February 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [sound:for-linus 23/27] (.init.text+0x134): multiple definition of `plat_irq_setup'; arch/sh/kernel/cpu/sh2/setup-sh7619.o:(.init.text+0x30): first defined here
by kbuild test robot 12 Feb '19

12 Feb '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: 00a399cad1a063e7665f06b6497a807db20441fd commit: c97617a81a7616d49bc3700959e08c6c6f447093 [23/27] ALSA: hda/ca0132 - Fix build error without CONFIG_PCI config: sh-allmodconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 8.2.0-11) 8.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout c97617a81a7616d49bc3700959e08c6c6f447093 # save the attached .config to linux build tree GCC_VERSION=8.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): sh4-linux-gnu-ld: arch/sh/boards/of-generic.o: in function `plat_irq_setup': >> (.init.text+0x134): multiple definition of `plat_irq_setup'; arch/sh/kernel/cpu/sh2/setup-sh7619.o:(.init.text+0x30): first defined here sh4-linux-gnu-ld: arch/sh/boards/of-generic.o: in function `arch_init_clk_ops': (.init.text+0x118): multiple definition of `arch_init_clk_ops'; arch/sh/kernel/cpu/sh2/clock-sh7619.o:(.init.text+0x0): first defined here --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [sound:for-linus 23/27] (.init.text+0x118): multiple definition of `arch_init_clk_ops'; arch/sh/kernel/cpu/sh2/clock-sh7619.o:(.init.text+0x0): first defined here
by kbuild test robot 12 Feb '19

12 Feb '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: 00a399cad1a063e7665f06b6497a807db20441fd commit: c97617a81a7616d49bc3700959e08c6c6f447093 [23/27] ALSA: hda/ca0132 - Fix build error without CONFIG_PCI config: sh-allmodconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 8.2.0-11) 8.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout c97617a81a7616d49bc3700959e08c6c6f447093 # save the attached .config to linux build tree GCC_VERSION=8.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): sh4-linux-gnu-ld: arch/sh/boards/of-generic.o: in function `plat_irq_setup': (.init.text+0x134): multiple definition of `plat_irq_setup'; arch/sh/kernel/cpu/sh2/setup-sh7619.o:(.init.text+0x30): first defined here sh4-linux-gnu-ld: arch/sh/boards/of-generic.o: in function `arch_init_clk_ops': >> (.init.text+0x118): multiple definition of `arch_init_clk_ops'; arch/sh/kernel/cpu/sh2/clock-sh7619.o:(.init.text+0x0): first defined here --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [RFC v1] tlv320aic32x4: delay i2c access by 1 ms after hardware reset
by Peter Seiderer 11 Feb '19

11 Feb '19

As stated in 'TLV320AIC3254 Application Reference Guide' ([1]): 3.2 Device Startup Lockout Times After the TLV320AIC3254 initializes through hardware reset at power-up or software reset, the internal registers initialize to default values. This initialization takes place within 1ms after pulling the RESET signal high. During this initialization phase, no register-read or register-write operation should be performed on ADC or DAC coefficient buffers. Also, no block within the codec should be powered up during the initialization phase. [1] http://www.ti.com/lit/an/slaa408a/slaa408a.pdf Signed-off-by: Peter Seiderer <ps.report(a)gmx.net> --- Note: This came up after the electronic departement took a deeper look at the i2c-bus/vcc/reset pins with a logic analyzer and the documentation because of a vcc/reset timing problem (now fixed on custom board). Not 100% sure if the 1 ms is needed (as restricted to ADC/DAC coefficent and codec block power up). --- sound/soc/codecs/tlv320aic32x4.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index e1bfba62fc08..b3beb0ca87c3 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -972,6 +972,8 @@ static int aic32x4_component_probe(struct snd_soc_component *component) gpio_set_value(aic32x4->rstn_gpio, 1); } + mdelay(1); + snd_soc_component_write(component, AIC32X4_RESET, 0x01); if (aic32x4->setup) -- 2.20.1

