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[alsa-devel] compress: adding support for hardware dependent params
by Banajit Goswami 03 Feb '14

03 Feb '14

Hi, We have a requirement, where we need to set a number of hardware specific parameters for compressed offload use-cases, before starting a stream. As these parameters need to be set once for the entire session, we would like to explore the best way to set these additional parameters somewhere before calling trigger() function. Below are few of the proposed solutions that we could think of at this point for getting this done- 1. Adding new IOCTL call: Add a new ioctl control in compress-offload.c file to handle hardware specific parameters setting. This new ioctl control will call into soc-compress.c and eventually call into platform driver ioctl to set those hardware specific parameters. In this case, we will add a new structure, let's say "snd_compr_hw_ctrl_params", with the same fields as "snd_compr_metadata" struct. 2. Adding a function hw_dependant_hw_params(), which will be called after hw_params() and before prepare() function. The new function will eventually call into the platform driver to get the hardware specific parameters set. Once implemented, this would look something like- hw_params(); hw_dependant_hw_params(); prepare(); 3. Add a function "hw_dependant_hw_params()", as _part_ of the hw_params() function. This new function will be called after the (existing) generic parameters are set. So once implemented, the proposed function calls would look like- hw_params() { generic_hw_params(); hw_dependant_hw_params(); } prepare(); We would like to get feedback on these proposed solutions and get a discussion going to see how best these requirements can be fitted in the current Compressed ALSA framework (for mainlining). Thanks and Regards, Banajit Goswami The Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation

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[alsa-devel] [PATCH v3 0/6] ASoC: tlv320aic32x4: DT support
by Markus Pargmann 03 Feb '14

03 Feb '14

Hi, The series adds basic DT support for the TI tlv320aic32x4 codec. Patch 3 ("ASoC: tlv320aic32x4: Use signed int mixer controls") depends on series "[PATCH 0/2] ASoC: core: volume control using signed register values". I added the remaining two patches of the "ASoC: tlv320aic32x4 bugfixes" series to the beginning of this series to have all tlv320aic32x4 patches in one series. Regards, Markus Changes in v3: - Rename property gpio-reset to reset-gpios - Use negative constants as minimum volume instead of casting to negative integers Changes in v2: - Regulator support - Fix documentation for gpio property Markus Pargmann (6): ASoC: tlv320aic32x4: Fix mono playback ASoC: tlv320aic32x4: Fix MICPGA input configuration ASoC: tlv320aic32x4: Use signed int mixer controls ASoC: tlv320aic32x4: DT support ASoC: tlv320aic32x4: Support for master clock ASoC: tlv320aic32x4: Support for regulators .../devicetree/bindings/sound/tlv320aic32x4.txt | 28 ++++ sound/soc/codecs/tlv320aic32x4.c | 154 ++++++++++++++++++--- sound/soc/codecs/tlv320aic32x4.h | 3 + 3 files changed, 164 insertions(+), 21 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt -- 1.8.5.2

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[alsa-devel] [PATCH 0/4] ASoC: davinci-mcasp: Fix master mode capture (w/o playback)
by Peter Ujfalusi 03 Feb '14

03 Feb '14

Hi, When McASP is master the TX clocks are used for capture clocking as well. Other parts of the code already does the configuration for both tx and rx - for different bits in xxFMT/xxFMCT registers we should do the same in other places. Along with this fix I have added three cleanup patch which came along when I was debugging the issue. Regards, Peter --- Peter Ujfalusi (4): ASoC: davinci-mcasp: Harmonize the sub hw_params function names ASoC: davinci-mcasp: Configure xxTDM, xxFMT and xxFMCT registers synchronously ASoC: davinci-mcasp: Return value handling cleanup for mcasp_common_hw_param() ASoC: davinci-mcasp: Code cleanup in davinci_mcasp_hw_params() sound/soc/davinci/davinci-mcasp.c | 81 +++++++++++++++++++-------------------- 1 file changed, 39 insertions(+), 42 deletions(-) -- 1.8.5.3

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Re: [alsa-devel] Got headphone out working via hda-verb setting on MacPro1, 1
by Takashi Iwai 03 Feb '14

03 Feb '14

At Fri, 31 Jan 2014 18:50:11 +0100, Nicolai Beuermann wrote: > > Am 22.01.2014 11:29, schrieb Takashi Iwai: > > At Tue, 21 Jan 2014 01:13:03 +0100, > > Nicolai Beuermann wrote: > >> > >> Hello, > >> hda-analyzer told me to write a note to this list when settings are > >> successful. > >> > >> My setup: > >> MacPro 1,1 > >> Kernel 3.10.17 (gentoo sources) > >> ALSA k3.10.17 > >> > >> Had to set > >> /usr/bin/hda-verb /dev/snd/hwC0D0 0x18 SET_PIN_WIDGET_CONTROL 0xc1 > >> to get the headphone out at the front working. > >> > >> Wrote a bit (in german) about it: > >> http://www.nico-beuermann.de/blogging/archives/117-Hier-gibt-es-was-auf-die… > > > > Could you check whether 3.13 still has the same problem? > > If yes, please give alsa-info.sh output. Better to run it with > > --no-upload and attach the output. > > 3.13 still has the same problem. > alsa-infos's output has been attached. Thanks. Could you try the patch below with 3.14-rc1? 3.13.1 still doesn't contain the fix for MBA 1,1 (which will be merged to stable kernel soon later). Alternatively, you can cherry-pick the commit e7729a415315fcd9516912050d85d5aaebcededc ALSA: hda - Fix silent output on MacBook Air 1,1 from Linus tree, and apply the patch below on its top. Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Fix missing VREF setup for Mac Pro 1,1 Mac Pro 1,1 with ALC889A codec needs the VREF setup on NID 0x18 to VREF50, in order to make the speaker working. The same fixup was already needed for MacBook Air 1,1, so we can reuse it. Reported-by: Nicolai Beuermann <mail(a)nico-beuermann.de> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cbeb67d98852..af7a0c372292 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1821,6 +1821,7 @@ enum { ALC889_FIXUP_IMAC91_VREF, ALC889_FIXUP_MBA11_VREF, ALC889_FIXUP_MBA21_VREF, + ALC889_FIXUP_MP11_VREF, ALC882_FIXUP_INV_DMIC, ALC882_FIXUP_NO_PRIMARY_HP, ALC887_FIXUP_ASUS_BASS, @@ -2190,6 +2191,12 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC889_FIXUP_MBP_VREF, }, + [ALC889_FIXUP_MP11_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc889_fixup_mba11_vref, + .chained = true, + .chain_id = ALC885_FIXUP_MACPRO_GPIO, + }, [ALC882_FIXUP_INV_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, @@ -2253,7 +2260,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF), - SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC885_FIXUP_MACPRO_GPIO), + SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC889_FIXUP_MP11_VREF), SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC889_FIXUP_MBP_VREF), -- 1.8.5.2

