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

[alsa-devel] [PATCHv5 0/5] Motorola Droid 4 Audio Support
by Sebastian Reichel 04 Apr '18

04 Apr '18

Hi, This adds audio support to Motorola Droid 4. I dropped the regulator from the DT binding as requested by Mark. Tested: - Playing via both DACs using Speaker, Earpiece, Headphone - Recording using internal Mic - Volume Controls Known not to work: - 3.5mm detection support (requires closed source firmware, needs further investigation) - Modem / Bluetooth Audio (this needs something more sophisticated than the audio-graph-card driver. I will have a look once the basic support has been merged) Changes since PATCHv4: * Replace verbose GPL text with SPDX header * Use "GPL v2" as MODULE_LICENSE * Drop "default MFD_CPCAP" for the codec driver * Use SOC_SINGLE instead of SOC_ENUM for Phase invert switches * Find DT node by node name instead of by compatible property - Drop Acked-by from Rob Herring * Add new patch for MFD driver to implement the mentioned DT change Changes since PATCHv3: * Drop regulator from DT binding, request VAUDIO from global regulator namespace instead Changes since PATCHv2: * Fix a whitespace issue * Fix const notes Takashi provided * Fix a DAPM route issue I accidently introduced in v2 Changes since PATCHv1: * Add patch from Tony exporting soc_dpcm_runtime_update * Integrate DT patch for vaudio initial mode * Split dt-binding from codec patch and add Ack from Rob * Fix CPCAP position in Kconfig/Makefile * Avoid "err +=" constructs * Simplify reset function * Drop cpcap_audio_read/write helpers * Do not use tertiary operator for mute register value * Update Input Mux logic * Switch from simple-audio-card to audio-graph-card -- Sebastian Sebastian Reichel (5): dt-bindings: mfd: motorola-cpcap: document audio-codec ASoC: codec: cpcap: new codec mfd: motorola-cpcap: Add audio-codec support ARM: dts: motorola-cpcap-mapphone: add audio-codec ARM: dts: omap4-droid4: add soundcard .../devicetree/bindings/mfd/motorola-cpcap.txt | 42 + arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 13 + arch/arm/boot/dts/omap4-droid4-xt894.dts | 78 + drivers/mfd/motorola-cpcap.c | 51 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cpcap.c | 1568 ++++++++++++++++++++ 7 files changed, 1757 insertions(+), 1 deletion(-) create mode 100644 sound/soc/codecs/cpcap.c -- 2.16.1

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[alsa-devel] Disable DAPM for click avoidance ?
by Ricard Wanderlof 27 Mar '18

27 Mar '18

I'm having an issue with a codec where the normal DAPM procedure of disabling most of the codec when there is no stream ongoing results in large clicks when a playback stream starts and stops, because the output jumps from 0V when the output is disabled to over a volt when the output is enabled. There appears to be a 'slow stop/start' feature in the codec, however, it takes about one second to ramp up and down the output voltage, and that is too long a delay when starting a stream. In our usecase the power consumtion of the codec is negligable compared to the rest of the electronics, so leaving the codec running at all times is perfectly feasible. Is there any way from a machine driver to disable the DAPM switching, i.e. leave parts of the codec running at all times? In this specific case, there are two internal PM controls which enable the output. One of them is controlled by an output mixer element (SND_SOC_DAPM_MIXER_E), and the other one from set_bias_level(). I've had a look in the core code, but it seems that the whole business of changing the bias level when a stream is started is always performed and that there is no way to say "Set the bias level to ON and leave it there". An alternative would be to augment the codec driver with an 'always on' feature, which effectively always enables PM for certain parts of the chip. But it seems like the wrong place to do this type of thing. Another use for this feature is that we have had a lot of problems over the years with buggy I2S interfaces misbehaving when streams are started and stopped, and there would be an advantage to letting the AIF run the whole time after startup. /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] [PATCH - JACK PCM plugin] jack: Do not Xrun the ALSA buffer
by twischer＠de.adit-jv.com 21 Mar '18