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[alsa-devel] pcm_dshare calculates size incorrectly on buffer rollover
by Brendan Shanks 11 Feb '19

11 Feb '19

I'm working on an embedded system based on an Ambarella H1 SoC (32-bit ARM Cortex A9). Audio playback through ALSA (and GStreamer) works fine when playing to the raw hw device. When playing through dshare (my normal configuration), playback stops after exactly 7 hours 16 minutes, for about 3 minutes. During the 3 minutes, the playback thread consumes an entire CPU core. After the 3 minutes, GStreamer reports an xrun, recovers from it, and playback goes back to normal. After some debugging, the problem seems to be in snd_pcm_dshare_sync_area(). When dshare->appl_ptr rolls over to 0, 'size' becomes huge, 3036676576. 'slave_size' is also much bigger than it should be, 1258291680. This is what 'size' is set to when the for loop starts, and I believe the for loop then spends ~3 minutes copying a huge amount of samples. This also explains the 7h16m time, it's linked to the PCM boundary which is 1258291200. At 48 kHz, 1258291200 samples takes 7h16m54s. I'm not sure what the fix should be though. Is this really a bug in dshare, or are bad values being set somewhere else? GStreamer? Or maybe the period/buffer size (480/9600) is causing problems? Any advice is appreciated, my modified snd_pcm_dshare_sync_area() and the output are below. Brendan Shanks $ more /proc/asound/card0/pcm1p/sub0/hw_params access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 4 rate: 48000 (48000/1) period_size: 480 buffer_size: 9600 $ more /proc/asound/card0/pcm1p/sub0/sw_params tstamp_mode: ENABLE period_step: 1 avail_min: 480 start_threshold: 1 stop_threshold: 1258291200 silence_threshold: 0 silence_size: 1258291200 boundary: 1258291200 --- output from running 'gst-launch-1.0 -e audiotestsrc ! alsasink' with prints included as below snd_pcm_dshare_sync_area size 3036676576, dshare->appl_ptr 0 dshare->last_appl_ptr 1258290720 slave_hw_ptr1 1258282080 slave_period_size 480 slave_hw_ptr2 1258282080 slave_buffer_size 9600 slave_hw_ptr3 1258291680 slave_boundary 1258291200 slave_hw_ptr4 1258291680 slave_appl_ptr 0 slave_size1 1258291680 size 1258291680 appl_ptr 9120 size 1258291680 boundary 1258291200 last_appl_ptr 0 slave_appl_ptr 0 dshare->slave_appl_ptr 480 exiting snd_pcm_dshare_sync_area size 3036676576, dshare->appl_ptr 0 dshare->last_appl_ptr 1258290720 slave_hw_ptr1 8965920 slave_period_size 480 slave_hw_ptr2 8965920 slave_buffer_size 9600 slave_hw_ptr3 8975520 slave_boundary 1258291200 slave_hw_ptr4 8975520 slave_appl_ptr 8975040 slave_size1 480 size 480 appl_ptr 9120 size 480 boundary 1258291200 last_appl_ptr 0 slave_appl_ptr 8640 dshare->slave_appl_ptr 8975520 exiting --- snd_pcm_dshare_sync_area() from alsa-lib-1.1.6 with prints added: static void snd_pcm_dshare_sync_area(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; snd_pcm_uframes_t slave_hw_ptr, slave_appl_ptr, slave_size; snd_pcm_uframes_t appl_ptr, size; const snd_pcm_channel_area_t *src_areas, *dst_areas; int print = 0; /* calculate the size to transfer */ size = dshare->appl_ptr - dshare->last_appl_ptr; if (! size) return; if (size > 9600) { printf("snd_pcm_dshare_sync_area size %lu, dshare->appl_ptr %lu dshare->last_appl_ptr %lu\n", size, dshare->appl_ptr, dshare->last_appl_ptr); print = 1; } slave_hw_ptr = dshare->slave_hw_ptr; if (print) printf("slave_hw_ptr1 %lu slave_period_size %lu", slave_hw_ptr, dshare->slave_period_size); /* don't write on the last active period - this area may be cleared * by the driver during write operation... */ slave_hw_ptr -= slave_hw_ptr % dshare->slave_period_size; if (print) printf("slave_hw_ptr2 %lu slave_buffer_size %lu", slave_hw_ptr, dshare->slave_buffer_size); slave_hw_ptr += dshare->slave_buffer_size; if (print) printf("slave_hw_ptr3 %lu slave_boundary %lu", slave_hw_ptr, dshare->slave_boundary); if (dshare->slave_hw_ptr > dshare->slave_boundary) slave_hw_ptr -= dshare->slave_boundary; if (print) printf("slave_hw_ptr4 %lu slave_appl_ptr %lu", slave_hw_ptr, dshare->slave_appl_ptr); if (slave_hw_ptr < dshare->slave_appl_ptr) slave_size = slave_hw_ptr + (dshare->slave_boundary - dshare->slave_appl_ptr); else slave_size = slave_hw_ptr - dshare->slave_appl_ptr; if (print) printf("slave_size1 %lu", slave_size); if (slave_size < size) size = slave_size; if (! size) return; if (print) printf("size %lu", size); /* add sample areas here */ src_areas = snd_pcm_mmap_areas(pcm); dst_areas = snd_pcm_mmap_areas(dshare->spcm); appl_ptr = dshare->last_appl_ptr % pcm->buffer_size; if (print) printf("appl_ptr %lu size %lu boundary %lu", appl_ptr, size, pcm->boundary); dshare->last_appl_ptr += size; dshare->last_appl_ptr %= pcm->boundary; if (print) printf("last_appl_ptr %lu", dshare->last_appl_ptr); slave_appl_ptr = dshare->slave_appl_ptr % dshare->slave_buffer_size; if (print) printf("slave_appl_ptr %lu", slave_appl_ptr); dshare->slave_appl_ptr += size; dshare->slave_appl_ptr %= dshare->slave_boundary; if (print) printf("dshare->slave_appl_ptr %lu", dshare->slave_appl_ptr); for (;;) { snd_pcm_uframes_t transfer = size; if (appl_ptr + transfer > pcm->buffer_size) transfer = pcm->buffer_size - appl_ptr; if (slave_appl_ptr + transfer > dshare->slave_buffer_size) transfer = dshare->slave_buffer_size - slave_appl_ptr; share_areas(dshare, src_areas, dst_areas, appl_ptr, slave_appl_ptr, transfer); size -= transfer; if (! size) break; slave_appl_ptr += transfer; slave_appl_ptr %= dshare->slave_buffer_size; appl_ptr += transfer; appl_ptr %= pcm->buffer_size; } if (print) printf("exiting");