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[alsa-devel] mentoring for open sound hardware project?
by Hubert Denkmair 03 Feb '14

03 Feb '14

Hello list, I'm currently pushing a project for free audio hardware (basically a digital audio interface / mixer with attached linux system). The details are not final yet, but the design involves a FPGA for audio routing and a ARM SoC board for recording, playback etc. Plan is to support quite a few channels on the linux side, say 20 in, 20 out at 96kHz/24bit. Of course, this needs a Alsa driver. My question is: Is there a "best way" to do the FPGA<->Linux connection? Using the linux GPMC system for communication is the best thing we came up with until now. Are there already drivers doing something alike that we could build upon? Since there is no kernel developer in our team (yet), is someone on this list willing to mentor our driver development? Greetings, Hubert Denkmair

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[alsa-devel] [RFC][PATCH 0/8] A new driver for OXFW970/971 based devices
by Takashi Sakamoto 03 Feb '14

03 Feb '14

This series of patch is based on my previous series below: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-December/070424.h… This series of patch add a new driver, snd-oxfw, for OXFW970/971 based devices. These chipsets are already supported by snd-firewire-speakers but this new driver aim to support recording equipment. Current supported devices: * Behringer F-Control Audio 202 * Mackie Onyx-i series (former model) * Mackie Onyx Satellite Devices possible to be supported if identifying IDs: * Mackie, d.2 pro * Mackie, d.4 pro * Mackie, U.420 * Mackie, U.420d * Mackie, Tapco Link.Firewire I worked with Behringer F-Control Audio 202. So I want someone to test with the other devices, especially about some quirks. Clemens, would you gime me your opinions about issues below? And if you know something about this chipset, would you share the information? 1.snd-firewire-speakers and snd-oxfw With this series of patches, ALSA has two drivers for the same chipsets. Merging these two drivers is a bit difficult because: - snd-firewire-speakers has control interface but snd-oxfw don't - snd-firewire-speakers don't support in-stream but snd-oxfw supports 2.currently snd-oxfw works with streams in SYT-Match Due to invalid sequence of 'presentation timestamp' in transmitted packets. In detail, please see 4th patch. 3.assumption of channnel formation Some commands may not be implemented. In detail, please see 3rd patch. 4.don't check source of clock All of supported devices (including can be) don't have the functionality to switch source of clock. So I want to omit this. For issue 3 and 4, Descriptor mechanism in 'Enhancements to the AV/C General Specification 3.0 Version 1.1' may be a best solution but it costs much to implement against the number of available devices. I want to avoid this huge mechanism if possible. Regards Takashi Sakamoto Takashi Sakamoto (8): oxfw: Add skelton for OXFW970/971 based devices oxfw: Read firmware version to name card oxfw: Add some AV/C commands for channel formation of AMDTP stream oxfw: Add connections and streams management oxfw: Add proc interface for debugging purpose oxfw: Add MIDI interface oxfw: Add PCM interface oxfw: Add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 3 +- sound/firewire/Kconfig | 17 + sound/firewire/Makefile | 1 + sound/firewire/oxfw/Makefile | 3 + sound/firewire/oxfw/oxfw.c | 243 ++++++++++++++ sound/firewire/oxfw/oxfw.h | 140 ++++++++ sound/firewire/oxfw/oxfw_command.c | 161 ++++++++++ sound/firewire/oxfw/oxfw_hwdep.c | 197 ++++++++++++ sound/firewire/oxfw/oxfw_midi.c | 156 +++++++++ sound/firewire/oxfw/oxfw_pcm.c | 421 ++++++++++++++++++++++++ sound/firewire/oxfw/oxfw_proc.c | 61 ++++ sound/firewire/oxfw/oxfw_stream.c | 639 +++++++++++++++++++++++++++++++++++++ 13 files changed, 2043 insertions(+), 2 deletions(-) create mode 100644 sound/firewire/oxfw/Makefile create mode 100644 sound/firewire/oxfw/oxfw.c create mode 100644 sound/firewire/oxfw/oxfw.h create mode 100644 sound/firewire/oxfw/oxfw_command.c create mode 100644 sound/firewire/oxfw/oxfw_hwdep.c create mode 100644 sound/firewire/oxfw/oxfw_midi.c create mode 100644 sound/firewire/oxfw/oxfw_pcm.c create mode 100644 sound/firewire/oxfw/oxfw_proc.c create mode 100644 sound/firewire/oxfw/oxfw_stream.c -- 1.8.3.2

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[alsa-devel] Audio artefacts but no xruns
by Mark Hills 02 Feb '14

02 Feb '14

Hi, I have several applications for realtime processing of audio in/out, using small buffer sizes. I often find that I can get audible time-based distoritions, yet no xrun is reported, and the application is unaware. This problem is something I have, for a long time, taken for granted with ALSA. But it doesn't seem right and I would like to understand what's going on here. I am testing using different USB audio devices; snd-usb-audio and snd-usb-caiaq. Often using plughw or hw. Is this some fundamental limitation of USB, that it is unaware of problems nearer to the hardware? Or is this some bug where it's worth me narrowing down a specific test case and trying to fix it? Or some limitation of the use of intermediate processing like plughw? The effects range from occasional subtle clicks through ringing or noticable gaps in the audio, on record or playback. At very least I would like to guarantee a notification in userland that the integrity of the recording or playback is affected. This is particularly prevalent on, eg. Raspberry Pi. My other tests are on x86 platforms. Thanks -- Mark

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Re: [alsa-devel] [FFADO-devel] Call for testing: ALSA driver for Fireworks/BeBoB based devices
by Maximilian Engelhardt 02 Feb '14

