May 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: control: fix a redundant-copy issue
by Wenwen Wang 13 May '18

13 May '18

In snd_ctl_elem_add_compat(), the fields of the struct 'data' need to be copied from the corresponding fields of the struct 'data32' in userspace. This is achieved by invoking copy_from_user() and get_user() functions. The problem here is that the 'type' field is copied twice. One is by copy_from_user() and one is by get_user(). Given that the 'type' field is not used between the two copies, the second copy is *completely* redundant and should be removed for better performance and cleanup. Also, these two copies can cause inconsistent data: as the struct 'data32' resides in userspace and a malicious userspace process can race to change the 'type' field between the two copies to cause inconsistent data. Depending on how the data is used in the future, such an inconsistency may cause potential security risks. For above reasons, we should take out the second copy. Signed-off-by: Wenwen Wang <wang6495(a)umn.edu> --- sound/core/control_compat.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index a848836..507fd52 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -396,8 +396,7 @@ static int snd_ctl_elem_add_compat(struct snd_ctl_file *file, if (copy_from_user(&data->id, &data32->id, sizeof(data->id)) || copy_from_user(&data->type, &data32->type, 3 * sizeof(u32))) goto error; - if (get_user(data->owner, &data32->owner) || - get_user(data->type, &data32->type)) + if (get_user(data->owner, &data32->owner)) goto error; switch (data->type) { case SNDRV_CTL_ELEM_TYPE_BOOLEAN: -- 2.7.4

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[alsa-devel] [PATCH v2 0/7] USB Audio Device Class 3.0 BADD profiles support
by Ruslan Bilovol 13 May '18

13 May '18

This patchset adds BADD profiles support from the USB Audio Device Class 3.0 spec [1]. BADD profile support is defined as mandatory feature of UAC3-compliant device, it should be implemented as a separate USB configuration. Notable issue with BADD configuration is that it misses class-specific descriptors (and it's mandatory as per spec), so host should guess them from BADD profile number and parameters of endpoints (type, number of endpoints and max packet size) This patchset adds support of all known/existing BADD profiles from the UAC3 specification. First 5 patches are refactoring and improvements, and last 2 patches actually implement UAC3 BADD profiles support. It's an alternative implementation comparing to [2], and doesn't build usb descriptors on the host but instead initializes alsa-usb structures with known parameters, so we don't need to keep whole class-specific descriptors in the driver since we anyway need to have BADD-specific logic. I've picked one Jorge's UAC1 patch-improvement and updated it to v4.17 wich contais recently introduced header's sanity checks. Remaining part is to add interrupt endpoint support so we will be able to detect jack insertion in the Headset Adapter profile. This has been tested on ARM and x86-64 machines with custom UAC3 gadget which I'll post later to linux-usb Comments and testing are welcome. v2: - split refactoring patch to more atomic and bisectable changes as suggested by Takashi - renamed "Side Tone"->"Sidetone" as suggested by Andrew - revorked main BADD patch to have table lookup for checking BADD channels validity and to reduce identation as suggested by Takashi - also removed code duplication in BADD patch by reusing common part of build_feature_ctl() func v1: http://mailman.alsa-project.org/pipermail/alsa-devel/2018-April/134412.html [1] http://www.usb.org/developers/docs/devclass_docs/USB_Audio_v3.0.zip [2] https://www.spinics.net/lists/alsa-devel/msg71614.html Jorge Sanjuan (1): ALSA: usb: Only get AudioControl header for UAC1 class. Ruslan Bilovol (6): ALSA: usb: stream: move audioformat alloc/init into separate function ALSA: usb: stream: refactor uac1/2 audio interface parsing ALSA: usb: stream: refactor uac3 audio interface parsing ALSA: usb: mixer: make string parsing independent of mixer_build state include: usb: audio-v3: add BADD-specific values ALSA: usb: add UAC3 BADD profiles support include/linux/usb/audio-v3.h | 26 ++ sound/usb/card.c | 53 ++-- sound/usb/clock.c | 9 +- sound/usb/mixer.c | 351 +++++++++++++++++++--- sound/usb/mixer_maps.c | 65 ++++ sound/usb/stream.c | 687 +++++++++++++++++++++++++------------------ sound/usb/usbaudio.h | 2 + 7 files changed, 849 insertions(+), 344 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v2] ALSA: hda - Add missing HD-Audio codec information
by Shrikant Dharwadkar 13 May '18

