate but rather it should be a bool. Also, like other sub-menu configs, it's more user-friendly to be default=y; it's merely a menu config and the actual drivers are configured in the sub-menu, after all. Fixes: 287599cf2d77 ("ALSA: add Intel HDMI LPE audio driver for BYT/CHT-T") Signed-off-by: Takashi Iwai --- sound/x86/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig index 84c8f8fc597c..8adf4d1bd46e 100644 --- a/sound/x86/Kconfig +++ b/sound/x86/Kconfig @@ -1,6 +1,7 @@ menuconfig SND_X86 - tristate "X86 sound devices" + bool "X86 sound devices" depends on X86 + default y ---help--- X86 sound devices that don't fall under SoC or PCI categories From c520ff3d03f0b5db7146d9beed6373ad5d2a5e0e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Mar 2017 13:56:04 +0100 Subject: [PATCH 25/31] ALSA: seq: Fix racy cell insertions during snd_seq_pool_done() When snd_seq_pool_done() is called, it marks the closing flag to refuse the further cell insertions. But snd_seq_pool_done() itself doesn't clear the cells but just waits until all cells are cleared by the caller side. That is, it's racy, and this leads to the endless stall as syzkaller spotted. This patch addresses the racy by splitting the setup of pool->closing flag out of snd_seq_pool_done(), and calling it properly before snd_seq_pool_done(). BugLink: http://lkml.kernel.org/r/CACT4Y+aqqy8bZA1fFieifNxR2fAfFQQABcBHj801+u5ePV0URw@mail.gmail.com Reported-and-tested-by: Dmitry Vyukov Cc: Signed-off-by: Takashi Iwai --- sound/core/seq/seq_clientmgr.c | 1 + sound/core/seq/seq_fifo.c | 3 +++ sound/core/seq/seq_memory.c | 17 +++++++++++++---- sound/core/seq/seq_memory.h | 1 + 4 files changed, 18 insertions(+), 4 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 4c935202ce23..f3b1d7f50b81 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1832,6 +1832,7 @@ static int snd_seq_ioctl_set_client_pool(struct snd_seq_client *client, info->output_pool != client->pool->size)) { if (snd_seq_write_pool_allocated(client)) { /* remove all existing cells */ + snd_seq_pool_mark_closing(client->pool); snd_seq_queue_client_leave_cells(client->number); snd_seq_pool_done(client->pool); } diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c index 448efd4e980e..33980d1c8037 100644 --- a/sound/core/seq/seq_fifo.c +++ b/sound/core/seq/seq_fifo.c @@ -72,6 +72,9 @@ void snd_seq_fifo_delete(struct snd_seq_fifo **fifo) return; *fifo = NULL; + if (f->pool) + snd_seq_pool_mark_closing(f->pool); + snd_seq_fifo_clear(f); /* wake up clients if any */ diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index 1a1acf3ddda4..d4c61ec9be13 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -415,6 +415,18 @@ int snd_seq_pool_init(struct snd_seq_pool *pool) return 0; } +/* refuse the further insertion to the pool */ +void snd_seq_pool_mark_closing(struct snd_seq_pool *pool) +{ + unsigned long flags; + + if (snd_BUG_ON(!pool)) + return; + spin_lock_irqsave(&pool->lock, flags); + pool->closing = 1; + spin_unlock_irqrestore(&pool->lock, flags); +} + /* remove events */ int snd_seq_pool_done(struct snd_seq_pool *pool) { @@ -425,10 +437,6 @@ int snd_seq_pool_done(struct snd_seq_pool *pool) return -EINVAL; /* wait for closing all threads */ - spin_lock_irqsave(&pool->lock, flags); - pool->closing = 1; - spin_unlock_irqrestore(&pool->lock, flags); - if (waitqueue_active(&pool->output_sleep)) wake_up(&pool->output_sleep); @@ -485,6 +493,7 @@ int snd_seq_pool_delete(struct snd_seq_pool **ppool) *ppool = NULL; if (pool == NULL) return 0; + snd_seq_pool_mark_closing(pool); snd_seq_pool_done(pool); kfree(pool); return 0; diff --git a/sound/core/seq/seq_memory.h b/sound/core/seq/seq_memory.h index 4a2ec779b8a7..32f959c17786 100644 --- a/sound/core/seq/seq_memory.h +++ b/sound/core/seq/seq_memory.h @@ -84,6 +84,7 @@ static inline int snd_seq_total_cells(struct snd_seq_pool *pool) int snd_seq_pool_init(struct snd_seq_pool *pool); /* done pool - free events */ +void snd_seq_pool_mark_closing(struct snd_seq_pool *pool); int snd_seq_pool_done(struct snd_seq_pool *pool); /* create pool */ From 3f307834e695f59dac4337a40316bdecfb9d0508 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Thu, 23 Mar 2017 10:00:25 +0800 Subject: [PATCH 26/31] ALSA: hda - Adding a group of pin definition to fix headset problem A new Dell laptop needs to apply ALC269_FIXUP_DELL1_MIC_NO_PRESENCE to fix the headset problem, and the pin definiton of this machine is not in the pin quirk table yet, now adding it to the table. Signed-off-by: Hui Wang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8d6b3703d0a2..7f989898cbd9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6102,6 +6102,8 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { ALC295_STANDARD_PINS, {0x17, 0x21014040}, {0x18, 0x21a19050}), + SND_HDA_PIN_QUIRK(0x10ec0295, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC295_STANDARD_PINS), SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC298_STANDARD_PINS, {0x17, 0x90170110}), From 2d7d54002e396c180db0c800c1046f0a3c471597 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 24 Mar 2017 17:07:57 +0100 Subject: [PATCH 27/31] ALSA: seq: Fix race during FIFO resize When a new event is queued while processing to resize the FIFO in snd_seq_fifo_clear(), it may lead to a use-after-free, as the old pool that is being queued gets removed. For avoiding this race, we need to close the pool to be deleted and sync its usage before actually deleting it. The issue was spotted by syzkaller. Reported-by: Dmitry Vyukov Cc: Signed-off-by: Takashi Iwai --- sound/core/seq/seq_fifo.