02 Feb '14

On Monday 14 October 2013 12:17:31 Takashi Sakamoto wrote: > Hi all, > > This is a call for testing my ALSA driver for Fireworks/BeBoB based devices. > > Please test 'snd-fireworks' for Fireworks and 'snd-bebob' for BeBoB if > you have some devices listed in the end of this mail. > > Status: > - still under development > - Without snd-dice and Clemens' development (I must do this later) > > Functionality: > - playback/capturing (full duplex) with PCM/MIDI interface > - hardware metering for some devices with CONTROL interface > - switching clock source/digital interface/digital mode with CONTROL > interface > - print hardware status with PROC interface > > Note: > - Don't use simultaneously 'ALSA PCM/MIDI playback/capture' and 'jackd > with Firewire (FFADO) backend'. Both of them try connecting to the > device when another is running. > - I add much modification into snd-firewire-lib for full duplex > synchronization of receive/transmit AMDTP stream. > > Requirement: > - Linux kernel 3.11 or later because of Juju (nickname of Firewire > stack) changing its API. > - Dynamic Kernel Module Support (DKMS) is reccomended for safely > installing/uninstalling > (I work with Ubuntu 13.10) > > Bug report: > - report with /proc/asound/cardX/#XXX > - please send your experiences to me with the output > > How to install (DKMS): > 1. $ git clone https://github.com/takaswie/snd-firewire-improve.git > 2. $ ln -s $(pwd)/snd-firewire-improve/ /usr/src/alsa-firewire-3.11 > (superuser) > 3. $ dkms install snd-firewire/3.11 (superuser) > > How to uninstall (DKMS): > 1. $ modprobe -r snd-bebob snd-fireworks snd-firewire-lib (superuser) > 2. $ dkms remove ans-firewire/3.11 --all (superuser) > 3. $ rm /usr/src/alsa-firewire-3.11 (superuser) > 4. $ rm snd-firewire-improve > > How to install (Manual): > 1. $ git clone https://github.com/takaswie/snd-firewire-improve.git > 2. $ cd snd-firewire-improve > 3. $ make > 4, backup system snd-firewire-lib/snd-firewire-speakers/snd-isight > (superuser) > 5. install > snd-firewire-lib/snd-firewire-speakers/snd-isight/snd-fireworks/snd-bebob > (superuser) 6. depmod -a (superuser) > > How to uninstall (Manual) > 1. modprobe -r snd-firewire-lib snd-firewire-speakers snd-isight > snd-fireworks snd-bebob (superuser) > 2. remove > snd-firewire-lib/snd-firewire-speakers/snd-isight/snd-fireworks/snd-bebob > (superuser) 3. recover snd-firewire-lib/snd-firewire-speakers/snd-isight > (superuser) 4. depmod -a (superuser) > > Confirmed to work: > - AudioFire4 > - AudioFirePre8 > - Ozonic > - Firewire Solo > - Firewire Audiophile > - Firewire 410 > > == Fireworks based devices > [Echo Audio] > AudioFire2 > AudioFire4 > AudioFirePre8 > AudioFire8 (till 2009) > AudioFire8 (since 2009) > AudioFire12 > > [Gibson] > RIP > > [Mackie] > Onyx 400F > Onyx 1200F > > == BeBoB based devices > [Yamaha] > GO44 > GO46 > > [M-Audio] > (to control mixer channels please use FFADO upstream) > Ozonic > Firewire 410 > Firewire Audiophile > Firewire Solo > NRV10 > ProFireLightbridge > > [Focusrite] > SaffirePro 26 I/O > SaffirePro 10 I/O > Saffire(LE) > > [Edirol] > FA-66 > FA-101 > > [TerraTecElectronic GmbH] > Phase88FW > PhaseX24FW > > [PreSonus] > FireBox > FirePod > > [Mackie] > OnyxFirewire > > [Tascam] > IF-FW/DM > > [Behringer] > X32 > > [ApogeeElectronics] > Rosetta200 > > [ESI] > Quatafire610 > > > Regards > > Takashi Sakamoto > o-takashi(a)sakamocchi.jp > > ---------------------------------------------------------------------------- > -- October Webinars: Code for Performance > Free Intel webinars can help you accelerate application performance. > Explore tips for MPI, OpenMP, advanced profiling, and more. Get the most > from the latest Intel processors and coprocessors. See abstracts and > register > > http://pubads.g.doubleclick.net/gampad/clk?id=60134071&iu=/4140/ostg.clktrk > _______________________________________________ > FFADO-devel mailing list > FFADO-devel(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/ffado-devel Hi Takashi, Today I finally had some time to try your patches. I have a Terratec Phase 24FW Card. I did use the latest version (as of today) from https://github.com/takaswie/snd-firewire-improve.git on a Debian testing system. I did not use dkms but loaded the driver manually: $ cd sound/firewire $ modprobe -r snd-firewire-lib $ insmod ./snd-firewire-lib.ko $ insmod ./bebob/snd-bebob.ko First I have to say the driver does work and I can play music through the device, so thanks for your work. Unfortunately I managed to crash my system with it. I was using mplayer2 to play sound files and it seems that if I let mplayer2 play multiple files the system crashed somewhere between the end of the last file and the start of the new file (probably a race or something). I only did some quick testes because I hat to reboot after every crash (even sysrq doesn't work anymore), but I think the crash doesn't happen every time mplayer2 switches to the next file, but most of the time. If I let it play only one file I didn't observe any crashes. Here is the mplayer2 command I did use for testing: $ mplayer -ao alsa:device=hw=1 <some files> And here the output of /proc/asound/cardX/#XXX $ cat /proc/asound/card1/#clock Sampling rate: 48000 Clock Source: Internal $ cat /proc/asound/card1/#firmware Manufacturer: bridgeCo Protocol Ver: 1 Build Ver: 0 GUID: 0x000AAC04005F9C07 Model ID: 0x04 Model Rev: 1 Firmware Date: 20060420 Firmware Time: 105241 Firmware ID: 0x4 Firmware Ver: 18681119 Base Addr: 0x20080000 Max Size: 1572864 Loader Date: 20040719 Loader Time: 134002 $ cat /proc/asound/card1/#formation Output Stream from device: Rate PCM MIDI 22050 0 0 24000 0 0 32000 4 1 44100 4 1 48000 4 1 88200 4 1 96000 4 1 176400 0 0 192000 4 1 Input Stream to device: Rate PCM MIDI 22050 0 0 24000 0 0 32000 6 1 44100 6 1 48000 6 1 88200 6 1 96000 6 1 176400 0 0 192000 6 1 Attached is the crash