21 Mar '18

From: Timo Wischer <twischer(a)de.adit-jv.com> when the JACK thread is requesting too many audio frames Playback: Without this commit the ALSA audio buffer will be played with endless repeats as long as the user application has not provided new audio data. Therefore this garbage will be played as long as the user application has not called snd_pcm_stop() after an Xrun. With this fix the rest of the JACK buffer will be filled with silence. Capture: Without this commit the audio data in the ALSA buffer would be overwritten. With this commit the new data from the JACK buffer will not be copied. Therefore the existing data in the ALSA buffer will not be overwritten. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- This patch is depending on http://mailman.alsa-project.org/pipermail/alsa-devel/2018-January/130986.ht… diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index c22a5d0..a4f35f4 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -54,6 +54,68 @@ typedef struct { static int snd_pcm_jack_stop(snd_pcm_ioplug_t *io); + +static inline snd_pcm_uframes_t snd_pcm_jack_playback_avail(const snd_pcm_ioplug_t* const io, + const snd_pcm_uframes_t hw_ptr, + const snd_pcm_uframes_t appl_ptr) +{ + const snd_pcm_jack_t* const jack = io->private_data; + + /* cannot use io->hw_ptr without calling snd_pcm_avail_update() + * because it is not guranteed that snd_pcm_jack_pointer() was already + * called + */ + snd_pcm_sframes_t avail; + avail = hw_ptr + io->buffer_size - appl_ptr; + if (avail < 0) + avail += jack->boundary; + else if ((snd_pcm_uframes_t) avail >= jack->boundary) + avail -= jack->boundary; + return avail; +} + +static inline snd_pcm_uframes_t snd_pcm_jack_capture_avail(const snd_pcm_ioplug_t* const io, + const snd_pcm_uframes_t hw_ptr, + const snd_pcm_uframes_t appl_ptr) +{ + const snd_pcm_jack_t* const jack = io->private_data; + + /* cannot use io->hw_ptr without calling snd_pcm_avail_update() + * because it is not guranteed that snd_pcm_jack_pointer() was already + * called + */ + snd_pcm_sframes_t avail; + avail = hw_ptr - appl_ptr; + if (avail < 0) + avail += jack->boundary; + return avail; +} + +static inline snd_pcm_uframes_t snd_pcm_jack_avail(const snd_pcm_ioplug_t* const io, + const snd_pcm_uframes_t hw_ptr, + const snd_pcm_uframes_t appl_ptr) +{ + return (io->stream == SND_PCM_STREAM_PLAYBACK) ? + snd_pcm_jack_playback_avail(io, hw_ptr, appl_ptr) : + snd_pcm_jack_capture_avail(io, hw_ptr, appl_ptr); +} + +static inline snd_pcm_uframes_t snd_pcm_jack_hw_avail(const snd_pcm_ioplug_t* const io, + const snd_pcm_uframes_t hw_ptr, + const snd_pcm_uframes_t appl_ptr) +{ + /* available data/space which can be transfered by the user application */ + const snd_pcm_uframes_t user_avail = snd_pcm_jack_avail(io, hw_ptr, + appl_ptr); + + if (user_avail > io->buffer_size) { + /* there was an Xrun */ + return 0; + } + /* available data/space which can be transfered by the DMA */ + return io->buffer_size - user_avail; +} + static int pcm_poll_block_check(snd_pcm_ioplug_t *io) { static char buf[32]; @@ -154,7 +216,6 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) { snd_pcm_jack_t *jack = io->private_data; snd_pcm_uframes_t hw_ptr; - const snd_pcm_channel_area_t *areas; snd_pcm_uframes_t xfer = 0; unsigned int channel; @@ -164,40 +225,57 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) jack->areas[channel].first = 0; jack->areas[channel].step = jack->sample_bits; } - - if (io->state != SND_PCM_STATE_RUNNING) { - if (io->stream == SND_PCM_STREAM_PLAYBACK) { - for (channel = 0; channel < io->channels; channel++) - snd_pcm_area_silence(&jack->areas[channel], 0, nframes, io->format); - return 0; - } - } hw_ptr = jack->hw_ptr; - areas = snd_pcm_ioplug_mmap_areas(io); - - while (xfer < nframes) { - snd_pcm_uframes_t frames = nframes - xfer; - snd_pcm_uframes_t offset = hw_ptr % io->buffer_size; - snd_pcm_uframes_t cont = io->buffer_size - offset; - - if (cont < frames) - frames = cont; + if (io->state == SND_PCM_STATE_RUNNING) { + const snd_pcm_channel_area_t *areas = snd_pcm_ioplug_mmap_areas(io); + + while (xfer < nframes) { + snd_pcm_uframes_t frames = nframes - xfer; + snd_pcm_uframes_t offset = hw_ptr % io->buffer_size; + snd_pcm_uframes_t cont = io->buffer_size - offset; + snd_pcm_uframes_t hw_avail = + snd_pcm_jack_hw_avail(io, hw_ptr, + io->appl_ptr); + + /* stop copying if there is no more data available */ + if (hw_avail <= 0) + break; + + /* split the snd_pcm_area_copy() function into two parts + * if the data to copy passes the buffer wrap around + */ + if (cont < frames) + frames = cont; + + if (hw_avail < frames) + frames = hw_avail; + + for (channel = 0; channel < io->channels; channel++) { + if (io->stream == SND_PCM_STREAM_PLAYBACK) + snd_pcm_area_copy(&jack->areas[channel], xfer, &areas[channel], offset, frames, io->format); + else + snd_pcm_area_copy(&areas[channel], offset, &jack->areas[channel], xfer, frames, io->format); + } - for (channel = 0; channel < io->channels; channel++) { - if (io->stream == SND_PCM_STREAM_PLAYBACK) - snd_pcm_area_copy(&jack->areas[channel], xfer, &areas[channel], offset, frames, io->format); - else - snd_pcm_area_copy(&areas[channel], offset, &jack->areas[channel], xfer, frames, io->format); + hw_ptr += frames; + if (hw_ptr >= jack->boundary) + hw_ptr -= jack->boundary; + xfer += frames; } - - hw_ptr += frames; - if (hw_ptr >= jack->boundary) - hw_ptr -= jack->boundary; - xfer += frames; } jack->hw_ptr = hw_ptr; + /* check if requested frames were copied */ + if (xfer < nframes) { + /* always fill the not yet written JACK buffer with silence */ + if (io->stream == SND_PCM_STREAM_PLAYBACK) { + const snd_pcm_uframes_t samples = nframes - xfer; + for (channel = 0; channel < io->channels; channel++) + snd_pcm_area_silence(&jack->areas[channel], xfer, samples, io->format); + } + } + pcm_poll_unblock_check(io); /* unblock socket for polling if needed */ return 0; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: hdac_hdmi: Program pin-port-mux during jack report event
by Guneshwor Singh 19 Mar '18