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c index 33980d1c8037..01c4cfe30c9f 100644 --- a/sound/core/seq/seq_fifo.c +++ b/sound/core/seq/seq_fifo.c @@ -267,6 +267,10 @@ int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize) /* NOTE: overflow flag is not cleared */ spin_unlock_irqrestore(&f->lock, flags); + /* close the old pool and wait until all users are gone */ + snd_seq_pool_mark_closing(oldpool); + snd_use_lock_sync(&f->use_lock); + /* release cells in old pool */ for (cell = oldhead; cell; cell = next) { next = cell->next; From 3c9d3f1bc2defd418b5933bbc928096c9c686d3b Mon Sep 17 00:00:00 2001 From: Arnaud Pouliquen Date: Thu, 23 Mar 2017 19:39:54 +0100 Subject: [PATCH 28/31] ASoC: STI: Fix reader substream pointer set reader->substream is used in IRQ handler for error case but is never set. Set value to pcm substream on DAI startup and clean it on dai shutdown. Signed-off-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- sound/soc/sti/uniperif_reader.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 5992c6ab3833..93a8df6ed880 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -349,6 +349,8 @@ static int uni_reader_startup(struct snd_pcm_substream *substream, struct uniperif *reader = priv->dai_data.uni; int ret; + reader->substream = substream; + if (!UNIPERIF_TYPE_IS_TDM(reader)) return 0; @@ -378,6 +380,7 @@ static void uni_reader_shutdown(struct snd_pcm_substream *substream, /* Stop the reader */ uni_reader_stop(reader); } + reader->substream = NULL; } static const struct snd_soc_dai_ops uni_reader_dai_ops = { From 971edb0a0087b65bd3726d23b2dffff405d48f72 Mon Sep 17 00:00:00 2001 From: Lucas Stach Date: Thu, 23 Mar 2017 15:05:26 +0100 Subject: [PATCH 29/31] ASoC: simple-card: fix simple_dai clk lookup The clock needs to be stored in the simple_dai structure, so it can be enabled later on. This has been broken during the conversion to use devm_* functions for the clk lookup. Fixes: e984fd61e860 (ASoC: simple-card: use devm_get_clk_from_child()) Signed-off-by: Lucas Stach Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 4924575d2e95..343b291fc372 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -115,6 +115,7 @@ int asoc_simple_card_parse_clk(struct device *dev, clk = devm_get_clk_from_child(dev, node, NULL); if (!IS_ERR(clk)) { simple_dai->sysclk = clk_get_rate(clk); + simple_dai->clk = clk; } else if (!of_property_read_u32(node, "system-clock-frequency", &val)) { simple_dai->sysclk = val; } else { From 2f726aec19a9d2c63bec9a8a53a3910ffdcd09f8 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Fri, 31 Mar 2017 10:31:40 +0800 Subject: [PATCH 30/31] ALSA: hda - fix a problem for lineout on a Dell AIO machine On this Dell AIO machine, the lineout jack does not work. We found the pin 0x1a is assigned to lineout on this machine, and in the past, we applied ALC298_FIXUP_DELL1_MIC_NO_PRESENCE to fix the heaset-set mic problem for this machine, this fixup will redefine the pin 0x1a to headphone-mic, as a result the lineout doesn't work anymore. After consulting with Dell, they told us this machine doesn't support microphone via headset jack, so we add a new fixup which only defines the pin 0x18 as the headset-mic. [rearranged the fixup insertion position by tiwai in order to make the merge with other branches easier -- tiwai] Fixes: 59ec4b57bcae ("ALSA: hda - Fix headset mic detection problem for two dell machines") Cc: Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7f989898cbd9..299835d1fbaa 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4858,6 +4858,7 @@ enum { ALC292_FIXUP_DISABLE_AAMIX, ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK, ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, ALC293_FIXUP_LENOVO_SPK_NOISE, @@ -5470,6 +5471,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, [ALC275_FIXUP_DELL_XPS] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -5542,7 +5552,7 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc298_fixup_speaker_volume, .chained = true, - .chain_id = ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, + .chain_id = ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, }, [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { .type = HDA_FIXUP_PINS, From 3d016d57fdc5e6caa4cd67896f4b081bccad6e2c Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 3 Apr 2017 21:13:40 +0900 Subject: [PATCH 31/31] ALSA: oxfw: fix regression to handle Stanton SCS.1m/1d At a commit 6c29230e2a5f ("ALSA: oxfw: delayed registration of sound card"), ALSA oxfw driver fails to handle SCS.1m/1d, due to -EBUSY at a call of snd_card_register(). The cause is that the driver manages to register two rawmidi instances with the same device number 0. This is a regression introduced since kernel 4.7. This commit fixes the regression, by fixing up device property after discovering stream formats. Fixes: 6c29230e2a5f ("ALSA: oxfw: delayed registration of sound card") Cc: # 4.7+ Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/oxfw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 74d7fb6efce6..413ab6313bb6 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -227,11 +227,11 @@ static void do_registration(struct work_struct *work) if (err < 0) goto error; - err = detect_quirks(oxfw); + err = snd_oxfw_stream_discover(oxfw); if (err < 0) goto error; - err = snd_oxfw_stream_discover(oxfw); + err = detect_quirks(oxfw); if (err < 0) goto error;