log as far as I could obtain it. pulseaudio[4885]: [pulseaudio] module-alsa-card.c: Failed to find a working profile. pulseaudio[4885]: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="firewire-0x000aac04005f9c07" card_name="alsa_card.firewire-0x000aac04005f9c07" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes card_properties="module-udev-detect.discovered=1""): initialization failed. kernel: [ 334.172898] BUG: unable to handle kernel paging request at ffffffffffffffff kernel: [ 334.172934] IP: [<ffffffffa0154670>] fw_iso_context_queue_flush+0x10/0x10 [firewire_core] kernel: [ 334.172977] PGD 180f067 PUD 1811067 PMD 0 kernel: [ 334.172998] Oops: 0000 [#1] SMP kernel: [ 334.173015] Modules linked in: snd_bebob(O) snd_firewire_lib(O) sit ebtable_nat tunnel4 ip_tunnel ebtables cpufreq_stats cpufreq_userspace cpufreq_conservative parport_pc cpufreq_powersave ppdev lp ip6table_raw parport nf_conntrack_ipv6 nf_defrag_ipv6 ip6table_filter ip6_tables xt_REDIRECT iptable_nat nf_nat_ipv4 nf_nat xt_CT iptable_raw ipt_REJECT xt_LOG xt_limit xt_tcpudp nf_conntrack_ipv4 nf_defrag_ipv4 xt_conntrack nf_conntrack iptable_filter ip_tables x_tables binfmt_misc deflate ctr twofish_generic twofish_avx_x86_64 twofish_x86_64_3way twofish_x86_64 twofish_common camellia_generic camellia_x86_64 serpent_avx_x86_64 serpent_sse2_x86_64 serpent_generic xts blowfish_generic blowfish_x86_64 blowfish_common cast5_generic cast_common des_generic cbc cmac xcbc rmd160 sha512_generic sha256_generic crypto_null af_key xfrm_algo arc4 ecb md4 hmac nls_utf8 cifs rpcsec_gss_krb5 nfsv4 dns_resolver nfsd auth_rpcgss oid_registry nfs_acl nfs lockd fscache sunrpc w83627ehf hwmon_vid loop fuse ecryptfs isl6423 stv6110x stv090x snd_hda_codec_hdmi snd_hda_codec_realtek iTCO_wdt iTCO_vendor_support stv0299 rc_hauppauge mxm_wmi x86_pkg_temp_thermal intel_powerclamp coretemp kvm_intel kvm crc32_pclmul crc32c_intel ghash_clmulni_intel aesni_intel aes_x86_64 ablk_helper cryptd lrw gf128mul glue_helper psmouse snd_hda_intel snd_hda_codec snd_hwdep snd_pcm_oss snd_mixer_oss snd_pcm snd_page_alloc snd_seq_midi snd_seq_midi_event pcspkr snd_rawmidi budget budget_ci serio_raw budget_core ttpci_eeprom snd_seq i2c_i801 saa7146 snd_seq_device snd_timer rc_core dvb_core lpc_ich mfd_core snd soundcore evdev i915 drm_kms_helper wmi drm i2c_algo_bit i2c_core video button processor mei_me mei ext4 crc16 mbcache jbd2 btrfs xor raid6_pq crc32c libcrc32c dm_mod raid1 md_mod sg sd_mod sr_mod cdrom crc_t10dif crct10dif_common hid_generic usbhid hid ahci firewire_ohci libahci firewire_core crc_itu_t libata scsi_mod ehci_pci ehci_hcd e1000e xhci_hcd ptp pps_core usbcore usb_common thermal fan thermal_sys kernel: [ 334.173881] CPU: 0 PID: 5343 Comm: mplayer Tainted: G O 3.12-1-amd64 #1 Debian 3.12.6-2 kernel: [ 334.173913] Hardware name: /DZ77BH-55K, BIOS BHZ7710H.86A.0100.2013.0517.0942 05/17/2013 kernel: [ 334.173947] task: ffff88040da7a0c0 ti: ffff8803da2e4000 task.ti: ffff8803da2e4000 kernel: [ 334.173973] RIP: 0010:[<ffffffffa0154670>] [<ffffffffa0154670>] fw_iso_context_queue_flush+0x10/0x10 [firewire_core] kernel: [ 334.174015] RSP: 0018:ffff8803da2e5d70 EFLAGS: 00010002 kernel: [ 334.174035] RAX: ffffffffa0acd040 RBX: ffff8800c99c95c8 RCX: 0000000000001c2c kernel: [ 334.174060] RDX: ffff88040dcff000 RSI: 0000000000000000 RDI: ffffffffffffffff kernel: [ 334.174086] RBP: ffff8803da2e5de8 R08: 0000000000000000 R09: 00000000000031a1 kernel: [ 334.174111] R10: 0000000052ed7c8e R11: 0000000000000246 R12: 00007fffa996f808 kernel: [ 334.174137] R13: 00007fffa996f808 R14: 0000000000000000 R15: ffff8803f70a8c00 kernel: [ 334.174162] FS: 00007f286ae957c0(0000) GS:ffff88041f200000(0000) knlGS:0000000000000000 kernel: [ 334.174190] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 kernel: [ 334.174211] CR2: ffffffffffffffff CR3: 000000040c1d3000 CR4: 00000000001407f0 kernel: [ 334.174236] Stack: kernel: [ 334.174245] ffffffffa0abbca9 ffff8803da3a7000 ffffffffa0569bc4 0000000000000000 kernel: [ 334.174277] 0000000c00000000 0000000081019435 0000000000000097 ffffffffa03e321b kernel: [ 334.174309] ffff8803edca3558 ffff8803edca3530 ffff8803f70a8c00 00007fffa996f808 kernel: [ 334.174340] Call Trace: kernel: [ 334.174352] [<ffffffffa0abbca9>] ? amdtp_stream_pcm_pointer+0x29/0x40 [snd_firewire_lib] kernel: [ 334.174388] [<ffffffffa0569bc4>] ? snd_pcm_update_hw_ptr0+0x34/0x3a0 [snd_pcm] kernel: [ 334.174419] [<ffffffffa03e321b>] ? snd_timer_notify+0x6b/0x150 [snd_timer] kernel: [ 334.174448] [<ffffffffa056363a>] ? snd_pcm_delay+0x5a/0x140 [snd_pcm] kernel: [ 334.174476] [<ffffffffa0565623>] ? snd_pcm_common_ioctl1+0x5f3/0xc90 [snd_pcm] kernel: [ 334.174506] [<ffffffffa056a5f3>] ? snd_pcm_lib_write1+0x3a3/0x3f0 [snd_pcm] kernel: [ 334.174536] [<ffffffffa0567b80>] ? pcm_chmap_ctl_private_free+0x30/0x30 [snd_pcm] kernel: [ 334.174567] [<ffffffffa0565db0>] ? snd_pcm_playback_ioctl1+0xf0/0x240 [snd_pcm] kernel: [ 334.174597] [<ffffffffa0565f26>] ? snd_pcm_playback_ioctl+0x26/0x30 [snd_pcm] kernel: [ 334.174625] [<ffffffff81184d94>] ? do_vfs_ioctl+0x2d4/0x4b0 kernel: [ 334.174648] [<ffffffff81184ff0>] ? SyS_ioctl+0x80/0xa0 kernel: [ 334.174669] [<ffffffff81499b39>] ? system_call_fastpath+0x16/0x1b kernel: [ 334.174691] Code: 00 48 8b 40 60 ff e0 0f 1f 40 00 48 8b 07 48 8b 00 48 8b 40 70 ff e0 0f 1f 40 00 48 8b 07 48 8b 00 48 8b 40 78 ff e0 0f 1f 40 00 <48> 8b 07 48 8b 00 48 8b 80 80 00 00 00 ff e0 90 48 8b 07 48 8b kernel: [ 334.174857] RIP [<ffffffffa0154670>] fw_iso_context_queue_flush+0x10/0x10 [firewire_core] kernel: [ 334.174893] RSP <ffff8803da2e5d70> kernel: [ 334.174906] CR2: ffffffffffffffff kernel: [ 334.184706] ---[ end trace 4fd37a42001c8a5d ]--- kernel: [ 355.198207] INFO: rcu_sched detected stalls on CPUs/tasks: { 0} (detected by 2, t=5252 jiffies, g=3743, c=3742, q=2720) kernel: [ 355.198212] sending NMI to all CPUs: kernel: [ 355.198213] NMI backtrace