13 May '18

This adds missing HD-Audio codec information such as GPIO info, power controls, volume capabilities from HDA spec. These additional info can be useful in debugging issues. Following are the info that are being added: Function group nodes -Add power state controls -Add subordinate node count parameters -Add function group type information -Add function group capabilities -Add unsolicited response controls -Add input converter SDI select information -Add GPIO enable control -Add GPIO direction information -Add GPIO wake enable information -Add GPIO unsolicited control information -Add GPIO sticky mask data -Add GPIO data sets associated with GPIO signals Widget nodes -Add pin widget controls -Add power state controls -Add converter stream and channel controls -Add processing state information -Add processing capabilities -Add volume knob capabilities -Add volume knob control information -Add digital converter control information -Add pin sense control information -Add EAPD/BTL enable controls -Add unsolicited response controls Signed-off-by: Shrikant Dharwadkar <shrikant.dharwadkar(a)intel.com> --- sound/hda/hdac_sysfs.c | 167 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 167 insertions(+) diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index fb2aa34..b1acff2 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -37,6 +37,9 @@ static ssize_t type##_show(struct device *dev, \ } \ static DEVICE_ATTR_RO(type) +#define CODEC_PRINT(verb) sprintf(buf, "0x%08x\n", \ + snd_hdac_read_parm(codec, nid, verb)) + CODEC_ATTR(type); CODEC_ATTR(vendor_id); CODEC_ATTR(subsystem_id); @@ -173,6 +176,12 @@ static ssize_t pin_caps_show(struct hdac_device *codec, hda_nid_t nid, snd_hdac_read_parm(codec, nid, AC_PAR_PIN_CAP)); } +static ssize_t pin_widget_ctl_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_PIN_WIDGET_CONTROL); +} + static ssize_t pin_cfg_show(struct hdac_device *codec, hda_nid_t nid, struct widget_attribute *attr, char *buf) { @@ -243,6 +252,14 @@ static ssize_t power_caps_show(struct hdac_device *codec, hda_nid_t nid, snd_hdac_read_parm(codec, nid, AC_PAR_POWER_STATE)); } +static ssize_t power_states_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + if (nid != codec->afg && !(get_wcaps(codec, nid) & AC_WCAP_POWER)) + return 0; + return CODEC_PRINT(AC_VERB_GET_POWER_STATE); +} + static ssize_t gpio_caps_show(struct hdac_device *codec, hda_nid_t nid, struct widget_attribute *attr, char *buf) { @@ -266,27 +283,166 @@ static ssize_t connections_show(struct hdac_device *codec, hda_nid_t nid, return ret; } +static ssize_t converter_stream_channel_show(struct hdac_device *codec, + hda_nid_t nid, struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_CONV); +} + +static ssize_t processing_states_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_PROC_STATE); +} + +static ssize_t subordinate_node_count_show(struct hdac_device *codec, + hda_nid_t nid, struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_PAR_NODE_COUNT); +} + +static ssize_t function_group_type_show(struct hdac_device *codec, + hda_nid_t nid, struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_PAR_FUNCTION_TYPE); +} + +static ssize_t processing_caps_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_PAR_PROC_CAP); +} + +static ssize_t volume_knob_caps_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_PAR_VOL_KNB_CAP); +} + +static ssize_t volume_knob_ctl_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_VOLUME_KNOB_CONTROL); +} + +static ssize_t digital_converter_ctl_show(struct