February 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [bug report] ASoC: hdac_hdmi: Add support for multiple ports to a PCM
by Dan Carpenter 22 Feb '17

22 Feb '17

Hello Jeeja KP, The patch e0e5d3e5a53b: "ASoC: hdac_hdmi: Add support for multiple ports to a PCM" from Feb 7, 2017, leads to the following static checker warning: sound/soc/codecs/hdac_hdmi.c:2011 hdac_hdmi_dev_remove() error: potential NULL dereference 'port'. sound/soc/codecs/hdac_hdmi.c 1995 static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev) 1996 { 1997 struct hdac_hdmi_priv *hdmi = edev->private_data; 1998 struct hdac_hdmi_pin *pin, *pin_next; 1999 struct hdac_hdmi_cvt *cvt, *cvt_next; 2000 struct hdac_hdmi_pcm *pcm, *pcm_next; 2001 struct hdac_hdmi_port *port; 2002 int i; 2003 2004 snd_soc_unregister_codec(&edev->hdac.dev); 2005 2006 list_for_each_entry_safe(pcm, pcm_next, &hdmi->pcm_list, head) { 2007 pcm->cvt = NULL; 2008 if (list_empty(&pcm->port_list)) 2009 continue; 2010 2011 list_for_each_entry(port, &pcm->port_list, head) 2012 port = NULL; It's not clear to me what this loop is supposed to do. To me it looks like it will Oops every single time without fail. 2013 2014 list_del(&pcm->head); 2015 kfree(pcm); 2016 } 2017 2018 list_for_each_entry_safe(cvt, cvt_next, &hdmi->cvt_list, head) { 2019 list_del(&cvt->head); 2020 kfree(cvt->name); 2021 kfree(cvt); 2022 } 2023 2024 list_for_each_entry_safe(pin, pin_next, &hdmi->pin_list, head) { 2025 for (i = 0; i < pin->num_ports; i++) 2026 pin->ports[i].pin = NULL; 2027 kfree(pin->ports); 2028 list_del(&pin->head); 2029 kfree(pin); 2030 } 2031 2032 return 0; 2033 } regards, dan carpenter

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[alsa-devel] [PATCH 0/5] ALSA: usb-audio: fixes for quirks of Tascam US-16x08
by Takashi Sakamoto 22 Feb '17

22 Feb '17

Hi, A commit d2bb390a2081 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk") was merged just now, while static code checker reports some apparent bugs. This patchset is to solve the bugs. Takashi Sakamoto (5): ALSA: usb-audio: localize one-referer variable ALSA: usb-audio: purge needless variable length array ALSA: usb-audio: localize function without external linkage ALSA: usb-audio: deallocate memory objects in error path ALSA: usb-audio: fix msleep timeout due to minimum resolution of task scheduling sound/usb/mixer_us16x08.c | 42 ++++++++++++++++++++++++------------------ 1 file changed, 24 insertions(+), 18 deletions(-) -- 2.9.3

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[alsa-devel] Custom USB sound card configuration
by Stephen Wilson 22 Feb '17