for cpu 2 kernel: [ 355.198215] CPU: 2 PID: 3929 Comm: collectd Tainted: G D O 3.12-1-amd64 #1 Debian 3.12.6-2 kernel: [ 355.198216] Hardware name: /DZ77BH-55K, BIOS BHZ7710H.86A.0100.2013.0517.0942 05/17/2013 kernel: [ 355.198226] task: ffff88040bcad0c0 ti: ffff88040bf76000 task.ti: ffff88040bf76000 kernel: [ 355.198227] RIP: 0010:[<ffffffff81275bdd>] [<ffffffff81275bdd>] find_next_bit+0x4d/0xd0 kernel: [ 355.198231] RSP: 0018:ffff88041f303df0 EFLAGS: 00000046 kernel: [ 355.198231] RAX: 0000000000000000 RBX: ffff88041f30de00 RCX: 0000000000000001 kernel: [ 355.198232] RDX: ffff88041f30de38 RSI: 0000000000000080 RDI: 0000000000000180 kernel: [ 355.198233] RBP: ffff88041f20de40 R08: ffff88041f30de08 R09: 0000000000000008 kernel: [ 355.198234] R10: 0000000000000000 R11: ffff88041f303bb6 R12: 000000000000de80 kernel: [ 355.198234] R13: 0000000000080000 R14: 0000000000000055 R15: 0000000000000002 kernel: [ 355.198235] FS: 00007f8149847700(0000) GS:ffff88041f300000(0000) knlGS:0000000000000000 kernel: [ 355.198236] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 kernel: [ 355.198237] CR2: 00007f8157da9000 CR3: 000000040d165000 CR4: 00000000001407e0 kernel: [ 355.198238] Stack: kernel: [ 355.198239] ffffffff81043637 0000000000000008 000000000000de40 0000000200000002 kernel: [ 355.198241] 0000000000000082 0000000000002710 ffffffff818419c0 ffffffff818419c0 kernel: [ 355.198242] ffff88040bf76000 0000000000000aa0 ffff88041f30ed80 ffffffff8103f88e kernel: [ 355.198243] Call Trace: kernel: [ 355.198244] <IRQ> kernel: [ 355.198245] [<ffffffff81043637>] ? __x2apic_send_IPI_mask+0xb7/0x190 kernel: [ 355.198251] [<ffffffff8103f88e>] ? arch_trigger_all_cpu_backtrace+0x4e/0x80 kernel: [ 355.198253] [<ffffffff810e1b48>] ? rcu_check_callbacks+0x5b8/0x600 kernel: [ 355.198256] [<ffffffff810b4c30>] ? tick_sched_handle.isra.15+0x60/0x60 kernel: [ 355.198258] [<ffffffff8106881b>] ? update_process_times+0x3b/0x70 kernel: [ 355.198259] [<ffffffff810b4beb>] ? tick_sched_handle.isra.15+0x1b/0x60 kernel: [ 355.198260] [<ffffffff810b4c67>] ? tick_sched_timer+0x37/0x60 kernel: [ 355.198263] [<ffffffff8107ea22>] ? __run_hrtimer+0x62/0x1b0 kernel: [ 355.198264] [<ffffffff8107f1c9>] ? hrtimer_interrupt+0xe9/0x220 kernel: [ 355.198266] [<ffffffff8103deb6>] ? smp_apic_timer_interrupt+0x36/0x50 kernel: [ 355.198269] [<ffffffff8149a71d>] ? apic_timer_interrupt+0x6d/0x80 kernel: [ 355.198269] <EOI> kernel: [ 355.198270] [<ffffffff810ba548>] ? generic_exec_single+0x68/0x90 kernel: [ 355.198273] [<ffffffff810ba564>] ? generic_exec_single+0x84/0x90 kernel: [ 355.198275] [<ffffffff81287cb0>] ? ucs2_strncmp+0x60/0x60 kernel: [ 355.198277] [<ffffffff810ba655>] ? smp_call_function_single+0xe5/0x190 kernel: [ 355.198278] [<ffffffff81287cb0>] ? ucs2_strncmp+0x60/0x60 kernel: [ 355.198280] [<ffffffff81287dd6>] ? rdmsr_on_cpu+0x46/0x60 kernel: [ 355.198295] [<ffffffffa06561f3>] ? show_temp+0x93/0xd0 [coretemp] kernel: [ 355.198298] [<ffffffff81342157>] ? dev_attr_show+0x17/0x50 kernel: [ 355.198300] [<ffffffff811decc9>] ? sysfs_read_file+0x99/0x170 kernel: [ 355.198302] [<ffffffff81172f94>] ? vfs_read+0x94/0x160 kernel: [ 355.198304] [<ffffffff81173a83>] ? SyS_read+0x43/0xa0 kernel: [ 355.198306] [<ffffffff81499b39>] ? system_call_fastpath+0x16/0x1b kernel: [ 355.198306] Code: 3f 75 6b 48 f7 c6 c0 ff ff ff 0f 84 93 00 00 00 49 8b 00 49 8d 50 08 48 85 c0 74 0f eb 3f 48 83 c2 08 48 8b 42 f8 48 85 c0 75 32 <48> 83 ee 40 48 83 c7 40 48 f7 c6 c0 ff ff ff 75 e2 48 85 f6 48 kernel: [ 355.198321] NMI backtrace for cpu 0 kernel: [ 355.198324] CPU: 0 PID: 5343 Comm: mplayer Tainted: G D O 3.12-1-amd64 #1 Debian 3.12.6-2 kernel: [ 355.198324] Hardware name: /DZ77BH-55K, BIOS BHZ7710H.86A.0100.2013.0517.0942 05/17/2013 kernel: [ 355.198325] task: ffff88040da7a0c0 ti: ffff8803da2e4000 task.ti: ffff8803da2e4000 kernel: [ 355.198326] RIP: 0010:[<ffffffff81492083>] [<ffffffff81492083>] _raw_spin_lock+0x23/0x30 kernel: [ 355.198329] RSP: 0018:ffff8803da2e5a30 EFLAGS: 00000097 kernel: [ 355.198330] RAX: 0000000000001c2c RBX: ffff8803f70a8c00 RCX: 0000000000001c2d kernel: [ 355.198331] RDX: 0000000000001c2d RSI: ffff88040b1b5580 RDI: ffff8803f70a8d70 kernel: [ 355.198331] RBP: ffff8803da3a7000 R08: 0000000000000000 R09: 0000000000000000 kernel: [ 355.198332] R10: ffff88040b1b5590 R11: 0000000000000117 R12: ffff88040b1b5580 kernel: [ 355.198333] R13: ffff88040dca7e40 R14: ffff88040b029d98 R15: ffff8803d9156330 kernel: [ 355.198334] FS: 0000000000000000(0000) GS:ffff88041f200000(0000) knlGS:0000000000000000 kernel: [ 355.198335] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 kernel: [ 355.198335] CR2: ffffffffffffffff CR3: 000000000180c000 CR4: 00000000001407f0 kernel: [ 355.198336] Stack: kernel: [ 355.198337] ffffffffa0563cb8 ffff8803f70a8c00 ffff88040b029c00 ffffffffa0563d2a kernel: [ 355.198338] ffff8803f70a8c00 ffff88040b029c00 ffffffffa0563e68 ffff88040b1b5580 kernel: [ 355.198340] 0000000000000008 ffff8803d9156330 ffff8803d3d7e780 ffff88040e6656e0 kernel: [ 355.198341] Call Trace: kernel: [ 355.198351] [<ffffffffa0563cb8>] ? snd_pcm_drop+0x48/0xb0 [snd_pcm] kernel: [ 355.198355] [<ffffffffa0563d2a>] ? snd_pcm_release_substream.part.29+0xa/0x90 [snd_pcm] kernel: [ 355.198359] [<ffffffffa0563e68>] ? snd_pcm_release+0x98/0xb0 [snd_pcm] kernel: [ 355.198361] [<ffffffff81174a9a>] ? __fput+0xca/0x220 kernel: [ 355.198363] [<ffffffff81078904>] ? task_work_run+0xb4/0xd0 kernel: [ 355.198365] [<ffffffff8105d8c3>] ? do_exit+0x2b3/0xa40 kernel: [ 355.198367] [<ffffffff814932b8>] ? oops_end+0x98/0xe0 kernel: [ 355.198369] [<ffffffff81489540>] ? no_context+0x26a/0x276 kernel: [ 355.198371] [<ffffffff81495a0e>] ? __do_page_fault+0x2fe/0x500 kernel: [ 355.198373] [<ffffffff8111abea>] ? __alloc_pages_nodemask+0x14a/0xa30 kernel: [ 355.198376] [<ffffffff81127fcb>] ? shmem_find_get_pages_and_swap.part.18+0xcb/0x120 [some part missing...] Greetings, Maxi