hdac_device *codec, + hda_nid_t nid, struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_DIGI_CONVERT_1); +} + +static ssize_t pin_sense_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_PIN_SENSE); +} + +static ssize_t EAPD_BTL_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_EAPD_BTLENABLE); +} + +static ssize_t unsol_caps_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_UNSOLICITED_RESPONSE); +} + +static ssize_t sdi_select_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_SDI_SELECT); +} + +static ssize_t gpio_enable_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_MASK); +} + +static ssize_t gpio_direction_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_DIRECTION); +} + +static ssize_t gpio_wake_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_WAKE_MASK); +} + +static ssize_t gpio_unsol_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK); +} + +static ssize_t gpio_sticky_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_STICKY_MASK); +} + +static ssize_t gpio_data_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_DATA); +} + static WIDGET_ATTR_RO(caps); static WIDGET_ATTR_RO(pin_caps); static WIDGET_ATTR_RO(pin_cfg); +static WIDGET_ATTR_RO(pin_widget_ctl); static WIDGET_ATTR_RO(pcm_caps); static WIDGET_ATTR_RO(pcm_formats); static WIDGET_ATTR_RO(amp_in_caps); static WIDGET_ATTR_RO(amp_out_caps); static WIDGET_ATTR_RO(power_caps); +static WIDGET_ATTR_RO(power_states); static WIDGET_ATTR_RO(gpio_caps); static WIDGET_ATTR_RO(connections); +static WIDGET_ATTR_RO(converter_stream_channel); +static WIDGET_ATTR_RO(processing_states); +static WIDGET_ATTR_RO(subordinate_node_count); +static WIDGET_ATTR_RO(function_group_type); +static WIDGET_ATTR_RO(processing_caps); +static WIDGET_ATTR_RO(volume_knob_caps); +static WIDGET_ATTR_RO(volume_knob_ctl); +static WIDGET_ATTR_RO(digital_converter_ctl); +static WIDGET_ATTR_RO(pin_sense); +static WIDGET_ATTR_RO(EAPD_BTL); +static WIDGET_ATTR_RO(unsol_caps); +static WIDGET_ATTR_RO(sdi_select); +static WIDGET_ATTR_RO(gpio_enable); +static WIDGET_ATTR_RO(gpio_direction); +static WIDGET_ATTR_RO(gpio_wake); +static WIDGET_ATTR_RO(gpio_unsol); +static WIDGET_ATTR_RO(gpio_sticky); +static WIDGET_ATTR_RO(gpio_data); static struct attribute *widget_node_attrs[] = { &wid_attr_caps.attr, &wid_attr_pin_caps.attr, + &wid_attr_pin_widget_ctl.attr, &wid_attr_pin_cfg.attr, &wid_attr_pcm_caps.attr, &wid_attr_pcm_formats.attr, &wid_attr_amp_in_caps.attr, &wid_attr_amp_out_caps.attr, &wid_attr_power_caps.attr, + &wid_attr_power_states.attr, &wid_attr_connections.attr, + &wid_attr_converter_stream_channel.attr, + &wid_attr_processing_states.attr, + &wid_attr_processing_caps.attr, + &wid_attr_volume_knob_caps.attr, + &wid_attr_volume_knob_ctl.attr, + &wid_attr_digital_converter_ctl.attr, + &wid_attr_pin_sense.attr, + &wid_attr_EAPD_BTL.attr, + &wid_attr_unsol_caps.attr, NULL, }; @@ -296,7 +452,18 @@ static struct attribute *widget_afg_attrs[] = { &wid_attr_amp_in_caps.attr, &wid_attr_amp_out_caps.attr, &wid_attr_power_caps.attr, + &wid_attr_power_states.attr, &wid_attr_gpio_caps.attr, + &wid_attr_subordinate_node_count.attr, + &wid_attr_function_group_type.attr, + &wid_attr_unsol_caps.attr, + &wid_attr_sdi_select.attr, + &wid_attr_gpio_enable.attr, + &wid_attr_gpio_direction.attr, + &wid_attr_gpio_wake.attr, + &wid_attr_gpio_unsol.attr, + &wid_attr_gpio_sticky.attr, + &wid_attr_gpio_data.attr, NULL, }; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: hisilicon: fix spelling mistake: "uknown" -> "unknown"
by Colin King 13 May '18