22 Feb '17

Hello, I'm developing an application using the pjSip library/framework using the ALSA library as the interface to the PCM audio devices. The audio device is a USB PIC presenting a single sound card with 4 devices (PIC code is still in development so ignore the examples names) - see output of aplay -l below... When I launch my pjSip application (and the pjSIP test application) only the first device on my sound card is listed - see below.... My thoughts are that I need a custom card configuration file for my sound card so all four devices are present as specific devices but I cannot work out how to do this from the ALSA Project documentation. I assume I need a new entry in the /usr/share/alsa/cards/aliases.conf file and an equivalent [example].conf file. What needs to go into both of these files? Any help would be greatly appreciated... Thanks Steve [root@##########]# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC662 rev1 Digital [ALC662 rev1 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Example [Harmony USB Headset Example], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Example [Harmony USB Headset Example], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Example [Harmony USB Headset Example], device 2: USB Audio [USB Audio #2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Example [Harmony USB Headset Example], device 3: USB Audio [USB Audio #3] Subdevices: 1/1 Subdevice #0: subdevice #0 [root@################]# pjsystest-i686-pc-linux-gnu 09:15:12.615 os_core_unix.c !pjlib 2.5.5 for POSIX initialized 09:15:12.621 sip_endpoint.c .Creating endpoint instance... 09:15:12.623 pjlib .select() I/O Queue created (0xa20aa00) 09:15:12.623 sip_endpoint.c .Module "mod-msg-print" registered 09:15:12.623 sip_transport. .Transport manager created. 09:15:12.624 pjsua_core.c .PJSUA state changed: NULL --> CREATED 09:15:12.624 sip_endpoint.c .Module "mod-pjsua-log" registered 09:15:12.624 sip_endpoint.c .Module "mod-tsx-layer" registered 09:15:12.624 sip_endpoint.c .Module "mod-stateful-util" registered 09:15:12.624 sip_endpoint.c .Module "mod-ua" registered 09:15:12.625 sip_endpoint.c .Module "mod-100rel" registered 09:15:12.625 sip_endpoint.c .Module "mod-pjsua" registered 09:15:12.625 sip_endpoint.c .Module "mod-invite" registered 09:15:12.897 alsa_dev.c ..ALSA driver found 18 devices 09:15:12.897 alsa_dev.c ..ALSA initialized 09:15:12.898 pjlib ..select() I/O Queue created (0xa23b4fc) WARNING: no real random source present! 09:15:12.910 sip_endpoint.c .Module "mod-evsub" registered 09:15:12.910 sip_endpoint.c .Module "mod-presence" registered 09:15:12.910 sip_endpoint.c .Module "mod-mwi" registered 09:15:12.910 sip_endpoint.c .Module "mod-refer" registered 09:15:12.910 sip_endpoint.c .Module "mod-pjsua-pres" registered 09:15:12.910 sip_endpoint.c .Module "mod-pjsua-im" registered 09:15:12.910 sip_endpoint.c .Module "mod-pjsua-options" registered 09:15:12.910 pjsua_core.c .1 SIP worker threads created 09:15:12.910 pjsua_core.c .pjsua version 2.5.5 for Linux-2.6.34.130/i686/glibc-2.12 initialized 09:15:12.910 pjsua_core.c .PJSUA state changed: CREATED --> INIT 09:15:12.910 pjsua_core.c PJSUA state changed: INIT --> STARTING 09:15:12.910 sip_endpoint.c .Module "mod-unsolicited-mwi" registered 09:15:12.911 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING M E N U : --------- 0: Tests 00: Run test wizard 01: Device Test 02: Play Tone 03: Play WAV File1 04: Play WAV File2 05: Record Audio 06: Latency Test 07: AEC/AES Test 09: Exit 1: Options 10: View Devices 11: View Settings Enter the menu number: 10 09:15:13.463 systest.c Running Audio Device List Audio Device List Found 18 devices 0: ALSA [default:CARD=Intel] (0/1) 1: ALSA [sysdefault:CARD=Intel] (0/1) 2: ALSA [front:CARD=Intel,DEV=0] (1/1) 3: ALSA [surround40:CARD=Intel,DEV=0] (1/1) 4: ALSA [surround51:CARD=Intel,DEV=0] (1/1) 5: ALSA [surround71:CARD=Intel,DEV=0] (1/1) 6: ALSA [iec958:CARD=Intel,DEV=0] (0/1) 7: ALSA [hdmi:CARD=Intel,DEV=0] (0/1) 8: ALSA [default:CARD=Example] (1/1) 9: ALSA [sysdefault:CARD=Example] (1/1) 10: ALSA [front:CARD=Example,DEV=0] (1/1) 11: ALSA [surround21:CARD=Example,DEV=0] (1/1) 12: ALSA [surround40:CARD=Example,DEV=0] (1/1) 13: ALSA [surround41:CARD=Example,DEV=0] (1/1) 14: ALSA [surround50:CARD=Example,DEV=0] (1/1) 15: ALSA [surround51:CARD=Example,DEV=0] (1/1) 16: ALSA [surround71:CARD=Example,DEV=0] (1/1) 17: ALSA [iec958:CARD=Example,DEV=0] (1/1)

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[alsa-devel] [PATCH v2] ALSA: usb-audio: purge needless variable length array
by Takashi Sakamoto 22 Feb '17