1 0

[alsa-devel] unsubscribe
by animan9＠gmail.com 01 Feb '14

01 Feb '14

unsubscribe Sent from Windows Mail

1 0

[alsa-devel] [PATCH 1/4] ASoC: wm8962: Hold a runtime PM reference while handling interrupts
by Mark Brown 31 Jan '14

31 Jan '14

From: Mark Brown <broonie(a)linaro.org> If the device is runtime suspended then we can't interact with it as it may have been powered off and the register map will be in cache only mode. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8962.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 97db3b45b411..aa7ae76d5e28 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3003,6 +3003,12 @@ static irqreturn_t wm8962_irq(int irq, void *data) unsigned int active; int reg, ret; + ret = pm_runtime_get_sync(dev); + if (ret < 0) { + dev_err(dev, "Failed to resume: %d\n", ret); + return ret; + } + ret = regmap_read(wm8962->regmap, WM8962_INTERRUPT_STATUS_2_MASK, &mask); if (ret != 0) { @@ -3019,8 +3025,10 @@ static irqreturn_t wm8962_irq(int irq, void *data) active &= ~mask; - if (!active) + if (!active) { + pm_runtime_put(dev); return IRQ_NONE; + } /* Acknowledge the interrupts */ ret = regmap_write(wm8962->regmap, WM8962_INTERRUPT_STATUS_2, active); @@ -3070,6 +3078,8 @@ static irqreturn_t wm8962_irq(int irq, void *data) msecs_to_jiffies(250)); } + pm_runtime_put(dev); + return IRQ_HANDLED; } -- 1.9.rc1

3 6

[alsa-devel] [PATCH 0/3] ASoC: davinci-mcasp/evm: PM related changes
by Peter Ujfalusi 31 Jan '14

31 Jan '14

Hi, In order to be able to benefit from the core's feature to handle the dai pinctrl default/sleep states the davinci-evm should set it's .pm section snd_soc_pm_ops. In the McASP driver consolidate the pm_runtime use to avoid unbalanced refcounting: trigger:start increased while trigger:stop did not decreased the pm counter. Move the pm callbacks from platfrom device to the soc_dai_driver. Now it is only a matter of dts update to enable the pin level power saving for McASP. Regards, Peter --- Peter Ujfalusi (3): ASoC: davinci-mcasp: Consolidate pm_runtime_get/put() use in the driver ASoC: davinci-evm: Add pm callbacks to platform driver ASoC: davinci-mcasp: Move pm callbacks from platform device to soc_dai_driver sound/soc/davinci/davinci-evm.c | 1 + sound/soc/davinci/davinci-mcasp.c | 97 +++++++++++++++++++-------------------- 2 files changed, 47 insertions(+), 51 deletions(-) -- 1.8.5.3