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[alsa-devel] [PATCH 21/22] ARM: dts: exynos5422-odroidxu3: Add support for secondary DAI
by Sylwester Nawrocki 11 Feb '19

11 Feb '19

This patch extends DAPM routing and adds secondary CPU DAI entry to support the secondary audio PCM interface on Odroid XU3. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- arch/arm/boot/dts/exynos5422-odroidxu3-audio.dtsi | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/arch/arm/boot/dts/exynos5422-odroidxu3-audio.dtsi b/arch/arm/boot/dts/exynos5422-odroidxu3-audio.dtsi index e84544b220b9..51a843bd65ed 100644 --- a/arch/arm/boot/dts/exynos5422-odroidxu3-audio.dtsi +++ b/arch/arm/boot/dts/exynos5422-odroidxu3-audio.dtsi @@ -24,7 +24,9 @@ "Headphone Jack", "MICBIAS", "IN1", "Headphone Jack", "Speakers", "SPKL", - "Speakers", "SPKR"; + "Speakers", "SPKR", + "I2S Playback", "Mixer DAI TX", + "HiFi Playback", "Mixer DAI TX"; assigned-clocks = <&clock CLK_MOUT_EPLL>, <&clock CLK_MOUT_MAU_EPLL>, @@ -51,7 +53,7 @@ <196608000>; cpu { - sound-dai = <&i2s0 0>; + sound-dai = <&i2s0 0>, <&i2s0 1>; }; codec { sound-dai = <&hdmi>, <&max98090>; -- 2.20.1