22 Feb '17

Variable length array is used in 'snd_us16x08_meter_get()', while there is no need. It's better to purge it because variable length array has overhead for stack handling. This commit replaces the array with static length. Sparse generated below warning. sound/usb/mixer_us16x08.c:714:18: warning: Variable length array is used. Fixes: d2bb390a2081 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/usb/mixer_us16x08.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index 7ac45ec..73a0b9a 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -711,7 +711,7 @@ static int snd_us16x08_meter_get(struct snd_kcontrol *kcontrol, struct snd_usb_audio *chip = elem->head.mixer->chip; struct snd_us16x08_meter_store *store = elem->private_data; u8 meter_urb[64]; - char tmp[max(sizeof(mix_init_msg1), sizeof(mix_init_msg2))]; + char tmp[sizeof(mix_init_msg2)] = {0}; if (elem) { store = (struct snd_us16x08_meter_store *) elem->private_data; @@ -721,8 +721,8 @@ static int snd_us16x08_meter_get(struct snd_kcontrol *kcontrol, switch (kcontrol->private_value) { case 0: - memcpy(tmp, mix_init_msg1, sizeof(mix_init_msg1)); - snd_us16x08_send_urb(chip, tmp, 4); + snd_us16x08_send_urb(chip, (char *)mix_init_msg1, + sizeof(mix_init_msg1)); snd_us16x08_recv_urb(chip, meter_urb, sizeof(meter_urb)); kcontrol->private_value++; @@ -740,7 +740,7 @@ static int snd_us16x08_meter_get(struct snd_kcontrol *kcontrol, case 3: memcpy(tmp, mix_init_msg2, sizeof(mix_init_msg2)); tmp[2] = snd_get_meter_comp_index(store); - snd_us16x08_send_urb(chip, tmp, 10); + snd_us16x08_send_urb(chip, tmp, sizeof(mix_init_msg2)); snd_us16x08_recv_urb(chip, meter_urb, sizeof(meter_urb)); kcontrol->private_value = 0; -- 2.9.3

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[alsa-devel] [PATCH v2 1/3] ASoC: zx-tdm: add bindings doc for zte's tdm controller
by Baoyou Xie 22 Feb '17

22 Feb '17

This patch adds dt-binding documentation for zte's tdm controller. Signed-off-by: Baoyou Xie <baoyou.xie(a)linaro.org> --- .../devicetree/bindings/sound/zte,tdm.txt | 30 ++++++++++++++++++++++ 1 file changed, 30 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/zte,tdm.txt diff --git a/Documentation/devicetree/bindings/sound/zte,tdm.txt b/Documentation/devicetree/bindings/sound/zte,tdm.txt new file mode 100644 index 0000000..8ffc39e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/zte,tdm.txt @@ -0,0 +1,30 @@ +ZTE TDM DAI driver + +Required properties: + +- compatible : should be one of the following. + * zte,zx296718-tdm +- reg : physical base address of the controller and length of memory mapped + region. +- clocks : Pairs of phandle and specifier referencing the controller's clocks. +- clock-names: "wclk" for the wclk. + "pclk" for the pclk. +-#clock-cells: should be 1. +- zte,tdm-dma-sysctrl : Reference to the sysctrl controller controlling + the dma. includes: + phandle of sysctrl. + register offset in sysctrl for control dma. + mask of the register that be written to sysctrl. + +Example: + + tdm: tdm@1487000 { + compatible = "zte,zx296718-tdm"; + reg = <0x01487000 0x1000>; + clocks = <&audiocrm AUDIO_TDM_WCLK>, <&audiocrm AUDIO_TDM_PCLK>; + clock-names = "wclk", "pclk"; + #clock-cells = <1>; + pinctrl-names = "default"; + pinctrl-0 = <&tdm_global_pin>; + zte,tdm-dma-sysctrl = <&sysctrl 0x10c 4>; + }; -- 2.7.4

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[alsa-devel] Applied "ASoC: rsnd: drop useles self-assignments" to the asoc tree
by Mark Brown 22 Feb '17