13 May '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err message text Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/hisilicon/hi6210-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 07a57209e055..53344a3b7a60 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -498,7 +498,7 @@ static int hi6210_i2s_trigger(struct snd_pcm_substream *substream, int cmd, hi6210_i2s_txctrl(cpu_dai, 0); break; default: - dev_err(cpu_dai->dev, "uknown cmd\n"); + dev_err(cpu_dai->dev, "unknown cmd\n"); return -EINVAL; } return 0; -- 2.17.0

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[alsa-devel] [PATCH] ASoC: Intel: bytcr_rt5640: Fix compile error
by Hans de Goede 13 May '18

13 May '18

Fix the compile error introduced by: "ASoC: Intel: bytcr_rt5640: Configure PLL1 before using it". Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Mark, feel free to squash this into the original commit --- sound/soc/intel/boards/bytcr_rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 493240194991..828be866fb00 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -210,14 +210,14 @@ static int byt_rt5640_prepare_and_enable_pll1(struct snd_soc_dai *codec_dai, } if (ret < 0) { - dev_err(codec_dai->codec->dev, "can't set pll: %d\n", ret); + dev_err(codec_dai->component->dev, "can't set pll: %d\n", ret); return ret; } ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1, rate * 512, SND_SOC_CLOCK_IN); if (ret < 0) { - dev_err(codec_dai->codec->dev, "can't set clock %d\n", ret); + dev_err(codec_dai->component->dev, "can't set clock %d\n", ret); return ret; } -- 2.17.0

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[alsa-devel] What is the meaning of #sound-dai-cells in a device tree?
by sdaau 11 May '18

11 May '18

Hi all, Apologies if this isn't the right forum (my thinking was, is that it is related to sound, and thus ALSA, more) - however, if it isn't, please let me know where I could forward this query (maybe devicetree mailing list ?) I have tried, but I really cannot find any documentation where I can read about, what does conceptually #sound-dai-cells in a device tree mean. Most I've found so far, is: linux_git$ grep -r 'sound-dai-cells' --exclude-dir='.git' arch/arm/boot/dts/sun7i-a20-cubietruck.dts: #sound-dai-cells = <0>; ... arch/arm/boot/dts/r8a7790.dtsi: * #sound-dai-cells is required arch/arm/boot/dts/r8a7790.dtsi: * Single DAI : #sound-dai-cells = <0>; <&rcar_sound>; arch/arm/boot/dts/r8a7790.dtsi: * Multi DAI : #sound-dai-cells = <1>; <&rcar_sound N>; ... Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.txt:- #sound-dai-cells: Should be set to 1 (for multi-dai) ... As far as I can tell: * if it is 1, then you can refer to "ports" (?) of a soundcard node, i.e. <&rcar_sound N>; * if it is 0, then you cannot refer to "ports", and you can only refer to <&rcar_sound> Also, it seems to appear only in these kernel source files: drivers/misc/atmel-ssc.c: if (!of_property_read_bool(np, "#sound-dai-cells")) drivers/misc/atmel-ssc.c: if (of_property_read_bool(ssc->pdev->dev.of_node, "#sound-dai-cells")) ... sound/soc/generic/simple-card.c:#define CELL "#sound-dai-cells" ... sound/soc/generic/simple-scu-card.c:#define CELL "#sound-dai-cells" ... sound/soc/soc-core.c: "#sound-dai-cells", 0, &args); sound/soc/soc-core.c: "#sound-dai-cells"); sound/soc/soc-core.c: "#sound-dai-cells", ... so, I guess, it only has meaning in what would be nodes under `target = <&sound>;` in a device tree - is that correct? But then - what is sound-dai-cells actually describing? Number of entries in `static struct snd_soc_dai_driver` of a (codec?) driver, if it is defined as an array? Or a number of DAI links? Could anyone offer a more proper conceptual description of what #sound-dai-cells is supposed to describe? Many thanks in advance for any pointers, s.

