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[alsa-devel] Possible GPL-licensed header included in alsa-lib
by Clemens Lang 27 Jul '15

27 Jul '15

Hi, I recently reviewed the alsa-lib 1.0.27.2 source code from a licensing perspective to ensure compliant usage and distribution. It seems the intention is for the libraries to be covered by LGPL. I had a look at include/list.h, which seems to be licensed under GPL-2.0. It does not have a license header or any other copyright information, but it states it was taken from Linux 2.4.0: include/list.h, line 5: | This code was taken from the Linux 2.4.0 kernel. [jaroslav] I'm not sure one could argue that the header was trivial, especially since it does not only contain constants, structures and interface definitions, but also implementations of inline functions. I found one previous inquiry about a similar issue[1], but it does not cover the include/list.h header. Two questions: 1. Have the copyright holders of the code in question agreed to re-license under a different, LGPL-preserving license and, if yes, is there some documentation about that? 2. If the header is indeed covered by GPL-2.0, would you accept patches to replace it with a different implementation, e.g. Rusty Russell's MIT-licensed list from ccan[2]? [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-March/038267.html [2] http://ccodearchive.net/info/list.html Thank you for your time, Clemens Lang -- BMW Car IT GmbH Clemens Lang Spezialist Entwicklung Lise-Meitner-Straße 14 89081 Ulm Mail: clemens.lang(a)bmw-carit.de Web: http://www.bmw-carit.de -------------------------------------------------------------------- BMW Car IT GmbH Geschäftsführer: Michael Würtenberger und Reinhard Stolle Sitz und Registergericht: München HRB 134810 --------------------------------------------------------------------

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[alsa-devel] snd_pcm_set_params faild on set_buffer_time_near and set_period_time_near
by Martin Geier 27 Jul '15

27 Jul '15

Hallo, I tried to configure alsa device with snd_pcm_set_params function, requested parameters was: channel 2, rate: 22050, soft_resample: 1, latency: 50000 Function snd_pcm_hw_params_set_buffer_time_near and snd_pcm_hw_param_set_near is called with: min: 50000, max: 50000, mindir: 0, maxdir: -1 and input params was: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 22050 PERIOD_TIME: (362 743039) PERIOD_SIZE: [8 16384] PERIOD_BYTES: [32 65536] PERIODS: [2 19] BUFFER_TIME: (725 1486078) BUFFER_SIZE: [16 32768] BUFFER_BYTES: [64 131072] TICK_TIME: ALL unfortunately, this function fails, on function snd_pcm_hw_param_set_first: ALSA ERROR hw_params: set_near (BUFFER_TIME) value = 50000 : Invalid argument ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 22050 PERIOD_TIME: (4580 5533) PERIOD_SIZE: NONE PERIOD_BYTES: [404 488] PERIODS: 10 BUFFER_TIME: (50022 50023) BUFFER_SIZE: 1103 BUFFER_BYTES: 4412 TICK_TIME: ALL This shouldn't be problem because in snd_pcm_set_params is snd_pcm_hw_params_set_period_time_near called. This function should be called with same params as snd_pcm_hw_params_set_buffer_time_near but the input params are: snd_pcm_hw_param_set_near: min: 12500, max: 12500, mindir: 0, maxdir: -1 hw_params: snd_pcm_hw_param_set_near (PERIOD_TIME) ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 22050 PERIOD_TIME: (4580 5533) PERIOD_SIZE: NONE PERIOD_BYTES: [404 488] PERIODS: 10 BUFFER_TIME: (50022 50023) BUFFER_SIZE: 1103 BUFFER_BYTES: 4412 TICK_TIME: ALL because in snd_pcm_hw_param_set_near is if (last) err = snd_pcm_hw_param_set_last(pcm, params, var, val, dir); else err = snd_pcm_hw_param_set_first(pcm, params, var, val, dir); if (err < 0) dump_hw_params(params, "set_near", var, *val, err); return err; and in error part is only debug output and not restore original parameters. Environment: alsa-lib 1.0.27.1 TI AM335x soc Simple fix is attached, but I am not sure if it is correct. BR Martin Geier -- *StreamUnlimited* High Tech Campus Vienna, Gutheil-Schoder-Gasse 10, A-1102 Vienna, Austria Levocska 9, 851 01 Bratislava, Slovakia Email: martin.geier(a)streamunlimited.com www.streamunlimited.com

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[alsa-devel] [PATCH] ALSA: hda - Apply fixup for another Toshiba Satellite S50D
by Takashi Iwai 27 Jul '15

27 Jul '15

Toshiba Satellite S50D has another model with a different PCI SSID (1179:fa93) while the previous fixup was for 1179:fa91. Adjust the fixup entry with SND_PCI_QUIRK_MASK() to match with both devices. Reported-by: Tim Sample <timsample(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index dcc7fe91244c..9d947aef2c8b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2920,7 +2920,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x148a, "HP Mini", STAC_92HD83XXX_HP_LED), SND_PCI_QUIRK_VENDOR(PCI_VENDOR_ID_HP, "HP", STAC_92HD83XXX_HP), - SND_PCI_QUIRK(PCI_VENDOR_ID_TOSHIBA, 0xfa91, + /* match both for 0xfa91 and 0xfa93 */ + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_TOSHIBA, 0xfffd, 0xfa91, "Toshiba Satellite S50D", STAC_92HD83XXX_GPIO10_EAPD), {} /* terminator */ }; -- 2.4.6

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[alsa-devel] [PATCH] ALSA: fireworks: add support for AudioFire2 quirk
by Takashi Sakamoto 27 Jul '15

27 Jul '15

Fireworks uses TSB43CB43(IceLynx-Micro) as its IEC 61883-1/6 interface. This chip includes ARM7 core, and loads and runs program. The firmware is stored in on-board memory and loaded every powering-on. Echo Audio ships several versions of firmwares for each model. These firmwares have each quirk and the quirk changes a sequence of packets. AudioFire2 has a quirk to transfer a first packet with non-zero in its dbc field. This causes ALSA Fireworks driver to detect discontinuity. As long as I investigated, firmware 5.7, 5.7.6 and 5.8 have this quirk. This commit adds a support for the quirk to handle AudioFire2 packets. For safe, CIP_SKIP_INIT_DBC_CHECK is applied to all versions of AudioFire2's firmwares. 02 00050002 90ffffff <- 42 0005000a 90013000 42 00050012 90014400 42 0005001a 90015800 02 0005001a 90ffffff 42 00050022 90019000 42 0005002a 9001a400 42 00050032 9001b800 02 00050032 90ffffff 42 0005003a 9001d000 42 00050042 9001e400 42 0005004a 9001f800 02 0005004a 90ffffff Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireworks/fireworks.c | 2 ++ sound/firewire/fireworks/fireworks.h | 1 + sound/firewire/fireworks/fireworks_stream.c | 3 +++ 3 files changed, 6 insertions(+) diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 2682e7e..c670db4 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -248,6 +248,8 @@ efw_probe(struct fw_unit *unit, err = get_hardware_info(efw); if (err < 0) goto error; + if (entry->model_id == MODEL_ECHO_AUDIOFIRE_2) + efw->is_af2 = true; if (entry->model_id == MODEL_ECHO_AUDIOFIRE_9) efw->is_af9 = true; diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index 4f0201a..c33252b 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -70,6 +70,7 @@ struct snd_efw { bool resp_addr_changable; /* for quirks */ + bool is_af2; bool is_af9; u32 firmware_version; diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index c55db1b..a0762dd 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -172,6 +172,9 @@ int snd_efw_stream_init_duplex(struct snd_efw *efw) efw->tx_stream.flags |= CIP_DBC_IS_END_EVENT; /* Fireworks reset dbc at bus reset. */ efw->tx_stream.flags |= CIP_SKIP_DBC_ZERO_CHECK; + /* AudioFire2 starts packets with non-zero dbc. */ + if (efw->is_af2) + efw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK; /* AudioFire9 always reports wrong dbs. */ if (efw->is_af9) efw->tx_stream.flags |= CIP_WRONG_DBS; -- 2.1.4

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[alsa-devel] [PATCH v3 0/4] i915 to call hda driver on HDMI plug/unplug
by David Henningsson 27 Jul '15

27 Jul '15

Changes since v2 is that the patch set has diminished to not transfer connector/eld information, since it might be better that such information to be transferred by a separate call in the ordinary direction. That could be added in a later patch set. The audio_ptr is now a void* (instead of a hdac_bus*) so that the audio side can easily refactor the ptr to point to the codec instead of the bus if we would find that beneficial. David Henningsson (4): drm/i915: Add audio hotplug info callback drm/i915: Call audio hotplug notify function ALSA: hda - Dispatch incoming HDMI hotplug i915 callback ALSA: hda - Wake the codec up on hotplug notify events drivers/gpu/drm/i915/i915_drv.h | 1 + drivers/gpu/drm/i915/intel_audio.c | 23 ++++++++++++++++++++--- include/drm/i915_component.h | 5 +++++ include/sound/hdaudio.h | 4 ++++ sound/hda/hdac_i915.c | 23 +++++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 13 ++++++++++++- 6 files changed, 65 insertions(+), 4 deletions(-) -- 1.7.9.5

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Re: [alsa-devel] [ALC668]: Asus N751JK - surround 21 and surround40 pulseaudio bug
by Raymond Yau 26 Jul '15

26 Jul '15

> > Alsa-info patched kernel: http://www.alsa-project.org/db/?f=cd23a3f0bb72e38d73224620e3e8ab57a73aa441 > > Chmap results: > > Type = FIXED, Channels = 2 > FL FR > Type = FIXED, Channels = 4 > FL FR NA LFE > > When unplugging the subwoofer, then chmap gives this and stays like this. > > Cannot open PCM stream hw:1 for PLAYBACK > > [Pavucontrol] -> [Configuration] -> [Built-in Audio] gives > > Analog Stereo Duplex > Analog Stereo Output > Analog Surround 2.1 Output + Analog Stereo Input > Analog Surround 2.1 Output > Analog Surround 4.0 Output + Analog Stereo Input > Analog Surround 4.0 Output > Analog Stereo Input > Off Seem pulseaudio bug since your laptop does not have surround40 chmap It is not clear how pulseaudio's 21 channel-map match with alsa's 2.1 chmap http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… [Mapping analog-surround-21] device-strings = surround21:%f channel-map = front-left,front-right,lfe paths-output = analog-output analog-output-lineout analog-output-speaker priority = 8 direction = output [Mapping analog-surround-40] device-strings = surround40:%f channel-map = front-left,front-right,rear-left,rear-right paths-output = analog-output analog-output-lineout analog-output-speaker priority = 7 direction = output > > Pulseaudio verbose log while plugging in the subwoofer: http://pastebin.com/Ya5u1qz6 https://bugs.freedesktop.org/enter_bug.cgi?product=PulseAudio

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[alsa-devel] [PATCH v3] ALSA: hda - Fix the headset mic that will not work on Dell desktop machine
by woodrow.shen＠canonical.com 25 Jul '15

25 Jul '15

From: Woodrow Shen <woodrow.shen(a)canonical.com> When the headset was plugged in the Dell desktop, the mic of headset can't be detected and workable. According to the alsa-info, we found the differece between alsa and init_pin_configs on the machine, so we need to add the pin configs to make headset work. Codec: Realtek ALC3234 Vendor Id: 0x10ec0255 Subsystem Id: 0x102806bb BugLink: https://bugs.launchpad.net/bugs/1477900 Signed-off-by: Woodrow Shen <woodrow.shen(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 742fc62..d94c0e3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5473,6 +5473,28 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1e, 0x411111f0}, {0x21, 0x0221103f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170150}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x02011020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221105f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170110}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x01014020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, {0x17, 0x90170140}, -- 2.1.4

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Re: [alsa-devel] ALSA: hda - Use generic parser for Cirrus codec driver
by Dan Carpenter 25 Jul '15

25 Jul '15

Hello Takashi Iwai, The patch 1077a024812d: "ALSA: hda - Use generic parser for Cirrus codec driver" from Dec 19, 2012, leads to the following static checker warning: sound/pci/hda/patch_cirrus.c:1004 cs4210_spdif_automute() warn: we tested 'spdif_present' before and it was 'true' sound/pci/hda/patch_cirrus.c 984 static void cs4210_spdif_automute(struct hda_codec *codec, 985 struct hda_jack_callback *tbl) 986 { 987 struct cs_spec *spec = codec->spec; 988 bool spdif_present = false; 989 hda_nid_t spdif_pin = spec->gen.autocfg.dig_out_pins[0]; 990 991 /* detect on spdif is specific to CS4210 */ 992 if (!spec->spdif_detect || 993 spec->vendor_nid != CS4210_VENDOR_NID) 994 return; 995 996 spdif_present = snd_hda_jack_detect(codec, spdif_pin); 997 if (spdif_present == spec->spdif_present) 998 return; 999 1000 spec->spdif_present = spdif_present; 1001 /* SPDIF TX on/off */ 1002 if (spdif_present) ^^^^^^^^^^^^^ 1003 snd_hda_set_pin_ctl(codec, spdif_pin, 1004 spdif_present ? PIN_OUT : 0); ^^^^^^^^^^^^^ I'm not positive but I think this is buggy. We should remove the if statement. 1005 1006 cs_automute(codec); 1007 } regards, dan carpenter

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[alsa-devel] [PATCH] ASoC: rt5677: Return error if devm_gpiod_get_optional return ERR_PTR
by Axel Lin 25 Jul '15

25 Jul '15

If devm_gpiod_get_optional() return ERR_PTR, it means something wrong so request gpio fails. We had better return error in such case. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5677.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 8d70911..3a074b6 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5133,14 +5133,14 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, if (IS_ERR(rt5677->pow_ldo2)) { ret = PTR_ERR(rt5677->pow_ldo2); dev_err(&i2c->dev, "Failed to request POW_LDO2: %d\n", ret); - rt5677->pow_ldo2 = 0; + return PTR_ERR(rt5677->pow_ldo2); } rt5677->reset_pin = devm_gpiod_get_optional(&i2c->dev, "realtek,reset", GPIOD_OUT_HIGH); if (IS_ERR(rt5677->reset_pin)) { ret = PTR_ERR(rt5677->reset_pin); dev_err(&i2c->dev, "Failed to request RESET: %d\n", ret); - rt5677->reset_pin = 0; + return PTR_ERR(rt5677->reset_pin); } if (rt5677->pow_ldo2 || rt5677->reset_pin) { -- 2.1.0

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Re: [alsa-devel] writing an alsa driver
by Marco Bodega 24 Jul '15

24 Jul '15

Hi all, I own an Asus x205ta too and I really want to help to make this audio card works. I'm using linux next with Michele Curti's patch (as of 23 june) Here kernel .config: CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH=m CONFIG_SND_SOC_INTEL_BYTCR_RT5645_MACH=m # CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH is no CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH=m # CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH is not set # CONFIG_SND_SOC_INTEL_BROADWELL_MACH is not set CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH=m CONFIG_SND_SOC_INTEL_BYTCR_RT5645_MACH=m bo@X205TA:~$ uname -a Linux X205TA 4.2.0-rc2-next-20150717+ #3 SMP Wed Jul 22 15:31:59 CEST 2015 x86_64 x86_64 x86_64 GNU/Linux Output of lsmod: snd_soc_sst_bytcr_rt5645 16384 0 snd_intel_sst_acpi 16384 1 snd_intel_sst_core 73728 1 snd_intel_sst_acpi snd_soc_rt5645 139264 1 snd_soc_sst_mfld_platform 90112 3 snd_intel_sst_core snd_pcm 102400 4 snd_soc_rt5645,snd_soc_core,snd_soc_sst_mfld_platform,snd_soc_sst_bytcr_rt5645 bo@X205TA:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: baytrailcraudio [baytrailcraudio], device 0: Baytrail Audio (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 bo@@X205TA:~$ aplay -D hw:0 -c 2 -r 48000 -f S16_LE /dev/urandomPlaying raw data '/dev/urandom' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo aplay: set_params:1297: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 48000 PERIOD_TIME: 125000 PERIOD_SIZE: 6000 PERIOD_BYTES: 24000 PERIODS: 4 BUFFER_TIME: 500000 BUFFER_SIZE: 24000 BUFFER_BYTES: 96000 TICK_TIME: 0 bo@X205TA:~$ cat /proc/interrupts | grep sst 6: 0 0 0 0 IO-APIC 28-fasteoi intel_sst_driver bo@X205TA:~$ ll /lib/firmware/intel/fw* -rw-r--r-- 1 root root 647078 lug 22 17:56 /lib/firmware/intel/fw_sst_0f28.bin -rw-r--r-- 1 root root 265684 lug 22 17:56 /lib/firmware/intel/fw_sst_0f28.bin-48kHz_i2s_master -rw-r--r-- 1 root root 701614 lug 22 17:56 /lib/firmware/intel/fw_sst_22a8.bin I read that the firmware used by bytx205_rt5645 is intel/fw_sst_0f28.bin: in fact in sst_acpi i have: {"10EC5648", "X205", "bytx205_rt5645", NULL, "intel/fw_sst_0f28.bin", &byt_rvp_platform_data }, but it's bounded to sst1 as i see from disassembling acpi data: cat dsdt.dsl: [...] Device (SSP1) { Name (_ADR, Zero) // _ADR: Address Name (_HID, "SSPX0000" /* Intel SSP Device */) // _HID: Hardware ID Name (_CID, "SSPX0000" /* Intel SSP Device */) // _CID: Compatible ID [...] Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x0000001D, } Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x00000018, } Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x00000019, } Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x0000001A, } Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x0000001B, } Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x0000001C, } [...] however the fw is tied to work with SSP0. What can i do? I tried to connect parts in routing: #DSP routing amixer -c0 sset 'codec_out0 mix 0 pcm0_in' on amixer -c0 sset 'media0_out mix 0 media1_in' on amixer -c0 sset 'media1_in Gain 0' 80% amixer -c0 sset 'media1_in Gain 0 Ramp Delay' 50 amixer -c0 sset 'media1_in Gain 0' off amixer -c0 sset 'pcm0_in Gain 0' 80% amixer -c0 sset 'pcm0_in Gain 0 Ramp Delay' 50 amixer -c0 sset 'pcm0_in Gain 0' off amixer -c0 sset 'codec_out0 Gain 0' 80% amixer -c0 sset 'codec_out0 Gain 0 Ramp Delay' 50 amixer -c0 sset 'codec_out0 Gain 0' off #Codec routin, if you have different one please modfiy amixer -c0 sset "DAC MIXL INF1" on amixer -c0 sset "DAC MIXR INF1" on amixer -c0 sset "Stereo DAC MIXL DAC L1" on amixer -c0 sset "Stereo DAC MIXR DAC R1" on amixer -c0 sset "SPK MIXL DAC L1" on amixer -c0 sset "SPK MIXR DAC R1" on amixer -c0 sset "SPOL MIX SPKVOL L" on amixer -c0 sset "SPOR MIX SPKVOL R" on amixer -c0 sset "SPOL MIX DAC L1" on amixer -c0 sset "SPOR MIX DAC R1" on amixer -c0 sset "Speaker L" on amixer -c0 sset "Speaker R" on amixer -c0 sset "Headphone" off amixer -c0 sset "Ext Spk" on and the attached dmesg log appeared I get no sound but at least no heat :) Is everything that I've done correct? I see in these output logs many of the errors and bugs that the other users found. Thanks for the time spent Marco Output of dmesg:

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[alsa-devel] [GIT PULL] ASoC fixes for v4.2
by Mark Brown 24 Jul '15

24 Jul '15

The following changes since commit 52721d9d3334c1cb1f76219a161084094ec634dc: Linux 4.2-rc3 (2015-07-19 14:45:02 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-fix-v4.2-rc3 for you to fetch changes up to 996034b117b467709dec7811ef134063934fa626: Merge remote-tracking branches 'asoc/fix/sgtl5000', 'asoc/fix/topology' and 'asoc/fix/zx' into asoc-linus (2015-07-24 16:18:22 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.2 A lot of small fixes here, a few to the core: - Fix for binding DAPM stream widgets on devices with prefixes assigned to them - Minor fixes for the newly added topology interfaces - Locking and memory leak fixes for DAPM - Driver specific fixes ---------------------------------------------------------------- Axel Lin (2): ASoC: sgtl5000: Fix up define for SGTL5000_SMALL_POP ASoC: pcm1681: Fix setting de-emphasis sampling rate selection Ben Zhang (1): ASoC: ssm4567: Keep TDM_BCLKS in ssm4567_set_dai_fmt Charles Keepax (1): ASoC: dapm: Fix deadlock on auto-disable mux controls Fang, Yang A (1): ASoC: Intel: fix incorrect widget name Jeeja KP (1): ASoC: topology: Fix to add dapm mixer info Juergen Borleis (1): ASoC: fsl-ssi: Fix bitclock calculation for master mode Koro Chen (2): ASoC: mediatek: Fix unbalanced calls to runtime suspend/resume ASoC: mediatek: Use platform_of_node for machine drivers Krzysztof Kozlowski (2): ASoC: zx: i2s: Fix devm_ioremap_resource return value check ASoC: zx: spdif: Fix devm_ioremap_resource return value check Lars-Peter Clausen (4): ASoC: dapm: Lock during userspace access ASoC: dapm: Fix kcontrol widget name memory management ASoC: Free card DAPM context on snd_soc_instantiate_card() error path ASoC: dapm: Don't add prefix to widget stream name Liam Girdwood (1): ASoC: topology: Fix TLV size calculation. Mark Brown (6): Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/pcm1681' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branch 'asoc/fix/ssm4567' into asoc-linus Merge remote-tracking branches 'asoc/fix/fsl-ssi', 'asoc/fix/intel', 'asoc/fix/intel-kconfig' and 'asoc/fix/mediatek' into asoc-linus Merge remote-tracking branches 'asoc/fix/sgtl5000', 'asoc/fix/topology' and 'asoc/fix/zx' into asoc-linus Nicolas Boichat (1): ASoC: rt5645: Check if codec is initialized in workqueue handler Shilpa Sreeramalu (1): ASoC: Intel: Get correct usage_count value to load firmware Vinod Koul (2): ASoC: topology: fix typos in topology header ASoC: Intel: use CONFIG_SND_SOC for intel boards .../devicetree/bindings/sound/mt8173-max98090.txt | 2 ++ .../bindings/sound/mt8173-rt5650-rt5676.txt | 2 ++ include/uapi/sound/asoc.h | 4 +-- sound/soc/codecs/pcm1681.c | 2 +- sound/soc/codecs/rt5645.c | 3 ++ sound/soc/codecs/sgtl5000.h | 2 +- sound/soc/codecs/ssm4567.c | 8 ++++- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/intel/Makefile | 2 +- sound/soc/intel/atom/sst/sst_drv_interface.c | 14 ++++----- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 4 +-- sound/soc/mediatek/mt8173-max98090.c | 17 ++++++++--- sound/soc/mediatek/mt8173-rt5650-rt5676.c | 19 +++++++++--- sound/soc/mediatek/mtk-afe-pcm.c | 2 ++ sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 35 +++++++++++----------- sound/soc/soc-topology.c | 23 +++++++------- sound/soc/zte/zx296702-i2s.c | 4 +-- sound/soc/zte/zx296702-spdif.c | 4 +-- 19 files changed, 93 insertions(+), 57 deletions(-)

2 1

[alsa-devel] [PATCH RFT] ASoC: cs4265: Fix setting dai format for Left/Right Justified
by Axel Lin 24 Jul '15

24 Jul '15

The settings in current code does not match the datasheet, fix it. DAC Control - Address 03h DAC Digital Interface Format (Bits 5:4) DAC_DIF1 DAC_DIF0 Description 0 0 Left Justified, up to 24-bit data (default) 0 1 I²S, up to 24-bit data 1 0 Right-Justified, 16-bit Data 1 1 Right-Justified, 24-bit Data Transmitter Control 2 - Address 12h Transmitter Digital Interface Format (Bits 7:6) Tx_DIF1 Tx_DIF0 Description Format Figure 0 0 Left Justified, up to 24-bit data (default) 0 1 I²S, up to 24-bit data 1 0 Right-Justified, 16-bit Data 1 1 Right-Justified, 24-bit Data Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi, I don't have the h/w to test, so please test if this patch works. Thanks, Axel sound/soc/codecs/cs4265.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index d1a77c7..e42a5038 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -457,14 +457,14 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, case SND_SOC_DAIFMT_RIGHT_J: if (params_width(params) == 16) { snd_soc_update_bits(codec, CS4265_DAC_CTL, - CS4265_DAC_CTL_DIF, (1 << 5)); + CS4265_DAC_CTL_DIF, (2 << 4)); snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, - CS4265_SPDIF_CTL2_DIF, (1 << 7)); + CS4265_SPDIF_CTL2_DIF, (2 << 6)); } else { snd_soc_update_bits(codec, CS4265_DAC_CTL, - CS4265_DAC_CTL_DIF, (3 << 5)); + CS4265_DAC_CTL_DIF, (3 << 4)); snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, - CS4265_SPDIF_CTL2_DIF, (1 << 7)); + CS4265_SPDIF_CTL2_DIF, (3 << 6)); } break; case SND_SOC_DAIFMT_LEFT_J: @@ -473,7 +473,7 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, CS4265_ADC_CTL, CS4265_ADC_DIF, 0); snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, - CS4265_SPDIF_CTL2_DIF, (1 << 6)); + CS4265_SPDIF_CTL2_DIF, 0); break; default: -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: cs4265: CS4265_INT_STATUS is readable register
by Axel Lin 24 Jul '15

24 Jul '15

CS4265_INT_STATUS was missed in cs4265_readable_register, fix it. The register 0x01 ~ 0x12 are readable, use CS4265_CHIP_ID ... CS4265_SPDIF_CTL2 syntax for better readability. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs4265.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index e42a5038..55db19d 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -60,23 +60,7 @@ static const struct reg_default cs4265_reg_defaults[] = { static bool cs4265_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS4265_PWRCTL: - case CS4265_DAC_CTL: - case CS4265_ADC_CTL: - case CS4265_MCLK_FREQ: - case CS4265_SIG_SEL: - case CS4265_CHB_PGA_CTL: - case CS4265_CHA_PGA_CTL: - case CS4265_ADC_CTL2: - case CS4265_DAC_CHA_VOL: - case CS4265_DAC_CHB_VOL: - case CS4265_DAC_CTL2: - case CS4265_SPDIF_CTL1: - case CS4265_SPDIF_CTL2: - case CS4265_INT_MASK: - case CS4265_STATUS_MODE_MSB: - case CS4265_STATUS_MODE_LSB: - case CS4265_CHIP_ID: + case CS4265_CHIP_ID ... CS4265_SPDIF_CTL2: return true; default: return false; -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: Use params_width() at appropriate places
by Axel Lin 24 Jul '15

24 Jul '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98925.c | 2 +- sound/soc/codecs/pcm512x.c | 4 ++-- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8776.c | 4 ++-- sound/soc/codecs/wm8996.c | 2 +- 6 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index ebb648a..5990de3 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -430,7 +430,7 @@ static int max98925_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); - switch (snd_pcm_format_width(params_format(params))) { + switch (params_width(params)) { case 16: regmap_update_bits(max98925->regmap, MAX98925_FORMAT, diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index de16429..047c489 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1117,7 +1117,7 @@ static int pcm512x_hw_params(struct snd_pcm_substream *substream, params_rate(params), params_channels(params)); - switch (snd_pcm_format_width(params_format(params))) { + switch (params_width(params)) { case 16: alen = PCM512x_ALEN_16; break; @@ -1132,7 +1132,7 @@ static int pcm512x_hw_params(struct snd_pcm_substream *substream, break; default: dev_err(codec->dev, "Bad frame size: %d\n", - snd_pcm_format_width(params_format(params))); + params_width(params)); return -EINVAL; } diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 35199fc..fd1439e 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1702,7 +1702,7 @@ static int wm2200_hw_params(struct snd_pcm_substream *substream, int *bclk_rates; /* Data sizes if not using TDM */ - wl = snd_pcm_format_width(params_format(params)); + wl = params_width(params); if (wl < 0) return wl; fl = snd_soc_params_to_frame_size(params); diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 8120b0c..c2cdcae 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1408,7 +1408,7 @@ static int wm5100_hw_params(struct snd_pcm_substream *substream, base = dai->driver->base; /* Data sizes if not using TDM */ - wl = snd_pcm_format_width(params_format(params)); + wl = params_width(params); if (wl < 0) return wl; fl = snd_soc_params_to_frame_size(params); diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 592866d..183c9a4 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -265,7 +265,7 @@ static int wm8776_hw_params(struct snd_pcm_substream *substream, } /* Set word length */ - switch (snd_pcm_format_width(params_format(params))) { + switch (params_width(params)) { case 16: iface = 0; break; @@ -280,7 +280,7 @@ static int wm8776_hw_params(struct snd_pcm_substream *substream, break; default: dev_err(codec->dev, "Unsupported sample size: %i\n", - snd_pcm_format_width(params_format(params))); + params_width(params)); return -EINVAL; } diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 2a696db..f7ccd9f 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1780,7 +1780,7 @@ static int wm8996_hw_params(struct snd_pcm_substream *substream, wm8996->rx_rate[dai->id] = params_rate(params); /* Needs looking at for TDM */ - bits = snd_pcm_format_width(params_format(params)); + bits = params_width(params); if (bits < 0) return bits; aifdata |= (bits << WM8996_AIF1TX_WL_SHIFT) | bits; -- 2.1.0

2 1

[alsa-devel] [PATCH v2] ALSA: hda - Fix the headset mic that will not work on Dell desktop machine
by woodrow.shen＠canonical.com 24 Jul '15

24 Jul '15

When the headset was plugged in the Dell desktop, the mic of headset can't be detected and workable. According to the alsa-info, we found the differece between alsa and init_pin_configs on the machine, so we need to add the pin configs to make headset work. Codec: Realtek ALC3234 Vendor Id: 0x10ec0255 Subsystem Id: 0x102806bb BugLink: https://bugs.launchpad.net/bugs/1477900 Signed-off-by: Woodrow Shen <woodrow.shen(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 742fc62..d94c0e3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5473,6 +5473,28 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1e, 0x411111f0}, {0x21, 0x0221103f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170150}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x02011020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221105f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170110}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x01014020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, {0x17, 0x90170140}, -- 2.1.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: pcm1681: Fix setting de-emphasis sampling rate selection
by Axel Lin 24 Jul '15

24 Jul '15

The de-emphasis sampling rate selection is controlled by BIT[3:4] of PCM1681_DEEMPH_CONTROL register. Do proper left shift to set it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/pcm1681.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index b2c990f..1011142 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -102,7 +102,7 @@ static int pcm1681_set_deemph(struct snd_soc_codec *codec) if (val != -1) { regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, - PCM1681_DEEMPH_RATE_MASK, val); + PCM1681_DEEMPH_RATE_MASK, val << 3); enable = 1; } else enable = 0; -- 2.1.0

3 9

[alsa-devel] [PATCH] ASoC: sta529: Remove redundant I2C_FUNC_SMBUS_BYTE_DATA functionality check
by Axel Lin 24 Jul '15

24 Jul '15

This checking is done by regmap_get_i2c_bus() which is called in devm_regmap_init_i2c(). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/sta529.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 3430f44..2cdaca9 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -339,9 +339,6 @@ static int sta529_i2c_probe(struct i2c_client *i2c, struct sta529 *sta529; int ret; - if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) - return -EINVAL; - sta529 = devm_kzalloc(&i2c->dev, sizeof(struct sta529), GFP_KERNEL); if (!sta529) return -ENOMEM; -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: ssm4567: Added ACPI entry for SSM4567 codec
by Harsha Priya 24 Jul '15

24 Jul '15

Added INT343B ACPI ID for the SSM4567 codec Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: M Naveen <naveen.m(a)intel.com> Signed-off-by: M R Sathya Prakash <sathya.prakash.m.r(a)intel.com> --- sound/soc/codecs/ssm4567.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index 938d2cb..af536ae 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -10,6 +10,7 @@ * Licensed under the GPL-2. */ +#include <linux/acpi.h> #include <linux/module.h> #include <linux/init.h> #include <linux/i2c.h> @@ -450,10 +451,21 @@ static const struct i2c_device_id ssm4567_i2c_ids[] = { }; MODULE_DEVICE_TABLE(i2c, ssm4567_i2c_ids); +#ifdef CONFIG_ACPI + +static const struct acpi_device_id ssm4567_acpi_match[] = { + { "INT343B", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, ssm4567_acpi_match); + +#endif + static struct i2c_driver ssm4567_driver = { .driver = { .name = "ssm4567", .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(ssm4567_acpi_match), }, .probe = ssm4567_i2c_probe, .remove = ssm4567_i2c_remove, -- 1.9.1

4 3

[alsa-devel] [PATCH v1] ALSA: hda- Add headset mic pin quirk for a Dell device
by woodrow.shen＠canonical.com 24 Jul '15

24 Jul '15

From: Woodrow Shen <woodrow.shen(a)canonical.com> Fix the headset mic that will not work on Dell desktop machine. BugLink: https://bugs.launchpad.net/bugs/1475553 Signed-off-by: Woodrow Shen <woodrow.shen(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 742fc62..d94c0e3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5473,6 +5473,28 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1e, 0x411111f0}, {0x21, 0x0221103f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170150}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x02011020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221105f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170110}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x01014020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, {0x17, 0x90170140}, -- 2.1.4

4 6

[alsa-devel] [ [PATCH] ALSA: hda- Add headset mic pin quirk for a Dell device]
by woodrow.shen＠canonical.com 24 Jul '15

24 Jul '15

From: woodrowshen <woodrow.shen(a)canonical.com> BugLink: https://bugs.launchpad.net/bugs/1475553 rigned-off-by: woodrowshen <woodrow.shen(a)canonical.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 742fc62..d94c0e3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5473,6 +5473,28 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1e, 0x411111f0}, {0x21, 0x0221103f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170150}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x02011020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221105f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170110}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x01014020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, {0x17, 0x90170140}, -- 2.1.4

1 0

[alsa-devel] [PATCH v2 00/19] Update ALSA, and au0828 drivers to use Managed Media Controller API
by Shuah Khan 23 Jul '15

23 Jul '15

This patch series updates ALSA driver, and au0828 core driver to use Managed Media controller API to share tuner. Please note that Managed Media Controller API and DVB and V4L2 drivers updates to use Media Controller API have been added in a prior patch series. Media Controller API is enhanced with two new interfaces to register and unregister entity_notify hooks to allow drivers to take appropriate actions when as new entities get added to the shared media device. Tested exclusion between digital, analog, and audio to ensure when tuner has an active link to DVB FE, analog, and audio will detect and honor the tuner busy conditions and vice versa. Changes since v1: Link to v1: http://www.spinics.net/lists/linux-media/msg91697.html 1. Fixed Open Issue: ALSA has makes media_entity_pipeline_start() call in irq path. I am seeing warnings that the graph_mutex is unsafe irq lock. Media Controller API updates to start/stop pipeline to be irq safe might be necessary. Maybe there are other MC interfaces that need to be irq safe, but I haven't seen any problems with my testing. graph_mutex is changed to a spinlock. Changed drivers that directly hold the lock for graph walks. media_entity_setup_link() might need to be made IRQ safe. I am running more tests to ensure there is no lock warns when dvb, video, and audio apps. run at the sametime and use MC API (that holds graph_lock) to check for tuner availability. Initial testing looked good so far. 2. Add enable_source handler field to struct media_device Add a new field to enable handler to find source entity for the sink entity and check if it is available, and activate the link using media_entity_setup_link() interface. Bridge driver is expected to implement and set the handler when media_device is registered or when bridge driver finds the media_device during probe. This is to enable the use-case to find tuner entity connected to the decoder entity and check if it is available, and activate the using media_entity_setup_link() if it is available. This hanlder can be invoked from media core (v4l-core, dvb-core) as well as other drivers such as ALSA that control the media device. 3. Changes to v4l2-core, ALSA to use enable_source handler 4. Changes to au0828 bridge driver to implement enable_source handler Note: This series includes a revert to a patch that added media controller entity framework enhancements that implemented entity ops for entity_notify functionality. Entity ops for entity_notify doesn't handle and cover entity create ordering variations that could occur during boot. entity_notify list has been moved to media device level which makes the entity_notify calls to work correctly. History: This patch series has been updated to address comments from 3 previous versions of this series. Links to v3 version for reference are: https://www.mail-archive.com/linux-media%40vger.kernel.org/msg89491.html https://www.mail-archive.com/linux-media@vger.kernel.org/msg89492.html https://www.mail-archive.com/linux-media%40vger.kernel.org/msg89493.html Shuah Khan (19): Revert "[media] media: media controller entity framework enhancements for ALSA" media: Media Controller register/unregister entity_notify API media: Add ALSA Media Controller devnodes media: au8522 change to create MC pad for ALSA Audio Out media: Convert graph_mutex to a spinlock and call it graph_lock media: platform exynos4-is: Update graph_mutex to graph_lock spinlock media: platform omap3isp: Update graph_mutex to graph_lock spinlock media: platform s3c-camif: Update graph_mutex to graph_lock spinlock media: platform vsp1: Update graph_mutex to graph_lock spinlock media: platform xilinx: Update graph_mutex to graph_lock spinlock staging media: davinci_vpfe: Update graph_mutex to graph_lock spinlock staging media: omap4iss: Update graph_mutex to graph_lock spinlock media: Add irq safe Media Controller start/stop pipeline API media: Add enable_source handler field to struct media_device media: v4l-core add v4l_enable_media_tuner() to check for tuner availability media: Change v4l-core to check for tuner availability media: dvb-frontend change to check for tuner availability from open media: au0828 change to use Managed Media Controller API sound/usb: Update ALSA driver to use Managed Media Controller API drivers/media/dvb-core/dvb_frontend.c | 43 ++-- drivers/media/dvb-frontends/au8522.h | 8 + drivers/media/dvb-frontends/au8522_decoder.c | 1 + drivers/media/dvb-frontends/au8522_priv.h | 8 - drivers/media/media-device.c | 60 +++++- drivers/media/media-entity.c | 88 ++++++-- drivers/media/platform/exynos4-is/fimc-isp-video.c | 8 +- drivers/media/platform/exynos4-is/fimc-lite.c | 8 +- drivers/media/platform/exynos4-is/media-dev.c | 14 +- drivers/media/platform/exynos4-is/media-dev.h | 4 +- drivers/media/platform/omap3isp/isp.c | 4 +- drivers/media/platform/omap3isp/ispvideo.c | 4 +- drivers/media/platform/s3c-camif/camif-core.c | 6 +- drivers/media/platform/vsp1/vsp1_video.c | 4 +- drivers/media/platform/xilinx/xilinx-dma.c | 4 +- drivers/media/usb/au0828/au0828-core.c | 184 ++++++++++++----- drivers/media/usb/au0828/au0828-video.c | 72 ++----- drivers/media/usb/au0828/au0828.h | 5 + drivers/media/v4l2-core/v4l2-dev.c | 17 ++ drivers/media/v4l2-core/v4l2-ioctl.c | 29 +++ drivers/media/v4l2-core/videobuf2-core.c | 5 + drivers/staging/media/davinci_vpfe/vpfe_video.c | 12 +- drivers/staging/media/omap4iss/iss.c | 4 +- drivers/staging/media/omap4iss/iss_video.c | 4 +- include/media/media-device.h | 41 +++- include/media/media-entity.h | 7 +- include/media/v4l2-dev.h | 3 + include/uapi/linux/media.h | 5 + sound/usb/Makefile | 15 +- sound/usb/card.c | 5 + sound/usb/card.h | 1 + sound/usb/media.c | 227 +++++++++++++++++++++ sound/usb/media.h | 52 +++++ sound/usb/pcm.c | 15 +- sound/usb/quirks-table.h | 1 + sound/usb/quirks.c | 9 +- sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 1 + 38 files changed, 765 insertions(+), 215 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.1.4

1 20

Re: [alsa-devel] linux-next: Tree for Jul 23 (sound/soc/codecs/sti-sas.c)
by Randy Dunlap 23 Jul '15

23 Jul '15

On 07/22/15 23:57, Stephen Rothwell wrote: > Hi all, > > Changes since 20150722: > on x86_64: ../sound/soc/codecs/sti-sas.c: In function 'stih416_dac_probe': ../sound/soc/codecs/sti-sas.c:227:2: error: implicit declaration of function 'devm_reset_control_get' [-Werror=implicit-function-declaration] dac->rst = devm_reset_control_get(codec->dev, "dac_rst"); ^ ../sound/soc/codecs/sti-sas.c:227:11: warning: assignment makes pointer from integer without a cast [enabled by default] dac->rst = devm_reset_control_get(codec->dev, "dac_rst"); ^ Full randconfig file is attached. -- ~Randy

1 0

[alsa-devel] [PATCH] ASoC: rl6231: add pll preset table
by Bard Liao 23 Jul '15

23 Jul '15

Currently, rl6231_pll_calc provide a working PLL parameters for given freq_in and freq_out. However, in some cases it is not the perfect parameter. For example if freq_in = 19200000 and freq_out = 24576000, the calculated parameter will gengrate 24.5647 MHz which is not exactly the same as what we need. But the PLL can output 24.576 MHz as exactly what we expect if we set the best PLL parameter. To improve it, we put the best match parameters in a preset table. We can search the preset table first, if there is no preset parameter for the given freq_in and freq_out, we can still calculate a working PLL parameter. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rl6231.c | 27 ++++++++++++++++++++++++++- 1 file changed, 26 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index 56650d6..96f3e90 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -43,6 +43,19 @@ int rl6231_calc_dmic_clk(int rate) } EXPORT_SYMBOL_GPL(rl6231_calc_dmic_clk); +struct pll_calc_map { + unsigned int pll_in; + unsigned int pll_out; + int k; + int n; + int m; + bool m_bp; +}; + +static const struct pll_calc_map pll_preset_table[] = { + {19200000, 24576000, 3, 30, 3, false}, +}; + /** * rl6231_pll_calc - Calcualte PLL M/N/K code. * @freq_in: external clock provided to codec. @@ -57,7 +70,7 @@ int rl6231_pll_calc(const unsigned int freq_in, const unsigned int freq_out, struct rl6231_pll_code *pll_code) { int max_n = RL6231_PLL_N_MAX, max_m = RL6231_PLL_M_MAX; - int k, red, n_t, pll_out, in_t, out_t; + int i, k, red, n_t, pll_out, in_t, out_t; int n = 0, m = 0, m_t = 0; int red_t = abs(freq_out - freq_in); bool bypass = false; @@ -65,6 +78,18 @@ int rl6231_pll_calc(const unsigned int freq_in, if (RL6231_PLL_INP_MAX < freq_in || RL6231_PLL_INP_MIN > freq_in) return -EINVAL; + for (i = 0; i < ARRAY_SIZE(pll_preset_table); i++) { + if (freq_in == pll_preset_table[i].pll_in && + freq_out == pll_preset_table[i].pll_out) { + k = pll_preset_table[i].k; + m = pll_preset_table[i].m; + n = pll_preset_table[i].n; + bypass = pll_preset_table[i].m_bp; + pr_debug("Use preset PLL parameter table\n"); + goto code_find; + } + } + k = 100000000 / freq_out - 2; if (k > RL6231_PLL_K_MAX) k = RL6231_PLL_K_MAX; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: xtensa: Remove unnecessary snd_pcm_lib_preallocate_free_for_all()
by Vaishali Thakkar 23 Jul '15

23 Jul '15

The ALSA core takes care that all preallocated memory is freed when the PCM itself is freed. There is no need to do this manually in the driver. Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/xtensa/xtfpga-i2s.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c index 1cfb19e..039f65e 100644 --- a/sound/soc/xtensa/xtfpga-i2s.c +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -474,11 +474,6 @@ static int xtfpga_pcm_new(struct snd_soc_pcm_runtime *rtd) card->dev, size, size); } -static void xtfpga_pcm_free(struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_free_for_all(pcm); -} - static const struct snd_pcm_ops xtfpga_pcm_ops = { .open = xtfpga_pcm_open, .close = xtfpga_pcm_close, @@ -490,7 +485,6 @@ static const struct snd_pcm_ops xtfpga_pcm_ops = { static const struct snd_soc_platform_driver xtfpga_soc_platform = { .pcm_new = xtfpga_pcm_new, - .pcm_free = xtfpga_pcm_free, .ops = &xtfpga_pcm_ops, }; -- 1.9.1

2 1

[alsa-devel] [PATCHv2 0/3] regmap: Use different lockdep class for each regmap init call
by Nicolas Boichat 23 Jul '15

23 Jul '15

Second version. Small fixup in the regmap patch itself [3/3] (the declaration of __devm_regmap_init_ac97 was missing lock_key/name parameters). Also, there are 2 drivers that are using function pointers named "regmap_init", which causes problems as it is now a macro: put these in parentheses to prevent the preprocessor from substituting them. Patches 1+2 should be pushed first to avoid breaking git bissect. Compile tested with allyesconfig on x86_64 and arm. Nicolas Boichat (3): mfd: vexpress: Add parentheses around bridge->ops->regmap_init call thermal: sti: Add parentheses around bridge->ops->regmap_init call regmap: Use different lockdep class for each regmap init call drivers/base/regmap/regmap-ac97.c | 22 +++-- drivers/base/regmap/regmap-i2c.c | 22 +++-- drivers/base/regmap/regmap-mmio.c | 27 ++++-- drivers/base/regmap/regmap-spi.c | 22 +++-- drivers/base/regmap/regmap-spmi.c | 44 +++++---- drivers/base/regmap/regmap.c | 31 +++--- drivers/bus/vexpress-config.c | 2 +- drivers/thermal/st/st_thermal.c | 2 +- include/linux/regmap.h | 192 ++++++++++++++++++++++++++++---------- 9 files changed, 252 insertions(+), 112 deletions(-) -- 2.4.3.573.g4eafbef

3 6

[alsa-devel] [PATCH] ASoC: rt5677: Remove NULL test for desc before gpiod_set_value_cansleep call
by Axel Lin 23 Jul '15

23 Jul '15

It's safe to call gpiod_set_value_cansleep() with NULL desc. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5677.c | 18 ++++++------------ 1 file changed, 6 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 016db7d..8d70911 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4763,10 +4763,8 @@ static int rt5677_remove(struct snd_soc_codec *codec) struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec); - if (rt5677->pow_ldo2) - gpiod_set_value_cansleep(rt5677->pow_ldo2, 0); - if (rt5677->reset_pin) - gpiod_set_value_cansleep(rt5677->reset_pin, 0); + gpiod_set_value_cansleep(rt5677->pow_ldo2, 0); + gpiod_set_value_cansleep(rt5677->reset_pin, 0); return 0; } @@ -4780,10 +4778,8 @@ static int rt5677_suspend(struct snd_soc_codec *codec) regcache_cache_only(rt5677->regmap, true); regcache_mark_dirty(rt5677->regmap); - if (rt5677->pow_ldo2) - gpiod_set_value_cansleep(rt5677->pow_ldo2, 0); - if (rt5677->reset_pin) - gpiod_set_value_cansleep(rt5677->reset_pin, 0); + gpiod_set_value_cansleep(rt5677->pow_ldo2, 0); + gpiod_set_value_cansleep(rt5677->reset_pin, 0); } return 0; @@ -4794,10 +4790,8 @@ static int rt5677_resume(struct snd_soc_codec *codec) struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); if (!rt5677->dsp_vad_en) { - if (rt5677->pow_ldo2) - gpiod_set_value_cansleep(rt5677->pow_ldo2, 1); - if (rt5677->reset_pin) - gpiod_set_value_cansleep(rt5677->reset_pin, 1); + gpiod_set_value_cansleep(rt5677->pow_ldo2, 1); + gpiod_set_value_cansleep(rt5677->reset_pin, 1); if (rt5677->pow_ldo2 || rt5677->reset_pin) msleep(10); -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: tas2552: Fix off-by-one for max_register setting
by Axel Lin 23 Jul '15

23 Jul '15

The latest valid register is TAS2552_VBAT_DATA. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/tas2552.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2552.h b/sound/soc/codecs/tas2552.h index 5746f8f..e34752b 100644 --- a/sound/soc/codecs/tas2552.h +++ b/sound/soc/codecs/tas2552.h @@ -42,7 +42,7 @@ #define TAS2552_BOOST_APT_CTRL 0x14 #define TAS2552_VER_NUM 0x16 #define TAS2552_VBAT_DATA 0x19 -#define TAS2552_MAX_REG 0x20 +#define TAS2552_MAX_REG TAS2552_VBAT_DATA /* CFG1 Register Masks */ #define TAS2552_DEV_RESET (1 << 0) -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: tas5086: Improve the logic for de-emphasis sampling rate selection
by Axel Lin 23 Jul '15

23 Jul '15

Slightly improve the logic for de-emphasis sampling rate selection by break out the loop if the rate is matched. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/tas5086.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 0a49fc8..d49d25d 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -266,10 +266,14 @@ static int tas5086_set_deemph(struct snd_soc_codec *codec) struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); int i, val = 0; - if (priv->deemph) - for (i = 0; i < ARRAY_SIZE(tas5086_deemph); i++) - if (tas5086_deemph[i] == priv->rate) + if (priv->deemph) { + for (i = 0; i < ARRAY_SIZE(tas5086_deemph); i++) { + if (tas5086_deemph[i] == priv->rate) { val = i; + break; + } + } + } return regmap_update_bits(priv->regmap, TAS5086_SYS_CONTROL_1, TAS5086_DEEMPH_MASK, val); -- 2.1.0

3 2

Re: [alsa-devel] Intel Baytrail Linux Audio
by Vinod Koul 23 Jul '15

23 Jul '15

On Thu, Jul 09, 2015 at 02:57:58PM -0400, Tyler Yeomans wrote: > Antonio/Vinod, > > I have made the change in the sst-acpi.c file for the atom specific boards. > I've attached the dmesg output here with dynamic debug enabled. > > To clarify- > > snd-intel-sst-core is loaded > FE-BE fix has been applied > I'm using the ssp0 firmware. > > Interrupts are still 0 > > Still no sound. > > Would leaving the device on and providing remote access (i.e though > teamviewer) assist in any way? The fw download seems to fail on your device, which can also be due to incorrect interrupt mapping... [ 20.386850] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 20.386862] intel_sst_acpi 80860F28:00: fw returned err -16 [ 20.386873] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 20.752559] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 20.752570] intel_sst_acpi 80860F28:00: fw returned err -16 [ 21.163378] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x0 fw_state 0x1 [ 21.163388] intel_sst_acpi 80860F28:00: fw download failed -16 Can you tell me which device you have, what was preinstalled on it -- ~Vinod

2 4

[alsa-devel] [PATCH] ASoC: mediatek: Add suspend/resume callbacks
by Koro Chen 23 Jul '15

23 Jul '15

This adds suspend/resume callbacks, which are common for each DAI. To be able to continue the last playback/capture after resume when suspend was done during a playback/capture, in the callbacks we do backup/restore of registers which were set before prepare stage. Registers to be backup/restore are defined in a backup list array. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-common.h | 8 ---- sound/soc/mediatek/mtk-afe-pcm.c | 77 +++++++++++++++++++++++++++++++++++++ 2 files changed, 77 insertions(+), 8 deletions(-) diff --git a/sound/soc/mediatek/mtk-afe-common.h b/sound/soc/mediatek/mtk-afe-common.h index a88b175..cc4393c 100644 --- a/sound/soc/mediatek/mtk-afe-common.h +++ b/sound/soc/mediatek/mtk-afe-common.h @@ -98,12 +98,4 @@ struct mtk_afe_memif { const struct mtk_afe_irq_data *irqdata; }; -struct mtk_afe { - /* address for ioremap audio hardware register */ - void __iomem *base_addr; - struct device *dev; - struct regmap *regmap; - struct mtk_afe_memif memif[MTK_AFE_MEMIF_NUM]; - struct clk *clocks[MTK_CLK_NUM]; -}; #endif diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index de38d2e..55471ee 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -45,18 +45,21 @@ /* Memory interface */ #define AFE_DL1_BASE 0x0040 #define AFE_DL1_CUR 0x0044 +#define AFE_DL1_END 0x0048 #define AFE_DL2_BASE 0x0050 #define AFE_DL2_CUR 0x0054 #define AFE_AWB_BASE 0x0070 #define AFE_AWB_CUR 0x007c #define AFE_VUL_BASE 0x0080 #define AFE_VUL_CUR 0x008c +#define AFE_VUL_END 0x0088 #define AFE_DAI_BASE 0x0090 #define AFE_DAI_CUR 0x009c #define AFE_MOD_PCM_BASE 0x0330 #define AFE_MOD_PCM_CUR 0x033c #define AFE_HDMI_OUT_BASE 0x0374 #define AFE_HDMI_OUT_CUR 0x0378 +#define AFE_HDMI_OUT_END 0x037c #define AFE_ADDA2_TOP_CON0 0x0600 @@ -127,6 +130,34 @@ enum afe_tdm_ch_start { AFE_TDM_CH_ZERO, }; +static const unsigned int mtk_afe_backup_list[] = { + AUDIO_TOP_CON0, + AFE_CONN1, + AFE_CONN2, + AFE_CONN7, + AFE_CONN8, + AFE_DAC_CON1, + AFE_DL1_BASE, + AFE_DL1_END, + AFE_VUL_BASE, + AFE_VUL_END, + AFE_HDMI_OUT_BASE, + AFE_HDMI_OUT_END, + AFE_HDMI_CONN0, + AFE_DAC_CON0, +}; + +struct mtk_afe { + /* address for ioremap audio hardware register */ + void __iomem *base_addr; + struct device *dev; + struct regmap *regmap; + struct mtk_afe_memif memif[MTK_AFE_MEMIF_NUM]; + struct clk *clocks[MTK_CLK_NUM]; + unsigned int backup_regs[ARRAY_SIZE(mtk_afe_backup_list)]; + bool suspended; +}; + static const struct snd_pcm_hardware mtk_afe_hardware = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP_VALID), @@ -722,11 +753,53 @@ static const struct snd_soc_dai_ops mtk_afe_hdmi_ops = { }; +static int mtk_afe_runtime_suspend(struct device *dev); +static int mtk_afe_runtime_resume(struct device *dev); + +static int mtk_afe_dai_suspend(struct snd_soc_dai *dai) +{ + struct mtk_afe *afe = snd_soc_dai_get_drvdata(dai); + int i; + + dev_dbg(afe->dev, "%s\n", __func__); + if (pm_runtime_status_suspended(afe->dev) || afe->suspended) + return 0; + + for (i = 0; i < ARRAY_SIZE(mtk_afe_backup_list); i++) + regmap_read(afe->regmap, mtk_afe_backup_list[i], + &afe->backup_regs[i]); + + afe->suspended = true; + mtk_afe_runtime_suspend(afe->dev); + return 0; +} + +static int mtk_afe_dai_resume(struct snd_soc_dai *dai) +{ + struct mtk_afe *afe = snd_soc_dai_get_drvdata(dai); + int i = 0; + + dev_dbg(afe->dev, "%s\n", __func__); + if (pm_runtime_status_suspended(afe->dev) || !afe->suspended) + return 0; + + mtk_afe_runtime_resume(afe->dev); + + for (i = 0; i < ARRAY_SIZE(mtk_afe_backup_list); i++) + regmap_write(afe->regmap, mtk_afe_backup_list[i], + afe->backup_regs[i]); + + afe->suspended = false; + return 0; +} + static struct snd_soc_dai_driver mtk_afe_pcm_dais[] = { /* FE DAIs: memory intefaces to CPU */ { .name = "DL1", /* downlink 1 */ .id = MTK_AFE_MEMIF_DL1, + .suspend = mtk_afe_dai_suspend, + .resume = mtk_afe_dai_resume, .playback = { .stream_name = "DL1", .channels_min = 1, @@ -738,6 +811,8 @@ static struct snd_soc_dai_driver mtk_afe_pcm_dais[] = { }, { .name = "VUL", /* voice uplink */ .id = MTK_AFE_MEMIF_VUL, + .suspend = mtk_afe_dai_suspend, + .resume = mtk_afe_dai_resume, .capture = { .stream_name = "VUL", .channels_min = 1, @@ -774,6 +849,8 @@ static struct snd_soc_dai_driver mtk_afe_hdmi_dais[] = { { .name = "HDMI", .id = MTK_AFE_MEMIF_HDMI, + .suspend = mtk_afe_dai_suspend, + .resume = mtk_afe_dai_resume, .playback = { .stream_name = "HDMI", .channels_min = 2, -- 1.8.1.1.dirty

2 1

[alsa-devel] [PATCH 1/2] ASoC: cs35l32: Fix define for CS35L32_SDOUT_3ST
by Axel Lin 23 Jul '15

23 Jul '15

According to the datasheet, the CS35L32_SDOUT_3ST is BIT(3). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs35l32.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l32.h b/sound/soc/codecs/cs35l32.h index 31ab804..1d6c250 100644 --- a/sound/soc/codecs/cs35l32.h +++ b/sound/soc/codecs/cs35l32.h @@ -80,7 +80,7 @@ struct cs35l32_platform_data { #define CS35L32_GAIN_MGR_MASK 0x08 #define CS35L32_ADSP_SHARE_MASK 0x08 #define CS35L32_ADSP_DATACFG_MASK 0x30 -#define CS35L32_SDOUT_3ST 0x80 +#define CS35L32_SDOUT_3ST 0x08 #define CS35L32_BATT_REC_MASK 0x0E #define CS35L32_BATT_THRESH_MASK 0x30 -- 2.1.0

3 4

[alsa-devel] [PATCH] ASoC: cs42l73: Fix mask for setting CS42L73_SP_3ST bit
by Axel Lin 23 Jul '15

23 Jul '15

CS42L73_SP_3ST is BIT(7), so the mask field is wrong. Fix it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l73.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 7cb1d70..be6477f 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1236,8 +1236,8 @@ static int cs42l73_set_tristate(struct snd_soc_dai *dai, int tristate) struct snd_soc_codec *codec = dai->codec; int id = dai->id; - return snd_soc_update_bits(codec, CS42L73_SPC(id), - 0x7F, tristate << 7); + return snd_soc_update_bits(codec, CS42L73_SPC(id), CS42L73_SP_3ST, + tristate << 7); } static const struct snd_pcm_hw_constraint_list constraints_12_24 = { -- 2.1.0

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[alsa-devel] [PATCH 0/4] i915 to call hda driver on HDMI plug/unplug
by David Henningsson 23 Jul '15

23 Jul '15

This patch set aims to resolve three problems: * The first - and most serious one - is that the audio driver is not woken up properly when in power save modes, especially not when the HDA controller is in D3. By having the i915 driver call directly into the hda driver, the HDA driver is always notified that an HDMI hotplug event has happened. * Second, there is currently no way for userspace to match an HDMI audio output with an HDMI video output. We fix this by sending connector_type and connector_type_id in the HDMI hotplug callback. * Third, writing ELD info to the hardware just so the HDA driver can read it from the hardware seems a bit inefficient. We could just pass that information in the callback, too. The patch in its current form fixes the first of these problems and provides most of the infrastructure for the second and third problem. The patch set is based on 4.2rc2 + my recent codec wakeup patch. So far, it has been tested (and working) on one Skylake machine. David Henningsson (4): drm/i915: Add audio hotplug info struct drm/i915: Call audio hotplug notify function ALSA: hda - Dispatch incoming HDMI hotplug i915 callback ALSA: hda - Wake the codec up on hotplug notify events drivers/gpu/drm/i915/i915_drv.h | 1 + drivers/gpu/drm/i915/intel_audio.c | 46 ++++++++++++++++++++++++++++++++++++ include/drm/i915_component.h | 19 +++++++++++++++ include/sound/hdaudio.h | 4 ++++ sound/hda/hdac_i915.c | 24 +++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 22 ++++++++++++++++- 6 files changed, 115 insertions(+), 1 deletion(-) -- 1.7.9.5

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[alsa-devel] [PATCH v2 0/4] i915 to call hda driver on HDMI plug/unplug
by David Henningsson 22 Jul '15

22 Jul '15

Changes since previous version: * Renames and refactorings according to Daniel's and Takashi's suggestions. * Debug message slightly improved. * Braswell now has all three pins too, Baytrail still has only one pin. It would still be good with some confirmation from Intel that this is correct. David Henningsson (4): drm/i915: Add audio hotplug info struct drm/i915: Call audio hotplug notify function ALSA: hda - Dispatch incoming HDMI hotplug i915 callback ALSA: hda - Wake the codec up on hotplug notify events drivers/gpu/drm/i915/i915_drv.h | 1 + drivers/gpu/drm/i915/intel_audio.c | 41 ++++++++++++++++++++++++++++++++++++ include/drm/i915_component.h | 18 ++++++++++++++++ include/sound/hdaudio.h | 4 ++++ sound/hda/hdac_i915.c | 26 +++++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 13 +++++++++++- 6 files changed, 102 insertions(+), 1 deletion(-) -- 1.7.9.5

1 4

[alsa-devel] [RFC][PATCH 00/37] ALSA: firewire: support AMDTP variants
by Takashi Sakamoto 22 Jul '15

22 Jul '15

Hi, This patchset is for three new drivers and related modification to firewire-lib module. Overview: * Patch 01-02: for firewire-lib, to split AMDTP layer to two parts * Patch 03-07: for firewire-lib, to add MIDI asynchronous transaction * Patch 08-16: for firewire-digi00x, a new driver for Digi 002/003 family * Patch 17-26: for firewire-tascam, a new driver for TASCAM FireWire series * Patch 27-28: for firewire-lib, to add SPH field in CIP header support * Patch 29-37: for firewire-motu, a new driver for MOTU FireWire series Status: * firewire-digi00x: * well-prepared, except for final testing with Digi 003 series * firewire-tascam: * well-prepared, except for conversion of controll messages to MIDI * firewire-motu: * experimental, for preview to developers As the other my RFCs, I prepare for a branch in my backport repository, to make it easy to test this patchset. https://github.com/takaswie/snd-firewire-improve/tree/amdtp-variants I'm happy if receiving your comments, reviews, tests and your patches. I have a plan to posting patches again to merging these well-prepared drivers till next merge window for Linux 4.3. Cc: Damien Zammit <damien(a)zamaudio.com> Cc: Robin Gareus <robin(a)gareus.org> Cc: Yoshifuji Hideaki <yoshfuji(a)linux-ipv6.org> Regards Takashi Sakamoto (37): ALSA: firewire-lib: rename 'amdtp' to 'amdtp-stream' for functional separation ALSA: firewire-lib: functional separation between packet stream and data block format ALSA: firewire-lib: add helper functions for asynchronous MIDI port ALSA: firewire-lib: serialize request transaction and response transaction ALSA: firewire-lib: schedule tasklet again when MIDI substream has rest of MIDI messages ALSA: firewire-lib: add throttle for MIDI data rate ALSA: firewire-lib: avoid endless loop to transfer MIDI messages at fatal error ALSA: firewire-digi00x: add skelton for Digi 002/003 family ALSA: firewire-digi00x: add protocol layer ALSA: firewire-digi00x: add stream functionality ALSA: firewire-digi00x: add proc node to show clock status ALSA: firewire-digi00x: add PCM functionality ALSA: firewire-digi00x: add MIDI functionality ALSA: firewire-digi00x: add hwdep interface ALSA: firewire-digi00x: add support for asynchronous messaging ALSA: firewire-digi00x: add support for MIDI ports for machine control ALSA: firewire-tascam: add skeleton for TASCAM FireWire series ALSA: firewire-tascam: add a structure for model-dependent parameters. ALSA: firewire-tascam: add proc node to show firmware information ALSA: firewire-tascam: add specific protocol layer ALSA: firewire-tascam: add streaming functionality ALSA: firewire-tascam: add PCM functionality ALSA: firewire-tascam: add transaction functionality ALSA: firewire-tascam: add MIDI functionality ALSA: firewire-tascam: add hwdep interface ALSA: firewire-tascam: add MMAP support to show status and control message ALSA: firewire-lib: add support for source packet header field in CIP header ALSA: firewire-lib: enable protocol layer to handle current cycle count ALSA: firewire-motu: add skeleton for Mark of the unicorn (MOTU) FireWire series ALSA: firewire-motu: add a structure for model-dependent parameters. ALSA: firewire-motu: add MOTU specific protocol layer ALSA: firewire-motu: add stream functionality ALSA: firewire-motu: add transaction functionality ALSA: firewire-motu: add proc node to show the status of internal clock ALSA: firewire-motu: add PCM functionality ALSA: firewire-motu: add MIDI functionality ALSA: firewire-motu: add hwdep interface include/uapi/sound/asound.h | 5 +- include/uapi/sound/firewire.h | 16 +- sound/firewire/Kconfig | 38 + sound/firewire/Makefile | 5 +- sound/firewire/amdtp-am824.c | 432 ++++++++++ sound/firewire/amdtp-am824.h | 45 ++ sound/firewire/amdtp-stream.c | 882 ++++++++++++++++++++ sound/firewire/amdtp-stream.h | 253 ++++++ sound/firewire/amdtp.c | 1107 -------------------------- sound/firewire/amdtp.h | 296 ------- sound/firewire/bebob/bebob.h | 2 +- sound/firewire/bebob/bebob_midi.c | 16 +- sound/firewire/bebob/bebob_pcm.c | 10 +- sound/firewire/bebob/bebob_stream.c | 34 +- sound/firewire/dice/dice-midi.c | 16 +- sound/firewire/dice/dice-pcm.c | 10 +- sound/firewire/dice/dice-stream.c | 41 +- sound/firewire/dice/dice.h | 2 +- sound/firewire/digi00x/Makefile | 4 + sound/firewire/digi00x/amdtp-dot.c | 425 ++++++++++ sound/firewire/digi00x/digi00x-hwdep.c | 200 +++++ sound/firewire/digi00x/digi00x-midi.c | 198 +++++ sound/firewire/digi00x/digi00x-pcm.c | 355 +++++++++ sound/firewire/digi00x/digi00x-proc.c | 99 +++ sound/firewire/digi00x/digi00x-stream.c | 422 ++++++++++ sound/firewire/digi00x/digi00x-transaction.c | 137 ++++ sound/firewire/digi00x/digi00x.c | 174 ++++ sound/firewire/digi00x/digi00x.h | 155 ++++ sound/firewire/fcp.c | 2 +- sound/firewire/fireworks/fireworks.c | 12 +- sound/firewire/fireworks/fireworks.h | 2 +- sound/firewire/fireworks/fireworks_midi.c | 16 +- sound/firewire/fireworks/fireworks_pcm.c | 8 +- sound/firewire/fireworks/fireworks_stream.c | 8 +- sound/firewire/lib.c | 127 +++ sound/firewire/lib.h | 52 ++ sound/firewire/motu/Makefile | 3 + sound/firewire/motu/amdtp-motu.c | 442 ++++++++++ sound/firewire/motu/motu-hwdep.c | 210 +++++ sound/firewire/motu/motu-midi.c | 151 ++++ sound/firewire/motu/motu-pcm.c | 397 +++++++++ sound/firewire/motu/motu-proc.c | 73 ++ sound/firewire/motu/motu-stream.c | 670 ++++++++++++++++ sound/firewire/motu/motu-transaction.c | 97 +++ sound/firewire/motu/motu.c | 311 ++++++++ sound/firewire/motu/motu.h | 127 +++ sound/firewire/oxfw/oxfw-midi.c | 16 +- sound/firewire/oxfw/oxfw-pcm.c | 8 +- sound/firewire/oxfw/oxfw-stream.c | 11 +- sound/firewire/oxfw/oxfw.h | 2 +- sound/firewire/tascam/Makefile | 4 + sound/firewire/tascam/amdtp-tascam.c | 236 ++++++ sound/firewire/tascam/tascam-hwdep.c | 232 ++++++ sound/firewire/tascam/tascam-midi.c | 158 ++++ sound/firewire/tascam/tascam-pcm.c | 306 +++++++ sound/firewire/tascam/tascam-proc.c | 88 ++ sound/firewire/tascam/tascam-stream.c | 512 ++++++++++++ sound/firewire/tascam/tascam-transaction.c | 234 ++++++ sound/firewire/tascam/tascam.c | 232 ++++++ sound/firewire/tascam/tascam.h | 141 ++++ 60 files changed, 8760 insertions(+), 1507 deletions(-) create mode 100644 sound/firewire/amdtp-am824.c create mode 100644 sound/firewire/amdtp-am824.h create mode 100644 sound/firewire/amdtp-stream.c create mode 100644 sound/firewire/amdtp-stream.h delete mode 100644 sound/firewire/amdtp.c delete mode 100644 sound/firewire/amdtp.h create mode 100644 sound/firewire/digi00x/Makefile create mode 100644 sound/firewire/digi00x/amdtp-dot.c create mode 100644 sound/firewire/digi00x/digi00x-hwdep.c create mode 100644 sound/firewire/digi00x/digi00x-midi.c create mode 100644 sound/firewire/digi00x/digi00x-pcm.c create mode 100644 sound/firewire/digi00x/digi00x-proc.c create mode 100644 sound/firewire/digi00x/digi00x-stream.c create mode 100644 sound/firewire/digi00x/digi00x-transaction.c create mode 100644 sound/firewire/digi00x/digi00x.c create mode 100644 sound/firewire/digi00x/digi00x.h create mode 100644 sound/firewire/motu/Makefile create mode 100644 sound/firewire/motu/amdtp-motu.c create mode 100644 sound/firewire/motu/motu-hwdep.c create mode 100644 sound/firewire/motu/motu-midi.c create mode 100644 sound/firewire/motu/motu-pcm.c create mode 100644 sound/firewire/motu/motu-proc.c create mode 100644 sound/firewire/motu/motu-stream.c create mode 100644 sound/firewire/motu/motu-transaction.c create mode 100644 sound/firewire/motu/motu.c create mode 100644 sound/firewire/motu/motu.h create mode 100644 sound/firewire/tascam/Makefile create mode 100644 sound/firewire/tascam/amdtp-tascam.c create mode 100644 sound/firewire/tascam/tascam-hwdep.c create mode 100644 sound/firewire/tascam/tascam-midi.c create mode 100644 sound/firewire/tascam/tascam-pcm.c create mode 100644 sound/firewire/tascam/tascam-proc.c create mode 100644 sound/firewire/tascam/tascam-stream.c create mode 100644 sound/firewire/tascam/tascam-transaction.c create mode 100644 sound/firewire/tascam/tascam.c create mode 100644 sound/firewire/tascam/tascam.h -- 2.1.4

4 50

[alsa-devel] pulseaudio external mic jack detection - ASoC based card
by maruthi srinivas 22 Jul '15

22 Jul '15

Hi, I have a issue with mic jack status in Ubuntu "sound settings" UI. When external mic is removed, UI still shows Mic. When compared to HD Azalia driver, UI shouldn't show any external Mic jack information, if it is removed. Codec driver I use reports jack status properly and uses snd_soc_jack_report(). /proc/interrupts, /dev/input/eventX reflects jack activity. But the same is not reflected in UI. I want to know how pulseaudio detect jack status for drivers which follow ALSA SoC architecture. Does it read from any event from /dev/input/eventX ? Please help in understanding. Regards, Maruthi

4 4

[alsa-devel] [PATCH RFT] ASoC: sgtl5000: Fix up define for SGTL5000_SMALL_POP
by Axel Lin 22 Jul '15

22 Jul '15

Currently, below code actually does not update any bit because SGTL5000_SMALL_POP is 0. snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, SGTL5000_SMALL_POP, 1); The SGTL5000_SMALL_POP should be BIT(0) rather than 0, fix it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi, I don't have h/w and datasheet, but the code looks obvious does not make sense. Can someone help to confirm and test if this patch? Thanks, Axel sound/soc/codecs/sgtl5000.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index bd7a344..1c317de 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -275,7 +275,7 @@ #define SGTL5000_BIAS_CTRL_MASK 0x000e #define SGTL5000_BIAS_CTRL_SHIFT 1 #define SGTL5000_BIAS_CTRL_WIDTH 3 -#define SGTL5000_SMALL_POP 0 +#define SGTL5000_SMALL_POP 1 /* * SGTL5000_CHIP_MIC_CTRL -- 2.1.0

5 5

[alsa-devel] [PATCH] ASoC: kirkwood: prevent double streaming
by Jean-Francois Moine 22 Jul '15

22 Jul '15

The kirkwood audio subsystem presents 2 PCM's for one source. Streaming on a second PCM while the first one is active cuts this last one. Then, ending the last stream gives a kernel trap in free_irq(). Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-dma.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 4cf2245..dbfdfe9 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -148,10 +148,14 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) dram = mv_mbus_dram_info(); addr = substream->dma_buffer.addr; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (priv->substream_play) + return -EBUSY; priv->substream_play = substream; kirkwood_dma_conf_mbus_windows(priv->io, KIRKWOOD_PLAYBACK_WIN, addr, dram); } else { + if (priv->substream_rec) + return -EBUSY; priv->substream_rec = substream; kirkwood_dma_conf_mbus_windows(priv->io, KIRKWOOD_RECORD_WIN, addr, dram); -- 2.1.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: dapm: Add helper function to free a widget
by Lars-Peter Clausen 22 Jul '15

22 Jul '15

snd_soc_tplg_widget_remove_all() has a verbatim copy of an older version of the widget freeing code from dapm_free_widgets(). Add a new helper function that takes care of freeing a widget and use it in both places. This removes the duplicated code and also makes sure that future changes to the widget freeing code only have to be made in one location. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Series depends on fix/dapm --- include/sound/soc-dapm.h | 1 + sound/soc/soc-dapm.c | 38 ++++++++++++++++++++++---------------- sound/soc/soc-topology.c | 25 +------------------------ 3 files changed, 24 insertions(+), 40 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 37d95a8..cadc7fc 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -397,6 +397,7 @@ int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num); int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num); +void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w); /* dapm events */ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1992568..7a57b01 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2315,30 +2315,36 @@ static void dapm_free_path(struct snd_soc_dapm_path *path) kfree(path); } +void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_path *p, *next_p; + + list_del(&w->list); + /* + * remove source and sink paths associated to this widget. + * While removing the path, remove reference to it from both + * source and sink widgets so that path is removed only once. + */ + list_for_each_entry_safe(p, next_p, &w->sources, list_sink) + dapm_free_path(p); + + list_for_each_entry_safe(p, next_p, &w->sinks, list_source) + dapm_free_path(p); + + kfree(w->kcontrols); + kfree(w->name); + kfree(w); +} + /* free all dapm widgets and resources */ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_widget *w, *next_w; - struct snd_soc_dapm_path *p, *next_p; list_for_each_entry_safe(w, next_w, &dapm->card->widgets, list) { if (w->dapm != dapm) continue; - list_del(&w->list); - /* - * remove source and sink paths associated to this widget. - * While removing the path, remove reference to it from both - * source and sink widgets so that path is removed only once. - */ - list_for_each_entry_safe(p, next_p, &w->sources, list_sink) - dapm_free_path(p); - - list_for_each_entry_safe(p, next_p, &w->sinks, list_source) - dapm_free_path(p); - - kfree(w->kcontrols); - kfree(w->name); - kfree(w); + snd_soc_dapm_free_widget(w); } } diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 4a838e4..df3ddf0 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1734,7 +1734,6 @@ void snd_soc_tplg_widget_remove_all(struct snd_soc_dapm_context *dapm, u32 index) { struct snd_soc_dapm_widget *w, *next_w; - struct snd_soc_dapm_path *p, *next_p; list_for_each_entry_safe(w, next_w, &dapm->card->widgets, list) { @@ -1746,31 +1745,9 @@ void snd_soc_tplg_widget_remove_all(struct snd_soc_dapm_context *dapm, if (w->dobj.index != index && w->dobj.index != SND_SOC_TPLG_INDEX_ALL) continue; - - list_del(&w->list); - - /* - * remove source and sink paths associated to this widget. - * While removing the path, remove reference to it from both - * source and sink widgets so that path is removed only once. - */ - list_for_each_entry_safe(p, next_p, &w->sources, list_sink) { - list_del(&p->list_sink); - list_del(&p->list_source); - list_del(&p->list); - kfree(p); - } - list_for_each_entry_safe(p, next_p, &w->sinks, list_source) { - list_del(&p->list_sink); - list_del(&p->list_source); - list_del(&p->list); - kfree(p); - } /* check and free and dynamic widget kcontrols */ snd_soc_tplg_widget_remove(w); - kfree(w->kcontrols); - kfree(w->name); - kfree(w); + snd_soc_dapm_free_widget(w); } } EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_remove_all); -- 2.1.4