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[alsa-devel] [PATCH 1/1] seq: arecordmidi: Add num-events option
by pmalani＠chromium.org 11 Feb '19

11 Feb '19

From: Prashant Malani <pmalani(a)chromium.org> Add a command line option to automatically exit after recording a fixed number of MIDI events. This allows a program using arecordmidi to expect a MIDI file to be written automatically when the specified number of events have been received, instead of having to send a SIGINT or SIGTERM programmatically. It also avoids the need to have the arecordmidi process running in the background, and then constantly stat the output file to check if any bytes have been written to it (this makes for less predictable and longer-running tests). This functionality finds use in Chrome OS functional testing, since having to send SIGTERM/SIGINT programmatically and then wait for the output file adds unpredictability and delay to the tests. The addition of this command-line option should (hopefully) not break any existing usage. Signed-off-by: Prashant Malani <pmalani(a)chromium.org> diff --git a/seq/aplaymidi/arecordmidi.c b/seq/aplaymidi/arecordmidi.c index 19dbb7d..ca25bfa 100644 --- a/seq/aplaymidi/arecordmidi.c +++ b/seq/aplaymidi/arecordmidi.c @@ -690,7 +690,8 @@ static void help(const char *argv0) " -t,--ticks=ticks resolution in ticks per beat or frame\n" " -s,--split-channels create a track for each channel\n" " -m,--metronome=client:port play a metronome signal\n" - " -i,--timesig=nn:dd time signature\n", + " -i,--timesig=nn:dd time signature\n" + " -n,--num-events=events fixed number of events to record, then exit\n", argv0); } @@ -706,7 +707,7 @@ static void sighandler(int sig) int main(int argc, char *argv[]) { - static const char short_options[] = "hVlp:b:f:t:sdm:i:"; + static const char short_options[] = "hVlp:b:f:t:sdm:i:n:"; static const struct option long_options[] = { {"help", 0, NULL, 'h'}, {"version", 0, NULL, 'V'}, @@ -719,6 +720,7 @@ int main(int argc, char *argv[]) {"dump", 0, NULL, 'd'}, {"metronome", 1, NULL, 'm'}, {"timesig", 1, NULL, 'i'}, + {"num-events", 1, NULL, 'n'}, { } }; @@ -727,6 +729,9 @@ int main(int argc, char *argv[]) struct pollfd *pfds; int npfds; int c, err; + /* If |num_events| isn't specified, leave it at 0. */ + int num_events = 0; + int events_received = 0; init_seq(); @@ -775,6 +780,11 @@ int main(int argc, char *argv[]) case 'i': time_signature(optarg); break; + case 'n': + num_events = atoi(optarg); + if (num_events <= 0) + fatal("num_events must be greater than 0"); + break; default: help(argv[0]); return 1; @@ -864,13 +874,20 @@ int main(int argc, char *argv[]) err = snd_seq_event_input(seq, &event); if (err < 0) break; - if (event) + if (event) { record_event(event); + events_received++; + } } while (err > 0); if (stop) break; + if (num_events && (events_received == num_events)) + break; } + if (num_events && events_received < num_events) + fputs("Warning: Received signal before num_events\n", stdout); + finish_tracks(); write_file(); -- 2.20.1.495.gaa96b0ce6b-goog

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Re: [alsa-devel] Driver for MOTU devices
by Takashi Sakamoto 10 Feb '19