2 6

[alsa-devel] [PATCH] ALSA: hda/hdmi - allow PIN_OUT to be dynamically enabled
by Stephen Warren 31 Jan '14

31 Jan '14

From: Stephen Warren <swarren(a)nvidia.com> Commit 384a48d71520 "ALSA: hda: HDMI: Support codecs with fewer cvts than pins" dynamically enabled each pin widget's PIN_OUT only when the pin was actively in use. This was required on certain NVIDIA CODECs for correct operation. Specifically, if multiple pin widgets each had their mux input select the same audio converter widget and each pin widget had PIN_OUT enabled, then only one of the pin widgets would actually receive the audio, and often not the one the user wanted! However, this apparently broke some Intel systems, and commit 6169b673618b "ALSA: hda - Always turn on pins for HDMI/DP" reverted the dynamic setting of PIN_OUT. This in turn broke the afore-mentioned NVIDIA CODECs. This change supports either dynamic or static handling of PIN_OUT, selected by a flag set up during CODEC initialization. This flag is enabled for all recent NVIDIA GPUs. Reported-by: Uosis <uosisl(a)gmail.com> Cc: <stable(a)vger.kernel.org> # v3.13 Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v3.12 or earlier stable require some manual back-porting effort, or perhaps just a variety of other commits to be backported. I'm investigating exactly which commits, and whether they're appropriate for stable. --- sound/pci/hda/patch_hdmi.c | 40 ++++++++++++++++++++++++++++++++++++---- 1 file changed, 36 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 64f0a5e73a25..5ef95034d041 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -132,6 +132,9 @@ struct hdmi_spec { struct hdmi_eld temp_eld; struct hdmi_ops ops; + + bool dyn_pin_out; + /* * Non-generic VIA/NVIDIA specific */ @@ -500,15 +503,25 @@ static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t pin_nid, static void hdmi_init_pin(struct hda_codec *codec, hda_nid_t pin_nid) { + struct hdmi_spec *spec = codec->spec; + int pin_out; + /* Unmute */ if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); - /* Enable pin out: some machines with GM965 gets broken output when - * the pin is disabled or changed while using with HDMI - */ + + if (spec->dyn_pin_out) + /* Disable pin out until stream is active */ + pin_out = 0; + else + /* Enable pin out: some machines with GM965 gets broken output + * when the pin is disabled or changed while using with HDMI + */ + pin_out = PIN_OUT; + snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + AC_VERB_SET_PIN_WIDGET_CONTROL, pin_out); } static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t cvt_nid) @@ -1735,6 +1748,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); hda_nid_t pin_nid = per_pin->pin_nid; bool non_pcm; + int pinctl; non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); mutex_lock(&per_pin->lock); @@ -1744,6 +1758,14 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hdmi_setup_audio_infoframe(codec, per_pin, non_pcm); mutex_unlock(&per_pin->lock); + if (spec->dyn_pin_out) { + pinctl = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl | PIN_OUT); + } + return spec->ops.setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } @@ -1763,6 +1785,7 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, int cvt_idx, pin_idx; struct hdmi_spec_per_cvt *per_cvt; struct hdmi_spec_per_pin *per_pin; + int pinctl; if (hinfo->nid) { cvt_idx = cvt_nid_to_cvt_index(spec, hinfo->nid); @@ -1779,6 +1802,14 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, return -EINVAL; per_pin = get_pin(spec, pin_idx); + if (spec->dyn_pin_out) { + pinctl = snd_hda_codec_read(codec, per_pin->pin_nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_codec_write(codec, per_pin->pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl & ~PIN_OUT); + } + snd_hda_spdif_ctls_unassign(codec, pin_idx); mutex_lock(&per_pin->lock); @@ -2840,6 +2871,7 @@ static int patch_nvhdmi(struct hda_codec *codec) return err; spec = codec->spec; + spec->dyn_pin_out = true; spec->ops.chmap_cea_alloc_validate_get_type = nvhdmi_chmap_cea_alloc_validate_get_type; -- 1.8.1.5

2 1

[alsa-devel] [PATCH v2 0/5] add a TDA998x CODEC
by Jean-Francois Moine 30 Jan '14

30 Jan '14

The TDA998x HDMI transmitter accepts audio input from either I2S or S/PDIF. Theses inputs have different intrinsic constraints and these constraints may be modified by the audio parameters of the connected video device. The choice of I2S or S/PDIF may be the done by the user or by automatic processing (DPCM?) at each audio starting time. This asks for a dynamic audio input switch in the HDMI driver. This patch series implements the TDA998x specific CODEC. A simple function call mechanism is used for exchanges between the CODEC and the HDMI driver. Note: the changes in the TDA998x I2C driver are based on my previous patch series: http://lists.freedesktop.org/archives/dri-devel/2014-January/052837.html - v2 - add ACLK setting and code optimization in patch 1 - from Mark Brown's remarks in patch 2: - don't compile the codec when CONFIG_ALL_CODECS - simplify the code about start/stop audio - fix coding style errors - add audio-port-names associated to audio-ports - add audio-port-names in patch 4 - add patch 5 'adjust the audio CTS_N pre-divider from audio format' for the Beaglebone-Black board (Jyri Sarha) Jean-Francois Moine (5): drm/i2c: tda998x: add a function for dynamic audio input switch ASoC: tda998x: add a codec driver for TDA998x ASoC: tda998x: add DT documentation ASoC: tda998x: adjust the audio hw parameters from EDID ASoC: tda998x: adjust the audio CTS_N pre-divider from audio format .../devicetree/bindings/sound/tda998x.txt | 16 + drivers/gpu/drm/i2c/tda998x_drv.c | 82 +++++- include/drm/i2c/tda998x.h | 11 +- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 325 +++++++++++++++++++++ 6 files changed, 436 insertions(+), 6 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tda998x.txt create mode 100644 sound/soc/codecs/tda998x.c -- 1.9.rc1

2 6

[alsa-devel] [PATCH RFC v3 0/8] Beaglebone-Black HDMI audio
by Jyri Sarha 30 Jan '14