2 3

[alsa-devel] [PATCH] ASoC: ssm4567: Keep TDM_BCLKS in ssm4567_set_dai_fmt
by Ben Zhang 22 Jul '15

22 Jul '15

The regmap_write in ssm4567_set_dai_fmt accidentally clears the TDM_BCLKS field which was set earlier by ssm4567_set_tdm_slot. This patch fixes it by using regmap_update_bits with proper mask. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/ssm4567.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index 4662587..a757ab8 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -315,7 +315,13 @@ static int ssm4567_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) if (invert_fclk) ctrl1 |= SSM4567_SAI_CTRL_1_FSYNC; - return regmap_write(ssm4567->regmap, SSM4567_REG_SAI_CTRL_1, ctrl1); + return regmap_update_bits(ssm4567->regmap, SSM4567_REG_SAI_CTRL_1, + SSM4567_SAI_CTRL_1_BCLK | + SSM4567_SAI_CTRL_1_FSYNC | + SSM4567_SAI_CTRL_1_LJ | + SSM4567_SAI_CTRL_1_TDM | + SSM4567_SAI_CTRL_1_PDM, + ctrl1); } static int ssm4567_set_power(struct ssm4567 *ssm4567, bool enable) -- 2.4.3.573.g4eafbef

4 3

[alsa-devel] [PATCH] ALSA: hda - Add headset mic pin quirk for a Dell device
by David Henningsson 22 Jul '15

22 Jul '15

Without this patch, the headset mic will not work on this machine. BugLink: https://bugs.launchpad.net/bugs/1476987 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index caba66b..11c8a74 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5463,6 +5463,17 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1d, 0x40700001}, {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170130}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x01014020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221103f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, {0x17, 0x90170140}, -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - remove one pin from ALC292_STANDARD_PINS
by Hui Wang 22 Jul '15

22 Jul '15

One more Dell laptop with alc293 codec needs ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, but the pin 0x1e does not match the corresponding one in the ALC292_STANDARD_PINS. To use this macro for this machine, we need to remove pin 0x1e from it. BugLink: https://bugs.launchpad.net/bugs/1476888 Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 26 +++++++++++++++++++------- 1 file changed, 19 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index caba66b..ee6f13a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5430,8 +5430,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {0x15, 0x0221401f}, \ {0x1a, 0x411111f0}, \ {0x1b, 0x411111f0}, \ - {0x1d, 0x40700001}, \ - {0x1e, 0x411111f0} + {0x1d, 0x40700001} #define ALC298_STANDARD_PINS \ {0x18, 0x411111f0}, \ @@ -5690,35 +5689,48 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x13, 0x411111f0}, {0x16, 0x01014020}, {0x18, 0x411111f0}, - {0x19, 0x01a19030}), + {0x19, 0x01a19030}, + {0x1e, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0292, 0x1028, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x12, 0x90a60140}, {0x13, 0x411111f0}, {0x16, 0x01014020}, {0x18, 0x02a19031}, - {0x19, 0x01a1903e}), + {0x19, 0x01a1903e}, + {0x1e, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0292, 0x1028, "Dell", ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x12, 0x90a60140}, {0x13, 0x411111f0}, {0x16, 0x411111f0}, {0x18, 0x411111f0}, - {0x19, 0x411111f0}), + {0x19, 0x411111f0}, + {0x1e, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x12, 0x40000000}, {0x13, 0x90a60140}, {0x16, 0x21014020}, {0x18, 0x411111f0}, - {0x19, 0x21a19030}), + {0x19, 0x21a19030}, + {0x1e, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x12, 0x40000000}, {0x13, 0x90a60140}, {0x16, 0x411111f0}, {0x18, 0x411111f0}, - {0x19, 0x411111f0}), + {0x19, 0x411111f0}, + {0x1e, 0x411111f0}), + SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC292_STANDARD_PINS, + {0x12, 0x40000000}, + {0x13, 0x90a60140}, + {0x16, 0x21014020}, + {0x18, 0x411111f0}, + {0x19, 0x21a19030}, + {0x1e, 0x411111ff}), SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC298_STANDARD_PINS, {0x12, 0x90a60130}, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: dapm: Don't add prefix to widget stream name
by Lars-Peter Clausen 21 Jul '15

21 Jul '15

Commit fdb6eb0a1287 ("ASoC: dapm: Modify widget stream name according to prefix") fixed the case where a DAPM route between a DAI widget and a DAC/ADC/AIF widget with a matching stream name was not created when the DAPM context was using a prefix. Unfortunately the patch introduced a few issues on its own like leaking the dynamically allocated stream name memory and also not checking whether the allocation succeeded in the first place. It is also incomplete in that it still does not handle the case where stream name of the widget is a substring of the stream name of the DAI, which is explicitly allowed and works fine if no DAPM prefix is used. Revert the commit and take a slightly different approach to solving the issue. Instead of comparing the widget's stream name to the name of the DAI widget compare it to the stream name of the DAI widget. The stream name of the DAI widget is identical to the name of the DAI widget except that it wont have the DAPM prefix added. So this approach behaves identical regardless to whether the DAPM context uses a prefix or not. We don't have to worry about potentially matching with a widget with the same stream name, but from a different DAPM context with a different prefix, since the code already makes sure that both the DAI widget and the matched widget are from the same DAPM context. Fixes: fdb6eb0a1287 ("ASoC: dapm: Modify widget stream name according to prefix") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1628f0c..1992568 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3344,16 +3344,10 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, } prefix = soc_dapm_prefix(dapm); - if (prefix) { + if (prefix) w->name = kasprintf(GFP_KERNEL, "%s %s", prefix, widget->name); - if (widget->sname) - w->sname = kasprintf(GFP_KERNEL, "%s %s", prefix, - widget->sname); - } else { + else w->name = kasprintf(GFP_KERNEL, "%s", widget->name); - if (widget->sname) - w->sname = kasprintf(GFP_KERNEL, "%s", widget->sname); - } if (w->name == NULL) { kfree(w); return NULL; @@ -3802,7 +3796,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) break; } - if (!w->sname || !strstr(w->sname, dai_w->name)) + if (!w->sname || !strstr(w->sname, dai_w->sname)) continue; if (dai_w->id == snd_soc_dapm_dai_in) { -- 2.1.4

3 4

[alsa-devel] [patch] ASoC: max98925: reading beyond the end of the array
by Dan Carpenter 21 Jul '15

21 Jul '15

Debug prints are seldom useful and this one has an annoying thing where it reads one space beyond the end of the array on error. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index e1753c8..ebb648a 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -271,8 +271,6 @@ static inline int max98925_rate_value(struct snd_soc_codec *codec, break; } } - dev_dbg(codec->dev, "%s: sample rate is %d, returning %d\n", - __func__, rate_table[i].rate, *value); return ret; }

2 4

Re: [alsa-devel] ASoC: rsnd: show debug message for SSI/SRC/DVC connection
by Dan Carpenter 21 Jul '15

21 Jul '15

Hello Kuninori Morimoto, This is a semi-automatic email about new static checker warnings. The patch 84e95355602c: "ASoC: rsnd: show debug message for SSI/SRC/DVC connection" from Jul 15, 2015, leads to the following Smatch complaint: sound/soc/sh/rcar/core.c:338 rsnd_dai_connect() warn: variable dereferenced before check 'mod' (see line 335) sound/soc/sh/rcar/core.c 334 { 335 struct rsnd_priv *priv = rsnd_mod_to_priv(mod); ^^^^^^^^^^^^^^^^^^^^^ Patch adds new dereference. 336 struct device *dev = rsnd_priv_to_dev(priv); 337 338 if (!mod) ^^^^ Existing code assumed "mod" could be NULL. 339 return -EIO; 340 regards, dan carpenter

3 2

[alsa-devel] [patch] ASoC: sti: error handling bug in sti_uniperiph_cpu_dai_of()
by Dan Carpenter 21 Jul '15

21 Jul '15

There is a stray '!' which means the condition is never true. Fixes: f3bd847eb0a7 ('ASoC: sti: Add uniperipheral dai driver') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index 51f745c..dffabf3 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -175,7 +175,7 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node, UNIPERIF_FIFO_DATA_OFFSET(uni); uni->irq = platform_get_irq(priv->pdev, 0); - if (!uni->irq < 0) { + if (uni->irq < 0) { dev_err(dev, "Failed to get IRQ resource"); return -ENXIO; }

2 1

[alsa-devel] [PATCH] ALSA: hda - Add new GPU codec ID 0x10de007d to snd-hda
by Aaron Plattner 21 Jul '15

21 Jul '15

Vendor ID 0x10de007d is used by a yet-to-be-named GPU chip. Signed-off-by: Aaron Plattner <aplattner(a)nvidia.com> --- This chip also has the 2-ch audio swapping bug, so patch_nvhdmi is appropriate here. sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 95158914cc6c..a97db5fc8a15 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3512,6 +3512,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0070, .name = "GPU 70 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0071, .name = "GPU 71 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0072, .name = "GPU 72 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de007d, .name = "GPU 7d HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, { .id = 0x11069f81, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, @@ -3576,6 +3577,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de0070"); MODULE_ALIAS("snd-hda-codec-id:10de0071"); MODULE_ALIAS("snd-hda-codec-id:10de0072"); +MODULE_ALIAS("snd-hda-codec-id:10de007d"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_ALIAS("snd-hda-codec-id:11069f80"); MODULE_ALIAS("snd-hda-codec-id:11069f81"); -- 2.4.6

2 2

[alsa-devel] [PATCH 0/7] Update ALSA, and au0828 drivers to use Managed Media Controller API
by Shuah Khan 20 Jul '15

20 Jul '15

s patch series updates ALSA driver, and au0828 core driver to use Managed Media controller API to share tuner. Please note that Managed Media Controller API and DVB and V4L2 drivers updates to use Media Controller API have been added in a prior patch series. Media Controller API is enhanced with two new interfaces to register and unregister entity_notify hooks to allow drivers to take appropriate actions when as new entities get added to the shared media device. Tested exclusion between digital, analog, and audio to ensure when tuner has an active link to DVB FE, analog, and audio will detect and honor the tuner busy conditions and vice versa. This patch series has been updated to address comments from 3 previous versions of this series. Link to v3 below for reference. https://www.mail-archive.com/linux-media%40vger.kernel.org/msg89491.html https://www.mail-archive.com/linux-media@vger.kernel.org/msg89492.html https://www.mail-archive.com/linux-media%40vger.kernel.org/msg89493.html Open Issues: ALSA has makes media_entity_pipeline_start() call in irq path. I am seeing warnings that the graph_mutex is unsafe irq lock. Media Controller API updates to start/stop pipeline to be irq safe might be necessary. Maybe there are other MC interfaces that need to be irq safe, but I haven't seen any problems with my testing. So as per options, graph_mutex could be changed to a spinlock. It looks like drivers hold this lock and it isn't abstracted to MC API. Unfortunate, this would require changes to drivers that directly hold the lock for graph walks if this mutex is changed to spinlock. This issue needs to be resolved. Note: This series includes a revert to a patch that added media controller entity framework enhancements that implemented entity ops for entity_notify functionality. Entity ops for entity_notify doesn't handle and cover entity create ordering variations that could occur during boot. entity_notify list has been moved to media device level which makes the entity_notify calls to work correctly. Shuah Khan (7): Revert "[media] media: media controller entity framework enhancements for ALSA" media: Media Controller register/unregister entity_notify API media: Add ALSA Media Controller devnodes media: change dvb-frontend to honor MC tuner enable error media: au8522 change to create MC pad for ALSA Audio Out media: au0828 change to use Managed Media Controller API sound/usb: Update ALSA driver to use Managed Media Controller API drivers/media/dvb-core/dvb_frontend.c | 5 +- drivers/media/dvb-frontends/au8522.h | 8 + drivers/media/dvb-frontends/au8522_decoder.c | 1 + drivers/media/dvb-frontends/au8522_priv.h | 8 - drivers/media/media-device.c | 46 ++++- drivers/media/usb/au0828/au0828-core.c | 132 ++++++++------ drivers/media/usb/au0828/au0828.h | 5 + include/media/media-device.h | 23 +++ include/media/media-entity.h | 4 - include/uapi/linux/media.h | 5 + sound/usb/Makefile | 15 +- sound/usb/card.c | 5 + sound/usb/card.h | 1 + sound/usb/media.c | 260 +++++++++++++++++++++++++++ sound/usb/media.h | 52 ++++++ sound/usb/pcm.c | 15 +- sound/usb/quirks-table.h | 1 + sound/usb/quirks.c | 9 +- sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 1 + 20 files changed, 523 insertions(+), 75 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.1.4

3 14

[alsa-devel] [asoc:topic/rockchip 2/2] warning: (SND_SOC_ROCKCHIP_MAX98090 && ..) selects SND_SOC_ROCKCHIP_I2S which has unmet direct dependencies (SOUND && ..)
by kbuild test robot 20 Jul '15

20 Jul '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rockchip head: 86059653ea7ca7b30ed25d6bec5807ba59a4f2e6 commit: 86059653ea7ca7b30ed25d6bec5807ba59a4f2e6 [2/2] ASoC: rockchip: Add machine driver for rt5645/rt5650 codec config: sparc64-allyesconfig (attached as .config) reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 86059653ea7ca7b30ed25d6bec5807ba59a4f2e6 # save the attached .config to linux build tree make.cross ARCH=sparc64 All warnings (new ones prefixed by >>): warning: (SND_SOC_ROCKCHIP_MAX98090 && SND_SOC_ROCKCHIP_RT5645) selects SND_SOC_ROCKCHIP_I2S which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && CLKDEV_LOOKUP && SND_SOC_ROCKCHIP) --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 1/2] ASoC: cs4349: Fix max_register setting for cs4349_regmap
by Axel Lin 20 Jul '15

20 Jul '15

The max_register should be the maximum valid register index rather than number of registers. Also remove unused defines. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs4349.c | 2 +- sound/soc/codecs/cs4349.h | 3 --- 2 files changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index a8df8a74..9273b06 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -268,7 +268,7 @@ static struct regmap_config cs4349_regmap = { .reg_bits = 8, .val_bits = 8, - .max_register = CS4349_NUMREGS, + .max_register = CS4349_MISC, .reg_defaults = cs4349_reg_defaults, .num_reg_defaults = ARRAY_SIZE(cs4349_reg_defaults), .readable_reg = cs4349_readable_register, diff --git a/sound/soc/codecs/cs4349.h b/sound/soc/codecs/cs4349.h index 3884a89..7effa0a 100644 --- a/sound/soc/codecs/cs4349.h +++ b/sound/soc/codecs/cs4349.h @@ -36,9 +36,6 @@ struct cs4349_platform_data { #define CS4349_RMPFLT 0x07 /* Ramp and Filter Control */ #define CS4349_MISC 0x08 /* Power Down,Freeze Control,Pop Stop*/ -#define CS4349_FIRSTREG 0x01 -#define CS4349_LASTREG 0x08 -#define CS4349_NUMREGS (CS4349_LASTREG - CS4349_FIRSTREG + 1) #define CS4349_I2C_INCR 0x80 -- 2.1.0

3 4

[alsa-devel] [PATCH] ASoC: rt5645: Remove return value in jack detect work
by Nicolas Boichat 20 Jul '15

20 Jul '15

"ASoC: rt5645: Check if codec is initialized in workqueue handler" adds a check if codec is NULL in rt5645_irq_detection, which returns an int. However, "ASoC: rt5645: Remove irq_jack_detection function" removes that function, and moves the code in jack_detect_work, which returns void. Remove the return value to fix compilation warning. Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> --- Mark, Thanks for replacing "ASoC: rt5645: Check if codec is initialized in workqueue handler" in fix/rt5645, that makes this a bit easier. This patch applies on top of broonie/sound.git/for-next, and should be applied in topic/rt5645, after merging fix/rt5645 into topic/rt5645. sound/soc/codecs/rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 3a6756d..9d54b34 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2928,7 +2928,7 @@ static void rt5645_jack_detect_work(struct work_struct *work) int val, btn_type, gpio_state = 0, report = 0; if (!rt5645->codec) - return -EINVAL; + return; switch (rt5645->pdata.jd_mode) { case 0: /* Not using rt5645 JD */ -- 2.4.3.573.g4eafbef

2 1

[alsa-devel] [PATCH v3 0/2] Add codec machine driver for rockchip platform
by Xing Zheng 20 Jul '15

20 Jul '15

From: zhengxing <zhengxing(a)rock-chips.com> Hi, The simple-card is not common at present, soc maybe need own machine driver for jack detection. Add drivers for two families of rockchip-bases chromebooks. These machine drives don't use simplecard because we need custom jack detection plumbing. - use ts3a227e for ext jack detection with max98090 - call rt5645_set_jack_detect function via rt5645 codec driver Thanks. Changes in v3: Signed-off-by: zhengxing <zhengxing(a)rock-chips.com> zhengxing (2): ASoC: rockchip: Add machine driver for max98090 codec ASoC: rockchip: Add machine driver for rt5645/rt5650 codec .../bindings/sound/rockchip-max98090.txt | 19 ++ .../devicetree/bindings/sound/rockchip-rt5645.txt | 17 ++ sound/soc/rockchip/Kconfig | 19 ++ sound/soc/rockchip/Makefile | 6 + sound/soc/rockchip/rockchip_max98090.c | 236 ++++++++++++++++++++ sound/soc/rockchip/rockchip_rt5645.c | 225 +++++++++++++++++++ 6 files changed, 522 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-max98090.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-rt5645.txt create mode 100644 sound/soc/rockchip/rockchip_max98090.c create mode 100644 sound/soc/rockchip/rockchip_rt5645.c -- 1.7.9.5

2 4

[alsa-devel] [PATCH] ASoC: ux500: adjust devm usage
by Julia Lawall 20 Jul '15

20 Jul '15

The explicit call to devm_regulator_put in the probe and remove functions does not seem to be necessary. In particular, the functions prcmu_qos_remove_requirement and ux500_msp_i2s_cleanup_msp in the remove function seem to do nothing that can interfere with devm_regulator_put, making it safe to allow devm_regulator_put to occur after the end of the remove function. Convert the calls to clk_get to devm_clk_get, and remove the corresponding calls to clk_put in the probe and remove functions. Replace various gotos by direct returns, and drop unneeded labels. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/soc/ux500/ux500_msp_dai.c | 29 ++++++++++------------------- 1 file changed, 10 insertions(+), 19 deletions(-) diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 978f2d7..f5df08d 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -773,20 +773,22 @@ static int ux500_msp_drv_probe(struct platform_device *pdev) } prcmu_qos_add_requirement(PRCMU_QOS_APE_OPP, (char *)pdev->name, 50); - drvdata->pclk = clk_get(&pdev->dev, "apb_pclk"); + drvdata->pclk = devm_clk_get(&pdev->dev, "apb_pclk"); if (IS_ERR(drvdata->pclk)) { ret = (int)PTR_ERR(drvdata->pclk); - dev_err(&pdev->dev, "%s: ERROR: clk_get of pclk failed (%d)!\n", + dev_err(&pdev->dev, + "%s: ERROR: devm_clk_get of pclk failed (%d)!\n", __func__, ret); - goto err_pclk; + return ret; } - drvdata->clk = clk_get(&pdev->dev, NULL); + drvdata->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(drvdata->clk)) { ret = (int)PTR_ERR(drvdata->clk); - dev_err(&pdev->dev, "%s: ERROR: clk_get failed (%d)!\n", + dev_err(&pdev->dev, + "%s: ERROR: devm_clk_get failed (%d)!\n", __func__, ret); - goto err_clk; + return ret; } ret = ux500_msp_i2s_init_msp(pdev, &drvdata->msp, @@ -795,7 +797,7 @@ static int ux500_msp_drv_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s: ERROR: Failed to init MSP-struct (%d)!", __func__, ret); - goto err_init_msp; + return ret; } dev_set_drvdata(&pdev->dev, drvdata); @@ -804,7 +806,7 @@ static int ux500_msp_drv_probe(struct platform_device *pdev) if (ret < 0) { dev_err(&pdev->dev, "Error: %s: Failed to register MSP%d!\n", __func__, drvdata->msp->id); - goto err_init_msp; + return ret; } ret = ux500_pcm_register_platform(pdev); @@ -819,13 +821,6 @@ static int ux500_msp_drv_probe(struct platform_device *pdev) err_reg_plat: snd_soc_unregister_component(&pdev->dev); -err_init_msp: - clk_put(drvdata->clk); -err_clk: - clk_put(drvdata->pclk); -err_pclk: - devm_regulator_put(drvdata->reg_vape); - return ret; } @@ -837,12 +832,8 @@ static int ux500_msp_drv_remove(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); - devm_regulator_put(drvdata->reg_vape); prcmu_qos_remove_requirement(PRCMU_QOS_APE_OPP, "ux500_msp_i2s"); - clk_put(drvdata->clk); - clk_put(drvdata->pclk); - ux500_msp_i2s_cleanup_msp(pdev, drvdata->msp); return 0;

3 2

[alsa-devel] Applied "ASoC: cs4349: include gpio/consumer.h" to the asoc tree
by Mark Brown 20 Jul '15

20 Jul '15

The patch ASoC: cs4349: include gpio/consumer.h 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 091571d071a435a517aec42c4f8c56883c5dc531 Mon Sep 17 00:00:00 2001 From: "akpm(a)linux-foundation.org" <akpm(a)linux-foundation.org> Date: Fri, 17 Jul 2015 15:43:46 -0700 Subject: [PATCH] ASoC: cs4349: include gpio/consumer.h s390 allmodconfig: sound/soc/codecs/cs4349.c: In function 'cs4349_i2c_probe': sound/soc/codecs/cs4349.c:300: error: implicit declaration of function 'devm_gpiod_get_optional' sound/soc/codecs/cs4349.c:301: error: 'GPIOD_OUT_LOW' undeclared (first use in this function) sound/soc/codecs/cs4349.c:301: error: (Each undeclared identifier is reported only once sound/soc/codecs/cs4349.c:301: error: for each function it appears in.) sound/soc/codecs/cs4349.c:306: error: implicit declaration of function 'gpiod_set_value_cansleep' Signed-off-by: Andrew Morton <akpm(a)linux-foundation.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cs4349.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index 2569010ee396..a7538aea7f99 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -16,6 +16,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/platform_device.h> #include <linux/pm.h> #include <linux/i2c.h> -- 2.1.4

1 0

[alsa-devel] Applied "ASoC: sti-uniperf: Fix implicit inclusion of pinctrl" to the asoc tree
by Mark Brown 20 Jul '15

20 Jul '15

The patch ASoC: sti-uniperf: Fix implicit inclusion of pinctrl 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 c7ab66836d2d31fe549b71c0b835c563d4419061 Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Mon, 20 Jul 2015 18:36:15 +0100 Subject: [PATCH] ASoC: sti-uniperf: Fix implicit inclusion of pinctrl Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sti/sti_uniperif.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index 83a301f2aa66..10e8c3a0b85f 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -6,6 +6,7 @@ */ #include <linux/module.h> +#include <linux/pinctrl.h> #include "uniperif.h" -- 2.1.4

1 0

[alsa-devel] [PATCH v2 1/4] ASoC: fsl_sai: Add driver suspend and resume to support MEGA Fast
by Zidan Wang 20 Jul '15

20 Jul '15

For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of modules during system suspend and resume procedure. Thus, SAI needs to save all the values of registers before the system suspend and restore them after the system resume. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5c73bea..62187e4 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -637,6 +637,8 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { + case FSL_SAI_TCSR: + case FSL_SAI_RCSR: case FSL_SAI_TFR: case FSL_SAI_RFR: case FSL_SAI_TDR: @@ -681,6 +683,7 @@ static const struct regmap_config fsl_sai_regmap_config = { .readable_reg = fsl_sai_readable_reg, .volatile_reg = fsl_sai_volatile_reg, .writeable_reg = fsl_sai_writeable_reg, + .cache_type = REGCACHE_RBTREE, }; static int fsl_sai_probe(struct platform_device *pdev) @@ -802,10 +805,40 @@ static const struct of_device_id fsl_sai_ids[] = { { /* sentinel */ } }; +#if CONFIG_PM_SLEEP +static int fsl_sai_suspend(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, true); + regcache_mark_dirty(sai->regmap); + + return 0; +} + +static int fsl_sai_resume(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, false); + regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR); + msleep(1); + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + return regcache_sync(sai->regmap); +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops fsl_sai_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(fsl_sai_suspend, fsl_sai_resume) +}; + static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, .driver = { .name = "fsl-sai", + .pm = &fsl_sai_pm_ops, .of_match_table = fsl_sai_ids, }, }; -- 1.9.1

3 6

Re: [alsa-devel] ASoC: Add max98925 codec driver
by Dan Carpenter 20 Jul '15

20 Jul '15

Hello Anish Kumar, The patch 1ff2765182d1: "ASoC: Add max98925 codec driver" from Mar 9, 2015, leads to the following static checker warning: sound/soc/codecs/max98925.c:274 max98925_rate_value() error: buffer overflow 'rate_table' 9 <= 9 sound/soc/codecs/max98925.c 259 static inline int max98925_rate_value(struct snd_soc_codec *codec, 260 int rate, int clock, int *value, int *n, int *m) 261 { 262 int ret = -EINVAL; 263 int i; 264 265 for (i = 0; i < ARRAY_SIZE(rate_table); i++) { 266 if (rate_table[i].rate >= rate) { 267 *value = rate_table[i].sr; 268 *n = rate_table[i].divisors[clock][0]; 269 *m = rate_table[i].divisors[clock][1]; 270 ret = 0; 271 break; 272 } 273 } 274 dev_dbg(codec->dev, "%s: sample rate is %d, returning %d\n", 275 __func__, rate_table[i].rate, *value); ^^^^^^^^^^^^^^^^^^ If we don't break then we're off by one. 276 return ret; 277 } regards, dan carpenter

2 2

[alsa-devel] [PATCH] ALSA: hda: add new AMD PCI IDs with proper driver caps
by Maruthi Srinivas Bayyavarapu 20 Jul '15

20 Jul '15

Fixes audio problems on newer asics Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Cc: stable(a)vger.kernel.org --- sound/pci/hda/hda_intel.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ca151b4..cb7e8f8 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2182,6 +2182,8 @@ static const struct pci_device_id azx_ids[] = { /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x1308), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0x157a), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0x793b), .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, { PCI_DEVICE(0x1002, 0x7919), @@ -2236,8 +2238,14 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaab0), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaac0), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaac8), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaad8), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaae8), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA | AZX_DCAPS_POSFIX_VIA }, -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ALSA: hda - Fix Skylake codec timeout
by David Henningsson 20 Jul '15

20 Jul '15

When the controller is powered up but the HDMI codec is powered down on Skylake, the power well is turned off. When the codec is then powered up again, we need to poke the codec a little extra to make sure it wakes up. Otherwise we'll get sad "no response from codec" messages and broken audio. This also changes azx_runtime_resume to actually call snd_hdac_set_codec_wakeup for Skylake (before STATETS read). (Otherwise it would only have been called for Haswell and Broadwell, which both do not need it, so this probably was not the author's intention.) Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/hda/hdac_i915.c | 5 ++++- sound/pci/hda/hda_intel.c | 18 ++++++++++-------- 2 files changed, 14 insertions(+), 9 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 442500e..5676b84 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -56,8 +56,11 @@ int snd_hdac_display_power(struct hdac_bus *bus, bool enable) enable ? "enable" : "disable"); if (enable) { - if (!bus->i915_power_refcount++) + if (!bus->i915_power_refcount++) { acomp->ops->get_power(acomp->dev); + snd_hdac_set_codec_wakeup(bus, true); + snd_hdac_set_codec_wakeup(bus, false); + } } else { WARN_ON(!bus->i915_power_refcount); if (!--bus->i915_power_refcount) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ca151b4..9962237 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -979,14 +979,16 @@ static int azx_runtime_resume(struct device *dev) if (!azx_has_pm_runtime(chip)) return 0; - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL - && hda->need_i915_power) { - bus = azx_bus(chip); - snd_hdac_display_power(bus, true); - haswell_set_bclk(hda); - /* toggle codec wakeup bit for STATESTS read */ - snd_hdac_set_codec_wakeup(bus, true); - snd_hdac_set_codec_wakeup(bus, false); + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { + bus = azx_bus(chip); + if (hda->need_i915_power) { + snd_hdac_display_power(bus, true); + haswell_set_bclk(hda); + } else { + /* toggle codec wakeup bit for STATESTS read */ + snd_hdac_set_codec_wakeup(bus, true); + snd_hdac_set_codec_wakeup(bus, false); + } } /* Read STATESTS before controller reset */ -- 1.9.1

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[alsa-devel] Beaglebone audio card problem diagnosis?
by g4＠novadsp.com 19 Jul '15

19 Jul '15

Hello all. I'm running mainline 4.1.0 on a Beaglebone Black (TI AM3359 SOC). The board has a standard TI TLV320AIC codec connected via the B3 rev B audio cape: (http://elinux.org/CircuitCo:Audio_Cape_RevB) Loading the nominally correct device tree overlay gets the following kernel log (1). An aplay -l listing (2) shows something recognized is present. Lsmod (3) shows what looks like a sane collection of prerequisite modules. Upon trying to play a test file the kernel gets a number of buffer underflow/underrun messages (4) There was a patch posted here (http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/083797.h tml) which appears to have made it into 4.1.0. Any thoughts on where to start looking to fix? More info can be provided if required and can also check hardware pin activity. Any pointers much appreciated. TAIA Jerry. -----------------------------------snip (1) root@arm:/home/debian# echo BB-BONE-AUDI-02 > /sys/devices/platform/bone_capemgr/slots [ 100.654019] bone_capemgr bone_capemgr: part_number 'BB-BONE-AUDI-02', version 'N/A' [ 100.662117] bone_capemgr bone_capemgr: slot #4: override [ 100.667840] bone_capemgr bone_capemgr: Using override eeprom data at slot 4 [ 100.675221] bone_capemgr bone_capemgr: slot #4: 'Override Board Name,00A0,Override Manuf,BB-BONE-AUDI-02' [ 100.712198] bone_capemgr bone_capemgr: slot #4: dtbo 'BB-BONE-AUDI-02-00A0.dtbo' loaded; overlay id #0 root@arm:/home/debian# [ 101.015982] davinci_evm ocp:sound: ASoC: CPU DAI (null) not registered [ 101.023017] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 101.056114] 2-0018 supply IOVDD not found, using dummy regulator [ 101.067907] 2-0018 supply DVDD not found, using dummy regulator [ 101.074535] 2-0018 supply AVDD not found, using dummy regulator [ 101.080746] 2-0018 supply DRVDD not found, using dummy regulator [ 101.089316] davinci_evm ocp:sound: ASoC: CODEC DAI tlv320aic3x-hifi not registered [ 101.097310] davinci_evm ocp:sound: snd_soc_register_card failed (-517) [ 101.114161] davinci_evm ocp:sound: tlv320aic3x-hifi <-> 48038000.mcasp mapping ok -----------------------------------snip (2) root@arm:/home/debian# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: EVM [DA830 EVM], device 0: AIC3X tlv320aic3x-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 -----------------------------------snip (3) root@arm:/home/debian# lsmod Module Size Used by snd_soc_davinci_mcasp 14407 2 snd_soc_tlv320aic3x 40077 1 snd_soc_edma 1150 1 snd_soc_davinci_mcasp snd_soc_evm 4802 0 snd_soc_omap 2573 1 snd_soc_davinci_mcasp snd_soc_core 156839 5 snd_soc_davinci_mcasp,snd_soc_edma,snd_soc_omap,snd_soc_tlv320aic3x,snd_soc_ evm snd_compress 11858 1 snd_soc_core snd_pcm_dmaengine 5065 2 snd_soc_core,snd_soc_omap snd_pcm 85645 5 snd_soc_davinci_mcasp,snd_soc_core,snd_soc_omap,snd_soc_tlv320aic3x,snd_pcm_ dmaengine snd_timer 17058 1 snd_pcm snd 58469 4 snd_soc_core,snd_timer,snd_pcm,snd_compress soundcore 6861 1 snd omap_aes 13065 0 omap_sham 19198 0 tda998x 11695 1 tilcdc 27899 0 omap_rng 4354 0 drm_kms_helper 106679 3 tda998x,tilcdc rng_core 7266 1 omap_rng uio_pdrv_genirq 3317 0 uio 8350 1 uio_pdrv_genirq leds_gpio 3110 0 -----------------------------------snip (4) debian@arm:~$ aplay sine2s.wav Playing WAVE 'sine2s.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo [ 618.068355] davinci-mcasp 48038000.mcasp: Transmit buffer underflow underrun!!! (at least 0.410 ms long) [ 618.092511] davinci-mcasp 48038000.mcasp: Transmit buffer underflow underrun!!! (at least 11.824 ms long) [ 618.121692] davinci-mcasp 48038000.mcasp: Transmit buffer underflow underrun!!! (at least 3.200 ms long) [ 618.134323] davinci-mcasp 48038000.mcasp: Transmit buffer underflow

1 0

[alsa-devel] [PATCH] ALSA: hda - Add headset mic support for Acer Aspire V5-573G
by Takashi Iwai 19 Jul '15

19 Jul '15

From: Mateusz Sylwestrzak <matisec7(a)gmail.com> Acer Aspire V5 with the ALC282 codec is given the wrong value for the 0x19 PIN by the laptop's BIOS. Overriding it with the correct value adds support for the headset microphone which would not otherwise be visible in the system. The fix is based on commit 7819717b1134 with a similar quirk for Acer Aspire with the ALC269 codec. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=96201 Cc: <stable(a)vger.kernel.org> Signed-off-by: Mateusz Sylwestrzak <matisec7(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d35cf506a7db..caba66b2311c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5061,7 +5061,7 @@ static const struct hda_fixup alc269_fixups[] = { { 0x14, 0x90170110 }, { 0x17, 0x40000008 }, { 0x18, 0x411111f0 }, - { 0x19, 0x411111f0 }, + { 0x19, 0x01a1913c }, { 0x1a, 0x411111f0 }, { 0x1b, 0x411111f0 }, { 0x1d, 0x40f89b2d }, -- 2.4.5

1 0

[alsa-devel] [PATCH v3 0/2] Add codec machine driver for rockchip platform
by Xing Zheng 19 Jul '15

19 Jul '15

From: zhengxing <zhengxing(a)rock-chips.com> Hi, The simple-card is not common at present, soc maybe need own machine driver for jack detection. Add drivers for two families of rockchip-bases chromebooks. These machine drives don't use simplecard because we need custom jack detection plumbing. - use ts3a227e for ext jack detection with max98090 - call rt5645_set_jack_detect function via rt5645 codec driver Thanks. Changes in v3: Signed-off-by: zhengxing <zhengxing(a)rock-chips.com> zhengxing (2): ASoC: rockchip: Add machine driver for max98090 codec ASoC: rockchip: Add machine driver for rt5645/rt5650 codec .../bindings/sound/rockchip-max98090.txt | 19 ++ .../devicetree/bindings/sound/rockchip-rt5645.txt | 17 ++ sound/soc/rockchip/Kconfig | 19 ++ sound/soc/rockchip/Makefile | 6 + sound/soc/rockchip/rockchip_max98090.c | 236 ++++++++++++++++++++ sound/soc/rockchip/rockchip_rt5645.c | 225 +++++++++++++++++++ 6 files changed, 522 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-max98090.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-rt5645.txt create mode 100644 sound/soc/rockchip/rockchip_max98090.c create mode 100644 sound/soc/rockchip/rockchip_rt5645.c -- 1.7.9.5

1 2

[alsa-devel] [PATCH v2 0/2] Add codec machine driver for rockchip platform
by Xing Zheng 19 Jul '15

19 Jul '15

From: zhengxing <zhengxing(a)rock-chips.com> Hi, The simple-card is not common at present, soc maybe need own machine driver for jack detection. Add drivers for two families of rockchip-bases chromebooks. These machine drives don't use simplecard because we need custom jack detection plumbing. - use ts3a227e for ext jack detection with max98090 - call rt5645_set_jack_detect function via rt5645 codec driver Thanks. Changes in v2: Signed-off-by: zhengxing <zhengxing(a)rock-chips.com> zhengxing (2): ASoC: rockchip: Add machine driver for max98090 codec ASoC: rockchip: Add machine driver for rt5645/rt5650 codec .../bindings/sound/rockchip-max98090.txt | 19 ++ .../devicetree/bindings/sound/rockchip-rt5645.txt | 17 ++ sound/soc/rockchip/Kconfig | 19 ++ sound/soc/rockchip/Makefile | 6 + sound/soc/rockchip/rockchip_max98090.c | 247 ++++++++++++++++++++ sound/soc/rockchip/rockchip_rt5645.c | 236 +++++++++++++++++++ 6 files changed, 544 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-max98090.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-rt5645.txt create mode 100644 sound/soc/rockchip/rockchip_max98090.c create mode 100644 sound/soc/rockchip/rockchip_rt5645.c -- 1.7.9.5

2 4

Re: [alsa-devel] After kernel upgrade to v ~~> 4.1.0, no audio output unless UNPLUG/RE-PLUGIN of audio jack on card; OK with earlier kernels
by robert.devanna＠nospammail.net 19 Jul '15

19 Jul '15

On Mon, Jul 6, 2015, at 01:46 AM, Takashi Iwai wrote: > It's a known problem with VIA codecs, and the fix was already merged > in 4.2-rc1, the commit 735c75cf4d434862e38c01dcfb2ce8d2fcb9035f. > The next stable 4.1.x kernel should contain the fix. Upgraded to uname -rm 4.1.2-1.g5ccacb4-desktop x86_64 Fyi, the problem remains unchanged. To get sound, a physical unplug/replug is required.

2 1

[alsa-devel] [PATCH 0/2] Add codec machine driver for rockchip platform
by Xing Zheng 18 Jul '15

18 Jul '15

From: zhengxing <zhengxing(a)rock-chips.com> Hi, The simple-card is not common at present, soc maybe need own machine driver for jack detection. Add drivers for two families of rockchip-bases chromebooks. These machine drives don't use simplecard because we need custom jack detection plumbing. - use ts3a227e for ext jack detection with max98090 - call rt5645_set_jack_detect function via rt5645 codec driver Thanks. zhengxing (2): ASoC: rockchip: Add machine driver for max98090 codec ASoC: rockchip: Add machine driver for rt5645/rt5650 codec sound/soc/rockchip/Kconfig | 19 +++ sound/soc/rockchip/Makefile | 6 + sound/soc/rockchip/rockchip_max98090.c | 246 ++++++++++++++++++++++++++++++++ sound/soc/rockchip/rockchip_rt5645.c | 235 ++++++++++++++++++++++++++++++ 4 files changed, 506 insertions(+) create mode 100644 sound/soc/rockchip/rockchip_max98090.c create mode 100644 sound/soc/rockchip/rockchip_rt5645.c -- 1.7.9.5

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[alsa-devel] [PATCH v2] arm64: dts: mt8173: Add afe device node
by Koro Chen 17 Jul '15

17 Jul '15

This adds afe (audio front end) device node to the MT8173 dtsi file. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- This patch is based on Matthias's tree: https://github.com/mbgg/linux-mediatek branch: v4.2-next/arm64 Changes since v1: - change node name to afe: audio-controller@11220000 --- arch/arm64/boot/dts/mediatek/mt8173.dtsi | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/arch/arm64/boot/dts/mediatek/mt8173.dtsi b/arch/arm64/boot/dts/mediatek/mt8173.dtsi index 0696f8f..ce9255a 100644 --- a/arch/arm64/boot/dts/mediatek/mt8173.dtsi +++ b/arch/arm64/boot/dts/mediatek/mt8173.dtsi @@ -14,6 +14,7 @@ #include <dt-bindings/clock/mt8173-clk.h> #include <dt-bindings/interrupt-controller/irq.h> #include <dt-bindings/interrupt-controller/arm-gic.h> +#include <dt-bindings/power/mt8173-power.h> #include <dt-bindings/reset-controller/mt8173-resets.h> #include "mt8173-pinfunc.h" @@ -393,6 +394,37 @@ #size-cells = <0>; status = "disabled"; }; + + afe: audio-controller@11220000 { + compatible = "mediatek,mt8173-afe-pcm"; + reg = <0 0x11220000 0 0x1000>; + interrupts = <GIC_SPI 134 IRQ_TYPE_EDGE_FALLING>; + power-domains = <&scpsys MT8173_POWER_DOMAIN_AUDIO>; + clocks = <&infracfg CLK_INFRA_AUDIO>, + <&topckgen CLK_TOP_AUDIO_SEL>, + <&topckgen CLK_TOP_AUD_INTBUS_SEL>, + <&topckgen CLK_TOP_APLL1_DIV0>, + <&topckgen CLK_TOP_APLL2_DIV0>, + <&topckgen CLK_TOP_I2S0_M_SEL>, + <&topckgen CLK_TOP_I2S1_M_SEL>, + <&topckgen CLK_TOP_I2S2_M_SEL>, + <&topckgen CLK_TOP_I2S3_M_SEL>, + <&topckgen CLK_TOP_I2S3_B_SEL>; + clock-names = "infra_sys_audio_clk", + "top_pdn_audio", + "top_pdn_aud_intbus", + "bck0", + "bck1", + "i2s0_m", + "i2s1_m", + "i2s2_m", + "i2s3_m", + "i2s3_b"; + assigned-clocks = <&topckgen CLK_TOP_AUD_1_SEL>, + <&topckgen CLK_TOP_AUD_2_SEL>; + assigned-clock-parents = <&topckgen CLK_TOP_APLL1>, + <&topckgen CLK_TOP_APLL2>; + }; }; }; -- 1.8.1.1.dirty

3 3

[alsa-devel] [PATCH] ALSA: sparc: Add missing kfree in error path
by Christophe JAILLET 17 Jul '15

17 Jul '15

If 'of_ioremap' fails, then 'amd' should be freed, otherwise, there is a memory leak. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/sparc/amd7930.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index 1b1a89e..784ceb8 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -956,6 +956,7 @@ static int snd_amd7930_create(struct snd_card *card, if (!amd->regs) { snd_printk(KERN_ERR "amd7930-%d: Unable to map chip registers.\n", dev); + kfree(amd); return -EIO; } -- 2.1.4

2 1

[alsa-devel] [PATCH 0/35] ASoC: rsnd: add CTU/MIX prototype support
by Kuninori Morimoto 17 Jul '15

17 Jul '15

Hi Mark, Simon These patches add CTU/MIX prototype support on Renesas sound card driver. It needs DPCM feature, so it is assuming to be used via rsrc-card sound card. These CTU/MIX is still prototype support. This Mixer works well if sound A and B are using same bit length, same sampling rate. 1) - 2) adds audio prefix support on DT which is needed on DPCM 3) - 29) cleanup/tidyup/workaround/prepare patches for CTU/MIX support 30) - 31) CTU/MIX support on driver 32) - 35) CTU/MIX support on SoC Simon, This time I didn't Cc:ed to SH-ML, since 32) - 35) need Document update (?). It is done on 30) - 31). I can post 32) - 35) if Mark accepted 1) - 31) Kuninori Morimoto (35): 1) ASoC: core: add snd_soc_of_parse_audio_prefix() 2) ASoC: rsrc-card: use snd_soc_of_parse_audio_route/prefix for routing 3) ASoC: rsnd: remove unnecessary 'out of memory' message from SSI 4) ASoC: rsnd: remove unnecessary 'out of memory' message from SRC 5) ASoC: rsnd: remove unnecessary 'out of memory' message from DVC 6) ASoC: rsnd: rename BUSIF_DALIGN to SSI_BUSIF_DALIGN 7) ASoC: rsnd: rename INT_ENABLE to SSI_INT_ENABLE 8) ASoC: rsnd: fixup each module counter on __rsnd_mod_call() 9) ASoC: rsnd: add workaround for SRC sync convert + DVC 10) ASoC: rsnd: rsnd_mod_id() return -1 if mod was NULL 11) ASoC: rsnd: move DVC specific macro into dvc.c 12) ASoC: rsnd: move SRC specific macro into src.c 13) ASoC: rsnd: dvc: make sure DVC soft reset 14) ASoC: rsnd: src: make sure SRC soft reset 15) ASoC: rsnd: enable module multi connection 16) ASoC: rsnd: rename rsnd_path_parse/break() into add/remove 17) ASoC: rsnd: add rsnd_path_parse() for CTU/MIX/DVC route setting 18) ASoC: rsnd: add rsnd_dvc_initialize_lock/unlock() 19) ASoC: rsnd: add rsnd_src_initialize_lock/unlock() 20) ASoC: rsnd: tidyup ADINR function name 21) ASoC: rsnd: add rsnd_get_adinr_chan() 22) ASoC: rsnd: tidyup data align position 23) ASoC: rsnd: show debug message for SSI/SRC/DVC connection 24) ASoC: rsnd: tidyup rsnd_dma_ops definition place 25) ASoC: rsnd: check the Gen1 at the beginning of DVC probe 26) ASoC: rsnd: dma: add DMA name on .ops 27) ASoC: rsnd: add rsnd_io_to_mod() 28) ASoC: rsnd: tidyup SRC position on each code 29) ASoC: rsnd: update Audio DMA path search method 30) ASoC: rsnd: add CTU (Channel Transfer Unit) prototype support 31) ASoC: rsnd: add MIX (Mixer) support 32) ARM: shmobile: r8a7790: Add Audio CTU support on DTSI 33) ARM: shmobile: r8a7790: Add Audio MIX support on DTSI 34) ARM: shmobile: r8a7791: Add Audio CTU support on DTSI 35) ARM: shmobile: r8a7791: Add Audio MIX support on DTSI Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 22 +++++++++++ Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 7 ++++ arch/arm/boot/dts/r8a7790.dtsi | 23 ++++++++++++ arch/arm/boot/dts/r8a7791.dtsi | 23 ++++++++++++ include/dt-bindings/clock/r8a7790-clock.h | 2 + include/dt-bindings/clock/r8a7791-clock.h | 2 + include/sound/rcar_snd.h | 14 +++++++ include/sound/soc.h | 4 ++ sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/core.c | 194 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++------------ sound/soc/sh/rcar/ctu.c | 171 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/dma.c | 126 ++++++++++++++++++++++++++++++++++++++++++-------------------- sound/soc/sh/rcar/dvc.c | 73 ++++++++++++++++-------------------- sound/soc/sh/rcar/gen.c | 17 ++++++++- sound/soc/sh/rcar/mix.c | 200 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 107 ++++++++++++++++++++++++++++++++++++---------------- sound/soc/sh/rcar/rsrc-card.c | 22 +++++++++-- sound/soc/sh/rcar/src.c | 117 +++++++++++++++++++++++++++++++-------------------------- sound/soc/sh/rcar/ssi.c | 4 +- sound/soc/soc-core.c | 20 ++++++++++ 20 files changed, 952 insertions(+), 198 deletions(-) Best regards --- Kuninori Morimoto

2 67

[alsa-devel] [PATCH] speaker-test: Add option to specify signal scale
by Julian Scheel 17 Jul '15

17 Jul '15

Allow generated signals (sine and noise) to have a specified signal scale instead of using hardcoded limits of 80%. This can be handy for debugging, ie when analysing clipping issues with audio drivers. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- speaker-test/speaker-test.c | 30 ++++++++++++++++++------------ 1 file changed, 18 insertions(+), 12 deletions(-) diff --git a/speaker-test/speaker-test.c b/speaker-test/speaker-test.c index ccf2671..25970af 100644 --- a/speaker-test/speaker-test.c +++ b/speaker-test/speaker-test.c @@ -100,6 +100,7 @@ static unsigned int period_time = 0; /* period time in us */ static unsigned int nperiods = 4; /* number of periods */ static double freq = 440.0; /* sinusoidal wave frequency in Hz */ static int test_type = TEST_PINK_NOISE; /* Test type. 1 = noise, 2 = sine wave */ +static float generator_scale = 0.8; /* Scale to use for sine volume */ static pink_noise_t pink; static snd_pcm_uframes_t buffer_size; static snd_pcm_uframes_t period_size; @@ -306,7 +307,7 @@ static void generate_sine(uint8_t *frames, int channel, int count, double *_phas switch (format) { case SND_PCM_FORMAT_S8: if (chn==channel) { - res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale * 0x7fffffff; ires = res; *samp8++ = ires >> 24; } else { @@ -315,7 +316,7 @@ static void generate_sine(uint8_t *frames, int channel, int count, double *_phas break; case SND_PCM_FORMAT_S16_LE: if (chn==channel) { - res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale * 0x7fffffff; ires = res; *samp16++ = LE_SHORT(ires >> 16); } else { @@ -324,7 +325,7 @@ static void generate_sine(uint8_t *frames, int channel, int count, double *_phas break; case SND_PCM_FORMAT_S16_BE: if (chn==channel) { - res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale * 0x7fffffff; ires = res; *samp16++ = BE_SHORT(ires >> 16); } else { @@ -333,7 +334,7 @@ static void generate_sine(uint8_t *frames, int channel, int count, double *_phas break; case SND_PCM_FORMAT_FLOAT_LE: if (chn==channel) { - res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0.75 ; /* Don't use MAX volume */ + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale; fres = res; *samp_f++ = fres; } else { @@ -342,7 +343,7 @@ static void generate_sine(uint8_t *frames, int channel, int count, double *_phas break; case SND_PCM_FORMAT_S32_LE: if (chn==channel) { - res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale * 0x7fffffff; ires = res; *samp32++ = LE_INT(ires); } else { @@ -351,7 +352,7 @@ static void generate_sine(uint8_t *frames, int channel, int count, double *_phas break; case SND_PCM_FORMAT_S32_BE: if (chn==channel) { - res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale * 0x7fffffff; ires = res; *samp32++ = BE_INT(ires); } else { @@ -389,7 +390,7 @@ static void generate_pink_noise( uint8_t *frames, int channel, int count) { switch (format) { case SND_PCM_FORMAT_S8: if (chn==channel) { - res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ + res = generate_pink_noise_sample(&pink) * generator_scale * 0x07fffffff; ires = res; *samp8++ = ires >> 24; } else { @@ -398,7 +399,7 @@ static void generate_pink_noise( uint8_t *frames, int channel, int count) { break; case SND_PCM_FORMAT_S16_LE: if (chn==channel) { - res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ + res = generate_pink_noise_sample(&pink) * generator_scale * 0x07fffffff; ires = res; *samp16++ = LE_SHORT(ires >> 16); } else { @@ -407,7 +408,7 @@ static void generate_pink_noise( uint8_t *frames, int channel, int count) { break; case SND_PCM_FORMAT_S16_BE: if (chn==channel) { - res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ + res = generate_pink_noise_sample(&pink) * generator_scale * 0x07fffffff; ires = res; *samp16++ = BE_SHORT(ires >> 16); } else { @@ -416,7 +417,7 @@ static void generate_pink_noise( uint8_t *frames, int channel, int count) { break; case SND_PCM_FORMAT_S32_LE: if (chn==channel) { - res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ + res = generate_pink_noise_sample(&pink) * generator_scale * 0x07fffffff; ires = res; *samp32++ = LE_INT(ires); } else { @@ -425,7 +426,7 @@ static void generate_pink_noise( uint8_t *frames, int channel, int count) { break; case SND_PCM_FORMAT_S32_BE: if (chn==channel) { - res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ + res = generate_pink_noise_sample(&pink) * generator_scale * 0x07fffffff; ires = res; *samp32++ = BE_INT(ires); } else { @@ -1017,6 +1018,7 @@ static void help(void) "-W,--wavdir Specify the directory containing WAV files\n" "-m,--chmap Specify the channel map to override\n" "-X,--force-frequency force frequencies outside the 30-8000hz range\n" + "-S,--scale Scale of generated test tones in percent (default=80)\n" "\n")); printf(_("Recognized sample formats are:")); for (fmt = supported_formats; *fmt >= 0; fmt++) { @@ -1060,6 +1062,7 @@ int main(int argc, char *argv[]) { {"wavdir", 1, NULL, 'W'}, {"debug", 0, NULL, 'd'}, {"force-frequency", 0, NULL, 'X'}, + {"scale", 1, NULL, 'S'}, #ifdef CONFIG_SUPPORT_CHMAP {"chmap", 1, NULL, 'm'}, #endif @@ -1081,7 +1084,7 @@ int main(int argc, char *argv[]) { while (1) { int c; - if ((c = getopt_long(argc, argv, "hD:r:c:f:F:b:p:P:t:l:s:w:W:d:X" + if ((c = getopt_long(argc, argv, "hD:r:c:f:F:b:p:P:t:l:s:w:W:d:XS:" #ifdef CONFIG_SUPPORT_CHMAP "m:" #endif @@ -1182,6 +1185,9 @@ int main(int argc, char *argv[]) { chmap = optarg; break; #endif + case 'S': + generator_scale = atoi(optarg) / 100.0; + break; default: fprintf(stderr, _("Unknown option '%c'\n"), c); exit(EXIT_FAILURE); -- 2.4.3

2 2

[alsa-devel] [PATCH 1/2 V5] regmap: Use reg_sequence for multi_reg_write / register_patch
by Nariman Poushin 17 Jul '15

17 Jul '15

Separate the functionality using sequences of register writes from the functions that take register defaults. This change renames the arguments in order to support the extension of reg_sequence to take an optional delay to be applied after any given register in a sequence is written. This avoids adding an int to all register defaults, which could substantially increase memory usage for regmaps with large default tables. This also updates all the clients of multi_reg_write/register_patch. Signed-off-by: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> --- drivers/base/regmap/internal.h | 2 +- drivers/base/regmap/regmap.c | 22 +++++++++++----------- drivers/gpu/drm/i2c/adv7511.c | 2 +- drivers/input/misc/drv260x.c | 6 +++--- drivers/input/misc/drv2665.c | 2 +- drivers/input/misc/drv2667.c | 4 ++-- drivers/mfd/arizona-core.c | 2 +- drivers/mfd/twl6040.c | 2 +- drivers/mfd/wm5102-tables.c | 6 +++--- drivers/mfd/wm5110-tables.c | 6 +++--- drivers/mfd/wm8994-core.c | 8 ++++---- drivers/mfd/wm8997-tables.c | 2 +- include/linux/regmap.h | 17 ++++++++++++++--- sound/soc/codecs/arizona.c | 2 +- sound/soc/codecs/cs35l32.c | 2 +- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/da7210.c | 4 ++-- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/rt5645.c | 4 ++-- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5670.c | 2 +- sound/soc/codecs/rt5677.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- 27 files changed, 62 insertions(+), 51 deletions(-) diff --git a/drivers/base/regmap/internal.h b/drivers/base/regmap/internal.h index b2b2849..873ddf9 100644 --- a/drivers/base/regmap/internal.h +++ b/drivers/base/regmap/internal.h @@ -136,7 +136,7 @@ struct regmap { /* if set, the HW registers are known to match map->reg_defaults */ bool no_sync_defaults; - struct reg_default *patch; + struct reg_sequence *patch; int patch_regs; /* if set, converts bulk rw to single rw */ diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index dd63bcb..0a849ee 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -1755,7 +1755,7 @@ EXPORT_SYMBOL_GPL(regmap_bulk_write); * relative. The page register has been written if that was neccessary. */ static int _regmap_raw_multi_reg_write(struct regmap *map, - const struct reg_default *regs, + const struct reg_sequence *regs, size_t num_regs) { int ret; @@ -1812,12 +1812,12 @@ static unsigned int _regmap_register_page(struct regmap *map, } static int _regmap_range_multi_paged_reg_write(struct regmap *map, - struct reg_default *regs, + struct reg_sequence *regs, size_t num_regs) { int ret; int i, n; - struct reg_default *base; + struct reg_sequence *base; unsigned int this_page = 0; /* * the set of registers are not neccessarily in order, but @@ -1855,7 +1855,7 @@ static int _regmap_range_multi_paged_reg_write(struct regmap *map, } static int _regmap_multi_reg_write(struct regmap *map, - const struct reg_default *regs, + const struct reg_sequence *regs, size_t num_regs) { int i; @@ -1907,8 +1907,8 @@ static int _regmap_multi_reg_write(struct regmap *map, struct regmap_range_node *range; range = _regmap_range_lookup(map, reg); if (range) { - size_t len = sizeof(struct reg_default)*num_regs; - struct reg_default *base = kmemdup(regs, len, + size_t len = sizeof(struct reg_sequence)*num_regs; + struct reg_sequence *base = kmemdup(regs, len, GFP_KERNEL); if (!base) return -ENOMEM; @@ -1941,7 +1941,7 @@ static int _regmap_multi_reg_write(struct regmap *map, * A value of zero will be returned on success, a negative errno will be * returned in error cases. */ -int regmap_multi_reg_write(struct regmap *map, const struct reg_default *regs, +int regmap_multi_reg_write(struct regmap *map, const struct reg_sequence *regs, int num_regs) { int ret; @@ -1974,7 +1974,7 @@ EXPORT_SYMBOL_GPL(regmap_multi_reg_write); * be returned in error cases. */ int regmap_multi_reg_write_bypassed(struct regmap *map, - const struct reg_default *regs, + const struct reg_sequence *regs, int num_regs) { int ret; @@ -2587,10 +2587,10 @@ EXPORT_SYMBOL_GPL(regmap_async_complete); * The caller must ensure that this function cannot be called * concurrently with either itself or regcache_sync(). */ -int regmap_register_patch(struct regmap *map, const struct reg_default *regs, +int regmap_register_patch(struct regmap *map, const struct reg_sequence *regs, int num_regs) { - struct reg_default *p; + struct reg_sequence *p; int ret; bool bypass; @@ -2599,7 +2599,7 @@ int regmap_register_patch(struct regmap *map, const struct reg_default *regs, return 0; p = krealloc(map->patch, - sizeof(struct reg_default) * (map->patch_regs + num_regs), + sizeof(struct reg_sequence) * (map->patch_regs + num_regs), GFP_KERNEL); if (p) { memcpy(p + map->patch_regs, regs, num_regs * sizeof(*regs)); diff --git a/drivers/gpu/drm/i2c/adv7511.c b/drivers/gpu/drm/i2c/adv7511.c index 2aaa3c8..00416f2 100644 --- a/drivers/gpu/drm/i2c/adv7511.c +++ b/drivers/gpu/drm/i2c/adv7511.c @@ -54,7 +54,7 @@ static struct adv7511 *encoder_to_adv7511(struct drm_encoder *encoder) } /* ADI recommended values for proper operation. */ -static const struct reg_default adv7511_fixed_registers[] = { +static const struct reg_sequence adv7511_fixed_registers[] = { { 0x98, 0x03 }, { 0x9a, 0xe0 }, { 0x9c, 0x30 }, diff --git a/drivers/input/misc/drv260x.c b/drivers/input/misc/drv260x.c index e5d60ec..f5c9cf2 100644 --- a/drivers/input/misc/drv260x.c +++ b/drivers/input/misc/drv260x.c @@ -313,14 +313,14 @@ static void drv260x_close(struct input_dev *input) gpiod_set_value(haptics->enable_gpio, 0); } -static const struct reg_default drv260x_lra_cal_regs[] = { +static const struct reg_sequence drv260x_lra_cal_regs[] = { { DRV260X_MODE, DRV260X_AUTO_CAL }, { DRV260X_CTRL3, DRV260X_NG_THRESH_2 }, { DRV260X_FEEDBACK_CTRL, DRV260X_FB_REG_LRA_MODE | DRV260X_BRAKE_FACTOR_4X | DRV260X_LOOP_GAIN_HIGH }, }; -static const struct reg_default drv260x_lra_init_regs[] = { +static const struct reg_sequence drv260x_lra_init_regs[] = { { DRV260X_MODE, DRV260X_RT_PLAYBACK }, { DRV260X_A_TO_V_CTRL, DRV260X_AUDIO_HAPTICS_PEAK_20MS | DRV260X_AUDIO_HAPTICS_FILTER_125HZ }, @@ -337,7 +337,7 @@ static const struct reg_default drv260x_lra_init_regs[] = { { DRV260X_CTRL4, DRV260X_AUTOCAL_TIME_500MS }, }; -static const struct reg_default drv260x_erm_cal_regs[] = { +static const struct reg_sequence drv260x_erm_cal_regs[] = { { DRV260X_MODE, DRV260X_AUTO_CAL }, { DRV260X_A_TO_V_MIN_INPUT, DRV260X_AUDIO_HAPTICS_MIN_IN_VOLT }, { DRV260X_A_TO_V_MAX_INPUT, DRV260X_AUDIO_HAPTICS_MAX_IN_VOLT }, diff --git a/drivers/input/misc/drv2665.c b/drivers/input/misc/drv2665.c index 0afaa33..924456e 100644 --- a/drivers/input/misc/drv2665.c +++ b/drivers/input/misc/drv2665.c @@ -132,7 +132,7 @@ static void drv2665_close(struct input_dev *input) "Failed to enter standby mode: %d\n", error); } -static const struct reg_default drv2665_init_regs[] = { +static const struct reg_sequence drv2665_init_regs[] = { { DRV2665_CTRL_2, 0 | DRV2665_10_MS_IDLE_TOUT }, { DRV2665_CTRL_1, DRV2665_25_VPP_GAIN }, }; diff --git a/drivers/input/misc/drv2667.c b/drivers/input/misc/drv2667.c index fc0fddf..047136a 100644 --- a/drivers/input/misc/drv2667.c +++ b/drivers/input/misc/drv2667.c @@ -262,14 +262,14 @@ static void drv2667_close(struct input_dev *input) "Failed to enter standby mode: %d\n", error); } -static const struct reg_default drv2667_init_regs[] = { +static const struct reg_sequence drv2667_init_regs[] = { { DRV2667_CTRL_2, 0 }, { DRV2667_CTRL_1, DRV2667_25_VPP_GAIN }, { DRV2667_WV_SEQ_0, 1 }, { DRV2667_WV_SEQ_1, 0 } }; -static const struct reg_default drv2667_page1_init[] = { +static const struct reg_sequence drv2667_page1_init[] = { { DRV2667_RAM_HDR_SZ, 0x05 }, { DRV2667_RAM_START_HI, 0x80 }, { DRV2667_RAM_START_LO, 0x06 }, diff --git a/drivers/mfd/arizona-core.c b/drivers/mfd/arizona-core.c index bebf58a..66d50be 100644 --- a/drivers/mfd/arizona-core.c +++ b/drivers/mfd/arizona-core.c @@ -392,7 +392,7 @@ err: * Register patch to some of the CODECs internal write sequences * to ensure a clean exit from the low power sleep state. */ -static const struct reg_default wm5110_sleep_patch[] = { +static const struct reg_sequence wm5110_sleep_patch[] = { { 0x337A, 0xC100 }, { 0x337B, 0x0041 }, { 0x3300, 0xA210 }, diff --git a/drivers/mfd/twl6040.c b/drivers/mfd/twl6040.c index c5265c1..583dc33 100644 --- a/drivers/mfd/twl6040.c +++ b/drivers/mfd/twl6040.c @@ -86,7 +86,7 @@ static const struct reg_default twl6040_defaults[] = { { 0x2E, 0x00 }, /* REG_STATUS (ro) */ }; -static struct reg_default twl6040_patch[] = { +static struct reg_sequence twl6040_patch[] = { /* * Select I2C bus access to dual access registers * Interrupt register is cleared on read diff --git a/drivers/mfd/wm5102-tables.c b/drivers/mfd/wm5102-tables.c index aeae6ec..423fb37 100644 --- a/drivers/mfd/wm5102-tables.c +++ b/drivers/mfd/wm5102-tables.c @@ -21,7 +21,7 @@ #define WM5102_NUM_AOD_ISR 2 #define WM5102_NUM_ISR 5 -static const struct reg_default wm5102_reva_patch[] = { +static const struct reg_sequence wm5102_reva_patch[] = { { 0x80, 0x0003 }, { 0x221, 0x0090 }, { 0x211, 0x0014 }, @@ -57,7 +57,7 @@ static const struct reg_default wm5102_reva_patch[] = { { 0x80, 0x0000 }, }; -static const struct reg_default wm5102_revb_patch[] = { +static const struct reg_sequence wm5102_revb_patch[] = { { 0x19, 0x0001 }, { 0x80, 0x0003 }, { 0x081, 0xE022 }, @@ -80,7 +80,7 @@ static const struct reg_default wm5102_revb_patch[] = { /* We use a function so we can use ARRAY_SIZE() */ int wm5102_patch(struct arizona *arizona) { - const struct reg_default *wm5102_patch; + const struct reg_sequence *wm5102_patch; int patch_size; switch (arizona->rev) { diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index 12cad94..26ce14f 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -21,7 +21,7 @@ #define WM5110_NUM_AOD_ISR 2 #define WM5110_NUM_ISR 5 -static const struct reg_default wm5110_reva_patch[] = { +static const struct reg_sequence wm5110_reva_patch[] = { { 0x80, 0x3 }, { 0x44, 0x20 }, { 0x45, 0x40 }, @@ -134,7 +134,7 @@ static const struct reg_default wm5110_reva_patch[] = { { 0x209, 0x002A }, }; -static const struct reg_default wm5110_revb_patch[] = { +static const struct reg_sequence wm5110_revb_patch[] = { { 0x80, 0x3 }, { 0x36e, 0x0210 }, { 0x370, 0x0210 }, @@ -224,7 +224,7 @@ static const struct reg_default wm5110_revb_patch[] = { { 0x80, 0x0 }, }; -static const struct reg_default wm5110_revd_patch[] = { +static const struct reg_sequence wm5110_revd_patch[] = { { 0x80, 0x3 }, { 0x80, 0x3 }, { 0x393, 0x27 }, diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index 53ae5af..0f4169a 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -243,21 +243,21 @@ static int wm8994_ldo_in_use(struct wm8994_pdata *pdata, int ldo) } #endif -static const struct reg_default wm8994_revc_patch[] = { +static const struct reg_sequence wm8994_revc_patch[] = { { 0x102, 0x3 }, { 0x56, 0x3 }, { 0x817, 0x0 }, { 0x102, 0x0 }, }; -static const struct reg_default wm8958_reva_patch[] = { +static const struct reg_sequence wm8958_reva_patch[] = { { 0x102, 0x3 }, { 0xcb, 0x81 }, { 0x817, 0x0 }, { 0x102, 0x0 }, }; -static const struct reg_default wm1811_reva_patch[] = { +static const struct reg_sequence wm1811_reva_patch[] = { { 0x102, 0x3 }, { 0x56, 0xc07 }, { 0x5d, 0x7e }, @@ -326,7 +326,7 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) { struct wm8994_pdata *pdata; struct regmap_config *regmap_config; - const struct reg_default *regmap_patch = NULL; + const struct reg_sequence *regmap_patch = NULL; const char *devname; int ret, i, patch_regs = 0; int pulls = 0; diff --git a/drivers/mfd/wm8997-tables.c b/drivers/mfd/wm8997-tables.c index c0c25d75..cab2c68 100644 --- a/drivers/mfd/wm8997-tables.c +++ b/drivers/mfd/wm8997-tables.c @@ -17,7 +17,7 @@ #include "arizona.h" -static const struct reg_default wm8997_reva_patch[] = { +static const struct reg_sequence wm8997_reva_patch[] = { { 0x80, 0x0003 }, { 0x214, 0x0008 }, { 0x458, 0x0000 }, diff --git a/include/linux/regmap.h b/include/linux/regmap.h index 519c962..4a67590 100644 --- a/include/linux/regmap.h +++ b/include/linux/regmap.h @@ -50,6 +50,17 @@ struct reg_default { unsigned int def; }; +/** + * Register/value pairs for sequences of writes + * + * @reg: Register address. + * @def: Register value. + */ +struct reg_sequence { + unsigned int reg; + unsigned int def; +}; + #ifdef CONFIG_REGMAP enum regmap_endian { @@ -410,10 +421,10 @@ int regmap_raw_write(struct regmap *map, unsigned int reg, const void *val, size_t val_len); int regmap_bulk_write(struct regmap *map, unsigned int reg, const void *val, size_t val_count); -int regmap_multi_reg_write(struct regmap *map, const struct reg_default *regs, +int regmap_multi_reg_write(struct regmap *map, const struct reg_sequence *regs, int num_regs); int regmap_multi_reg_write_bypassed(struct regmap *map, - const struct reg_default *regs, + const struct reg_sequence *regs, int num_regs); int regmap_raw_write_async(struct regmap *map, unsigned int reg, const void *val, size_t val_len); @@ -452,7 +463,7 @@ void regcache_mark_dirty(struct regmap *map); bool regmap_check_range_table(struct regmap *map, unsigned int reg, const struct regmap_access_table *table); -int regmap_register_patch(struct regmap *map, const struct reg_default *regs, +int regmap_register_patch(struct regmap *map, const struct reg_sequence *regs, int num_regs); int regmap_parse_val(struct regmap *map, const void *buf, unsigned int *val); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 802e05e..5edd33f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1366,7 +1366,7 @@ static void arizona_wm5102_set_dac_comp(struct snd_soc_codec *codec, { struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); struct arizona *arizona = priv->arizona; - struct reg_default dac_comp[] = { + struct reg_sequence dac_comp[] = { { 0x80, 0x3 }, { ARIZONA_DAC_COMP_1, 0 }, { ARIZONA_DAC_COMP_2, 0 }, diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 8f40025..2813a1b 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -276,7 +276,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_cs35l32 = { }; /* Current and threshold powerup sequence Pg37 in datasheet */ -static const struct reg_default cs35l32_monitor_patch[] = { +static const struct reg_sequence cs35l32_monitor_patch[] = { { 0x00, 0x99 }, { 0x48, 0x17 }, diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 4de52c9..8b2d059 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1118,7 +1118,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_cs42l52 = { }; /* Current and threshold powerup sequence Pg37 */ -static const struct reg_default cs42l52_threshold_patch[] = { +static const struct reg_sequence cs42l52_threshold_patch[] = { { 0x00, 0x99 }, { 0x3E, 0xBA }, diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 21810e5..bf0fb3d 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1182,7 +1182,7 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { #if IS_ENABLED(CONFIG_I2C) -static struct reg_default da7210_regmap_i2c_patch[] = { +static struct reg_sequence da7210_regmap_i2c_patch[] = { /* System controller master disable */ { DA7210_STARTUP1, 0x00 }, @@ -1269,7 +1269,7 @@ static struct i2c_driver da7210_i2c_driver = { #if defined(CONFIG_SPI_MASTER) -static struct reg_default da7210_regmap_spi_patch[] = { +static struct reg_sequence da7210_regmap_spi_patch[] = { /* Dummy read to give two pulses over nCS for SPI */ { DA7210_AUX2, 0x00 }, { DA7210_AUX2, 0x00 }, diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 9bc78e5..1ed1f88 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -51,7 +51,7 @@ static const struct regmap_range_cfg rt5640_ranges[] = { .window_len = 0x1, }, }; -static const struct reg_default init_list[] = { +static const struct reg_sequence init_list[] = { {RT5640_PR_BASE + 0x3d, 0x3600}, {RT5640_PR_BASE + 0x12, 0x0aa8}, {RT5640_PR_BASE + 0x14, 0x0aaa}, diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9ce311e..c0f4be4 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -54,7 +54,7 @@ static const struct regmap_range_cfg rt5645_ranges[] = { }, }; -static const struct reg_default init_list[] = { +static const struct reg_sequence init_list[] = { {RT5645_PR_BASE + 0x3d, 0x3600}, {RT5645_PR_BASE + 0x1c, 0xfd20}, {RT5645_PR_BASE + 0x20, 0x611f}, @@ -63,7 +63,7 @@ static const struct reg_default init_list[] = { }; #define RT5645_INIT_REG_LEN ARRAY_SIZE(init_list) -static const struct reg_default rt5650_init_list[] = { +static const struct reg_sequence rt5650_init_list[] = { {0xf6, 0x0100}, }; diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index a3506e1..db9b866 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -46,7 +46,7 @@ static const struct regmap_range_cfg rt5651_ranges[] = { .window_len = 0x1, }, }; -static struct reg_default init_list[] = { +static struct reg_sequence init_list[] = { {RT5651_PR_BASE + 0x3d, 0x3e00}, }; diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index a9123d4..462a91f 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -51,7 +51,7 @@ static const struct regmap_range_cfg rt5670_ranges[] = { .window_len = 0x1, }, }; -static const struct reg_default init_list[] = { +static const struct reg_sequence init_list[] = { { RT5670_PR_BASE + 0x14, 0x9a8a }, { RT5670_PR_BASE + 0x38, 0x3ba1 }, { RT5670_PR_BASE + 0x3d, 0x3640 }, diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 31d969a..b897752 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -54,7 +54,7 @@ static const struct regmap_range_cfg rt5677_ranges[] = { }, }; -static const struct reg_default init_list[] = { +static const struct reg_sequence init_list[] = { {RT5677_ASRC_12, 0x0018}, {RT5677_PR_BASE + 0x3d, 0x364d}, {RT5677_PR_BASE + 0x17, 0x4fc0}, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index a7cf19b..83ae1eb 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1668,7 +1668,7 @@ static const struct i2c_device_id aic3x_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); -static const struct reg_default aic3007_class_d[] = { +static const struct reg_sequence aic3007_class_d[] = { /* Class-D speaker driver init; datasheet p. 46 */ { AIC3X_PAGE_SELECT, 0x0D }, { 0xD, 0x0D }, diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index c830832..6c60792 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -897,7 +897,7 @@ static bool wm2200_readable_register(struct device *dev, unsigned int reg) } } -static const struct reg_default wm2200_reva_patch[] = { +static const struct reg_sequence wm2200_reva_patch[] = { { 0x07, 0x0003 }, { 0x102, 0x0200 }, { 0x203, 0x0084 }, diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 4c10cd8..26d79bb 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1247,7 +1247,7 @@ static const struct snd_soc_dapm_route wm5100_dapm_routes[] = { { "PWM2", NULL, "PWM2 Driver" }, }; -static const struct reg_default wm5100_reva_patches[] = { +static const struct reg_sequence wm5100_reva_patches[] = { { WM5100_AUDIO_IF_1_10, 0 }, { WM5100_AUDIO_IF_1_11, 1 }, { WM5100_AUDIO_IF_1_12, 2 }, diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index c5748fd..05492e8 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3495,7 +3495,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8962 = { }; /* Improve power consumption for IN4 DC measurement mode */ -static const struct reg_default wm8962_dc_measure[] = { +static const struct reg_sequence wm8962_dc_measure[] = { { 0xfd, 0x1 }, { 0xcc, 0x40 }, { 0xfd, 0 }, diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 8a8db86..52ec64d 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1595,7 +1595,7 @@ static int wm8993_resume(struct snd_soc_codec *codec) #endif /* Tune DC servo configuration */ -static struct reg_default wm8993_regmap_patch[] = { +static struct reg_sequence wm8993_regmap_patch[] = { { 0x44, 3 }, { 0x56, 3 }, { 0x44, 0 }, -- 2.1.4

2 3

[alsa-devel] Applied "ASoC: wm5110: Use reg_sequence for multi_reg_write/register_patch" to the asoc tree
by Mark Brown 17 Jul '15