10 Feb '19

Hi, On Fri, 8 Feb 2019, Peter Sjöberg wrote: > Hello! > > I'm contacting you as the developer of the kernel driver for MOTU firewire devices. > > First I want to thank you for your time and effort to develop the driver - I have the Audio Express which was unusable in Linux for a long time so this was a > really nice surprise! > > I want to ask you a question about the code, if it's the wrong place to ask please redirect me. > In the file amdtp-motu.c line 139 the code thas converts 24 to 32 bits reads: > *dst = (byte[0] << 24) | (byte[1] << 16) | byte[2]; > ..however I am puzzled by the conversion of the third byte here. In my mind it should be: > *dst = (byte[0] << 24) | (byte[1] << 16) | (byte[2] << 8); > > The conversion in the other direction is written like this (line 167-169): > byte[0] = (*src >> 24) & 0xff; > byte[1] = (*src >> 16) & 0xff; > byte[2] = (*src >> 8) & 0xff; > > I just want to help out so please inform me if I'm wrong and why :) Oops. It's a bug since v4.12 or later, indeed. Unfortunately, I'm on vacation. I'll post the patch to fix it in next week after testing it with MOTU models I own. Thanks for your report ;) By the way, I guess that you see many kernel messages with 'unsolicited response' from firewire-core driver, like. fireire_core 0000:03:00.0: unsolicited response (source ffc0, tlabel 1f) As long as I know, AudioExpress often transfers invalid response code for any asynchronous transaction[1]. This brings failures of read/write transaction and ALSA MOTU driver sometimes fails to handle the device. For your information. [1] https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/soun… Regards Takashi Sakamoto

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[alsa-devel] [PATCH] ASoC: codecs: jz4725b: Remove unnecessary const qualifier
by Nathan Chancellor 09 Feb '19

09 Feb '19

Clang warns: sound/soc/codecs/jz4725b.c:177:14: warning: duplicate 'const' declaration specifier [-Wduplicate-decl-specifier] static const SOC_VALUE_ENUM_SINGLE_DECL(jz4725b_codec_adc_src_enum, ^ include/sound/soc.h:356:2: note: expanded from macro 'SOC_VALUE_ENUM_SINGLE_DECL' SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) ^ include/sound/soc.h:353:2: note: expanded from macro 'SOC_VALUE_ENUM_DOUBLE_DECL' const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \ ^ As it points out, SOC_VALUE_ENUM_DOUBLE_DECL has the const attribute in its definition so remove it here. Fixes: e9d97b05a80f ("ASoC: codecs: Add jz4725b-codec driver") Link: https://github.com/ClangBuiltLinux/linux/issues/354 Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> --- sound/soc/codecs/jz4725b.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c index e3dba92f30bd..5cc8c7ca24c6 100644 --- a/sound/soc/codecs/jz4725b.c +++ b/sound/soc/codecs/jz4725b.c @@ -174,12 +174,12 @@ static const char * const jz4725b_codec_adc_src_texts[] = { "Mic 1", "Mic 2", "Line In", "Mixer", }; static const unsigned int jz4725b_codec_adc_src_values[] = { 0, 1, 2, 3, }; -static const SOC_VALUE_ENUM_SINGLE_DECL(jz4725b_codec_adc_src_enum, - JZ4725B_CODEC_REG_CR3, - REG_CR3_INSEL_OFFSET, - REG_CR3_INSEL_MASK, - jz4725b_codec_adc_src_texts, - jz4725b_codec_adc_src_values); +static SOC_VALUE_ENUM_SINGLE_DECL(jz4725b_codec_adc_src_enum, + JZ4725B_CODEC_REG_CR3, + REG_CR3_INSEL_OFFSET, + REG_CR3_INSEL_MASK, + jz4725b_codec_adc_src_texts, + jz4725b_codec_adc_src_values); static const struct snd_kcontrol_new jz4725b_codec_adc_src_ctrl = SOC_DAPM_ENUM("Route", jz4725b_codec_adc_src_enum); -- 2.20.1

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Re: [alsa-devel] [PATCH] ALSA: pcm: Revert capture stream behavior change in blocking mode
by Ricardo Biehl Pasquali 08 Feb '19