19 Mar '18

From: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> During d3/d0 cycle, the connection selection index of all pins points to the default value. This needs to be restored to ensure audio is restored after d3/d0 cycle. So store the connection selection index and program it during jack report event which gets invoked in cases like d3/d0 cycle, hot plug detection when multiple displays are connected. Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale(a)intel.com> Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index dba6f4c5074a..cabcb75bf4fc 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -32,6 +32,7 @@ #include <sound/hda_chmap.h> #include "../../hda/local.h" #include "hdac_hdmi.h" +#include <sound/hda_regmap.h> #define NAME_SIZE 32 @@ -83,6 +84,7 @@ struct hdac_hdmi_pin { struct list_head head; hda_nid_t nid; bool mst_capable; + int conn_index; struct hdac_hdmi_port *ports; int num_ports; struct hdac_ext_device *edev; @@ -141,6 +143,9 @@ struct hdac_hdmi_priv { #define hdev_to_hdmi_priv(_hdev) ((to_ehdac_device(_hdev))->private_data) +static int hdac_hdmi_port_select_set(struct hdac_ext_device *edev, + struct hdac_hdmi_port *port); + static struct hdac_hdmi_pcm * hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi, struct hdac_hdmi_cvt *cvt) @@ -159,6 +164,7 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, struct hdac_hdmi_port *port, bool is_connect) { struct hdac_ext_device *edev = port->pin->edev; + int cmd, err; if (is_connect) snd_soc_dapm_enable_pin(port->dapm, port->jack_pin); @@ -166,6 +172,29 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, snd_soc_dapm_disable_pin(port->dapm, port->jack_pin); if (is_connect) { + /* set the device if pin is mst_capable */ + if (hdac_hdmi_port_select_set(edev, port) < 0) { + dev_err(&edev->hdev.dev, + "port %d device select fail\n", port->id); + return; + } + /* + * Restore the connection selection index of the + * respective pin. + */ + if (port->pin->conn_index > 0) { + cmd = snd_hdac_regmap_encode_verb(port->pin->nid, + AC_VERB_SET_CONNECT_SEL); + err = snd_hdac_regmap_write_raw(&edev->hdev, cmd, + port->pin->conn_index - 1); + if (err < 0) { + dev_err(&edev->hdev.dev, + "pin %d conn select index fail %d\n", + port->pin->nid, err); + return; + } + } + /* * Report Jack connect event when a device is connected * for the first time where same PCM is attached to multiple @@ -891,6 +920,9 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol, } } + if (ucontrol->value.enumerated.item[0] > 0) + port->pin->conn_index = ucontrol->value.enumerated.item[0]; + /* * Jack status is not reported during device probe as the * PCMs are not registered by then. So report it here. -- 2.16.1