22 Feb '17

The patch ASoC: rsnd: drop useles self-assignments has been applied to the asoc tree at git://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 34157f7bec8276b4296cf2ec172fc13385ac8af7 Mon Sep 17 00:00:00 2001 From: Wolfram Sang <wsa+renesas(a)sang-engineering.com> Date: Mon, 20 Feb 2017 21:58:34 +0100 Subject: [PATCH] ASoC: rsnd: drop useles self-assignments Coverity reported (CID 1397992) this self-assignment. I think the code stays readable even with the assignments removed. Signed-off-by: Wolfram Sang <wsa+renesas(a)sang-engineering.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/core.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 948c5ec87980..72966bdd3daa 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -674,12 +674,10 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* set clock inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_IF: - rdai->bit_clk_inv = rdai->bit_clk_inv; rdai->frm_clk_inv = !rdai->frm_clk_inv; break; case SND_SOC_DAIFMT_IB_NF: rdai->bit_clk_inv = !rdai->bit_clk_inv; - rdai->frm_clk_inv = rdai->frm_clk_inv; break; case SND_SOC_DAIFMT_IB_IF: rdai->bit_clk_inv = !rdai->bit_clk_inv; -- 2.11.0

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[alsa-devel] Applied "ASoC: rsnd: check return value of init function" to the asoc tree
by Mark Brown 22 Feb '17

22 Feb '17

The patch ASoC: rsnd: check return value of init function has been applied to the asoc tree at git://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 56d2c61d611a50e58dba521be1325dc90f9cc933 Mon Sep 17 00:00:00 2001 From: Wolfram Sang <wsa+renesas(a)sang-engineering.com> Date: Mon, 20 Feb 2017 22:05:07 +0100 Subject: [PATCH] ASoC: rsnd: check return value of init function Currently, this function cannot fail for the ADG case. Still, let's apply defensive programming techniques to make sure we fail gracefully whenever rsnd_mod_init() gets extended with another failure case. Reported by Coverity (CID 1397893). Signed-off-by: Wolfram Sang <wsa+renesas(a)sang-engineering.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/adg.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 85a33ac0a5c4..54146f66538c 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -564,6 +564,7 @@ int rsnd_adg_probe(struct rsnd_priv *priv) struct rsnd_adg *adg; struct device *dev = rsnd_priv_to_dev(priv); struct device_node *np = dev->of_node; + int ret; adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL); if (!adg) { @@ -571,8 +572,10 @@ int rsnd_adg_probe(struct rsnd_priv *priv) return -ENOMEM; } - rsnd_mod_init(priv, &adg->mod, &adg_ops, + ret = rsnd_mod_init(priv, &adg->mod, &adg_ops, NULL, NULL, 0, 0); + if (ret) + return ret; rsnd_adg_get_clkin(priv, adg); rsnd_adg_get_clkout(priv, adg); -- 2.11.0

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[alsa-devel] [PATCH] ASoC: mediatek: add I2C dependency for CS42XX8
by Arnd Bergmann 22 Feb '17

22 Feb '17

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 <arnd(a)arndb.de> --- 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 -- 2.9.0

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[alsa-devel] [PATCH] ALSA: x86: hdmi: select CONFIG_SND_PCM
by Arnd Bergmann 21 Feb '17

21 Feb '17

The newly added driver uses the PCM helpers and fails to link if they are disabled: sound/built-in.o: In function `hdmi_lpe_audio_runtime_suspend': intel_hdmi_audio.c:(.text+0x15906): undefined reference to `snd_pcm_suspend' sound/built-in.o: In function `had_pcm_hw_params': intel_hdmi_audio.c:(.text+0x15ac7): undefined reference to `snd_pcm_lib_malloc_pages' sound/built-in.o: In function `had_pcm_open': intel_hdmi_audio.c:(.text+0x15d49): undefined reference to `snd_pcm_hw_constraint_integer' This uses a Kconfig 'select' statement to make sure they are enabled. Fixes: 5dab11d89777 ("ALSA: x86: hdmi: Add audio support for BYT and CHT") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/x86/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig index 30d066e4885e..84c8f8fc597c 100644 --- a/sound/x86/Kconfig +++ b/sound/x86/Kconfig @@ -9,6 +9,7 @@ if SND_X86 config HDMI_LPE_AUDIO tristate "HDMI audio without HDaudio on Intel Atom platforms" depends on DRM_I915 + select SND_PCM help Choose this option to support HDMI LPE Audio mode -- 2.9.0

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[alsa-devel] [PATCH] pcm_rate: Do not discard slave reported delay in status result.
by Alan Young 21 Feb '17

21 Feb '17

Similar to recent dshare patch.

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