08 Feb '19

Takashi Iwai wrote: > e.g. one thread goes to a write loop in blocking mode while another > thread controls the start/stop of the stream manually. You meant a read loop. Agreed. What about adding a comment like: /* * If size < start_threshold, wait indefinitely. Another * thread may start capture */ So no one will bring this idea again. pasquali

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[alsa-devel] ASoC: regulator notifier registration should be managed
by Guennadi Liakhovetski 08 Feb '19

08 Feb '19

From: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Regulator notifiers, that were registered during codec driver probing, must be unregistered during driver release, or device managed versions have to be used. This patch fixes codec drivers, that weren't explicitly unregistering notifiers and simplifies those, that did that manually. Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/soc/codecs/max9860.c | 3 ++- sound/soc/codecs/pcm512x.c | 5 +++-- sound/soc/codecs/tlv320aic31xx.c | 16 +++------------- sound/soc/codecs/tlv320aic3x.c | 25 ++++--------------------- sound/soc/codecs/wm8770.c | 18 +++--------------- sound/soc/codecs/wm8962.c | 9 +++------ sound/soc/codecs/wm8995.c | 29 +++++++---------------------- sound/soc/codecs/wm8996.c | 9 +++------ 8 files changed, 28 insertions(+), 86 deletions(-) diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c index de3d44e9199b..8be636fe6552 100644 --- a/sound/soc/codecs/max9860.c +++ b/sound/soc/codecs/max9860.c @@ -615,7 +615,8 @@ static int max9860_probe(struct i2c_client *i2c) max9860->dvddio_nb.notifier_call = max9860_dvddio_event; - ret = regulator_register_notifier(max9860->dvddio, &max9860->dvddio_nb); + ret = devm_regulator_register_notifier(max9860->dvddio, + &max9860->dvddio_nb); if (ret) dev_err(dev, "Failed to register DVDDIO notifier: %d\n", ret); diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 4cc24a5d5c31..18f414be849f 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1520,8 +1520,9 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) pcm512x->supply_nb[2].notifier_call = pcm512x_regulator_event_2; for (i = 0; i < ARRAY_SIZE(pcm512x->supplies); i++) { - ret = regulator_register_notifier(pcm512x->supplies[i].consumer, - &pcm512x->supply_nb[i]); + ret = devm_regulator_register_notifier( + pcm512x->supplies[i].consumer, + &pcm512x->supply_nb[i]); if (ret != 0) { dev_err(dev, "Failed to register regulator notifier: %d\n", diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 608ad49ad978..7a4a607cca3c 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1274,8 +1274,9 @@ static int aic31xx_codec_probe(struct snd_soc_component *component) aic31xx->disable_nb[i].nb.notifier_call = aic31xx_regulator_event; aic31xx->disable_nb[i].aic31xx = aic31xx; - ret = regulator_register_notifier(aic31xx->supplies[i].consumer, - &aic31xx->disable_nb[i].nb); + ret = devm_regulator_register_notifier( + aic31xx->supplies[i].consumer, + &aic31xx->disable_nb[i].nb); if (ret) { dev_err(component->dev, "Failed to request regulator notifier: %d\n", @@ -1298,19 +1299,8 @@ static int aic31xx_codec_probe(struct snd_soc_component *component) return 0; } -static void aic31xx_codec_remove(struct snd_soc_component *component) -{ - struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); - int i; - - for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) - regulator_unregister_notifier(aic31xx->supplies[i].consumer, - &aic31xx->disable_nb[i].nb); -} - static const struct snd_soc_component_driver soc_codec_driver_aic31xx = { .probe = aic31xx_codec_probe, - .remove = aic31xx_codec_remove, .set_bias_level = aic31xx_set_bias_level, .controls = common31xx_snd_controls, .num_controls = ARRAY_SIZE(common31xx_snd_controls), diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6a271e6e6b8f..6dcc3ace9d70 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1605,13 +1605,14 @@ static int aic3x_probe(struct snd_soc_component *component) for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) { aic3x->disable_nb[i].nb.notifier_call = aic3x_regulator_event; aic3x->disable_nb[i].aic3x = aic3x; - ret = regulator_register_notifier(aic3x->supplies[i].consumer, - &aic3x->disable_nb[i].nb); + ret = devm_regulator_register_notifier( + aic3x->supplies[i].consumer, + &aic3x->disable_nb[i].nb); if (ret) { dev_err(component->dev, "Failed to request regulator notifier: %d\n", ret); - goto err_notif; + return ret; } } @@ -1669,29 +1670,11 @@ static int aic3x_probe(struct snd_soc_component *component) aic3x_add_widgets(component); return 0; - -err_notif: - while (i--) - regulator_unregister_notifier(aic3x->supplies[i].consumer, - &aic3x->disable_nb[i].nb); - return ret; -} - -static void aic3x_remove(struct snd_soc_component *component) -{ - struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component); - int i; - - list_del(&aic3x->list); - for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) - regulator_unregister_notifier(aic3x->supplies[i].consumer, - &aic3x->disable_nb[i].nb); } static const struct snd_soc_component_driver soc_component_dev_aic3x = { .set_bias_level = aic3x_set_bias_level, .probe = aic3x_probe, - .remove = aic3x_remove, .controls = aic3x_snd_controls, .num_controls = ARRAY_SIZE(aic3x_snd_controls), .dapm_widgets = aic3x_dapm_widgets, diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 806245c70f8b..37467c512597 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -666,8 +666,9 @@ static int wm8770_spi_probe(struct spi_device *spi) /* This should really be moved into the regulator core */ for (i = 0; i < ARRAY_SIZE(wm8770->supplies); i++) { - ret = regulator_register_notifier(wm8770->supplies[i].consumer, - &wm8770->disable_nb[i]); + ret = devm_regulator_register_notifier( + wm8770->supplies[i].consumer, + &wm8770->disable_nb[i]); if (ret) { dev_err(&spi->dev, "Failed to register regulator notifier: %d\n", @@ -687,25 +688,12 @@ static int wm8770_spi_probe(struct spi_device *spi) return ret; } -static int wm8770_spi_remove(struct spi_device *spi) -{ - struct wm8770_priv *wm8770 = spi_get_drvdata(spi); - int i; - - for (i = 0; i < ARRAY_SIZE(wm8770->supplies); ++i) - regulator_unregister_notifier(wm8770->supplies[i].consumer, - &wm8770->disable_nb[i]); - - return 0; -} - static struct spi_driver wm8770_spi_driver = { .driver = { .name = "wm8770", .of_match_table = wm8770_of_match, }, .probe = wm8770_spi_probe, - .remove = wm8770_spi_remove }; module_spi_driver(wm8770_spi_driver); diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index efd8910b1ff7..467ed78dd2df 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3424,8 +3424,9 @@ static int wm8962_probe(struct snd_soc_component *component) /* This should really be moved into the regulator core */ for (i = 0; i < ARRAY_SIZE(wm8962->supplies); i++) { - ret = regulator_register_notifier(wm8962->supplies[i].consumer, - &wm8962->disable_nb[i]); + ret = devm_regulator_register_notifier( + wm8962->supplies[i].consumer, + &wm8962->disable_nb[i]); if (ret != 0) { dev_err(component->dev, "Failed to register regulator notifier: %d\n", @@ -3467,15 +3468,11 @@ static int wm8962_probe(struct snd_soc_component *component) static void wm8962_remove(struct snd_soc_component *component) { struct wm8962_priv *wm8962 = snd_soc_component_get_drvdata(component); - int i; cancel_delayed_work_sync(&wm8962->mic_work); wm8962_free_gpio(component); wm8962_free_beep(component); - for (i = 0; i < ARRAY_SIZE(wm8962->supplies); i++) - regulator_unregister_notifier(wm8962->supplies[i].consumer, - &wm8962->disable_nb[i]); } static