17 Jul '15

The patch ASoC: wm5110: Use reg_sequence for multi_reg_write/register_patch 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 dc606e05468fc5e3b125bffdd54253f853bfbb8e Mon Sep 17 00:00:00 2001 From: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> Date: Fri, 17 Jul 2015 15:11:10 +0100 Subject: [PATCH] ASoC: wm5110: Use reg_sequence for multi_reg_write/register_patch Introduced by: commit 8019ff6cfc04 ("regmap: Use reg_sequence for multi_reg_write / register_patch") Interacting with: commit d1acd31883d7 ("ASoC: wm5110: Add special DRE on/off handling for the headphone path") Signed-off-by: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm5110.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index b5c201b1e454..1f29991c1f71 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -185,7 +185,7 @@ static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w, return 0; } -static const struct reg_default wm5110_no_dre_left_enable[] = { +static const struct reg_sequence wm5110_no_dre_left_enable[] = { { 0x3024, 0xE410 }, { 0x3025, 0x0056 }, { 0x301B, 0x0224 }, @@ -203,7 +203,7 @@ static const struct reg_default wm5110_no_dre_left_enable[] = { { 0x3039, 0x3080 }, }; -static const struct reg_default wm5110_dre_left_enable[] = { +static const struct reg_sequence wm5110_dre_left_enable[] = { { 0x3024, 0x0231 }, { 0x3025, 0x0B00 }, { 0x301B, 0x0227 }, @@ -221,7 +221,7 @@ static const struct reg_default wm5110_dre_left_enable[] = { { 0x3039, 0x0B00 }, }; -static const struct reg_default wm5110_no_dre_right_enable[] = { +static const struct reg_sequence wm5110_no_dre_right_enable[] = { { 0x3074, 0xE414 }, { 0x3075, 0x0056 }, { 0x306B, 0x0224 }, @@ -239,7 +239,7 @@ static const struct reg_default wm5110_no_dre_right_enable[] = { { 0x3089, 0x3080 }, }; -static const struct reg_default wm5110_dre_right_enable[] = { +static const struct reg_sequence wm5110_dre_right_enable[] = { { 0x3074, 0x0231 }, { 0x3075, 0x0B00 }, { 0x306B, 0x0227 }, @@ -263,7 +263,7 @@ static int wm5110_hp_pre_enable(struct snd_soc_dapm_widget *w) struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); struct arizona *arizona = priv->arizona; unsigned int val = snd_soc_read(codec, ARIZONA_DRE_ENABLE); - const struct reg_default *wseq; + const struct reg_sequence *wseq; int nregs; switch (w->shift) { @@ -354,7 +354,7 @@ static int wm5110_hp_ev(struct snd_soc_dapm_widget *w, static int wm5110_clear_pga_volume(struct arizona *arizona, int output) { - struct reg_default clear_pga = { + struct reg_sequence clear_pga = { ARIZONA_OUTPUT_PATH_CONFIG_1L + output * 4, 0x80 }; int ret; -- 2.1.4

1 0

[alsa-devel] [PATCH] ALSA: hda - Check the return value from pm_runtime_get/put*()
by Takashi Iwai 17 Jul '15

17 Jul '15

This patch changes the return type of snd_hdac_power_up/down() and variants to pass the error code from the underlying pm_runtime_get/put() calls. Currently they are ignored, but in most places, these should be handled properly. As an example, the regmap handler is updated to check the return value and accesses the register only when the wakeup succeeds. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hdaudio.h | 16 ++++++++-------- sound/hda/hdac_device.c | 26 ++++++++++++++++++-------- sound/hda/hdac_regmap.c | 10 ++++++---- 3 files changed, 32 insertions(+), 20 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 4caf1fde8a4f..288c7fa509d8 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -164,15 +164,15 @@ static inline int snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, } #ifdef CONFIG_PM -void snd_hdac_power_up(struct hdac_device *codec); -void snd_hdac_power_down(struct hdac_device *codec); -void snd_hdac_power_up_pm(struct hdac_device *codec); -void snd_hdac_power_down_pm(struct hdac_device *codec); +int snd_hdac_power_up(struct hdac_device *codec); +int snd_hdac_power_down(struct hdac_device *codec); +int snd_hdac_power_up_pm(struct hdac_device *codec); +int snd_hdac_power_down_pm(struct hdac_device *codec); #else -static inline void snd_hdac_power_up(struct hdac_device *codec) {} -static inline void snd_hdac_power_down(struct hdac_device *codec) {} -static inline void snd_hdac_power_up_pm(struct hdac_device *codec) {} -static inline void snd_hdac_power_down_pm(struct hdac_device *codec) {} +static inline int snd_hdac_power_up(struct hdac_device *codec) { return 0; } +static inline int snd_hdac_power_down(struct hdac_device *codec) { return 0; } +static inline int snd_hdac_power_up_pm(struct hdac_device *codec) { return 0; } +static inline int snd_hdac_power_down_pm(struct hdac_device *codec) { return 0; } #endif /* diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index cdee7103f649..df7039ebe1fa 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -501,23 +501,27 @@ EXPORT_SYMBOL_GPL(snd_hdac_get_connections); * This function calls the runtime PM helper to power up the given codec. * Unlike snd_hdac_power_up_pm(), you should call this only for the code * path that isn't included in PM path. Otherwise it gets stuck. + * + * Returns zero if successful, or a negative error code. */ -void snd_hdac_power_up(struct hdac_device *codec) +int snd_hdac_power_up(struct hdac_device *codec) { - pm_runtime_get_sync(&codec->dev); + return pm_runtime_get_sync(&codec->dev); } EXPORT_SYMBOL_GPL(snd_hdac_power_up); /** * snd_hdac_power_down - power down the codec * @codec: the codec object + * + * Returns zero if successful, or a negative error code. */ -void snd_hdac_power_down(struct hdac_device *codec) +int snd_hdac_power_down(struct hdac_device *codec) { struct device *dev = &codec->dev; pm_runtime_mark_last_busy(dev); - pm_runtime_put_autosuspend(dev); + return pm_runtime_put_autosuspend(dev); } EXPORT_SYMBOL_GPL(snd_hdac_power_down); @@ -529,11 +533,14 @@ EXPORT_SYMBOL_GPL(snd_hdac_power_down); * which may be called by PM suspend/resume again. OTOH, if a power-up * call must wake up the sleeper (e.g. in a kctl callback), use * snd_hdac_power_up() instead. + * + * Returns zero if successful, or a negative error code. */ -void snd_hdac_power_up_pm(struct hdac_device *codec) +int snd_hdac_power_up_pm(struct hdac_device *codec) { if (!atomic_inc_not_zero(&codec->in_pm)) - snd_hdac_power_up(codec); + return snd_hdac_power_up(codec); + return 0; } EXPORT_SYMBOL_GPL(snd_hdac_power_up_pm); @@ -543,11 +550,14 @@ EXPORT_SYMBOL_GPL(snd_hdac_power_up_pm); * * Like snd_hdac_power_up_pm(), this function is used in a recursive * code path like init code which may be called by PM suspend/resume again. + * + * Returns zero if successful, or a negative error code. */ -void snd_hdac_power_down_pm(struct hdac_device *codec) +int snd_hdac_power_down_pm(struct hdac_device *codec) { if (atomic_dec_if_positive(&codec->in_pm) < 0) - snd_hdac_power_down(codec); + return snd_hdac_power_down(codec); + return 0; } EXPORT_SYMBOL_GPL(snd_hdac_power_down_pm); #endif diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 1eabcdf69457..b0ed870ffb88 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -410,8 +410,9 @@ int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg, err = reg_raw_write(codec, reg, val); if (err == -EAGAIN) { - snd_hdac_power_up_pm(codec); - err = reg_raw_write(codec, reg, val); + err = snd_hdac_power_up_pm(codec); + if (!err) + err = reg_raw_write(codec, reg, val); snd_hdac_power_down_pm(codec); } return err; @@ -442,8 +443,9 @@ int snd_hdac_regmap_read_raw(struct hdac_device *codec, unsigned int reg, err = reg_raw_read(codec, reg, val); if (err == -EAGAIN) { - snd_hdac_power_up_pm(codec); - err = reg_raw_read(codec, reg, val); + err = snd_hdac_power_up_pm(codec); + if (!err) + err = reg_raw_read(codec, reg, val); snd_hdac_power_down_pm(codec); } return err; -- 2.4.5

1 0

[alsa-devel] [PATCH] ALSA: pcm: Fix lockdep warning with nonatomic PCM ops
by Takashi Iwai 17 Jul '15

17 Jul '15

With the nonatomic PCM ops, the system may spew lockdep warnings like: ============================================= [ INFO: possible recursive locking detected ] 4.2.0-rc1-jeejaval3 #12 Not tainted --------------------------------------------- aplay/4029 is trying to acquire lock: (snd_pcm_link_rwsem){.+.+.+}, at: [<ffffffff816fd473>] snd_pcm_stream_lock+0x43/0x60 but task is already holding lock: (snd_pcm_link_rwsem){.+.+.+}, at: [<ffffffff816fcf29>] snd_pcm_action_nonatomic+0x29/0x80 other info that might help us debug this: Possible unsafe locking scenario: CPU0 ---- lock(snd_pcm_link_rwsem); lock(snd_pcm_link_rwsem); Although this is false-positive as the rwsem is taken always as read-only for these code paths, it's certainly annoying to see this at any occasion. A simple fix is to use down_read_nested() in snd_pcm_stream_lock() that can be called inside another lock. Reported-by: Vinod Koul <vinod.koul(a)intel.com> Reported-by: Jeeja Kp <jeeja.kp(a)intel.com> Tested-by: Jeeja Kp <jeeja.kp(a)intel.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index d126c03361ae..75888dd38a7f 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -85,7 +85,7 @@ static DECLARE_RWSEM(snd_pcm_link_rwsem); void snd_pcm_stream_lock(struct snd_pcm_substream *substream) { if (substream->pcm->nonatomic) { - down_read(&snd_pcm_link_rwsem); + down_read_nested(&snd_pcm_link_rwsem, SINGLE_DEPTH_NESTING); mutex_lock(&substream->self_group.mutex); } else { read_lock(&snd_pcm_link_rwlock); -- 2.4.5

1 0

[alsa-devel] [asoc:for-next 6/18] sound/soc/codecs/rt5645.c:2931:3: warning: 'return' with a value, in function returning void
by kbuild test robot 17 Jul '15

17 Jul '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: db91b047f7131c7e8fe00ca9e38d14c0c103363f commit: 947d4dc4cd5be5da8e5e82f314d5cfb85a3eb3cc [6/18] Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next config: i386-randconfig-x009-201528 (attached as .config) reproduce: git checkout 947d4dc4cd5be5da8e5e82f314d5cfb85a3eb3cc # save the attached .config to linux build tree make ARCH=i386 All warnings (new ones prefixed by >>): sound/soc/codecs/rt5645.c: In function 'rt5645_jack_detect_work': >> sound/soc/codecs/rt5645.c:2931:3: warning: 'return' with a value, in function returning void return -EINVAL; ^ vim +/return +2931 sound/soc/codecs/rt5645.c 6e747d53 Bard Liao 2015-04-28 2915 regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1, 6e747d53 Bard Liao 2015-04-28 2916 RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); 6e747d53 Bard Liao 2015-04-28 2917 } 345b0f50 John Lin 2015-05-18 2918 rt5645_irq(0, rt5645); f3fa1bbd Oder Chiou 2014-09-19 2919 f3fa1bbd Oder Chiou 2014-09-19 2920 return 0; f3fa1bbd Oder Chiou 2014-09-19 2921 } f3fa1bbd Oder Chiou 2014-09-19 2922 EXPORT_SYMBOL_GPL(rt5645_set_jack_detect); f3fa1bbd Oder Chiou 2014-09-19 2923 cd6e82b8 Oder Chiou 2014-10-07 2924 static void rt5645_jack_detect_work(struct work_struct *work) cd6e82b8 Oder Chiou 2014-10-07 2925 { cd6e82b8 Oder Chiou 2014-10-07 2926 struct rt5645_priv *rt5645 = cd6e82b8 Oder Chiou 2014-10-07 2927 container_of(work, struct rt5645_priv, jack_detect_work.work); 6e747d53 Bard Liao 2015-04-28 2928 int val, btn_type, gpio_state = 0, report = 0; 6e747d53 Bard Liao 2015-04-28 2929 f2a5ded3 Nicolas Boichat 2015-07-17 2930 if (!rt5645->codec) f2a5ded3 Nicolas Boichat 2015-07-17 @2931 return -EINVAL; f2a5ded3 Nicolas Boichat 2015-07-17 2932 6e747d53 Bard Liao 2015-04-28 2933 switch (rt5645->pdata.jd_mode) { 6e747d53 Bard Liao 2015-04-28 2934 case 0: /* Not using rt5645 JD */ 0b0cefc8 Oder Chiou 2015-06-10 2935 if (rt5645->gpiod_hp_det) { 0b0cefc8 Oder Chiou 2015-06-10 2936 gpio_state = gpiod_get_value(rt5645->gpiod_hp_det); 0b0cefc8 Oder Chiou 2015-06-10 2937 dev_dbg(rt5645->codec->dev, "gpio_state = %d\n", 0b0cefc8 Oder Chiou 2015-06-10 2938 gpio_state); 0b0cefc8 Oder Chiou 2015-06-10 2939 report = rt5645_jack_detect(rt5645->codec, gpio_state); :::::: The code at line 2931 was first introduced by commit :::::: f2a5ded38592e5936a099ea6535ad5d3addcbc9d ASoC: rt5645: Check if codec is initialized in workqueue handler :::::: TO: Nicolas Boichat <drinkcat(a)chromium.org> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] HDMI hotplug on Skylake when power well is off
by David Henningsson 17 Jul '15

17 Jul '15

Hi, I'm trying to debug an issue here where the HDMI hotplug events are not delivered to the audio side when the power well is off. This is on a Skylake machine (running in HDA mode). I'm not sure whether the problem is upstream or due to my own patches while testing, so I was wondering how this is supposed to be working, so I can troubleshoot further? Should there be an IRQ on the HDA controller even if the power well is off, and if not, how should the audio driver be notified that an HDMI hotplug event has happened? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

3 13

[alsa-devel] [PATCH v4] ASoC: rt5645: Check if codec is initialized in workqueue handler
by Nicolas Boichat 17 Jul '15

17 Jul '15

This fixes kernel panic on boot, if rt5645->codec is NULL when rt5645_jack_detect_work is first called. rt5645_jack_detect_work needs rt5645->codec to be initialized to setup dapm pins. Also, reporting jack events is useless, as the jacks cannot be set before the codec is ready. Since we manually call the interrupt handler in rt5645_set_jack_detect, the initial jack state will be reported correctly, and dapm pins will be setup at that time. Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> --- This replaces 3c2234569cfff83718792f810a9bb855b78c2473, and applies on top of Linux 4.2-rc1, so that this patch can be applied in its own branch, without triggering a compile warning. Also, reworded the message to make it more obvious this can cause a kernel panic. Thanks! sound/soc/codecs/rt5645.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9ce311e..e9cc3aa 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2943,6 +2943,9 @@ static int rt5645_irq_detection(struct rt5645_priv *rt5645) { int val, btn_type, gpio_state = 0, report = 0; + if (!rt5645->codec) + return -EINVAL; + switch (rt5645->pdata.jd_mode) { case 0: /* Not using rt5645 JD */ if (rt5645->gpiod_hp_det) { -- 2.4.3.573.g4eafbef

2 3

[alsa-devel] [PATCH] ALSA: hda: fix kstrdup return value
by Heloise NH 17 Jul '15

17 Jul '15

From: kernelpatch_update <kernelpatch_update(a)163.com> In kstrdup we should return -ENOMEM when it reports an memory allocation failure, while the -ENODEV is referred to a failure in finding the cpu node in the device tree. Signed-off-by: Heloise NH <kernelpatch_update(a)163.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5de3c5d..d78fa71 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -975,7 +975,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, if (codec->bus->modelname) { codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL); if (!codec->modelname) { - err = -ENODEV; + err = -ENOMEM; goto error; } } -- 1.9.1

2 1

[alsa-devel] bug-report: Play Bad Sound After Test Runs for 6 hrs
by Yadi 17 Jul '15

17 Jul '15

Hi, everyone, I encounter a strange issue that sound would be bad after running below testcase on intel target over 7 hours. composite: libsdl1.2-dev 1.2.14-4ubuntu1.1 libsdl-mixer1.2 1.2.8-6build1 alsa-lib-1.0.29 this issue would disappear if I try to attach process by gdb or adding print in alsa-lib/pcm. any comments would be appreciated. Yadi

2 1

[alsa-devel] drm/edid: sound/core: pcm_drm_eld.c compilation error in 4.2
by Jean-Francois Moine 17 Jul '15

17 Jul '15

The file sound/core/pcm_drm_eld.c which was added by the commit 838d1631b766529213684f07dd71cdf2e92f0623 ALSA: pcm: add DRM ELD helper lacks the function drm_eld_sad() which was defined by Russell's patch http://article.gmane.org/gmane.linux.ports.arm.kernel/411574 drm/edid: add function to help find SADs This patch has not been applied. -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

3 2

[alsa-devel] [PATCH] ASoC: rt5645: remove unused variable
by Sudip Mukherjee 17 Jul '15

17 Jul '15

We were getting a build warning about unused variable. This variable was a leftover from: a4e3c5fa955c ("ASoC: rt5645: Simplify rt5645_enable_push_button_irq") Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/codecs/rt5645.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index d24c979..ff091d6 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2763,7 +2763,6 @@ static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, bool enable) { struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); - struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); if (enable) { snd_soc_dapm_force_enable_pin(dapm, "ADC L power"); -- 1.8.1.2

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: rt5645: Fix missing free_irq
by Koro Chen 17 Jul '15

17 Jul '15

The driver does not free irq when snd_soc_register_codec returns error. It does not return error when request irq failed, either. Add return when request irq failed, and free_irq if snd_soc_register_codec failed. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- Change since v1: - use free_irq instead of devm_request_threaded_irq --- sound/soc/codecs/rt5645.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 2cedf0d..56e8c31 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3401,12 +3401,23 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5645", rt5645); - if (ret) + if (ret) { dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + return ret; + } } - return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, - rt5645_dai, ARRAY_SIZE(rt5645_dai)); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, + rt5645_dai, ARRAY_SIZE(rt5645_dai)); + if (ret) + goto err_irq; + + return 0; + +err_irq: + if (rt5645->i2c->irq) + free_irq(rt5645->i2c->irq, rt5645); + return ret; } static int rt5645_i2c_remove(struct i2c_client *i2c) -- 1.8.1.1.dirty

1 1

[alsa-devel] [PATCH] ASoC: rt5645: Remove unused rt5645 variable
by Nicolas Boichat 17 Jul '15

17 Jul '15

"ASoC: rt5645: Simplify rt5645_enable_push_button_irq" removes the test that accessed rt5645->pdata.jd_mode (that test is now done in rt5645_jack_detect only), so we do not need that variable anymore. Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> --- Applies on top of topic/rt5645. Sorry for not catching this earlier. sound/soc/codecs/rt5645.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 2cedf0d..c959ee0 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2763,7 +2763,6 @@ static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, bool enable) { struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); - struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); if (enable) { snd_soc_dapm_force_enable_pin(dapm, "ADC L power"); -- 2.4.3.573.g4eafbef

1 0

[alsa-devel] [PATCH] ASoC: Constify dev_pm_ops variables
by Axel Lin 17 Jul '15

17 Jul '15

The dev_pm_ops variables are not modified after initialization in these drivers, so make them const. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index cdaa175..878d43a 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2481,7 +2481,7 @@ static int wm2200_runtime_resume(struct device *dev) } #endif -static struct dev_pm_ops wm2200_pm = { +static const struct dev_pm_ops wm2200_pm = { SET_RUNTIME_PM_OPS(wm2200_runtime_suspend, wm2200_runtime_resume, NULL) }; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index e5572d2..eb494d0 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2706,7 +2706,7 @@ static int wm5100_runtime_resume(struct device *dev) } #endif -static struct dev_pm_ops wm5100_pm = { +static const struct dev_pm_ops wm5100_pm = { SET_RUNTIME_PM_OPS(wm5100_runtime_suspend, wm5100_runtime_resume, NULL) }; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index c67e304..cbd1886 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3859,7 +3859,7 @@ static int wm8962_runtime_suspend(struct device *dev) } #endif -static struct dev_pm_ops wm8962_pm = { +static const struct dev_pm_ops wm8962_pm = { SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; -- 2.1.0

1 0

[alsa-devel] Applied "ASoC: sti: sti_uniperiph_dai_create_ctrl() can be static" to the asoc tree
by Mark Brown 17 Jul '15

17 Jul '15

The patch ASoC: sti: sti_uniperiph_dai_create_ctrl() can be static 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 f2da4542268503d22869617b7dbcb40d364c78bb Mon Sep 17 00:00:00 2001 From: kbuild test robot <fengguang.wu(a)intel.com> Date: Fri, 17 Jul 2015 07:44:09 +0800 Subject: [PATCH] ASoC: sti: sti_uniperiph_dai_create_ctrl() can be static Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sti/sti_uniperif.c | 2 +- sound/soc/sti/uniperif_player.c | 4 ++-- sound/soc/sti/uniperif_reader.c | 4 ++-- 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index 749e6b294184..83a301f2aa66 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -14,7 +14,7 @@ * This function is used to create Ctrl associated to DAI but also pcm device. * Request is done by front end to associate ctrl with pcm device id */ -int sti_uniperiph_dai_create_ctrl(struct snd_soc_dai *dai) +static int sti_uniperiph_dai_create_ctrl(struct snd_soc_dai *dai) { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct uniperif *uni = priv->dai_data.uni; diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index d990d2c81a5d..f609089a6e00 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -43,7 +43,7 @@ * Note: snd_pcm_hardware is linked to DMA controller but is declared here to * integrate DAI_CPU capability in term of rate and supported channels */ -const struct snd_pcm_hardware uni_player_pcm_hw = { +static const struct snd_pcm_hardware uni_player_pcm_hw = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID, @@ -175,7 +175,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) return ret; } -int uni_player_clk_set_rate(struct uniperif *player, unsigned long rate) +static int uni_player_clk_set_rate(struct uniperif *player, unsigned long rate) { int rate_adjusted, rate_achieved, delta, ret; int adjustment = player->clk_adj; diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 7d83827ba660..c502626f339b 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -17,7 +17,7 @@ * Note: snd_pcm_hardware is linked to DMA controller but is declared here to * integrate unireader capability in term of rate and supported channels */ -const struct snd_pcm_hardware uni_reader_pcm_hw = { +static const struct snd_pcm_hardware uni_reader_pcm_hw = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID, @@ -324,7 +324,7 @@ static int uni_reader_parse_dt(struct platform_device *pdev, return 0; } -const struct snd_soc_dai_ops uni_reader_dai_ops = { +static const struct snd_soc_dai_ops uni_reader_dai_ops = { .shutdown = uni_reader_shutdown, .prepare = uni_reader_prepare, .trigger = uni_reader_trigger, -- 2.1.4

1 0

[alsa-devel] [PATCH v2 0/12] Discover and probe dependencies
by Tomeu Vizoso 17 Jul '15

17 Jul '15

Hi, this is version 2 of a series that probes devices in dependency order so as to avoid deferred probes. While deferred probing is a powerful solution that makes sure that you eventually get a working system at the end of the boot, can make it very time consuming to find out why a device didn't probe and can also introduce big delays in when a device actually probes by sending it to the end of the deferred queue. So far I have only tested on a Tegra124 Chromebook. Thanks, Tomeu Changes in v2: - Instead of delaying all probes until late_initcall, only delay matches of platform devices that have a firmware node attached. - Allow bindings implementations register a function instead of using class callbacks, as not only subsystems implement firmware bindings. - Move strends to string.h - Document that consumers of backlight devices can use the 'backlight' property to hold a phandle to the backlight device. - Allocate the list of dependencies and pass it to the function that fills it. Tomeu Vizoso (12): device: property: delay device-driver matches device: property: find dependencies of a firmware node string: Introduce strends() gpio: register dependency parser for firmware nodes gpu: host1x: register dependency parser for firmware nodes backlight: Document consumers of backlight nodes backlight: register dependency parser for firmware nodes USB: EHCI: register dependency parser for firmware nodes regulator: register dependency parser for firmware nodes pwm: register dependency parser for firmware nodes ASoC: tegra: register dependency parser for firmware nodes driver-core: probe dependencies before probing .../bindings/video/backlight/backlight.txt | 22 ++++ drivers/base/dd.c | 139 +++++++++++++++++++++ drivers/base/property.c | 120 ++++++++++++++++++ drivers/gpio/gpiolib.c | 54 ++++++++ drivers/gpu/host1x/dev.c | 26 ++++ drivers/pwm/core.c | 28 +++++ drivers/regulator/core.c | 27 ++++ drivers/usb/host/ehci-tegra.c | 16 +++ drivers/video/backlight/backlight.c | 16 +++ include/linux/fwnode.h | 5 + include/linux/property.h | 12 ++ include/linux/string.h | 13 ++ sound/soc/tegra/tegra_max98090.c | 42 ++++++- 13 files changed, 519 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/video/backlight/backlight.txt -- 2.4.1

4 28

[alsa-devel] [PATCH 1/4] ASoC: max9877: Make driver global regmap struct local
by Lars-Peter Clausen 16 Jul '15

16 Jul '15

Use a stack local variable to handle function local state rather than a global static variable. The later has a potential for race conditions if the probe function runs for two devices concurrently. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/max9877.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index e1df06f..b469e1c 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -20,8 +20,6 @@ #include "max9877.h" -static struct regmap *regmap; - static const struct reg_default max9877_regs[] = { { 0, 0x40 }, { 1, 0x00 }, @@ -145,6 +143,7 @@ static const struct regmap_config max9877_regmap = { static int max9877_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { + struct regmap *regmap; int i; regmap = devm_regmap_init_i2c(client, &max9877_regmap); -- 2.1.4

2 7

[alsa-devel] [PATCH 0/6] ASoC: Add audio for sti platforms round 2
by Arnaud Pouliquen 16 Jul '15

16 Jul '15

New set of patches after partial integration of [PATCH v3 0/9] ASoC: Add audio for sti platforms New patches: ASoC: sti: minor corrections for uniplayer Patches rebased from [PATCH v3 0/9] asoc: Add audio for sti platforms ASoC: sti: Add CPU DAI driver for capture ( IRQ enable /disable cleaned) ASoC: sti: Add uniperipheral dai driver ASoC: Add ability to build sti drivers ASoC: sti: Add clock adjustement control ASoC: sti: Add IEC control Arnaud Pouliquen (6): ASoC: sti: minor corrections for uniplayer ASoC: sti: Add CPU DAI driver for capture ASoC: sti: Add uniperipheral dai driver ASoC: Add ability to build sti drivers ASoC: sti: Add clock adjustement control ASoC: sti: Add IEC control sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/sti/Kconfig | 11 ++ sound/soc/sti/Makefile | 4 + sound/soc/sti/sti_uniperif.c | 253 ++++++++++++++++++++++++++++ sound/soc/sti/uniperif.h | 14 ++ sound/soc/sti/uniperif_player.c | 226 ++++++++++++++++++++++++- sound/soc/sti/uniperif_reader.c | 362 ++++++++++++++++++++++++++++++++++++++++ 8 files changed, 869 insertions(+), 3 deletions(-) create mode 100644 sound/soc/sti/Kconfig create mode 100644 sound/soc/sti/Makefile create mode 100644 sound/soc/sti/sti_uniperif.c create mode 100644 sound/soc/sti/uniperif_reader.c -- 1.9.1

2 10

[alsa-devel] [PATCH 0/7] ASoC: Intel: Skylake: Add code loader DMA support
by Vinod Koul 16 Jul '15

16 Jul '15

This series add code loader DMA support to Skylake driver and then adds the IPC init routines along with few smaller changes Omair M Abdullah (1): ASoC: Intel: Skylake: Print error code in IPC for SKL Subhransu S. Prusty (3): ASoC: Intel: Skylake: Add dsp loader ops ASoC: Intel: Skylake: Process code loader DMA interrupt ASoC: Intel: Skylake: Add SKL DSP initialization Vinod Koul (3): ASoC: Intel: Skylake: Add code loader DMA registers ASoC: Intel: Skylake: add code loader DMA operations ASoC: Intel: Skylake: Add code loader DMA APIs sound/soc/intel/common/sst-dsp-priv.h | 3 + sound/soc/intel/skylake/Makefile | 3 +- sound/soc/intel/skylake/skl-sst-cldma.c | 327 ++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst-cldma.h | 251 ++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst-dsp.c | 5 + sound/soc/intel/skylake/skl-sst-dsp.h | 26 +++ sound/soc/intel/skylake/skl-sst-ipc.c | 11 ++ sound/soc/intel/skylake/skl-sst.c | 280 +++++++++++++++++++++++++++ 8 files changed, 905 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/skylake/skl-sst-cldma.c create mode 100644 sound/soc/intel/skylake/skl-sst-cldma.h create mode 100644 sound/soc/intel/skylake/skl-sst.c -- 1.9.1

2 15

[alsa-devel] [PATCH v4] ASoC: cs4349: Add support for Cirrus Logic CS4349
by timothyc.howe＠gmail.com 16 Jul '15

16 Jul '15

From: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- Documentation/devicetree/bindings/sound/cs4349.txt | 19 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4349.c | 401 +++++++++++++++++++++ sound/soc/codecs/cs4349.h | 146 ++++++++ 5 files changed, 574 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs4349.txt create mode 100644 sound/soc/codecs/cs4349.c create mode 100644 sound/soc/codecs/cs4349.h diff --git a/Documentation/devicetree/bindings/sound/cs4349.txt b/Documentation/devicetree/bindings/sound/cs4349.txt new file mode 100644 index 0000000..54c117b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4349.txt @@ -0,0 +1,19 @@ +CS4349 audio CODEC + +Required properties: + + - compatible : "cirrus,cs4349" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + +Example: + +codec: cs4349@48 { + compatible = "cirrus,cs4349"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7223c7f..4a78606 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -54,6 +54,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_I2C if I2C select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C + select SND_SOC_CS4349 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -413,6 +414,11 @@ config SND_SOC_CS42XX8_I2C select SND_SOC_CS42XX8 select REGMAP_I2C +# Cirrus Logic CS4349 HiFi DAC +config SND_SOC_CS4349 + tristate "Cirrus Logic CS4349 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9f7136c..36a9f37 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -46,6 +46,7 @@ snd-soc-cs4271-i2c-objs := cs4271-i2c.o snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o +snd-soc-cs4349-objs := cs4349.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -237,6 +238,7 @@ obj-$(CONFIG_SND_SOC_CS4271_I2C) += snd-soc-cs4271-i2c.o obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o +obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c new file mode 100644 index 0000000..a8df8a74 --- /dev/null +++ b/sound/soc/codecs/cs4349.c @@ -0,0 +1,401 @@ +/* + * cs4349.c -- CS4349 ALSA Soc Audio driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Authors: Tim Howe <Tim.Howe(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs4349.h" + + +static const struct reg_default cs4349_reg_defaults[] = { + { 2, 0x00 }, /* r02 - Mode Control */ + { 3, 0x09 }, /* r03 - Volume, Mixing and Inversion Control */ + { 4, 0x81 }, /* r04 - Mute Control */ + { 5, 0x00 }, /* r05 - Channel A Volume Control */ + { 6, 0x00 }, /* r06 - Channel B Volume Control */ + { 7, 0xB1 }, /* r07 - Ramp and Filter Control */ + { 8, 0x1C }, /* r08 - Misc. Control */ +}; + +/* Private data for the CS4349 */ +struct cs4349_private { + struct regmap *regmap; + struct cs4349_platform_data pdata; + struct gpio_desc *reset_gpio; + unsigned int mode; + int rate; +}; + +static bool cs4349_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS4349_CHIPID: + case CS4349_MODE: + case CS4349_VMI: + case CS4349_MUTE: + case CS4349_VOLA: + case CS4349_VOLB: + case CS4349_RMPFLT: + case CS4349_MISC: + return true; + default: + return false; + } +} + +static int cs4349_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + unsigned int fmt; + + fmt = format & SND_SOC_DAIFMT_FORMAT_MASK; + + switch (fmt) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + cs4349->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs4349_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + int mode, fmt, ret; + + mode = snd_soc_read(codec, CS4349_MODE); + cs4349->rate = params_rate(params); + + switch (cs4349->mode) { + case SND_SOC_DAIFMT_I2S: + mode |= MODE_FORMAT(DIF_I2S); + break; + case SND_SOC_DAIFMT_LEFT_J: + mode |= MODE_FORMAT(DIF_LEFT_JST); + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_width(params)) { + case 16: + fmt = DIF_RGHT_JST16; + break; + case 24: + fmt = DIF_RGHT_JST24; + break; + default: + return -EINVAL; + } + mode |= MODE_FORMAT(fmt); + break; + default: + return -EINVAL; + } + + ret = snd_soc_write(codec, CS4349_MODE, mode); + if (ret < 0) + return ret; + + return 0; +} + +static int cs4349_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int reg; + + reg = 0; + if (mute) + reg = MUTE_AB_MASK; + + return snd_soc_update_bits(codec, CS4349_MUTE, MUTE_AB_MASK, reg); +} + +static DECLARE_TLV_DB_SCALE(dig_tlv, -12750, 50, 0); + +static const char * const chan_mix_texts[] = { + "Mute", "MuteA", "MuteA SwapB", "MuteA MonoB", "SwapA MuteB", + "BothR", "Swap", "SwapA MonoB", "MuteB", "Normal", "BothL", + "MonoB", "MonoA MuteB", "MonoA", "MonoA SwapB", "Mono", + /*Normal == Channel A = Left, Channel B = Right*/ +}; + +static const char * const fm_texts[] = { + "Auto", "Single", "Double", "Quad", +}; + +static const char * const deemph_texts[] = { + "None", "44.1k", "48k", "32k", +}; + +static const char * const softr_zeroc_texts[] = { + "Immediate", "Zero Cross", "Soft Ramp", "SR on ZC", +}; + +static int deemph_values[] = { + 0, 4, 8, 12, +}; + +static int softr_zeroc_values[] = { + 0, 64, 128, 192, +}; + +static const struct soc_enum chan_mix_enum = + SOC_ENUM_SINGLE(CS4349_VMI, 0, + ARRAY_SIZE(chan_mix_texts), + chan_mix_texts); + +static const struct soc_enum fm_mode_enum = + SOC_ENUM_SINGLE(CS4349_MODE, 0, + ARRAY_SIZE(fm_texts), + fm_texts); + +static SOC_VALUE_ENUM_SINGLE_DECL(deemph_enum, CS4349_MODE, 0, DEM_MASK, + deemph_texts, deemph_values); + +static SOC_VALUE_ENUM_SINGLE_DECL(softr_zeroc_enum, CS4349_RMPFLT, 0, + SR_ZC_MASK, softr_zeroc_texts, + softr_zeroc_values); + +static const struct snd_kcontrol_new cs4349_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", + CS4349_VOLA, CS4349_VOLB, 0, 0xFF, 1, dig_tlv), + SOC_ENUM("Functional Mode", fm_mode_enum), + SOC_ENUM("De-Emphasis Control", deemph_enum), + SOC_ENUM("Soft Ramp Zero Cross Control", softr_zeroc_enum), + SOC_ENUM("Channel Mixer", chan_mix_enum), + SOC_SINGLE("VolA = VolB Switch", CS4349_VMI, 7, 1, 0), + SOC_SINGLE("InvertA Switch", CS4349_VMI, 6, 1, 0), + SOC_SINGLE("InvertB Switch", CS4349_VMI, 5, 1, 0), + SOC_SINGLE("Auto-Mute Switch", CS4349_MUTE, 7, 1, 0), + SOC_SINGLE("MUTEC A = B Switch", CS4349_MUTE, 5, 1, 0), + SOC_SINGLE("Soft Ramp Up Switch", CS4349_RMPFLT, 5, 1, 0), + SOC_SINGLE("Soft Ramp Down Switch", CS4349_RMPFLT, 4, 1, 0), + SOC_SINGLE("Slow Roll Off Filter Switch", CS4349_RMPFLT, 2, 1, 0), + SOC_SINGLE("Freeze Switch", CS4349_MISC, 5, 1, 0), + SOC_SINGLE("Popguard Switch", CS4349_MISC, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget cs4349_dapm_widgets[] = { + SND_SOC_DAPM_DAC("HiFi DAC", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_OUTPUT("OutputA"), + SND_SOC_DAPM_OUTPUT("OutputB"), +}; + +static const struct snd_soc_dapm_route cs4349_routes[] = { + {"DAC Playback", NULL, "OutputA"}, + {"DAC Playback", NULL, "OutputB"}, + + {"OutputA", NULL, "HiFi DAC"}, + {"OutputB", NULL, "HiFi DAC"}, +}; + +#define CS4349_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define CS4349_PCM_RATES SNDRV_PCM_RATE_8000_192000 + +static const struct snd_soc_dai_ops cs4349_dai_ops = { + .hw_params = cs4349_pcm_hw_params, + .set_fmt = cs4349_set_dai_fmt, + .digital_mute = cs4349_digital_mute, +}; + +static struct snd_soc_dai_driver cs4349_dai = { + .name = "cs4349_hifi", + .playback = { + .stream_name = "DAC Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS4349_PCM_RATES, + .formats = CS4349_PCM_FORMATS, + }, + .ops = &cs4349_dai_ops, + .symmetric_rates = 1, +}; + +static struct snd_soc_codec_driver soc_codec_dev_cs4349 = { + .controls = cs4349_snd_controls, + .num_controls = ARRAY_SIZE(cs4349_snd_controls), + + .dapm_widgets = cs4349_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4349_dapm_widgets), + .dapm_routes = cs4349_routes, + .num_dapm_routes = ARRAY_SIZE(cs4349_routes), +}; + +static struct regmap_config cs4349_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS4349_NUMREGS, + .reg_defaults = cs4349_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4349_reg_defaults), + .readable_reg = cs4349_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs4349_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs4349_private *cs4349; + struct cs4349_platform_data *pdata = dev_get_platdata(&client->dev); + int ret = 0; + + cs4349 = devm_kzalloc(&client->dev, sizeof(*cs4349), GFP_KERNEL); + if (!cs4349) + return -ENOMEM; + + cs4349->regmap = devm_regmap_init_i2c(client, &cs4349_regmap); + if (IS_ERR(cs4349->regmap)) { + ret = PTR_ERR(cs4349->regmap); + dev_err(&client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + if (pdata) + cs4349->pdata = *pdata; + + /* Reset the Device */ + cs4349->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs4349->reset_gpio)) + return PTR_ERR(cs4349->reset_gpio); + + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + + i2c_set_clientdata(client, cs4349); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4349, + &cs4349_dai, 1); +} + +static int cs4349_i2c_remove(struct i2c_client *client) +{ + struct cs4349_private *cs4349 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + /* Hold down reset */ + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int cs4349_runtime_suspend(struct device *dev) +{ + struct cs4349_private *cs4349 = dev_get_drvdata(dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + int ret; + + ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 1); + if (ret < 0) + return ret; + + regcache_cache_only(cs4349->regmap, true); + + /* Hold down reset */ + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + + return 0; +} + +static int cs4349_runtime_resume(struct device *dev) +{ + struct cs4349_private *cs4349 = dev_get_drvdata(dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + int ret; + + ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 0); + if (ret < 0) + return ret; + + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + + regcache_cache_only(cs4349->regmap, false); + regcache_sync(cs4349->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs4349_runtime_pm = { + SET_RUNTIME_PM_OPS(cs4349_runtime_suspend, cs4349_runtime_resume, + NULL) +}; + +static const struct of_device_id cs4349_of_match[] = { + { .compatible = "cirrus,cs4349", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs4349_of_match); + +static const struct i2c_device_id cs4349_i2c_id[] = { + {"cs4349", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs4349_i2c_id); + +static struct i2c_driver cs4349_i2c_driver = { + .driver = { + .name = "cs4349", + .owner = THIS_MODULE, + .of_match_table = cs4349_of_match, + }, + .id_table = cs4349_i2c_id, + .probe = cs4349_i2c_probe, + .remove = cs4349_i2c_remove, +}; + +module_i2c_driver(cs4349_i2c_driver); + +MODULE_AUTHOR("Tim Howe <tim.howe(a)cirrus.com>"); +MODULE_DESCRIPTION("Cirrus Logic CS4349 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4349.h b/sound/soc/codecs/cs4349.h new file mode 100644 index 0000000..3884a89 --- /dev/null +++ b/sound/soc/codecs/cs4349.h @@ -0,0 +1,146 @@ +/* + * ALSA SoC CS4349 codec driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Tim Howe <Tim.Howe(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + */ + +#ifndef __CS4349_H__ +#define __CS4349_H__ + +struct cs4349_platform_data { + + /* GPIO for Reset */ + unsigned int gpio_nreset; + +}; + +/* CS4349 registers addresses */ +#define CS4349_CHIPID 0x01 /* Device and Rev ID, Read Only */ +#define CS4349_MODE 0x02 /* Mode Control */ +#define CS4349_VMI 0x03 /* Volume, Mixing, Inversion Control */ +#define CS4349_MUTE 0x04 /* Mute Control */ +#define CS4349_VOLA 0x05 /* DAC Channel A Volume Control */ +#define CS4349_VOLB 0x06 /* DAC Channel B Volume Control */ +#define CS4349_RMPFLT 0x07 /* Ramp and Filter Control */ +#define CS4349_MISC 0x08 /* Power Down,Freeze Control,Pop Stop*/ + +#define CS4349_FIRSTREG 0x01 +#define CS4349_LASTREG 0x08 +#define CS4349_NUMREGS (CS4349_LASTREG - CS4349_FIRSTREG + 1) +#define CS4349_I2C_INCR 0x80 + + +/* Device and Revision ID */ +#define CS4349_REVA 0xF0 /* Rev A */ +#define CS4349_REVB 0xF1 /* Rev B */ +#define CS4349_REVC2 0xFF /* Rev C2 */ + + +/* PDN_DONE Poll Maximum + * If soft ramp is set it will take much longer to power down + * the system. + */ +#define PDN_POLL_MAX 900 + + +/* Bitfield Definitions */ + +/* CS4349_MODE */ +/* (Digital Interface Format, De-Emphasis Control, Functional Mode */ +#define DIF2 (1 << 6) +#define DIF1 (1 << 5) +#define DIF0 (1 << 4) +#define DEM1 (1 << 3) +#define DEM0 (1 << 2) +#define FM1 (1 << 1) +#define DIF_LEFT_JST 0x00 +#define DIF_I2S 0x01 +#define DIF_RGHT_JST16 0x02 +#define DIF_RGHT_JST24 0x03 +#define DIF_TDM0 0x04 +#define DIF_TDM1 0x05 +#define DIF_TDM2 0x06 +#define DIF_TDM3 0x07 +#define DIF_MASK 0x70 +#define MODE_FORMAT(x) (((x)&7)<<4) +#define DEM_MASK 0x0C +#define NO_DEM 0x00 +#define DEM_441 0x04 +#define DEM_48K 0x08 +#define DEM_32K 0x0C +#define FM_AUTO 0x00 +#define FM_SNGL 0x01 +#define FM_DBL 0x02 +#define FM_QUAD 0x03 +#define FM_SNGL_MIN 30000 +#define FM_SNGL_MAX 54000 +#define FM_DBL_MAX 108000 +#define FM_QUAD_MAX 216000 +#define FM_MASK 0x03 + +/* CS4349_VMI (VMI = Volume, Mixing and Inversion Controls) */ +#define VOLBISA (1 << 7) +#define VOLAISB (1 << 7) +/* INVERT_A only available for Left Jstfd, Right Jstfd16 and Right Jstfd24 */ +#define INVERT_A (1 << 6) +/* INVERT_B only available for Left Jstfd, Right Jstfd16 and Right Jstfd24 */ +#define INVERT_B (1 << 5) +#define ATAPI3 (1 << 3) +#define ATAPI2 (1 << 2) +#define ATAPI1 (1 << 1) +#define ATAPI0 (1 << 0) +#define MUTEAB 0x00 +#define MUTEA_RIGHTB 0x01 +#define MUTEA_LEFTB 0x02 +#define MUTEA_SUMLRDIV2B 0x03 +#define RIGHTA_MUTEB 0x04 +#define RIGHTA_RIGHTB 0x05 +#define RIGHTA_LEFTB 0x06 +#define RIGHTA_SUMLRDIV2B 0x07 +#define LEFTA_MUTEB 0x08 +#define LEFTA_RIGHTB 0x09 /* Default */ +#define LEFTA_LEFTB 0x0A +#define LEFTA_SUMLRDIV2B 0x0B +#define SUMLRDIV2A_MUTEB 0x0C +#define SUMLRDIV2A_RIGHTB 0x0D +#define SUMLRDIV2A_LEFTB 0x0E +#define SUMLRDIV2_AB 0x0F +#define CHMIX_MASK 0x0F + +/* CS4349_MUTE */ +#define AUTOMUTE (1 << 7) +#define MUTEC_AB (1 << 5) +#define MUTE_A (1 << 4) +#define MUTE_B (1 << 3) +#define MUTE_AB_MASK 0x18 + +/* CS4349_RMPFLT (Ramp and Filter Control) */ +#define SCZ1 (1 << 7) +#define SCZ0 (1 << 6) +#define RMP_UP (1 << 5) +#define RMP_DN (1 << 4) +#define FILT_SEL (1 << 2) +#define IMMDT_CHNG 0x31 +#define ZEROCRSS 0x71 +#define SOFT_RMP 0xB1 +#define SFTRMP_ZEROCRSS 0xF1 +#define SR_ZC_MASK 0xC0 + +/* CS4349_MISC */ +#define PWR_DWN (1 << 7) +#define FREEZE (1 << 5) +#define POPG_EN (1 << 4) + +#endif /* __CS4349_H__ */ -- 2.4.5

2 1

[alsa-devel] [PATCH 1/2] V4 regmap: Use reg_sequence for multi_reg_write / register_patch
by Nariman Poushin 16 Jul '15

16 Jul '15

Add support for writing sequences of registers / patches with specified delays (in microseconds). Logically separates the functionality using sequences of register writes from the functions that take register defaults, as adding a delay field on the reg_defaults can increase memory usage substantially. This also updates all the clients of multi_reg_write/register_patch. Signed-off-by: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> --- drivers/base/regmap/internal.h | 2 +- drivers/base/regmap/regmap.c | 22 +++++++++++----------- drivers/gpu/drm/i2c/adv7511.c | 2 +- drivers/input/misc/drv260x.c | 6 +++--- drivers/input/misc/drv2665.c | 2 +- drivers/input/misc/drv2667.c | 4 ++-- drivers/mfd/arizona-core.c | 2 +- drivers/mfd/twl6040.c | 2 +- drivers/mfd/wm5102-tables.c | 6 +++--- drivers/mfd/wm5110-tables.c | 6 +++--- drivers/mfd/wm8994-core.c | 8 ++++---- drivers/mfd/wm8997-tables.c | 2 +- include/linux/regmap.h | 21 ++++++++++++++++++--- sound/soc/codecs/arizona.c | 2 +- sound/soc/codecs/cs35l32.c | 2 +- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/da7210.c | 4 ++-- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/rt5645.c | 4 ++-- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5670.c | 2 +- sound/soc/codecs/rt5677.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- 27 files changed, 66 insertions(+), 51 deletions(-) diff --git a/drivers/base/regmap/internal.h b/drivers/base/regmap/internal.h index b2b2849..873ddf9 100644 --- a/drivers/base/regmap/internal.h +++ b/drivers/base/regmap/internal.h @@ -136,7 +136,7 @@ struct regmap { /* if set, the HW registers are known to match map->reg_defaults */ bool no_sync_defaults; - struct reg_default *patch; + struct reg_sequence *patch; int patch_regs; /* if set, converts bulk rw to single rw */ diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index dd63bcb..0a849ee 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -1755,7 +1755,7 @@ EXPORT_SYMBOL_GPL(regmap_bulk_write); * relative. The page register has been written if that was neccessary. */ static int _regmap_raw_multi_reg_write(struct regmap *map, - const struct reg_default *regs, + const struct reg_sequence *regs, size_t num_regs) { int ret; @@ -1812,12 +1812,12 @@ static unsigned int _regmap_register_page(struct regmap *map, } static int _regmap_range_multi_paged_reg_write(struct regmap *map, - struct reg_default *regs, + struct reg_sequence *regs, size_t num_regs) { int ret; int i, n; - struct reg_default *base; + struct reg_sequence *base; unsigned int this_page = 0; /* * the set of registers are not neccessarily in order, but @@ -1855,7 +1855,7 @@ static int _regmap_range_multi_paged_reg_write(struct regmap *map, } static int _regmap_multi_reg_write(struct regmap *map, - const struct reg_default *regs, + const struct reg_sequence *regs, size_t num_regs) { int i; @@ -1907,8 +1907,8 @@ static int _regmap_multi_reg_write(struct regmap *map, struct regmap_range_node *range; range = _regmap_range_lookup(map, reg); if (range) { - size_t len = sizeof(struct reg_default)*num_regs; - struct reg_default *base = kmemdup(regs, len, + size_t len = sizeof(struct reg_sequence)*num_regs; + struct reg_sequence *base = kmemdup(regs, len, GFP_KERNEL); if (!base) return -ENOMEM; @@ -1941,7 +1941,7 @@ static int _regmap_multi_reg_write(struct regmap *map, * A value of zero will be returned on success, a negative errno will be * returned in error cases. */ -int regmap_multi_reg_write(struct regmap *map, const struct reg_default *regs, +int regmap_multi_reg_write(struct regmap *map, const struct reg_sequence *regs, int num_regs) { int ret; @@ -1974,7 +1974,7 @@ EXPORT_SYMBOL_GPL(regmap_multi_reg_write); * be returned in error cases. */ int regmap_multi_reg_write_bypassed(struct regmap *map, - const struct reg_default *regs, + const struct reg_sequence *regs, int num_regs) { int ret; @@ -2587,10 +2587,10 @@ EXPORT_SYMBOL_GPL(regmap_async_complete); * The caller must ensure that this function cannot be called * concurrently with either itself or regcache_sync(). */ -int regmap_register_patch(struct regmap *map, const struct reg_default *regs, +int regmap_register_patch(struct regmap *map, const struct reg_sequence *regs, int num_regs) { - struct reg_default *p; + struct reg_sequence *p; int ret; bool bypass; @@ -2599,7 +2599,7 @@ int regmap_register_patch(struct regmap *map, const struct reg_default *regs, return 0; p = krealloc(map->patch, - sizeof(struct reg_default) * (map->patch_regs + num_regs), + sizeof(struct reg_sequence) * (map->patch_regs + num_regs), GFP_KERNEL); if (p) { memcpy(p + map->patch_regs, regs, num_regs * sizeof(*regs)); diff --git a/drivers/gpu/drm/i2c/adv7511.c b/drivers/gpu/drm/i2c/adv7511.c index 2aaa3c8..00416f2 100644 --- a/drivers/gpu/drm/i2c/adv7511.c +++ b/drivers/gpu/drm/i2c/adv7511.c @@ -54,7 +54,7 @@ static struct adv7511 *encoder_to_adv7511(struct drm_encoder *encoder) } /* ADI recommended values for proper operation. */ -static const struct reg_default adv7511_fixed_registers[] = { +static const struct reg_sequence adv7511_fixed_registers[] = { { 0x98, 0x03 }, { 0x9a, 0xe0 }, { 0x9c, 0x30 }, diff --git a/drivers/input/misc/drv260x.c b/drivers/input/misc/drv260x.c index e5d60ec..f5c9cf2 100644 --- a/drivers/input/misc/drv260x.c +++ b/drivers/input/misc/drv260x.c @@ -313,14 +313,14 @@ static void drv260x_close(struct input_dev *input) gpiod_set_value(haptics->enable_gpio, 0); } -static const struct reg_default drv260x_lra_cal_regs[] = { +static const struct reg_sequence drv260x_lra_cal_regs[] = { { DRV260X_MODE, DRV260X_AUTO_CAL }, { DRV260X_CTRL3, DRV260X_NG_THRESH_2 }, { DRV260X_FEEDBACK_CTRL, DRV260X_FB_REG_LRA_MODE | DRV260X_BRAKE_FACTOR_4X | DRV260X_LOOP_GAIN_HIGH }, }; -static const struct reg_default drv260x_lra_init_regs[] = { +static const struct reg_sequence drv260x_lra_init_regs[] = { { DRV260X_MODE, DRV260X_RT_PLAYBACK }, { DRV260X_A_TO_V_CTRL, DRV260X_AUDIO_HAPTICS_PEAK_20MS | DRV260X_AUDIO_HAPTICS_FILTER_125HZ }, @@ -337,7 +337,7 @@ static const struct reg_default drv260x_lra_init_regs[] = { { DRV260X_CTRL4, DRV260X_AUTOCAL_TIME_500MS }, }; -static const struct reg_default drv260x_erm_cal_regs[] = { +static const struct reg_sequence drv260x_erm_cal_regs[] = { { DRV260X_MODE, DRV260X_AUTO_CAL }, { DRV260X_A_TO_V_MIN_INPUT, DRV260X_AUDIO_HAPTICS_MIN_IN_VOLT }, { DRV260X_A_TO_V_MAX_INPUT, DRV260X_AUDIO_HAPTICS_MAX_IN_VOLT }, diff --git a/drivers/input/misc/drv2665.c b/drivers/input/misc/drv2665.c index 0afaa33..924456e 100644 --- a/drivers/input/misc/drv2665.c +++ b/drivers/input/misc/drv2665.c @@ -132,7 +132,7 @@ static void drv2665_close(struct input_dev *input) "Failed to enter standby mode: %d\n", error); } -static const struct reg_default drv2665_init_regs[] = { +static const struct reg_sequence drv2665_init_regs[] = { { DRV2665_CTRL_2, 0 | DRV2665_10_MS_IDLE_TOUT }, { DRV2665_CTRL_1, DRV2665_25_VPP_GAIN }, }; diff --git a/drivers/input/misc/drv2667.c b/drivers/input/misc/drv2667.c index fc0fddf..047136a 100644 --- a/drivers/input/misc/drv2667.c +++ b/drivers/input/misc/drv2667.c @@ -262,14 +262,14 @@ static void drv2667_close(struct input_dev *input) "Failed to enter standby mode: %d\n", error); } -static const struct reg_default drv2667_init_regs[] = { +static const struct reg_sequence drv2667_init_regs[] = { { DRV2667_CTRL_2, 0 }, { DRV2667_CTRL_1, DRV2667_25_VPP_GAIN }, { DRV2667_WV_SEQ_0, 1 }, { DRV2667_WV_SEQ_1, 0 } }; -static const struct reg_default drv2667_page1_init[] = { +static const struct reg_sequence drv2667_page1_init[] = { { DRV2667_RAM_HDR_SZ, 0x05 }, { DRV2667_RAM_START_HI, 0x80 }, { DRV2667_RAM_START_LO, 0x06 }, diff --git a/drivers/mfd/arizona-core.c b/drivers/mfd/arizona-core.c index bebf58a..66d50be 100644 --- a/drivers/mfd/arizona-core.c +++ b/drivers/mfd/arizona-core.c @@ -392,7 +392,7 @@ err: * Register patch to some of the CODECs internal write sequences * to ensure a clean exit from the low power sleep state. */ -static const struct reg_default wm5110_sleep_patch[] = { +static const struct reg_sequence wm5110_sleep_patch[] = { { 0x337A, 0xC100 }, { 0x337B, 0x0041 }, { 0x3300, 0xA210 }, diff --git a/drivers/mfd/twl6040.c b/drivers/mfd/twl6040.c index c5265c1..583dc33 100644 --- a/drivers/mfd/twl6040.c +++ b/drivers/mfd/twl6040.c @@ -86,7 +86,7 @@ static const struct reg_default twl6040_defaults[] = { { 0x2E, 0x00 }, /* REG_STATUS (ro) */ }; -static struct reg_default twl6040_patch[] = { +static struct reg_sequence twl6040_patch[] = { /* * Select I2C bus access to dual access registers * Interrupt register is cleared on read diff --git a/drivers/mfd/wm5102-tables.c b/drivers/mfd/wm5102-tables.c index aeae6ec..423fb37 100644 --- a/drivers/mfd/wm5102-tables.c +++ b/drivers/mfd/wm5102-tables.c @@ -21,7 +21,7 @@ #define WM5102_NUM_AOD_ISR 2 #define WM5102_NUM_ISR 5 -static const struct reg_default wm5102_reva_patch[] = { +static const struct reg_sequence wm5102_reva_patch[] = { { 0x80, 0x0003 }, { 0x221, 0x0090 }, { 0x211, 0x0014 }, @@ -57,7 +57,7 @@ static const struct reg_default wm5102_reva_patch[] = { { 0x80, 0x0000 }, }; -static const struct reg_default wm5102_revb_patch[] = { +static const struct reg_sequence wm5102_revb_patch[] = { { 0x19, 0x0001 }, { 0x80, 0x0003 }, { 0x081, 0xE022 }, @@ -80,7 +80,7 @@ static const struct reg_default wm5102_revb_patch[] = { /* We use a function so we can use ARRAY_SIZE() */ int wm5102_patch(struct arizona *arizona) { - const struct reg_default *wm5102_patch; + const struct reg_sequence *wm5102_patch; int patch_size; switch (arizona->rev) { diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index 12cad94..26ce14f 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -21,7 +21,7 @@ #define WM5110_NUM_AOD_ISR 2 #define WM5110_NUM_ISR 5 -static const struct reg_default wm5110_reva_patch[] = { +static const struct reg_sequence wm5110_reva_patch[] = { { 0x80, 0x3 }, { 0x44, 0x20 }, { 0x45, 0x40 }, @@ -134,7 +134,7 @@ static const struct reg_default wm5110_reva_patch[] = { { 0x209, 0x002A }, }; -static const struct reg_default wm5110_revb_patch[] = { +static const struct reg_sequence wm5110_revb_patch[] = { { 0x80, 0x3 }, { 0x36e, 0x0210 }, { 0x370, 0x0210 }, @@ -224,7 +224,7 @@ static const struct reg_default wm5110_revb_patch[] = { { 0x80, 0x0 }, }; -static const struct reg_default wm5110_revd_patch[] = { +static const struct reg_sequence wm5110_revd_patch[] = { { 0x80, 0x3 }, { 0x80, 0x3 }, { 0x393, 0x27 }, diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index 53ae5af..0f4169a 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -243,21 +243,21 @@ static int wm8994_ldo_in_use(struct wm8994_pdata *pdata, int ldo) } #endif -static const struct reg_default wm8994_revc_patch[] = { +static const struct reg_sequence wm8994_revc_patch[] = { { 0x102, 0x3 }, { 0x56, 0x3 }, { 0x817, 0x0 }, { 0x102, 0x0 }, }; -static const struct reg_default wm8958_reva_patch[] = { +static const struct reg_sequence wm8958_reva_patch[] = { { 0x102, 0x3 }, { 0xcb, 0x81 }, { 0x817, 0x0 }, { 0x102, 0x0 }, }; -static const struct reg_default wm1811_reva_patch[] = { +static const struct reg_sequence wm1811_reva_patch[] = { { 0x102, 0x3 }, { 0x56, 0xc07 }, { 0x5d, 0x7e }, @@ -326,7 +326,7 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) { struct wm8994_pdata *pdata; struct regmap_config *regmap_config; - const struct reg_default *regmap_patch = NULL; + const struct reg_sequence *regmap_patch = NULL; const char *devname; int ret, i, patch_regs = 0; int pulls = 0; diff --git a/drivers/mfd/wm8997-tables.c b/drivers/mfd/wm8997-tables.c index c0c25d75..cab2c68 100644 --- a/drivers/mfd/wm8997-tables.c +++ b/drivers/mfd/wm8997-tables.c @@ -17,7 +17,7 @@ #include "arizona.h" -static const struct reg_default wm8997_reva_patch[] = { +static const struct reg_sequence wm8997_reva_patch[] = { { 0x80, 0x0003 }, { 0x214, 0x0008 }, { 0x458, 0x0000 }, diff --git a/include/linux/regmap.h b/include/linux/regmap.h index 519c962..ee5106e 100644 --- a/include/linux/regmap.h +++ b/include/linux/regmap.h @@ -50,6 +50,21 @@ struct reg_default { unsigned int def; }; +/** + * Register / Value pairs for sequences of writes, incorporating an optional + * delay in microseconds. + * + * @reg: Register address. + * @def: Register default value. + * @delay_us: Delay in microseconds + */ + +struct reg_sequence { + unsigned reg; + unsigned def; + unsigned delay_us; +}; + #ifdef CONFIG_REGMAP enum regmap_endian { @@ -410,10 +425,10 @@ int regmap_raw_write(struct regmap *map, unsigned int reg, const void *val, size_t val_len); int regmap_bulk_write(struct regmap *map, unsigned int reg, const void *val, size_t val_count); -int regmap_multi_reg_write(struct regmap *map, const struct reg_default *regs, +int regmap_multi_reg_write(struct regmap *map, const struct reg_sequence *regs, int num_regs); int regmap_multi_reg_write_bypassed(struct regmap *map, - const struct reg_default *regs, + const struct reg_sequence *regs, int num_regs); int regmap_raw_write_async(struct regmap *map, unsigned int reg, const void *val, size_t val_len); @@ -452,7 +467,7 @@ void regcache_mark_dirty(struct regmap *map); bool regmap_check_range_table(struct regmap *map, unsigned int reg, const struct regmap_access_table *table); -int regmap_register_patch(struct regmap *map, const struct reg_default *regs, +int regmap_register_patch(struct regmap *map, const struct reg_sequence *regs, int num_regs); int regmap_parse_val(struct regmap *map, const void *buf, unsigned int *val); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 802e05e..5edd33f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1366,7 +1366,7 @@ static void arizona_wm5102_set_dac_comp(struct snd_soc_codec *codec, { struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); struct arizona *arizona = priv->arizona; - struct reg_default dac_comp[] = { + struct reg_sequence dac_comp[] = { { 0x80, 0x3 }, { ARIZONA_DAC_COMP_1, 0 }, { ARIZONA_DAC_COMP_2, 0 }, diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 8f40025..2813a1b 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -276,7 +276,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_cs35l32 = { }; /* Current and threshold powerup sequence Pg37 in datasheet */ -static const struct reg_default cs35l32_monitor_patch[] = { +static const struct reg_sequence cs35l32_monitor_patch[] = { { 0x00, 0x99 }, { 0x48, 0x17 }, diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 4de52c9..8b2d059 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1118,7 +1118,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_cs42l52 = { }; /* Current and threshold powerup sequence Pg37 */ -static const struct reg_default cs42l52_threshold_patch[] = { +static const struct reg_sequence cs42l52_threshold_patch[] = { { 0x00, 0x99 }, { 0x3E, 0xBA }, diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 21810e5..bf0fb3d 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1182,7 +1182,7 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { #if IS_ENABLED(CONFIG_I2C) -static struct reg_default da7210_regmap_i2c_patch[] = { +static struct reg_sequence da7210_regmap_i2c_patch[] = { /* System controller master disable */ { DA7210_STARTUP1, 0x00 }, @@ -1269,7 +1269,7 @@ static struct i2c_driver da7210_i2c_driver = { #if defined(CONFIG_SPI_MASTER) -static struct reg_default da7210_regmap_spi_patch[] = { +static struct reg_sequence da7210_regmap_spi_patch[] = { /* Dummy read to give two pulses over nCS for SPI */ { DA7210_AUX2, 0x00 }, { DA7210_AUX2, 0x00 }, diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 9bc78e5..1ed1f88 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -51,7 +51,7 @@ static const struct regmap_range_cfg rt5640_ranges[] = { .window_len = 0x1, }, }; -static const struct reg_default init_list[] = { +static const struct reg_sequence init_list[] = { {RT5640_PR_BASE + 0x3d, 0x3600}, {RT5640_PR_BASE + 0x12, 0x0aa8}, {RT5640_PR_BASE + 0x14, 0x0aaa}, diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9ce311e..c0f4be4 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -54,7 +54,7 @@ static const struct regmap_range_cfg rt5645_ranges[] = { }, }; -static const struct reg_default init_list[] = { +static const struct reg_sequence init_list[] = { {RT5645_PR_BASE + 0x3d, 0x3600}, {RT5645_PR_BASE + 0x1c, 0xfd20}, {RT5645_PR_BASE + 0x20, 0x611f}, @@ -63,7 +63,7 @@ static const struct reg_default init_list[] = { }; #define RT5645_INIT_REG_LEN ARRAY_SIZE(init_list) -static const struct reg_default rt5650_init_list[] = { +static const struct reg_sequence rt5650_init_list[] = { {0xf6, 0x0100}, }; diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index a3506e1..db9b866 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -46,7 +46,7 @@ static const struct regmap_range_cfg rt5651_ranges[] = { .window_len = 0x1, }, }; -static struct reg_default init_list[] = { +static struct reg_sequence init_list[] = { {RT5651_PR_BASE + 0x3d, 0x3e00}, }; diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index a9123d4..462a91f 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -51,7 +51,7 @@ static const struct regmap_range_cfg rt5670_ranges[] = { .window_len = 0x1, }, }; -static const struct reg_default init_list[] = { +static const struct reg_sequence init_list[] = { { RT5670_PR_BASE + 0x14, 0x9a8a }, { RT5670_PR_BASE + 0x38, 0x3ba1 }, { RT5670_PR_BASE + 0x3d, 0x3640 }, diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 31d969a..b897752 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -54,7 +54,7 @@ static const struct regmap_range_cfg rt5677_ranges[] = { }, }; -static const struct reg_default init_list[] = { +static const struct reg_sequence init_list[] = { {RT5677_ASRC_12, 0x0018}, {RT5677_PR_BASE + 0x3d, 0x364d}, {RT5677_PR_BASE + 0x17, 0x4fc0}, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index a7cf19b..83ae1eb 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1668,7 +1668,7 @@ static const struct i2c_device_id aic3x_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); -static const struct reg_default aic3007_class_d[] = { +static const struct reg_sequence aic3007_class_d[] = { /* Class-D speaker driver init; datasheet p. 46 */ { AIC3X_PAGE_SELECT, 0x0D }, { 0xD, 0x0D }, diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index c830832..6c60792 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -897,7 +897,7 @@ static bool wm2200_readable_register(struct device *dev, unsigned int reg) } } -static const struct reg_default wm2200_reva_patch[] = { +static const struct reg_sequence wm2200_reva_patch[] = { { 0x07, 0x0003 }, { 0x102, 0x0200 }, { 0x203, 0x0084 }, diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 4c10cd8..26d79bb 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1247,7 +1247,7 @@ static const struct snd_soc_dapm_route wm5100_dapm_routes[] = { { "PWM2", NULL, "PWM2 Driver" }, }; -static const struct reg_default wm5100_reva_patches[] = { +static const struct reg_sequence wm5100_reva_patches[] = { { WM5100_AUDIO_IF_1_10, 0 }, { WM5100_AUDIO_IF_1_11, 1 }, { WM5100_AUDIO_IF_1_12, 2 }, diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index c5748fd..05492e8 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3495,7 +3495,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8962 = { }; /* Improve power consumption for IN4 DC measurement mode */ -static const struct reg_default wm8962_dc_measure[] = { +static const struct reg_sequence wm8962_dc_measure[] = { { 0xfd, 0x1 }, { 0xcc, 0x40 }, { 0xfd, 0 }, diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 8a8db86..52ec64d 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1595,7 +1595,7 @@ static int wm8993_resume(struct snd_soc_codec *codec) #endif /* Tune DC servo configuration */ -static struct reg_default wm8993_regmap_patch[] = { +static struct reg_sequence wm8993_regmap_patch[] = { { 0x44, 3 }, { 0x56, 3 }, { 0x44, 0 }, -- 2.1.4

3 5

[alsa-devel] [asoc:fix/rt5645 1/1] sound/soc/codecs/rt5645.c:2947:3: warning: 'return' with no value, in function returning non-void
by kbuild test robot 16 Jul '15

16 Jul '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git fix/rt5645 head: 3c2234569cfff83718792f810a9bb855b78c2473 commit: 3c2234569cfff83718792f810a9bb855b78c2473 [1/1] ASoC: rt5645: Check if codec is initialized in workqueue handler config: i386-randconfig-x009-201528 (attached as .config) reproduce: git checkout 3c2234569cfff83718792f810a9bb855b78c2473 # save the attached .config to linux build tree make ARCH=i386 All warnings (new ones prefixed by >>): sound/soc/codecs/rt5645.c: In function 'rt5645_irq_detection': >> sound/soc/codecs/rt5645.c:2947:3: warning: 'return' with no value, in function returning non-void [-Wreturn-type] return; ^ vim +/return +2947 sound/soc/codecs/rt5645.c 2931 { 2932 int btn_type, val; 2933 2934 val = snd_soc_read(codec, RT5650_4BTN_IL_CMD1); 2935 pr_debug("val=0x%x\n", val); 2936 btn_type = val & 0xfff0; 2937 snd_soc_write(codec, RT5650_4BTN_IL_CMD1, val); 2938 2939 return btn_type; 2940 } 2941 2942 static int rt5645_irq_detection(struct rt5645_priv *rt5645) 2943 { 2944 int val, btn_type, gpio_state = 0, report = 0; 2945 2946 if (!rt5645->codec) > 2947 return; 2948 2949 switch (rt5645->pdata.jd_mode) { 2950 case 0: /* Not using rt5645 JD */ 2951 if (rt5645->gpiod_hp_det) { 2952 gpio_state = gpiod_get_value(rt5645->gpiod_hp_det); 2953 dev_dbg(rt5645->codec->dev, "gpio_state = %d\n", 2954 gpio_state); 2955 report = rt5645_jack_detect(rt5645->codec, gpio_state); --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

3 2

[alsa-devel] [PATCH] ALSA: hda/realtek: Enable headphone amp on HP Folio 9480m
by Keith Packard 16 Jul '15

16 Jul '15

This laptop needs GPIO4 pulled high to enable the headphone amplifier. I modelled the patch on the existing GPIO4 code which pulls the line low for the same purpose. Signed-off-by: Keith Packard <keithp(a)keithp.com> --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d01045..9060f1f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3436,6 +3436,23 @@ static void alc280_fixup_hp_gpio4(struct hda_codec *codec, } } +static void alc280_fixup_hp_gpio4_hp_amp(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + /* Pull GPIO4 high to enable headphone amp */ + static const struct hda_verb gpio_init[] = { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x10 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x10 }, + { 0x01, AC_VERB_SET_GPIO_DATA, 0x10 }, + {} + }; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + snd_hda_add_verbs(codec, gpio_init); + } +} + static void gpio2_mic_hotkey_event(struct hda_codec *codec, struct hda_jack_callback *event) { @@ -4512,6 +4529,7 @@ enum { ALC286_FIXUP_HP_GPIO_LED, ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, ALC280_FIXUP_HP_DOCK_PINS, + ALC280_FIXUP_HP_GPIO4_HP_AMP, ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, ALC288_FIXUP_DELL_XPS_13_GPIO6, @@ -5012,6 +5030,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC280_FIXUP_HP_GPIO4 }, + [ALC280_FIXUP_HP_GPIO4_HP_AMP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc280_fixup_hp_gpio4_hp_amp, + }, [ALC288_FIXUP_DELL_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_dell_alc288, @@ -5103,6 +5125,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x22b7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22bf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22cf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22db, "HP", ALC280_FIXUP_HP_GPIO4_HP_AMP), SND_PCI_QUIRK(0x103c, 0x22dc, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x22fb, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), /* ALC290 */ -- 2.1.4

2 7

[alsa-devel] [asoc:topic/rt5645 7/7] sound/soc/codecs/rt5645.c:2766:22: warning: unused variable 'rt5645'
by kbuild test robot 16 Jul '15

16 Jul '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rt5645 head: a4e3c5fa955ce0934774f3f3684377dbf2c20eca commit: a4e3c5fa955ce0934774f3f3684377dbf2c20eca [7/7] ASoC: rt5645: Simplify rt5645_enable_push_button_irq config: i386-randconfig-x009-201528 (attached as .config) reproduce: git checkout a4e3c5fa955ce0934774f3f3684377dbf2c20eca # save the attached .config to linux build tree make ARCH=i386 All warnings (new ones prefixed by >>): sound/soc/codecs/rt5645.c: In function 'rt5645_enable_push_button_irq': >> sound/soc/codecs/rt5645.c:2766:22: warning: unused variable 'rt5645' [-Wunused-variable] struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); ^ vim +/rt5645 +2766 sound/soc/codecs/rt5645.c d12d6c4e John Lin 2015-05-12 2750 regmap_write(rt5645->regmap, RT5645_DEPOP_M2, 0x1140); d12d6c4e John Lin 2015-05-12 2751 regmap_write(rt5645->regmap, RT5645_DEPOP_M1, 0x0000); d12d6c4e John Lin 2015-05-12 2752 regmap_write(rt5645->regmap, RT5645_GEN_CTRL2, 0x4020); d12d6c4e John Lin 2015-05-12 2753 regmap_write(rt5645->regmap, RT5645_PWR_ANLG2, 0x0006); d12d6c4e John Lin 2015-05-12 2754 regmap_write(rt5645->regmap, RT5645_PWR_DIG2, 0x0000); d12d6c4e John Lin 2015-05-12 2755 msleep(350); d12d6c4e John Lin 2015-05-12 2756 d12d6c4e John Lin 2015-05-12 2757 regcache_cache_bypass(rt5645->regmap, false); d12d6c4e John Lin 2015-05-12 2758 d12d6c4e John Lin 2015-05-12 2759 return ret; d12d6c4e John Lin 2015-05-12 2760 } d12d6c4e John Lin 2015-05-12 2761 6e747d53 Bard Liao 2015-04-28 2762 static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, 6e747d53 Bard Liao 2015-04-28 2763 bool enable) f3fa1bbd Oder Chiou 2014-09-19 2764 { e2ada818 Lars-Peter Clausen 2015-07-06 2765 struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); f3fa1bbd Oder Chiou 2014-09-19 @2766 struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); f3fa1bbd Oder Chiou 2014-09-19 2767 6e747d53 Bard Liao 2015-04-28 2768 if (enable) { a4e3c5fa Nicolas Boichat 2015-07-16 2769 snd_soc_dapm_force_enable_pin(dapm, "ADC L power"); a4e3c5fa Nicolas Boichat 2015-07-16 2770 snd_soc_dapm_force_enable_pin(dapm, "ADC R power"); a4e3c5fa Nicolas Boichat 2015-07-16 2771 snd_soc_dapm_sync(dapm); 22f5d9f8 Nicolas Boichat 2015-06-10 2772 6e747d53 Bard Liao 2015-04-28 2773 snd_soc_update_bits(codec, 6e747d53 Bard Liao 2015-04-28 2774 RT5645_INT_IRQ_ST, 0x8, 0x8); :::::: The code at line 2766 was first introduced by commit :::::: f3fa1bbd836a7d6efb2abd506ed8e24096f39062 ASoC: rt5645: Add headset detect function :::::: TO: Oder Chiou <oder_chiou(a)realtek.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH v3] ASoC: cs4349: Add support for Cirrus Logic CS4349
by timothyc.howe＠gmail.com 16 Jul '15