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[alsa-devel] [BUG] Kernel crash on Allwinner H3 due to sound core changes
by Jernej Škrabec 19 Mar '18

19 Mar '18

Hi all, with todays linux-next (next-20180228), kernel on Allwinner H3 SoC crashes with dmesg like that: https://pastebin.com/raw/0D5JeaJ8 I bisected the kernel and first offending commit is: be7ee5f32a9a ("ASoC: soc-generic-dmaengine-pcm: replace platform to component") I know that crash message is completely unrelated to sound subsystem, but it turns out that if I disable CONFIG_SND_SUN4I_CODEC kernel works ok, but this way I lose analog audio output. Any suggestions what can be the issue? Best regards, Jernej

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[alsa-devel] [PATCH v1 0/6] ALSA: usb: UAC3. Add support for Basic Audio Device (BADD)
by Jorge Sanjuan 19 Mar '18

19 Mar '18

This adds functionality for the Basic Audio Device (BADD) subset that is defined in the USB Audio Class 3 (UAC3). In a BADD compliant driver, all Class Specific descriptors are inferred and so they should all be generated in the *host* as the device wont expose those. This also applies to the new HIGH CAPABILITY descriptor as the GET request is not to be supported in a BADD compliant driver. The new class requires the device to have 3 usb configurations as follows: 1: Legacy Mode: UAC1 or UAC2. 2: BADD device with a prefined topology. (Minimum). 3: UAC3 device for more detailed description or more complex devices that can't be covered by the BADD profile. This patch series also includes some minor fixes to sound/usb. It has been tested with an actual UAC3 compliant device in different host machines and all configs (UAC1/2, BADD and UAC3) work. Also, this has been implemented on top of the the patch which adds UAC3 support to the usb sound card driver: commit ddd452d7b04b86fb5f9285a19ac54deca9264ac1 Author: Ruslan Bilovol <ruslan.bilovol(a)gmail.com> Date: Tue Nov 7 04:01:20 2017 +0200 Jorge Sanjuan (6): usb: audio: Add bi-directional terminal types. usb: audio: Fix variable length field to be variable. ALSA: usb: UAC3. Add initial mixer support. ALSA: usb: Only get AudioControl header for UAC1 class. ALSA: usb: Use Class Specific EP for UAC3 devices. ALSA: usb: ADC3: Add initial BADD spec support include/linux/usb/audio-v3.h | 2 +- include/uapi/linux/usb/audio.h | 21 +- sound/usb/Makefile | 3 +- sound/usb/badd.c | 518 +++++++++++++++++++++++++++++++++++++++++ sound/usb/badd.h | 14 ++ sound/usb/card.c | 25 +- sound/usb/mixer.c | 24 ++ sound/usb/stream.c | 165 ++++++++----- sound/usb/usbaudio.h | 1 + 9 files changed, 702 insertions(+), 71 deletions(-) create mode 100644 sound/usb/badd.c create mode 100644 sound/usb/badd.h -- 2.11.0

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[alsa-devel] [PATCH 0/2] TDA7419 audio processor driver
by Matt Porter 18 Mar '18

18 Mar '18

This series adds an ASoC component driver for the ST TDA7419 audio processor which is commonly used in automotive audio applications. The datasheet can be found at http://www.st.com/resource/en/datasheet/tda7419.pdf Matt Porter (2): ASoC: add tda7419 audio processor binding ASoC: add tda7419 audio processor driver .../devicetree/bindings/sound/tda7419.txt | 15 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda7419.c | 571 +++++++++++++++++++++ 4 files changed, 594 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tda7419.txt create mode 100644 sound/soc/codecs/tda7419.c -- 2.11.0

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[alsa-devel] [PATCH v2] sound: omap-twl4030: do not complain on -EPROBE_DEFER when initializing
by Ladislav Michl 14 Mar '18

14 Mar '18

Don't complain on -EPROBE_DEFER when initializing, the driver probe will be retried later. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- Changes: -v2: fixed typo in commit log, added Jarkko Nikula's ack sound/soc/omap/omap-twl4030.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index cccc316743fa..7c1f71884768 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -322,8 +322,10 @@ static int omap_twl4030_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, priv); ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, "devm_snd_soc_register_card() failed: %d\n", - ret); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "devm_snd_soc_register_card() failed: %d\n", + ret); return ret; } -- 2.16.1