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[alsa-devel] [PATCH v6 00/13] soundwire: Add stream support
by Vinod Koul 11 May '18

11 May '18

This series adds support in SoundWire subsystem for: - Documentation for stream support - stream management - data port management - DAI ops in cadence and Intel drivers - ASoC API to propagate SDW stream Updates in v6: - Fix return error as pointed by Sanyog - Update stream state enum documentation as pointed by Pierre and few other nits Updated in v5: - Add todo for device-device changes in future and some comment updates based on feedback from Pierre Updates in v4: - Add more comments for sdw_release_master_stream() - Make sure ports are freed upon sdw_stream_remove_slave() - Clarify snd_soc_dai_set_sdw_stream() and setting of NULL upon release. Updates in v3: - Remove stream state transition checks and redefine stream states - Update bansk switch routines - Other nitpicks pointed out by Pierre. Updates in v2: - Make ASoC API inlined - Make stream states as states and not action - Update the direction enum - Fix some typos and comment updates *** SUBJECT HERE *** *** BLURB HERE *** Sanyog Kale (7): Documentation: soundwire: Add more documentation soundwire: Add support for SoundWire stream management soundwire: Add support for port management soundwire: Add Master and Slave port programming soundwire: Add helpers for ports operations soundwire: Add bank switch routine soundwire: Add stream configuration APIs Shreyas NC (2): ASoC: Add SoundWire stream programming interface soundwire: Remove cdns_master_ops Vinod Koul (4): soundwire: cdns: Add port routines soundwire: cdns: Add stream routines soundwire: intel: Add stream initialization soundwire: intel: Add audio DAI ops .../driver-api/soundwire/error_handling.rst | 65 + Documentation/driver-api/soundwire/index.rst | 3 + Documentation/driver-api/soundwire/locking.rst | 106 ++ Documentation/driver-api/soundwire/stream.rst | 372 +++++ drivers/soundwire/Kconfig | 2 +- drivers/soundwire/Makefile | 2 +- drivers/soundwire/bus.c | 43 + drivers/soundwire/bus.h | 72 + drivers/soundwire/cadence_master.c | 451 +++++- drivers/soundwire/cadence_master.h | 151 ++ drivers/soundwire/intel.c | 525 ++++++- drivers/soundwire/intel.h | 4 + drivers/soundwire/intel_init.c | 3 + drivers/soundwire/stream.c | 1479 ++++++++++++++++++++ include/linux/soundwire/sdw.h | 332 ++++- include/linux/soundwire/sdw_intel.h | 14 + include/sound/soc-dai.h | 23 + 17 files changed, 3633 insertions(+), 14 deletions(-) create mode 100644 Documentation/driver-api/soundwire/error_handling.rst create mode 100644 Documentation/driver-api/soundwire/locking.rst create mode 100644 Documentation/driver-api/soundwire/stream.rst create mode 100644 drivers/soundwire/stream.c -- 2.7.4

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Only disable inactive serializer
by Vishal Thanki 11 May '18

11 May '18

As a side effect of the following commit, the active TX serializer may get disabled which may result in distorted audio output. ASoC: davinci-mcasp: Add support for multichannel playback (2952b27e2e463b28d5c0f04000f96b968137ca42) For example, if a 4 channel I2S playback with two TX serializers is activated. Later on, if a recording of 2 channels, with only 1 RX serializer is started, which will also disable one of the TX serializer because max_active_serializers is only calculated for RX (recording) stream. This patch fixes this issue. Signed-off-by: Vishal Thanki <vishalthanki(a)gmail.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 79fc4aa..1f96c9d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -789,7 +789,7 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, rx_ser < max_active_serializers) { mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; - } else { + } else if (mcasp->serial_dir[i] == INACTIVE_MODE) { mcasp_mod_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), SRMOD_INACTIVE, SRMOD_MASK); } -- 2.7.4