16 Jul '15

From: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- Documentation/devicetree/bindings/sound/cs4349.txt | 19 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4349.c | 401 +++++++++++++++++++++ sound/soc/codecs/cs4349.h | 146 ++++++++ 5 files changed, 574 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs4349.txt create mode 100644 sound/soc/codecs/cs4349.c create mode 100644 sound/soc/codecs/cs4349.h diff --git a/Documentation/devicetree/bindings/sound/cs4349.txt b/Documentation/devicetree/bindings/sound/cs4349.txt new file mode 100644 index 0000000..54c117b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4349.txt @@ -0,0 +1,19 @@ +CS4349 audio CODEC + +Required properties: + + - compatible : "cirrus,cs4349" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + +Example: + +codec: cs4349@48 { + compatible = "cirrus,cs4349"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7223c7f..4a78606 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -54,6 +54,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_I2C if I2C select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C + select SND_SOC_CS4349 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -413,6 +414,11 @@ config SND_SOC_CS42XX8_I2C select SND_SOC_CS42XX8 select REGMAP_I2C +# Cirrus Logic CS4349 HiFi DAC +config SND_SOC_CS4349 + tristate "Cirrus Logic CS4349 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9f7136c..36a9f37 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -46,6 +46,7 @@ snd-soc-cs4271-i2c-objs := cs4271-i2c.o snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o +snd-soc-cs4349-objs := cs4349.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -237,6 +238,7 @@ obj-$(CONFIG_SND_SOC_CS4271_I2C) += snd-soc-cs4271-i2c.o obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o +obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c new file mode 100644 index 0000000..fb8d1e1 --- /dev/null +++ b/sound/soc/codecs/cs4349.c @@ -0,0 +1,401 @@ +/* + * cs4349.c -- CS4349 ALSA Soc Audio driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Authors: Tim Howe <Tim.Howe(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs4349.h" + + +static const struct reg_default cs4349_reg_defaults[] = { + { 2, 0x00 }, /* r02 - Mode Control */ + { 3, 0x09 }, /* r03 - Volume, Mixing and Inversion Control */ + { 4, 0x81 }, /* r04 - Mute Control */ + { 5, 0x00 }, /* r05 - Channel A Volume Control */ + { 6, 0x00 }, /* r06 - Channel B Volume Control */ + { 7, 0xB1 }, /* r07 - Ramp and Filter Control */ + { 8, 0x1C }, /* r08 - Misc. Control */ +}; + +/* Private data for the CS4349 */ +struct cs4349_private { + struct regmap *regmap; + struct cs4349_platform_data pdata; + struct gpio_desc *reset_gpio; + unsigned int mode; + int rate; +}; + +static bool cs4349_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS4349_CHIPID: + case CS4349_MODE: + case CS4349_VMI: + case CS4349_MUTE: + case CS4349_VOLA: + case CS4349_VOLB: + case CS4349_RMPFLT: + case CS4349_MISC: + return true; + default: + return false; + } +} + +static int cs4349_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + unsigned int fmt; + + fmt = format & SND_SOC_DAIFMT_FORMAT_MASK; + + switch (fmt) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + cs4349->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs4349_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + int mode, fmt, ret; + + mode = snd_soc_read(codec, CS4349_MODE); + cs4349->rate = params_rate(params); + + switch (cs4349->mode) { + case SND_SOC_DAIFMT_I2S: + mode |= MODE_FORMAT(DIF_I2S); + break; + case SND_SOC_DAIFMT_LEFT_J: + mode |= MODE_FORMAT(DIF_LEFT_JST); + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_width(params)) { + case 16: + fmt = DIF_RGHT_JST16; + break; + case 24: + fmt = DIF_RGHT_JST24; + break; + default: + return -EINVAL; + } + mode |= MODE_FORMAT(fmt); + break; + default: + return -EINVAL; + } + + ret = snd_soc_write(codec, CS4349_MODE, mode); + if (ret < 0) + return ret; + + return 0; +} + +static int cs4349_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int reg; + + reg = 0; + if (mute) + reg = MUTE_AB_MASK; + + return snd_soc_update_bits(codec, CS4349_MUTE, MUTE_AB_MASK, reg); +} + +static DECLARE_TLV_DB_SCALE(dig_tlv, -12750, 50, 0); + +static const char * const chan_mix_texts[] = { + "Mute", "MuteA", "MuteA SwapB", "MuteA MonoB", "SwapA MuteB", + "BothR", "Swap", "SwapA MonoB", "MuteB", "Normal", "BothL", + "MonoB", "MonoA MuteB", "MonoA", "MonoA SwapB", "Mono", + /*Normal == Channel A = Left, Channel B = Right*/ +}; + +static const char * const fm_texts[] = { + "Auto", "Single", "Double", "Quad", +}; + +static const char * const deemph_texts[] = { + "None", "44.1k", "48k", "32k", +}; + +static const char * const softr_zeroc_texts[] = { + "Immediate", "Zero Cross", "Soft Ramp", "SR on ZC", +}; + +static int deemph_values[] = { + 0, 4, 8, 12, +}; + +static int softr_zeroc_values[] = { + 0, 64, 128, 192, +}; + +static const struct soc_enum chan_mix_enum = + SOC_ENUM_SINGLE(CS4349_VMI, 0, + ARRAY_SIZE(chan_mix_texts), + chan_mix_texts); + +static const struct soc_enum fm_mode_enum = + SOC_ENUM_SINGLE(CS4349_MODE, 0, + ARRAY_SIZE(fm_texts), + fm_texts); + +static SOC_VALUE_ENUM_SINGLE_DECL(deemph_enum, CS4349_MODE, 0, DEM_MASK, + deemph_texts, deemph_values); + +static SOC_VALUE_ENUM_SINGLE_DECL(softr_zeroc_enum, CS4349_RMPFLT, 0, + SR_ZC_MASK, softr_zeroc_texts, + softr_zeroc_values); + +static const struct snd_kcontrol_new cs4349_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", + CS4349_VOLA, CS4349_VOLB, 0, 0xFF, 1, dig_tlv), + SOC_ENUM("Functional Mode", fm_mode_enum), + SOC_ENUM("De-Emphasis Control", deemph_enum), + SOC_ENUM("Soft Ramp Zero Cross Control", softr_zeroc_enum), + SOC_ENUM("Channel Mixer", chan_mix_enum), + SOC_SINGLE("VolA = VolB Switch", CS4349_VMI, 7, 1, 0), + SOC_SINGLE("InvertA Switch", CS4349_VMI, 6, 1, 0), + SOC_SINGLE("InvertB Switch", CS4349_VMI, 5, 1, 0), + SOC_SINGLE("Auto-Mute Switch", CS4349_MUTE, 7, 1, 0), + SOC_SINGLE("MUTEC A = B Switch", CS4349_MUTE, 5, 1, 0), + SOC_SINGLE("Soft Ramp Up Switch", CS4349_RMPFLT, 5, 1, 0), + SOC_SINGLE("Soft Ramp Down Switch", CS4349_RMPFLT, 4, 1, 0), + SOC_SINGLE("Slow Roll Off Filter Switch", CS4349_RMPFLT, 2, 1, 0), + SOC_SINGLE("Freeze Switch", CS4349_MISC, 5, 1, 0), + SOC_SINGLE("Popguard Switch", CS4349_MISC, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget cs4349_dapm_widgets[] = { + SND_SOC_DAPM_DAC("HiFi DAC", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_OUTPUT("OutputA"), + SND_SOC_DAPM_OUTPUT("OutputB"), +}; + +static const struct snd_soc_dapm_route cs4349_routes[] = { + {"DAC Playback", NULL, "OutputA"}, + {"DAC Playback", NULL, "OutputB"}, + + {"OutputA", NULL, "HiFi DAC"}, + {"OutputB", NULL, "HiFi DAC"}, +}; + +#define CS4349_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) + +#define CS4349_PCM_RATES SNDRV_PCM_RATE_8000_192000 + +static const struct snd_soc_dai_ops cs4349_dai_ops = { + .hw_params = cs4349_pcm_hw_params, + .set_fmt = cs4349_set_dai_fmt, + .digital_mute = cs4349_digital_mute, +}; + +static struct snd_soc_dai_driver cs4349_dai = { + .name = "cs4349_hifi", + .playback = { + .stream_name = "DAC Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS4349_PCM_RATES, + .formats = CS4349_PCM_FORMATS, + }, + .ops = &cs4349_dai_ops, + .symmetric_rates = 1, +}; + +static struct snd_soc_codec_driver soc_codec_dev_cs4349 = { + .controls = cs4349_snd_controls, + .num_controls = ARRAY_SIZE(cs4349_snd_controls), + + .dapm_widgets = cs4349_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4349_dapm_widgets), + .dapm_routes = cs4349_routes, + .num_dapm_routes = ARRAY_SIZE(cs4349_routes), +}; + +static struct regmap_config cs4349_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS4349_NUMREGS, + .reg_defaults = cs4349_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4349_reg_defaults), + .readable_reg = cs4349_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs4349_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs4349_private *cs4349; + struct cs4349_platform_data *pdata = dev_get_platdata(&client->dev); + int ret = 0; + + cs4349 = devm_kzalloc(&client->dev, sizeof(*cs4349), GFP_KERNEL); + if (!cs4349) + return -ENOMEM; + + cs4349->regmap = devm_regmap_init_i2c(client, &cs4349_regmap); + if (IS_ERR(cs4349->regmap)) { + ret = PTR_ERR(cs4349->regmap); + dev_err(&client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + if (pdata) + cs4349->pdata = *pdata; + + /* Reset the Device */ + cs4349->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs4349->reset_gpio)) + return PTR_ERR(cs4349->reset_gpio); + + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + + i2c_set_clientdata(client, cs4349); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4349, + &cs4349_dai, 1); +} + +static int cs4349_i2c_remove(struct i2c_client *client) +{ + struct cs4349_private *cs4349 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + /* Hold down reset */ + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int cs4349_runtime_suspend(struct device *dev) +{ + struct cs4349_private *cs4349 = dev_get_drvdata(dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + int ret; + + ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 1); + if (ret < 0) + return ret; + + regcache_sync(cs4349->regmap); + regcache_cache_only(cs4349->regmap, true); + + /* Hold down reset */ + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + + return 0; +} + +static int cs4349_runtime_resume(struct device *dev) +{ + struct cs4349_private *cs4349 = dev_get_drvdata(dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + int ret; + + ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 0); + if (ret < 0) + return ret; + + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + + regcache_cache_only(cs4349->regmap, false); + regcache_sync(cs4349->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs4349_runtime_pm = { + SET_RUNTIME_PM_OPS(cs4349_runtime_suspend, cs4349_runtime_resume, + NULL) +}; + +static const struct of_device_id cs4349_of_match[] = { + { .compatible = "cirrus,cs4349", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs4349_of_match); + +static const struct i2c_device_id cs4349_i2c_id[] = { + {"cs4349", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs4349_i2c_id); + +static struct i2c_driver cs4349_i2c_driver = { + .driver = { + .name = "cs4349", + .owner = THIS_MODULE, + .of_match_table = cs4349_of_match, + }, + .id_table = cs4349_i2c_id, + .probe = cs4349_i2c_probe, + .remove = cs4349_i2c_remove, +}; + +module_i2c_driver(cs4349_i2c_driver); + +MODULE_AUTHOR("Tim Howe <tim.howe(a)cirrus.com>"); +MODULE_DESCRIPTION("Cirrus Logic CS4349 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4349.h b/sound/soc/codecs/cs4349.h new file mode 100644 index 0000000..3884a89 --- /dev/null +++ b/sound/soc/codecs/cs4349.h @@ -0,0 +1,146 @@ +/* + * ALSA SoC CS4349 codec driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Tim Howe <Tim.Howe(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + */ + +#ifndef __CS4349_H__ +#define __CS4349_H__ + +struct cs4349_platform_data { + + /* GPIO for Reset */ + unsigned int gpio_nreset; + +}; + +/* CS4349 registers addresses */ +#define CS4349_CHIPID 0x01 /* Device and Rev ID, Read Only */ +#define CS4349_MODE 0x02 /* Mode Control */ +#define CS4349_VMI 0x03 /* Volume, Mixing, Inversion Control */ +#define CS4349_MUTE 0x04 /* Mute Control */ +#define CS4349_VOLA 0x05 /* DAC Channel A Volume Control */ +#define CS4349_VOLB 0x06 /* DAC Channel B Volume Control */ +#define CS4349_RMPFLT 0x07 /* Ramp and Filter Control */ +#define CS4349_MISC 0x08 /* Power Down,Freeze Control,Pop Stop*/ + +#define CS4349_FIRSTREG 0x01 +#define CS4349_LASTREG 0x08 +#define CS4349_NUMREGS (CS4349_LASTREG - CS4349_FIRSTREG + 1) +#define CS4349_I2C_INCR 0x80 + + +/* Device and Revision ID */ +#define CS4349_REVA 0xF0 /* Rev A */ +#define CS4349_REVB 0xF1 /* Rev B */ +#define CS4349_REVC2 0xFF /* Rev C2 */ + + +/* PDN_DONE Poll Maximum + * If soft ramp is set it will take much longer to power down + * the system. + */ +#define PDN_POLL_MAX 900 + + +/* Bitfield Definitions */ + +/* CS4349_MODE */ +/* (Digital Interface Format, De-Emphasis Control, Functional Mode */ +#define DIF2 (1 << 6) +#define DIF1 (1 << 5) +#define DIF0 (1 << 4) +#define DEM1 (1 << 3) +#define DEM0 (1 << 2) +#define FM1 (1 << 1) +#define DIF_LEFT_JST 0x00 +#define DIF_I2S 0x01 +#define DIF_RGHT_JST16 0x02 +#define DIF_RGHT_JST24 0x03 +#define DIF_TDM0 0x04 +#define DIF_TDM1 0x05 +#define DIF_TDM2 0x06 +#define DIF_TDM3 0x07 +#define DIF_MASK 0x70 +#define MODE_FORMAT(x) (((x)&7)<<4) +#define DEM_MASK 0x0C +#define NO_DEM 0x00 +#define DEM_441 0x04 +#define DEM_48K 0x08 +#define DEM_32K 0x0C +#define FM_AUTO 0x00 +#define FM_SNGL 0x01 +#define FM_DBL 0x02 +#define FM_QUAD 0x03 +#define FM_SNGL_MIN 30000 +#define FM_SNGL_MAX 54000 +#define FM_DBL_MAX 108000 +#define FM_QUAD_MAX 216000 +#define FM_MASK 0x03 + +/* CS4349_VMI (VMI = Volume, Mixing and Inversion Controls) */ +#define VOLBISA (1 << 7) +#define VOLAISB (1 << 7) +/* INVERT_A only available for Left Jstfd, Right Jstfd16 and Right Jstfd24 */ +#define INVERT_A (1 << 6) +/* INVERT_B only available for Left Jstfd, Right Jstfd16 and Right Jstfd24 */ +#define INVERT_B (1 << 5) +#define ATAPI3 (1 << 3) +#define ATAPI2 (1 << 2) +#define ATAPI1 (1 << 1) +#define ATAPI0 (1 << 0) +#define MUTEAB 0x00 +#define MUTEA_RIGHTB 0x01 +#define MUTEA_LEFTB 0x02 +#define MUTEA_SUMLRDIV2B 0x03 +#define RIGHTA_MUTEB 0x04 +#define RIGHTA_RIGHTB 0x05 +#define RIGHTA_LEFTB 0x06 +#define RIGHTA_SUMLRDIV2B 0x07 +#define LEFTA_MUTEB 0x08 +#define LEFTA_RIGHTB 0x09 /* Default */ +#define LEFTA_LEFTB 0x0A +#define LEFTA_SUMLRDIV2B 0x0B +#define SUMLRDIV2A_MUTEB 0x0C +#define SUMLRDIV2A_RIGHTB 0x0D +#define SUMLRDIV2A_LEFTB 0x0E +#define SUMLRDIV2_AB 0x0F +#define CHMIX_MASK 0x0F + +/* CS4349_MUTE */ +#define AUTOMUTE (1 << 7) +#define MUTEC_AB (1 << 5) +#define MUTE_A (1 << 4) +#define MUTE_B (1 << 3) +#define MUTE_AB_MASK 0x18 + +/* CS4349_RMPFLT (Ramp and Filter Control) */ +#define SCZ1 (1 << 7) +#define SCZ0 (1 << 6) +#define RMP_UP (1 << 5) +#define RMP_DN (1 << 4) +#define FILT_SEL (1 << 2) +#define IMMDT_CHNG 0x31 +#define ZEROCRSS 0x71 +#define SOFT_RMP 0xB1 +#define SFTRMP_ZEROCRSS 0xF1 +#define SR_ZC_MASK 0xC0 + +/* CS4349_MISC */ +#define PWR_DWN (1 << 7) +#define FREEZE (1 << 5) +#define POPG_EN (1 << 4) + +#endif /* __CS4349_H__ */ -- 2.4.5

2 1

[alsa-devel] [PATCH] ASoC: rt298: remove meanless pr_info
by Bard Liao 16 Jul '15

16 Jul '15

This line doesn't include any information. So, remove it. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt298.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 7c4bcb6..6ff68a3 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1144,8 +1144,6 @@ static int rt298_i2c_probe(struct i2c_client *i2c, const struct acpi_device_id *acpiid; int i, ret; - pr_info("%s\n", __func__); - rt298 = devm_kzalloc(&i2c->dev, sizeof(*rt298), GFP_KERNEL); if (NULL == rt298) -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt5645: Update dapm pins when the card is not instantiated yet
by Nicolas Boichat 16 Jul '15

16 Jul '15

This makes sure the dapm state is consistent when the card is instantiated. However, if the card is not instantiated yet, we still update the registers manually in the "jack in" case, so that we can immediately report if a mic is present or not. Disabling "Mic Det Power" after detection, and on jack out, can wait until the card gets instantiated. Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> --- sound/soc/codecs/rt5645.c | 36 +++++++++++------------------------- 1 file changed, 11 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 093a41d..a56bca6 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2805,12 +2805,11 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) if (jack_insert) { regmap_write(rt5645->regmap, RT5645_CHARGE_PUMP, 0x0006); - if (codec->component.card->instantiated) { - /* for jack type detect */ - snd_soc_dapm_force_enable_pin(dapm, "LDO2"); - snd_soc_dapm_force_enable_pin(dapm, "Mic Det Power"); - snd_soc_dapm_sync(dapm); - } else { + /* for jack type detect */ + snd_soc_dapm_force_enable_pin(dapm, "LDO2"); + snd_soc_dapm_force_enable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_sync(dapm); + if (!dapm->card->instantiated) { /* Power up necessary bits for JD if dapm is not ready yet */ regmap_update_bits(rt5645->regmap, RT5645_PWR_ANLG1, @@ -2841,12 +2840,8 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) rt5645_enable_push_button_irq(codec, true); } } else { - if (codec->component.card->instantiated) { - snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); - snd_soc_dapm_sync(dapm); - } else - regmap_update_bits(rt5645->regmap, - RT5645_PWR_VOL, RT5645_PWR_MIC_DET, 0); + snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_sync(dapm); rt5645->jack_type = SND_JACK_HEADPHONE; } @@ -2855,19 +2850,10 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) if (rt5645->en_button_func) rt5645_enable_push_button_irq(codec, false); else { - if (codec->component.card->instantiated) { - if (rt5645->pdata.jd_mode == 0) - snd_soc_dapm_disable_pin(dapm, "LDO2"); - snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); - snd_soc_dapm_sync(dapm); - } else { - if (rt5645->pdata.jd_mode == 0) - regmap_update_bits(rt5645->regmap, - RT5645_PWR_MIXER, - RT5645_PWR_LDO2, 0); - regmap_update_bits(rt5645->regmap, - RT5645_PWR_VOL, RT5645_PWR_MIC_DET, 0); - } + if (rt5645->pdata.jd_mode == 0) + snd_soc_dapm_disable_pin(dapm, "LDO2"); + snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_sync(dapm); } } -- 2.4.3.573.g4eafbef

2 3

[alsa-devel] Applied "ASoC: rt5645: Check if codec is initialized in workqueue handler" to the asoc tree
by Mark Brown 16 Jul '15

16 Jul '15

The patch ASoC: rt5645: Check if codec is initialized in workqueue handler 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 3c2234569cfff83718792f810a9bb855b78c2473 Mon Sep 17 00:00:00 2001 From: Nicolas Boichat <drinkcat(a)chromium.org> Date: Thu, 16 Jul 2015 11:30:58 +0800 Subject: [PATCH] ASoC: rt5645: Check if codec is initialized in workqueue handler rt5645_jack_detect_work needs rt5645->codec to be initialized to setup dapm pins. Also, reporting jack events is useless, as the jacks cannot be set before the codec is ready. Since we manually call the interrupt handler in rt5645_set_jack_detect, the initial jack state will be reported correctly, and dapm pins will be setup at that time. Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5645.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9ce311e..d1d49e8 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2943,6 +2943,9 @@ static int rt5645_irq_detection(struct rt5645_priv *rt5645) { int val, btn_type, gpio_state = 0, report = 0; + if (!rt5645->codec) + return; + switch (rt5645->pdata.jd_mode) { case 0: /* Not using rt5645 JD */ if (rt5645->gpiod_hp_det) { -- 2.1.4

1 0

[alsa-devel] [PATCH] ALSA: hda: fix kstrdup return value
by kernelpatch_update 16 Jul '15

16 Jul '15

In kstrdup we should return -ENOMEM when it reports an memory allocation failure, while the -ENODEV is referred to a failure in finding the cpu node in the device tree. Signed-off-by: kernelpatch_update <kernelpatch_update(a)163.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5de3c5d..d78fa71 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -975,7 +975,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, if (codec->bus->modelname) { codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL); if (!codec->modelname) { - err = -ENODEV; + err = -ENOMEM; goto error; } } -- 1.9.1

2 2

[alsa-devel] [PATCH 1/1] ASoC: sti-sas: fix x86 compilation issue.
by Arnaud Pouliquen 16 Jul '15

16 Jul '15

Fixes for compilation warnings and errors reported by Kbuild test robot for x86_64 and i386. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- sound/soc/codecs/sti-sas.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index 32db2c2..39f3c8b 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -227,8 +227,8 @@ static int stih416_dac_probe(struct snd_soc_dai *dai) dac->rst = devm_reset_control_get(codec->dev, "dac_rst"); if (IS_ERR(dac->rst)) { dev_err(dai->codec->dev, - "%s: ERROR: DAC reset control not defined (%d)!\n", - __func__, (int)dac->rst); + "%s: ERROR: DAC reset control not defined !\n", + __func__); dac->rst = NULL; return -EFAULT; } @@ -547,6 +547,7 @@ static int sti_sas_driver_probe(struct platform_device *pdev) { struct device_node *pnode = pdev->dev.of_node; struct sti_sas_data *drvdata; + const struct of_device_id *of_id; /* Allocate device structure */ drvdata = devm_kzalloc(&pdev->dev, sizeof(struct sti_sas_data), @@ -555,12 +556,13 @@ static int sti_sas_driver_probe(struct platform_device *pdev) return -ENOMEM; /* Populate data structure depending on compatibility */ - if (!of_match_node(sti_sas_dev_match, pnode)->data) { + of_id = of_match_node(sti_sas_dev_match, pnode); + if (!of_id->data) { dev_err(&pdev->dev, "data associated to device is missing"); return -EINVAL; } - drvdata->dev_data = of_match_node(sti_sas_dev_match, pnode)->data; + drvdata->dev_data = (struct sti_sas_dev_data *)of_id->data; /* Initialise device structure */ drvdata->dev = &pdev->dev; -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: fsi: Remove obsolete sh_fsi2 platform_device_id entry
by Geert Uytterhoeven 16 Jul '15

16 Jul '15

Since the removal of the r8a7740 legacy board code in commit 1fa59bda21c7fa36 ("ARM: shmobile: Remove legacy board code for Armadillo-800 EVA"), all former users of the "sh_fsi2" platform device name are only supported in generic DT-only ARM multi-platform builds. The driver doesn't need to match platform devices by name anymore, hence remove the corresponding platform_device_id entry. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- The commit mentioned above is in -next, destined for v4.3. --- sound/soc/sh/fsi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 142c066eaee2ecb1..0215c78cbddf5491 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1911,7 +1911,6 @@ MODULE_DEVICE_TABLE(of, fsi_of_match); static const struct platform_device_id fsi_id_table[] = { { "sh_fsi", (kernel_ulong_t)&fsi1_core }, - { "sh_fsi2", (kernel_ulong_t)&fsi2_core }, {}, }; MODULE_DEVICE_TABLE(platform, fsi_id_table); -- 1.9.1

4 3

[alsa-devel] [PATCH v3] ASoC: rt5645: Check if codec is initialized in workqueue handler
by Nicolas Boichat 16 Jul '15

16 Jul '15

rt5645_jack_detect_work needs rt5645->codec to be initialized to setup dapm pins. Also, reporting jack events is useless, as the jacks cannot be set before the codec is ready. Since we manually call the interrupt handler in rt5645_set_jack_detect, the initial jack state will be reported correctly, and dapm pins will be setup at that time. Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> --- sound/soc/codecs/rt5645.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 093a41d..ff75d04 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2918,6 +2918,9 @@ static void rt5645_jack_detect_work(struct work_struct *work) container_of(work, struct rt5645_priv, jack_detect_work.work); int val, btn_type, gpio_state = 0, report = 0; + if (!rt5645->codec) + return; + switch (rt5645->pdata.jd_mode) { case 0: /* Not using rt5645 JD */ if (rt5645->gpiod_hp_det) { -- 2.4.3.573.g4eafbef

1 0

[alsa-devel] [PATCH v2 0/3] ASoC: rt5645: Remove codec dependency in workqueue handler
by Nicolas Boichat 16 Jul '15

16 Jul '15

Same issue as https://patchwork.kernel.org/patch/6694351/, but reworked fix. The first 2 patches are preparatory work for the 3rd one, which fixes occasional kernel panics on boot. Nicolas Boichat (3): ASoC: rt5645: Simplify rt5645_enable_push_button_irq ASoC: rt5645: Remove irq_jack_detection function ASoC: rt5645: Remove codec dependency in workqueue handler sound/soc/codecs/rt5645.c | 185 +++++++++++++++++++++++----------------------- sound/soc/codecs/rt5645.h | 1 + 2 files changed, 95 insertions(+), 91 deletions(-) -- 2.4.3.573.g4eafbef

3 14

[alsa-devel] [PATCH] ALSA: pcm_dmaengine: Run complete cb only if runtime check passes
by Misael Lopez Cruz 15 Jul '15

15 Jul '15

The DMA complete callback uses substream's runtime data, so make sure the runtime check passes before going forward. The callback could run while the stream is closing or after it's already closed. By then, the substream's runtime data might already been freed and cause NULL pointer dereferences. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Reviewed-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/core/pcm_dmaengine.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index fba365a..061a8f7 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -133,7 +133,12 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_set_config_from_dai_data); static void dmaengine_pcm_dma_complete(void *arg) { struct snd_pcm_substream *substream = arg; - struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct dmaengine_pcm_runtime_data *prtd; + + if (PCM_RUNTIME_CHECK(substream)) + return; + + prtd = substream_to_prtd(substream); prtd->pos += snd_pcm_lib_period_bytes(substream); if (prtd->pos >= snd_pcm_lib_buffer_bytes(substream)) -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: Intel: Get correct usage_count value to load firmware
by yang.a.fang＠intel.com 15 Jul '15

15 Jul '15

From: Shilpa Sreeramalu <shilpa.sreeramalu(a)intel.com> The usage_count variable was read before it was set to the correct value, due to which the firmware load was failing. Because of this IPC messages sent to the firmware were timing out causing a delay of about 1 second while playing audio from the internal speakers. With this patch the usage_count is read after the function call pm_runtime_get_sync which will increment the usage_count variable and the firmware load is successful and all the IPC messages are processed correctly. Cc: stable(a)vger.kernel.org Signed-off-by: Shilpa Sreeramalu <shilpa.sreeramalu(a)intel.com> --- sound/soc/intel/atom/sst/sst_drv_interface.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 620da1d..440b953 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -42,6 +42,11 @@ #define MIN_FRAGMENT_SIZE (50 * 1024) #define MAX_FRAGMENT_SIZE (1024 * 1024) #define SST_GET_BYTES_PER_SAMPLE(pcm_wd_sz) (((pcm_wd_sz + 15) >> 4) << 1) +#ifdef PM_CONFIG +#define GET_USAGE_COUNT(dev) (atomic_read(&dev->power.usage_count)) +#else +#define GET_USAGE_COUNT(dev) 1 +#endif int free_stream_context(struct intel_sst_drv *ctx, unsigned int str_id) { @@ -141,15 +146,9 @@ static int sst_power_control(struct device *dev, bool state) int ret = 0; int usage_count = 0; -#ifdef CONFIG_PM - usage_count = atomic_read(&dev->power.usage_count); -#else - usage_count = 1; -#endif - if (state == true) { ret = pm_runtime_get_sync(dev); - + usage_count = GET_USAGE_COUNT(dev); dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count); if (ret < 0) { dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret); @@ -164,6 +163,7 @@ static int sst_power_control(struct device *dev, bool state) } } } else { + usage_count = GET_USAGE_COUNT(dev); dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", usage_count); return sst_pm_runtime_put(ctx); } -- 1.7.9.5

4 6

[alsa-devel] [PATCH] Softvol: Allow per process volume control.
by Jimmy 15 Jul '15

15 Jul '15

This change uses ephemeral pcm controls to provide per process volume control in conjunction with dmix. It basically generates a new control every time a softvol PCM is opened and deletes it afterwards. It also needs to generate a unique name for the control and I have included one method of doing that. This patch is totally non-breaking and backwards compatible, apart from one call to snd_ctl_elem_unlock (see below). Changes to asoundrc: * pcm.softvol.per_process BOOL has been added to enable these changes, defaults to false. * when per_process is true then bits of text in pcm.softvol.control.name surrounded by curly braces ('{' and '}') are treated as substitution variables (eg "softvol.{pid}") so that uniquely named controls can be generated. Changes to pcm_softvol.c * new int (really a bool) in snd_pcm_softvol_t that is read in _...open() and passed through to the internal open method because that seemed to be the thing to do and then used in softvol_load_control(). * a big ugly per_proc_format_ctl_name() function has been added that generates the new hopefully unique names. It talks to /proc/ and does a bunch string manipulation. There probably are no off-by-one errors left in there. * per process controls are closed along with the pcm. * I added a call to snd_ctl_elem_unlock() when a new control is created and *didn't* put a guard on it because I don't understand why anyone would want the control locked anyway ... It seems that the first time that softvol is opened you can't use the control so you have to close and open it again. Presumably there is a reason for this because unlock isn't called anywhere else, but I don't know what it is. Nothing else calls unlock either, maybe nothing else makes new on-demand ctls. I'm open to suggestions like "get rid of the templating" and "use better config variable names" but as far as I can tell the basic idea is useful and simple. --- src/pcm/pcm_softvol.c | 148 ++++++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 144 insertions(+), 4 deletions(-) diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c index c6cfd8896b26..a03b672a6f42 100644 --- a/src/pcm/pcm_softvol.c +++ b/src/pcm/pcm_softvol.c @@ -31,6 +31,11 @@ #include "pcm_local.h" #include "pcm_plugin.h" +#include <sys/types.h> +#include <unistd.h> +#include <errno.h> +#include <libgen.h> //basename + #ifndef PIC /* entry for static linking */ const char *_snd_module_pcm_softvol = ""; @@ -51,6 +56,7 @@ typedef struct { double min_dB; double max_dB; unsigned int *dB_value; + int per_process; } snd_pcm_softvol_t; #define VOL_SCALE_SHIFT 16 @@ -408,6 +414,17 @@ static void softvol_free(snd_pcm_softvol_t *svol) static int snd_pcm_softvol_close(snd_pcm_t *pcm) { snd_pcm_softvol_t *svol = pcm->private_data; + snd_ctl_elem_info_t *cinfo; + + if (svol->per_process) { + snd_ctl_elem_info_alloca(&cinfo); + snd_ctl_elem_info_set_id(cinfo, &svol->elem.id); + // Ignore non-existence at this point. + if (!snd_ctl_elem_info(svol->ctl, cinfo)) { + snd_ctl_elem_remove(svol->ctl, &cinfo->id); + } + } + softvol_free(svol); return 0; } @@ -689,6 +706,92 @@ static int add_user_ctl(snd_pcm_softvol_t *svol, snd_ctl_elem_info_t *cinfo, int return snd_ctl_elem_write(svol->ctl, &svol->elem); } +/* Takes a template string of the form "softvol {cmd} ({pid})" and writes to + * outbuf a the same string with the text within curly braces replaced with + * relevant information. Returns the number of chars written to outbuf + * including a terminating '\0' or a negative number on error. + * + * Currently supported values for substition are: + * pid The current processes pid + * cmd The basename of the processes argv[0], from /proc/self/cmdline + * exe The basename of the processes binary, from following /proc/self/exe + * { The literal open curly brace '{' + * ob The literal open curly brace '{' + * cb The literal open curly brace '}' + */ +static int per_proc_format_ctl_name(char *outbuf, size_t outbuf_len, char *template) { + int template_len = strlen(template), n, proc_fd; + char *outbuf_p = outbuf, *open_b, *close_b, *template_p = template; + char proc_buf[50]; // max pid can be up to 22 bits ... + char resolved_path[PATH_MAX]; + + // walk through template_p pushing onto outbuf_p + while (outbuf_p < outbuf+outbuf_len) { + open_b = strchr(template_p, '{'); + if (open_b == NULL) { + // Point the end of the current literal to the end + open_b = template + template_len; + } + memcpy(outbuf_p, template_p, open_b-template_p); + outbuf_p += open_b-template_p; + template_p = open_b+1; + + if (template_p >= template + template_len - 1) { + *outbuf_p = '\0'; + break; + } + + if ((close_b = strchr(open_b, '}')) == NULL) { + SNDERR("Invalid control name template: Unmatched '{'"); + return -EINVAL; + } + *close_b = '\0'; + if (strcmp(template_p, "pid") == 0) { + n = snprintf(outbuf_p, outbuf+outbuf_len-outbuf_p, "%d", getpid()); + } else if (strcmp(template_p, "cmd") == 0) { + n = snprintf(proc_buf, 49, "/proc/%d/cmdline", getpid()); + proc_fd = open(proc_buf, O_RDONLY); + if (proc_fd < 0) { + // Maybe a short lived process? + n = errno; + SNDERR("Unable to complete control name template: open: %s: %s", strerror(errno), proc_buf); + return -n; + } + n = read(proc_fd, proc_buf, 49); + proc_buf[n] = '\0'; + n = basename(proc_buf)-proc_buf; + open_b = strchr(proc_buf+n, ' '); + if (open_b != NULL) *open_b = '\0'; + n = snprintf(outbuf_p, outbuf+outbuf_len-outbuf_p, "%s", proc_buf+n); + } else if (strcmp(template_p, "exe") == 0) { + n = snprintf(proc_buf, 49, "/proc/%d/exe", getpid()); + if (realpath(proc_buf, resolved_path) == NULL) { + // Maybe a short lived process? + n = errno; + SNDERR("Unable to complete control name template: open: %s: %s", strerror(errno), proc_buf); + return -n; + } + n = basename(resolved_path)-resolved_path; + n = snprintf(outbuf_p, outbuf+outbuf_len-outbuf_p, "%s", basename(resolved_path)); + } else if (strcmp(template_p, "{") == 0 || strcmp(template_p, "ob") == 0) { + n = 1; + *outbuf_p = '{'; + } else if (strcmp(template_p, "cb") == 0) { + n = 1; + *outbuf_p = '}'; + } else if (strlen(template_p) == 0) { + n = 0; + } else { + SNDERR("Invalid control name template: Unknown substitution: %s", template_p); + return -EINVAL; + } + + outbuf_p = outbuf_p + n; + template_p = close_b + 1; + } + return strlen(outbuf)+1; +} + /* * load and set up user-control * returns 0 if the user-control is found or created, @@ -700,7 +803,7 @@ static int softvol_load_control(snd_pcm_t *pcm, snd_pcm_softvol_t *svol, int cchannels, double min_dB, double max_dB, int resolution) { - char tmp_name[32]; + char tmp_name[32], tmp_template[sizeof(ctl_id->name)]; snd_pcm_info_t *info; snd_ctl_elem_info_t *cinfo; int err; @@ -724,6 +827,15 @@ static int softvol_load_control(snd_pcm_t *pcm, snd_pcm_softvol_t *svol, return err; } + if (svol->per_process) { + err = per_proc_format_ctl_name(tmp_template, sizeof(ctl_id->name), ctl_id->name); + if (err < 1) { // Shouldn't be zero length output... + SNDERR("Cannot open CTL %s: Control name template error %d", tmp_name, err); + return -EINVAL; + } + memcpy(&ctl_id->name, tmp_template, sizeof(ctl_id->name)); + } + svol->elem.id = *ctl_id; svol->max_val = resolution - 1; svol->min_dB = min_dB; @@ -747,6 +859,7 @@ static int softvol_load_control(snd_pcm_t *pcm, snd_pcm_softvol_t *svol, SNDERR("Cannot add a control"); return err; } + snd_ctl_elem_unlock(svol->ctl, &cinfo->id); } else { if (! (cinfo->access & SNDRV_CTL_ELEM_ACCESS_USER)) { /* hardware control exists */ @@ -836,6 +949,10 @@ static const snd_pcm_ops_t snd_pcm_softvol_ops = { * \param resolution resolution of control * \param slave Slave PCM handle * \param close_slave When set, the slave PCM handle is closed with copy PCM + * \param per_process When set, a new ephemeral control is created for each + * softvol PCM. This allows per process volume control when used with + * dmix. Make sure the control name has eg {pid}, {cmd}, {exe} in it so + * the generated names are unique. * \retval zero on success otherwise a negative error code * \warning Using of this function might be dangerous in the sense * of compatibility reasons. The prototype might be freely @@ -846,7 +963,7 @@ int snd_pcm_softvol_open(snd_pcm_t **pcmp, const char *name, int ctl_card, snd_ctl_elem_id_t *ctl_id, int cchannels, double min_dB, double max_dB, int resolution, - snd_pcm_t *slave, int close_slave) + snd_pcm_t *slave, int close_slave, int per_process) { snd_pcm_t *pcm; snd_pcm_softvol_t *svol; @@ -862,6 +979,7 @@ int snd_pcm_softvol_open(snd_pcm_t **pcmp, const char *name, svol = calloc(1, sizeof(*svol)); if (! svol) return -ENOMEM; + svol->per_process = per_process; err = softvol_load_control(slave, svol, ctl_card, ctl_id, cchannels, min_dB, max_dB, resolution); if (err < 0) { @@ -929,6 +1047,17 @@ If the control already exists and it's a system control (i.e. no user-defined control), the plugin simply passes its slave without any changes. +If per_process is set you can (and should) user certain substition variables +in the control name so that uniquely named controls can be generated per-process. +Substition variables are within curly braces, everything else is used literaly, eg +{cmd} ({pid}). Currently supported variables are: +- pid The current processes pid +- cmd The basename of the processes argv[0], from /proc/self/cmdline +- exe The basename of the processes binary, from following /proc/self/exe +- { The literal open curly brace '{' +- ob The literal open curly brace '{' +- cb The literal open curly brace '}' + \code pcm.name { type softvol # Soft Volume conversion PCM @@ -953,6 +1082,8 @@ pcm.name { [max_dB REAL] # maximal dB value (default: 0.0) [resolution INT] # resolution (default: 256) # resolution = 2 means a mute switch + [per_process BOOL] # generate one control per process (default: false) + # useful with dmix. } \endcode @@ -992,7 +1123,7 @@ int _snd_pcm_softvol_open(snd_pcm_t **pcmp, const char *name, int resolution = PRESET_RESOLUTION; double min_dB = PRESET_MIN_DB; double max_dB = ZERO_DB; - int card = -1, cchannels = 2; + int card = -1, cchannels = 2, per_process = 0; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); @@ -1035,6 +1166,15 @@ int _snd_pcm_softvol_open(snd_pcm_t **pcmp, const char *name, } continue; } + if (strcmp(id, "per_process") == 0) { + err = snd_config_get_bool(n); + if (err < 0) { + SNDERR("Invalid boolean value in per_process"); + return err; + } + per_process = err; + continue; + } SNDERR("Unknown field %s", id); return -EINVAL; } @@ -1092,7 +1232,7 @@ int _snd_pcm_softvol_open(snd_pcm_t **pcmp, const char *name, return err; } err = snd_pcm_softvol_open(pcmp, name, sformat, card, ctl_id, cchannels, - min_dB, max_dB, resolution, spcm, 1); + min_dB, max_dB, resolution, spcm, 1, per_process); if (err < 0) snd_pcm_close(spcm); } -- 2.1.4

3 2

[alsa-devel] [PATCH 1/2][linux-next] ASoC: Fix warning while make xmldocs caused by soc-core.c
by Masanari Iida 15 Jul '15

15 Jul '15

This patch fix following warning while make xmldocs. Warning(.//sound/soc/soc-core.c:2148): No description found for parameter 'ratio' Add missing ":" Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8c8ce65..691fcc4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2140,7 +2140,7 @@ EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); /** * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. * @dai: DAI - * @ratio Ratio of BCLK to Sample rate. + * @ratio: Ratio of BCLK to Sample rate. * * Configures the DAI for a preset BCLK to sample rate ratio. */ -- 2.5.0.rc2.13.g961abca

1 1

[alsa-devel] [PATCH] Drop owner assignment from i2c_driver (and platform left-overs)
by Krzysztof Kozlowski 15 Jul '15

15 Jul '15

3 6

[alsa-devel] [patch] ASoC: fsl: fsl_spdif: signedness bug in fsl_spdif_startup()
by Dan Carpenter 15 Jul '15

15 Jul '15

We need "i" to be signed or it leads to a forever loop in the error handling. Fixes: fa3be9208dcb ('ASoC: fsl: fsl_spdif: Check for clk_prepare_enable() error') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 3ee4539..7a92f30 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -454,7 +454,8 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream, struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); struct platform_device *pdev = spdif_priv->pdev; struct regmap *regmap = spdif_priv->regmap; - u32 scr, mask, i; + u32 scr, mask; + int i; int ret; /* Reset module and interrupts only for first initialization */

3 2

[alsa-devel] [PATCH] ASoC: Constify snd_soc_dai_ops variables
by Axel Lin 15 Jul '15

15 Jul '15

The snd_soc_dai_ops variables are not modified after initialization in these drivers, so make them const. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/cs42l56.c | 2 +- sound/soc/codecs/isabelle.c | 8 ++++---- sound/soc/codecs/lm49453.c | 10 +++++----- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/max98357a.c | 2 +- sound/soc/codecs/mc13783.c | 6 +++--- sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5670.c | 2 +- sound/soc/codecs/rt5677.c | 2 +- sound/soc/codecs/si476x.c | 2 +- sound/soc/codecs/tas2552.c | 2 +- sound/soc/codecs/tlv320aic31xx.c | 2 +- 13 files changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 4de52c9..c8f4af2 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -919,7 +919,7 @@ static int cs42l52_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE) -static struct snd_soc_dai_ops cs42l52_ops = { +static const struct snd_soc_dai_ops cs42l52_ops = { .hw_params = cs42l52_pcm_hw_params, .digital_mute = cs42l52_digital_mute, .set_fmt = cs42l52_set_fmt, diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 1e11ba4..22bc2f3 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -989,7 +989,7 @@ static int cs42l56_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops cs42l56_ops = { +static const struct snd_soc_dai_ops cs42l56_ops = { .hw_params = cs42l56_pcm_hw_params, .digital_mute = cs42l56_digital_mute, .set_fmt = cs42l56_set_dai_fmt, diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index ac7dfe7..4013864 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1016,25 +1016,25 @@ static int isabelle_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) #define ISABELLE_FORMATS (SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops isabelle_hs_dai_ops = { +static const struct snd_soc_dai_ops isabelle_hs_dai_ops = { .hw_params = isabelle_hw_params, .set_fmt = isabelle_set_dai_fmt, .digital_mute = isabelle_hs_mute, }; -static struct snd_soc_dai_ops isabelle_hf_dai_ops = { +static const struct snd_soc_dai_ops isabelle_hf_dai_ops = { .hw_params = isabelle_hw_params, .set_fmt = isabelle_set_dai_fmt, .digital_mute = isabelle_hf_mute, }; -static struct snd_soc_dai_ops isabelle_line_dai_ops = { +static const struct snd_soc_dai_ops isabelle_line_dai_ops = { .hw_params = isabelle_hw_params, .set_fmt = isabelle_set_dai_fmt, .digital_mute = isabelle_line_mute, }; -static struct snd_soc_dai_ops isabelle_ul_dai_ops = { +static const struct snd_soc_dai_ops isabelle_ul_dai_ops = { .hw_params = isabelle_hw_params, .set_fmt = isabelle_set_dai_fmt, }; diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index ef432aa..4e76992 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1287,35 +1287,35 @@ static int lm49453_set_bias_level(struct snd_soc_codec *codec, #define LM49453_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops lm49453_headset_dai_ops = { +static const struct snd_soc_dai_ops lm49453_headset_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, .digital_mute = lm49453_hp_mute, }; -static struct snd_soc_dai_ops lm49453_speaker_dai_ops = { +static const struct snd_soc_dai_ops lm49453_speaker_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, .digital_mute = lm49453_ls_mute, }; -static struct snd_soc_dai_ops lm49453_haptic_dai_ops = { +static const struct snd_soc_dai_ops lm49453_haptic_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, .digital_mute = lm49453_ha_mute, }; -static struct snd_soc_dai_ops lm49453_ep_dai_ops = { +static const struct snd_soc_dai_ops lm49453_ep_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, .digital_mute = lm49453_ep_mute, }; -static struct snd_soc_dai_ops lm49453_lineout_dai_ops = { +static const struct snd_soc_dai_ops lm49453_lineout_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 1697340..a2cdf52 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2386,7 +2386,7 @@ EXPORT_SYMBOL_GPL(max98090_mic_detect); #define MAX98090_RATES SNDRV_PCM_RATE_8000_96000 #define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops max98090_dai_ops = { +static const struct snd_soc_dai_ops max98090_dai_ops = { .set_sysclk = max98090_dai_set_sysclk, .set_fmt = max98090_dai_set_fmt, .set_tdm_slot = max98090_set_tdm_slot, diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index e88119a..f5e3dce 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -79,7 +79,7 @@ static struct snd_soc_codec_driver max98357a_codec_driver = { .num_dapm_routes = ARRAY_SIZE(max98357a_dapm_routes), }; -static struct snd_soc_dai_ops max98357a_dai_ops = { +static const struct snd_soc_dai_ops max98357a_dai_ops = { .trigger = max98357a_daiops_trigger, }; diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 3d44fc5..3e770cb 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -650,14 +650,14 @@ static int mc13783_remove(struct snd_soc_codec *codec) #define MC13783_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops mc13783_ops_dac = { +static const struct snd_soc_dai_ops mc13783_ops_dac = { .hw_params = mc13783_pcm_hw_params_dac, .set_fmt = mc13783_set_fmt_async, .set_sysclk = mc13783_set_sysclk_dac, .set_tdm_slot = mc13783_set_tdm_slot_dac, }; -static struct snd_soc_dai_ops mc13783_ops_codec = { +static const struct snd_soc_dai_ops mc13783_ops_codec = { .hw_params = mc13783_pcm_hw_params_codec, .set_fmt = mc13783_set_fmt_async, .set_sysclk = mc13783_set_sysclk_codec, @@ -698,7 +698,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = { }, }; -static struct snd_soc_dai_ops mc13783_ops_sync = { +static const struct snd_soc_dai_ops mc13783_ops_sync = { .hw_params = mc13783_pcm_hw_params_sync, .set_fmt = mc13783_set_fmt_sync, .set_sysclk = mc13783_set_sysclk_sync, diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9dfa431..9b1ad6b 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3106,7 +3106,7 @@ static int rt5645_resume(struct snd_soc_codec *codec) #define RT5645_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -static struct snd_soc_dai_ops rt5645_aif_dai_ops = { +static const struct snd_soc_dai_ops rt5645_aif_dai_ops = { .hw_params = rt5645_hw_params, .set_fmt = rt5645_set_dai_fmt, .set_sysclk = rt5645_set_dai_sysclk, diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index a791d7e..132d012 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2720,7 +2720,7 @@ static int rt5670_resume(struct snd_soc_codec *codec) #define RT5670_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -static struct snd_soc_dai_ops rt5670_aif_dai_ops = { +static const struct snd_soc_dai_ops rt5670_aif_dai_ops = { .hw_params = rt5670_hw_params, .set_fmt = rt5670_set_dai_fmt, .set_sysclk = rt5670_set_dai_sysclk, diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 13b871f..151db60 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4861,7 +4861,7 @@ static int rt5677_write(void *context, unsigned int reg, unsigned int val) #define RT5677_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -static struct snd_soc_dai_ops rt5677_aif_dai_ops = { +static const struct snd_soc_dai_ops rt5677_aif_dai_ops = { .hw_params = rt5677_hw_params, .set_fmt = rt5677_set_dai_fmt, .set_sysclk = rt5677_set_dai_sysclk, diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 3e72964..a8402d0 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -208,7 +208,7 @@ out: return err; } -static struct snd_soc_dai_ops si476x_dai_ops = { +static const struct snd_soc_dai_ops si476x_dai_ops = { .hw_params = si476x_codec_hw_params, .set_fmt = si476x_codec_set_dai_fmt, }; diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 71cfae0..f2ae302 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -520,7 +520,7 @@ static const struct dev_pm_ops tas2552_pm = { NULL) }; -static struct snd_soc_dai_ops tas2552_speaker_dai_ops = { +static const struct snd_soc_dai_ops tas2552_speaker_dai_ops = { .hw_params = tas2552_hw_params, .prepare = tas2552_prepare, .set_sysclk = tas2552_set_dai_sysclk, diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index c4c960f..df01fec 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1121,7 +1121,7 @@ static struct snd_soc_codec_driver soc_codec_driver_aic31xx = { .num_dapm_routes = ARRAY_SIZE(aic31xx_audio_map), }; -static struct snd_soc_dai_ops aic31xx_dai_ops = { +static const struct snd_soc_dai_ops aic31xx_dai_ops = { .hw_params = aic31xx_hw_params, .set_sysclk = aic31xx_set_dai_sysclk, .set_fmt = aic31xx_set_dai_fmt, -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: da732x: Merge da732x_dai1_ops and da732x_dai2_ops
by Axel Lin 15 Jul '15

15 Jul '15

da732x_dai1_ops is identical to da732x_dai2_ops, so merge them to da732x_dai_ops. The da732x_dai_ops is not modified after initialization, so make it const. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/da732x.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index b1cb465..29251b0 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1196,13 +1196,7 @@ static int da732x_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, #define DA732X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops da732x_dai1_ops = { - .hw_params = da732x_hw_params, - .set_fmt = da732x_set_dai_fmt, - .set_sysclk = da732x_set_dai_sysclk, -}; - -static struct snd_soc_dai_ops da732x_dai2_ops = { +static const struct snd_soc_dai_ops da732x_dai_ops = { .hw_params = da732x_hw_params, .set_fmt = da732x_set_dai_fmt, .set_sysclk = da732x_set_dai_sysclk, @@ -1227,7 +1221,7 @@ static struct snd_soc_dai_driver da732x_dai[] = { .rates = DA732X_RATES, .formats = DA732X_FORMATS, }, - .ops = &da732x_dai1_ops, + .ops = &da732x_dai_ops, }, { .name = "DA732X_AIFB", @@ -1247,7 +1241,7 @@ static struct snd_soc_dai_driver da732x_dai[] = { .rates = DA732X_RATES, .formats = DA732X_FORMATS, }, - .ops = &da732x_dai2_ops, + .ops = &da732x_dai_ops, }, }; -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: sirf-audio-codec: Staticise local symbols
by Axel Lin 15 Jul '15

15 Jul '15

Also make sirf_audio_codec_dai_ops const. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/sirf-audio-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index 29cb442..6bfd25c 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -370,11 +370,11 @@ static int sirf_audio_codec_trigger(struct snd_pcm_substream *substream, return 0; } -struct snd_soc_dai_ops sirf_audio_codec_dai_ops = { +static const struct snd_soc_dai_ops sirf_audio_codec_dai_ops = { .trigger = sirf_audio_codec_trigger, }; -struct snd_soc_dai_driver sirf_audio_codec_dai = { +static struct snd_soc_dai_driver sirf_audio_codec_dai = { .name = "sirf-audio-codec", .playback = { .stream_name = "Playback", -- 2.1.0

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: Drop owner assignment from i2c_driver
by Krzysztof Kozlowski 15 Jul '15

15 Jul '15

i2c_driver does not need to set an owner because i2c_register_driver() will set it. Signed-off-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> --- The coccinelle script which generated the patch was sent here: http://www.spinics.net/lists/kernel/msg2029903.html Changes since v1: 1. Rebased on next-20150714 2. Removed owner from rt298.c --- sound/soc/codecs/adau1373.c | 1 - sound/soc/codecs/adau1701.c | 1 - sound/soc/codecs/adau1761-i2c.c | 1 - sound/soc/codecs/adau1781-i2c.c | 1 - sound/soc/codecs/adau1977-i2c.c | 1 - sound/soc/codecs/adav803.c | 1 - sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/ak4641.c | 1 - sound/soc/codecs/ak4642.c | 1 - sound/soc/codecs/ak4671.c | 1 - sound/soc/codecs/alc5623.c | 1 - sound/soc/codecs/alc5632.c | 1 - sound/soc/codecs/cs35l32.c | 1 - sound/soc/codecs/cs4265.c | 1 - sound/soc/codecs/cs4270.c | 1 - sound/soc/codecs/cs4271-i2c.c | 1 - sound/soc/codecs/cs42l51-i2c.c | 1 - sound/soc/codecs/cs42l52.c | 1 - sound/soc/codecs/cs42l56.c | 1 - sound/soc/codecs/cs42l73.c | 1 - sound/soc/codecs/cs42xx8-i2c.c | 1 - sound/soc/codecs/da7210.c | 1 - sound/soc/codecs/da7213.c | 1 - sound/soc/codecs/da732x.c | 1 - sound/soc/codecs/da9055.c | 1 - sound/soc/codecs/isabelle.c | 1 - sound/soc/codecs/lm4857.c | 1 - sound/soc/codecs/lm49453.c | 1 - sound/soc/codecs/max9768.c | 1 - sound/soc/codecs/max98088.c | 1 - sound/soc/codecs/max98090.c | 1 - sound/soc/codecs/max98095.c | 1 - sound/soc/codecs/max9850.c | 1 - sound/soc/codecs/max9877.c | 1 - sound/soc/codecs/max98925.c | 1 - sound/soc/codecs/ml26124.c | 1 - sound/soc/codecs/pcm1681.c | 1 - sound/soc/codecs/pcm512x-i2c.c | 1 - sound/soc/codecs/rt286.c | 1 - sound/soc/codecs/rt298.c | 1 - sound/soc/codecs/rt5631.c | 1 - sound/soc/codecs/rt5640.c | 1 - sound/soc/codecs/rt5645.c | 1 - sound/soc/codecs/rt5651.c | 1 - sound/soc/codecs/rt5670.c | 1 - sound/soc/codecs/rt5677.c | 1 - sound/soc/codecs/sgtl5000.c | 1 - sound/soc/codecs/ssm2518.c | 1 - sound/soc/codecs/ssm2602-i2c.c | 1 - sound/soc/codecs/ssm4567.c | 1 - sound/soc/codecs/sta32x.c | 1 - sound/soc/codecs/sta350.c | 1 - sound/soc/codecs/sta529.c | 1 - sound/soc/codecs/tas2552.c | 1 - sound/soc/codecs/tas5086.c | 1 - sound/soc/codecs/tfa9879.c | 1 - sound/soc/codecs/tlv320aic31xx.c | 1 - sound/soc/codecs/tlv320aic32x4.c | 1 - sound/soc/codecs/tlv320aic3x.c | 1 - sound/soc/codecs/tlv320dac33.c | 1 - sound/soc/codecs/tpa6130a2.c | 1 - sound/soc/codecs/ts3a227e.c | 1 - sound/soc/codecs/uda1380.c | 1 - sound/soc/codecs/wm1250-ev1.c | 1 - sound/soc/codecs/wm2000.c | 1 - sound/soc/codecs/wm2200.c | 1 - sound/soc/codecs/wm5100.c | 1 - sound/soc/codecs/wm8510.c | 1 - sound/soc/codecs/wm8523.c | 1 - sound/soc/codecs/wm8580.c | 1 - sound/soc/codecs/wm8711.c | 1 - sound/soc/codecs/wm8728.c | 1 - sound/soc/codecs/wm8731.c | 1 - sound/soc/codecs/wm8737.c | 1 - sound/soc/codecs/wm8741.c | 1 - sound/soc/codecs/wm8750.c | 1 - sound/soc/codecs/wm8753.c | 1 - sound/soc/codecs/wm8776.c | 1 - sound/soc/codecs/wm8804-i2c.c | 1 - sound/soc/codecs/wm8900.c | 1 - sound/soc/codecs/wm8903.c | 1 - sound/soc/codecs/wm8904.c | 1 - sound/soc/codecs/wm8940.c | 1 - sound/soc/codecs/wm8955.c | 1 - sound/soc/codecs/wm8960.c | 1 - sound/soc/codecs/wm8961.c | 1 - sound/soc/codecs/wm8962.c | 1 - sound/soc/codecs/wm8971.c | 1 - sound/soc/codecs/wm8974.c | 1 - sound/soc/codecs/wm8978.c | 1 - sound/soc/codecs/wm8983.c | 1 - sound/soc/codecs/wm8985.c | 1 - sound/soc/codecs/wm8988.c | 1 - sound/soc/codecs/wm8990.c | 1 - sound/soc/codecs/wm8991.c | 1 - sound/soc/codecs/wm8993.c | 1 - sound/soc/codecs/wm8995.c | 1 - sound/soc/codecs/wm8996.c | 1 - sound/soc/codecs/wm9081.c | 1 - sound/soc/codecs/wm9090.c | 1 - 100 files changed, 100 deletions(-) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index a43160254929..6c96860f46de 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1534,7 +1534,6 @@ MODULE_DEVICE_TABLE(i2c, adau1373_i2c_id); static struct i2c_driver adau1373_i2c_driver = { .driver = { .name = "adau1373", - .owner = THIS_MODULE, }, .probe = adau1373_i2c_probe, .remove = adau1373_i2c_remove, diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index ff7f846e3b76..de53c0d7bf10 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -915,7 +915,6 @@ MODULE_DEVICE_TABLE(i2c, adau1701_i2c_id); static struct i2c_driver adau1701_i2c_driver = { .driver = { .name = "adau1701", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(adau1701_dt_ids), }, .probe = adau1701_i2c_probe, diff --git a/sound/soc/codecs/adau1761-i2c.c b/sound/soc/codecs/adau1761-i2c.c index 862796dec693..348ccb17d3cc 100644 --- a/sound/soc/codecs/adau1761-i2c.c +++ b/sound/soc/codecs/adau1761-i2c.c @@ -47,7 +47,6 @@ MODULE_DEVICE_TABLE(i2c, adau1761_i2c_ids); static struct i2c_driver adau1761_i2c_driver = { .driver = { .name = "adau1761", - .owner = THIS_MODULE, }, .probe = adau1761_i2c_probe, .remove = adau1761_i2c_remove, diff --git a/sound/soc/codecs/adau1781-i2c.c b/sound/soc/codecs/adau1781-i2c.c index 2ce4362ccec1..0e32bba92339 100644 --- a/sound/soc/codecs/adau1781-i2c.c +++ b/sound/soc/codecs/adau1781-i2c.c @@ -45,7 +45,6 @@ MODULE_DEVICE_TABLE(i2c, adau1781_i2c_ids); static struct i2c_driver adau1781_i2c_driver = { .driver = { .name = "adau1781", - .owner = THIS_MODULE, }, .probe = adau1781_i2c_probe, .remove = adau1781_i2c_remove, diff --git a/sound/soc/codecs/adau1977-i2c.c b/sound/soc/codecs/adau1977-i2c.c index 9700e8c838c9..21e7394a972a 100644 --- a/sound/soc/codecs/adau1977-i2c.c +++ b/sound/soc/codecs/adau1977-i2c.c @@ -46,7 +46,6 @@ MODULE_DEVICE_TABLE(i2c, adau1977_i2c_ids); static struct i2c_driver adau1977_i2c_driver = { .driver = { .name = "adau1977", - .owner = THIS_MODULE, }, .probe = adau1977_i2c_probe, .remove = adau1977_i2c_remove, diff --git a/sound/soc/codecs/adav803.c b/sound/soc/codecs/adav803.c index 66d9fce34e62..52881faedcf6 100644 --- a/sound/soc/codecs/adav803.c +++ b/sound/soc/codecs/adav803.c @@ -36,7 +36,6 @@ static int adav803_remove(struct i2c_client *client) static struct i2c_driver adav803_driver = { .driver = { .name = "adav803", - .owner = THIS_MODULE, }, .probe = adav803_probe, .remove = adav803_remove, diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 8670861e5bec..54428c64387b 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -444,7 +444,6 @@ MODULE_DEVICE_TABLE(i2c, ak4535_i2c_id); static struct i2c_driver ak4535_i2c_driver = { .driver = { .name = "ak4535", - .owner = THIS_MODULE, }, .probe = ak4535_i2c_probe, .remove = ak4535_i2c_remove, diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 2d0ff4595ea0..b14176f8d884 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -609,7 +609,6 @@ MODULE_DEVICE_TABLE(i2c, ak4641_i2c_id); static struct i2c_driver ak4641_i2c_driver = { .driver = { .name = "ak4641", - .owner = THIS_MODULE, }, .probe = ak4641_i2c_probe, .remove = ak4641_i2c_remove, diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index fe963e17ceb4..4a90143d0e90 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -645,7 +645,6 @@ MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); static struct i2c_driver ak4642_i2c_driver = { .driver = { .name = "ak4642-codec", - .owner = THIS_MODULE, .of_match_table = ak4642_of_match, }, .probe = ak4642_i2c_probe, diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 0e59063aeb6f..c73a9f6914b6 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -663,7 +663,6 @@ MODULE_DEVICE_TABLE(i2c, ak4671_i2c_id); static struct i2c_driver ak4671_i2c_driver = { .driver = { .name = "ak4671-codec", - .owner = THIS_MODULE, }, .probe = ak4671_i2c_probe, .remove = ak4671_i2c_remove, diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 0fc24e0d518c..cf99c4e90acc 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -1085,7 +1085,6 @@ MODULE_DEVICE_TABLE(of, alc5623_of_match); static struct i2c_driver alc5623_i2c_driver = { .driver = { .name = "alc562x-codec", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(alc5623_of_match), }, .probe = alc5623_i2c_probe, diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 29a26455ce25..ef6de511dc7e 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1183,7 +1183,6 @@ MODULE_DEVICE_TABLE(of, alc5632_of_match); static struct i2c_driver alc5632_i2c_driver = { .driver = { .name = "alc5632", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(alc5632_of_match), }, .probe = alc5632_i2c_probe, diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 8f40025b7e7c..76564dc752d3 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -607,7 +607,6 @@ MODULE_DEVICE_TABLE(i2c, cs35l32_id); static struct i2c_driver cs35l32_i2c_driver = { .driver = { .name = "cs35l32", - .owner = THIS_MODULE, .pm = &cs35l32_runtime_pm, .of_match_table = cs35l32_of_match, }, diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index d7ec4756e45b..d1a77c7489d8 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -658,7 +658,6 @@ MODULE_DEVICE_TABLE(i2c, cs4265_id); static struct i2c_driver cs4265_i2c_driver = { .driver = { .name = "cs4265", - .owner = THIS_MODULE, .of_match_table = cs4265_of_match, }, .id_table = cs4265_id, diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index e6d4ff9fd992..e07807d96b68 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -751,7 +751,6 @@ MODULE_DEVICE_TABLE(i2c, cs4270_id); static struct i2c_driver cs4270_i2c_driver = { .driver = { .name = "cs4270", - .owner = THIS_MODULE, .of_match_table = cs4270_of_match, }, .id_table = cs4270_id, diff --git a/sound/soc/codecs/cs4271-i2c.c b/sound/soc/codecs/cs4271-i2c.c index b264da030340..dcb3223d7d8f 100644 --- a/sound/soc/codecs/cs4271-i2c.c +++ b/sound/soc/codecs/cs4271-i2c.c @@ -48,7 +48,6 @@ MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id); static struct i2c_driver cs4271_i2c_driver = { .driver = { .name = "cs4271", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(cs4271_dt_ids), }, .probe = cs4271_i2c_probe, diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c index c40428f25ba5..9bad478474fa 100644 --- a/sound/soc/codecs/cs42l51-i2c.c +++ b/sound/soc/codecs/cs42l51-i2c.c @@ -45,7 +45,6 @@ static int cs42l51_i2c_remove(struct i2c_client *i2c) static struct i2c_driver cs42l51_i2c_driver = { .driver = { .name = "cs42l51", - .owner = THIS_MODULE, .of_match_table = cs42l51_of_match, }, .probe = cs42l51_i2c_probe, diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 4de52c9957ac..b82d8e5b76ed 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1285,7 +1285,6 @@ MODULE_DEVICE_TABLE(i2c, cs42l52_id); static struct i2c_driver cs42l52_i2c_driver = { .driver = { .name = "cs42l52", - .owner = THIS_MODULE, .of_match_table = cs42l52_of_match, }, .id_table = cs42l52_id, diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 1e11ba45a79f..4ae793365985 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -1408,7 +1408,6 @@ MODULE_DEVICE_TABLE(i2c, cs42l56_id); static struct i2c_driver cs42l56_i2c_driver = { .driver = { .name = "cs42l56", - .owner = THIS_MODULE, .of_match_table = cs42l56_of_match, }, .id_table = cs42l56_id, diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index b7853b9d3a60..7cb1d7091dae 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1491,7 +1491,6 @@ MODULE_DEVICE_TABLE(i2c, cs42l73_id); static struct i2c_driver cs42l73_i2c_driver = { .driver = { .name = "cs42l73", - .owner = THIS_MODULE, .of_match_table = cs42l73_of_match, }, .id_table = cs42l73_id, diff --git a/sound/soc/codecs/cs42xx8-i2c.c b/sound/soc/codecs/cs42xx8-i2c.c index 5a71c9e5b441..800c1d549347 100644 --- a/sound/soc/codecs/cs42xx8-i2c.c +++ b/sound/soc/codecs/cs42xx8-i2c.c @@ -49,7 +49,6 @@ MODULE_DEVICE_TABLE(i2c, cs42xx8_i2c_id); static struct i2c_driver cs42xx8_i2c_driver = { .driver = { .name = "cs42xx8", - .owner = THIS_MODULE, .pm = &cs42xx8_pm, .of_match_table = cs42xx8_of_match, }, diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a44ead615b67..457ed82d3e10 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1259,7 +1259,6 @@ MODULE_DEVICE_TABLE(i2c, da7210_i2c_id); static struct i2c_driver da7210_i2c_driver = { .driver = { .name = "da7210", - .owner = THIS_MODULE, }, .probe = da7210_i2c_probe, .remove = da7210_i2c_remove, diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index baa36f6fd405..47fc3bec8a9c 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1585,7 +1585,6 @@ MODULE_DEVICE_TABLE(i2c, da7213_i2c_id); static struct i2c_driver da7213_i2c_driver = { .driver = { .name = "da7213", - .owner = THIS_MODULE, }, .probe = da7213_i2c_probe, .remove = da7213_remove, diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index b1cb4656a49c..0868568429a8 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1572,7 +1572,6 @@ MODULE_DEVICE_TABLE(i2c, da732x_i2c_id); static struct i2c_driver da732x_i2c_driver = { .driver = { .name = "da7320", - .owner = THIS_MODULE, }, .probe = da732x_i2c_probe, .remove = da732x_i2c_remove, diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 925d82278249..04c76f37e1fd 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1538,7 +1538,6 @@ static const struct of_device_id da9055_of_match[] = { static struct i2c_driver da9055_i2c_driver = { .driver = { .name = "da9055-codec", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(da9055_of_match), }, .probe = da9055_i2c_probe, diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index ac7dfe79c53b..34dc0f362482 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1149,7 +1149,6 @@ MODULE_DEVICE_TABLE(i2c, isabelle_i2c_id); static struct i2c_driver isabelle_i2c_driver = { .driver = { .name = "isabelle", - .owner = THIS_MODULE, }, .probe = isabelle_i2c_probe, .remove = isabelle_i2c_remove, diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 99ffc49aa779..558de1053f73 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -142,7 +142,6 @@ MODULE_DEVICE_TABLE(i2c, lm4857_i2c_id); static struct i2c_driver lm4857_i2c_driver = { .driver = { .name = "lm4857", - .owner = THIS_MODULE, }, .probe = lm4857_i2c_probe, .id_table = lm4857_i2c_id, diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index ef432aa8c639..6d3e8dbe34f5 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1456,7 +1456,6 @@ MODULE_DEVICE_TABLE(i2c, lm49453_i2c_id); static struct i2c_driver lm49453_i2c_driver = { .driver = { .name = "lm49453", - .owner = THIS_MODULE, }, .probe = lm49453_i2c_probe, .remove = lm49453_i2c_remove, diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index e3ccea229160..f987497a8411 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -242,7 +242,6 @@ MODULE_DEVICE_TABLE(i2c, max9768_i2c_id); static struct i2c_driver max9768_i2c_driver = { .driver = { .name = "max9768", - .owner = THIS_MODULE, }, .probe = max9768_i2c_probe, .remove = max9768_i2c_remove, diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index d0f45348bfbb..99c2daa0eebf 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -2011,7 +2011,6 @@ MODULE_DEVICE_TABLE(i2c, max98088_i2c_id); static struct i2c_driver max98088_i2c_driver = { .driver = { .name = "max98088", - .owner = THIS_MODULE, }, .probe = max98088_i2c_probe, .remove = max98088_i2c_remove, diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 1697340c3842..af6d282c5e2d 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2717,7 +2717,6 @@ MODULE_DEVICE_TABLE(acpi, max98090_acpi_match); static struct i2c_driver max98090_i2c_driver = { .driver = { .name = "max98090", - .owner = THIS_MODULE, .pm = &max98090_pm, .of_match_table = of_match_ptr(max98090_of_match), .acpi_match_table = ACPI_PTR(max98090_acpi_match), diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 66afd2b781e9..7e3814e65c87 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2434,7 +2434,6 @@ MODULE_DEVICE_TABLE(of, max98095_of_match); static struct i2c_driver max98095_i2c_driver = { .driver = { .name = "max98095", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(max98095_of_match), }, .probe = max98095_i2c_probe, diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 481d58f1cb3f..7c990521638c 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -352,7 +352,6 @@ MODULE_DEVICE_TABLE(i2c, max9850_i2c_id); static struct i2c_driver max9850_i2c_driver = { .driver = { .name = "max9850", - .owner = THIS_MODULE, }, .probe = max9850_i2c_probe, .remove = max9850_i2c_remove, diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 8df99fb69ab1..e1df06fab756 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -174,7 +174,6 @@ MODULE_DEVICE_TABLE(i2c, max9877_i2c_id); static struct i2c_driver max9877_i2c_driver = { .driver = { .name = "max9877", - .owner = THIS_MODULE, }, .probe = max9877_i2c_probe, .remove = max9877_i2c_remove, diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index aad664225dc3..ce551eecbf95 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -639,7 +639,6 @@ MODULE_DEVICE_TABLE(of, max98925_of_match); static struct i2c_driver max98925_i2c_driver = { .driver = { .name = "max98925", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(max98925_of_match), .pm = NULL, }, diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 1e8ede8454af..f561c78b9e0e 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -597,7 +597,6 @@ MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id); static struct i2c_driver ml26124_i2c_driver = { .driver = { .name = "ml26124", - .owner = THIS_MODULE, }, .probe = ml26124_i2c_probe, .remove = ml26124_i2c_remove, diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 477e13d30971..b2c990f08aa3 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -330,7 +330,6 @@ static int pcm1681_i2c_remove(struct i2c_client *client) static struct i2c_driver pcm1681_i2c_driver = { .driver = { .name = "pcm1681", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(pcm1681_dt_ids), }, .id_table = pcm1681_i2c_id, diff --git a/sound/soc/codecs/pcm512x-i2c.c b/sound/soc/codecs/pcm512x-i2c.c index dcdfac0ffeb1..dbff416e38be 100644 --- a/sound/soc/codecs/pcm512x-i2c.c +++ b/sound/soc/codecs/pcm512x-i2c.c @@ -67,7 +67,6 @@ static struct i2c_driver pcm512x_i2c_driver = { .id_table = pcm512x_i2c_id, .driver = { .name = "pcm512x", - .owner = THIS_MODULE, .of_match_table = pcm512x_of_match, .pm = &pcm512x_pm_ops, }, diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 76d88bad195c..bd9365885f73 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1259,7 +1259,6 @@ static int rt286_i2c_remove(struct i2c_client *i2c) static struct i2c_driver rt286_i2c_driver = { .driver = { .name = "rt286", - .owner = THIS_MODULE, .acpi_match_table = ACPI_PTR(rt286_acpi_match), }, .probe = rt286_i2c_probe, diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 7c4bcb65ef2c..75e5679dfef8 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1259,7 +1259,6 @@ static int rt298_i2c_remove(struct i2c_client *i2c) static struct i2c_driver rt298_i2c_driver = { .driver = { .name = "rt298", - .owner = THIS_MODULE, .acpi_match_table = ACPI_PTR(rt298_acpi_match), }, .probe = rt298_i2c_probe, diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 058167c80d71..d8771020299d 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1725,7 +1725,6 @@ static int rt5631_i2c_remove(struct i2c_client *client) static struct i2c_driver rt5631_i2c_driver = { .driver = { .name = "rt5631", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(rt5631_i2c_dt_ids), }, .probe = rt5631_i2c_probe, diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 0cbdac653041..187470bf926d 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2242,7 +2242,6 @@ static int rt5640_i2c_remove(struct i2c_client *i2c) static struct i2c_driver rt5640_i2c_driver = { .driver = { .name = "rt5640", - .owner = THIS_MODULE, .acpi_match_table = ACPI_PTR(rt5640_acpi_match), .of_match_table = of_match_ptr(rt5640_of_match), }, diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9dfa431b337d..b4e9b6b60778 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3455,7 +3455,6 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) static struct i2c_driver rt5645_i2c_driver = { .driver = { .name = "rt5645", - .owner = THIS_MODULE, .acpi_match_table = ACPI_PTR(rt5645_acpi_match), }, .probe = rt5645_i2c_probe, diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 9f3bc5a5b7b9..2c9eb8983042 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1806,7 +1806,6 @@ static int rt5651_i2c_remove(struct i2c_client *i2c) static struct i2c_driver rt5651_i2c_driver = { .driver = { .name = "rt5651", - .owner = THIS_MODULE, }, .probe = rt5651_i2c_probe, .remove = rt5651_i2c_remove, diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index a791d7e133f2..e84e223b4b86 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -3043,7 +3043,6 @@ static int rt5670_i2c_remove(struct i2c_client *i2c) static struct i2c_driver rt5670_i2c_driver = { .driver = { .name = "rt5670", - .owner = THIS_MODULE, .acpi_match_table = ACPI_PTR(rt5670_acpi_match), }, .probe = rt5670_i2c_probe, diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 13b871f11023..c69351dfa03a 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5241,7 +5241,6 @@ static int rt5677_i2c_remove(struct i2c_client *i2c) static struct i2c_driver rt5677_i2c_driver = { .driver = { .name = "rt5677", - .owner = THIS_MODULE, }, .probe = rt5677_i2c_probe, .remove = rt5677_i2c_remove, diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index e673f6ceb521..f3e646757215 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1601,7 +1601,6 @@ MODULE_DEVICE_TABLE(of, sgtl5000_dt_ids); static struct i2c_driver sgtl5000_i2c_driver = { .driver = { .name = "sgtl5000", - .owner = THIS_MODULE, .of_match_table = sgtl5000_dt_ids, }, .probe = sgtl5000_i2c_probe, diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index f30de7639bb9..5d94d6c7ad33 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -815,7 +815,6 @@ MODULE_DEVICE_TABLE(i2c, ssm2518_i2c_ids); static struct i2c_driver ssm2518_driver = { .driver = { .name = "ssm2518", - .owner = THIS_MODULE, }, .probe = ssm2518_i2c_probe, .remove = ssm2518_i2c_remove, diff --git a/sound/soc/codecs/ssm2602-i2c.c b/sound/soc/codecs/ssm2602-i2c.c index 0d9779d6bfda..173ba85ff59e 100644 --- a/sound/soc/codecs/ssm2602-i2c.c +++ b/sound/soc/codecs/ssm2602-i2c.c @@ -52,7 +52,6 @@ MODULE_DEVICE_TABLE(of, ssm2602_of_match); static struct i2c_driver ssm2602_i2c_driver = { .driver = { .name = "ssm2602", - .owner = THIS_MODULE, .of_match_table = ssm2602_of_match, }, .probe = ssm2602_i2c_probe, diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index 938d2cb6d78b..f3f1f6874c72 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -453,7 +453,6 @@ MODULE_DEVICE_TABLE(i2c, ssm4567_i2c_ids); static struct i2c_driver ssm4567_driver = { .driver = { .name = "ssm4567", - .owner = THIS_MODULE, }, .probe = ssm4567_i2c_probe, .remove = ssm4567_i2c_remove, diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 60eff36260cb..a9844b2ac829 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -1144,7 +1144,6 @@ MODULE_DEVICE_TABLE(i2c, sta32x_i2c_id); static struct i2c_driver sta32x_i2c_driver = { .driver = { .name = "sta32x", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(st32x_dt_ids), }, .probe = sta32x_i2c_probe, diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c index bd819a3f205a..33a4612f0a07 100644 --- a/sound/soc/codecs/sta350.c +++ b/sound/soc/codecs/sta350.c @@ -1264,7 +1264,6 @@ MODULE_DEVICE_TABLE(i2c, sta350_i2c_id); static struct i2c_driver sta350_i2c_driver = { .driver = { .name = "sta350", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(st350_dt_ids), }, .probe = sta350_i2c_probe, diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 4f70378b2cfb..3430f444c1ae 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -379,7 +379,6 @@ MODULE_DEVICE_TABLE(i2c, sta529_i2c_id); static struct i2c_driver sta529_i2c_driver = { .driver = { .name = "sta529", - .owner = THIS_MODULE, }, .probe = sta529_i2c_probe, .remove = sta529_i2c_remove, diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 71cfae0de3ee..d36afcbbd354 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -769,7 +769,6 @@ MODULE_DEVICE_TABLE(of, tas2552_of_match); static struct i2c_driver tas2552_i2c_driver = { .driver = { .name = "tas2552", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(tas2552_of_match), .pm = &tas2552_pm, }, diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 32942bed34b1..0a49fc851577 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -994,7 +994,6 @@ static int tas5086_i2c_remove(struct i2c_client *i2c) static struct i2c_driver tas5086_i2c_driver = { .driver = { .name = "tas5086", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(tas5086_dt_ids), }, .id_table = tas5086_i2c_id, diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c index bf7596ba49f3..cb5310d89c0f 100644 --- a/sound/soc/codecs/tfa9879.c +++ b/sound/soc/codecs/tfa9879.c @@ -314,7 +314,6 @@ MODULE_DEVICE_TABLE(i2c, tfa9879_i2c_id); static struct i2c_driver tfa9879_i2c_driver = { .driver = { .name = "tfa9879", - .owner = THIS_MODULE, }, .probe = tfa9879_i2c_probe, .remove = tfa9879_i2c_remove, diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index c4c960f592a1..48dd9b2cd0fa 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1283,7 +1283,6 @@ MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); static struct i2c_driver aic31xx_i2c_driver = { .driver = { .name = "tlv320aic31xx-codec", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(tlv320aic31xx_of_match), }, .probe = aic31xx_i2c_probe, diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index ad6cb90e5f9b..f2d3191961e1 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -871,7 +871,6 @@ MODULE_DEVICE_TABLE(of, aic32x4_of_id); static struct i2c_driver aic32x4_i2c_driver = { .driver = { .name = "tlv320aic32x4", - .owner = THIS_MODULE, .of_match_table = aic32x4_of_id, }, .probe = aic32x4_i2c_probe, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index a7cf19b53fb2..125a93517cdb 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1825,7 +1825,6 @@ MODULE_DEVICE_TABLE(of, tlv320aic3x_of_match); static struct i2c_driver aic3x_i2c_driver = { .driver = { .name = "tlv320aic3x-codec", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(tlv320aic3x_of_match), }, .probe = aic3x_i2c_probe, diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index d67a311f0e75..781398fb2841 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1585,7 +1585,6 @@ MODULE_DEVICE_TABLE(i2c, tlv320dac33_i2c_id); static struct i2c_driver tlv320dac33_i2c_driver = { .driver = { .name = "tlv320dac33-codec", - .owner = THIS_MODULE, }, .probe = dac33_i2c_probe, .remove = dac33_i2c_remove, diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 6fac9e034c48..265c4c38804d 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -488,7 +488,6 @@ MODULE_DEVICE_TABLE(of, tpa6130a2_of_match); static struct i2c_driver tpa6130a2_i2c_driver = { .driver = { .name = "tpa6130a2", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(tpa6130a2_of_match), }, .probe = tpa6130a2_probe, diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index ffc6f3028cc5..43568435c208 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -377,7 +377,6 @@ MODULE_DEVICE_TABLE(of, ts3a227e_of_match); static struct i2c_driver ts3a227e_driver = { .driver = { .name = "ts3a227e", - .owner = THIS_MODULE, .pm = &ts3a227e_pm, .of_match_table = of_match_ptr(ts3a227e_of_match), }, diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 6e159f59d219..5cafb16ccf1a 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -810,7 +810,6 @@ MODULE_DEVICE_TABLE(i2c, uda1380_i2c_id); static struct i2c_driver uda1380_i2c_driver = { .driver = { .name = "uda1380-codec", - .owner = THIS_MODULE, }, .probe = uda1380_i2c_probe, .remove = uda1380_i2c_remove, diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index 048f00568260..ec45c5b220b1 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -251,7 +251,6 @@ MODULE_DEVICE_TABLE(i2c, wm1250_ev1_i2c_id); static struct i2c_driver wm1250_ev1_i2c_driver = { .driver = { .name = "wm1250-ev1", - .owner = THIS_MODULE, }, .probe = wm1250_ev1_probe, .remove = wm1250_ev1_remove, diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 21d5402e343f..786abd02b140 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -942,7 +942,6 @@ MODULE_DEVICE_TABLE(i2c, wm2000_i2c_id); static struct i2c_driver wm2000_i2c_driver = { .driver = { .name = "wm2000", - .owner = THIS_MODULE, }, .probe = wm2000_i2c_probe, .remove = wm2000_i2c_remove, diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 01aeb75dd774..cdaa175283df 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2495,7 +2495,6 @@ MODULE_DEVICE_TABLE(i2c, wm2200_i2c_id); static struct i2c_driver wm2200_i2c_driver = { .driver = { .name = "wm2200", - .owner = THIS_MODULE, .pm = &wm2200_pm, }, .probe = wm2200_i2c_probe, diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 98495dd61239..e5572d28395d 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2720,7 +2720,6 @@ MODULE_DEVICE_TABLE(i2c, wm5100_i2c_id); static struct i2c_driver wm5100_i2c_driver = { .driver = { .name = "wm5100", - .owner = THIS_MODULE, .pm = &wm5100_pm, }, .probe = wm5100_i2c_probe, diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index dac5beb4d023..3cff5a699e57 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -690,7 +690,6 @@ MODULE_DEVICE_TABLE(i2c, wm8510_i2c_id); static struct i2c_driver wm8510_i2c_driver = { .driver = { .name = "wm8510", - .owner = THIS_MODULE, .of_match_table = wm8510_of_match, }, .probe = wm8510_i2c_probe, diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 43ea8ae5f871..5f8fde56e68b 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -534,7 +534,6 @@ MODULE_DEVICE_TABLE(i2c, wm8523_i2c_id); static struct i2c_driver wm8523_i2c_driver = { .driver = { .name = "wm8523", - .owner = THIS_MODULE, .of_match_table = wm8523_of_match, }, .probe = wm8523_i2c_probe, diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 759a7928ac3e..abf60355f7f7 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -978,7 +978,6 @@ MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id); static struct i2c_driver wm8580_i2c_driver = { .driver = { .name = "wm8580", - .owner = THIS_MODULE, .of_match_table = wm8580_of_match, }, .probe = wm8580_i2c_probe, diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index cc8251f09f8a..44b9e0ae7451 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -478,7 +478,6 @@ MODULE_DEVICE_TABLE(i2c, wm8711_i2c_id); static struct i2c_driver wm8711_i2c_driver = { .driver = { .name = "wm8711", - .owner = THIS_MODULE, .of_match_table = wm8711_of_match, }, .probe = wm8711_i2c_probe, diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index f1a173e6ec33..cd7b02413ccf 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -319,7 +319,6 @@ MODULE_DEVICE_TABLE(i2c, wm8728_i2c_id); static struct i2c_driver wm8728_i2c_driver = { .driver = { .name = "wm8728", - .owner = THIS_MODULE, .of_match_table = wm8728_of_match, }, .probe = wm8728_i2c_probe, diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 628d50c40cc4..c8622329f86d 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -802,7 +802,6 @@ MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id); static struct i2c_driver wm8731_i2c_driver = { .driver = { .name = "wm8731", - .owner = THIS_MODULE, .of_match_table = wm8731_of_match, }, .probe = wm8731_i2c_probe, diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 6ad606fd8b69..b54a7b78f562 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -657,7 +657,6 @@ MODULE_DEVICE_TABLE(i2c, wm8737_i2c_id); static struct i2c_driver wm8737_i2c_driver = { .driver = { .name = "wm8737", - .owner = THIS_MODULE, .of_match_table = wm8737_of_match, }, .probe = wm8737_i2c_probe, diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index b34623786e35..430fa7ded713 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -633,7 +633,6 @@ MODULE_DEVICE_TABLE(i2c, wm8741_i2c_id); static struct i2c_driver wm8741_i2c_driver = { .driver = { .name = "wm8741", - .owner = THIS_MODULE, .of_match_table = wm8741_of_match, }, .probe = wm8741_i2c_probe, diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 56d89b0865fa..873933a7966f 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -826,7 +826,6 @@ MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id); static struct i2c_driver wm8750_i2c_driver = { .driver = { .name = "wm8750", - .owner = THIS_MODULE, .of_match_table = wm8750_of_match, }, .probe = wm8750_i2c_probe, diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index feb2997a377a..b5e50ffbf99f 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1609,7 +1609,6 @@ MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id); static struct i2c_driver wm8753_i2c_driver = { .driver = { .name = "wm8753", - .owner = THIS_MODULE, .of_match_table = wm8753_of_match, }, .probe = wm8753_i2c_probe, diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index ece9b4456767..592866dd60c7 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -536,7 +536,6 @@ MODULE_DEVICE_TABLE(i2c, wm8776_i2c_id); static struct i2c_driver wm8776_i2c_driver = { .driver = { .name = "wm8776", - .owner = THIS_MODULE, .of_match_table = wm8776_of_match, }, .probe = wm8776_i2c_probe, diff --git a/sound/soc/codecs/wm8804-i2c.c b/sound/soc/codecs/wm8804-i2c.c index 6596f5f3a0c3..f27464c2c5ba 100644 --- a/sound/soc/codecs/wm8804-i2c.c +++ b/sound/soc/codecs/wm8804-i2c.c @@ -49,7 +49,6 @@ MODULE_DEVICE_TABLE(of, wm8804_of_match); static struct i2c_driver wm8804_i2c_driver = { .driver = { .name = "wm8804", - .owner = THIS_MODULE, .pm = &wm8804_pm, .of_match_table = wm8804_of_match, }, diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index f3759ec5a863..98900aa66dc3 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1312,7 +1312,6 @@ MODULE_DEVICE_TABLE(i2c, wm8900_i2c_id); static struct i2c_driver wm8900_i2c_driver = { .driver = { .name = "wm8900", - .owner = THIS_MODULE, }, .probe = wm8900_i2c_probe, .remove = wm8900_i2c_remove, diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b5322c1544fb..b011253459af 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2193,7 +2193,6 @@ MODULE_DEVICE_TABLE(i2c, wm8903_i2c_id); static struct i2c_driver wm8903_i2c_driver = { .driver = { .name = "wm8903", - .owner = THIS_MODULE, .of_match_table = wm8903_of_match, }, .probe = wm8903_i2c_probe, diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 07dc40022d4f..b783743dc97e 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2294,7 +2294,6 @@ MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id); static struct i2c_driver wm8904_i2c_driver = { .driver = { .name = "wm8904", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(wm8904_of_match), }, .probe = wm8904_i2c_probe, diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 98ef0ba5c2a4..f6f9395ea38e 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -787,7 +787,6 @@ MODULE_DEVICE_TABLE(i2c, wm8940_i2c_id); static struct i2c_driver wm8940_i2c_driver = { .driver = { .name = "wm8940", - .owner = THIS_MODULE, }, .probe = wm8940_i2c_probe, .remove = wm8940_i2c_remove, diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 2d591c24704b..12e4435f00f8 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -1009,7 +1009,6 @@ MODULE_DEVICE_TABLE(i2c, wm8955_i2c_id); static struct i2c_driver wm8955_i2c_driver = { .driver = { .name = "wm8955", - .owner = THIS_MODULE, }, .probe = wm8955_i2c_probe, .remove = wm8955_i2c_remove, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 94c5c4681ce5..1ed0720b41f0 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1216,7 +1216,6 @@ MODULE_DEVICE_TABLE(of, wm8960_of_match); static struct i2c_driver wm8960_i2c_driver = { .driver = { .name = "wm8960", - .owner = THIS_MODULE, .of_match_table = wm8960_of_match, }, .probe = wm8960_i2c_probe, diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index a057662632ff..20a010344fd5 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -982,7 +982,6 @@ MODULE_DEVICE_TABLE(i2c, wm8961_i2c_id); static struct i2c_driver wm8961_i2c_driver = { .driver = { .name = "wm8961", - .owner = THIS_MODULE, }, .probe = wm8961_i2c_probe, .remove = wm8961_i2c_remove, diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8edc6d203404..c67e3047f022 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3878,7 +3878,6 @@ MODULE_DEVICE_TABLE(of, wm8962_of_match); static struct i2c_driver wm8962_i2c_driver = { .driver = { .name = "wm8962", - .owner = THIS_MODULE, .of_match_table = wm8962_of_match, .pm = &wm8962_pm, }, diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index b51184c4e816..2cdde32c43c6 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -710,7 +710,6 @@ MODULE_DEVICE_TABLE(i2c, wm8971_i2c_id); static struct i2c_driver wm8971_i2c_driver = { .driver = { .name = "wm8971", - .owner = THIS_MODULE, }, .probe = wm8971_i2c_probe, .remove = wm8971_i2c_remove, diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 33b16a7ba82e..0a60677397b3 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -634,7 +634,6 @@ MODULE_DEVICE_TABLE(i2c, wm8974_i2c_id); static struct i2c_driver wm8974_i2c_driver = { .driver = { .name = "wm8974", - .owner = THIS_MODULE, }, .probe = wm8974_i2c_probe, .remove = wm8974_i2c_remove, diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index cfc8cdf49970..d36d6001fbb7 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -1072,7 +1072,6 @@ MODULE_DEVICE_TABLE(i2c, wm8978_i2c_id); static struct i2c_driver wm8978_i2c_driver = { .driver = { .name = "wm8978", - .owner = THIS_MODULE, }, .probe = wm8978_i2c_probe, .remove = wm8978_i2c_remove, diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 2fdd2c6cc09d..f6861cc4c321 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -1133,7 +1133,6 @@ MODULE_DEVICE_TABLE(i2c, wm8983_i2c_id); static struct i2c_driver wm8983_i2c_driver = { .driver = { .name = "wm8983", - .owner = THIS_MODULE, }, .probe = wm8983_i2c_probe, .remove = wm8983_i2c_remove, diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 8a85f5004d41..9c3c1517a4f3 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -1144,7 +1144,6 @@ MODULE_DEVICE_TABLE(i2c, wm8985_i2c_id); static struct i2c_driver wm8985_i2c_driver = { .driver = { .name = "wm8985", - .owner = THIS_MODULE, }, .probe = wm8985_i2c_probe, .remove = wm8985_i2c_remove, diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index f13a995af277..c88ce99ce9e1 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -919,7 +919,6 @@ MODULE_DEVICE_TABLE(i2c, wm8988_i2c_id); static struct i2c_driver wm8988_i2c_driver = { .driver = { .name = "wm8988", - .owner = THIS_MODULE, }, .probe = wm8988_i2c_probe, .remove = wm8988_i2c_remove, diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 1993fd2a6f15..83d72d8bc51b 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1356,7 +1356,6 @@ MODULE_DEVICE_TABLE(i2c, wm8990_i2c_id); static struct i2c_driver wm8990_i2c_driver = { .driver = { .name = "wm8990", - .owner = THIS_MODULE, }, .probe = wm8990_i2c_probe, .remove = wm8990_i2c_remove, diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 44a677720828..709d16e338d9 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1363,7 +1363,6 @@ MODULE_DEVICE_TABLE(i2c, wm8991_i2c_id); static struct i2c_driver wm8991_i2c_driver = { .driver = { .name = "wm8991", - .owner = THIS_MODULE, }, .probe = wm8991_i2c_probe, .remove = wm8991_i2c_remove, diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index f2d5211a9a96..9638accb74b9 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1742,7 +1742,6 @@ MODULE_DEVICE_TABLE(i2c, wm8993_i2c_id); static struct i2c_driver wm8993_i2c_driver = { .driver = { .name = "wm8993", - .owner = THIS_MODULE, }, .probe = wm8993_i2c_probe, .remove = wm8993_i2c_remove, diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 505b65f5734f..eda52a96c1fa 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2298,7 +2298,6 @@ MODULE_DEVICE_TABLE(i2c, wm8995_i2c_id); static struct i2c_driver wm8995_i2c_driver = { .driver = { .name = "wm8995", - .owner = THIS_MODULE, }, .probe = wm8995_i2c_probe, .remove = wm8995_i2c_remove, diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index c600c8cdba0b..2a696dbb73af 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -3098,7 +3098,6 @@ MODULE_DEVICE_TABLE(i2c, wm8996_i2c_id); static struct i2c_driver wm8996_i2c_driver = { .driver = { .name = "wm8996", - .owner = THIS_MODULE, }, .probe = wm8996_i2c_probe, .remove = wm8996_i2c_remove, diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index ec91ce8a7ca8..45223a20331f 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1378,7 +1378,6 @@ MODULE_DEVICE_TABLE(i2c, wm9081_i2c_id); static struct i2c_driver wm9081_i2c_driver = { .driver = { .name = "wm9081", - .owner = THIS_MODULE, }, .probe = wm9081_i2c_probe, .remove = wm9081_i2c_remove, diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 13d23fc797db..12baf725d4a5 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -636,7 +636,6 @@ MODULE_DEVICE_TABLE(i2c, wm9090_id); static struct i2c_driver wm9090_i2c_driver = { .driver = { .name = "wm9090", - .owner = THIS_MODULE, }, .probe = wm9090_i2c_probe, .remove = wm9090_i2c_remove, -- 1.9.1