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[alsa-devel] [Xen-devel][PATCH 0/2] sndif: add explicit back and front synchronization
by Oleksandr Andrushchenko 14 Mar '18

14 Mar '18

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> Hi, all! Foreword ======== This change is aimed to add support for explicit back and front synchronization during playback and capture in response to comments raised during upstream attempt of the para-virtualized sound frontend driver for Xen [1], [2] and gather opinions from the relevant communities (ALSA, Xen) on the change. The relevant backend is implemented as a user-space application [3] and uses accompanying helper library [4]. Both frontend driver and backend were tested on real HW running Xen hypervisor (Renesas R-Car ARM based H3/M3 boards, x86) to make sure the proposed solution does work. Rationale ========= During the first attempt to upstream the Linux front driver [5] number of comments and concerns were raised, one of the biggest flaws in the design were questioned by both Clemens Ladisch [6] and Takashi Sakamoto [7]: the absence of synchronization between frontend and backend during capture/playback. Two options were discussed: “In design of ALSA PCM core, drivers are expected to synchronize to actual hardwares for semi-realtime data transmission. The synchronization is done by two points: 1) Interrupts to respond events from actual hardwares. 2) Positions of actual data transmission in any serial sound interfaces of actual hardwares. “ and finally a change to the existing protocol was suggested: “In 'include/xen/interface/io/sndif.h', there's no functionalities I described the above: 1. notifications from DomU to Dom0 about the size of period for interrupts from actual hardwares. Or no way from Dom0 to DomU about the configured size of the period. 2. notifications of the interrupts from actual hardwares to DomU.” This is implemented as a change to the sndif protocol and allows removing period emulation: 1. Introduced a new event channel from back to front 2. New event with number of bytes played/captured (XENSND_EVT_CUR_POS, to be used for sending snd_pcm_period_elapsed at frontend (in Linux implementation). Sent in bytes, not frames to make the protocol generic and consistent) 3. New request for playback/capture control (XENSND_OP_TRIGGER) with start/pause/stop/resume sub-ops 4. Playback/capture buffer size is set on the backend side via XENSND_FIELD_BUFFER_SIZE XenStore entry Waiting for your valuable comments, Thank you, Oleksandr [1] https://github.com/andr2000/linux/commits/snd_upstream_v1 [2] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/inc… [3] https://github.com/xen-troops/snd_be [4] https://github.com/xen-troops/libxenbe [5] https://lkml.org/lkml/2017/8/7/363 [6] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-August/123617.html [7] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-August/123744.html Oleksandr Andrushchenko (2): sndif: introduce protocol version sndif: add explicit back and front synchronization xen/include/public/io/sndif.h | 173 +++++++++++++++++++++++++++++++++++++++++- 1 file changed, 170 insertions(+), 3 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: mt8173-rt5650: fix child-node lookup
by Johan Hovold 12 Mar '18

12 Mar '18

This driver used the wrong OF-helper when looking up the optional capture-codec child node during probe. Instead of searching just children of the sound node, a tree-wide depth-first search starting at the unrelated platform node was done. Not only could this end up matching an unrelated node or no node at all; the platform node could also be prematurely freed since of_find_node_by_name() drops a reference to its first argument. This particular pattern has been observed leading to crashes after probe deferrals in other drivers. Fix this by dropping the broken call to of_find_node_by_name() and keeping only the second, correct lookup using of_get_child_by_name() while taking care not to bail out if the optional node is missing. Note that this also addresses two capture-codec node-reference leaks (one for each of the original helper calls). Compile tested only. Fixes: d349caeb0510 ("ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver") Cc: PC Liao <pc.liao(a)mediatek.com> Signed-off-by: Johan Hovold <johan(a)kernel.org> --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index 40ebefd625c1..679fc8bea0a3 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -272,15 +272,10 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) } mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node; - if (of_find_node_by_name(platform_node, "codec-capture")) { - np = of_get_child_by_name(pdev->dev.of_node, "codec-capture"); - if (!np) { - dev_err(&pdev->dev, - "%s: Can't find codec-capture DT node\n", - __func__); - return -EINVAL; - } + np = of_get_child_by_name(pdev->dev.of_node, "codec-capture"); + if (np) { ret = snd_soc_of_get_dai_name(np, &codec_capture_dai); + of_node_put(np); if (ret < 0) { dev_err(&pdev->dev, "%s codec_capture_dai name fail %d\n", -- 2.13.6

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