const struct snd_soc_component_driver soc_component_dev_wm8962 = { diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 68c99fe37097..fe2156a1220a 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1995,20 +1995,6 @@ static int wm8995_set_bias_level(struct snd_soc_component *component, return 0; } -static void wm8995_remove(struct snd_soc_component *component) -{ - struct wm8995_priv *wm8995; - int i; - - wm8995 = snd_soc_component_get_drvdata(component); - - for (i = 0; i < ARRAY_SIZE(wm8995->supplies); ++i) - regulator_unregister_notifier(wm8995->supplies[i].consumer, - &wm8995->disable_nb[i]); - - regulator_bulk_free(ARRAY_SIZE(wm8995->supplies), wm8995->supplies); -} - static int wm8995_probe(struct snd_soc_component *component) { struct wm8995_priv *wm8995; @@ -2021,8 +2007,9 @@ static int wm8995_probe(struct snd_soc_component *component) for (i = 0; i < ARRAY_SIZE(wm8995->supplies); i++) wm8995->supplies[i].supply = wm8995_supply_names[i]; - ret = regulator_bulk_get(component->dev, ARRAY_SIZE(wm8995->supplies), - wm8995->supplies); + ret = devm_regulator_bulk_get(component->dev, + ARRAY_SIZE(wm8995->supplies), + wm8995->supplies); if (ret) { dev_err(component->dev, "Failed to request supplies: %d\n", ret); return ret; @@ -2039,8 +2026,9 @@ static int wm8995_probe(struct snd_soc_component *component) /* This should really be moved into the regulator core */ for (i = 0; i < ARRAY_SIZE(wm8995->supplies); i++) { - ret = regulator_register_notifier(wm8995->supplies[i].consumer, - &wm8995->disable_nb[i]); + ret = devm_regulator_register_notifier( + wm8995->supplies[i].consumer, + &wm8995->disable_nb[i]); if (ret) { dev_err(component->dev, "Failed to register regulator notifier: %d\n", @@ -2052,7 +2040,7 @@ static int wm8995_probe(struct snd_soc_component *component) wm8995->supplies); if (ret) { dev_err(component->dev, "Failed to enable supplies: %d\n", ret); - goto err_reg_get; + retturn ret; } ret = snd_soc_component_read32(component, WM8995_SOFTWARE_RESET); @@ -2099,8 +2087,6 @@ static int wm8995_probe(struct snd_soc_component *component) err_reg_enable: regulator_bulk_disable(ARRAY_SIZE(wm8995->supplies), wm8995->supplies); -err_reg_get: - regulator_bulk_free(ARRAY_SIZE(wm8995->supplies), wm8995->supplies); return ret; } @@ -2188,7 +2174,6 @@ static struct snd_soc_dai_driver wm8995_dai[] = { static const struct snd_soc_component_driver soc_component_dev_wm8995 = { .probe = wm8995_probe, - .remove = wm8995_remove, .set_bias_level = wm8995_set_bias_level, .controls = wm8995_snd_controls, .num_controls = ARRAY_SIZE(wm8995_snd_controls), diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 91711f8958c5..ab04ea18c312 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2801,8 +2801,9 @@ static int wm8996_i2c_probe(struct i2c_client *i2c, /* This should really be moved into the regulator core */ for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) { - ret = regulator_register_notifier(wm8996->supplies[i].consumer, - &wm8996->disable_nb[i]); + ret = devm_regulator_register_notifier( + wm8996->supplies[i].consumer, + &wm8996->disable_nb[i]); if (ret != 0) { dev_err(&i2c->dev, "Failed to register regulator notifier: %d\n", @@ -3071,16 +3072,12 @@ static int wm8996_i2c_probe(struct i2c_client *i2c, static int wm8996_i2c_remove(struct i2c_client *client) { struct wm8996_priv *wm8996 = i2c_get_clientdata(client); - int i; wm8996_free_gpio(wm8996); if (wm8996->pdata.ldo_ena > 0) { gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); gpio_free(wm8996->pdata.ldo_ena); } - for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) - regulator_unregister_notifier(wm8996->supplies[i].consumer, - &wm8996->disable_nb[i]); return 0; } -- 2.17.1

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