2 4

[alsa-devel] [PATCH v2] ASoC: cs4349: Add support for Cirrus Logic CS4349
by timothyc.howe＠gmail.com 15 Jul '15

15 Jul '15

From: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- Documentation/devicetree/bindings/sound/cs4349.txt | 19 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4349.c | 414 +++++++++++++++++++++ sound/soc/codecs/cs4349.h | 146 ++++++++ 5 files changed, 587 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs4349.txt create mode 100644 sound/soc/codecs/cs4349.c create mode 100644 sound/soc/codecs/cs4349.h diff --git a/Documentation/devicetree/bindings/sound/cs4349.txt b/Documentation/devicetree/bindings/sound/cs4349.txt new file mode 100644 index 0000000..54c117b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4349.txt @@ -0,0 +1,19 @@ +CS4349 audio CODEC + +Required properties: + + - compatible : "cirrus,cs4349" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + +Example: + +codec: cs4349@48 { + compatible = "cirrus,cs4349"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7223c7f..4a78606 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -54,6 +54,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_I2C if I2C select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C + select SND_SOC_CS4349 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -413,6 +414,11 @@ config SND_SOC_CS42XX8_I2C select SND_SOC_CS42XX8 select REGMAP_I2C +# Cirrus Logic CS4349 HiFi DAC +config SND_SOC_CS4349 + tristate "Cirrus Logic CS4349 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9f7136c..36a9f37 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -46,6 +46,7 @@ snd-soc-cs4271-i2c-objs := cs4271-i2c.o snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o +snd-soc-cs4349-objs := cs4349.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -237,6 +238,7 @@ obj-$(CONFIG_SND_SOC_CS4271_I2C) += snd-soc-cs4271-i2c.o obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o +obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c new file mode 100644 index 0000000..ee6b676 --- /dev/null +++ b/sound/soc/codecs/cs4349.c @@ -0,0 +1,414 @@ +/* + * cs4349.c -- CS4349 ALSA Soc Audio driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Authors: Tim Howe <Tim.Howe(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs4349.h" + + +static const struct reg_default cs4349_reg_defaults[] = { + { 2, 0x00 }, /* r02 - Mode Control */ + { 3, 0x09 }, /* r03 - Volume, Mixing and Inversion Control */ + { 4, 0x81 }, /* r04 - Mute Control */ + { 5, 0x00 }, /* r05 - Channel A Volume Control */ + { 6, 0x00 }, /* r06 - Channel B Volume Control */ + { 7, 0xB1 }, /* r07 - Ramp and Filter Control */ + { 8, 0x1C }, /* r08 - Misc. Control */ +}; + +/* Private data for the CS4349 */ +struct cs4349_private { + struct regmap *regmap; + struct cs4349_platform_data pdata; + struct gpio_desc *reset_gpio; + unsigned int mclk; + unsigned int mode; + int rate; +}; + +static bool cs4349_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS4349_CHIPID: + case CS4349_MODE: + case CS4349_VMI: + case CS4349_MUTE: + case CS4349_VOLA: + case CS4349_VOLB: + case CS4349_RMPFLT: + case CS4349_MISC: + return true; + default: + return false; + } +} + +static int cs4349_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + + cs4349->mclk = freq; + + return 0; +} + +static int cs4349_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + unsigned int fmt; + + fmt = format & SND_SOC_DAIFMT_FORMAT_MASK; + + switch (fmt) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + cs4349->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs4349_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + int mode, fmt, ret; + + mode = snd_soc_read(codec, CS4349_MODE); + cs4349->rate = params_rate(params); + + switch (cs4349->mode) { + case SND_SOC_DAIFMT_I2S: + mode |= MODE_FORMAT(DIF_I2S); + break; + case SND_SOC_DAIFMT_LEFT_J: + mode |= MODE_FORMAT(DIF_LEFT_JST); + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_width(params)) { + case 16: + fmt = DIF_RGHT_JST16; + break; + case 24: + fmt = DIF_RGHT_JST24; + break; + default: + return -EINVAL; + } + mode |= MODE_FORMAT(fmt); + break; + default: + return -EINVAL; + } + + ret = snd_soc_write(codec, CS4349_MODE, mode); + if (ret < 0) + return ret; + + return 0; +} + +static int cs4349_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int reg; + + reg = 0; + if (mute) + reg = MUTE_AB_MASK; + + return snd_soc_update_bits(codec, CS4349_MUTE, MUTE_AB_MASK, reg); +} + +static DECLARE_TLV_DB_SCALE(dig_tlv, -12750, 50, 0); + +static const char * const chan_mix_texts[] = { + "Mute", "MuteA", "MuteA SwapB", "MuteA MonoB", "SwapA MuteB", + "BothR", "Swap", "SwapA MonoB", "MuteB", "Normal", "BothL", + "MonoB", "MonoA MuteB", "MonoA", "MonoA SwapB", "Mono", + /*Normal == Channel A = Left, Channel B = Right*/ +}; + +static const char * const fm_texts[] = { + "Auto", "Single", "Double", "Quad", +}; + +static const char * const deemph_texts[] = { + "None", "44.1k", "48k", "32k", +}; + +static const char * const softr_zeroc_texts[] = { + "Immediate", "Zero Cross", "Soft Ramp", "SR on ZC", +}; + +static int deemph_values[] = { + 0, 4, 8, 12, +}; + +static int softr_zeroc_values[] = { + 0, 64, 128, 192, +}; + +static const struct soc_enum chan_mix_enum = + SOC_ENUM_SINGLE(CS4349_VMI, 0, + ARRAY_SIZE(chan_mix_texts), + chan_mix_texts); + +static const struct soc_enum fm_mode_enum = + SOC_ENUM_SINGLE(CS4349_MODE, 0, + ARRAY_SIZE(fm_texts), + fm_texts); + +static SOC_VALUE_ENUM_SINGLE_DECL(deemph_enum, CS4349_MODE, 0, DEM_MASK, + deemph_texts, deemph_values); + +static SOC_VALUE_ENUM_SINGLE_DECL(softr_zeroc_enum, CS4349_RMPFLT, 0, + SR_ZC_MASK, softr_zeroc_texts, + softr_zeroc_values); + +static const struct snd_kcontrol_new cs4349_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", + CS4349_VOLA, CS4349_VOLB, 0, 0xFF, 1, dig_tlv), + SOC_ENUM("Functional Mode", fm_mode_enum), + SOC_ENUM("De-Emphasis Control", deemph_enum), + SOC_ENUM("Soft Ramp Zero Cross Control", softr_zeroc_enum), + SOC_ENUM("Channel Mixer", chan_mix_enum), + SOC_SINGLE("VolA = VolB Switch", CS4349_VMI, 7, 1, 0), + SOC_SINGLE("InvertA Switch", CS4349_VMI, 6, 1, 0), + SOC_SINGLE("InvertB Switch", CS4349_VMI, 5, 1, 0), + SOC_SINGLE("Auto-Mute Switch", CS4349_MUTE, 7, 1, 0), + SOC_SINGLE("MUTEC A = B Switch", CS4349_MUTE, 5, 1, 0), + SOC_SINGLE("Soft Ramp Up Switch", CS4349_RMPFLT, 5, 1, 0), + SOC_SINGLE("Soft Ramp Down Switch", CS4349_RMPFLT, 4, 1, 0), + SOC_SINGLE("Slow Roll Off Filter Switch", CS4349_RMPFLT, 2, 1, 0), + SOC_SINGLE("Freeze Switch", CS4349_MISC, 5, 1, 0), + SOC_SINGLE("Popguard Switch", CS4349_MISC, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget cs4349_dapm_widgets[] = { + SND_SOC_DAPM_DAC("HiFi DAC", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_OUTPUT("OutputA"), + SND_SOC_DAPM_OUTPUT("OutputB"), +}; + +static const struct snd_soc_dapm_route cs4349_routes[] = { + {"DAC Playback", NULL, "OutputA"}, + {"DAC Playback", NULL, "OutputB"}, + + {"OutputA", NULL, "HiFi DAC"}, + {"OutputB", NULL, "HiFi DAC"}, +}; + +#define CS4349_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) + +#define CS4349_PCM_RATES SNDRV_PCM_RATE_8000_192000 + +static const struct snd_soc_dai_ops cs4349_dai_ops = { + .hw_params = cs4349_pcm_hw_params, + .set_fmt = cs4349_set_dai_fmt, + .digital_mute = cs4349_digital_mute, +}; + +static struct snd_soc_dai_driver cs4349_dai = { + .name = "cs4349_hifi", + .playback = { + .stream_name = "DAC Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS4349_PCM_RATES, + .formats = CS4349_PCM_FORMATS, + }, + .ops = &cs4349_dai_ops, + .symmetric_rates = 1, +}; + +static struct snd_soc_codec_driver soc_codec_dev_cs4349 = { + .set_sysclk = cs4349_codec_set_sysclk, + + .controls = cs4349_snd_controls, + .num_controls = ARRAY_SIZE(cs4349_snd_controls), + + .dapm_widgets = cs4349_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4349_dapm_widgets), + .dapm_routes = cs4349_routes, + .num_dapm_routes = ARRAY_SIZE(cs4349_routes), +}; + +static struct regmap_config cs4349_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS4349_NUMREGS, + .reg_defaults = cs4349_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4349_reg_defaults), + .readable_reg = cs4349_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs4349_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs4349_private *cs4349; + struct cs4349_platform_data *pdata = dev_get_platdata(&client->dev); + int ret = 0; + + cs4349 = devm_kzalloc(&client->dev, sizeof(*cs4349), GFP_KERNEL); + if (!cs4349) + return -ENOMEM; + + cs4349->regmap = devm_regmap_init_i2c(client, &cs4349_regmap); + if (IS_ERR(cs4349->regmap)) { + ret = PTR_ERR(cs4349->regmap); + dev_err(&client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + if (pdata) + cs4349->pdata = *pdata; + + /* Reset the Device */ + cs4349->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs4349->reset_gpio)) + return PTR_ERR(cs4349->reset_gpio); + + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + + i2c_set_clientdata(client, cs4349); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4349, + &cs4349_dai, 1); +} + +static int cs4349_i2c_remove(struct i2c_client *client) +{ + struct cs4349_private *cs4349 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + /* Hold down reset */ + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int cs4349_runtime_suspend(struct device *dev) +{ + struct cs4349_private *cs4349 = dev_get_drvdata(dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + int ret; + + ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 1); + if (ret < 0) + return ret; + + regcache_cache_only(cs4349->regmap, true); + regcache_sync(cs4349->regmap); + + /* Hold down reset */ + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + + return 0; +} + +static int cs4349_runtime_resume(struct device *dev) +{ + struct cs4349_private *cs4349 = dev_get_drvdata(dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + int ret; + + ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 0); + if (ret < 0) + return ret; + + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + + regcache_cache_only(cs4349->regmap, false); + regcache_sync(cs4349->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs4349_runtime_pm = { + SET_RUNTIME_PM_OPS(cs4349_runtime_suspend, cs4349_runtime_resume, + NULL) +}; + +static const struct of_device_id cs4349_of_match[] = { + { .compatible = "cirrus,cs4349", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs4349_of_match); + +static const struct i2c_device_id cs4349_i2c_id[] = { + {"cs4349", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs4349_i2c_id); + +static struct i2c_driver cs4349_i2c_driver = { + .driver = { + .name = "cs4349", + .owner = THIS_MODULE, + .of_match_table = cs4349_of_match, + }, + .id_table = cs4349_i2c_id, + .probe = cs4349_i2c_probe, + .remove = cs4349_i2c_remove, +}; + +module_i2c_driver(cs4349_i2c_driver); + +MODULE_AUTHOR("Tim Howe <tim.howe(a)cirrus.com>"); +MODULE_DESCRIPTION("Cirrus Logic CS4349 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4349.h b/sound/soc/codecs/cs4349.h new file mode 100644 index 0000000..3884a89 --- /dev/null +++ b/sound/soc/codecs/cs4349.h @@ -0,0 +1,146 @@ +/* + * ALSA SoC CS4349 codec driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Tim Howe <Tim.Howe(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + */ + +#ifndef __CS4349_H__ +#define __CS4349_H__ + +struct cs4349_platform_data { + + /* GPIO for Reset */ + unsigned int gpio_nreset; + +}; + +/* CS4349 registers addresses */ +#define CS4349_CHIPID 0x01 /* Device and Rev ID, Read Only */ +#define CS4349_MODE 0x02 /* Mode Control */ +#define CS4349_VMI 0x03 /* Volume, Mixing, Inversion Control */ +#define CS4349_MUTE 0x04 /* Mute Control */ +#define CS4349_VOLA 0x05 /* DAC Channel A Volume Control */ +#define CS4349_VOLB 0x06 /* DAC Channel B Volume Control */ +#define CS4349_RMPFLT 0x07 /* Ramp and Filter Control */ +#define CS4349_MISC 0x08 /* Power Down,Freeze Control,Pop Stop*/ + +#define CS4349_FIRSTREG 0x01 +#define CS4349_LASTREG 0x08 +#define CS4349_NUMREGS (CS4349_LASTREG - CS4349_FIRSTREG + 1) +#define CS4349_I2C_INCR 0x80 + + +/* Device and Revision ID */ +#define CS4349_REVA 0xF0 /* Rev A */ +#define CS4349_REVB 0xF1 /* Rev B */ +#define CS4349_REVC2 0xFF /* Rev C2 */ + + +/* PDN_DONE Poll Maximum + * If soft ramp is set it will take much longer to power down + * the system. + */ +#define PDN_POLL_MAX 900 + + +/* Bitfield Definitions */ + +/* CS4349_MODE */ +/* (Digital Interface Format, De-Emphasis Control, Functional Mode */ +#define DIF2 (1 << 6) +#define DIF1 (1 << 5) +#define DIF0 (1 << 4) +#define DEM1 (1 << 3) +#define DEM0 (1 << 2) +#define FM1 (1 << 1) +#define DIF_LEFT_JST 0x00 +#define DIF_I2S 0x01 +#define DIF_RGHT_JST16 0x02 +#define DIF_RGHT_JST24 0x03 +#define DIF_TDM0 0x04 +#define DIF_TDM1 0x05 +#define DIF_TDM2 0x06 +#define DIF_TDM3 0x07 +#define DIF_MASK 0x70 +#define MODE_FORMAT(x) (((x)&7)<<4) +#define DEM_MASK 0x0C +#define NO_DEM 0x00 +#define DEM_441 0x04 +#define DEM_48K 0x08 +#define DEM_32K 0x0C +#define FM_AUTO 0x00 +#define FM_SNGL 0x01 +#define FM_DBL 0x02 +#define FM_QUAD 0x03 +#define FM_SNGL_MIN 30000 +#define FM_SNGL_MAX 54000 +#define FM_DBL_MAX 108000 +#define FM_QUAD_MAX 216000 +#define FM_MASK 0x03 + +/* CS4349_VMI (VMI = Volume, Mixing and Inversion Controls) */ +#define VOLBISA (1 << 7) +#define VOLAISB (1 << 7) +/* INVERT_A only available for Left Jstfd, Right Jstfd16 and Right Jstfd24 */ +#define INVERT_A (1 << 6) +/* INVERT_B only available for Left Jstfd, Right Jstfd16 and Right Jstfd24 */ +#define INVERT_B (1 << 5) +#define ATAPI3 (1 << 3) +#define ATAPI2 (1 << 2) +#define ATAPI1 (1 << 1) +#define ATAPI0 (1 << 0) +#define MUTEAB 0x00 +#define MUTEA_RIGHTB 0x01 +#define MUTEA_LEFTB 0x02 +#define MUTEA_SUMLRDIV2B 0x03 +#define RIGHTA_MUTEB 0x04 +#define RIGHTA_RIGHTB 0x05 +#define RIGHTA_LEFTB 0x06 +#define RIGHTA_SUMLRDIV2B 0x07 +#define LEFTA_MUTEB 0x08 +#define LEFTA_RIGHTB 0x09 /* Default */ +#define LEFTA_LEFTB 0x0A +#define LEFTA_SUMLRDIV2B 0x0B +#define SUMLRDIV2A_MUTEB 0x0C +#define SUMLRDIV2A_RIGHTB 0x0D +#define SUMLRDIV2A_LEFTB 0x0E +#define SUMLRDIV2_AB 0x0F +#define CHMIX_MASK 0x0F + +/* CS4349_MUTE */ +#define AUTOMUTE (1 << 7) +#define MUTEC_AB (1 << 5) +#define MUTE_A (1 << 4) +#define MUTE_B (1 << 3) +#define MUTE_AB_MASK 0x18 + +/* CS4349_RMPFLT (Ramp and Filter Control) */ +#define SCZ1 (1 << 7) +#define SCZ0 (1 << 6) +#define RMP_UP (1 << 5) +#define RMP_DN (1 << 4) +#define FILT_SEL (1 << 2) +#define IMMDT_CHNG 0x31 +#define ZEROCRSS 0x71 +#define SOFT_RMP 0xB1 +#define SFTRMP_ZEROCRSS 0xF1 +#define SR_ZC_MASK 0xC0 + +/* CS4349_MISC */ +#define PWR_DWN (1 << 7) +#define FREEZE (1 << 5) +#define POPG_EN (1 << 4) + +#endif /* __CS4349_H__ */ -- 2.4.5

2 1

[alsa-devel] armv7 raspberry pi kernel compilation and snd_usb_us122l tascam us122 module requires x86
by James Miller 15 Jul '15

15 Jul '15

hi - I am compiling a rt kernel for my raspberry pi, and I notice I am unable to select the us122l driver for inclusion as a module as it is listed as requiring x86 to be selected. Many of the other modules are available and use the same codebase as the us122l driver, so I was wondering if the x86 requirement is strictly necessary? Is it possible to check that the us122l module requires x86 and is incompatible with arm architecture please. Many thanks James -- James Stewart Miller Bsc(hons) Psych.

2 1

[alsa-devel] Issue in alsa when dma complete race with pcm release
by Shengjiu Wang 15 Jul '15

15 Jul '15

Hi alsa-devel There maybe a issue in ALSA when dma complete race with snd_pcm_release. The pcm release and dma complete are in different thread. There is occasion that dmaengine_pcm_dma_complete() is called too late, some memory has been freed, the prtd is null. Then there is kernel dump. Is there any solution for this issue? Thanks. Best regards wang shengjiu

2 9

[alsa-devel] [PATCH] ALSA: hda - Fix Skylake codec timeouts
by David Henningsson 15 Jul '15

15 Jul '15

When the controller is powered up but the HDMI codec is powered down on Skylake, the power well is turned off. When the codec is then powered up again, we need to poke the codec a little extra to make sure it wakes up. Otherwise we'll get sad "no response from codec" messages and broken audio. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> Tested-by: Woodrow Shen <woodrow.shen(a)canonical.com> --- * It would good to have an ack from Intel on this patch, since they have better hardware knowledge than I. * Also I haven't really kept track of all recent reorganisation of the hda driver so I'm not totally sure how many kernels back (if any) that this applies to sound/pci/hda/hda_intel.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ca151b4..872e9a7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -567,9 +567,17 @@ static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev) /* Enable/disable i915 display power for the link */ static int azx_intel_link_power(struct azx *chip, bool enable) { + int err; struct hdac_bus *bus = azx_bus(chip); - return snd_hdac_display_power(bus, enable); + err = snd_hdac_display_power(bus, enable); + if (err < 0) + return err; + if (enable && (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)) { + snd_hdac_set_codec_wakeup(bus, true); + snd_hdac_set_codec_wakeup(bus, false); + } + return 0; } /* -- 1.9.1

2 3

[alsa-devel] [PATCH v2] ASoC: cs4349: Add support for Cirrus Logic CS4349
by timothyc.howe＠gmail.com 15 Jul '15

15 Jul '15

From: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- sound/soc/codecs/cs4349.c | 27 ++++++++------------------- 1 file changed, 8 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index c9a96b6..ee6b676 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -305,19 +305,8 @@ static int cs4349_i2c_probe(struct i2c_client *client, return ret; } - if (pdata) { + if (pdata) cs4349->pdata = *pdata; - } else { - pdata = devm_kzalloc(&client->dev, - sizeof(struct cs4349_platform_data), - GFP_KERNEL); - if (!pdata) { - dev_err(&client->dev, - "could not allocate pdata\n"); - return -ENOMEM; - } - cs4349->pdata = *pdata; - } /* Reset the Device */ cs4349->reset_gpio = devm_gpiod_get_optional(&client->dev, @@ -330,9 +319,6 @@ static int cs4349_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, cs4349); - dev_info(&client->dev, - "Cirrus Logic CS4349\n"); - return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4349, &cs4349_dai, 1); } @@ -357,14 +343,17 @@ static int cs4349_runtime_suspend(struct device *dev) struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); int ret; - /* Hold down reset */ - if (cs4349->reset_gpio) - gpiod_set_value_cansleep(cs4349->reset_gpio, 0); - ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 1); if (ret < 0) return ret; + regcache_cache_only(cs4349->regmap, true); + regcache_sync(cs4349->regmap); + + /* Hold down reset */ + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + return 0; } -- 2.4.5

2 1

[alsa-devel] [GIT PULL] Braswell firmware update
by Vinod Koul 14 Jul '15

14 Jul '15

Hello Linux firmware maintainers, Please PULL to receive the updated BSW firmware binary which contains power management fixes. We have also updated the version of binary in WHENCE file The following changes since commit 3161bfa479d5e9ed4f46b57df9bcecbbc4f8eb3c: linux-firmware: Update firmware patch for Intel Bluetooth 7260 (B3/B4) (2015-05-13 11:12:48 -0400) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git BSW for you to fetch changes up to e5fa14afd09909124fc7618fc4a0ac00fa4f9347: linux-firmware: update audio firmware for Braswell platform (2015-06-17 11:21:47 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: update audio firmware for Braswell platform WHENCE | 1 + intel/fw_sst_22a8.bin | Bin 647390 -> 701614 bytes 2 files changed, 1 insertion(+) Thanks -- ~Vinod

2 2

[alsa-devel] [PATCH] cht_bsw_rt5645: Fix Kernel crash with strcpy
by Jenny TC 14 Jul '15

14 Jul '15

Kernel crash is observed when strcpy is used with const char*. The crash is reported only if driver is loaded as a module or the probe is called in the late stage of Kernel boot (EPROBE_DEFER). This is because Kernel enables read only protection near the end. Issue is addressed by replacing the strcpy with pointer assignment Signed-off-by: Jenny TC <jenny.tc(a)intel.com> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index bdcaf46..3665d6c 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -367,7 +367,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) card->dev = &pdev->dev; sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id); /* set correct codec name */ - strcpy((char *)card->dai_link[2].codec_name, codec_name); + card->dai_link[2].codec_name = codec_name; snd_soc_card_set_drvdata(card, drv); ret_val = devm_snd_soc_register_card(&pdev->dev, card); if (ret_val) { -- 1.7.9.5

1 0

[alsa-devel] [PATCH] ALSA: line6: Fix -EBUSY error during active monitoring
by Takashi Iwai 14 Jul '15

14 Jul '15

When a monitor stream is active, the next PCM stream access results in EBUSY error because of the check in line6_stream_start(). Fix this by just skipping the submission of pending URBs when the stream is already running instead. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=101431 Cc: <stable(a)vger.kernel.org> # v4.0+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/line6/pcm.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c index 8461d6bf992f..204cc074adb9 100644 --- a/sound/usb/line6/pcm.c +++ b/sound/usb/line6/pcm.c @@ -186,12 +186,8 @@ static int line6_stream_start(struct snd_line6_pcm *line6pcm, int direction, int ret = 0; spin_lock_irqsave(&pstr->lock, flags); - if (!test_and_set_bit(type, &pstr->running)) { - if (pstr->active_urbs || pstr->unlink_urbs) { - ret = -EBUSY; - goto error; - } - + if (!test_and_set_bit(type, &pstr->running) && + !(pstr->active_urbs || pstr->unlink_urbs)) { pstr->count = 0; /* Submit all currently available URBs */ if (direction == SNDRV_PCM_STREAM_PLAYBACK) @@ -199,7 +195,6 @@ static int line6_stream_start(struct snd_line6_pcm *line6pcm, int direction, else ret = line6_submit_audio_in_all_urbs(line6pcm); } - error: if (ret < 0) clear_bit(type, &pstr->running); spin_unlock_irqrestore(&pstr->lock, flags); -- 2.4.5

1 0

[alsa-devel] ASoC Multiple outputs
by Matt Redfearn 14 Jul '15

14 Jul '15

Hi list, I am writing an ALSA driver for an SoC / board which has a single I2S source. The source may be connected internally to either a codec outputting to analogue headphones, or to an HDMI transmitter. At the moment I have set up two snd_soc_dai_link within the same card connecting the source to each sink. static struct snd_soc_dai_link ci20_dai_link[] = { { .name = "ci20", .stream_name = "headphones", .cpu_dai_name = "jz4780-i2s", .platform_name = "jz4780-i2s", .codec_dai_name = "jz4780-hifi", .codec_name = "jz4780-codec", .init = ci20_init, .ops = &ci20_audio_dai_ops, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, }, { .name = "ci20 HDMI", .stream_name = "hdmi", .cpu_dai_name = "jz4780-i2s", .platform_name = "jz4780-i2s", .codec_dai_name = "dw-hdmi-hifi", .codec_name = "dw-hdmi-audio", .init = ci20_hdmi_init, .ops = &ci20_audio_dai_ops, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, } }; This works and creates 2 devices, and I am able to get mplayer to play sound out to either the headphones or HDMI using the argument "-ao alsa:device=hw=0.0" or "-ao alsa:device=hw=0.1" aplay -l **** List of PLAYBACK Hardware Devices **** card 0: ci20 [ci20], device 0: headphones jz4780-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: ci20 [ci20], device 1: hdmi jz4780-hdmi-hifi-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 However, alsamixer does not show anything about the second device (HDMI), only the analogue output associated with device 0. Please could someone advise what is the "correct" way to set up multiple links in such a system such that the headphones / HDMI outputs appear in the mixer and are available to pulseaudio. Should I create multiple cards? Should I create subdevices for the different outputs? If so, what structures should be set up within the driver to achieve this? Suggestions of any documentation / websites that may help would be great! Thanks! Matt

2 1

[alsa-devel] [PATCH] ASoC: topology: Fix to add dapm mixer info
by Subhransu S. Prusty 14 Jul '15

14 Jul '15

From: Jeeja KP <jeeja.kp(a)intel.com> Mixer control for widgets can't be created if the info is NULL. So assign the correct info for this. Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index f38f111..70bcfe9 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -144,7 +144,7 @@ static const struct snd_soc_tplg_kcontrol_ops io_ops[] = { {SND_SOC_TPLG_CTL_STROBE, snd_soc_get_strobe, snd_soc_put_strobe, NULL}, {SND_SOC_TPLG_DAPM_CTL_VOLSW, snd_soc_dapm_get_volsw, - snd_soc_dapm_put_volsw, NULL}, + snd_soc_dapm_put_volsw, snd_soc_info_volsw}, {SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE, snd_soc_dapm_get_enum_double, snd_soc_dapm_put_enum_double, snd_soc_info_enum_double}, {SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT, snd_soc_dapm_get_enum_double, -- 2.4.3

2 1

[alsa-devel] [PATCH] ASoC: max98925: Remove control_data initialization
by Lars-Peter Clausen 14 Jul '15

14 Jul '15

The control_data field of the snd_soc_codec struct is no longer used by ASoC drivers using standard regmap IO. There is no need to initialize the field to point to the drivers regmap struct, so drop that. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/max98925.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index aad6642..eddf8bc 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -523,7 +523,6 @@ static int max98925_probe(struct snd_soc_codec *codec) struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); max98925->codec = codec; - codec->control_data = max98925->regmap; regmap_write(max98925->regmap, MAX98925_GLOBAL_ENABLE, 0x00); /* It's not the default but we need to set DAI_DLY */ regmap_write(max98925->regmap, -- 2.1.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: max98357a: Make 'sdmode-gpios' dts property optional
by Anatol Pomozov 14 Jul '15

14 Jul '15

The option is not needed if chip is always on or managed by some other part of system like platform card driver. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- Documentation/devicetree/bindings/sound/max98357a.txt | 6 +++++- sound/soc/codecs/max98357a.c | 5 ++++- 2 files changed, 9 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/max98357a.txt b/Documentation/devicetree/bindings/sound/max98357a.txt index a7a149a..28645a2 100644 --- a/Documentation/devicetree/bindings/sound/max98357a.txt +++ b/Documentation/devicetree/bindings/sound/max98357a.txt @@ -4,7 +4,11 @@ This node models the Maxim MAX98357A DAC. Required properties: - compatible : "maxim,max98357a" -- sdmode-gpios : GPIO specifier for the GPIO -> DAC SDMODE pin + +Optional properties: +- sdmode-gpios : GPIO specifier for the chip's SD_MODE pin. + If this option is not specified then driver does not manage + the pin state (e.g. chip is always on). Example: diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 3a2fda0..fa1b793 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -31,6 +31,9 @@ static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, { struct gpio_desc *sdmode = snd_soc_dai_get_drvdata(dai); + if (!sdmode) + return 0; + switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: @@ -60,7 +63,7 @@ static int max98357a_codec_probe(struct snd_soc_codec *codec) { struct gpio_desc *sdmode; - sdmode = devm_gpiod_get(codec->dev, "sdmode", GPIOD_OUT_LOW); + sdmode = devm_gpiod_get_optional(codec->dev, "sdmode", GPIOD_OUT_LOW); if (IS_ERR(sdmode)) { dev_err(codec->dev, "%s() unable to get sdmode GPIO: %ld\n", __func__, PTR_ERR(sdmode)); -- 2.4.5

3 6

[alsa-devel] [PATCH - CS4349 2/2] Add devicetree bindings for Cirrus Logic CS4349
by timothyc.howe＠gmail.com 13 Jul '15

13 Jul '15

From: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- Documentation/devicetree/bindings/sound/cs4349.txt | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs4349.txt diff --git a/Documentation/devicetree/bindings/sound/cs4349.txt b/Documentation/devicetree/bindings/sound/cs4349.txt new file mode 100644 index 0000000..54c117b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4349.txt @@ -0,0 +1,19 @@ +CS4349 audio CODEC + +Required properties: + + - compatible : "cirrus,cs4349" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + +Example: + +codec: cs4349@48 { + compatible = "cirrus,cs4349"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; +}; -- 2.4.5

2 1

[alsa-devel] [PATCH - CS4349 1/2] Add support for Cirrus Logic CS4349
by timothyc.howe＠gmail.com 13 Jul '15

13 Jul '15

From: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4349.c | 425 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs4349.h | 146 ++++++++++++++++ 4 files changed, 579 insertions(+) create mode 100644 sound/soc/codecs/cs4349.c create mode 100644 sound/soc/codecs/cs4349.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7223c7f..4a78606 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -54,6 +54,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_I2C if I2C select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C + select SND_SOC_CS4349 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -413,6 +414,11 @@ config SND_SOC_CS42XX8_I2C select SND_SOC_CS42XX8 select REGMAP_I2C +# Cirrus Logic CS4349 HiFi DAC +config SND_SOC_CS4349 + tristate "Cirrus Logic CS4349 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9f7136c..36a9f37 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -46,6 +46,7 @@ snd-soc-cs4271-i2c-objs := cs4271-i2c.o snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o +snd-soc-cs4349-objs := cs4349.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -237,6 +238,7 @@ obj-$(CONFIG_SND_SOC_CS4271_I2C) += snd-soc-cs4271-i2c.o obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o +obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c new file mode 100644 index 0000000..81fc2a0 --- /dev/null +++ b/sound/soc/codecs/cs4349.c @@ -0,0 +1,425 @@ +/* + * cs4349.c -- CS4349 ALSA Soc Audio driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Authors: Tim Howe <Tim.Howe(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs4349.h" + + +static const struct reg_default cs4349_reg_defaults[] = { + { 2, 0x00 }, /* r02 - Mode Control */ + { 3, 0x09 }, /* r03 - Volume, Mixing and Inversion Control */ + { 4, 0x81 }, /* r04 - Mute Control */ + { 5, 0x00 }, /* r05 - Channel A Volume Control */ + { 6, 0x00 }, /* r06 - Channel B Volume Control */ + { 7, 0xB1 }, /* r07 - Ramp and Filter Control */ + { 8, 0x1C }, /* r08 - Misc. Control */ +}; + +/* Private data for the CS4349 */ +struct cs4349_private { + struct regmap *regmap; + struct cs4349_platform_data pdata; + struct gpio_desc *reset_gpio; + unsigned int mclk; + unsigned int mode; + int rate; +}; + +static bool cs4349_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS4349_CHIPID: + case CS4349_MODE: + case CS4349_VMI: + case CS4349_MUTE: + case CS4349_VOLA: + case CS4349_VOLB: + case CS4349_RMPFLT: + case CS4349_MISC: + return true; + default: + return false; + } +} + +static int cs4349_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + + cs4349->mclk = freq; + + return 0; +} + +static int cs4349_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + unsigned int fmt; + + fmt = format & SND_SOC_DAIFMT_FORMAT_MASK; + + switch (fmt) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + cs4349->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs4349_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + int mode, fmt, ret; + + mode = snd_soc_read(codec, CS4349_MODE); + cs4349->rate = params_rate(params); + + switch (cs4349->mode) { + case SND_SOC_DAIFMT_I2S: + mode |= MODE_FORMAT(DIF_I2S); + break; + case SND_SOC_DAIFMT_LEFT_J: + mode |= MODE_FORMAT(DIF_LEFT_JST); + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_width(params)) { + case 16: + fmt = DIF_RGHT_JST16; + break; + case 24: + fmt = DIF_RGHT_JST24; + break; + default: + return -EINVAL; + } + mode |= MODE_FORMAT(fmt); + break; + default: + return -EINVAL; + } + + ret = snd_soc_write(codec, CS4349_MODE, mode); + if (ret < 0) + return ret; + + return 0; +} + +static int cs4349_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int reg; + + reg = 0; + if (mute) + reg = MUTE_AB_MASK; + + return snd_soc_update_bits(codec, CS4349_MUTE, MUTE_AB_MASK, reg); +} + +static DECLARE_TLV_DB_SCALE(dig_tlv, -12750, 50, 0); + +static const char * const chan_mix_texts[] = { + "Mute", "MuteA", "MuteA SwapB", "MuteA MonoB", "SwapA MuteB", + "BothR", "Swap", "SwapA MonoB", "MuteB", "Normal", "BothL", + "MonoB", "MonoA MuteB", "MonoA", "MonoA SwapB", "Mono", + /*Normal == Channel A = Left, Channel B = Right*/ +}; + +static const char * const fm_texts[] = { + "Auto", "Single", "Double", "Quad", +}; + +static const char * const deemph_texts[] = { + "None", "44.1k", "48k", "32k", +}; + +static const char * const softr_zeroc_texts[] = { + "Immediate", "Zero Cross", "Soft Ramp", "SR on ZC", +}; + +static int deemph_values[] = { + 0, 4, 8, 12, +}; + +static int softr_zeroc_values[] = { + 0, 64, 128, 192, +}; + +static const struct soc_enum chan_mix_enum = + SOC_ENUM_SINGLE(CS4349_VMI, 0, + ARRAY_SIZE(chan_mix_texts), + chan_mix_texts); + +static const struct soc_enum fm_mode_enum = + SOC_ENUM_SINGLE(CS4349_MODE, 0, + ARRAY_SIZE(fm_texts), + fm_texts); + +static SOC_VALUE_ENUM_SINGLE_DECL(deemph_enum, CS4349_MODE, 0, DEM_MASK, + deemph_texts, deemph_values); + +static SOC_VALUE_ENUM_SINGLE_DECL(softr_zeroc_enum, CS4349_RMPFLT, 0, + SR_ZC_MASK, softr_zeroc_texts, + softr_zeroc_values); + +static const struct snd_kcontrol_new cs4349_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", + CS4349_VOLA, CS4349_VOLB, 0, 0xFF, 1, dig_tlv), + SOC_ENUM("Functional Mode", fm_mode_enum), + SOC_ENUM("De-Emphasis Control", deemph_enum), + SOC_ENUM("Soft Ramp Zero Cross Control", softr_zeroc_enum), + SOC_ENUM("Channel Mixer", chan_mix_enum), + SOC_SINGLE("VolA = VolB Switch", CS4349_VMI, 7, 1, 0), + SOC_SINGLE("InvertA Switch", CS4349_VMI, 6, 1, 0), + SOC_SINGLE("InvertB Switch", CS4349_VMI, 5, 1, 0), + SOC_SINGLE("Auto-Mute Switch", CS4349_MUTE, 7, 1, 0), + SOC_SINGLE("MUTEC A = B Switch", CS4349_MUTE, 5, 1, 0), + SOC_SINGLE("Soft Ramp Up Switch", CS4349_RMPFLT, 5, 1, 0), + SOC_SINGLE("Soft Ramp Down Switch", CS4349_RMPFLT, 4, 1, 0), + SOC_SINGLE("Slow Roll Off Filter Switch", CS4349_RMPFLT, 2, 1, 0), + SOC_SINGLE("Freeze Switch", CS4349_MISC, 5, 1, 0), + SOC_SINGLE("Popguard Switch", CS4349_MISC, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget cs4349_dapm_widgets[] = { + SND_SOC_DAPM_DAC("HiFi DAC", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_OUTPUT("OutputA"), + SND_SOC_DAPM_OUTPUT("OutputB"), +}; + +static const struct snd_soc_dapm_route cs4349_routes[] = { + {"DAC Playback", NULL, "OutputA"}, + {"DAC Playback", NULL, "OutputB"}, + + {"OutputA", NULL, "HiFi DAC"}, + {"OutputB", NULL, "HiFi DAC"}, +}; + +#define CS4349_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) + +#define CS4349_PCM_RATES SNDRV_PCM_RATE_8000_192000 + +static const struct snd_soc_dai_ops cs4349_dai_ops = { + .hw_params = cs4349_pcm_hw_params, + .set_fmt = cs4349_set_dai_fmt, + .digital_mute = cs4349_digital_mute, +}; + +static struct snd_soc_dai_driver cs4349_dai = { + .name = "cs4349_hifi", + .playback = { + .stream_name = "DAC Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS4349_PCM_RATES, + .formats = CS4349_PCM_FORMATS, + }, + .ops = &cs4349_dai_ops, + .symmetric_rates = 1, +}; + +static struct snd_soc_codec_driver soc_codec_dev_cs4349 = { + .set_sysclk = cs4349_codec_set_sysclk, + + .controls = cs4349_snd_controls, + .num_controls = ARRAY_SIZE(cs4349_snd_controls), + + .dapm_widgets = cs4349_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4349_dapm_widgets), + .dapm_routes = cs4349_routes, + .num_dapm_routes = ARRAY_SIZE(cs4349_routes), +}; + +static struct regmap_config cs4349_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS4349_NUMREGS, + .reg_defaults = cs4349_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4349_reg_defaults), + .readable_reg = cs4349_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs4349_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs4349_private *cs4349; + struct cs4349_platform_data *pdata = dev_get_platdata(&client->dev); + int ret = 0; + + cs4349 = devm_kzalloc(&client->dev, sizeof(*cs4349), GFP_KERNEL); + if (!cs4349) + return -ENOMEM; + + cs4349->regmap = devm_regmap_init_i2c(client, &cs4349_regmap); + if (IS_ERR(cs4349->regmap)) { + ret = PTR_ERR(cs4349->regmap); + dev_err(&client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + if (pdata) { + cs4349->pdata = *pdata; + } else { + pdata = devm_kzalloc(&client->dev, + sizeof(struct cs4349_platform_data), + GFP_KERNEL); + if (!pdata) { + dev_err(&client->dev, + "could not allocate pdata\n"); + return -ENOMEM; + } + cs4349->pdata = *pdata; + } + + /* Reset the Device */ + cs4349->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs4349->reset_gpio)) + return PTR_ERR(cs4349->reset_gpio); + + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + + i2c_set_clientdata(client, cs4349); + + dev_info(&client->dev, + "Cirrus Logic CS4349\n"); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4349, + &cs4349_dai, 1); +} + +static int cs4349_i2c_remove(struct i2c_client *client) +{ + struct cs4349_private *cs4349 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + /* Hold down reset */ + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int cs4349_runtime_suspend(struct device *dev) +{ + struct cs4349_private *cs4349 = dev_get_drvdata(dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + int ret; + + /* Hold down reset */ + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + + ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 1); + if (ret < 0) + return ret; + + return 0; +} + +static int cs4349_runtime_resume(struct device *dev) +{ + struct cs4349_private *cs4349 = dev_get_drvdata(dev); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + int ret; + + ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 0); + if (ret < 0) + return ret; + + if (cs4349->reset_gpio) + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + + regcache_cache_only(cs4349->regmap, false); + regcache_sync(cs4349->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs4349_runtime_pm = { + SET_RUNTIME_PM_OPS(cs4349_runtime_suspend, cs4349_runtime_resume, + NULL) +}; + +static const struct of_device_id cs4349_of_match[] = { + { .compatible = "cirrus,cs4349", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs4349_of_match); + +static const struct i2c_device_id cs4349_i2c_id[] = { + {"cs4349", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs4349_i2c_id); + +static struct i2c_driver cs4349_i2c_driver = { + .driver = { + .name = "cs4349", + .owner = THIS_MODULE, + .of_match_table = cs4349_of_match, + }, + .id_table = cs4349_i2c_id, + .probe = cs4349_i2c_probe, + .remove = cs4349_i2c_remove, +}; + +module_i2c_driver(cs4349_i2c_driver); + +MODULE_AUTHOR("Tim Howe <tim.howe(a)cirrus.com>"); +MODULE_DESCRIPTION("Cirrus Logic CS4349 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4349.h b/sound/soc/codecs/cs4349.h new file mode 100644 index 0000000..3884a89 --- /dev/null +++ b/sound/soc/codecs/cs4349.h @@ -0,0 +1,146 @@ +/* + * ALSA SoC CS4349 codec driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Tim Howe <Tim.Howe(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + */ + +#ifndef __CS4349_H__ +#define __CS4349_H__ + +struct cs4349_platform_data { + + /* GPIO for Reset */ + unsigned int gpio_nreset; + +}; + +/* CS4349 registers addresses */ +#define CS4349_CHIPID 0x01 /* Device and Rev ID, Read Only */ +#define CS4349_MODE 0x02 /* Mode Control */ +#define CS4349_VMI 0x03 /* Volume, Mixing, Inversion Control */ +#define CS4349_MUTE 0x04 /* Mute Control */ +#define CS4349_VOLA 0x05 /* DAC Channel A Volume Control */ +#define CS4349_VOLB 0x06 /* DAC Channel B Volume Control */ +#define CS4349_RMPFLT 0x07 /* Ramp and Filter Control */ +#define CS4349_MISC 0x08 /* Power Down,Freeze Control,Pop Stop*/ + +#define CS4349_FIRSTREG 0x01 +#define CS4349_LASTREG 0x08 +#define CS4349_NUMREGS (CS4349_LASTREG - CS4349_FIRSTREG + 1) +#define CS4349_I2C_INCR 0x80 + + +/* Device and Revision ID */ +#define CS4349_REVA 0xF0 /* Rev A */ +#define CS4349_REVB 0xF1 /* Rev B */ +#define CS4349_REVC2 0xFF /* Rev C2 */ + + +/* PDN_DONE Poll Maximum + * If soft ramp is set it will take much longer to power down + * the system. + */ +#define PDN_POLL_MAX 900 + + +/* Bitfield Definitions */ + +/* CS4349_MODE */ +/* (Digital Interface Format, De-Emphasis Control, Functional Mode */ +#define DIF2 (1 << 6) +#define DIF1 (1 << 5) +#define DIF0 (1 << 4) +#define DEM1 (1 << 3) +#define DEM0 (1 << 2) +#define FM1 (1 << 1) +#define DIF_LEFT_JST 0x00 +#define DIF_I2S 0x01 +#define DIF_RGHT_JST16 0x02 +#define DIF_RGHT_JST24 0x03 +#define DIF_TDM0 0x04 +#define DIF_TDM1 0x05 +#define DIF_TDM2 0x06 +#define DIF_TDM3 0x07 +#define DIF_MASK 0x70 +#define MODE_FORMAT(x) (((x)&7)<<4) +#define DEM_MASK 0x0C +#define NO_DEM 0x00 +#define DEM_441 0x04 +#define DEM_48K 0x08 +#define DEM_32K 0x0C +#define FM_AUTO 0x00 +#define FM_SNGL 0x01 +#define FM_DBL 0x02 +#define FM_QUAD 0x03 +#define FM_SNGL_MIN 30000 +#define FM_SNGL_MAX 54000 +#define FM_DBL_MAX 108000 +#define FM_QUAD_MAX 216000 +#define FM_MASK 0x03 + +/* CS4349_VMI (VMI = Volume, Mixing and Inversion Controls) */ +#define VOLBISA (1 << 7) +#define VOLAISB (1 << 7) +/* INVERT_A only available for Left Jstfd, Right Jstfd16 and Right Jstfd24 */ +#define INVERT_A (1 << 6) +/* INVERT_B only available for Left Jstfd, Right Jstfd16 and Right Jstfd24 */ +#define INVERT_B (1 << 5) +#define ATAPI3 (1 << 3) +#define ATAPI2 (1 << 2) +#define ATAPI1 (1 << 1) +#define ATAPI0 (1 << 0) +#define MUTEAB 0x00 +#define MUTEA_RIGHTB 0x01 +#define MUTEA_LEFTB 0x02 +#define MUTEA_SUMLRDIV2B 0x03 +#define RIGHTA_MUTEB 0x04 +#define RIGHTA_RIGHTB 0x05 +#define RIGHTA_LEFTB 0x06 +#define RIGHTA_SUMLRDIV2B 0x07 +#define LEFTA_MUTEB 0x08 +#define LEFTA_RIGHTB 0x09 /* Default */ +#define LEFTA_LEFTB 0x0A +#define LEFTA_SUMLRDIV2B 0x0B +#define SUMLRDIV2A_MUTEB 0x0C +#define SUMLRDIV2A_RIGHTB 0x0D +#define SUMLRDIV2A_LEFTB 0x0E +#define SUMLRDIV2_AB 0x0F +#define CHMIX_MASK 0x0F + +/* CS4349_MUTE */ +#define AUTOMUTE (1 << 7) +#define MUTEC_AB (1 << 5) +#define MUTE_A (1 << 4) +#define MUTE_B (1 << 3) +#define MUTE_AB_MASK 0x18 + +/* CS4349_RMPFLT (Ramp and Filter Control) */ +#define SCZ1 (1 << 7) +#define SCZ0 (1 << 6) +#define RMP_UP (1 << 5) +#define RMP_DN (1 << 4) +#define FILT_SEL (1 << 2) +#define IMMDT_CHNG 0x31 +#define ZEROCRSS 0x71 +#define SOFT_RMP 0xB1 +#define SFTRMP_ZEROCRSS 0xF1 +#define SR_ZC_MASK 0xC0 + +/* CS4349_MISC */ +#define PWR_DWN (1 << 7) +#define FREEZE (1 << 5) +#define POPG_EN (1 << 4) + +#endif /* __CS4349_H__ */ -- 2.4.5

2 1

[alsa-devel] randconfig build error with next-20150713, in sound/soc/codecs/sti-sas.c
by Jim Davis 13 Jul '15

13 Jul '15

Building with the attached random configuration file, warning: (INTEL_SOC_DTS_IOSF_CORE && PUNIT_ATOM_DEBUG) selects IOSF_MBI which has unmet direct dependencies (PCI) warning: (SND_SOC_INTEL_BYTCR_RT5640_MACH && SND_SOC_INTEL_CHT_BSW_RT5672_MACH && SND_SOC_INTEL_CHT_BSW_RT5645_MACH && SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) selects SND_SST_IPC_ACPI which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && ACPI) warning: (SND_SOC_ALL_CODECS) selects SND_SOC_STI_SAS which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && SND_SOC_STI) sound/soc/codecs/sti-sas.c: In function ‘stih416_dac_probe’: sound/soc/codecs/sti-sas.c:227:13: error: implicit declaration of function ‘devm_reset_control_get’ [-Werror=implicit-function-declaration] dac->rst = devm_reset_control_get(codec->dev, "dac_rst"); ^ sound/soc/codecs/sti-sas.c:227:11: warning: assignment makes pointer from integer without a cast [-Wint-conversion] dac->rst = devm_reset_control_get(codec->dev, "dac_rst"); ^ sound/soc/codecs/sti-sas.c:231:14: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast] __func__, (int)dac->rst); ^ cc1: some warnings being treated as errors

1 0

[alsa-devel] [PATCH 1/2] ASoC: rt5645: Fix missing free_irq
by Koro Chen 13 Jul '15

13 Jul '15

The driver does not free irq if snd_soc_register_codec fails. It does not return error when request irq failed, either. Fix this by using devm_request_threaded_irq(), and returns when error. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/codecs/rt5645.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9dfa431..f9f2db8 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3427,11 +3427,15 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); if (rt5645->i2c->irq) { - ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq, - IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING - | IRQF_ONESHOT, "rt5645", rt5645); - if (ret) + ret = devm_request_threaded_irq(&i2c->dev, rt5645->i2c->irq, + NULL, rt5645_irq, + IRQF_TRIGGER_RISING | + IRQF_TRIGGER_FALLING | + IRQF_ONESHOT, "rt5645", rt5645); + if (ret) { dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + return ret; + } } return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, @@ -3442,9 +3446,6 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) { struct rt5645_priv *rt5645 = i2c_get_clientdata(i2c); - if (i2c->irq) - free_irq(i2c->irq, rt5645); - cancel_delayed_work_sync(&rt5645->jack_detect_work); snd_soc_unregister_codec(&i2c->dev); -- 1.8.1.1.dirty