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[alsa-devel] [asoc:for-4.18 139/159] sound/soc/intel/boards/bytcr_rt5640.c:177:20: error: 'struct snd_soc_dai' has no member named 'codec'
by kbuild test robot 11 May '18

11 May '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.18 head: 89027d9ebb5948392dd5a4ccc72d5a4eaa865537 commit: bcd9a325f0b0f407c4559779a94e802977c67274 [139/159] ASoC: Intel: bytcr_rt5640: Configure PLL1 before using it config: x86_64-federa-25 (attached as .config) compiler: gcc-7 (Debian 7.3.0-16) 7.3.0 reproduce: git checkout bcd9a325f0b0f407c4559779a94e802977c67274 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): sound/soc/intel/boards/bytcr_rt5640.c: In function 'byt_rt5640_prepare_and_enable_pll1': >> sound/soc/intel/boards/bytcr_rt5640.c:177:20: error: 'struct snd_soc_dai' has no member named 'codec' dev_err(codec_dai->codec->dev, "can't set pll: %d\n", ret); ^~ sound/soc/intel/boards/bytcr_rt5640.c:184:20: error: 'struct snd_soc_dai' has no member named 'codec' dev_err(codec_dai->codec->dev, "can't set clock %d\n", ret); ^~ vim +177 sound/soc/intel/boards/bytcr_rt5640.c 143 144 static int byt_rt5640_prepare_and_enable_pll1(struct snd_soc_dai *codec_dai, 145 int rate) 146 { 147 int ret; 148 149 /* Configure the PLL before selecting it */ 150 if (!(byt_rt5640_quirk & BYT_RT5640_MCLK_EN)) { 151 /* use bitclock as PLL input */ 152 if ((byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) || 153 (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) { 154 /* 2x16 bit slots on SSP0 */ 155 ret = snd_soc_dai_set_pll(codec_dai, 0, 156 RT5640_PLL1_S_BCLK1, 157 rate * 32, rate * 512); 158 } else { 159 /* 2x15 bit slots on SSP2 */ 160 ret = snd_soc_dai_set_pll(codec_dai, 0, 161 RT5640_PLL1_S_BCLK1, 162 rate * 50, rate * 512); 163 } 164 } else { 165 if (byt_rt5640_quirk & BYT_RT5640_MCLK_25MHZ) { 166 ret = snd_soc_dai_set_pll(codec_dai, 0, 167 RT5640_PLL1_S_MCLK, 168 25000000, rate * 512); 169 } else { 170 ret = snd_soc_dai_set_pll(codec_dai, 0, 171 RT5640_PLL1_S_MCLK, 172 19200000, rate * 512); 173 } 174 } 175 176 if (ret < 0) { > 177 dev_err(codec_dai->codec->dev, "can't set pll: %d\n", ret); 178 return ret; 179 } 180 181 ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1, 182 rate * 512, SND_SOC_CLOCK_IN); 183 if (ret < 0) { 184 dev_err(codec_dai->codec->dev, "can't set clock %d\n", ret); 185 return ret; 186 } 187 188 return 0; 189 } 190 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 1/4] ASoC: compress: Only assign compr->ops->copy once
by Charles Keepax 11 May '18

11 May '18

There are only one set of ops on the compressed stream so no need to reassign the copy callback repeatedly, stop after copy is seen to be necessary. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/soc-compress.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index ba56f87f96d4c..62875c6a93a14 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -973,6 +973,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) continue; compr->ops->copy = soc_compr_copy; + break; } -- 2.11.0

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