2 6

[alsa-devel] [PATCH] ASoC: add rt298 codec driver
by Bard Liao 13 Jul '15

13 Jul '15

It is the initial version of ALC298 codec driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- include/sound/rt298.h | 20 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt298.c | 1274 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt298.h | 206 ++++++++ 5 files changed, 1509 insertions(+) create mode 100644 include/sound/rt298.h create mode 100644 sound/soc/codecs/rt298.c create mode 100644 sound/soc/codecs/rt298.h diff --git a/include/sound/rt298.h b/include/sound/rt298.h new file mode 100644 index 0000000..7fffeaa --- /dev/null +++ b/include/sound/rt298.h @@ -0,0 +1,20 @@ +/* + * linux/sound/rt286.h -- Platform data for RT286 + * + * Copyright 2013 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT298_H +#define __LINUX_SND_RT298_H + +struct rt298_platform_data { + bool cbj_en; /*combo jack enable*/ + bool gpio2_en; /*GPIO2 enable*/ + bool suspend_power_off; /* power is off during suspend */ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index efaafce..76125a2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -83,6 +83,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER select SND_SOC_RT286 if I2C + select SND_SOC_RT298 if I2C select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_RT5645 if I2C @@ -512,12 +513,18 @@ config SND_SOC_RL6231 config SND_SOC_RL6347A tristate default y if SND_SOC_RT286=y + default y if SND_SOC_RT298=y default m if SND_SOC_RT286=m + default m if SND_SOC_RT298=m config SND_SOC_RT286 tristate depends on I2C +config SND_SOC_RT298 + tristate + depends on I2C + config SND_SOC_RT5631 tristate "Realtek ALC5631/RT5631 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cf160d9..3b58c45 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -79,6 +79,7 @@ snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt286-objs := rt286.o +snd-soc-rt298-objs := rt298.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o @@ -266,6 +267,7 @@ obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o +obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c new file mode 100644 index 0000000..7c4bcb6 --- /dev/null +++ b/sound/soc/codecs/rt298.c @@ -0,0 +1,1274 @@ +/* + * rt298.c -- RT298 ALSA SoC audio codec driver + * + * Copyright 2015 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <linux/workqueue.h> +#include <sound/rt298.h> +#include <sound/hda_verbs.h> + +#include "rl6347a.h" +#include "rt298.h" + +#define RT298_VENDOR_ID 0x10ec0298 + +struct rt298_priv { + struct reg_default *index_cache; + int index_cache_size; + struct regmap *regmap; + struct snd_soc_codec *codec; + struct rt298_platform_data pdata; + struct i2c_client *i2c; + struct snd_soc_jack *jack; + struct delayed_work jack_detect_work; + int sys_clk; + int clk_id; + int is_hp_in; +}; + +static struct reg_default rt298_index_def[] = { + { 0x01, 0xaaaa }, + { 0x02, 0x8aaa }, + { 0x03, 0x0002 }, + { 0x04, 0xaf01 }, + { 0x08, 0x000d }, + { 0x09, 0xd810 }, + { 0x0a, 0x0120 }, + { 0x0b, 0x0000 }, + { 0x0d, 0x2800 }, + { 0x0f, 0x0000 }, + { 0x19, 0x0a17 }, + { 0x20, 0x0020 }, + { 0x33, 0x0208 }, + { 0x46, 0x0300 }, + { 0x49, 0x0004 }, + { 0x4f, 0x50e9 }, + { 0x50, 0x2000 }, + { 0x63, 0x2902 }, + { 0x67, 0x1111 }, + { 0x68, 0x1016 }, + { 0x69, 0x273f }, +}; +#define INDEX_CACHE_SIZE ARRAY_SIZE(rt298_index_def) + +static const struct reg_default rt298_reg[] = { + { 0x00170500, 0x00000400 }, + { 0x00220000, 0x00000031 }, + { 0x00239000, 0x0000007f }, + { 0x0023a000, 0x0000007f }, + { 0x00270500, 0x00000400 }, + { 0x00370500, 0x00000400 }, + { 0x00870500, 0x00000400 }, + { 0x00920000, 0x00000031 }, + { 0x00935000, 0x000000c3 }, + { 0x00936000, 0x000000c3 }, + { 0x00970500, 0x00000400 }, + { 0x00b37000, 0x00000097 }, + { 0x00b37200, 0x00000097 }, + { 0x00b37300, 0x00000097 }, + { 0x00c37000, 0x00000000 }, + { 0x00c37100, 0x00000080 }, + { 0x01270500, 0x00000400 }, + { 0x01370500, 0x00000400 }, + { 0x01371f00, 0x411111f0 }, + { 0x01439000, 0x00000080 }, + { 0x0143a000, 0x00000080 }, + { 0x01470700, 0x00000000 }, + { 0x01470500, 0x00000400 }, + { 0x01470c00, 0x00000000 }, + { 0x01470100, 0x00000000 }, + { 0x01837000, 0x00000000 }, + { 0x01870500, 0x00000400 }, + { 0x02050000, 0x00000000 }, + { 0x02139000, 0x00000080 }, + { 0x0213a000, 0x00000080 }, + { 0x02170100, 0x00000000 }, + { 0x02170500, 0x00000400 }, + { 0x02170700, 0x00000000 }, + { 0x02270100, 0x00000000 }, + { 0x02370100, 0x00000000 }, + { 0x01870700, 0x00000020 }, + { 0x00830000, 0x000000c3 }, + { 0x00930000, 0x000000c3 }, + { 0x01270700, 0x00000000 }, +}; + +static bool rt298_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT298_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT298_GET_HP_SENSE: + case RT298_GET_MIC1_SENSE: + case RT298_PROC_COEF: + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT298_MIC1, 0): + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT298_SPK_OUT, 0): + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT298_HP_OUT, 0): + return true; + default: + return true; + } + + +} + +static bool rt298_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT298_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT298_GET_HP_SENSE: + case RT298_GET_MIC1_SENSE: + case RT298_SET_AUDIO_POWER: + case RT298_SET_HPO_POWER: + case RT298_SET_SPK_POWER: + case RT298_SET_DMIC1_POWER: + case RT298_SPK_MUX: + case RT298_HPO_MUX: + case RT298_ADC0_MUX: + case RT298_ADC1_MUX: + case RT298_SET_MIC1: + case RT298_SET_PIN_HPO: + case RT298_SET_PIN_SPK: + case RT298_SET_PIN_DMIC1: + case RT298_SPK_EAPD: + case RT298_SET_AMP_GAIN_HPO: + case RT298_SET_DMIC2_DEFAULT: + case RT298_DACL_GAIN: + case RT298_DACR_GAIN: + case RT298_ADCL_GAIN: + case RT298_ADCR_GAIN: + case RT298_MIC_GAIN: + case RT298_SPOL_GAIN: + case RT298_SPOR_GAIN: + case RT298_HPOL_GAIN: + case RT298_HPOR_GAIN: + case RT298_F_DAC_SWITCH: + case RT298_F_RECMIX_SWITCH: + case RT298_REC_MIC_SWITCH: + case RT298_REC_I2S_SWITCH: + case RT298_REC_LINE_SWITCH: + case RT298_REC_BEEP_SWITCH: + case RT298_DAC_FORMAT: + case RT298_ADC_FORMAT: + case RT298_COEF_INDEX: + case RT298_PROC_COEF: + case RT298_SET_AMP_GAIN_ADC_IN1: + case RT298_SET_AMP_GAIN_ADC_IN2: + case RT298_SET_POWER(RT298_DAC_OUT1): + case RT298_SET_POWER(RT298_DAC_OUT2): + case RT298_SET_POWER(RT298_ADC_IN1): + case RT298_SET_POWER(RT298_ADC_IN2): + case RT298_SET_POWER(RT298_DMIC2): + case RT298_SET_POWER(RT298_MIC1): + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT298_MIC1, 0): + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT298_SPK_OUT, 0): + case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT298_HP_OUT, 0): + return true; + default: + return false; + } +} + +#ifdef CONFIG_PM +static void rt298_index_sync(struct snd_soc_codec *codec) +{ + struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < INDEX_CACHE_SIZE; i++) { + snd_soc_write(codec, rt298->index_cache[i].reg, + rt298->index_cache[i].def); + } +} +#endif + +static int rt298_support_power_controls[] = { + RT298_DAC_OUT1, + RT298_DAC_OUT2, + RT298_ADC_IN1, + RT298_ADC_IN2, + RT298_MIC1, + RT298_DMIC1, + RT298_DMIC2, + RT298_SPK_OUT, + RT298_HP_OUT, +}; +#define RT298_POWER_REG_LEN ARRAY_SIZE(rt298_support_power_controls) + +static int rt298_jack_detect(struct rt298_priv *rt298, bool *hp, bool *mic) +{ + struct snd_soc_dapm_context *dapm; + unsigned int val, buf; + + *hp = false; + *mic = false; + + if (!rt298->codec) + return -EINVAL; + + dapm = snd_soc_codec_get_dapm(rt298->codec); + + if (rt298->pdata.cbj_en) { + regmap_read(rt298->regmap, RT298_GET_HP_SENSE, &buf); + *hp = buf & 0x80000000; + if (*hp == rt298->is_hp_in) + return -1; + rt298->is_hp_in = *hp; + if (*hp) { + /* power on HV,VERF */ + regmap_update_bits(rt298->regmap, + RT298_DC_GAIN, 0x200, 0x200); + + snd_soc_dapm_force_enable_pin(dapm, "HV"); + snd_soc_dapm_force_enable_pin(dapm, "VREF"); + /* power LDO1 */ + snd_soc_dapm_force_enable_pin(dapm, "LDO1"); + snd_soc_dapm_sync(dapm); + + regmap_write(rt298->regmap, RT298_SET_MIC1, 0x24); + msleep(50); + + regmap_update_bits(rt298->regmap, + RT298_CBJ_CTRL1, 0xfcc0, 0xd400); + msleep(300); + regmap_read(rt298->regmap, RT298_CBJ_CTRL2, &val); + + if (0x0070 == (val & 0x0070)) { + *mic = true; + } else { + regmap_update_bits(rt298->regmap, + RT298_CBJ_CTRL1, 0xfcc0, 0xe400); + msleep(300); + regmap_read(rt298->regmap, + RT298_CBJ_CTRL2, &val); + if (0x0070 == (val & 0x0070)) + *mic = true; + else + *mic = false; + } + regmap_update_bits(rt298->regmap, + RT298_DC_GAIN, 0x200, 0x0); + + } else { + *mic = false; + regmap_write(rt298->regmap, RT298_SET_MIC1, 0x20); + } + } else { + regmap_read(rt298->regmap, RT298_GET_HP_SENSE, &buf); + *hp = buf & 0x80000000; + regmap_read(rt298->regmap, RT298_GET_MIC1_SENSE, &buf); + *mic = buf & 0x80000000; + } + + snd_soc_dapm_disable_pin(dapm, "HV"); + snd_soc_dapm_disable_pin(dapm, "VREF"); + if (!*hp) + snd_soc_dapm_disable_pin(dapm, "LDO1"); + snd_soc_dapm_sync(dapm); + + pr_debug("*hp = %d *mic = %d\n", *hp, *mic); + + return 0; +} + +static void rt298_jack_detect_work(struct work_struct *work) +{ + struct rt298_priv *rt298 = + container_of(work, struct rt298_priv, jack_detect_work.work); + int status = 0; + bool hp = false; + bool mic = false; + + if (rt298_jack_detect(rt298, &hp, &mic) < 0) + return; + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt298->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); +} + +int rt298_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + + rt298->jack = jack; + + /* Send an initial empty report */ + snd_soc_jack_report(rt298->jack, 0, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + return 0; +} +EXPORT_SYMBOL_GPL(rt298_mic_detect); + +static int is_mclk_mode(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + + if (rt298->clk_id == RT298_SCLK_S_MCLK) + return 1; + else + return 0; +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static const struct snd_kcontrol_new rt298_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT298_DACL_GAIN, + RT298_DACR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT298_ADCL_GAIN, + RT298_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_SINGLE_TLV("AMIC Volume", RT298_MIC_GAIN, + 0, 0x3, 0, mic_vol_tlv), + SOC_DOUBLE_R("Speaker Playback Switch", RT298_SPOL_GAIN, + RT298_SPOR_GAIN, RT298_MUTE_SFT, 1, 1), +}; + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt298_front_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", RT298_F_DAC_SWITCH, + RT298_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIX Switch", RT298_F_RECMIX_SWITCH, + RT298_MUTE_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt298_rec_mix[] = { + SOC_DAPM_SINGLE("Mic1 Switch", RT298_REC_MIC_SWITCH, + RT298_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("I2S Switch", RT298_REC_I2S_SWITCH, + RT298_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Line1 Switch", RT298_REC_LINE_SWITCH, + RT298_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Beep Switch", RT298_REC_BEEP_SWITCH, + RT298_MUTE_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spo_enable_control = + SOC_DAPM_SINGLE("Switch", RT298_SET_PIN_SPK, + RT298_SET_PIN_SFT, 1, 0); + +static const struct snd_kcontrol_new hpol_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT298_HPOL_GAIN, + RT298_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hpor_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT298_HPOR_GAIN, + RT298_MUTE_SFT, 1, 1); + +/* ADC0 source */ +static const char * const rt298_adc_src[] = { + "Mic", "RECMIX", "Dmic" +}; + +static const int rt298_adc_values[] = { + 0, 4, 5, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt298_adc0_enum, RT298_ADC0_MUX, RT298_ADC_SEL_SFT, + RT298_ADC_SEL_MASK, rt298_adc_src, rt298_adc_values); + +static const struct snd_kcontrol_new rt298_adc0_mux = + SOC_DAPM_ENUM("ADC 0 source", rt298_adc0_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt298_adc1_enum, RT298_ADC1_MUX, RT298_ADC_SEL_SFT, + RT298_ADC_SEL_MASK, rt298_adc_src, rt298_adc_values); + +static const struct snd_kcontrol_new rt298_adc1_mux = + SOC_DAPM_ENUM("ADC 1 source", rt298_adc1_enum); + +static const char * const rt298_dac_src[] = { + "Front", "Surround" +}; +/* HP-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt298_hpo_enum, RT298_HPO_MUX, + 0, rt298_dac_src); + +static const struct snd_kcontrol_new rt298_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt298_hpo_enum); + +/* SPK-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt298_spo_enum, RT298_SPK_MUX, + 0, rt298_dac_src); + +static const struct snd_kcontrol_new rt298_spo_mux = +SOC_DAPM_ENUM("SPO source", rt298_spo_enum); + +static int rt298_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, + RT298_SPK_EAPD, RT298_SET_EAPD_HIGH); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, + RT298_SPK_EAPD, RT298_SET_EAPD_LOW); + break; + + default: + return 0; + } + + return 0; +} + +static int rt298_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, RT298_SET_PIN_DMIC1, 0x20); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, RT298_SET_PIN_DMIC1, 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt298_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + unsigned int nid; + + nid = (w->reg >> 20) & 0xff; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7000); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7080); + break; + default: + return 0; + } + + return 0; +} + +static int rt298_mic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, + RT298_A_BIAS_CTRL3, 0xc000, 0x8000); + snd_soc_update_bits(codec, + RT298_A_BIAS_CTRL2, 0xc000, 0x8000); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, + RT298_A_BIAS_CTRL3, 0xc000, 0x0000); + snd_soc_update_bits(codec, + RT298_A_BIAS_CTRL2, 0xc000, 0x0000); + break; + default: + return 0; + } + + return 0; +} + +static int rt298_vref_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, + RT298_CBJ_CTRL1, 0x0400, 0x0000); + mdelay(50); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt298_dapm_widgets[] = { + + SND_SOC_DAPM_SUPPLY_S("HV", 1, RT298_POWER_CTRL1, + 12, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("VREF", RT298_POWER_CTRL1, + 0, 1, rt298_vref_event, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY_S("BG_MBIAS", 1, RT298_POWER_CTRL2, + 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("LDO1", 1, RT298_POWER_CTRL2, + 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("LDO2", 1, RT298_POWER_CTRL2, + 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("VREF1", 1, RT298_POWER_CTRL2, + 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("LV", 2, RT298_POWER_CTRL1, + 13, 1, NULL, 0), + + + SND_SOC_DAPM_SUPPLY("MCLK MODE", RT298_PLL_CTRL1, + 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC1 Input Buffer", SND_SOC_NOPM, + 0, 0, rt298_mic1_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("Beep"), + + /* DMIC */ + SND_SOC_DAPM_PGA_E("DMIC1", RT298_SET_POWER(RT298_DMIC1), 0, 1, + NULL, 0, rt298_set_dmic1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA("DMIC2", RT298_SET_POWER(RT298_DMIC2), 0, 1, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0, + rt298_rec_mix, ARRAY_SIZE(rt298_rec_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT298_SET_POWER(RT298_ADC_IN1), 0, 1, + &rt298_adc0_mux, rt298_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT298_SET_POWER(RT298_ADC_IN2), 0, 1, + &rt298_adc1_mux, rt298_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* Output Mux */ + SND_SOC_DAPM_MUX("SPK Mux", SND_SOC_NOPM, 0, 0, &rt298_spo_mux), + SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt298_hpo_mux), + + SND_SOC_DAPM_SUPPLY("HP Power", RT298_SET_PIN_HPO, + RT298_SET_PIN_SFT, 0, NULL, 0), + + /* Output Mixer */ + SND_SOC_DAPM_MIXER("Front", RT298_SET_POWER(RT298_DAC_OUT1), 0, 1, + rt298_front_mix, ARRAY_SIZE(rt298_front_mix)), + SND_SOC_DAPM_PGA("Surround", RT298_SET_POWER(RT298_DAC_OUT2), 0, 1, + NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_SWITCH_E("SPO", SND_SOC_NOPM, 0, 0, + &spo_enable_control, rt298_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH("HPO L", SND_SOC_NOPM, 0, 0, + &hpol_enable_control), + SND_SOC_DAPM_SWITCH("HPO R", SND_SOC_NOPM, 0, 0, + &hpor_enable_control), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), +}; + +static const struct snd_soc_dapm_route rt298_dapm_routes[] = { + + {"ADC 0", NULL, "MCLK MODE", is_mclk_mode}, + {"ADC 1", NULL, "MCLK MODE", is_mclk_mode}, + {"Front", NULL, "MCLK MODE", is_mclk_mode}, + {"Surround", NULL, "MCLK MODE", is_mclk_mode}, + + {"HP Power", NULL, "LDO1"}, + {"HP Power", NULL, "LDO2"}, + {"HP Power", NULL, "LV"}, + {"HP Power", NULL, "VREF1"}, + {"HP Power", NULL, "BG_MBIAS"}, + + {"MIC1", NULL, "LDO1"}, + {"MIC1", NULL, "LDO2"}, + {"MIC1", NULL, "HV"}, + {"MIC1", NULL, "LV"}, + {"MIC1", NULL, "VREF"}, + {"MIC1", NULL, "VREF1"}, + {"MIC1", NULL, "BG_MBIAS"}, + {"MIC1", NULL, "MIC1 Input Buffer"}, + + {"SPO", NULL, "LDO1"}, + {"SPO", NULL, "LDO2"}, + {"SPO", NULL, "HV"}, + {"SPO", NULL, "LV"}, + {"SPO", NULL, "VREF"}, + {"SPO", NULL, "VREF1"}, + {"SPO", NULL, "BG_MBIAS"}, + + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + {"DMIC1", NULL, "DMIC Receiver"}, + {"DMIC2", NULL, "DMIC Receiver"}, + + {"RECMIX", "Beep Switch", "Beep"}, + {"RECMIX", "Line1 Switch", "LINE1"}, + {"RECMIX", "Mic1 Switch", "MIC1"}, + + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "RECMIX", "RECMIX"}, + {"ADC 0 Mux", "Mic", "MIC1"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "RECMIX", "RECMIX"}, + {"ADC 1 Mux", "Mic", "MIC1"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TX", NULL, "ADC 0"}, + {"AIF2TX", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RX"}, + {"DAC 1", NULL, "AIF2RX"}, + + {"Front", "DAC Switch", "DAC 0"}, + {"Front", "RECMIX Switch", "RECMIX"}, + + {"Surround", NULL, "DAC 1"}, + + {"SPK Mux", "Front", "Front"}, + {"SPK Mux", "Surround", "Surround"}, + + {"HPO Mux", "Front", "Front"}, + {"HPO Mux", "Surround", "Surround"}, + + {"SPO", "Switch", "SPK Mux"}, + {"HPO L", "Switch", "HPO Mux"}, + {"HPO R", "Switch", "HPO Mux"}, + {"HPO L", NULL, "HP Power"}, + {"HPO R", NULL, "HP Power"}, + + {"SPOL", NULL, "SPO"}, + {"SPOR", NULL, "SPO"}, + {"HPO Pin", NULL, "HPO L"}, + {"HPO Pin", NULL, "HPO R"}, +}; + +static int rt298_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt298->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt298->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt298->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + d_len_code = 0; + switch (params_width(params)) { + /* bit 6:4 Bits per Sample */ + case 16: + d_len_code = 0; + val |= (0x1 << 4); + break; + case 32: + d_len_code = 2; + val |= (0x4 << 4); + break; + case 20: + d_len_code = 1; + val |= (0x2 << 4); + break; + case 24: + d_len_code = 2; + val |= (0x3 << 4); + break; + case 8: + d_len_code = 3; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, + RT298_I2S_CTRL1, 0x0018, d_len_code << 3); + dev_dbg(codec->dev, "format val = 0x%x\n", val); + + snd_soc_update_bits(codec, RT298_DAC_FORMAT, 0x407f, val); + snd_soc_update_bits(codec, RT298_ADC_FORMAT, 0x407f, val); + + return 0; +} + +static int rt298_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, + RT298_I2S_CTRL1, 0x800, 0x800); + break; + case SND_SOC_DAIFMT_CBS_CFS: + snd_soc_update_bits(codec, + RT298_I2S_CTRL1, 0x800, 0x0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_soc_update_bits(codec, + RT298_I2S_CTRL1, 0x300, 0x0); + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_update_bits(codec, + RT298_I2S_CTRL1, 0x300, 0x1 << 8); + break; + case SND_SOC_DAIFMT_DSP_A: + snd_soc_update_bits(codec, + RT298_I2S_CTRL1, 0x300, 0x2 << 8); + break; + case SND_SOC_DAIFMT_DSP_B: + snd_soc_update_bits(codec, + RT298_I2S_CTRL1, 0x300, 0x3 << 8); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + snd_soc_update_bits(codec, RT298_DAC_FORMAT, 0x8000, 0); + snd_soc_update_bits(codec, RT298_ADC_FORMAT, 0x8000, 0); + + return 0; +} + +static int rt298_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq); + + if (RT298_SCLK_S_MCLK == clk_id) { + snd_soc_update_bits(codec, + RT298_I2S_CTRL2, 0x0100, 0x0); + snd_soc_update_bits(codec, + RT298_PLL_CTRL1, 0x20, 0x20); + } else { + snd_soc_update_bits(codec, + RT298_I2S_CTRL2, 0x0100, 0x0100); + snd_soc_update_bits(codec, + RT298_PLL_CTRL, 0x4, 0x4); + snd_soc_update_bits(codec, + RT298_PLL_CTRL1, 0x20, 0x0); + } + + switch (freq) { + case 19200000: + if (RT298_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT298_I2S_CTRL2, 0x40, 0x40); + break; + case 24000000: + if (RT298_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT298_I2S_CTRL2, 0x40, 0x0); + break; + case 12288000: + case 11289600: + snd_soc_update_bits(codec, + RT298_I2S_CTRL2, 0x8, 0x0); + snd_soc_update_bits(codec, + RT298_CLK_DIV, 0xfc1e, 0x0004); + break; + case 24576000: + case 22579200: + snd_soc_update_bits(codec, + RT298_I2S_CTRL2, 0x8, 0x8); + snd_soc_update_bits(codec, + RT298_CLK_DIV, 0xfc1e, 0x5406); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt298->sys_clk = freq; + rt298->clk_id = clk_id; + + return 0; +} + +static int rt298_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + if (50 == ratio) + snd_soc_update_bits(codec, + RT298_I2S_CTRL1, 0x1000, 0x1000); + else + snd_soc_update_bits(codec, + RT298_I2S_CTRL1, 0x1000, 0x0); + + + return 0; +} + +static int rt298_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == + snd_soc_codec_get_bias_level(codec)) { + snd_soc_write(codec, + RT298_SET_AUDIO_POWER, AC_PWRST_D0); + snd_soc_update_bits(codec, 0x0d, 0x200, 0x200); + snd_soc_update_bits(codec, 0x52, 0x80, 0x0); + mdelay(20); + snd_soc_update_bits(codec, 0x0d, 0x200, 0x0); + snd_soc_update_bits(codec, 0x52, 0x80, 0x80); + } + break; + + case SND_SOC_BIAS_ON: + mdelay(30); + snd_soc_update_bits(codec, + RT298_CBJ_CTRL1, 0x0400, 0x0400); + + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, + RT298_SET_AUDIO_POWER, AC_PWRST_D3); + snd_soc_update_bits(codec, + RT298_CBJ_CTRL1, 0x0400, 0x0000); + break; + + default: + break; + } + + return 0; +} + +static irqreturn_t rt298_irq(int irq, void *data) +{ + struct rt298_priv *rt298 = data; + bool hp = false; + bool mic = false; + int ret, status = 0; + + ret = rt298_jack_detect(rt298, &hp, &mic); + + /* Clear IRQ */ + regmap_update_bits(rt298->regmap, RT298_IRQ_CTRL, 0x1, 0x1); + + if (ret == 0) { + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt298->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + pm_wakeup_event(&rt298->i2c->dev, 300); + } + + return IRQ_HANDLED; +} + +static int rt298_probe(struct snd_soc_codec *codec) +{ + struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + + rt298->codec = codec; + + if (rt298->i2c->irq) { + regmap_update_bits(rt298->regmap, + RT298_IRQ_CTRL, 0x2, 0x2); + + INIT_DELAYED_WORK(&rt298->jack_detect_work, + rt298_jack_detect_work); + schedule_delayed_work(&rt298->jack_detect_work, + msecs_to_jiffies(1250)); + } + + return 0; +} + +static int rt298_remove(struct snd_soc_codec *codec) +{ + struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + + cancel_delayed_work_sync(&rt298->jack_detect_work); + + return 0; +} + +#ifdef CONFIG_PM +static int rt298_suspend(struct snd_soc_codec *codec) +{ + struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + + rt298->is_hp_in = -1; + regcache_cache_only(rt298->regmap, true); + regcache_mark_dirty(rt298->regmap); + + return 0; +} + +static int rt298_resume(struct snd_soc_codec *codec) +{ + struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt298->regmap, false); + rt298_index_sync(codec); + regcache_sync(rt298->regmap); + + return 0; +} +#else +#define rt298_suspend NULL +#define rt298_resume NULL +#endif + +#define RT298_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT298_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt298_aif_dai_ops = { + .hw_params = rt298_hw_params, + .set_fmt = rt298_set_dai_fmt, + .set_sysclk = rt298_set_dai_sysclk, + .set_bclk_ratio = rt298_set_bclk_ratio, +}; + +static struct snd_soc_dai_driver rt298_dai[] = { + { + .name = "rt298-aif1", + .id = RT298_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT298_STEREO_RATES, + .formats = RT298_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT298_STEREO_RATES, + .formats = RT298_FORMATS, + }, + .ops = &rt298_aif_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "rt298-aif2", + .id = RT298_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT298_STEREO_RATES, + .formats = RT298_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT298_STEREO_RATES, + .formats = RT298_FORMATS, + }, + .ops = &rt298_aif_dai_ops, + .symmetric_rates = 1, + }, + +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt298 = { + .probe = rt298_probe, + .remove = rt298_remove, + .suspend = rt298_suspend, + .resume = rt298_resume, + .set_bias_level = rt298_set_bias_level, + .idle_bias_off = true, + .controls = rt298_snd_controls, + .num_controls = ARRAY_SIZE(rt298_snd_controls), + .dapm_widgets = rt298_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt298_dapm_widgets), + .dapm_routes = rt298_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt298_dapm_routes), +}; + +static const struct regmap_config rt298_regmap = { + .reg_bits = 32, + .val_bits = 32, + .max_register = 0x02370100, + .volatile_reg = rt298_volatile_register, + .readable_reg = rt298_readable_register, + .reg_write = rl6347a_hw_write, + .reg_read = rl6347a_hw_read, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt298_reg, + .num_reg_defaults = ARRAY_SIZE(rt298_reg), +}; + +static const struct i2c_device_id rt298_i2c_id[] = { + {"rt298", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt298_i2c_id); + +static const struct acpi_device_id rt298_acpi_match[] = { + { "INT343A", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt298_acpi_match); + +static int rt298_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt298_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt298_priv *rt298; + struct device *dev = &i2c->dev; + const struct acpi_device_id *acpiid; + int i, ret; + + pr_info("%s\n", __func__); + + rt298 = devm_kzalloc(&i2c->dev, sizeof(*rt298), + GFP_KERNEL); + if (NULL == rt298) + return -ENOMEM; + + rt298->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt298_regmap); + if (IS_ERR(rt298->regmap)) { + ret = PTR_ERR(rt298->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt298->regmap, + RT298_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret); + if (ret != RT298_VENDOR_ID) { + dev_err(&i2c->dev, + "Device with ID register %#x is not rt298\n", ret); + return -ENODEV; + } + + rt298->index_cache = rt298_index_def; + rt298->index_cache_size = INDEX_CACHE_SIZE; + rt298->i2c = i2c; + i2c_set_clientdata(i2c, rt298); + + /* restore codec default */ + for (i = 0; i < INDEX_CACHE_SIZE; i++) + regmap_write(rt298->regmap, rt298->index_cache[i].reg, + rt298->index_cache[i].def); + for (i = 0; i < ARRAY_SIZE(rt298_reg); i++) + regmap_write(rt298->regmap, rt298_reg[i].reg, + rt298_reg[i].def); + + if (pdata) + rt298->pdata = *pdata; + + /* enable jack combo mode on supported devices */ + acpiid = acpi_match_device(dev->driver->acpi_match_table, dev); + if (acpiid) { + rt298->pdata = *(struct rt298_platform_data *) + acpiid->driver_data; + } + + /* VREF Charging */ + regmap_update_bits(rt298->regmap, 0x04, 0x80, 0x80); + regmap_update_bits(rt298->regmap, 0x1b, 0x860, 0x860); + /* Vref2 */ + regmap_update_bits(rt298->regmap, 0x08, 0x20, 0x20); + + regmap_write(rt298->regmap, RT298_SET_AUDIO_POWER, AC_PWRST_D3); + + for (i = 0; i < RT298_POWER_REG_LEN; i++) + regmap_write(rt298->regmap, + RT298_SET_POWER(rt298_support_power_controls[i]), + AC_PWRST_D1); + + if (!rt298->pdata.cbj_en) { + regmap_write(rt298->regmap, RT298_CBJ_CTRL2, 0x0000); + regmap_write(rt298->regmap, RT298_MIC1_DET_CTRL, 0x0816); + regmap_update_bits(rt298->regmap, + RT298_CBJ_CTRL1, 0xf000, 0xb000); + } else { + regmap_update_bits(rt298->regmap, + RT298_CBJ_CTRL1, 0xf000, 0x5000); + } + + mdelay(10); + + if (!rt298->pdata.gpio2_en) + regmap_write(rt298->regmap, RT298_SET_DMIC2_DEFAULT, 0x4000); + else + regmap_write(rt298->regmap, RT298_SET_DMIC2_DEFAULT, 0); + + mdelay(10); + + regmap_write(rt298->regmap, RT298_MISC_CTRL1, 0x0000); + regmap_update_bits(rt298->regmap, + RT298_WIND_FILTER_CTRL, 0x0082, 0x0082); + regmap_update_bits(rt298->regmap, RT298_IRQ_CTRL, 0x2, 0x2); + rt298->is_hp_in = -1; + + if (rt298->i2c->irq) { + ret = request_threaded_irq(rt298->i2c->irq, NULL, rt298_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt298", rt298); + if (ret != 0) { + dev_err(&i2c->dev, + "Failed to reguest IRQ: %d\n", ret); + return ret; + } + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt298, + rt298_dai, ARRAY_SIZE(rt298_dai)); + + return ret; +} + +static int rt298_i2c_remove(struct i2c_client *i2c) +{ + struct rt298_priv *rt298 = i2c_get_clientdata(i2c); + + if (i2c->irq) + free_irq(i2c->irq, rt298); + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + + +static struct i2c_driver rt298_i2c_driver = { + .driver = { + .name = "rt298", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt298_acpi_match), + }, + .probe = rt298_i2c_probe, + .remove = rt298_i2c_remove, + .id_table = rt298_i2c_id, +}; + +module_i2c_driver(rt298_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT298 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt298.h b/sound/soc/codecs/rt298.h new file mode 100644 index 0000000..31da162 --- /dev/null +++ b/sound/soc/codecs/rt298.h @@ -0,0 +1,206 @@ +/* + * rt298.h -- RT298 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu <johnnyhsu(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT298_H__ +#define __RT298_H__ + +#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) + +#define RT298_AUDIO_FUNCTION_GROUP 0x01 +#define RT298_DAC_OUT1 0x02 +#define RT298_DAC_OUT2 0x03 +#define RT298_DIG_CVT 0x06 +#define RT298_ADC_IN1 0x09 +#define RT298_ADC_IN2 0x08 +#define RT298_MIXER_IN 0x0b +#define RT298_MIXER_OUT1 0x0c +#define RT298_MIXER_OUT2 0x0d +#define RT298_DMIC1 0x12 +#define RT298_DMIC2 0x13 +#define RT298_SPK_OUT 0x14 +#define RT298_MIC1 0x18 +#define RT298_LINE1 0x1a +#define RT298_BEEP 0x1d +#define RT298_SPDIF 0x1e +#define RT298_VENDOR_REGISTERS 0x20 +#define RT298_HP_OUT 0x21 +#define RT298_MIXER_IN1 0x22 +#define RT298_MIXER_IN2 0x23 + +#define RT298_SET_PIN_SFT 6 +#define RT298_SET_PIN_ENABLE 0x40 +#define RT298_SET_PIN_DISABLE 0 +#define RT298_SET_EAPD_HIGH 0x2 +#define RT298_SET_EAPD_LOW 0 + +#define RT298_MUTE_SFT 7 + +/* Verb commands */ +#define RT298_GET_PARAM(NID, PARAM) VERB_CMD(AC_VERB_PARAMETERS, NID, PARAM) +#define RT298_SET_POWER(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, 0) +#define RT298_SET_AUDIO_POWER RT298_SET_POWER(RT298_AUDIO_FUNCTION_GROUP) +#define RT298_SET_HPO_POWER RT298_SET_POWER(RT298_HP_OUT) +#define RT298_SET_SPK_POWER RT298_SET_POWER(RT298_SPK_OUT) +#define RT298_SET_DMIC1_POWER RT298_SET_POWER(RT298_DMIC1) +#define RT298_SPK_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT298_SPK_OUT, 0) +#define RT298_HPO_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT298_HP_OUT, 0) +#define RT298_ADC0_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT298_MIXER_IN1, 0) +#define RT298_ADC1_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT298_MIXER_IN2, 0) +#define RT298_SET_MIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT298_MIC1, 0) +#define RT298_SET_PIN_HPO\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT298_HP_OUT, 0) +#define RT298_SET_PIN_SPK\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT298_SPK_OUT, 0) +#define RT298_SET_PIN_DMIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT298_DMIC1, 0) +#define RT298_SET_PIN_SPDIF\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT298_SPDIF, 0) +#define RT298_SET_PIN_DIG_CVT\ + VERB_CMD(AC_VERB_SET_DIGI_CONVERT_1, RT298_DIG_CVT, 0) +#define RT298_SPK_EAPD\ + VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT298_SPK_OUT, 0) +#define RT298_SET_AMP_GAIN_HPO\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_HP_OUT, 0) +#define RT298_SET_AMP_GAIN_ADC_IN1\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_ADC_IN1, 0) +#define RT298_SET_AMP_GAIN_ADC_IN2\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_ADC_IN2, 0) +#define RT298_GET_HP_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT298_HP_OUT, 0) +#define RT298_GET_MIC1_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT298_MIC1, 0) +#define RT298_SET_DMIC2_DEFAULT\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT298_DMIC2, 0) +#define RT298_SET_SPDIF_DEFAULT\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT298_SPDIF, 0) +#define RT298_DACL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_DAC_OUT1, 0xa000) +#define RT298_DACR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_DAC_OUT1, 0x9000) +#define RT298_ADCL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_ADC_IN1, 0x6000) +#define RT298_ADCR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_ADC_IN1, 0x5000) +#define RT298_MIC_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_MIC1, 0x7000) +#define RT298_SPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_SPK_OUT, 0xa000) +#define RT298_SPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_SPK_OUT, 0x9000) +#define RT298_HPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_HP_OUT, 0xa000) +#define RT298_HPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_HP_OUT, 0x9000) +#define RT298_F_DAC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_MIXER_OUT1, 0x7000) +#define RT298_F_RECMIX_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_MIXER_OUT1, 0x7100) +#define RT298_REC_MIC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_MIXER_IN, 0x7000) +#define RT298_REC_I2S_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_MIXER_IN, 0x7100) +#define RT298_REC_LINE_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_MIXER_IN, 0x7200) +#define RT298_REC_BEEP_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT298_MIXER_IN, 0x7300) +#define RT298_DAC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT298_DAC_OUT1, 0) +#define RT298_ADC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT298_ADC_IN1, 0) +#define RT298_COEF_INDEX\ + VERB_CMD(AC_VERB_SET_COEF_INDEX, RT298_VENDOR_REGISTERS, 0) +#define RT298_PROC_COEF\ + VERB_CMD(AC_VERB_SET_PROC_COEF, RT298_VENDOR_REGISTERS, 0) + +/* Index registers */ +#define RT298_A_BIAS_CTRL1 0x01 +#define RT298_A_BIAS_CTRL2 0x02 +#define RT298_POWER_CTRL1 0x03 +#define RT298_A_BIAS_CTRL3 0x04 +#define RT298_POWER_CTRL2 0x08 +#define RT298_I2S_CTRL1 0x09 +#define RT298_I2S_CTRL2 0x0a +#define RT298_CLK_DIV 0x0b +#define RT298_DC_GAIN 0x0d +#define RT298_POWER_CTRL3 0x0f +#define RT298_MIC1_DET_CTRL 0x19 +#define RT298_MISC_CTRL1 0x20 +#define RT298_IRQ_CTRL 0x33 +#define RT298_WIND_FILTER_CTRL 0x46 +#define RT298_PLL_CTRL1 0x49 +#define RT298_CBJ_CTRL1 0x4f +#define RT298_CBJ_CTRL2 0x50 +#define RT298_PLL_CTRL 0x63 +#define RT298_DEPOP_CTRL1 0x66 +#define RT298_DEPOP_CTRL2 0x67 +#define RT298_DEPOP_CTRL3 0x68 +#define RT298_DEPOP_CTRL4 0x69 + +/* SPDIF (0x06) */ +#define RT298_SPDIF_SEL_SFT 0 +#define RT298_SPDIF_SEL_PCM0 0 +#define RT298_SPDIF_SEL_PCM1 1 +#define RT298_SPDIF_SEL_SPOUT 2 +#define RT298_SPDIF_SEL_PP 3 + +/* RECMIX (0x0b) */ +#define RT298_M_REC_BEEP_SFT 0 +#define RT298_M_REC_LINE1_SFT 1 +#define RT298_M_REC_MIC1_SFT 2 +#define RT298_M_REC_I2S_SFT 3 + +/* Front (0x0c) */ +#define RT298_M_FRONT_DAC_SFT 0 +#define RT298_M_FRONT_REC_SFT 1 + +/* SPK-OUT (0x14) */ +#define RT298_M_SPK_MUX_SFT 14 +#define RT298_SPK_SEL_MASK 0x1 +#define RT298_SPK_SEL_SFT 0 +#define RT298_SPK_SEL_F 0 +#define RT298_SPK_SEL_S 1 + +/* HP-OUT (0x21) */ +#define RT298_M_HP_MUX_SFT 14 +#define RT298_HP_SEL_MASK 0x1 +#define RT298_HP_SEL_SFT 0 +#define RT298_HP_SEL_F 0 +#define RT298_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT298_ADC_SEL_MASK 0x7 +#define RT298_ADC_SEL_SFT 0 +#define RT298_ADC_SEL_SURR 0 +#define RT298_ADC_SEL_FRONT 1 +#define RT298_ADC_SEL_DMIC 2 +#define RT298_ADC_SEL_BEEP 4 +#define RT298_ADC_SEL_LINE1 5 +#define RT298_ADC_SEL_I2S 6 +#define RT298_ADC_SEL_MIC1 7 + +#define RT298_SCLK_S_MCLK 0 +#define RT298_SCLK_S_PLL 1 + +enum { + RT298_AIF1, + RT298_AIF2, + RT298_AIFS, +}; + +int rt298_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + +#endif /* __RT298_H__ */ + -- 1.8.1.1.439.g50a6b54

2 4

[alsa-devel] [PATCH] terminate readlink result string
by Tobias Stoeckmann 13 Jul '15

13 Jul '15

readlink does not guarantee that its result string is nul-terminated. Instead, increase the buffer by one byte to make sure that we can add '\0' at the end. --- alsactl/init_sysfs.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/alsactl/init_sysfs.c b/alsactl/init_sysfs.c index 0cbada2..5c789b6 100644 --- a/alsactl/init_sysfs.c +++ b/alsactl/init_sysfs.c @@ -108,11 +108,11 @@ static char *sysfs_attr_get_value(const char *devpath, const char *attr_name) if (S_ISLNK(statbuf.st_mode)) { /* links return the last element of the target path */ - char link_target[PATH_SIZE]; + char link_target[PATH_SIZE + 1]; int len; const char *pos; - len = readlink(path_full, link_target, sizeof(link_target)); + len = readlink(path_full, link_target, sizeof(link_target) - 1); if (len > 0) { link_target[len] = '\0'; pos = strrchr(link_target, '/'); -- 2.4.5

2 1

[alsa-devel] [PATCH 1/5] regmap: Add better support for devices without readback support
by Lars-Peter Clausen 13 Jul '15

13 Jul '15

Currently regmap requires that a reg_read callback is supplied, otherwise a warning is emitted each time regmap_read() is called. This means a device or bus without readback support needs to supply dummy reg_read callback. Apart from that regmap_read() will still work fine if a cache is used. Remove the warning and let regmap_readable() return false if not reg_read callback is supplied. This means a device no longer has to supply a dummy callback if it does not support readback and it also doesn't have to have a readable_reg callback that always returns false since this is now implicit. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- drivers/base/regmap/regmap.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index dd63bcb..8acc837 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -93,6 +93,9 @@ bool regmap_writeable(struct regmap *map, unsigned int reg) bool regmap_readable(struct regmap *map, unsigned int reg) { + if (!map->reg_read) + return false; + if (map->max_register && reg > map->max_register) return false; @@ -2109,8 +2112,6 @@ static int _regmap_read(struct regmap *map, unsigned int reg, int ret; void *context = _regmap_map_get_context(map); - WARN_ON(!map->reg_read); - if (!map->cache_bypass) { ret = regcache_read(map, reg, val); if (ret == 0) -- 2.1.4

2 9

[alsa-devel] [PATCH 1/2] ASoC: sti-sas: Fix checking return value for ERR_PTR
by Axel Lin 13 Jul '15

13 Jul '15

Both devm_regmap_init and syscon_regmap_lookup_by_phandle return ERR_PTR on failure. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/sti-sas.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index 32db2c2..4168b88 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -568,17 +568,17 @@ static int sti_sas_driver_probe(struct platform_device *pdev) /* Request the DAC & SPDIF registers memory region */ drvdata->dac.virt_regmap = devm_regmap_init(&pdev->dev, NULL, drvdata, drvdata->dev_data->regmap); - if (!drvdata->dac.virt_regmap) { + if (IS_ERR(drvdata->dac.virt_regmap)) { dev_err(&pdev->dev, "audio registers not enabled\n"); - return -EFAULT; + return PTR_ERR(drvdata->dac.virt_regmap); } /* Request the syscon region */ drvdata->dac.regmap = syscon_regmap_lookup_by_phandle(pnode, "st,syscfg"); - if (!drvdata->dac.regmap) { + if (IS_ERR(drvdata->dac.regmap)) { dev_err(&pdev->dev, "syscon registers not available\n"); - return -EFAULT; + return PTR_ERR(drvdata->dac.regmap); } drvdata->spdif.regmap = drvdata->dac.regmap; -- 2.1.0

2 3

[alsa-devel] [PATCH] ASoC: sti-sas: fix platform_no_drv_owner.cocci warnings
by kbuild test robot 13 Jul '15

13 Jul '15

sound/soc/codecs/sti-sas.c:616:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci CC: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- sti-sas.c | 1 - 1 file changed, 1 deletion(-) --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -613,7 +613,6 @@ static int sti_sas_driver_remove(struct static struct platform_driver sti_sas_platform_driver = { .driver = { .name = "sti-sas-codec", - .owner = THIS_MODULE, .of_match_table = sti_sas_dev_match, }, .probe = sti_sas_driver_probe,

2 1

[alsa-devel] Applied "ASoC: sti-sas: Remove spurious dependency on SND_SOC_STI" to the asoc tree
by Mark Brown 13 Jul '15

13 Jul '15

The patch ASoC: sti-sas: Remove spurious dependency on SND_SOC_STI 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 a2f3a8ca5219651292447f0e1124b3412bc3abff Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Mon, 13 Jul 2015 11:53:08 +0100 Subject: [PATCH] ASoC: sti-sas: Remove spurious dependency on SND_SOC_STI Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 46802eff292e..e20834818bcb 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -613,7 +613,6 @@ config SND_SOC_STAC9766 config SND_SOC_STI_SAS tristate "codec Audio support for STI SAS codec" - depends on SND_SOC_STI config SND_SOC_TAS2552 tristate "Texas Instruments TAS2552 Mono Audio amplifier" -- 2.1.4

1 0

[alsa-devel] Applied "ASoC: zx: i2s: Fix devm_ioremap_resource return value check" to the asoc tree
by Mark Brown 13 Jul '15

13 Jul '15

The patch ASoC: zx: i2s: Fix devm_ioremap_resource return value check 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 a18da49ff3ec849c4584ae3abe2b83ff50705ace Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski <k.kozlowski.k(a)gmail.com> Date: Thu, 9 Jul 2015 22:19:28 +0900 Subject: [PATCH] ASoC: zx: i2s: Fix devm_ioremap_resource return value check Value returned by devm_ioremap_resource() was checked for non-NULL but devm_ioremap_resource() returns IOMEM_ERR_PTR, not NULL. In case of error this could lead to dereference of ERR_PTR. Signed-off-by: Krzysztof Kozlowski <k.kozlowski.k(a)gmail.com> Reviewed-by: Jun Nie <jun.nie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/zte/zx296702-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/zte/zx296702-i2s.c b/sound/soc/zte/zx296702-i2s.c index 98d96e1b17e0..1930c42e1f55 100644 --- a/sound/soc/zte/zx296702-i2s.c +++ b/sound/soc/zte/zx296702-i2s.c @@ -393,9 +393,9 @@ static int zx_i2s_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); zx_i2s->mapbase = res->start; zx_i2s->reg_base = devm_ioremap_resource(&pdev->dev, res); - if (!zx_i2s->reg_base) { + if (IS_ERR(zx_i2s->reg_base)) { dev_err(&pdev->dev, "ioremap failed!\n"); - return -EIO; + return PTR_ERR(zx_i2s->reg_base); } writel_relaxed(0, zx_i2s->reg_base + ZX_I2S_FIFO_CTRL); -- 2.1.4

1 0

[alsa-devel] Applied "ASoC: zx: spdif: Fix devm_ioremap_resource return value check" to the asoc tree
by Mark Brown 13 Jul '15

13 Jul '15

The patch ASoC: zx: spdif: Fix devm_ioremap_resource return value check 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 2dfbe9afcda3243dd309826cbbb8ae9e9f602006 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski <k.kozlowski.k(a)gmail.com> Date: Thu, 9 Jul 2015 22:19:29 +0900 Subject: [PATCH] ASoC: zx: spdif: Fix devm_ioremap_resource return value check Value returned by devm_ioremap_resource() was checked for non-NULL but devm_ioremap_resource() returns IOMEM_ERR_PTR, not NULL. In case of error this could lead to dereference of ERR_PTR. Signed-off-by: Krzysztof Kozlowski <k.kozlowski.k(a)gmail.com> Reviewed-by: Jun Nie <jun.nie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/zte/zx296702-spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/zte/zx296702-spdif.c b/sound/soc/zte/zx296702-spdif.c index 11a0e46a1156..26265ce4caca 100644 --- a/sound/soc/zte/zx296702-spdif.c +++ b/sound/soc/zte/zx296702-spdif.c @@ -322,9 +322,9 @@ static int zx_spdif_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); zx_spdif->mapbase = res->start; zx_spdif->reg_base = devm_ioremap_resource(&pdev->dev, res); - if (!zx_spdif->reg_base) { + if (IS_ERR(zx_spdif->reg_base)) { dev_err(&pdev->dev, "ioremap failed!\n"); - return -EIO; + return PTR_ERR(zx_spdif->reg_base); } zx_spdif_dev_init(zx_spdif->reg_base); -- 2.1.4

1 0

[alsa-devel] [PATCH 1/3] ASoC: max98357a: Make 'sdmode-gpios' dts property optional
by Anatol Pomozov 12 Jul '15

12 Jul '15

The option is not needed if chip is always on or managed by some other part of system like platform card driver. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- Documentation/devicetree/bindings/sound/max98357a.txt | 6 +++++- sound/soc/codecs/max98357a.c | 5 ++++- 2 files changed, 9 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/max98357a.txt b/Documentation/devicetree/bindings/sound/max98357a.txt index a7a149a..28645a2 100644 --- a/Documentation/devicetree/bindings/sound/max98357a.txt +++ b/Documentation/devicetree/bindings/sound/max98357a.txt @@ -4,7 +4,11 @@ This node models the Maxim MAX98357A DAC. Required properties: - compatible : "maxim,max98357a" -- sdmode-gpios : GPIO specifier for the GPIO -> DAC SDMODE pin + +Optional properties: +- sdmode-gpios : GPIO specifier for the chip's SD_MODE pin. + If this option is not specified then driver does not manage + the pin state (e.g. chip is always on). Example: diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 3a2fda0..6d4246a 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -31,6 +31,9 @@ static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, { struct gpio_desc *sdmode = snd_soc_dai_get_drvdata(dai); + if (!sdmode) + return 0; + switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: @@ -61,7 +64,7 @@ static int max98357a_codec_probe(struct snd_soc_codec *codec) struct gpio_desc *sdmode; sdmode = devm_gpiod_get(codec->dev, "sdmode", GPIOD_OUT_LOW); - if (IS_ERR(sdmode)) { + if (IS_ERR(sdmode) && (PTR_ERR(sdmode) != -ENOENT)) { dev_err(codec->dev, "%s() unable to get sdmode GPIO: %ld\n", __func__, PTR_ERR(sdmode)); return PTR_ERR(sdmode); -- 2.4.5

2 5

[alsa-devel] When select 2.1 profile in pulseaudio, how do pulseaudio select notebook 's internal subwoofer or 5.1 external speaker ?
by Raymond Yau 12 Jul '15

12 Jul '15

> > > > Are you sure that it is alsa driver bug since pulseaudio seem not using > > line out jack to mute internal speakers ? > > No, I'm not. I set all my jacks to line out and it works now. > Do pulseaudio 2.1 profile select the internal subwoofer or your 5.1 external speaker ? diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 03b7399..c9e45e7 100644 @@ -952,6 +960,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, const struct snd_pci_quirk *q; int id = HDA_FIXUP_ID_NOT_SET; const char *name = NULL; + int i; if (codec->fixup_id != HDA_FIXUP_ID_NOT_SET) return; @@ -966,6 +975,9 @@ void snd_hda_pick_fixup(struct hda_codec *codec, return; } + if (!snd_hda_get_int_hint(codec, "alt_config", &i)) + codec->alt_config = i; + if (codec->modelname && models) { while (models->name) { if (!strcmp(codec->modelname, models->name)) { diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 12837ab..38841db 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -283,6 +283,8 @@ struct hda_codec { /* additional init verbs */ struct snd_array verbs; + int alt_config; + int combo_jack; }; #define dev_to_hda_codec(_dev) container_of(_dev, struct hda_codec, core.dev) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index ac0db16..12ee7f7 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -39,7 +39,6 @@ #include "hda_beep.h" #include "hda_generic.h" - /** * snd_hda_gen_spec_init - initialize hda_gen_spec struct * @spec: hda_gen_spec object to initialize @@ -180,10 +179,10 @@ static void parse_user_hints(struct hda_codec *codec) /* the following two are just for compatibility */ val = snd_hda_get_bool_hint(codec, "add_out_jack_modes"); if (val >= 0) - spec->add_jack_modes = !!val; + spec->add_out_jack_modes = !!val; val = snd_hda_get_bool_hint(codec, "add_in_jack_modes"); if (val >= 0) - spec->add_jack_modes = !!val; + spec->add_in_jack_modes = !!val; val = snd_hda_get_bool_hint(codec, "add_jack_modes"); if (val >= 0) spec->add_jack_modes = !!val; @@ -2648,7 +2720,7 @@ static void get_jack_mode_name(struct hda_codec *codec, hda_nid_t pin, static int get_out_jack_num_items(struct hda_codec *codec, hda_nid_t pin) { struct hda_gen_spec *spec = codec->spec; - if (spec->add_jack_modes) { + if (spec->add_jack_modes || spec->add_out_jack_modes) { unsigned int pincap = snd_hda_query_pin_caps(codec, pin); if ((pincap & AC_PINCAP_OUT) && (pincap & AC_PINCAP_HP_DRV)) return 2; @@ -2791,7 +2863,7 @@ static int get_in_jack_num_items(struct hda_codec *codec, hda_nid_t pin) { struct hda_gen_spec *spec = codec->spec; int nitems = 0; - if (spec->add_jack_modes) + if (spec->add_jack_modes || spec->add_in_jack_modes) nitems = hweight32(get_vref_caps(codec, pin)); return nitems ? nitems : 1; } @@ -3287,7 +3359,7 @@ static int create_input_ctls(struct hda_codec *codec) if (err < 0) return err; - if (spec->add_jack_modes) { + if (spec->add_jack_modes || spec->add_in_jack_modes) { err = create_in_jack_mode(codec, pin); if (err < 0) return err; @@ -4895,13 +4971,13 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, */ if (spec->hp_mic_pin && (spec->auto_mic || spec->input_mux.num_items == 1 || - spec->add_jack_modes)) { + spec->add_jack_modes || spec->add_in_jack_modes)) { err = create_hp_mic_jack_mode(codec, spec->hp_mic_pin); if (err < 0) return err; } - if (spec->add_jack_modes) { + if (spec->add_jack_modes || spec->add_out_jack_modes) { if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { err = create_out_jack_modes(codec, cfg->line_outs, cfg->line_out_pins); diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 56e4139..f10c55b 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -227,6 +229,8 @@ struct hda_gen_spec { unsigned int prefer_hp_amp:1; /* enable HP amp for speaker if any */ unsigned int add_stereo_mix_input:2; /* add aamix as a capture src */ unsigned int add_jack_modes:1; /* add i/o jack mode enum ctls */ + unsigned int add_in_jack_modes:1; /* add in jack mode enum ctls */ + unsigned int add_out_jack_modes:1; /* add out jack mode enum ctls */ unsigned int power_down_unused:1; /* power down unused widgets */ unsigned int dac_min_mute:1; /* minimal = mute for DACs */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8e02cdf..b13f588 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1914,6 +1914,62 @@ static void alc889_fixup_mba21_vref(struct hda_codec *codec, alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids)); } +static const struct snd_pcm_chmap_elem acer_pcm_chmaps[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_LFE, SNDRV_CHMAP_NA } }, /* LFE only on left */ + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { .channels = 6, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR, + SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE } }, + { } +}; + +static void alc888_fixup_acer_aspire_4930g(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const struct hda_pintbl subwoofer_cfg[] = { + { 0x16, 0x90170111 }, /* internal subwoofer */ + {} + }; + static const struct hda_pintbl front_cfg[] = { + { 0x15, 0x02014010}, + {} + }; + static const struct hda_pintbl rear_cfg[] = { + { 0x1a, 0x02014011}, + {} + }; + static const struct hda_pintbl clfe_cfg[] = { + { 0x18, 0x02014012}, + {} + }; + struct alc_spec *spec = codec->spec; + switch(action){ + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, subwoofer_cfg); + spec->gen.add_out_jack_modes = true; + switch(codec->alt_config){ + case 3: + snd_hda_apply_pincfgs(codec, clfe_cfg); + case 2: + snd_hda_apply_pincfgs(codec, rear_cfg); + case 1: + snd_hda_apply_pincfgs(codec, front_cfg); + break; + } + break; + case HDA_FIXUP_ACT_BUILD: + spec->gen.pcm_rec[0]->stream[0].chmap = acer_pcm_chmaps; + break; + } +} + /* Don't take HP output as primary * Strangely, the speaker output doesn't work on Vaio Z and some Vaio * all-in-one desktop PCs (for example VGC-LN51JGB) through DAC 0x05 @@ -2048,12 +2104,8 @@ static const struct hda_fixup alc882_fixups[] = { .v.func = alc889_fixup_coef, }, [ALC882_FIXUP_ACER_ASPIRE_4930G] = { - .type = HDA_FIXUP_PINS, - .v.pins = (const struct hda_pintbl[]) { - { 0x16, 0x99130111 }, /* CLFE speaker */ - { 0x17, 0x99130112 }, /* surround speaker */ - { } - }, + .type = HDA_FIXUP_FUNC, + .v.func = alc888_fixup_acer_aspire_4930g, .chained = true, .chain_id = ALC882_FIXUP_GPIO1, },

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[alsa-devel] How to activate DAI link to dummy codec?
by Anatol Pomozov 12 Jul '15

12 Jul '15

Hi I have following hardware configuration [SoC] <=> [RT5677] <=> [MAX98357A] MAX98357A is a simple speaker amplifier with I2S data interface and no I2C control. The chip automatically configures itself (rate/block frequency) when I2S is started streaming to the chip. This chip does not need a driver (there is nothing to configure) and we can use "dummy-codec" for it. There is a DAI link between RT5677 and MAX and it should be configured from RT5677 side. RT5677 is master here and it need to set I2S bus parameters. Configuration happens in rt5677 hw_params() function and it is triggered by dai link activation. My question is how to tell DAPM that I need to activate this link? If MAX98357A was a real codec then it would have an OUTPUT widget that I can activate, this would trigger dai link in between to be activated as well. But dummy codec does not have any widgets, so how I suppose to activate link to dummy codec?

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[alsa-devel] [asoc:topic/sti-sas 1/1] sound/soc/codecs/sti-sas.c:227:2: error: implicit declaration of function 'devm_reset_control_get'
by kbuild test robot 11 Jul '15

11 Jul '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/sti-sas head: 32a726b2e089ec1851965290a610c4ae9cab3303 commit: 32a726b2e089ec1851965290a610c4ae9cab3303 [1/1] ASoC: sti-sas: Add sti platform codec config: i386-randconfig-x0-07112203 (attached as .config) reproduce: git checkout 32a726b2e089ec1851965290a610c4ae9cab3303 # save the attached .config to linux build tree make ARCH=i386 All error/warnings (new ones prefixed by >>): sound/soc/codecs/sti-sas.c: In function 'stih416_dac_probe': >> sound/soc/codecs/sti-sas.c:227:2: error: implicit declaration of function 'devm_reset_control_get' [-Werror=implicit-function-declaration] dac->rst = devm_reset_control_get(codec->dev, "dac_rst"); ^ >> sound/soc/codecs/sti-sas.c:227:11: warning: assignment makes pointer from integer without a cast dac->rst = devm_reset_control_get(codec->dev, "dac_rst"); ^ cc1: some warnings being treated as errors vim +/devm_reset_control_get +227 sound/soc/codecs/sti-sas.c 221 { 222 struct snd_soc_codec *codec = dai->codec; 223 struct sti_sas_data *drvdata = dev_get_drvdata(codec->dev); 224 struct sti_dac_audio *dac = &drvdata->dac; 225 226 /* Get reset control */ > 227 dac->rst = devm_reset_control_get(codec->dev, "dac_rst"); 228 if (IS_ERR(dac->rst)) { 229 dev_err(dai->codec->dev, 230 "%s: ERROR: DAC reset control not defined (%d)!\n", --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:topic/sti-sas 1/1] sound/soc/codecs/sti-sas.c:558:2: note: in expansion of macro 'if'
by kbuild test robot 11 Jul '15

11 Jul '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/sti-sas head: 32a726b2e089ec1851965290a610c4ae9cab3303 commit: 32a726b2e089ec1851965290a610c4ae9cab3303 [1/1] ASoC: sti-sas: Add sti platform codec config: i386-randconfig-x0-07111645 (attached as .config) reproduce: git checkout 32a726b2e089ec1851965290a610c4ae9cab3303 # save the attached .config to linux build tree make ARCH=i386 All warnings (new ones prefixed by >>): In file included from include/uapi/linux/stddef.h:1:0, from include/linux/stddef.h:4, from include/uapi/linux/posix_types.h:4, from include/uapi/linux/types.h:13, from include/linux/types.h:5, from include/linux/io.h:21, from sound/soc/codecs/sti-sas.c:8: sound/soc/codecs/sti-sas.c: In function 'sti_sas_driver_probe': sound/soc/codecs/sti-sas.c:558:46: warning: dereferencing 'void *' pointer if (!of_match_node(sti_sas_dev_match, pnode)->data) { ^ include/linux/compiler.h:147:28: note: in definition of macro '__trace_if' if (__builtin_constant_p((cond)) ? !!(cond) : \ ^ >> sound/soc/codecs/sti-sas.c:558:2: note: in expansion of macro 'if' if (!of_match_node(sti_sas_dev_match, pnode)->data) { ^ sound/soc/codecs/sti-sas.c:558:46: error: request for member 'data' in something not a structure or union if (!of_match_node(sti_sas_dev_match, pnode)->data) { ^ include/linux/compiler.h:147:28: note: in definition of macro '__trace_if' if (__builtin_constant_p((cond)) ? !!(cond) : \ ^ >> sound/soc/codecs/sti-sas.c:558:2: note: in expansion of macro 'if' if (!of_match_node(sti_sas_dev_match, pnode)->data) { ^ sound/soc/codecs/sti-sas.c:558:46: warning: dereferencing 'void *' pointer if (!of_match_node(sti_sas_dev_match, pnode)->data) { ^ include/linux/compiler.h:147:40: note: in definition of macro '__trace_if' if (__builtin_constant_p((cond)) ? !!(cond) : \ ^ >> sound/soc/codecs/sti-sas.c:558:2: note: in expansion of macro 'if' if (!of_match_node(sti_sas_dev_match, pnode)->data) { ^ sound/soc/codecs/sti-sas.c:558:46: error: request for member 'data' in something not a structure or union if (!of_match_node(sti_sas_dev_match, pnode)->data) { ^ include/linux/compiler.h:147:40: note: in definition of macro '__trace_if' if (__builtin_constant_p((cond)) ? !!(cond) : \ ^ >> sound/soc/codecs/sti-sas.c:558:2: note: in expansion of macro 'if' if (!of_match_node(sti_sas_dev_match, pnode)->data) { ^ sound/soc/codecs/sti-sas.c:558:46: warning: dereferencing 'void *' pointer if (!of_match_node(sti_sas_dev_match, pnode)->data) { ^ include/linux/compiler.h:158:16: note: in definition of macro '__trace_if' ______r = !!(cond); \ ^ >> sound/soc/codecs/sti-sas.c:558:2: note: in expansion of macro 'if' if (!of_match_node(sti_sas_dev_match, pnode)->data) { ^ sound/soc/codecs/sti-sas.c:558:46: error: request for member 'data' in something not a structure or union if (!of_match_node(sti_sas_dev_match, pnode)->data) { ^ include/linux/compiler.h:158:16: note: in definition of macro '__trace_if' ______r = !!(cond); \ ^ >> sound/soc/codecs/sti-sas.c:558:2: note: in expansion of macro 'if' if (!of_match_node(sti_sas_dev_match, pnode)->data) { ^ sound/soc/codecs/sti-sas.c:563:61: warning: dereferencing 'void *' pointer drvdata->dev_data = of_match_node(sti_sas_dev_match, pnode)->data; ^ sound/soc/codecs/sti-sas.c:563:61: error: request for member 'data' in something not a structure or union vim +/if +558 sound/soc/codecs/sti-sas.c 542 }, 543 {}, 544 }; 545 546 static int sti_sas_driver_probe(struct platform_device *pdev) 547 { 548 struct device_node *pnode = pdev->dev.of_node; 549 struct sti_sas_data *drvdata; 550 551 /* Allocate device structure */ 552 drvdata = devm_kzalloc(&pdev->dev, sizeof(struct sti_sas_data), 553 GFP_KERNEL); 554 if (!drvdata) 555 return -ENOMEM; 556 557 /* Populate data structure depending on compatibility */ > 558 if (!of_match_node(sti_sas_dev_match, pnode)->data) { 559 dev_err(&pdev->dev, "data associated to device is missing"); 560 return -EINVAL; 561 } 562 563 drvdata->dev_data = of_match_node(sti_sas_dev_match, pnode)->data; 564 565 /* Initialise device structure */ 566 drvdata->dev = &pdev->dev; --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:topic/sti-sas 1/1] sound/soc/codecs/sti-sas.c:558:46: error: request for member 'data' in something not a structure or union
by kbuild test robot 11 Jul '15

11 Jul '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/sti-sas head: 32a726b2e089ec1851965290a610c4ae9cab3303 commit: 32a726b2e089ec1851965290a610c4ae9cab3303 [1/1] ASoC: sti-sas: Add sti platform codec config: x86_64-randconfig-a0-07111354 (attached as .config) reproduce: git checkout 32a726b2e089ec1851965290a610c4ae9cab3303 # save the attached .config to linux build tree make ARCH=x86_64 All error/warnings (new ones prefixed by >>): sound/soc/codecs/sti-sas.c: In function 'stih416_dac_probe': sound/soc/codecs/sti-sas.c:231:14: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast] __func__, (int)dac->rst); ^ sound/soc/codecs/sti-sas.c: In function 'sti_sas_driver_probe': >> sound/soc/codecs/sti-sas.c:558:46: warning: dereferencing 'void *' pointer if (!of_match_node(sti_sas_dev_match, pnode)->data) { ^ >> sound/soc/codecs/sti-sas.c:558:46: error: request for member 'data' in something not a structure or union sound/soc/codecs/sti-sas.c:563:61: warning: dereferencing 'void *' pointer drvdata->dev_data = of_match_node(sti_sas_dev_match, pnode)->data; ^ sound/soc/codecs/sti-sas.c:563:61: error: request for member 'data' in something not a structure or union vim +/data +558 sound/soc/codecs/sti-sas.c 552 drvdata = devm_kzalloc(&pdev->dev, sizeof(struct sti_sas_data), 553 GFP_KERNEL); 554 if (!drvdata) 555 return -ENOMEM; 556 557 /* Populate data structure depending on compatibility */ > 558 if (!of_match_node(sti_sas_dev_match, pnode)->data) { 559 dev_err(&pdev->dev, "data associated to device is missing"); 560 return -EINVAL; 561 } --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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Re: [alsa-devel] [pulseaudio-discuss] Strange hardware: XFX nForce 780i SLI MCP
by Raymond Yau 11 Jul '15

11 Jul '15

> Yesterday I got an interesting email related to dcaenc support. > > The person has an "XFX nForce 780i SLI" MCP board: > http://www.nvidia.com/object/product_nforce_780i_sli_us.html , > http://www.evga.com/support/manuals/files/132-CK-NF78.pdf > > As you see on page 18 of the manual, this board has no HDMI output, but > there is an optical SPDIF. Strangely enough, this SPDIF is his > "hdmi:CARD=NVidia,DEV=0" device. > > Given the recent discussions related to dynamically creating PCM devices > for HDMI, I think you will be interested in this alsa-info output. > > Is there anything here that would need special treatment by ALSA or > PulseAudio? Or, is this just a misnumbered/misnamed codec? > swap dig_out_pins of autocfg when type are different diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 03b7399..3ff16e8 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -405,6 +405,19 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, sort(cfg->inputs, cfg->num_inputs, sizeof(cfg->inputs[0]), compare_input_type, NULL); + if (cfg->dig_outs == 2) { + if (cfg->dig_out_type[0] != cfg->dig_out_type[1]) { + if (cfg->dig_out_type[0] == HDA_PCM_TYPE_HDMI) { + hda_nid_t temp = cfg->dig_out_pins[0]; + int temp_type = cfg->dig_out_type[0]; + cfg->dig_out_pins[0] = cfg->dig_out_pins[1]; + cfg->dig_out_type[0] = cfg->dig_out_type[1]; + cfg->dig_out_pins[1] = temp; + cfg->dig_out_type[1] = temp_type; + } + } + } + /* * debug prints of the parsed results */ > >

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Re: [alsa-devel] [RFT PATCH 1/2] ASoC: zx: i2s: Fix devm_ioremap_resource return value check
by Jun Nie 11 Jul '15

11 Jul '15

2015-07-09 21:19 GMT+08:00 Krzysztof Kozlowski <k.kozlowski.k(a)gmail.com>: > Value returned by devm_ioremap_resource() was checked for non-NULL but > devm_ioremap_resource() returns IOMEM_ERR_PTR, not NULL. In case of > error this could lead to dereference of ERR_PTR. > > Signed-off-by: Krzysztof Kozlowski <k.kozlowski.k(a)gmail.com> > Cc: <stable(a)vger.kernel.org> > Fixes: e5d4cd87800c ("ASoC: zx: Add ZTE zx296702 I2S DAI driver") > > --- > > Patch only compile tested. > --- > sound/soc/zte/zx296702-i2s.c | 4 ++-- > 1 file changed, 2 insertions(+), 2 deletions(-) > > diff --git a/sound/soc/zte/zx296702-i2s.c b/sound/soc/zte/zx296702-i2s.c > index 98d96e1b17e0..1930c42e1f55 100644 > --- a/sound/soc/zte/zx296702-i2s.c > +++ b/sound/soc/zte/zx296702-i2s.c > @@ -393,9 +393,9 @@ static int zx_i2s_probe(struct platform_device *pdev) > res = platform_get_resource(pdev, IORESOURCE_MEM, 0); > zx_i2s->mapbase = res->start; > zx_i2s->reg_base = devm_ioremap_resource(&pdev->dev, res); > - if (!zx_i2s->reg_base) { > + if (IS_ERR(zx_i2s->reg_base)) { > dev_err(&pdev->dev, "ioremap failed!\n"); > - return -EIO; > + return PTR_ERR(zx_i2s->reg_base); > } > > writel_relaxed(0, zx_i2s->reg_base + ZX_I2S_FIFO_CTRL); > -- > 2.1.4 > Reviewed-by: Jun Nie <jun.nie(a)linaro.org>

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Re: [alsa-devel] [RFT PATCH 2/2] ASoC: zx: spdif: Fix devm_ioremap_resource return value check
by Jun Nie 11 Jul '15

11 Jul '15

2015-07-09 21:19 GMT+08:00 Krzysztof Kozlowski <k.kozlowski.k(a)gmail.com>: > Value returned by devm_ioremap_resource() was checked for non-NULL but > devm_ioremap_resource() returns IOMEM_ERR_PTR, not NULL. In case of > error this could lead to dereference of ERR_PTR. > > Signed-off-by: Krzysztof Kozlowski <k.kozlowski.k(a)gmail.com> > Cc: <stable(a)vger.kernel.org> > Fixes: 6fc3d24d4277 ("ASoC: zx: Add zx296702 SPDIF support") > > --- > > Patch only compile tested. > --- > sound/soc/zte/zx296702-spdif.c | 4 ++-- > 1 file changed, 2 insertions(+), 2 deletions(-) > > diff --git a/sound/soc/zte/zx296702-spdif.c b/sound/soc/zte/zx296702-spdif.c > index 11a0e46a1156..26265ce4caca 100644 > --- a/sound/soc/zte/zx296702-spdif.c > +++ b/sound/soc/zte/zx296702-spdif.c > @@ -322,9 +322,9 @@ static int zx_spdif_probe(struct platform_device *pdev) > res = platform_get_resource(pdev, IORESOURCE_MEM, 0); > zx_spdif->mapbase = res->start; > zx_spdif->reg_base = devm_ioremap_resource(&pdev->dev, res); > - if (!zx_spdif->reg_base) { > + if (IS_ERR(zx_spdif->reg_base)) { > dev_err(&pdev->dev, "ioremap failed!\n"); > - return -EIO; > + return PTR_ERR(zx_spdif->reg_base); > } > > zx_spdif_dev_init(zx_spdif->reg_base); > -- > 2.1.4 > Reviewed-by: Jun Nie <jun.nie(a)linaro.org>

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Re: [alsa-devel] Fwd: Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 11 Jul '15

11 Jul '15

> > > > http://www.alsa-project.org/db/?f=30b3f0087374b914a20dbe20a618fb892a5d6fd5 > > > > > > Are there any offical specificaion ? > > > > > > > > So far most review only mention stereo speakers and subwoofer, the > > > > service > > > > > > guide only show how to replace two internal speakers > > > > > > Service manual mentions > > > > > > "Dolby®-certified surround sound system with two built-in stereo > > > > speakers and > > > > > one subwoofer supporting low-frequency effects" > > > > > > so all additional channels what I'm hearing in Windows is emulated. I hope > > > that it is clear now that my laptop has two internal speakers and one > > > subwoofer. > > > > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pc > > i/hda?id=8e38395360844806041ea69ab9690f5f174bc40c > > > > You need to define your 2.1 chmap if the subwoofer only use left channel > > > > Compiling and running PulseAudio from git > > > > http://colin.guthr.ie/2010/09/compiling-and-running-pulseaudio-from-git/ > > In Fedora 22 workaround with setting "Line out (front)" for 0x15 pin doesn't > work. > > Sound on my system is totally fucked up. http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… Are you sure that it is alsa driver bug since pulseaudio seem not using line out jack to mute internal speakers ? by adding new hint alt_config , you can use alt_config to select different set of pin config diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 03b7399..683a0ab 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -952,6 +952,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, const struct snd_pci_quirk *q; int id = HDA_FIXUP_ID_NOT_SET; const char *name = NULL; + int i; if (codec->fixup_id != HDA_FIXUP_ID_NOT_SET) return; @@ -966,6 +967,9 @@ void snd_hda_pick_fixup(struct hda_codec *codec, return; } + if (!snd_hda_get_int_hint(codec, "alt_config", &i)) + codec->alt_config = i; + if (codec->modelname && models) { while (models->name) { if (!strcmp(codec->modelname, models->name)) { diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8e02cdf..f406835 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1914,6 +1914,31 @@ static void alc889_fixup_mba21_vref(struct hda_codec *codec, alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids)); } +static void alc888_fixup_acer_aspire_4930g(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct hda_pintbl subwoofer_config[] = { + { 0x16, 0x90170111 }, /* internal subwoofer */ + {} + }; + struct hda_pintbl alt_config[] = { + { 0x15, 0x02014010}, + { 0x1a, 0x02014011}, + { 0x18, 0x02014012}, + {} + }; + snd_hda_apply_pincfgs(codec, subwoofer_config); + switch(codec->alt_config){ + case 1: + codec_info(codec, "5.1 External Speaker\n"); + snd_hda_apply_pincfgs(codec, alt_config); + break; + default: + codec_info(codec, "use [Hint]\nalt_config=1 for 5.1 External Speaker\n"); + break; + } +} + /* Don't take HP output as primary * Strangely, the speaker output doesn't work on Vaio Z and some Vaio * all-in-one desktop PCs (for example VGC-LN51JGB) through DAC 0x05 @@ -2048,12 +2073,8 @@ static const struct hda_fixup alc882_fixups[] = { .v.func = alc889_fixup_coef, }, [ALC882_FIXUP_ACER_ASPIRE_4930G] = { - .type = HDA_FIXUP_PINS, - .v.pins = (const struct hda_pintbl[]) { - { 0x16, 0x99130111 }, /* CLFE speaker */ - { 0x17, 0x99130112 }, /* surround speaker */ - { } - }, + .type = HDA_FIXUP_FUNC, + .v.func = alc888_fixup_acer_aspire_4930g, .chained = true, .chain_id = ALC882_FIXUP_GPIO1, },

1 0

[alsa-devel] [Alsa-user] HBR audio not working with radeon/caicos below 60Hz
by Dominique Dumont 10 Jul '15

10 Jul '15

Hello Unfortunately, I cannot get HBR audio (DTS-HD and dolby TrueHD) on my system with all refresh rates. Only 59 and 60 Hz refresh rates are working more or less properly: there's a dropout every now and then. Refresh rates below that are silent: no error is reported by kodi or by the kernel but my yamaha amp does not detect the audio stream. My yamaha amplifier is connected by HDMI to a radeon Caicos card. See [1] for the output of alsa-info.sh. DTS, dolby digital and DD-plus are working fine. This is a known issue [2] for kodi people. A bug [3] was reported on freedesktop.org to no avail. Can this issue be solved by alsa people or is this more an xorg bug ? Feel free to get back to me if you need more info. All the best -- https://github.com/dod38fr/config-model/ -o- http://search.cpan.org/~ddumont/ http://ddumont.wordpress.com/ -o- irc: dod at irc.debian.org [1] http://www.alsa-project.org/db/?f=4c59e10da7600d45ccff18ea0f655e2e97ce9ae6 [2] http://forum.kodi.tv/showthread.php?tid=174854 [3] https://bugs.freedesktop.org/show_bug.cgi?id=75064

1 0

[alsa-devel] [PATCH] ASoC: arizona: Fix error path in codec probe
by Charles Keepax 10 Jul '15

10 Jul '15

If we fail to add some DSPs or fail to add the controls we should call wm_adsp2_codec_remove for all the cores we have already added. This patch fixes this up on the wm5102 and wm5110. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 7 ++++++- sound/soc/codecs/wm5110.c | 10 ++++++++-- 2 files changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 6ddee99..64637d1 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1879,7 +1879,7 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) ret = snd_soc_add_codec_controls(codec, arizona_adsp2_rate_controls, 1); if (ret) - return ret; + goto err_adsp2_codec_probe; arizona_init_spk(codec); arizona_init_gpio(codec); @@ -1889,6 +1889,11 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) priv->core.arizona->dapm = dapm; return 0; + +err_adsp2_codec_probe: + wm_adsp2_codec_remove(&priv->core.adsp[0], codec); + + return ret; } static int wm5102_codec_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 429be44..df6a998 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1874,18 +1874,24 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) for (i = 0; i < WM5110_NUM_ADSP; ++i) { ret = wm_adsp2_codec_probe(&priv->core.adsp[i], codec); if (ret) - return ret; + goto err_adsp2_codec_probe; } ret = snd_soc_add_codec_controls(codec, arizona_adsp2_rate_controls, WM5110_NUM_ADSP); if (ret) - return ret; + goto err_adsp2_codec_probe; snd_soc_dapm_disable_pin(dapm, "HAPTICS"); return 0; + +err_adsp2_codec_probe: + for (--i; i >= 0; --i) + wm_adsp2_codec_remove(&priv->core.adsp[i], codec); + + return ret; } static int wm5110_codec_remove(struct snd_soc_codec *codec) -- 1.7.2.5

2 1

[alsa-devel] [PATCH 0/7] asoc: Add audio for sti platforms
by Arnaud Pouliquen 10 Jul '15

10 Jul '15

version 1: - First patches submission Set of patches concerns the audio part of the STi platform Hardware is split in two main blocks: Unipheriperal IPs and codec IPs Uniperipheral IPs: Ips in charge of the I2S buses Uniperipheral IPs are in charge the PCM rendering/capturing on/from I2S. - 4 uniperipheral player IPs receive the PCM audio data or audio encoded bit stream data from the memory (via FDMA)and performs optional IEC-60958 or IEC-61937 formatting on the audio data and sends them out on the I2S output. - 2 uniperipheral reader IPS receive the PCM audio or encoded bit stream on the I2S input interface and writes the processed data in memory (via FDMA) Codec IPs: - HDMI_TX: format and generate HDMI signal. - SPDIF: format and generate biphase signal for SPDIF output - DAC: format and generate stereo analogic signal - HDMI_RX: receive HDMI signal from external source. In addition 2 I2S/TDM interfaces are available to connect external ICs - PCM_IN to external capture codec (ADC, BT...) - PCM_OUT for external player codec (DAC, BT..) On Sti platform audio driver is splitted in 2 part to support Hw architecture - sti_audio_platform for uniperipheral IPs mmannagement - sti-sas codec for codec IP management +-----------------------+ +-----------------+ | +-----------------+ | | | | | +---------+ | | | +---------+ | | | |Uniperif | | | | | HDMI | | +---+ | | |player 0 +---------------> TX +-------> | | | +---------+ | | | +---------+ | +---+ | | +---------+ | | | | | | |Uniperif | | | | | +---+ | | |player 1 +---------------------------------> | | | +---------+ | | | | +---+ | | +---------+ | | | +---------+ | +---+ | | |Uniperif | | | | | DAC | | | | | | |player 2 +---------------> +------^----+ | | +---------+ | | | +---------+ | | | +---------+ | | | +---------+ | | | |Uniperif | | | | | SPDIF | | +---+ | | |player 3 +---------------> +-------> | | | +---------+ | | | +---------+ | +---+ | | +---------+ | | | | | | |Uniperif | | | | | +---+ | | |reader 0 <---------------------------------+ | | | +---------+ | | | | +---+ | | +---------+ | | | +---------+ | | | |Uniperif | | | | | HDMIF | | +---+ | | |reader 1 <---------------+ RX <-------+ | | | +---------+ | | | +---------+ | +---+ | | | | | | | | CPU DAI | | | | | +-----------------+ | | | | STI platform | | SAS CODEC | +-----------------------+ +-----------------+ In the first version of the driver only SPDIF and DAC are supported. in future HDMI TX, HDMI_RX and support for TDM mode should be added. Arnaud Pouliquen (7): ASoC: sti: add binding for ASoc driver Asoc: sti: add uniperipheral header file Asoc: sti: add CPU DAI driver for playback Asoc: sti: add CPU DAI driver for capture Asoc: sti: Add platform driver ASoc: Add ability to build sti drivers ASoc: Codec: add sti platform codec .../devicetree/bindings/sound/st,sti-asoc-card.txt | 176 +++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sti-sas.c | 663 ++++++++++ sound/soc/sti/Kconfig | 17 + sound/soc/sti/Makefile | 5 + sound/soc/sti/sti_platform.c | 643 +++++++++ sound/soc/sti/uniperif.h | 1236 ++++++++++++++++++ sound/soc/sti/uniperif_player.c | 1367 ++++++++++++++++++++ sound/soc/sti/uniperif_reader.c | 493 +++++++ 12 files changed, 4608 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt create mode 100644 sound/soc/codecs/sti-sas.c create mode 100644 sound/soc/sti/Kconfig create mode 100644 sound/soc/sti/Makefile create mode 100644 sound/soc/sti/sti_platform.c create mode 100644 sound/soc/sti/uniperif.h create mode 100644 sound/soc/sti/uniperif_player.c create mode 100644 sound/soc/sti/uniperif_reader.c -- 1.9.1

2 18

Re: [alsa-devel] Intel Baytrail Linux Audio
by Vinod Koul 10 Jul '15

10 Jul '15

On Wed, Jul 08, 2015 at 10:32:01PM +0200, Luka Karinja wrote: > Hello guys. I have been trying to get sound on my Asus T100TAF with rt5642. > I have been following your various threads but i'm not succesfull > with getting sound working. > Hoppfully you can give me a hand > > The changes i made to the kernel: > > diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c > index bb19b58..b45cf3e 100644 > --- a/sound/soc/intel/atom/sst/sst_acpi.c > +++ b/sound/soc/intel/atom/sst/sst_acpi.c > @@ -118,7 +118,7 @@ static const struct sst_res_info byt_rvp_res_info = { > .mbox_size = 0x1000, > .acpi_lpe_res_index = 0, > .acpi_ddr_index = 2, > - .acpi_ipc_irq_index = 5, > + .acpi_ipc_irq_index = 0, irq index in deafult shoudl work for t100, you should see interrupts > }; > static struct sst_platform_info byt_rvp_platform_data = { > @@ -345,6 +345,8 @@ static int sst_acpi_remove(struct platform_device *pdev) > static struct sst_machines sst_acpi_bytcr[] = { > {"10EC5640", "T100", "bytt100_rt5640", NULL, "intel/fw_sst_0f28.bin", > &byt_rvp_platform_data }, > + {"10EC5642", "T100", "bytt100_rt5640", NULL, "intel/fw_sst_0f28_ssp0.bin", > + &byt_rvp_platform_data }, That wont work, I have verfied with T100, it is SSP2 based, so SSP0 will nto work here. I am trying to get this tested on latest driver and provide a working config. Will update shortly -- ~Vinod

2 1

Re: [alsa-devel] Possible Bug Found with Xonar Stx Driver
by Clemens Ladisch 10 Jul '15

10 Jul '15

nick wrote: > After running the latest version of Linus's tree I am noticed way more buffering What exactly do you mean with the word "buffering"? > when playing music with my Xonar STX card when building kernels then > under the Ubuntu kernel version of 3.19.16. Sounds like a change in the disk access patterns. The audio buffer size is chosen by the playing application (whatever it is). > I have tried bisecting it between the releases of 3.18 and 4.0 > and was unable to narrow it down to a buggy commit. There was no relevant change in the sound driver. Regards, Clemens

2 1

[alsa-devel] Applied "regmap: add force_write option on _regmap_update_bits()" to the asoc tree
by Mark Brown 10 Jul '15

10 Jul '15

The patch regmap: add force_write option on _regmap_update_bits() 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 7ff0589c7bff4ca31b255ac2028f633f14047762 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 16 Jun 2015 08:52:22 +0000 Subject: [PATCH] regmap: add force_write option on _regmap_update_bits() Sometimes we want to write data even though it doesn't change value. Then, force_write option on _regmap_update_bits() helps this purpose. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/base/regmap/regmap.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index 7111d04f2621..69ec411ce722 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -34,7 +34,7 @@ static int _regmap_update_bits(struct regmap *map, unsigned int reg, unsigned int mask, unsigned int val, - bool *change); + bool *change, bool force_write); static int _regmap_bus_reg_read(void *context, unsigned int reg, unsigned int *val); @@ -1178,7 +1178,7 @@ static int _regmap_select_page(struct regmap *map, unsigned int *reg, ret = _regmap_update_bits(map, range->selector_reg, range->selector_mask, win_page << range->selector_shift, - &page_chg); + &page_chg, false); map->work_buf = orig_work_buf; @@ -2327,7 +2327,7 @@ EXPORT_SYMBOL_GPL(regmap_bulk_read); static int _regmap_update_bits(struct regmap *map, unsigned int reg, unsigned int mask, unsigned int val, - bool *change) + bool *change, bool force_write) { int ret; unsigned int tmp, orig; @@ -2339,7 +2339,7 @@ static int _regmap_update_bits(struct regmap *map, unsigned int reg, tmp = orig & ~mask; tmp |= val & mask; - if (tmp != orig) { + if (force_write || (tmp != orig)) { ret = _regmap_write(map, reg, tmp); if (change) *change = true; @@ -2367,7 +2367,7 @@ int regmap_update_bits(struct regmap *map, unsigned int reg, int ret; map->lock(map->lock_arg); - ret = _regmap_update_bits(map, reg, mask, val, NULL); + ret = _regmap_update_bits(map, reg, mask, val, NULL, false); map->unlock(map->lock_arg); return ret; @@ -2398,7 +2398,7 @@ int regmap_update_bits_async(struct regmap *map, unsigned int reg, map->async = true; - ret = _regmap_update_bits(map, reg, mask, val, NULL); + ret = _regmap_update_bits(map, reg, mask, val, NULL, false); map->async = false; @@ -2427,7 +2427,7 @@ int regmap_update_bits_check(struct regmap *map, unsigned int reg, int ret; map->lock(map->lock_arg); - ret = _regmap_update_bits(map, reg, mask, val, change); + ret = _regmap_update_bits(map, reg, mask, val, change, false); map->unlock(map->lock_arg); return ret; } @@ -2460,7 +2460,7 @@ int regmap_update_bits_check_async(struct regmap *map, unsigned int reg, map->async = true; - ret = _regmap_update_bits(map, reg, mask, val, change); + ret = _regmap_update_bits(map, reg, mask, val, change, false); map->async = false; -- 2.1.4

1 0

[alsa-devel] Applied "regmap: add regmap_write_bits()" to the asoc tree
by Mark Brown 10 Jul '15

10 Jul '15

The patch regmap: add regmap_write_bits() 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 fd4b7286ccc469bf5dde22db6b8fcc455c3c4a66 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 16 Jun 2015 08:52:39 +0000 Subject: [PATCH] regmap: add regmap_write_bits() regmap_write_bits() is similar to regmap_update_bits(), but regmap_write_bits() write data to register even though it is same value. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/base/regmap/regmap.c | 23 +++++++++++++++++++++++ include/linux/regmap.h | 9 +++++++++ 2 files changed, 32 insertions(+) diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index 69ec411ce722..d93bb9a8ab98 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -2375,6 +2375,29 @@ int regmap_update_bits(struct regmap *map, unsigned int reg, EXPORT_SYMBOL_GPL(regmap_update_bits); /** + * regmap_write_bits: Perform a read/modify/write cycle on the register map + * + * @map: Register map to update + * @reg: Register to update + * @mask: Bitmask to change + * @val: New value for bitmask + * + * Returns zero for success, a negative number on error. + */ +int regmap_write_bits(struct regmap *map, unsigned int reg, + unsigned int mask, unsigned int val) +{ + int ret; + + map->lock(map->lock_arg); + ret = _regmap_update_bits(map, reg, mask, val, NULL, true); + map->unlock(map->lock_arg); + + return ret; +} +EXPORT_SYMBOL_GPL(regmap_write_bits); + +/** * regmap_update_bits_async: Perform a read/modify/write cycle on the register * map asynchronously * diff --git a/include/linux/regmap.h b/include/linux/regmap.h index 59c55ea0f0b5..e4b9ad4f05ef 100644 --- a/include/linux/regmap.h +++ b/include/linux/regmap.h @@ -424,6 +424,8 @@ int regmap_bulk_read(struct regmap *map, unsigned int reg, void *val, size_t val_count); int regmap_update_bits(struct regmap *map, unsigned int reg, unsigned int mask, unsigned int val); +int regmap_write_bits(struct regmap *map, unsigned int reg, + unsigned int mask, unsigned int val); int regmap_update_bits_async(struct regmap *map, unsigned int reg, unsigned int mask, unsigned int val); int regmap_update_bits_check(struct regmap *map, unsigned int reg, @@ -645,6 +647,13 @@ static inline int regmap_update_bits(struct regmap *map, unsigned int reg, return -EINVAL; } +static inline int regmap_write_bits(struct regmap *map, unsigned int reg, + unsigned int mask, unsigned int val) +{ + WARN_ONCE(1, "regmap API is disabled"); + return -EINVAL; +} + static inline int regmap_update_bits_async(struct regmap *map, unsigned int reg, unsigned int mask, unsigned int val) -- 2.1.4

1 0

[alsa-devel] Applied "regmap: add regmap_fields_force_write()" to the asoc tree
by Mark Brown 10 Jul '15

10 Jul '15

The patch regmap: add regmap_fields_force_write() 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 e874e6c7edc43436f73cf84157d9221f8b807c36 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 16 Jun 2015 08:52:55 +0000 Subject: [PATCH] regmap: add regmap_fields_force_write() regmap_fields_force_write() is similar to regmap_fields_write(), but regmap_fields_force_write() write data to register even though it is same value. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/base/regmap/regmap.c | 12 ++++++++++++ include/linux/regmap.h | 2 ++ 2 files changed, 14 insertions(+) diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index d93bb9a8ab98..dd63bcbbf8a5 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -1624,6 +1624,18 @@ int regmap_fields_write(struct regmap_field *field, unsigned int id, } EXPORT_SYMBOL_GPL(regmap_fields_write); +int regmap_fields_force_write(struct regmap_field *field, unsigned int id, + unsigned int val) +{ + if (id >= field->id_size) + return -EINVAL; + + return regmap_write_bits(field->regmap, + field->reg + (field->id_offset * id), + field->mask, val << field->shift); +} +EXPORT_SYMBOL_GPL(regmap_fields_force_write); + /** * regmap_fields_update_bits(): Perform a read/modify/write cycle * on the register field diff --git a/include/linux/regmap.h b/include/linux/regmap.h index e4b9ad4f05ef..519c96231a91 100644 --- a/include/linux/regmap.h +++ b/include/linux/regmap.h @@ -505,6 +505,8 @@ int regmap_field_update_bits(struct regmap_field *field, int regmap_fields_write(struct regmap_field *field, unsigned int id, unsigned int val); +int regmap_fields_force_write(struct regmap_field *field, unsigned int id, + unsigned int val); int regmap_fields_read(struct regmap_field *field, unsigned int id, unsigned int *val); int regmap_fields_update_bits(struct regmap_field *field, unsigned int id, -- 2.1.4

1 0

[alsa-devel] Applied "ASoC: rsnd: gen: add rsnd_force_write()" to the asoc tree
by Mark Brown 10 Jul '15

10 Jul '15

The patch ASoC: rsnd: gen: add rsnd_force_write() 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 7b47ab47b3938e2274834dbde7915df98fc74368 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 16 Jun 2015 08:53:11 +0000 Subject: [PATCH] ASoC: rsnd: gen: add rsnd_force_write() rsnd_force_write() is similar to rsnd_write(), but rsnd_force_write() write data to register even though it is same value. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/gen.c | 16 ++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 4 ++++ 2 files changed, 20 insertions(+) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 8c7dc51b1c4f..48f704b1d80e 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -103,6 +103,22 @@ void rsnd_write(struct rsnd_priv *priv, regmap_fields_write(gen->regs[reg], rsnd_mod_id(mod), data); } +void rsnd_force_write(struct rsnd_priv *priv, + struct rsnd_mod *mod, + enum rsnd_reg reg, u32 data) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + + if (!rsnd_is_accessible_reg(priv, gen, reg)) + return; + + dev_dbg(dev, "w %s[%d] - %4d : %08x\n", + rsnd_mod_name(mod), rsnd_mod_id(mod), reg, data); + + regmap_fields_force_write(gen->regs[reg], rsnd_mod_id(mod), data); +} + void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, u32 mask, u32 data) { diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 09fcc54a8ee0..f729646e3d18 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -157,12 +157,16 @@ struct rsnd_dai_stream; rsnd_read(rsnd_mod_to_priv(m), m, RSND_REG_##r) #define rsnd_mod_write(m, r, d) \ rsnd_write(rsnd_mod_to_priv(m), m, RSND_REG_##r, d) +#define rsnd_mod_force_write(m, r, d) \ + rsnd_force_write(rsnd_mod_to_priv(m), m, RSND_REG_##r, d) #define rsnd_mod_bset(m, r, s, d) \ rsnd_bset(rsnd_mod_to_priv(m), m, RSND_REG_##r, s, d) u32 rsnd_read(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg); void rsnd_write(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, u32 data); +void rsnd_force_write(struct rsnd_priv *priv, struct rsnd_mod *mod, + enum rsnd_reg reg, u32 data); void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, u32 mask, u32 data); u32 rsnd_get_adinr(struct rsnd_mod *mod, struct rsnd_dai_stream *io); -- 2.1.4

1 0

[alsa-devel] [PATCH] ALSA: aoa: Drop owner assignment from i2c_driver
by Krzysztof Kozlowski 10 Jul '15

10 Jul '15

i2c_driver does not need to set an owner because i2c_register_driver() will set it. Signed-off-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> --- The coccinelle script which generated the patch was sent here: http://www.spinics.net/lists/kernel/msg2029903.html --- sound/aoa/codecs/onyx.c | 1 - sound/aoa/codecs/tas.c | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 23c371ecfb6b..a04edff8b729 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -1050,7 +1050,6 @@ MODULE_DEVICE_TABLE(i2c,onyx_i2c_id); static struct i2c_driver onyx_driver = { .driver = { .name = "aoa_codec_onyx", - .owner = THIS_MODULE, }, .probe = onyx_i2c_probe, .remove = onyx_i2c_remove, diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index 364c7c4416e8..78ed1ffbf786 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -939,7 +939,6 @@ MODULE_DEVICE_TABLE(i2c,tas_i2c_id); static struct i2c_driver tas_driver = { .driver = { .name = "aoa_codec_tas", - .owner = THIS_MODULE, }, .probe = tas_i2c_probe, .remove = tas_i2c_remove, -- 1.9.1

2 1

Re: [alsa-devel] Intel Baytrail Linux Audio
by Tyler Yeomans 10 Jul '15

10 Jul '15

Vinod, I am using the ssp0 firmware from: https://git.kernel.org/cgit/linux/kernel/git/vkoul/firmware.git/tree/intel?…. Thanks- Tyler On 07/06/2015 11:19 PM, Vinod Koul wrote: > On Mon, Jul 06, 2015 at 10:29:34AM -0400, Tyler Yeomans wrote: >> Hello all, >> >> Apparently I'm in the same boat as Nicolas is. >> I've attached my dmesg output. >> Everything seems to be generating but I am not able to get any sound >> output. >> Between the 37 second mark and the 617 second mark I had routed the >> FE to BE via the commands: >> >> amixer -c0 sset 'codec_out0 mix 0 pcm0_in' on >> amixer -c0 sset 'media0_out mix 0 media1_in' on > ok that seems fine >> Although it appears that did not work: >> [ 719.834922] intel_sst_acpi 80860F28:00: Wait timed-out >> condition:0x0, msg_id:0x1 fw_state 0x3 >> [ 719.834932] intel_sst_acpi 80860F28:00: fw returned err -16 > I don't think so, since you were able to start the card, that part worked :) > > The error is coming from FW here and seems unresponsive. > > Which firmware are you using? > > Second suspicion is for interrupts, if they are not mapped well we might see > these. > > Jarkko I remember that some board have to be configured properly for > interrupts, can you explaian the subtleties involved here > > -- > ~Vinod

5 11

[alsa-devel] [PATCH v3 0/4] ASoC: Intel: Skylake - Add IPC routines
by Vinod Koul 10 Jul '15

10 Jul '15

The SKL aDSP needs to send and receive IPCs to and from aDSP. To aid this we add IPC and aDSP management routines as a library for driver to use. For most of IPC handling we use the common IPC Intel code in place and add the messaging and message parsing bits on top. First 2 patches are against common IPC code and can be applied independently Last two add IPC handling and aDSP management routines v3 changes: - sst_dsp_register_poll uses opportunistic polling and polls for first few instances using mdelay and later sleeps - s/skl_ipc_reply_find_msg/skl_ipc_reply_get_msg - return IRQ_NONE when interrupt in not handled - removed trace_printk and superfluous cast to/from void * Subhransu S. Prusty (4): ASoC: Intel: Add helper to poll register for DSP status ASoC: Intel: Add helper to update register bits with attr RWC ASoC: Intel: Add Skylake IPC library ASoC: Intel: Add DSP init and boot up functionality for SKL sound/soc/intel/Kconfig | 1 + sound/soc/intel/common/sst-dsp-priv.h | 8 + sound/soc/intel/common/sst-dsp.c | 71 ++++ sound/soc/intel/common/sst-dsp.h | 6 + sound/soc/intel/skylake/Makefile | 5 + sound/soc/intel/skylake/skl-sst-dsp.c | 337 +++++++++++++++ sound/soc/intel/skylake/skl-sst-dsp.h | 57 +++ sound/soc/intel/skylake/skl-sst-ipc.c | 760 ++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 125 ++++++ 9 files changed, 1370 insertions(+) create mode 100644 sound/soc/intel/skylake/skl-sst-dsp.c create mode 100644 sound/soc/intel/skylake/skl-sst-ipc.c create mode 100644 sound/soc/intel/skylake/skl-sst-ipc.h -- 1.9.1

2 8

Re: [alsa-devel] [RFT PATCH 1/2] ASoC: zx: i2s: Fix devm_ioremap_resource return value check
by Mark Brown 09 Jul '15

09 Jul '15

On Thu, Jul 09, 2015 at 10:19:28PM +0900, Krzysztof Kozlowski wrote: > Signed-off-by: Krzysztof Kozlowski <k.kozlowski.k(a)gmail.com> > Cc: <stable(a)vger.kernel.org> Why are you adding a Cc to stable? The zx driver was added in the merge window...

1 0

[alsa-devel] [PATCH alsa-plugins] include speexdsp_types.h, not speex_types.h
by Tanu Kaskinen 09 Jul '15

09 Jul '15

Speexdsp was separated from speex in 1.2rc2. speex_types.h is not shipped by speexdsp, so alsa-plugins shouldn't use that file. speexdsp has speexdsp_types.h, which has the same contents as speex_types.h. speexdsp_types.h is a new file introduced in 1.2rc2, so this change bumps the minimum supported speexdsp version. The version check in configure.ac will actually break if speexdsp 1.2 ever gets released, because pkg-config thinks that "1.2" < "1.2rc2", but I think it's useful to fail if the installed speexdsp version is 1.2rc1 (which I believe is very common on current distributions). If a non-rc version of speexdsp will ever get released, I hope version number 1.2 will be skipped for this reason. (A non-rc version seems unlikely, since 1.2rc1 was released years ago, so it's pretty likely that the project is stuck on so called "release candidates" forever...) Signed-off-by: Tanu Kaskinen <tanu.kaskinen(a)linux.intel.com> --- configure.ac | 2 +- pph/arch.h | 2 +- pph/speex_resampler.h | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/configure.ac b/configure.ac index 902a6d7..c554d22 100644 --- a/configure.ac +++ b/configure.ac @@ -108,7 +108,7 @@ AC_SUBST(AVCODEC_CFLAGS) AC_SUBST(AVCODEC_LIBS) AC_SUBST(AVCODEC_HEADER) -PKG_CHECK_MODULES(speexdsp, [speexdsp >= 1.2], [HAVE_SPEEXDSP="yes"], [HAVE_SPEEXDSP=""]) +PKG_CHECK_MODULES(speexdsp, [speexdsp >= 1.2rc2], [HAVE_SPEEXDSP="yes"], [HAVE_SPEEXDSP=""]) AM_CONDITIONAL(HAVE_SPEEXDSP, test "$HAVE_SPEEXDSP" = "yes") AC_ARG_WITH([speex], diff --git a/pph/arch.h b/pph/arch.h index e2d731a..a07d0d9 100644 --- a/pph/arch.h +++ b/pph/arch.h @@ -36,7 +36,7 @@ #define ARCH_H #ifndef OUTSIDE_SPEEX -#include "speex/speex_types.h" +#include "speex/speexdsp_types.h" #endif #define ABS(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute integer value. */ diff --git a/pph/speex_resampler.h b/pph/speex_resampler.h index c44fbcd..aa85abb 100644 --- a/pph/speex_resampler.h +++ b/pph/speex_resampler.h @@ -82,7 +82,7 @@ #else /* OUTSIDE_SPEEX */ -#include "speex/speex_types.h" +#include "speex/speexdsp_types.h" #endif /* OUTSIDE_SPEEX */ -- 1.9.3

2 2

[alsa-devel] [PATCH v2 0/6] ASoC: Intel: Sklyake - add IPC routines
by Vinod Koul 09 Jul '15

09 Jul '15

The SKL aDSP needs to send and receive IPCs to and from aDSP. To aid this we add IPC and aDSP management routines as a library for driver to use. For most of IPC handling we use the common IPC Intel code in place and add the messaging and message parsing bits on top. First 4 patches are against common IPC code and can be applied independently Last two add IPC handling and aDSP management routines, they are not dependent upon anything and can be applied independently but they are generated against SKL PCM driver series so Makefile updates expect those to be there v2 folds few fixes seen and make function name consistent Subhransu S. Prusty (6): ASoC: Intel: Add helper to poll register for DSP status ASoC: Intel: Define SKL ADSP IPC and general purpose registers ASoC: Intel: Reorganize the common dsp structure ASoC: Intel: Add helper to update register bits with attr RWC ASoC: Intel: Add Skylake IPC library ASoC: Intel: Add DSP init and boot up functionality for SKL sound/soc/intel/Kconfig | 1 + sound/soc/intel/common/sst-dsp-priv.h | 20 +- sound/soc/intel/common/sst-dsp.c | 51 +++ sound/soc/intel/common/sst-dsp.h | 6 + sound/soc/intel/skylake/Makefile | 5 + sound/soc/intel/skylake/skl-sst-dsp.c | 337 +++++++++++++++ sound/soc/intel/skylake/skl-sst-dsp.h | 119 ++++++ sound/soc/intel/skylake/skl-sst-ipc.c | 752 ++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 125 ++++++ 9 files changed, 1412 insertions(+), 4 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-sst-dsp.c create mode 100644 sound/soc/intel/skylake/skl-sst-dsp.h create mode 100644 sound/soc/intel/skylake/skl-sst-ipc.c create mode 100644 sound/soc/intel/skylake/skl-sst-ipc.h -- 1.9.1

2 17

[alsa-devel] debugging kernel module: ASoC sun7i A20 I2S DAI driver for simple sound card setup (with UDA1380 codec)
by Andrea Venturi 09 Jul '15

09 Jul '15

hello, i'm writing here because i'm facing an issue on a new driver and like a feedback from the community. wht place better then this ML! i'm using an Allwinner A20 based card with I2S/I2C wires exposed (Olimex A20 SOM + EVB), and an external UDA1380 codec (dirty cheap purchase from those china techno junk sites) i'd like to make this bundle work together; my aim is to produce the mainline kernel DAI module for the actually missing I2S HW feature for Allwinner SOC (at least sun4i A10 and sun7i A20, don't know if the ) i gathered as more info as possible about Alsa ASoC implementation; but, boys, is not very easy to catch on as there's a huge bulk of documentation scattered all over the web with varying degree of obsolescence in place.. and usually the newer stuff is the less documented :-) anyway, as my "awareness" of the whole Alsa scenario went increasing (at least in my opinion..), i found the Freescale SAI (see .../sound/soc/fsl/fsl_sai.c) module to be the nearest/better/more modern "example" from which starting modeling the sunxi I2S module. so, now, i've shaped a Sunxi I2S kernel module that loads someway on my HW testbed. all the tech details (with source and tests) are published here: https://gist.github.com/aventuri/c40d08b946b0d1da5081 you'll find: - sunxi-i2s.c - a DTS with the sound entry for a simple sound card - a slightly modded uda1380.c codec driver; as a sidenote, i had to modify for params loading from DTS. if more details are needed, i'll add stuff on request, of course.. the starting mainline kernel is linux 4.1.rc sunxi-wip from here: https://github.com/jwrdegoede/linux-sunxi/tree/sunxi-wip where the i2s clocks and dmaengine (and the analog codec of AW sunxi, but it's not needed here) are already in place. the "gory details" about HW registers are coming from the my reading of A20 PDF and allwinner I2S linux-sunxi legacy I2S driver for 3.4: https://github.com/linux-sunxi/linux-sunxi/tree/sunxi-3.4/sound/soc/sunxi/i… these registers CAN be way off as i'm still debugging a bit before starting checking them carefully.. now, the kernel modules (snd-soc-simple-card snd-soc-sunxi-i2s and snd-soc-uda1380) are loaded ok and doing someway a proper initialization of HW and kernel data structures (check on the gist before). i can see the alsa device nodes # ls -lR /dev/snd /dev/snd: total 0 crw-rw---- 1 root root 116, 0 Jan 1 03:00 controlC0 crw-rw---- 1 root root 116, 24 Jan 1 03:00 pcmC0D0c crw-rw---- 1 root root 116, 16 Jan 1 03:00 pcmC0D0p crw-rw---- 1 root root 116, 33 Jan 1 03:00 timer and alsamixer is more or less showing all the controllers (many are from the UDA codec, of course, already working..) the problem i'm facing now is that the when i launch aplay with a wav file, when it accesses the dev node "/dev/snd/pcmC0D0p" is spitting out: "open error: Invalid argument" these are the last few rows of a "strace aplay wav/equinox-48KHz.wav" execution: # aplay wav/equinox-48KHz.wav [ 9018.638837] [I2S]Entered sunxi_i2s_startup: hw is subdevice #0, stream is: playback [ 9018.646858] [I2S]Entered sunxi_i2s_shutdown, need to clear some HW regs? aplay: main:722: audio open error: Invalid argument # ===================== ... open("/dev/snd/controlC0", O_RDONLY|O_LARGEFILE|O_CLOEXEC) = 3 fcntl64(3, F_SETFD, FD_CLOEXEC) = 0 ioctl(3, SNDRV_CTL_IOCTL_CARD_INFO or UI_DEV_CREATE, 0xbeac43b8) = 0 close(3) = 0 open("/dev/snd/controlC0", O_RDWR|O_LARGEFILE|O_CLOEXEC) = 3 fcntl64(3, F_SETFD, FD_CLOEXEC) = 0 ioctl(3, SNDRV_CTL_IOCTL_PVERSION or USBDEVFS_CONTROL, 0xbeac4540) = 0 ioctl(3, SNDRV_CTL_IOCTL_PCM_PREFER_SUBDEVICE, 0xbeac457c) = 0 open("/dev/snd/pcmC0D0p", O_RDWR|O_NONBLOCK|O_LARGEFILE|O_CLOEXEC) = -1 EINVAL (Invalid argument) close(3) = 0 write(2, "aplay: main:722: ", 17aplay: main:722: ) = 17 write(2, "audio open error: Invalid argume"..., 34audio open error: Invalid argument) = 34 write(2, "\n", 1 ) = 1 exit_group(1) = ? +++ exited with 1 +++ # ================== now my question is: how can i start debugging this step? which kernel tools i could use to pinpoint quickly who is answering back this "invalid argument" in the called function chain. i suppose some "sound params" in the my own kernel driver is not filled correctly; i tried with many wav files (44.1 48 etc..) but always get this reply. i tried with the "kernel function tracing" tool (/sys/kernel/debug/tracing/...) but it doesn't help a lot.. any suggestion? any indication on documents about effective Alsa ASoC debugging or architecture? bests andrea

1 0

[alsa-devel] [PATCH] ALSA: hda: Delete an unnecessary check before the function call "kobject_put"
by SF Markus Elfring 09 Jul '15

09 Jul '15

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 8 Jul 2015 21:26:02 +0200 The kobject_put() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/hda/hdac_sysfs.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index 0a6ce3b..089b35f 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -321,8 +321,7 @@ static void widget_tree_free(struct hdac_device *codec) free_widget_node(*p, &widget_node_group); kfree(tree->nodes); } - if (tree->root) - kobject_put(tree->root); + kobject_put(tree->root); kfree(tree); codec->widgets = NULL; } -- 2.4.5

2 1

[alsa-devel] [PATCH] ASoC: mediatek: Use platform_of_node for machine drivers
by Koro Chen 09 Jul '15

09 Jul '15

This replaces the platform_name in snd_soc_dai_link by device tree node. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- .../devicetree/bindings/sound/mt8173-max98090.txt | 2 ++ .../bindings/sound/mt8173-rt5650-rt5676.txt | 2 ++ sound/soc/mediatek/mt8173-max98090.c | 17 +++++++++++++---- sound/soc/mediatek/mt8173-rt5650-rt5676.c | 19 +++++++++++++++---- 4 files changed, 32 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8173-max98090.txt b/Documentation/devicetree/bindings/sound/mt8173-max98090.txt index 829bd26..519e97c 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-max98090.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-max98090.txt @@ -3,11 +3,13 @@ MT8173 with MAX98090 CODEC Required properties: - compatible : "mediatek,mt8173-max98090" - mediatek,audio-codec: the phandle of the MAX98090 audio codec +- mediatek,platform: the phandle of MT8173 ASoC platform Example: sound { compatible = "mediatek,mt8173-max98090"; mediatek,audio-codec = <&max98090>; + mediatek,platform = <&afe>; }; diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt index 61e98c9..f205ce9 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt @@ -3,11 +3,13 @@ MT8173 with RT5650 RT5676 CODECS Required properties: - compatible : "mediatek,mt8173-rt5650-rt5676" - mediatek,audio-codec: the phandles of rt5650 and rt5676 codecs +- mediatek,platform: the phandle of MT8173 ASoC platform Example: sound { compatible = "mediatek,mt8173-rt5650-rt5676"; mediatek,audio-codec = <&rt5650 &rt5676>; + mediatek,platform = <&afe>; }; diff --git a/sound/soc/mediatek/mt8173-max98090.c b/sound/soc/mediatek/mt8173-max98090.c index 4d44b58..2d2536a 100644 --- a/sound/soc/mediatek/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173-max98090.c @@ -103,7 +103,6 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = { .name = "MAX98090 Playback", .stream_name = "MAX98090 Playback", .cpu_dai_name = "DL1", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -114,7 +113,6 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = { .name = "MAX98090 Capture", .stream_name = "MAX98090 Capture", .cpu_dai_name = "VUL", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -125,7 +123,6 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = { { .name = "Codec", .cpu_dai_name = "I2S", - .platform_name = "11220000.mt8173-afe-pcm", .no_pcm = 1, .codec_dai_name = "HiFi", .init = mt8173_max98090_init, @@ -152,9 +149,21 @@ static struct snd_soc_card mt8173_max98090_card = { static int mt8173_max98090_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_max98090_card; - struct device_node *codec_node; + struct device_node *codec_node, *platform_node; int ret, i; + platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!platform_node) { + dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + return -EINVAL; + } + for (i = 0; i < card->num_links; i++) { + if (mt8173_max98090_dais[i].platform_name) + continue; + mt8173_max98090_dais[i].platform_of_node = platform_node; + } + codec_node = of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0); if (!codec_node) { diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c index 0940553..6f52eca 100644 --- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c @@ -138,7 +138,6 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { .name = "rt5650_rt5676 Playback", .stream_name = "rt5650_rt5676 Playback", .cpu_dai_name = "DL1", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -149,7 +148,6 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { .name = "rt5650_rt5676 Capture", .stream_name = "rt5650_rt5676 Capture", .cpu_dai_name = "VUL", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -161,7 +159,6 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { { .name = "Codec", .cpu_dai_name = "I2S", - .platform_name = "11220000.mt8173-afe-pcm", .no_pcm = 1, .codecs = mt8173_rt5650_rt5676_codecs, .num_codecs = 2, @@ -209,7 +206,21 @@ static struct snd_soc_card mt8173_rt5650_rt5676_card = { static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_rt5676_card; - int ret; + struct device_node *platform_node; + int i, ret; + + platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!platform_node) { + dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < card->num_links; i++) { + if (mt8173_rt5650_rt5676_dais[i].platform_name) + continue; + mt8173_rt5650_rt5676_dais[i].platform_of_node = platform_node; + } mt8173_rt5650_rt5676_codecs[0].of_node = of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0); -- 1.8.1.1.dirty

2 1

[alsa-devel] [PATCH] ASoC: arizona: Use the more idiomatic params_width
by Charles Keepax 09 Jul '15

09 Jul '15

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a88202d..c246fbd 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1504,7 +1504,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, else rates = &arizona_48k_bclk_rates[0]; - wl = snd_pcm_format_width(params_format(params)); + wl = params_width(params); if (tdm_slots) { arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n", -- 1.7.2.5

2 1

[alsa-devel] [PATCH v8 0/4] ASoC: Intel: Add skylake PCM driver
by Vinod Koul 09 Jul '15

09 Jul '15

This patch series adds the SKL PCM driver in first two patches and then adds decoupling of the controller for splitting the links in last patch this version fixes - superflous cast from/to void *, both gcc and sparse are happy! - remove buffer memset - check return of pm_runtime_get_sync - set dma_data to NULL on free - remove ifdef PM for pm_runtime_active - fix pci_set_drvdata calls Jeeja KP (4): ASoC: Intel: Add Skylake HDA platform driver ASoC: Intel: Add Skylake HDA audio driver ASoC: Intel: Add makefile support for SKL driver ASoC: Intel: Add support for decoupled mode in skl driver sound/soc/intel/Kconfig | 4 + sound/soc/intel/Makefile | 1 + sound/soc/intel/skylake/Makefile | 3 + sound/soc/intel/skylake/skl-pcm.c | 916 ++++++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl.c | 536 ++++++++++++++++++++++ sound/soc/intel/skylake/skl.h | 71 +++ 6 files changed, 1531 insertions(+) create mode 100644 sound/soc/intel/skylake/Makefile create mode 100644 sound/soc/intel/skylake/skl-pcm.c create mode 100644 sound/soc/intel/skylake/skl.c create mode 100644 sound/soc/intel/skylake/skl.h -- 1.9.1

2 8

[alsa-devel] [PATCH 0/6] ASoC: Intel: fix kconfig, symbols and sparse
by Vinod Koul 09 Jul '15

09 Jul '15

So this series has two parts part 1, first three patches which fix the issue of BYT driver not getting compiled in -next and then cleans up symbol usage in various driver Second part fixes (next three) spare warnings on drivers Vinod Koul (6): ASoC: Intel: use CONFIG_SND_SOC for intel boards ASoC: Intel: kconfig - Move DW_DMAC_CORE dependency to machines ASoC: intel - kconfig: remove SND_SOC_INTEL_SST prompt ASOC: intel - hsw: fix sparse unused varible warn ASoC: intel - atom: statify sst_slot_enum_info ASoC: Intel -atom: fix sst_platform_compr_ops declaration sound/soc/intel/Kconfig | 24 +++++++++++++----------- sound/soc/intel/Makefile | 2 +- sound/soc/intel/atom/sst-atom-controls.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 1 - sound/soc/intel/atom/sst-mfld-platform.h | 1 + sound/soc/intel/haswell/sst-haswell-pcm.c | 1 - 6 files changed, 16 insertions(+), 15 deletions(-) -- 1.9.1

3 13

[alsa-devel] [PATCH 1/3] Revert "ASoC: wm0010: Add missing IRQF_ONESHOT"
by Axel Lin 09 Jul '15

09 Jul '15

This reverts 58d468328646 ("ASoC: wm0010: Add missing IRQF_ONESHOT"). The coccinelle warnings is false positive because the original code does set IRQF_ONESHOT by "trigger |= IRQF_ONESHOT;". Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm0010.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 6560a66..ff924032 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -953,7 +953,7 @@ static int wm0010_spi_probe(struct spi_device *spi) trigger = IRQF_TRIGGER_FALLING; trigger |= IRQF_ONESHOT; - ret = request_threaded_irq(irq, NULL, wm0010_irq, trigger | IRQF_ONESHOT, + ret = request_threaded_irq(irq, NULL, wm0010_irq, trigger, "wm0010", wm0010); if (ret) { dev_err(wm0010->dev, "Failed to request IRQ %d: %d\n", -- 2.1.0

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[alsa-devel] [PATCH] ASoC: dapm: Fix kcontrol widget name memory management
by Lars-Peter Clausen 09 Jul '15

09 Jul '15

The name field of the widget template is only used inside snd_soc_dapm_new_control_unlocked() which allocates a copy for the actual widget. This means we need to free the name allocated for the template in dapm_kcontrol_data_alloc() and not the name of the actual widget in dapm_kcontrol_free(). Otherwise we get a double free on the widget name and a memory leak on the template name. Fixes: 773da9b358bf ("ASoC: dapm: Append "Autodisable" to autodisable widget names") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6bac491..1628f0c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -358,9 +358,10 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, data->widget = snd_soc_dapm_new_control_unlocked(widget->dapm, &template); + kfree(name); if (!data->widget) { ret = -ENOMEM; - goto err_name; + goto err_data; } } break; @@ -391,9 +392,10 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, data->widget = snd_soc_dapm_new_control_unlocked( widget->dapm, &template); + kfree(name); if (!data->widget) { ret = -ENOMEM; - goto err_name; + goto err_data; } snd_soc_dapm_add_path(widget->dapm, data->widget, @@ -408,8 +410,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, return 0; -err_name: - kfree(name); err_data: kfree(data); return ret; @@ -418,8 +418,6 @@ err_data: static void dapm_kcontrol_free(struct snd_kcontrol *kctl) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kctl); - if (data->widget) - kfree(data->widget->name); kfree(data->wlist); kfree(data); } -- 2.1.4

3 2

Re: [alsa-devel] Fw: Using loopback card to Connect GSM two way call to the real sound card UDA1345TS
by Srinivasan S 09 Jul '15

09 Jul '15

Dear Clemens, Many Thanks a lot for all your support w.r.t this thread so far Am facing overrun & underrun issues, when I run the the above GSM application with the attached asound.conf I have some understanding by googling & found why this overrun & underrun issues occurs ie., unable to transfer audio frames from hardware buffers to application buffers(for capture) & the viceversa(for playback), As I googled, I didn't get detailed information with bits & pieces here & there, could you please help me out in resolving this overrun & underrun issue ie., how to set this buffer size or only buffer_size is causing this issue or any other parameters like period_size, period_time values should also be considered inorder to fix this overrun & underrun issue Now I have randomly set the buffer size ie., 4097 for 8Khz sampling rate But I have no idea logically how this buffer size 4097 is working for 8Khz & when I copied the same buffer size to 48Khz, the audio gets cut & again overrun & underrun issue is seen Could you please help me out how to fix this overrun & underrun issue logically, rather than I have that an understanding of programming some random values like 4097 Is it possible to fix this issue in asound.conf itself, rather than using multithreading in GSM application So that fixing these types of issues henceforth will be faster & addressed logically & will be helpful for others as well Kindly do the needful as early as possible Many Thanks in advance once again ________________________________________ From: Srinivasan S Sent: Wednesday, July 1, 2015 12:54 PM To: Clemens Ladisch Cc: alsa-devel(a)alsa-project.org; linux-audio-dev(a)lists.linuxaudio.org Subject: Re: [alsa-devel] Fw: Using loopback card to Connect GSM two way call to the real sound card UDA1345TS Dear Clemens, Many Thanks a lot for all your support w.r.t this thread so far Am facing overrun & underrun issues on TI AM335x, when I run the the above GSM application with the attached asound.conf I have some understanding by googling & found why this overrun & underrun issues occurs ie., unable to transfer audio frames from hardware buffers to application buffers(for capture) & the viceversa(for playback), As I googled, I didn't get detailed information with bits & pieces here & there, could you please help me out in resolving this overrun & underrun issue ie., how to set this buffer size or only buffer_size is causing this issue or any other parameters like period_size, period_time values should also be considered inorder to fix this overrun & underrun issue Now I have randomly set the buffer size ie., 4097 for 8Khz sampling rate But I have no idea logically how this buffer size 4097 is working for 8Khz & when I copied the same buffer size to 48Khz, the audio gets cut & again overrun & underrun issue is seen Could you please help me out how to fix this overrun & underrun issue logically, rather than I have that an understanding of programming some random values like 4097 Is it possible to fix this issue in asound.conf itself, rather than using multithreading in GSM application So that fixing these types of issues henceforth will be faster & addressed logically & will be helpful for others as well Kindly do the needful as early as possible Many Thanks in advance once again ________________________________________ From: alsa-devel-bounces(a)alsa-project.org <alsa-devel-bounces(a)alsa-project.org> on behalf of Srinivasan S <srinivasan.s(a)tataelxsi.co.in> Sent: Wednesday, June 3, 2015 11:35 AM To: Clemens Ladisch Cc: alsa-devel(a)alsa-project.org; linux-audio-dev(a)lists.linuxaudio.org Subject: Re: [alsa-devel] Fw: Using loopback card to Connect GSM two way call to the real sound card UDA1345TS Dear Clemens As I didn't find Jack mailing lists by googling, could you please loop Jack mailing lists to this mail chain Kindly do the needful as early as possible Many Thanks in advance ________________________________________ From: Clemens Ladisch <clemens(a)ladisch.de> Sent: Monday, June 1, 2015 8:09 PM To: Srinivasan S Cc: alsa-devel(a)alsa-project.org; linux-audio-dev(a)lists.linuxaudio.org Subject: Re: [alsa-devel] Fw: Using loopback card to Connect GSM two way call to the real sound card UDA1345TS Srinivasan S wrote: > 2. Could you please let me know, I have downloaded jack-1.9.10.tar.bz2, how this needs to be installed in my rootfs IIRC Jack uses some non-standard build system. Try asking on the Jack mailing list how to cross-compile it. Please note that the ALSA Jack plugin is part of the alsa-plugins package. And as I already mentioned, it is unlikely that Jack will use less CPU than dshare. Regards, Clemens _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

2 3

[alsa-devel] Tascam US-16x08 0644:8047
by Greg 09 Jul '15

09 Jul '15

Greetings, I am fairly new to USB dev (in linux in particular, but also in general), but I would very much like to try to get support for the above device working in snd-usb-audio. - Is this an appropriate place to discuss snd-usb-audio? - Are there any recommended reading pointers for the quirk table? - or any highly recommended USB reading material that I might not have found? I patched parse_audio_format_rates_v2(), get_sample_rate_v2(), and set_sample_rate_v2(), and through some sort of beginner luck was able to get aplay audio out of the first two channels. That was incomplete hackery though (eg fixed sample rate), and I would like to learn how to properly add quirk support. There have been other reports that this device worked OOTB, but I fail to see how! I've also been examining the traffic to the device with wireshark and a win7 vm, but the learning curve for USB is a bit steep, so I am digesting. (: If anyone can provide suggestions on lsusb output alone, here's what I have: http://pastebin.com/pA9MLQet cheers, Greg

1 0

[alsa-devel] [PATCH v7 0/4] ASoC: intel - add skylake PCM driver
by Vinod Koul 09 Jul '15

09 Jul '15

This patch series adds the SKL PCM driver in first two patches and then adds decoupling of the controller for splitting the links in last patch This version is rebased on 4.2-rc1 and has fixed the style issues identfied in last review Jeeja KP (3): ASoC: Intel: add Skylake HDA platform driver ASoC: Intel - add Skylake HDA audio driver ASoC: Intel - add makefile support for SKL driver Vinod Koul (1): ASoC: intel - adds support for decoupled mode in skl driver sound/soc/intel/Kconfig | 7 + sound/soc/intel/Makefile | 1 + sound/soc/intel/skylake/Makefile | 3 + sound/soc/intel/skylake/skl-pcm.c | 936 ++++++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl.c | 543 ++++++++++++++++++++++ sound/soc/intel/skylake/skl.h | 71 +++ 6 files changed, 1561 insertions(+) create mode 100644 sound/soc/intel/skylake/Makefile create mode 100644 sound/soc/intel/skylake/skl-pcm.c create mode 100644 sound/soc/intel/skylake/skl.c create mode 100644 sound/soc/intel/skylake/skl.h -- 1.9.1

2 8

[alsa-devel] [PATCH 0/2] ASoC: Intel: skylake - match braces for if-else
by Vinod Koul 09 Jul '15

09 Jul '15

Coding style mandates that if-else should match braces even if one of them is a single line statement. Fix it in two places missed earlier Mark, you can also fold these two patches with SKL driver series (1st and 2nd patch can fold in 1st and 2nd patch respectively) or apply independetly, either way okay Vinod Koul (2): ASoC: Intel: skl-pcm.c match braces for if-else ASoC: Intel: skl.c match braces for if-else sound/soc/intel/skylake/skl-pcm.c | 3 ++- sound/soc/intel/skylake/skl.c | 4 ++-- 2 files changed, 4 insertions(+), 3 deletions(-) -- 1.9.1

2 4

Re: [alsa-devel] [ANNOUNCE] Media Controller workshop in Helsinki
by Lars-Peter Clausen 08 Jul '15

08 Jul '15

Cc: ALSA ML On 07/08/2015 01:16 PM, Mauro Carvalho Chehab wrote: > Hi, > > As discussed on our IRC #v4l channel at Freenode, we'll be running a 3-day > Media Controller workshop in Helsinki between July 29-31. > > The main goal of this workshop is to fixup the problems with the Media > Controller API to properly represent device nodes and the control tree, as > this is a requirement for the Media Controller to be used by DVB, ALSA and > radio devices. > > This is an invitation-only event, as we want to be sure that we'll have > there people that are actively working with the Media Controller, DVB > and/or ALSA. So, if you think this is for you, please send us an e-mail. > > Also, even if you're not able to participate, feel free to submit > relevant topics related to the Media Controller, in order to help us to > build the workshop's agenda and take other needs into account when working > on the needed API changes. > > As usual, we'll be using the media-workshop(a)linuxtv.org ML for the > specific discussions about that, so the ones interested on participate on > such discussions and/or be present there are requested to subscribe it, > and to submit themes of interest via the mailing lists. > > For those interested on this theme, I'll add a list of the most relevant > past discussions about this theme. > > Hope to see you there! > > Regards, Mauro > > --- > > The DVB needs with regards to MC are mapped on those two blog entries: > http://blogs.s-osg.org/media-controller-support-for-dvb-part-1/ > http://blogs.s-osg.org/the-role-of-dtv-network-interfaces-in-media-controll… > > We had two discussions about MC on our last mini-summit, back on March, > 26. The discussions and slides can be found at: > http://linuxtv.org/news.php?entry=2015-05-05.mchehab > http://www.linuxtv.org/downloads/presentations/media_summit_2015_US/dtv_med… > > A summary of the issues can also be seen on the emails below: > https://www.mail-archive.com/linux-media@vger.kernel.org/msg85910.html > https://www.mail-archive.com/linux-media@vger.kernel.org/msg85979.html > https://www.mail-archive.com/linux-media@vger.kernel.org/msg83883.html > https://www.mail-archive.com/linux-media@vger.kernel.org/msg83884.html > > Patches that minimally expose DVB via MC were merged on Kernel 4.1, > although it is currently marked as BROKEN, in order to tag those API > extensions as likely to change on future versions. > > The initial patchset proposal for ALSA support on MC is there at: > http://www.spinics.net/lists/linux-media/msg89574.html > > The most recent proposal for such fixup can be seen at: > http://www.spinics.net/lists/linux-media/msg91365.html

2 1

[alsa-devel] [PATCH] ASoC: Free card DAPM context on snd_soc_instantiate_card() error path
by Lars-Peter Clausen 08 Jul '15

08 Jul '15

Make sure the to free the card DAPM context if snd_soc_instantiate_card() fails, otherwise the memory allocated for the DAPM widgets is leaked. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2fc461c..8c8ce65 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1728,6 +1728,7 @@ card_probe_error: if (card->remove) card->remove(card); + snd_soc_dapm_free(&card->dapm); soc_cleanup_card_debugfs(card); snd_card_free(card->snd_card); -- 2.1.4

2 1

[alsa-devel] [PATCH 1/3] mfd: wm5110: Add registers for custom write sequence triggers
by Charles Keepax 08 Jul '15

08 Jul '15

This register will be needed as part of some additional support for the headphone path on wm5110, so this patch adds the register and sets up its regmap config. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- I have based the series on Mark's ASoC since that is where most of the work is. The MFD patches are pretty trivial but I am happy to send a series based on the MFD tree if preferred. Thanks, Charles drivers/mfd/wm5110-tables.c | 2 + include/linux/mfd/arizona/registers.h | 37 +++++++++++++++++++++++++++++++++ 2 files changed, 39 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index 12cad94..62a4aa1 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -676,6 +676,7 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000032, 0x0100 }, /* R50 - PWM Drive 3 */ { 0x00000040, 0x0000 }, /* R64 - Wake control */ { 0x00000041, 0x0000 }, /* R65 - Sequence control */ + { 0x00000042, 0x0000 }, /* R66 - Spare Triggers */ { 0x00000061, 0x01FF }, /* R97 - Sample Rate Sequence Select 1 */ { 0x00000062, 0x01FF }, /* R98 - Sample Rate Sequence Select 2 */ { 0x00000063, 0x01FF }, /* R99 - Sample Rate Sequence Select 3 */ @@ -1716,6 +1717,7 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_PWM_DRIVE_3: case ARIZONA_WAKE_CONTROL: case ARIZONA_SEQUENCE_CONTROL: + case ARIZONA_SPARE_TRIGGERS: case ARIZONA_SAMPLE_RATE_SEQUENCE_SELECT_1: case ARIZONA_SAMPLE_RATE_SEQUENCE_SELECT_2: case ARIZONA_SAMPLE_RATE_SEQUENCE_SELECT_3: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index aacc10d..f43aa56 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -39,6 +39,7 @@ #define ARIZONA_PWM_DRIVE_3 0x32 #define ARIZONA_WAKE_CONTROL 0x40 #define ARIZONA_SEQUENCE_CONTROL 0x41 +#define ARIZONA_SPARE_TRIGGERS 0x42 #define ARIZONA_SAMPLE_RATE_SEQUENCE_SELECT_1 0x61 #define ARIZONA_SAMPLE_RATE_SEQUENCE_SELECT_2 0x62 #define ARIZONA_SAMPLE_RATE_SEQUENCE_SELECT_3 0x63 @@ -1431,6 +1432,42 @@ #define ARIZONA_WSEQ_ENA_JD2_RISE_WIDTH 1 /* WSEQ_ENA_JD2_RISE */ /* + * R66 (0x42) - Spare Triggers + */ +#define ARIZONA_WS_TRG8 0x0080 /* WS_TRG8 */ +#define ARIZONA_WS_TRG8_MASK 0x0080 /* WS_TRG8 */ +#define ARIZONA_WS_TRG8_SHIFT 7 /* WS_TRG8 */ +#define ARIZONA_WS_TRG8_WIDTH 1 /* WS_TRG8 */ +#define ARIZONA_WS_TRG7 0x0040 /* WS_TRG7 */ +#define ARIZONA_WS_TRG7_MASK 0x0040 /* WS_TRG7 */ +#define ARIZONA_WS_TRG7_SHIFT 6 /* WS_TRG7 */ +#define ARIZONA_WS_TRG7_WIDTH 1 /* WS_TRG7 */ +#define ARIZONA_WS_TRG6 0x0020 /* WS_TRG6 */ +#define ARIZONA_WS_TRG6_MASK 0x0020 /* WS_TRG6 */ +#define ARIZONA_WS_TRG6_SHIFT 5 /* WS_TRG6 */ +#define ARIZONA_WS_TRG6_WIDTH 1 /* WS_TRG6 */ +#define ARIZONA_WS_TRG5 0x0010 /* WS_TRG5 */ +#define ARIZONA_WS_TRG5_MASK 0x0010 /* WS_TRG5 */ +#define ARIZONA_WS_TRG5_SHIFT 4 /* WS_TRG5 */ +#define ARIZONA_WS_TRG5_WIDTH 1 /* WS_TRG5 */ +#define ARIZONA_WS_TRG4 0x0008 /* WS_TRG4 */ +#define ARIZONA_WS_TRG4_MASK 0x0008 /* WS_TRG4 */ +#define ARIZONA_WS_TRG4_SHIFT 3 /* WS_TRG4 */ +#define ARIZONA_WS_TRG4_WIDTH 1 /* WS_TRG4 */ +#define ARIZONA_WS_TRG3 0x0004 /* WS_TRG3 */ +#define ARIZONA_WS_TRG3_MASK 0x0004 /* WS_TRG3 */ +#define ARIZONA_WS_TRG3_SHIFT 2 /* WS_TRG3 */ +#define ARIZONA_WS_TRG3_WIDTH 1 /* WS_TRG3 */ +#define ARIZONA_WS_TRG2 0x0002 /* WS_TRG2 */ +#define ARIZONA_WS_TRG2_MASK 0x0002 /* WS_TRG2 */ +#define ARIZONA_WS_TRG2_SHIFT 1 /* WS_TRG2 */ +#define ARIZONA_WS_TRG2_WIDTH 1 /* WS_TRG2 */ +#define ARIZONA_WS_TRG1 0x0001 /* WS_TRG1 */ +#define ARIZONA_WS_TRG1_MASK 0x0001 /* WS_TRG1 */ +#define ARIZONA_WS_TRG1_SHIFT 0 /* WS_TRG1 */ +#define ARIZONA_WS_TRG1_WIDTH 1 /* WS_TRG1 */ + +/* * R97 (0x61) - Sample Rate Sequence Select 1 */ #define ARIZONA_WSEQ_SAMPLE_RATE_DETECT_A_SEQ_ADDR_MASK 0x01FF /* WSEQ_SAMPLE_RATE_DETECT_A_SEQ_ADDR - [8:0] */ -- 1.7.2.5

3 6

[alsa-devel] [PATCH v3 1/3] mfd: wm5110: Add registers for custom write sequence triggers
by Charles Keepax 08 Jul '15

08 Jul '15

This register will be needed as part of some additional support for the headphone path on wm5110, so this patch adds the register and sets up its regmap config. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Acked-by: Lee Jones <lee.jones(a)linaro.org> --- drivers/mfd/wm5110-tables.c | 2 + include/linux/mfd/arizona/registers.h | 37 +++++++++++++++++++++++++++++++++ 2 files changed, 39 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index 12cad94..62a4aa1 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -676,6 +676,7 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000032, 0x0100 }, /* R50 - PWM Drive 3 */ { 0x00000040, 0x0000 }, /* R64 - Wake control */ { 0x00000041, 0x0000 }, /* R65 - Sequence control */ + { 0x00000042, 0x0000 }, /* R66 - Spare Triggers */ { 0x00000061, 0x01FF }, /* R97 - Sample Rate Sequence Select 1 */ { 0x00000062, 0x01FF }, /* R98 - Sample Rate Sequence Select 2 */ { 0x00000063, 0x01FF }, /* R99 - Sample Rate Sequence Select 3 */ @@ -1716,6 +1717,7 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_PWM_DRIVE_3: case ARIZONA_WAKE_CONTROL: case ARIZONA_SEQUENCE_CONTROL: + case ARIZONA_SPARE_TRIGGERS: case ARIZONA_SAMPLE_RATE_SEQUENCE_SELECT_1: case ARIZONA_SAMPLE_RATE_SEQUENCE_SELECT_2: case ARIZONA_SAMPLE_RATE_SEQUENCE_SELECT_3: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index 3499d36..11affb3 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -39,6 +39,7 @@ #define ARIZONA_PWM_DRIVE_3 0x32 #define ARIZONA_WAKE_CONTROL 0x40 #define ARIZONA_SEQUENCE_CONTROL 0x41 +#define ARIZONA_SPARE_TRIGGERS 0x42 #define ARIZONA_SAMPLE_RATE_SEQUENCE_SELECT_1 0x61 #define ARIZONA_SAMPLE_RATE_SEQUENCE_SELECT_2 0x62 #define ARIZONA_SAMPLE_RATE_SEQUENCE_SELECT_3 0x63 @@ -1431,6 +1432,42 @@ #define ARIZONA_WSEQ_ENA_JD2_RISE_WIDTH 1 /* WSEQ_ENA_JD2_RISE */ /* + * R66 (0x42) - Spare Triggers + */ +#define ARIZONA_WS_TRG8 0x0080 /* WS_TRG8 */ +#define ARIZONA_WS_TRG8_MASK 0x0080 /* WS_TRG8 */ +#define ARIZONA_WS_TRG8_SHIFT 7 /* WS_TRG8 */ +#define ARIZONA_WS_TRG8_WIDTH 1 /* WS_TRG8 */ +#define ARIZONA_WS_TRG7 0x0040 /* WS_TRG7 */ +#define ARIZONA_WS_TRG7_MASK 0x0040 /* WS_TRG7 */ +#define ARIZONA_WS_TRG7_SHIFT 6 /* WS_TRG7 */ +#define ARIZONA_WS_TRG7_WIDTH 1 /* WS_TRG7 */ +#define ARIZONA_WS_TRG6 0x0020 /* WS_TRG6 */ +#define ARIZONA_WS_TRG6_MASK 0x0020 /* WS_TRG6 */ +#define ARIZONA_WS_TRG6_SHIFT 5 /* WS_TRG6 */ +#define ARIZONA_WS_TRG6_WIDTH 1 /* WS_TRG6 */ +#define ARIZONA_WS_TRG5 0x0010 /* WS_TRG5 */ +#define ARIZONA_WS_TRG5_MASK 0x0010 /* WS_TRG5 */ +#define ARIZONA_WS_TRG5_SHIFT 4 /* WS_TRG5 */ +#define ARIZONA_WS_TRG5_WIDTH 1 /* WS_TRG5 */ +#define ARIZONA_WS_TRG4 0x0008 /* WS_TRG4 */ +#define ARIZONA_WS_TRG4_MASK 0x0008 /* WS_TRG4 */ +#define ARIZONA_WS_TRG4_SHIFT 3 /* WS_TRG4 */ +#define ARIZONA_WS_TRG4_WIDTH 1 /* WS_TRG4 */ +#define ARIZONA_WS_TRG3 0x0004 /* WS_TRG3 */ +#define ARIZONA_WS_TRG3_MASK 0x0004 /* WS_TRG3 */ +#define ARIZONA_WS_TRG3_SHIFT 2 /* WS_TRG3 */ +#define ARIZONA_WS_TRG3_WIDTH 1 /* WS_TRG3 */ +#define ARIZONA_WS_TRG2 0x0002 /* WS_TRG2 */ +#define ARIZONA_WS_TRG2_MASK 0x0002 /* WS_TRG2 */ +#define ARIZONA_WS_TRG2_SHIFT 1 /* WS_TRG2 */ +#define ARIZONA_WS_TRG2_WIDTH 1 /* WS_TRG2 */ +#define ARIZONA_WS_TRG1 0x0001 /* WS_TRG1 */ +#define ARIZONA_WS_TRG1_MASK 0x0001 /* WS_TRG1 */ +#define ARIZONA_WS_TRG1_SHIFT 0 /* WS_TRG1 */ +#define ARIZONA_WS_TRG1_WIDTH 1 /* WS_TRG1 */ + +/* * R97 (0x61) - Sample Rate Sequence Select 1 */ #define ARIZONA_WSEQ_SAMPLE_RATE_DETECT_A_SEQ_ADDR_MASK 0x01FF /* WSEQ_SAMPLE_RATE_DETECT_A_SEQ_ADDR - [8:0] */ -- 1.7.2.5

3 4

[alsa-devel] [PATCH 1/2] ASoC: Prevent components from being bound to multiple cards
by Lars-Peter Clausen 08 Jul '15

08 Jul '15

A component can only be bound to a single card at a time. Binding it to card while it is already bound to another will result in undefined behavior. As the undefined behavior might only manifest itself later on it is not necessarily always straight forward to find the cause. To prevent this add a check that refuses to bind a component to multiple cards as well as prints a error describing the problem. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8baf87d..d9ae826 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1104,9 +1104,19 @@ static int soc_probe_component(struct snd_soc_card *card, struct snd_soc_dai *dai; int ret; - if (!strcmp(component->name, "snd-soc-dummy") || component->probed) + if (!strcmp(component->name, "snd-soc-dummy")) return 0; + if (component->probed) { + if (component->card != card) { + dev_err(component->dev, + "Trying to bind component to card \"%s\" but is already bound to card \"%s\"\n", + card->name, component->card->name); + return -ENODEV; + } + return 0; + } + component->card = card; dapm->card = card; soc_set_name_prefix(card, component); -- 2.1.4

2 4

[alsa-devel] Getting stable PCM device id
by Alexandre Bique 08 Jul '15

08 Jul '15

Hi, I wonder how to get a PCM identifier that would be stable if you reboot or plug/unplug sound cards in a random order. I want to be able to store a sound card configuration and be able to restore it without relying on the card index or the device name, because if you have two times the same sound card it is not going to be unique. I'm not an expert in hardware ids, yet I imagine that we could use the hardware serial number or equivalent? Thank you very much, -- Alexandre Bique

2 2

[alsa-devel] [PATCH v2] ASoC: wm8960: update pll and clock setting function
by Zidan Wang 08 Jul '15

08 Jul '15

Add sysclk auto mode. When it's sysclk auto mode, if the MCLK is available for clock configure, using MCLK to provide sysclk directly, otherwise, search a available pll out frequcncy and set pll. TX and RX share the same sysclk and bclk divider register, and have different DAC/ADC divier register. When playback and capture simultaneously, the second stream should not set pll, should not change the sysclk and bclk, just need set DAC/ADC divider to support different TX and RX sample rate. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 256 ++++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/wm8960.h | 1 + 2 files changed, 214 insertions(+), 43 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 94c5c46..e170bdf 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -48,6 +48,9 @@ #define WM8960_DISOP 0x40 #define WM8960_DRES_MASK 0x30 +static bool is_pll_freq_available(unsigned int source, unsigned int target); +static int wm8960_set_pll(struct snd_soc_dai *codec_dai, + unsigned int freq_in, unsigned int freq_out); /* * wm8960 register cache * We can't read the WM8960 register space when we are @@ -126,9 +129,12 @@ struct wm8960_priv { struct snd_soc_dapm_widget *rout1; struct snd_soc_dapm_widget *out3; bool deemph; - int playback_fs; - int bclk; int sysclk; + int clk_id; + int freq_in; + int fs[2]; + int dac_div[2]; + bool is_stream_in_use[2]; struct wm8960_data pdata; }; @@ -164,8 +170,8 @@ static int wm8960_set_deemph(struct snd_soc_codec *codec) if (wm8960->deemph) { best = 1; for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { - if (abs(deemph_settings[i] - wm8960->playback_fs) < - abs(deemph_settings[best] - wm8960->playback_fs)) + if (abs(deemph_settings[i] - wm8960->fs[1]) < + abs(deemph_settings[best] - wm8960->fs[1])) best = i; } @@ -565,6 +571,9 @@ static struct { { 8000, 5 }, }; +/* -1 for reserved value */ +static const int sysclk_divs[] = { 1, -1, 2, -1 }; + /* Multiply 256 for internal 256 div */ static const int dac_divs[] = { 256, 384, 512, 768, 1024, 1408, 1536 }; @@ -574,61 +583,166 @@ static const int bclk_divs[] = { 120, 160, 220, 240, 320, 320, 320 }; -static void wm8960_configure_clocking(struct snd_soc_codec *codec, - bool tx, int lrclk) +static int wm8960_configure_clocking(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct snd_soc_codec *codec = dai->codec; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + unsigned int sample_rate = params_rate(params); + unsigned int channels = params_channels(params); + unsigned int sysclk, bclk, freq_out, freq_in; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u16 iface1 = snd_soc_read(codec, WM8960_IFACE1); u16 iface2 = snd_soc_read(codec, WM8960_IFACE2); - u32 sysclk; - int i, j; + int i, j, k; if (!(iface1 & (1<<6))) { dev_dbg(codec->dev, "Codec is slave mode, no need to configure clock\n"); - return; + return 0; + } + + /* + * playback and capture using the same registers of bclk and sysclk + * div, and different registers for dac/adc div. If playback and capture + * simultaneously, the second should not set pll, should not change + * the sysclk and bclk, only dac/adc div can be set to support different + * frame clock. + */ + if (wm8960->is_stream_in_use[!tx]) { + /* + * If ADCLRC configure as GPIO pin, DACLRC pin is used as a + * frame clock for ADCs and DACs. + */ + if (iface2 & (1 << 6)) { + snd_soc_update_bits(codec, WM8960_CLOCK1, + 0x7 << 3, wm8960->dac_div[!tx] << 3); + return 0; + } + + /* + * If the different TX and RX sample rate can be support, + * setting corresponding DAC or ADC divider. + */ + for (i = 0; i < ARRAY_SIZE(dac_divs); ++i) { + if (wm8960->fs[!tx] * dac_divs[wm8960->dac_div[!tx]] + == wm8960->fs[tx] * dac_divs[i]) { + if (tx) + snd_soc_update_bits(codec, + WM8960_CLOCK1, + 0x7 << 3, dac_divs[i] << 3); + else + snd_soc_update_bits(codec, + WM8960_CLOCK1, + 0x7 << 6, dac_divs[i] << 6); + wm8960->dac_div[tx] = dac_divs[i]; + return 0; + } + } + if (i == ARRAY_SIZE(dac_divs)) { + dev_err(codec->dev, + "can't support TX=%d RX=%d\n simultaneously", + wm8960->fs[1], wm8960->fs[0]); + return -EINVAL; + } + } + + if (wm8960->clk_id != WM8960_SYSCLK_MCLK && !wm8960->freq_in) { + dev_err(codec->dev, "No MCLK configured\n"); + return -EINVAL; } - if (!wm8960->sysclk) { - dev_dbg(codec->dev, "No SYSCLK configured\n"); - return; + freq_in = wm8960->freq_in; + + bclk = snd_soc_params_to_bclk(params); + if (channels == 1) + bclk *= 2; + + /* + * If it's sysclk auto mode, check if the MCLK can provide sysclk or + * not. If MCLK can provide sysclk, using MCLK to provide sysclk + * directly. Otherwise, auto select a available pll out frequency + * and set PLL. + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO) { + /* disable the PLL and using MCLK to provide sysclk */ + wm8960_set_pll(dai, 0, 0); + freq_out = freq_in; + } else if (wm8960->sysclk) + freq_out = wm8960->sysclk; + else { + dev_err(codec->dev, "No SYSCLK configured\n"); + return -EINVAL; } - if (!wm8960->bclk || !lrclk) { - dev_dbg(codec->dev, "No audio clocks configured\n"); - return; + /* check if the sysclk frequency is available. */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + sysclk = freq_out / sysclk_divs[i]; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + if (sysclk == dac_divs[j] * sample_rate) { + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) + if (sysclk == bclk * bclk_divs[k] / 10) + break; + if (k != ARRAY_SIZE(bclk_divs)) + break; + } + } + if (j != ARRAY_SIZE(dac_divs)) + break; } - for (i = 0; i < ARRAY_SIZE(dac_divs); ++i) { - if (wm8960->sysclk == lrclk * dac_divs[i]) { - for (j = 0; j < ARRAY_SIZE(bclk_divs); ++j) { - sysclk = wm8960->bclk * bclk_divs[j] / 10; - if (wm8960->sysclk == sysclk) + if (i != ARRAY_SIZE(sysclk_divs)) + goto configure_clock; + else if (wm8960->clk_id != WM8960_SYSCLK_AUTO) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; + } + /* get a available pll out frequency and set pll */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + sysclk = sample_rate * dac_divs[j]; + freq_out = sysclk * sysclk_divs[i]; + + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) { + if (sysclk == bclk * bclk_divs[k] / 10 && + is_pll_freq_available(freq_in, freq_out)) { + wm8960_set_pll(dai, freq_in, freq_out); break; + } else + continue; } - if(j != ARRAY_SIZE(bclk_divs)) + if (k != ARRAY_SIZE(bclk_divs)) break; } + if (j != ARRAY_SIZE(dac_divs)) + break; } - if (i == ARRAY_SIZE(dac_divs)) { - dev_err(codec->dev, "Unsupported sysclk %d\n", wm8960->sysclk); - return; + if (i == ARRAY_SIZE(sysclk_divs)) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; } - /* - * configure frame clock. If ADCLRC configure as GPIO pin, DACLRC - * pin is used as a frame clock for ADCs and DACs. - */ - if (iface2 & (1<<6)) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); +configure_clock: + snd_soc_update_bits(codec, WM8960_CLOCK1, 3 << 1, i << 1); + + if (iface2 & (1 << 6)) + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, j << 3); else if (tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, j << 3); else if (!tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, i << 6); + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, j << 6); /* configure bit clock */ - snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, j); + snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, k); + + wm8960->dac_div[tx] = j; + + return 0; } static int wm8960_hw_params(struct snd_pcm_substream *substream, @@ -641,10 +755,6 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int i; - wm8960->bclk = snd_soc_params_to_bclk(params); - if (params_channels(params) == 1) - wm8960->bclk *= 2; - /* bit size */ switch (params_width(params)) { case 16: @@ -667,11 +777,11 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + wm8960->fs[tx] = params_rate(params); /* Update filters for the new rate */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - wm8960->playback_fs = params_rate(params); + if (tx) wm8960_set_deemph(codec); - } else { + else { for (i = 0; i < ARRAY_SIZE(alc_rates); i++) if (alc_rates[i].rate == params_rate(params)) snd_soc_update_bits(codec, @@ -682,7 +792,27 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, /* set iface */ snd_soc_write(codec, WM8960_IFACE1, iface); - wm8960_configure_clocking(codec, tx, params_rate(params)); + wm8960->is_stream_in_use[tx] = true; + + return wm8960_configure_clocking(substream, params, dai); +} + +static int wm8960_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + u16 clock1 = snd_soc_read(codec, WM8960_CLOCK1); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + wm8960->is_stream_in_use[tx] = false; + /* + * If it's sysclk auto mode, no stream in use, and the pll is enabled, + * disable the pll + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO && (clock1 & 0x1) && + !wm8960->is_stream_in_use[!tx]) + wm8960_set_pll(dai, 0, 0); return 0; } @@ -892,6 +1022,28 @@ struct _pll_div { u32 k:24; }; +static bool is_pll_freq_available(unsigned int source, unsigned int target) +{ + unsigned int Ndiv; + + if (source == 0 || target == 0) + return false; + + /* Scale up target to PLL operating frequency */ + target *= 4; + Ndiv = target / source; + + if (Ndiv < 6) { + source >>= 1; + Ndiv = target / source; + } + + if ((Ndiv < 6) || (Ndiv > 12)) + return false; + + return true; +} + /* The size in bits of the pll divide multiplied by 10 * to allow rounding later */ #define FIXED_PLL_SIZE ((1 << 24) * 10) @@ -943,8 +1095,8 @@ static int pll_factors(unsigned int source, unsigned int target, return 0; } -static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) +static int wm8960_set_pll(struct snd_soc_dai *codec_dai, + unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; u16 reg; @@ -986,6 +1138,20 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, return 0; } +static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + + wm8960->freq_in = freq_in; + + if (wm8960->clk_id == WM8960_SYSCLK_AUTO) + return 0; + + return wm8960_set_pll(codec_dai, freq_in, freq_out); +} + static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { @@ -1043,11 +1209,14 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, snd_soc_update_bits(codec, WM8960_CLOCK1, 0x1, WM8960_SYSCLK_PLL); break; + case WM8960_SYSCLK_AUTO: + break; default: return -EINVAL; } wm8960->sysclk = freq; + wm8960->clk_id = clk_id; return 0; } @@ -1060,6 +1229,7 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, static const struct snd_soc_dai_ops wm8960_dai_ops = { .hw_params = wm8960_hw_params, + .hw_free = wm8960_hw_free, .digital_mute = wm8960_mute, .set_fmt = wm8960_set_dai_fmt, .set_clkdiv = wm8960_set_dai_clkdiv, diff --git a/sound/soc/codecs/wm8960.h b/sound/soc/codecs/wm8960.h index 2d8163d..ab3220d 100644 --- a/sound/soc/codecs/wm8960.h +++ b/sound/soc/codecs/wm8960.h @@ -82,6 +82,7 @@ #define WM8960_SYSCLK_MCLK (0 << 0) #define WM8960_SYSCLK_PLL (1 << 0) +#define WM8960_SYSCLK_AUTO (2 << 0) #define WM8960_DAC_DIV_1 (0 << 3) #define WM8960_DAC_DIV_1_5 (1 << 3) -- 1.9.1

2 1

[alsa-devel] Choosing the best PCM access
by Alexandre Bique 08 Jul '15

08 Jul '15

Hi, I just ran test/pcm.c and found that MMAP_NONINTERLEAVED was faster than MMAP_INTERLEAVED in my case. My test was to run it like: ./a.out -c 2 -b 3000 -p 1500 -m direct_nointerleaved And I got a lot more crackling in interleaved than in noninterleaved. Is it something very hardware/driver dependent and if so, how to know which is the best access? Thank you, -- Alexandre Bique

1 0

[alsa-devel] [PATCH v2 01/13] topology: uapi: Add UAPI headers for topology ABI
by Liam Girdwood 08 Jul '15

08 Jul '15

Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/asoc.h | 387 +++++++++++++++++++++++++++++++++++++++++++++++++++ include/sound/tlv.h | 23 +++ 2 files changed, 410 insertions(+) create mode 100644 include/sound/asoc.h create mode 100644 include/sound/tlv.h diff --git a/include/sound/asoc.h b/include/sound/asoc.h new file mode 100644 index 0000000..498cd0c --- /dev/null +++ b/include/sound/asoc.h @@ -0,0 +1,387 @@ +/* + * uapi/sound/asoc.h -- ALSA SoC Firmware Controls and DAPM + * + * Copyright (C) 2012 Texas Instruments Inc. + * Copyright (C) 2015 Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Simple file API to load FW that includes mixers, coefficients, DAPM graphs, + * algorithms, equalisers, DAIs, widgets etc. +*/ + +#ifndef __LINUX_UAPI_SND_ASOC_H +#define __LINUX_UAPI_SND_ASOC_H + +/* + * Maximum number of channels topology kcontrol can represent. + */ +#define SND_SOC_TPLG_MAX_CHAN 8 + +/* + * Maximum number of PCM formats capability + */ +#define SND_SOC_TPLG_MAX_FORMATS 16 + +/* + * Maximum number of PCM stream configs + */ +#define SND_SOC_TPLG_STREAM_CONFIG_MAX 8 + +/* individual kcontrol info types - can be mixed with other types */ +#define SND_SOC_TPLG_CTL_VOLSW 1 +#define SND_SOC_TPLG_CTL_VOLSW_SX 2 +#define SND_SOC_TPLG_CTL_VOLSW_XR_SX 3 +#define SND_SOC_TPLG_CTL_ENUM 4 +#define SND_SOC_TPLG_CTL_BYTES 5 +#define SND_SOC_TPLG_CTL_ENUM_VALUE 6 +#define SND_SOC_TPLG_CTL_RANGE 7 +#define SND_SOC_TPLG_CTL_STROBE 8 + + +/* individual widget kcontrol info types - can be mixed with other types */ +#define SND_SOC_TPLG_DAPM_CTL_VOLSW 64 +#define SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE 65 +#define SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT 66 +#define SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE 67 +#define SND_SOC_TPLG_DAPM_CTL_PIN 68 + +/* DAPM widget types - add new items to the end */ +#define SND_SOC_TPLG_DAPM_INPUT 0 +#define SND_SOC_TPLG_DAPM_OUTPUT 1 +#define SND_SOC_TPLG_DAPM_MUX 2 +#define SND_SOC_TPLG_DAPM_MIXER 3 +#define SND_SOC_TPLG_DAPM_PGA 4 +#define SND_SOC_TPLG_DAPM_OUT_DRV 5 +#define SND_SOC_TPLG_DAPM_ADC 6 +#define SND_SOC_TPLG_DAPM_DAC 7 +#define SND_SOC_TPLG_DAPM_SWITCH 8 +#define SND_SOC_TPLG_DAPM_PRE 9 +#define SND_SOC_TPLG_DAPM_POST 10 +#define SND_SOC_TPLG_DAPM_AIF_IN 11 +#define SND_SOC_TPLG_DAPM_AIF_OUT 12 +#define SND_SOC_TPLG_DAPM_DAI_IN 13 +#define SND_SOC_TPLG_DAPM_DAI_OUT 14 +#define SND_SOC_TPLG_DAPM_DAI_LINK 15 +#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DAI_LINK + +/* Header magic number and string sizes */ +#define SND_SOC_TPLG_MAGIC 0x41536F43 /* ASoC */ + +/* string sizes */ +#define SND_SOC_TPLG_NUM_TEXTS 16 + +/* ABI version */ +#define SND_SOC_TPLG_ABI_VERSION 0x2 + +/* Max size of TLV data */ +#define SND_SOC_TPLG_TLV_SIZE 32 + +/* + * File and Block header data types. + * Add new generic and vendor types to end of list. + * Generic types are handled by the core whilst vendors types are passed + * to the component drivers for handling. + */ +#define SND_SOC_TPLG_TYPE_MIXER 1 +#define SND_SOC_TPLG_TYPE_BYTES 2 +#define SND_SOC_TPLG_TYPE_ENUM 3 +#define SND_SOC_TPLG_TYPE_DAPM_GRAPH 4 +#define SND_SOC_TPLG_TYPE_DAPM_WIDGET 5 +#define SND_SOC_TPLG_TYPE_DAI_LINK 6 +#define SND_SOC_TPLG_TYPE_PCM 7 +#define SND_SOC_TPLG_TYPE_MANIFEST 8 +#define SND_SOC_TPLG_TYPE_CODEC_LINK 9 +#define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_TYPE_CODEC_LINK + +/* vendor block IDs - please add new vendor types to end */ +#define SND_SOC_TPLG_TYPE_VENDOR_FW 1000 +#define SND_SOC_TPLG_TYPE_VENDOR_CONFIG 1001 +#define SND_SOC_TPLG_TYPE_VENDOR_COEFF 1002 +#define SND_SOC_TPLG_TYPEVENDOR_CODEC 1003 + +#define SND_SOC_TPLG_STREAM_PLAYBACK 0 +#define SND_SOC_TPLG_STREAM_CAPTURE 1 + +/* + * Block Header. + * This header precedes all object and object arrays below. + */ +struct snd_soc_tplg_hdr { + __le32 magic; /* magic number */ + __le32 abi; /* ABI version */ + __le32 version; /* optional vendor specific version details */ + __le32 type; /* SND_SOC_TPLG_TYPE_ */ + __le32 size; /* size of this structure */ + __le32 vendor_type; /* optional vendor specific type info */ + __le32 payload_size; /* data bytes, excluding this header */ + __le32 index; /* identifier for block */ + __le32 count; /* number of elements in block */ +} __attribute__((packed)); + +/* + * Private data. + * All topology objects may have private data that can be used by the driver or + * firmware. Core will ignore this data. + */ +struct snd_soc_tplg_private { + __le32 size; /* in bytes of private data */ + char data[0]; +} __attribute__((packed)); + +/* + * Kcontrol TLV data. + */ +struct snd_soc_tplg_ctl_tlv { + __le32 size; /* in bytes aligned to 4 */ + __le32 numid; /* control element numeric identification */ + __le32 count; /* number of elem in data array */ + __le32 data[SND_SOC_TPLG_TLV_SIZE]; +} __attribute__((packed)); + +/* + * Kcontrol channel data + */ +struct snd_soc_tplg_channel { + __le32 size; /* in bytes of this structure */ + __le32 reg; + __le32 shift; + __le32 id; /* ID maps to Left, Right, LFE etc */ +} __attribute__((packed)); + +/* + * Kcontrol Operations IDs + */ +struct snd_soc_tplg_kcontrol_ops_id { + __le32 get; + __le32 put; + __le32 info; +} __attribute__((packed)); + +/* + * kcontrol header + */ +struct snd_soc_tplg_ctl_hdr { + __le32 size; /* in bytes of this structure */ + __le32 type; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 access; + struct snd_soc_tplg_kcontrol_ops_id ops; + __le32 tlv_size; /* non zero means control has TLV data */ +} __attribute__((packed)); + +/* + * Stream Capabilities + */ +struct snd_soc_tplg_stream_caps { + __le32 size; /* in bytes of this structure */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le64 formats[SND_SOC_TPLG_MAX_FORMATS]; /* supported formats SNDRV_PCM_FMTBIT_* */ + __le32 rates; /* supported rates SNDRV_PCM_RATE_* */ + __le32 rate_min; /* min rate */ + __le32 rate_max; /* max rate */ + __le32 channels_min; /* min channels */ + __le32 channels_max; /* max channels */ + __le32 periods_min; /* min number of periods */ + __le32 periods_max; /* max number of periods */ + __le32 period_size_min; /* min period size bytes */ + __le32 period_size_max; /* max period size bytes */ + __le32 buffer_size_min; /* min buffer size bytes */ + __le32 buffer_size_max; /* max buffer size bytes */ +} __attribute__((packed)); + +/* + * FE or BE Stream configuration supported by SW/FW + */ +struct snd_soc_tplg_stream { + __le32 size; /* in bytes of this structure */ + __le64 format; /* SNDRV_PCM_FMTBIT_* */ + __le32 rate; /* SNDRV_PCM_RATE_* */ + __le32 period_bytes; /* size of period in bytes */ + __le32 buffer_bytes; /* size of buffer in bytes */ + __le32 channels; /* channels */ + __le32 tdm_slot; /* optional BE bitmask of supported TDM slots */ + __le32 dai_fmt; /* SND_SOC_DAIFMT_ */ +} __attribute__((packed)); + +/* + * Duplex stream configuration supported by SW/FW. + */ +struct snd_soc_tplg_stream_config { + __le32 size; /* in bytes of this structure */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + struct snd_soc_tplg_stream playback; + struct snd_soc_tplg_stream capture; +} __attribute__((packed)); + +/* + * Manifest. List totals for each payload type. Not used in parsing, but will + * be passed to the component driver before any other objects in order for any + * global component resource allocations. + * + * File block representation for manifest :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_manifest | 1 | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_manifest { + __le32 size; /* in bytes of this structure */ + __le32 control_elems; /* number of control elements */ + __le32 widget_elems; /* number of widget elements */ + __le32 graph_elems; /* number of graph elements */ + __le32 dai_elems; /* number of DAI elements */ + __le32 dai_link_elems; /* number of DAI link elements */ + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * Mixer kcontrol. + * + * File block representation for mixer kcontrol :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_mixer_control | N | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_mixer_control { + struct snd_soc_tplg_ctl_hdr hdr; + __le32 size; /* in bytes of this structure */ + __le32 min; + __le32 max; + __le32 platform_max; + __le32 invert; + __le32 num_channels; + struct snd_soc_tplg_channel channel[SND_SOC_TPLG_MAX_CHAN]; + struct snd_soc_tplg_ctl_tlv tlv; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * Enumerated kcontrol + * + * File block representation for enum kcontrol :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_enum_control | N | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_enum_control { + struct snd_soc_tplg_ctl_hdr hdr; + __le32 size; /* in bytes of this structure */ + __le32 num_channels; + struct snd_soc_tplg_channel channel[SND_SOC_TPLG_MAX_CHAN]; + __le32 items; + __le32 mask; + __le32 count; + char texts[SND_SOC_TPLG_NUM_TEXTS][SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 values[SND_SOC_TPLG_NUM_TEXTS * SNDRV_CTL_ELEM_ID_NAME_MAXLEN / 4]; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * Bytes kcontrol + * + * File block representation for bytes kcontrol :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_bytes_control | N | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_bytes_control { + struct snd_soc_tplg_ctl_hdr hdr; + __le32 size; /* in bytes of this structure */ + __le32 max; + __le32 mask; + __le32 base; + __le32 num_regs; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * DAPM Graph Element + * + * File block representation for DAPM graph elements :- + * +-------------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-------------------------------------+----+ + * | struct snd_soc_tplg_dapm_graph_elem | N | + * +-------------------------------------+----+ + */ +struct snd_soc_tplg_dapm_graph_elem { + char sink[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char control[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char source[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; +} __attribute__((packed)); + +/* + * DAPM Widget. + * + * File block representation for DAPM widget :- + * +-------------------------------------+-----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-------------------------------------+-----+ + * | struct snd_soc_tplg_dapm_widget | N | + * +-------------------------------------+-----+ + * | struct snd_soc_tplg_enum_control | 0|1 | + * | struct snd_soc_tplg_mixer_control | 0|N | + * +-------------------------------------+-----+ + * + * Optional enum or mixer control can be appended to the end of each widget + * in the block. + */ +struct snd_soc_tplg_dapm_widget { + __le32 size; /* in bytes of this structure */ + __le32 id; /* SND_SOC_DAPM_CTL */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char sname[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + __le32 reg; /* negative reg = no direct dapm */ + __le32 shift; /* bits to shift */ + __le32 mask; /* non-shifted mask */ + __le32 subseq; /* sort within widget type */ + __u32 invert; /* invert the power bit */ + __u32 ignore_suspend; /* kept enabled over suspend */ + __u16 event_flags; + __u16 event_type; + __u16 num_kcontrols; + struct snd_soc_tplg_private priv; + /* + * kcontrols that relate to this widget + * follow here after widget private data + */ +} __attribute__((packed)); + +struct snd_soc_tplg_pcm_cfg_caps { + struct snd_soc_tplg_stream_caps caps; + struct snd_soc_tplg_stream_config configs[SND_SOC_TPLG_STREAM_CONFIG_MAX]; + __le32 num_configs; /* number of configs */ +} __attribute__((packed)); + +/* + * Describes SW/FW specific features of PCM or DAI link. + * + * File block representation for PCM/DAI-Link :- + * +-----------------------------------+-----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+-----+ + * | struct snd_soc_tplg_dapm_pcm_dai | N | + * +-----------------------------------+-----+ + */ +struct snd_soc_tplg_pcm_dai { + __le32 size; /* in bytes of this structure */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 id; /* unique ID - used to match */ + __le32 playback; /* supports playback mode */ + __le32 capture; /* supports capture mode */ + __le32 compress; /* 1 = compressed; 0 = PCM */ + struct snd_soc_tplg_pcm_cfg_caps capconf[2]; /* capabilities and configs */ +} __attribute__((packed)); + +#endif diff --git a/include/sound/tlv.h b/include/sound/tlv.h new file mode 100644 index 0000000..33d747d --- /dev/null +++ b/include/sound/tlv.h @@ -0,0 +1,23 @@ +/* + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#ifndef __UAPI_SOUND_TLV_H +#define __UAPI_SOUND_TLV_H + +#define SNDRV_CTL_TLVT_CONTAINER 0 /* one level down - group of TLVs */ +#define SNDRV_CTL_TLVT_DB_SCALE 1 /* dB scale */ +#define SNDRV_CTL_TLVT_DB_LINEAR 2 /* linear volume */ +#define SNDRV_CTL_TLVT_DB_RANGE 3 /* dB range container */ +#define SNDRV_CTL_TLVT_DB_MINMAX 4 /* dB scale with min/max */ +#define SNDRV_CTL_TLVT_DB_MINMAX_MUTE 5 /* dB scale with min/max with mute */ + +#endif -- 2.1.4

3 25

[alsa-devel] [PATCH] regmap: Use different lockdep class for each regmap init call
by Nicolas Boichat 08 Jul '15

08 Jul '15

Lockdep validator complains about recursive locking and deadlock when two different regmap instances are called in a nested order. That happens anytime a regmap read/write call needs to access another regmap. This is because, for performance reason, lockdep groups all locks initialized by the same mutex_init() in the same lock class. Therefore all regmap mutexes are in the same lock class, leading to lockdep "nested locking" warnings if a regmap accesses another regmap. In general, it is impossible to establish in advance the hierarchy of regmaps, so we make sure that each regmap init call initializes its own static lock_class_key. This is done by wrapping all regmap_init calls into macros. This also allows us to give meaningful names to the lock_class_key. For example, in rt5677 case, we have in /proc/lockdep_chains: irq_context: 0 [ffffffc0018d2198] &dev->mutex [ffffffc0018d2198] &dev->mutex [ffffffc001bd7f60] rt5677:5104:(&rt5677_regmap)->_lock [ffffffc001bd7f58] rt5677:5096:(&rt5677_regmap_physical)->_lock [ffffffc001b95448] &(&base->lock)->rlock The above would have resulted in a lockdep recursive warning previously. This is not the case anymore as the lockdep validator now clearly identifies the 2 regmaps as separate. Signed-off-by: Nicolas Boichat <drinkcat(a)chromium.org> --- Following up on the discussion here: http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/094124.html , and Lars-Peter's idea on how to define the macros, this converts all regmap_init calls to macros. One thing that is left over is the documentation. As it stands, regmap_init is documented just before __regmap_init definition in drivers/base/regmap.c (this also applies to all other regmap_init functions...). We do not want to document the lock_key/name parameters, so we could either leave it like that, or move all the documentation to regmap.h, in front of the respective macros. Thanks! drivers/base/regmap/regmap-ac97.c | 22 +++-- drivers/base/regmap/regmap-i2c.c | 22 +++-- drivers/base/regmap/regmap-mmio.c | 27 ++++-- drivers/base/regmap/regmap-spi.c | 22 +++-- drivers/base/regmap/regmap-spmi.c | 44 +++++---- drivers/base/regmap/regmap.c | 31 ++++--- include/linux/regmap.h | 190 ++++++++++++++++++++++++++++---------- 7 files changed, 248 insertions(+), 110 deletions(-) diff --git a/drivers/base/regmap/regmap-ac97.c b/drivers/base/regmap/regmap-ac97.c index 8d304e2..aa631be 100644 --- a/drivers/base/regmap/regmap-ac97.c +++ b/drivers/base/regmap/regmap-ac97.c @@ -87,12 +87,15 @@ static const struct regmap_bus ac97_regmap_bus = { * The return value will be an ERR_PTR() on error or a valid pointer to * a struct regmap. */ -struct regmap *regmap_init_ac97(struct snd_ac97 *ac97, - const struct regmap_config *config) +struct regmap *__regmap_init_ac97(struct snd_ac97 *ac97, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name) { - return regmap_init(&ac97->dev, &ac97_regmap_bus, ac97, config); + return __regmap_init(&ac97->dev, &ac97_regmap_bus, ac97, config, + lock_key, lock_name); } -EXPORT_SYMBOL_GPL(regmap_init_ac97); +EXPORT_SYMBOL_GPL(__regmap_init_ac97); /** * devm_regmap_init_ac97(): Initialise AC'97 register map @@ -104,11 +107,14 @@ EXPORT_SYMBOL_GPL(regmap_init_ac97); * to a struct regmap. The regmap will be automatically freed by the * device management code. */ -struct regmap *devm_regmap_init_ac97(struct snd_ac97 *ac97, - const struct regmap_config *config) +struct regmap *__devm_regmap_init_ac97(struct snd_ac97 *ac97, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name) { - return devm_regmap_init(&ac97->dev, &ac97_regmap_bus, ac97, config); + return __devm_regmap_init(&ac97->dev, &ac97_regmap_bus, ac97, config, + lock_key, lock_name); } -EXPORT_SYMBOL_GPL(devm_regmap_init_ac97); +EXPORT_SYMBOL_GPL(__devm_regmap_init_ac97); MODULE_LICENSE("GPL v2"); diff --git a/drivers/base/regmap/regmap-i2c.c b/drivers/base/regmap/regmap-i2c.c index 4b76e33..3163b22 100644 --- a/drivers/base/regmap/regmap-i2c.c +++ b/drivers/base/regmap/regmap-i2c.c @@ -242,17 +242,20 @@ static const struct regmap_bus *regmap_get_i2c_bus(struct i2c_client *i2c, * The return value will be an ERR_PTR() on error or a valid pointer to * a struct regmap. */ -struct regmap *regmap_init_i2c(struct i2c_client *i2c, - const struct regmap_config *config) +struct regmap *__regmap_init_i2c(struct i2c_client *i2c, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name) { const struct regmap_bus *bus = regmap_get_i2c_bus(i2c, config); if (IS_ERR(bus)) return ERR_CAST(bus); - return regmap_init(&i2c->dev, bus, &i2c->dev, config); + return __regmap_init(&i2c->dev, bus, &i2c->dev, config, + lock_key, lock_name); } -EXPORT_SYMBOL_GPL(regmap_init_i2c); +EXPORT_SYMBOL_GPL(__regmap_init_i2c); /** * devm_regmap_init_i2c(): Initialise managed register map @@ -264,16 +267,19 @@ EXPORT_SYMBOL_GPL(regmap_init_i2c); * to a struct regmap. The regmap will be automatically freed by the * device management code. */ -struct regmap *devm_regmap_init_i2c(struct i2c_client *i2c, - const struct regmap_config *config) +struct regmap *__devm_regmap_init_i2c(struct i2c_client *i2c, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name) { const struct regmap_bus *bus = regmap_get_i2c_bus(i2c, config); if (IS_ERR(bus)) return ERR_CAST(bus); - return devm_regmap_init(&i2c->dev, bus, &i2c->dev, config); + return __devm_regmap_init(&i2c->dev, bus, &i2c->dev, config, + lock_key, lock_name); } -EXPORT_SYMBOL_GPL(devm_regmap_init_i2c); +EXPORT_SYMBOL_GPL(__devm_regmap_init_i2c); MODULE_LICENSE("GPL"); diff --git a/drivers/base/regmap/regmap-mmio.c b/drivers/base/regmap/regmap-mmio.c index 04a329a..a1b2b27 100644 --- a/drivers/base/regmap/regmap-mmio.c +++ b/drivers/base/regmap/regmap-mmio.c @@ -307,9 +307,11 @@ err_free: * The return value will be an ERR_PTR() on error or a valid pointer to * a struct regmap. */ -struct regmap *regmap_init_mmio_clk(struct device *dev, const char *clk_id, - void __iomem *regs, - const struct regmap_config *config) +struct regmap *__regmap_init_mmio_clk(struct device *dev, const char *clk_id, + void __iomem *regs, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name) { struct regmap_mmio_context *ctx; @@ -317,9 +319,10 @@ struct regmap *regmap_init_mmio_clk(struct device *dev, const char *clk_id, if (IS_ERR(ctx)) return ERR_CAST(ctx); - return regmap_init(dev, &regmap_mmio, ctx, config); + return __regmap_init(dev, &regmap_mmio, ctx, config, + lock_key, lock_name); } -EXPORT_SYMBOL_GPL(regmap_init_mmio_clk); +EXPORT_SYMBOL_GPL(__regmap_init_mmio_clk); /** * devm_regmap_init_mmio_clk(): Initialise managed register map with clock @@ -333,9 +336,12 @@ EXPORT_SYMBOL_GPL(regmap_init_mmio_clk); * to a struct regmap. The regmap will be automatically freed by the * device management code. */ -struct regmap *devm_regmap_init_mmio_clk(struct device *dev, const char *clk_id, - void __iomem *regs, - const struct regmap_config *config) +struct regmap *__devm_regmap_init_mmio_clk(struct device *dev, + const char *clk_id, + void __iomem *regs, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name) { struct regmap_mmio_context *ctx; @@ -343,8 +349,9 @@ struct regmap *devm_regmap_init_mmio_clk(struct device *dev, const char *clk_id, if (IS_ERR(ctx)) return ERR_CAST(ctx); - return devm_regmap_init(dev, &regmap_mmio, ctx, config); + return __devm_regmap_init(dev, &regmap_mmio, ctx, config, + lock_key, lock_name); } -EXPORT_SYMBOL_GPL(devm_regmap_init_mmio_clk); +EXPORT_SYMBOL_GPL(__devm_regmap_init_mmio_clk); MODULE_LICENSE("GPL v2"); diff --git a/drivers/base/regmap/regmap-spi.c b/drivers/base/regmap/regmap-spi.c index 53d1148..4c7850d 100644 --- a/drivers/base/regmap/regmap-spi.c +++ b/drivers/base/regmap/regmap-spi.c @@ -122,12 +122,15 @@ static struct regmap_bus regmap_spi = { * The return value will be an ERR_PTR() on error or a valid pointer to * a struct regmap. */ -struct regmap *regmap_init_spi(struct spi_device *spi, - const struct regmap_config *config) +struct regmap *__regmap_init_spi(struct spi_device *spi, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name) { - return regmap_init(&spi->dev, &regmap_spi, &spi->dev, config); + return __regmap_init(&spi->dev, &regmap_spi, &spi->dev, config, + lock_key, lock_name); } -EXPORT_SYMBOL_GPL(regmap_init_spi); +EXPORT_SYMBOL_GPL(__regmap_init_spi); /** * devm_regmap_init_spi(): Initialise register map @@ -139,11 +142,14 @@ EXPORT_SYMBOL_GPL(regmap_init_spi); * to a struct regmap. The map will be automatically freed by the * device management code. */ -struct regmap *devm_regmap_init_spi(struct spi_device *spi, - const struct regmap_config *config) +struct regmap *__devm_regmap_init_spi(struct spi_device *spi, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name) { - return devm_regmap_init(&spi->dev, &regmap_spi, &spi->dev, config); + return __devm_regmap_init(&spi->dev, &regmap_spi, &spi->dev, config, + lock_key, lock_name); } -EXPORT_SYMBOL_GPL(devm_regmap_init_spi); +EXPORT_SYMBOL_GPL(__devm_regmap_init_spi); MODULE_LICENSE("GPL"); diff --git a/drivers/base/regmap/regmap-spmi.c b/drivers/base/regmap/regmap-spmi.c index d7026dc..7f50f58 100644 --- a/drivers/base/regmap/regmap-spmi.c +++ b/drivers/base/regmap/regmap-spmi.c @@ -99,12 +99,15 @@ static struct regmap_bus regmap_spmi_base = { * The return value will be an ERR_PTR() on error or a valid pointer to * a struct regmap. */ -struct regmap *regmap_init_spmi_base(struct spmi_device *sdev, - const struct regmap_config *config) +struct regmap *__regmap_init_spmi_base(struct spmi_device *sdev, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name) { - return regmap_init(&sdev->dev, &regmap_spmi_base, sdev, config); + return __regmap_init(&sdev->dev, &regmap_spmi_base, sdev, config, + lock_key, lock_name); } -EXPORT_SYMBOL_GPL(regmap_init_spmi_base); +EXPORT_SYMBOL_GPL(__regmap_init_spmi_base); /** * devm_regmap_init_spmi_base(): Create managed regmap for Base register space @@ -115,12 +118,15 @@ EXPORT_SYMBOL_GPL(regmap_init_spmi_base); * to a struct regmap. The regmap will be automatically freed by the * device management code. */ -struct regmap *devm_regmap_init_spmi_base(struct spmi_device *sdev, - const struct regmap_config *config) +struct regmap *__devm_regmap_init_spmi_base(struct spmi_device *sdev, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name) { - return devm_regmap_init(&sdev->dev, &regmap_spmi_base, sdev, config); + return __devm_regmap_init(&sdev->dev, &regmap_spmi_base, sdev, config, + lock_key, lock_name); } -EXPORT_SYMBOL_GPL(devm_regmap_init_spmi_base); +EXPORT_SYMBOL_GPL(__devm_regmap_init_spmi_base); static int regmap_spmi_ext_read(void *context, const void *reg, size_t reg_size, @@ -230,12 +236,15 @@ static struct regmap_bus regmap_spmi_ext = { * The return value will be an ERR_PTR() on error or a valid pointer to * a struct regmap. */ -struct regmap *regmap_init_spmi_ext(struct spmi_device *sdev, - const struct regmap_config *config) +struct regmap *__regmap_init_spmi_ext(struct spmi_device *sdev, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name) { - return regmap_init(&sdev->dev, &regmap_spmi_ext, sdev, config); + return __regmap_init(&sdev->dev, &regmap_spmi_ext, sdev, config, + lock_key, lock_name); } -EXPORT_SYMBOL_GPL(regmap_init_spmi_ext); +EXPORT_SYMBOL_GPL(__regmap_init_spmi_ext); /** * devm_regmap_init_spmi_ext(): Create managed regmap for Ext register space @@ -246,11 +255,14 @@ EXPORT_SYMBOL_GPL(regmap_init_spmi_ext); * to a struct regmap. The regmap will be automatically freed by the * device management code. */ -struct regmap *devm_regmap_init_spmi_ext(struct spmi_device *sdev, - const struct regmap_config *config) +struct regmap *__devm_regmap_init_spmi_ext(struct spmi_device *sdev, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name) { - return devm_regmap_init(&sdev->dev, &regmap_spmi_ext, sdev, config); + return __devm_regmap_init(&sdev->dev, &regmap_spmi_ext, sdev, config, + lock_key, lock_name); } -EXPORT_SYMBOL_GPL(devm_regmap_init_spmi_ext); +EXPORT_SYMBOL_GPL(__devm_regmap_init_spmi_ext); MODULE_LICENSE("GPL"); diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index 7111d04..b9fddcc 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -527,10 +527,12 @@ EXPORT_SYMBOL_GPL(regmap_get_val_endian); * a struct regmap. This function should generally not be called * directly, it should be called by bus-specific init functions. */ -struct regmap *regmap_init(struct device *dev, - const struct regmap_bus *bus, - void *bus_context, - const struct regmap_config *config) +struct regmap *__regmap_init(struct device *dev, + const struct regmap_bus *bus, + void *bus_context, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name) { struct regmap *map; int ret = -EINVAL; @@ -556,10 +558,14 @@ struct regmap *regmap_init(struct device *dev, spin_lock_init(&map->spinlock); map->lock = regmap_lock_spinlock; map->unlock = regmap_unlock_spinlock; + lockdep_set_class_and_name(&map->spinlock, + lock_key, lock_name); } else { mutex_init(&map->mutex); map->lock = regmap_lock_mutex; map->unlock = regmap_unlock_mutex; + lockdep_set_class_and_name(&map->mutex, + lock_key, lock_name); } map->lock_arg = map; } @@ -899,7 +905,7 @@ err_map: err: return ERR_PTR(ret); } -EXPORT_SYMBOL_GPL(regmap_init); +EXPORT_SYMBOL_GPL(__regmap_init); static void devm_regmap_release(struct device *dev, void *res) { @@ -919,10 +925,12 @@ static void devm_regmap_release(struct device *dev, void *res) * directly, it should be called by bus-specific init functions. The * map will be automatically freed by the device management code. */ -struct regmap *devm_regmap_init(struct device *dev, - const struct regmap_bus *bus, - void *bus_context, - const struct regmap_config *config) +struct regmap *__devm_regmap_init(struct device *dev, + const struct regmap_bus *bus, + void *bus_context, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name) { struct regmap **ptr, *regmap; @@ -930,7 +938,8 @@ struct regmap *devm_regmap_init(struct device *dev, if (!ptr) return ERR_PTR(-ENOMEM); - regmap = regmap_init(dev, bus, bus_context, config); + regmap = __regmap_init(dev, bus, bus_context, config, + lock_key, lock_name); if (!IS_ERR(regmap)) { *ptr = regmap; devres_add(dev, ptr); @@ -940,7 +949,7 @@ struct regmap *devm_regmap_init(struct device *dev, return regmap; } -EXPORT_SYMBOL_GPL(devm_regmap_init); +EXPORT_SYMBOL_GPL(__devm_regmap_init); static void regmap_field_init(struct regmap_field *rm_field, struct regmap *regmap, struct reg_field reg_field) diff --git a/include/linux/regmap.h b/include/linux/regmap.h index 59c55ea..02b4d01 100644 --- a/include/linux/regmap.h +++ b/include/linux/regmap.h @@ -17,6 +17,7 @@ #include <linux/rbtree.h> #include <linux/err.h> #include <linux/bug.h> +#include <linux/lockdep.h> struct module; struct device; @@ -324,46 +325,145 @@ struct regmap_bus { enum regmap_endian val_format_endian_default; }; -struct regmap *regmap_init(struct device *dev, - const struct regmap_bus *bus, - void *bus_context, - const struct regmap_config *config); -int regmap_attach_dev(struct device *dev, struct regmap *map, - const struct regmap_config *config); -struct regmap *regmap_init_i2c(struct i2c_client *i2c, - const struct regmap_config *config); -struct regmap *regmap_init_spi(struct spi_device *dev, - const struct regmap_config *config); -struct regmap *regmap_init_spmi_base(struct spmi_device *dev, - const struct regmap_config *config); -struct regmap *regmap_init_spmi_ext(struct spmi_device *dev, - const struct regmap_config *config); -struct regmap *regmap_init_mmio_clk(struct device *dev, const char *clk_id, - void __iomem *regs, - const struct regmap_config *config); -struct regmap *regmap_init_ac97(struct snd_ac97 *ac97, - const struct regmap_config *config); - -struct regmap *devm_regmap_init(struct device *dev, - const struct regmap_bus *bus, - void *bus_context, - const struct regmap_config *config); -struct regmap *devm_regmap_init_i2c(struct i2c_client *i2c, - const struct regmap_config *config); -struct regmap *devm_regmap_init_spi(struct spi_device *dev, - const struct regmap_config *config); -struct regmap *devm_regmap_init_spmi_base(struct spmi_device *dev, - const struct regmap_config *config); -struct regmap *devm_regmap_init_spmi_ext(struct spmi_device *dev, - const struct regmap_config *config); -struct regmap *devm_regmap_init_mmio_clk(struct device *dev, const char *clk_id, - void __iomem *regs, - const struct regmap_config *config); -struct regmap *devm_regmap_init_ac97(struct snd_ac97 *ac97, - const struct regmap_config *config); +/* + * __regmap_init functions. + * + * These functions take a lock key and name parameter, and should not be called + * directly. Instead, use the regmap_init macros that generate a key and name + * for each call. + */ +struct regmap *__regmap_init(struct device *dev, + const struct regmap_bus *bus, + void *bus_context, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name); +struct regmap *__regmap_init_i2c(struct i2c_client *i2c, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name); +struct regmap *__regmap_init_spi(struct spi_device *dev, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name); +struct regmap *__regmap_init_spmi_base(struct spmi_device *dev, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name); +struct regmap *__regmap_init_spmi_ext(struct spmi_device *dev, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name); +struct regmap *__regmap_init_mmio_clk(struct device *dev, const char *clk_id, + void __iomem *regs, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name); +struct regmap *__regmap_init_ac97(struct snd_ac97 *ac97, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name); + +struct regmap *__devm_regmap_init(struct device *dev, + const struct regmap_bus *bus, + void *bus_context, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name); +struct regmap *__devm_regmap_init_i2c(struct i2c_client *i2c, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name); +struct regmap *__devm_regmap_init_spi(struct spi_device *dev, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name); +struct regmap *__devm_regmap_init_spmi_base(struct spmi_device *dev, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name); +struct regmap *__devm_regmap_init_spmi_ext(struct spmi_device *dev, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name); +struct regmap *__devm_regmap_init_mmio_clk(struct device *dev, + const char *clk_id, + void __iomem *regs, + const struct regmap_config *config, + struct lock_class_key *lock_key, + const char *lock_name); +struct regmap *__devm_regmap_init_ac97(struct snd_ac97 *ac97, + const struct regmap_config *config); +/* + * Wrapper for regmap_init macros to include a unique lockdep key and name + * for each call. No-op if CONFIG_LOCKDEP is not set. + * + * @fn: Real function to call (in the form __[*_]regmap_init[_*]) + * @name: Config variable name (#config in the calling macro) + **/ +#ifdef CONFIG_LOCKDEP +#define __regmap_lockdep_wrapper(fn, name, ...) \ +( \ + ({ \ + static struct lock_class_key _key; \ + fn(__VA_ARGS__, &_key, \ + KBUILD_BASENAME ":" \ + __stringify(__LINE__) ":" \ + "(" name ")->lock"); \ + }) \ +) +#else +#define __regmap_lockdep_wrapper(fn, name, ...) fn(__VA_ARGS__, NULL, NULL) +#endif + +#define regmap_init(dev, bus, bus_context, config) \ + __regmap_lockdep_wrapper(__regmap_init, #config, \ + dev, bus, bus_context, config) +int regmap_attach_dev(struct device *dev, struct regmap *map, + const struct regmap_config *config); +#define regmap_init_i2c(i2c, config) \ + __regmap_lockdep_wrapper(__regmap_init_i2c, #config, \ + i2c, config) +#define regmap_init_spi(dev, config) \ + __regmap_lockdep_wrapper(__regmap_init_spi, #config, \ + dev, config) +#define regmap_init_spmi_base(dev, config) \ + __regmap_lockdep_wrapper(__regmap_init_spmi_base, #config, \ + dev, config) +#define regmap_init_spmi_ext(dev, config) \ + __regmap_lockdep_wrapper(__regmap_init_spmi_ext, #config, \ + dev, config) +#define regmap_init_mmio_clk(dev, clk_id, regs, config) \ + __regmap_lockdep_wrapper(__regmap_init_mmio_clk, #config, \ + dev, clk_id, regs, config) +#define regmap_init_ac97(ac97, config) \ + __regmap_lockdep_wrapper(__regmap_init_ac97, #config, \ + ac97, config) bool regmap_ac97_default_volatile(struct device *dev, unsigned int reg); +#define devm_regmap_init(dev, bus, bus_context, config) \ + __regmap_lockdep_wrapper(__devm_regmap_init, #config, \ + dev, bus, bus_context, config) +#define devm_regmap_init_i2c(i2c, config) \ + __regmap_lockdep_wrapper(__devm_regmap_init_i2c, #config, \ + i2c, config) +#define devm_regmap_init_spi(dev, config) \ + __regmap_lockdep_wrapper(__devm_regmap_init_spi, #config, \ + dev, config) +#define devm_regmap_init_spmi_base(dev, config) \ + __regmap_lockdep_wrapper(__devm_regmap_init_spmi_base, #config, \ + dev, config) +#define devm_regmap_init_spmi_ext(dev, config) \ + __regmap_lockdep_wrapper(__devm_regmap_init_spmi_ext, #config, \ + dev, config) +#define devm_regmap_init_mmio_clk(dev, clk_id, regs, config) \ + __regmap_lockdep_wrapper(__devm_regmap_init_mmio_clk, #config, \ + dev, clk_id, regs, config) +#define devm_regmap_init_ac97(ac97, config) \ + __regmap_lockdep_wrapper(__devm_regmap_init_ac97, #config, \ + ac97, config) + /** * regmap_init_mmio(): Initialise register map * @@ -374,12 +474,8 @@ bool regmap_ac97_default_volatile(struct device *dev, unsigned int reg); * The return value will be an ERR_PTR() on error or a valid pointer to * a struct regmap. */ -static inline struct regmap *regmap_init_mmio(struct device *dev, - void __iomem *regs, - const struct regmap_config *config) -{ - return regmap_init_mmio_clk(dev, NULL, regs, config); -} +#define regmap_init_mmio(dev, regs, config) \ + regmap_init_mmio_clk(dev, NULL, regs, config) /** * devm_regmap_init_mmio(): Initialise managed register map @@ -392,12 +488,8 @@ static inline struct regmap *regmap_init_mmio(struct device *dev, * to a struct regmap. The regmap will be automatically freed by the * device management code. */ -static inline struct regmap *devm_regmap_init_mmio(struct device *dev, - void __iomem *regs, - const struct regmap_config *config) -{ - return devm_regmap_init_mmio_clk(dev, NULL, regs, config); -} +#define devm_regmap_init_mmio(dev, regs, config) \ + devm_regmap_init_mmio_clk(dev, NULL, regs, config) void regmap_exit(struct regmap *map); int regmap_reinit_cache(struct regmap *map, -- 2.4.3.573.g4eafbef

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Re: [alsa-devel] [pulseaudio-discuss] Strange hardware: XFX nForce 780i SLI MCP
by Raymond Yau 08 Jul '15

08 Jul '15

> > I can confirm that there is no such setting in the BIOS. > Node 0x11 [Pin Complex] wcaps 0x400300: Mono Digital Control: name="HDMI Phantom Jack", index=0, device=0 Pincap 0x00000010: OUT Pin Default 0x18567530: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Yellow DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x10 Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Control: name="SPDIF Phantom Jack", index=0, device=0 Pincap 0x00000010: OUT Pin Default 0x01441520: [Jack] SPDIF Out at Ext Rear Conn = RCA, Color = Black DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 The easy workaround is use hdajackretask change the internal spdif connector node 0x11 as not connected if you only use spdif out at rear panel node 0x1e Otherwise , you need to sort autocfg.dig_out_pins by def assoc or location so the pin with spdif out at ext instead of the internal spdif connector digitak out at Int is first digital pin https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci…

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[alsa-devel] [PATCH 1/3] emu10k1: remove unused AC'97 mixer controls on Audigy
by Maciej S. Szmigiero 08 Jul '15

08 Jul '15

AC'97 Headphone output and EAPD control aren't used on Audigy so remove them from mixer. Also remove AC'97 3D control as the driver is already doing for Audigys with 1361T ADC. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/pci/emu10k1/emumixer.c | 11 ++++++----- 1 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 55e5716..f1cb7b3 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1830,10 +1830,16 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, "Capture Switch", "Capture Volume", "Mic Select", + "Headphone Playback Switch", + "Headphone Playback Volume", + "3D Control - Center", + "3D Control - Depth", + "3D Control - Switch", "Video Playback Switch", "Video Playback Volume", "Mic Playback Switch", "Mic Playback Volume", + "External Amplifier", NULL }; static char *audigy_rename_ctls[] = { @@ -1996,11 +2002,6 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, rename_ctl(card, "Analog Mix Capture Volume", "Line2 Capture Volume"); rename_ctl(card, "Aux2 Capture Volume", "Line3 Capture Volume"); rename_ctl(card, "Mic Capture Volume", "Unknown1 Capture Volume"); - remove_ctl(card, "Headphone Playback Switch"); - remove_ctl(card, "Headphone Playback Volume"); - remove_ctl(card, "3D Control - Center"); - remove_ctl(card, "3D Control - Depth"); - remove_ctl(card, "3D Control - Switch"); } if ((kctl = emu->ctl_send_routing = snd_ctl_new1(&snd_emu10k1_send_routing_control, emu)) == NULL) return -ENOMEM;

2 1

Re: [alsa-devel] [pulseaudio-discuss] Strange hardware: XFX nForce 780i SLI MCP
by Raymond Yau 08 Jul '15

08 Jul '15

> > Yesterday I got an interesting email related to dcaenc support. > > The person has an "XFX nForce 780i SLI" MCP board: http://www.nvidia.com/object/product_nforce_780i_sli_us.html , http://www.evga.com/support/manuals/files/132-CK-NF78.pdf > > As you see on page 18 of the manual, this board has no HDMI output, but there is an optical SPDIF. Strangely enough, this SPDIF is his "hdmi:CARD=NVidia,DEV=0" device. > > Given the recent discussions related to dynamically creating PCM devices for HDMI, I think you will be interested in this alsa-info output. > > Is there anything here that would need special treatment by ALSA or PulseAudio? Or, is this just a misnumbered/misnamed codec? > Check whether BIOS set the internal spdif as HDMI as pin default The alsa driver prefer HDMI instead of SPDIF when there are two Digital Outs

2 2

[alsa-devel] [PATCH] ASoC: ts3a227e: do not report jack status when there is i2c read err
by yang.a.fang＠intel.com 07 Jul '15

07 Jul '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> After suspend -> resume the ts3a227e_interrupt sometimes comes before i2c controller resume is called .regmap_read will return incorrect status and report a wrong jack status.We should return if there is read err,the interrupt will come again since it is level triggered and we are not yet clear the interrupt. In addtion,cht_bsw_max98090_ti machine driver registered additional notifier base on jack event which will program the audio codec.there will be codec timeout err if such event occurs prior to i2c controller is resumed. Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/codecs/ts3a227e.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index 12232d7..12d0f2a 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -23,6 +23,7 @@ #include "ts3a227e.h" struct ts3a227e { + struct device *dev; struct regmap *regmap; struct snd_soc_jack *jack; bool plugged; @@ -189,16 +190,28 @@ static irqreturn_t ts3a227e_interrupt(int irq, void *data) struct ts3a227e *ts3a227e = (struct ts3a227e *)data; struct regmap *regmap = ts3a227e->regmap; unsigned int int_reg, kp_int_reg, acc_reg, i; + struct device *dev = ts3a227e->dev; + int ret; /* Check for plug/unplug. */ - regmap_read(regmap, TS3A227E_REG_INTERRUPT, &int_reg); + ret = regmap_read(regmap, TS3A227E_REG_INTERRUPT, &int_reg); + if (ret) { + dev_err(dev, "failed to clear interrupt ret=%d\n", ret); + return IRQ_HANDLED; + } + if (int_reg & (DETECTION_COMPLETE_EVENT | INS_REM_EVENT)) { regmap_read(regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg); ts3a227e_new_jack_state(ts3a227e, acc_reg); } /* Report any key events. */ - regmap_read(regmap, TS3A227E_REG_KP_INTERRUPT, &kp_int_reg); + ret = regmap_read(regmap, TS3A227E_REG_KP_INTERRUPT, &kp_int_reg); + if (ret) { + dev_err(dev, "failed to clear key interrupt ret=%d\n", ret); + return IRQ_HANDLED; + } + for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) { if (kp_int_reg & PRESS_MASK(i)) ts3a227e->buttons_held |= (1 << i); @@ -283,6 +296,7 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, ts3a227e); + ts3a227e->dev = dev; ts3a227e->regmap = devm_regmap_init_i2c(i2c, &ts3a227e_regmap_config); if (IS_ERR(ts3a227e->regmap)) -- 1.7.9.5

5 8

[alsa-devel] [PATCH 3/3] emu10k1: rename Audigy Analog Capture Boost control
by Maciej S. Szmigiero 07 Jul '15

07 Jul '15

Audigy has "Analog Capture Boost" mixer control, however now this only controls mic level, not other analog sources. It applies also both to playback and capture, so rename it to something more descriptive. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/pci/emu10k1/emumixer.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 567b5cb..076b117 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1741,7 +1741,7 @@ static int snd_audigy_capture_boost_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new snd_audigy_capture_boost = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Capture Boost", + .name = "Mic Extra Boost", .info = snd_audigy_capture_boost_info, .get = snd_audigy_capture_boost_get, .put = snd_audigy_capture_boost_put

1 0

[alsa-devel] [PATCH 2/3] emu10k1: enable TAD mic out on Audigy
by Maciej S. Szmigiero 07 Jul '15

07 Jul '15

Enable TAD output on Audigy naming it "Phone Output", to be consistent with TAD input which is called "Phone". According to Creative doc ( http://support.creative.com/kb/ShowArticle.aspx?sid=3026 ) this should output just mic signal. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/pci/emu10k1/emumixer.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index f1cb7b3..567b5cb 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1819,8 +1819,6 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, * the Philips ADC for 24bit capture */ "PCM Playback Switch", "PCM Playback Volume", - "Master Mono Playback Switch", - "Master Mono Playback Volume", "Master Playback Switch", "Master Playback Volume", "PCM Out Path & Mute", @@ -1848,6 +1846,8 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, /* "Wave Capture Volume", "PCM Capture Volume", */ "Wave Master Playback Volume", "Master Playback Volume", "AMic Playback Volume", "Mic Playback Volume", + "Master Mono Playback Switch", "Phone Output Playback Switch", + "Master Mono Playback Volume", "Phone Output Playback Volume", NULL }; static char *audigy_rename_ctls_i2c_adc[] = { @@ -1873,8 +1873,6 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, * the Philips ADC for 24bit capture */ "PCM Playback Switch", "PCM Playback Volume", - "Master Mono Playback Switch", - "Master Mono Playback Volume", "Capture Source", "Capture Switch", "Capture Volume", @@ -1906,7 +1904,8 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, "Aux Playback Volume", "Aux Capture Volume", "Video Playback Switch", "Video Capture Switch", "Video Playback Volume", "Video Capture Volume", - + "Master Mono Playback Switch", "Phone Output Playback Switch", + "Master Mono Playback Volume", "Phone Output Playback Volume", NULL }; @@ -1941,6 +1940,9 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, snd_ac97_write_cache(emu->ac97, AC97_MASTER, 0x0000); /* set capture source to mic */ snd_ac97_write_cache(emu->ac97, AC97_REC_SEL, 0x0000); + /* set mono output (TAD) to mic */ + snd_ac97_update_bits(emu->ac97, AC97_GENERAL_PURPOSE, + 0x0200, 0x0200); if (emu->card_capabilities->adc_1361t) c = audigy_remove_ctls_1361t_adc; else

1 0

[alsa-devel] [PATCH 1/4] ASoC: max98090: Check for clk_prepare_enable() error
by Fabio Estevam 07 Jul '15

07 Jul '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> clk_prepare_enable() may fail, so we should better check its return value and propagate it in the case of error. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/max98090.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 78268f05..1697340 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1801,10 +1801,13 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec, if (IS_ERR(max98090->mclk)) break; - if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON) + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON) { clk_disable_unprepare(max98090->mclk); - else - clk_prepare_enable(max98090->mclk); + } else { + ret = clk_prepare_enable(max98090->mclk); + if (ret) + return ret; + } break; case SND_SOC_BIAS_STANDBY: -- 1.9.1

2 4

[alsa-devel] [PATCH V2] ASoC: core: Don't probe the component which is dummy
by Shengjiu Wang 07 Jul '15

07 Jul '15

Dummy dai can be used by multiple sound card. But it only belong to one card's dapm list. If another card use it, there will be dapm_assert_locked warning. [ 20.015782] WARNING: CPU: 1 PID: 661 at sound/soc/soc-dapm.c:124 dapm_assert_locked.isra.36+0x4c/0x58() [ 20.025249] Modules linked in: [ 20.028349] CPU: 1 PID: 661 Comm: aplay Not tainted 4.1.0-rc6-next-20150605-00004-gaee05d8-dirty #92 [ 20.037528] Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) [ 20.044110] Backtrace: [ 20.046614] [<80012e00>] (dump_backtrace) from [<80012fa0>] (show_stack+0x18/0x1c) [ 20.054229] r6:809e8060 r5:00000000 r4:00000000 r3:00000000 [ 20.060002] [<80012f88>] (show_stack) from [<807a0f74>] (dump_stack+0x80/0x9c) [ 20.067293] [<807a0ef4>] (dump_stack) from [<8002b144>] (warn_slowpath_common+0x7c/0xb4) [ 20.075427] r5:0000007c r4:00000000 [ 20.079065] [<8002b0c8>] (warn_slowpath_common) from [<8002b1a0>] (warn_slowpath_null+0x24/0x2c) [ 20.087898] r8:00000001 r7:88007c28 r6:ed94a680 r5:809e83e4 r4:ed83d6c0 [ 20.094747] [<8002b17c>] (warn_slowpath_null) from [<8058403c>] (dapm_assert_locked.isra.36+0x4c/0x58) [ 20.104101] [<80583ff0>] (dapm_assert_locked.isra.36) from [<805842ec>] (dapm_mark_dirty+0x64/0xa4) [ 20.113165] [<80584288>] (dapm_mark_dirty) from [<805853a8>] (soc_dapm_dai_stream_event.isra.42+0x30/0xc8) [ 20.122863] r8:ed9b5dbc r7:00000000 r6:00000001 r5:00000001 r4:ed83d6c0 [ 20.129706] [<80585378>] (soc_dapm_dai_stream_event.isra.42) from [<80587e28>] (snd_soc_dapm_stream_event+0x78/0xa0) [ 20.140264] r5:ee2ee62c r4:00000001 [ 20.143918] [<80587db0>] (snd_soc_dapm_stream_event) from [<8058957c>] (soc_pcm_prepare+0x138/0x21c) [ 20.153058] r8:ed8d9480 r7:00000000 r6:ed9b0e00 r5:00000001 r4:ee2ee62c r3:00000000 ... This patch is to not probe the dummy component in soc_probe_component. Then there is no widget created for dummy DAI, and also don't need to check the dummy dai in dapm_connect_dai_link_widgets(). Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> --- changes in v2 - don't probe the component which is dummy - remove the dummy check in dapm_connect_dai_link_widgets(). sound/soc/soc-core.c | 2 +- sound/soc/soc-dapm.c | 5 ----- 2 files changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3a4a5c0..6ce6217 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1102,7 +1102,7 @@ static int soc_probe_component(struct snd_soc_card *card, struct snd_soc_dai *dai; int ret; - if (component->probed) + if (!strcmp(component->name, "snd-soc-dummy") || component->probed) return 0; component->card = card; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index aa327c9..37ab6b9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3820,11 +3820,6 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, for (i = 0; i < rtd->num_codecs; i++) { struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; - /* there is no point in connecting BE DAI links with dummies */ - if (snd_soc_dai_is_dummy(codec_dai) || - snd_soc_dai_is_dummy(cpu_dai)) - continue; - /* connect BE DAI playback if widgets are valid */ if (codec_dai->playback_widget && cpu_dai->playback_widget) { source = cpu_dai->playback_widget; -- 1.7.9.5

2 1

[alsa-devel] Rane SL-4 USB sound card
by Tuukka Pasanen 07 Jul '15

07 Jul '15

Hello, I've been hacking to get Rane SL-4 USB audio (http://dj.rane.com/products/sl4-for-serato-scratch-live) which have 4 Input and 4 output channel to work with ALSA. After digging the lsusb and snoops I have separated endpoints for input (0x82) and output (0x06). They are just USB Isochronous I/O. I have compiled kernel and tried to get USB-quirks to work with this but then I have to admit that even Reloop Play seems be very similar it doesn't do the trick (No error but card doesn't come up eather) nor using Ranel SL-1 approach which just shoots the error [ 3555.009349] usb 1-5: Audio class v2 interfaces need an interface association [ 3555.009373] snd-usb-audio: probe of 1-5:1.0 failed with error -5 [ 3555.009394] usb 1-5: cannot find UAC_HEADER [ 3555.009401] snd-usb-audio: probe of 1-5:1.1 failed with error -5 [ 3555.009421] usb 1-5: cannot find UAC_HEADER [ 3555.009427] snd-usb-audio: probe of 1-5:1.2 failed with error -5 If I need interface association how to do that? Trying to adjust mixer values in mixer_maps.c. If someone have a time to give some guidance if this doable with Quirks or do I have to write some more specific driver? I attach lsusb below here. HID device seems to be adjusting buffer size and 48 Khz and 96 Khz output. Bus 001 Device 003: ID 1cc5:000f Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 255 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x1cc5 idProduct 0x000f bcdDevice 1.00 iManufacturer 1 Rane Corporation iProduct 2 Rane SL 4 iSerial 3 RH.01.00 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 235 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 32 iInterface 4 SL 4 Audio ** UNRECOGNIZED: 09 24 01 00 02 0a 4b 00 00 ** UNRECOGNIZED: 08 24 0a 05 01 01 00 00 ** UNRECOGNIZED: 11 24 02 01 01 01 00 05 0a 00 00 00 00 05 00 00 00 ** UNRECOGNIZED: 0c 24 03 02 02 06 00 01 05 00 00 00 ** UNRECOGNIZED: 11 24 02 03 02 06 00 05 0a 00 00 00 00 0f 00 00 00 ** UNRECOGNIZED: 0c 24 03 04 01 01 00 03 05 00 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 32 iInterface 0 ** UNRECOGNIZED: 10 24 01 01 00 01 01 00 00 00 0a 00 00 00 00 05 ** UNRECOGNIZED: 06 24 02 01 04 18 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x06 EP 6 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0208 1x 520 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 32 iInterface 0 ** UNRECOGNIZED: 10 24 01 03 00 01 01 00 00 00 0a 00 00 00 00 0f ** UNRECOGNIZED: 06 24 02 01 04 18 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x0208 1x 520 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.10 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 28 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 1 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 255 bDeviceProtocol 255 bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0000 (Bus Powered) Thanks you for your time and any tips would be nice. Sincerely, Tuukka

2 10

[alsa-devel] [PATCH] ASoC: topology: Fix TLV size calculation.
by Liam Girdwood 07 Jul '15

07 Jul '15

TLV size calculation was incorrectly calculated. Fix this according to include/sound/tlv.h Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-topology.c | 21 ++++++++++----------- 1 file changed, 10 insertions(+), 11 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index d096068..7a19df3 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -580,27 +580,26 @@ static int soc_tplg_init_kcontrol(struct soc_tplg *tplg, } static int soc_tplg_create_tlv(struct soc_tplg *tplg, - struct snd_kcontrol_new *kc, u32 tlv_size) + struct snd_kcontrol_new *kc, struct snd_soc_tplg_ctl_tlv *tplg_tlv) { - struct snd_soc_tplg_ctl_tlv *tplg_tlv; struct snd_ctl_tlv *tlv; + int size; - if (tlv_size == 0) + if (tplg_tlv->count == 0) return 0; - tplg_tlv = (struct snd_soc_tplg_ctl_tlv *) tplg->pos; - tplg->pos += tlv_size; - - tlv = kzalloc(sizeof(*tlv) + tlv_size, GFP_KERNEL); + size = ((tplg_tlv->count + (sizeof(unsigned int) - 1)) & + ~(sizeof(unsigned int) - 1)); + tlv = kzalloc(sizeof(*tlv) + size, GFP_KERNEL); if (tlv == NULL) return -ENOMEM; dev_dbg(tplg->dev, " created TLV type %d size %d bytes\n", - tplg_tlv->numid, tplg_tlv->size); + tplg_tlv->numid, size); tlv->numid = tplg_tlv->numid; - tlv->length = tplg_tlv->size; - memcpy(tlv->tlv, tplg_tlv + 1, tplg_tlv->size); + tlv->length = size; + memcpy(&tlv->tlv[0], tplg_tlv->data, size); kc->tlv.p = (void *)tlv; return 0; @@ -773,7 +772,7 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count, } /* create any TLV data */ - soc_tplg_create_tlv(tplg, &kc, mc->hdr.tlv_size); + soc_tplg_create_tlv(tplg, &kc, &mc->tlv); /* register control here */ err = soc_tplg_add_kcontrol(tplg, &kc, -- 2.1.4

2 7

[alsa-devel] [PATCH 0/2] ASoc: simple card: extend for MCLK support
by Arnaud Pouliquen 07 Jul '15

07 Jul '15

Patches propose to extend the management of the mclk-fs ratio. This extension allows to support following card constraints: - mclk is generated by cpu_dai, depending on frame rate. cpu_dai needs to be informed on the ratio supported by codec. - Each codec needs a specific ratio. In this case mclk-fs needs to be defined for the dai_link. Arnaud Pouliquen (2): ASoC: simple card: Add mclk-fs property in dai-link ASoC: simple card: set cpu-dai sysclk with mclk-fs .../devicetree/bindings/sound/simple-card.txt | 6 ++++- sound/soc/generic/simple-card.c | 27 +++++++++++++++++++--- 2 files changed, 29 insertions(+), 4 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH v2] ASoC: wm8731: initialize the hardware when loading the codec driver
by Sergey Kiselev 07 Jul '15

07 Jul '15

From: Sergey Kiselev <sergey.kiselev(a)intel.com> This patch moves the requesting supplies, hardware reset and initialization from wm8731_probe to wm8731_i2c_probe and wm8731_spi_probe. So that the codec hardware is initialized when loading the codec driver, and not when loading the machine driver. This avoids unnecesary hardware resets and re-initializations when re-loading the machine driver. Signed-off-by: Sergey Kiselev <sergey.kiselev(a)intel.com> --- sound/soc/codecs/wm8731.c | 74 +++++++++++++++++++++++++++-------------------- 1 file changed, 42 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 915ea11..f22935a 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -84,7 +84,7 @@ static bool wm8731_writeable(struct device *dev, unsigned int reg) return reg <= WM8731_RESET; } -#define wm8731_reset(c) snd_soc_write(c, WM8731_RESET, 0) +#define wm8731_reset(m) regmap_write(m, WM8731_RESET, 0) static const char *wm8731_input_select[] = {"Line In", "Mic"}; @@ -571,69 +571,63 @@ static struct snd_soc_dai_driver wm8731_dai = { .symmetric_rates = 1, }; -static int wm8731_probe(struct snd_soc_codec *codec) +static int wm8731_request_supplies(struct device *dev, + struct wm8731_priv *wm8731) { - struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); int ret = 0, i; for (i = 0; i < ARRAY_SIZE(wm8731->supplies); i++) wm8731->supplies[i].supply = wm8731_supply_names[i]; - ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8731->supplies), + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(wm8731->supplies), wm8731->supplies); if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + dev_err(dev, "Failed to request supplies: %d\n", ret); return ret; } ret = regulator_bulk_enable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); if (ret != 0) { - dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + dev_err(dev, "Failed to enable supplies: %d\n", ret); return ret; } - ret = wm8731_reset(codec); + return 0; +} + +static int wm8731_hw_init(struct device *dev, struct wm8731_priv *wm8731) +{ + int ret = 0; + + ret = wm8731_reset(wm8731->regmap); if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset: %d\n", ret); + dev_err(dev, "Failed to issue reset: %d\n", ret); goto err_regulator_enable; } - snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Clear POWEROFF, keep everything else disabled */ + regmap_write(wm8731->regmap, WM8731_PWR, 0x7f); /* Latch the update bits */ - snd_soc_update_bits(codec, WM8731_LOUT1V, 0x100, 0); - snd_soc_update_bits(codec, WM8731_ROUT1V, 0x100, 0); - snd_soc_update_bits(codec, WM8731_LINVOL, 0x100, 0); - snd_soc_update_bits(codec, WM8731_RINVOL, 0x100, 0); + regmap_update_bits(wm8731->regmap, WM8731_LOUT1V, 0x100, 0); + regmap_update_bits(wm8731->regmap, WM8731_ROUT1V, 0x100, 0); + regmap_update_bits(wm8731->regmap, WM8731_LINVOL, 0x100, 0); + regmap_update_bits(wm8731->regmap, WM8731_RINVOL, 0x100, 0); /* Disable bypass path by default */ - snd_soc_update_bits(codec, WM8731_APANA, 0x8, 0); + regmap_update_bits(wm8731->regmap, WM8731_APANA, 0x8, 0); - /* Regulators will have been enabled by bias management */ - regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); - - return 0; + regcache_mark_dirty(wm8731->regmap); err_regulator_enable: + /* Regulators will be enabled by bias management */ regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); return ret; } -/* power down chip */ -static int wm8731_remove(struct snd_soc_codec *codec) -{ - struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); - - regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); - - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_wm8731 = { - .probe = wm8731_probe, - .remove = wm8731_remove, .set_bias_level = wm8731_set_bias_level, .suspend_bias_off = true, @@ -690,6 +684,12 @@ static int wm8731_spi_probe(struct spi_device *spi) mutex_init(&wm8731->lock); + spi_set_drvdata(spi, wm8731); + + ret = wm8731_request_supplies(&spi->dev, wm8731); + if (ret != 0) + return ret; + wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); @@ -698,7 +698,9 @@ static int wm8731_spi_probe(struct spi_device *spi) return ret; } - spi_set_drvdata(spi, wm8731); + ret = wm8731_hw_init(&spi->dev, wm8731); + if (ret != 0) + return ret; ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8731, &wm8731_dai, 1); @@ -754,6 +756,12 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, mutex_init(&wm8731->lock); + i2c_set_clientdata(i2c, wm8731); + + ret = wm8731_request_supplies(&i2c->dev, wm8731); + if (ret != 0) + return ret; + wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); @@ -762,7 +770,9 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, return ret; } - i2c_set_clientdata(i2c, wm8731); + ret = wm8731_hw_init(&i2c->dev, wm8731); + if (ret != 0) + return ret; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8731, &wm8731_dai, 1); -- 1.9.1

3 5

[alsa-devel] [PATCH] ASoC: rt5645: move RT5645 muxes to rt5645_specific_dapm_widgets
by Oder Chiou 07 Jul '15

07 Jul '15

From: Bard Liao <bardliao(a)realtek.com> This is a similar patch to "move RT5650 muxes to rt5650_specific_ dapm_widgets" patch. The purpose is to silence the "has no paths" warnings. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 39 ++++++++++++++++++++++----------------- 1 file changed, 22 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index d0750be..8c8a2b6 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -1697,15 +1697,6 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), /* IF1 2 Mux */ - SND_SOC_DAPM_MUX("RT5645 IF1 ADC1 Swap Mux", SND_SOC_NOPM, - 0, 0, &rt5645_if1_adc1_in_mux), - SND_SOC_DAPM_MUX("RT5645 IF1 ADC2 Swap Mux", SND_SOC_NOPM, - 0, 0, &rt5645_if1_adc2_in_mux), - SND_SOC_DAPM_MUX("RT5645 IF1 ADC3 Swap Mux", SND_SOC_NOPM, - 0, 0, &rt5645_if1_adc3_in_mux), - SND_SOC_DAPM_MUX("RT5645 IF1 ADC Mux", SND_SOC_NOPM, - 0, 0, &rt5645_if1_adc_in_mux), - SND_SOC_DAPM_MUX("IF2 ADC Mux", SND_SOC_NOPM, 0, 0, &rt5645_if2_adc_in_mux), @@ -1716,14 +1707,6 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 DAC2", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 DAC3", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_MUX("RT5645 IF1 DAC1 L Mux", SND_SOC_NOPM, 0, 0, - &rt5645_if1_dac0_tdm_sel_mux), - SND_SOC_DAPM_MUX("RT5645 IF1 DAC1 R Mux", SND_SOC_NOPM, 0, 0, - &rt5645_if1_dac1_tdm_sel_mux), - SND_SOC_DAPM_MUX("RT5645 IF1 DAC2 L Mux", SND_SOC_NOPM, 0, 0, - &rt5645_if1_dac2_tdm_sel_mux), - SND_SOC_DAPM_MUX("RT5645 IF1 DAC2 R Mux", SND_SOC_NOPM, 0, 0, - &rt5645_if1_dac3_tdm_sel_mux), SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -1856,6 +1839,25 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("SPOR"), }; +static const struct snd_soc_dapm_widget rt5645_specific_dapm_widgets[] = { + SND_SOC_DAPM_MUX("RT5645 IF1 DAC1 L Mux", SND_SOC_NOPM, 0, 0, + &rt5645_if1_dac0_tdm_sel_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 DAC1 R Mux", SND_SOC_NOPM, 0, 0, + &rt5645_if1_dac1_tdm_sel_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 DAC2 L Mux", SND_SOC_NOPM, 0, 0, + &rt5645_if1_dac2_tdm_sel_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 DAC2 R Mux", SND_SOC_NOPM, 0, 0, + &rt5645_if1_dac3_tdm_sel_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 ADC Mux", SND_SOC_NOPM, + 0, 0, &rt5645_if1_adc_in_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 ADC1 Swap Mux", SND_SOC_NOPM, + 0, 0, &rt5645_if1_adc1_in_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 ADC2 Swap Mux", SND_SOC_NOPM, + 0, 0, &rt5645_if1_adc2_in_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 ADC3 Swap Mux", SND_SOC_NOPM, + 0, 0, &rt5645_if1_adc3_in_mux), +}; + static const struct snd_soc_dapm_widget rt5650_specific_dapm_widgets[] = { SND_SOC_DAPM_MUX("A DAC1 L Mux", SND_SOC_NOPM, 0, 0, &rt5650_a_dac1_l_mux), @@ -3050,6 +3052,9 @@ static int rt5645_probe(struct snd_soc_codec *codec) switch (rt5645->codec_type) { case CODEC_TYPE_RT5645: + snd_soc_dapm_new_controls(&codec->dapm, + rt5645_specific_dapm_widgets, + ARRAY_SIZE(rt5645_specific_dapm_widgets)); snd_soc_dapm_add_routes(&codec->dapm, rt5645_specific_dapm_routes, ARRAY_SIZE(rt5645_specific_dapm_routes)); -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: codecs: max98088: fix typo in constant
by Nik Nyby 07 Jul '15

07 Jul '15

This fixes a typo in the M98088_REG_02_JACK_STATUS constant. Signed-off-by: Nik Nyby <nikolas(a)gnu.org> --- sound/soc/codecs/max98088.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98088.h b/sound/soc/codecs/max98088.h index be89a4f..efa39bf 100644 --- a/sound/soc/codecs/max98088.h +++ b/sound/soc/codecs/max98088.h @@ -16,7 +16,7 @@ */ #define M98088_REG_00_IRQ_STATUS 0x00 #define M98088_REG_01_MIC_STATUS 0x01 -#define M98088_REG_02_JACK_STAUS 0x02 +#define M98088_REG_02_JACK_STATUS 0x02 #define M98088_REG_03_BATTERY_VOLTAGE 0x03 #define M98088_REG_0F_IRQ_ENABLE 0x0F #define M98088_REG_10_SYS_CLK 0x10 -- 2.1.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: arizona: Implement stability check for EQ coefficients
by Charles Keepax 07 Jul '15

07 Jul '15

Specifying unstable coefficients for the EQ can have a severe impact on the audio. This patchs adds a stability check on the coefficients written to the EQ, for this it is necessary to merge the mode control and the coefficients as some coefficients may only be unstable with a certain mode setting so it is ideal if these are always updated in sync. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 59 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 10 +++++++ sound/soc/codecs/wm5102.c | 12 +++------ sound/soc/codecs/wm5110.c | 12 +++------ sound/soc/codecs/wm8997.c | 12 +++------ 5 files changed, 81 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 802e05e..3996786 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2313,6 +2313,65 @@ const struct snd_kcontrol_new arizona_adsp2_rate_controls[] = { }; EXPORT_SYMBOL_GPL(arizona_adsp2_rate_controls); +static bool arizona_eq_filter_unstable(bool mode, __be16 _a, __be16 _b) +{ + s16 a = be16_to_cpu(_a); + s16 b = be16_to_cpu(_b); + + if (!mode) { + return abs(a) >= 4096; + } else { + if (abs(b) >= 4096) + return true; + + return (abs((a << 16) / (4096 - b)) >= 4096 << 4); + } +} + +int arizona_eq_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + struct soc_bytes *params = (void *)kcontrol->private_value; + unsigned int val; + __be16 *data; + int len; + int ret; + + len = params->num_regs * regmap_get_val_bytes(arizona->regmap); + + data = kmemdup(ucontrol->value.bytes.data, len, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + data[0] &= cpu_to_be16(ARIZONA_EQ1_B1_MODE); + + if (arizona_eq_filter_unstable(!!data[0], data[1], data[2]) || + arizona_eq_filter_unstable(true, data[4], data[5]) || + arizona_eq_filter_unstable(true, data[8], data[9]) || + arizona_eq_filter_unstable(true, data[12], data[13]) || + arizona_eq_filter_unstable(false, data[16], data[17])) { + dev_err(arizona->dev, "Rejecting unstable EQ coefficients\n"); + ret = -EINVAL; + goto out; + } + + ret = regmap_read(arizona->regmap, params->base, &val); + if (ret != 0) + goto out; + + val &= ~ARIZONA_EQ1_B1_MODE; + data[0] |= cpu_to_be16(val); + + ret = regmap_raw_write(arizona->regmap, params->base, data, len); + +out: + kfree(data); + return ret; +} +EXPORT_SYMBOL_GPL(arizona_eq_coeff_put); + MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 43deb04..ebe50cf 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -194,6 +194,13 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; ARIZONA_MIXER_ROUTES(name " Preloader", name "L"), \ ARIZONA_MIXER_ROUTES(name " Preloader", name "R") +#define ARIZONA_EQ_CONTROL(xname, xbase) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \ + .put = arizona_eq_coeff_put, .private_value = \ + ((unsigned long)&(struct soc_bytes) { .base = xbase, \ + .num_regs = 20, .mask = ~ARIZONA_EQ1_B1_MODE }) } + #define ARIZONA_RATE_ENUM_SIZE 4 extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE]; extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE]; @@ -229,6 +236,9 @@ extern int arizona_hp_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +extern int arizona_eq_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index d097f09e5..556bd98 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -788,8 +788,7 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), -SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ1 Coefficients", ARIZONA_EQ1_2), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -801,8 +800,7 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), -SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ2 Coefficients", ARIZONA_EQ2_2), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -814,8 +812,7 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), -SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ3 Coefficients", ARIZONA_EQ3_2), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -827,8 +824,7 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), -SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ4 Coefficients", ARIZONA_EQ4_2), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 709fcc6..dfa4d4b 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -247,8 +247,7 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), -SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ1 Coefficients", ARIZONA_EQ1_2), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -260,8 +259,7 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), -SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ2 Coefficients", ARIZONA_EQ2_2), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -273,8 +271,7 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), -SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ3 Coefficients", ARIZONA_EQ3_2), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -286,8 +283,7 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), -SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ4 Coefficients", ARIZONA_EQ4_2), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 4134dc7..53e0a46 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -174,8 +174,7 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), -SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ1 Coefficients", ARIZONA_EQ1_2), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -187,8 +186,7 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), -SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ2 Coefficients", ARIZONA_EQ2_2), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -200,8 +198,7 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), -SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ3 Coefficients", ARIZONA_EQ3_2), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -213,8 +210,7 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), -SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ4 Coefficients", ARIZONA_EQ4_2), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: fsl: Add dedicated DMA buffer size for each cpu dai
by Shengjiu Wang 07 Jul '15

07 Jul '15

As the ssi is not the only cpu dai, there are esai, spdif, sai. and imx_pcm_dma can be used by all of them. Especially ESAI need a larger DMA buffer size. So Add dedicated DMA buffer for each cpu dai. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> --- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/fsl/fsl_spdif.c | 2 +- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/fsl/imx-pcm-dma.c | 25 +++++++++++++++++++++---- sound/soc/fsl/imx-pcm.h | 9 +++++++-- sound/soc/fsl/imx-ssi.c | 2 +- 7 files changed, 33 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 5c75971..8c2ddc1 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -839,7 +839,7 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } - ret = imx_pcm_dma_init(pdev); + ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE); if (ret) dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5c73bea..a18fd92 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -791,7 +791,7 @@ static int fsl_sai_probe(struct platform_device *pdev) return ret; if (sai->sai_on_imx) - return imx_pcm_dma_init(pdev); + return imx_pcm_dma_init(pdev, IMX_SAI_DMABUF_SIZE); else return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); } diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 8e93221..d1e9be7 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1255,7 +1255,7 @@ static int fsl_spdif_probe(struct platform_device *pdev) return ret; } - ret = imx_pcm_dma_init(pdev); + ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE); if (ret) dev_err(&pdev->dev, "imx_pcm_dma_init failed: %d\n", ret); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c7647e0..e122dab 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1257,7 +1257,7 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, if (ret) goto error_pcm; } else { - ret = imx_pcm_dma_init(pdev); + ret = imx_pcm_dma_init(pdev, IMX_SSI_DMABUF_SIZE); if (ret) goto error_pcm; } diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 0db94f49..1fc01ed 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -40,7 +40,7 @@ static const struct snd_pcm_hardware imx_pcm_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, - .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, + .buffer_bytes_max = IMX_DEFAULT_DMABUF_SIZE, .period_bytes_min = 128, .period_bytes_max = 65535, /* Limited by SDMA engine */ .periods_min = 2, @@ -52,13 +52,30 @@ static const struct snd_dmaengine_pcm_config imx_dmaengine_pcm_config = { .pcm_hardware = &imx_pcm_hardware, .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, .compat_filter_fn = filter, - .prealloc_buffer_size = IMX_SSI_DMABUF_SIZE, + .prealloc_buffer_size = IMX_DEFAULT_DMABUF_SIZE, }; -int imx_pcm_dma_init(struct platform_device *pdev) +int imx_pcm_dma_init(struct platform_device *pdev, size_t size) { + struct snd_dmaengine_pcm_config *config; + struct snd_pcm_hardware *pcm_hardware; + + config = devm_kzalloc(&pdev->dev, + sizeof(struct snd_dmaengine_pcm_config), GFP_KERNEL); + *config = imx_dmaengine_pcm_config; + if (size) + config->prealloc_buffer_size = size; + + pcm_hardware = devm_kzalloc(&pdev->dev, + sizeof(struct snd_pcm_hardware), GFP_KERNEL); + *pcm_hardware = imx_pcm_hardware; + if (size) + pcm_hardware->buffer_bytes_max = size; + + config->pcm_hardware = pcm_hardware; + return devm_snd_dmaengine_pcm_register(&pdev->dev, - &imx_dmaengine_pcm_config, + config, SND_DMAENGINE_PCM_FLAG_COMPAT); } EXPORT_SYMBOL_GPL(imx_pcm_dma_init); diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index c79cb27..133c4470a 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -20,6 +20,11 @@ */ #define IMX_SSI_DMABUF_SIZE (64 * 1024) +#define IMX_DEFAULT_DMABUF_SIZE (64 * 1024) +#define IMX_SAI_DMABUF_SIZE (64 * 1024) +#define IMX_SPDIF_DMABUF_SIZE (64 * 1024) +#define IMX_ESAI_DMABUF_SIZE (256 * 1024) + static inline void imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data, int dma, enum sdma_peripheral_type peripheral_type) @@ -39,9 +44,9 @@ struct imx_pcm_fiq_params { }; #if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_DMA) -int imx_pcm_dma_init(struct platform_device *pdev); +int imx_pcm_dma_init(struct platform_device *pdev, size_t size); #else -static inline int imx_pcm_dma_init(struct platform_device *pdev) +static inline int imx_pcm_dma_init(struct platform_device *pdev, size_t size) { return -ENODEV; } diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 461ce27..48b2d24 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -603,7 +603,7 @@ static int imx_ssi_probe(struct platform_device *pdev) ssi->fiq_params.dma_params_tx = &ssi->dma_params_tx; ssi->fiq_init = imx_pcm_fiq_init(pdev, &ssi->fiq_params); - ssi->dma_init = imx_pcm_dma_init(pdev); + ssi->dma_init = imx_pcm_dma_init(pdev, IMX_SSI_DMABUF_SIZE); if (ssi->fiq_init && ssi->dma_init) { ret = ssi->fiq_init; -- 1.7.9.5

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[alsa-devel] [PATCH v2] ASoC: fsl: fsl_asrc: Check for clk_prepare_enable() error
by Fabio Estevam 07 Jul '15

07 Jul '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> clk_prepare_enable() may fail, so we should better check its return value and propagate it in the case of error. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Also handle the error for asrck_clk[i] as suggested by Nicolin sound/soc/fsl/fsl_asrc.c | 25 ++++++++++++++++++++----- 1 file changed, 20 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index c068494..9f087d4 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -931,14 +931,29 @@ static int fsl_asrc_probe(struct platform_device *pdev) static int fsl_asrc_runtime_resume(struct device *dev) { struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); - int i; + int i, ret; - clk_prepare_enable(asrc_priv->mem_clk); - clk_prepare_enable(asrc_priv->ipg_clk); - for (i = 0; i < ASRC_CLK_MAX_NUM; i++) - clk_prepare_enable(asrc_priv->asrck_clk[i]); + ret = clk_prepare_enable(asrc_priv->mem_clk); + if (ret) + return ret; + ret = clk_prepare_enable(asrc_priv->ipg_clk); + if (ret) + goto disable_mem_clk; + for (i = 0; i < ASRC_CLK_MAX_NUM; i++) { + ret = clk_prepare_enable(asrc_priv->asrck_clk[i]); + if (ret) + goto disable_asrck_clk; + } return 0; + +disable_asrck_clk: + for (i--; i >= 0; i--) + clk_disable_unprepare(asrc_priv->asrck_clk[i]); + clk_disable_unprepare(asrc_priv->ipg_clk); +disable_mem_clk: + clk_disable_unprepare(asrc_priv->mem_clk); + return ret; } static int fsl_asrc_runtime_suspend(struct device *dev) -- 1.9.1

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