30 Jan '14

Changes since RFC v2 version of the patches: - Dropped out already applied: ASoC: hdmi-codec: Add devicetree binding with documentation - Addresses Mark's comments here: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-December/070605.h… - ASoC: davinci-evm: Add named clock reference to DT bindings - Get rid of unnecessary castings - Add mclk NULL checks - Use devm_clk_get() - Change clock name from "ti,codec-clock" to "mclk" - Address Mark's comments here: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-December/070606.h… - ASoC: davinci-evm: HDMI audio support for TDA998x trough McASP I2S bus - Get rid of unnecessary castings - Update commit message - Add: ASoC: davinci-mcasp: Set BCLK divider if McASP is BCLK master - Use snd_soc_params_to_bclk() Changes since the first RFC version of the patches: - Drop out already applied: ASoC: hdmi-codec: Add SNDRV_PCM_FMTBIT_32_LE playback format - Change sound node's compatible property form: "ti,am33xx-beaglebone-black" to "ti,am33xx-beaglebone-black-audio" - Some minor style issue fixes from TI internal review Jyri Sarha (8): clk: add gpio controlled clock ASoC: davinci-evm: Add named clock reference to DT bindings ASoC: davinci-mcasp: Set BCLK divider if McASP is BCLK master ASoC: davinci-evm: HDMI audio support for TDA998x trough McASP I2S bus ASoC: davinci: HDMI audio build for AM33XX and TDA998x drm/tilcdc: Add I2C HDMI audio config for tda998x ARM: OMAP2+: omap2plus_defconfig: Enable tilcdc and TDA998X HDMI support ARM: OMAP2+: omap2plus_defconfig: Enable BeagleBone Black HDMI audio support .../devicetree/bindings/clock/gpio-clock.txt | 21 ++ .../bindings/sound/davinci-evm-audio.txt | 13 +- arch/arm/configs/omap2plus_defconfig | 5 + drivers/clk/Makefile | 1 + drivers/clk/clk-gpio.c | 210 +++++++++++++++++++ drivers/gpu/drm/tilcdc/tilcdc_slave.c | 24 ++- include/linux/clk-provider.h | 25 +++ sound/soc/davinci/Kconfig | 12 ++ sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 211 +++++++++++++++++++- sound/soc/davinci/davinci-mcasp.c | 20 ++ 11 files changed, 534 insertions(+), 9 deletions(-) create mode 100644 Documentation/devicetree/bindings/clock/gpio-clock.txt create mode 100644 drivers/clk/clk-gpio.c -- 1.7.9.5

4 22

[alsa-devel] [PATCH 00/11] HD-audio patches for 3.15
by Takashi Iwai 30 Jan '14

30 Jan '14

Hi, here is a series of patches for HD-audio for 3.15, i.e. no urgent fixes but mainly more enhancements and cleanups of static quirks. The patches are found in sound-unstable git tree topic/hda branch for now. They'll be merged to the main sound git tree once when the merge window is closed. Takashi

1 12

[alsa-devel] [PATCH v4 01/19] ALSA: Oxygen: Add the separate SPI waiting function
by Roman Volkov 30 Jan '14

30 Jan '14

The oxygen_wait_spi() function now performs waiting when the SPI bus completes a transaction. Introduce the timeout error checking and increase timeout to 200 from 40. Signed-off-by: Roman Volkov <v1ron(a)mail.ru> --- sound/pci/oxygen/oxygen_io.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 521eae4..471e64a 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -194,6 +194,21 @@ void oxygen_write_ac97_masked(struct oxygen *chip, unsigned int codec, } EXPORT_SYMBOL(oxygen_write_ac97_masked); +static int oxygen_wait_spi(struct oxygen *chip) +{ + /* Higher timeout to be sure: 200 us, + Actual transaction should not need more than 40 us */ + unsigned int count; + for (count = 50; count > 0; count--) { + udelay(4); + if ((oxygen_read8(chip, OXYGEN_SPI_CONTROL) & + OXYGEN_SPI_BUSY) == 0) + return 0; + } + snd_printk(KERN_ERR "Oxygen: SPI wait timeout\n"); + return -EIO; +} + void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data) { unsigned int count; -- 1.7.10.4

4 22

[alsa-devel] [PATCH 1/2] ASoC: Samsung: Remove invalid dependencies
by Sachin Kamat 30 Jan '14

30 Jan '14

These symbols got eliminated when non-DT support for Exynos was removed. Remove them. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/samsung/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 1f369c69469c..c006d4aafd5a 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -145,7 +145,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310 || MACH_SMDKC210) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ALSA: hda - add headset mic detect quirks for another Dell laptop
by Hui Wang 30 Jan '14

30 Jan '14

When we plug a 3-ring headset on the Dell machine (Vendor ID: 0x10ec0255, Subsystem ID: 0x1028064d), the headset mic can't be detected, after apply this patch, the headset mic can work well. BugLink: https://bugs.launchpad.net/bugs/1260303 Cc: David Henningsson <david.henningsson(a)canonical.com> Tested-by: Doro Wu <fan-cheng.wu(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d723c1f..56a8f18 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4294,6 +4294,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0640, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x064d, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0651, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0652, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0653, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.8.1.2

2 1

[alsa-devel] [PATCH 1/2] ASoC: fsl: imx-sgtl5000: No need to initialize local variable data
by Philipp Zabel 30 Jan '14

30 Jan '14

The first access to this variable is an assignment to the result of a kzalloc function call. Remove the superfluous initialisation. Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> --- sound/soc/fsl/imx-sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index f2beae7..3e60b4c 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -62,7 +62,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) struct device_node *ssi_np, *codec_np; struct platform_device *ssi_pdev; struct i2c_client *codec_dev; - struct imx_sgtl5000_data *data = NULL; + struct imx_sgtl5000_data *data; int int_port, ext_port; int ret; -- 1.8.5.3

3 4

[alsa-devel] Regression due to 6169b673618b "ALSA: hda - Always turn on pins for HDMI/DP"
by Stephen Warren 30 Jan '14

30 Jan '14

Takashi, It looks like commit 6169b673618b "ALSA: hda - Always turn on pins for HDMI/DP" has caused a regression on NVIDIA graphics cards. It's been reported to us by at least two separate users. On NVIDIA GPUs, it appears that if multiple pin complex widgets have their input mux set to the same audio output widget and PIN_OUT, then only one of the pin complex widgets will actually receive the audio. That's the reason we need to dynamically toggle PIN_OUT on and off, so that only actually active pin complex widgets are enabled. Without this, audio will be routed to the same pin complex all the time, rather than the user-selected device. I'm not sure if this is the way HDA CODECs are meant to work, but it certainly is how they do work:-) So, I'd like to basically revert 6169b673618b "ALSA: hda - Always turn on pins for HDMI/DP", and return the code to the state it was after commit 384a48d71520 "ALSA: hda: HDMI: Support codecs with fewer cvts than pins". However, I obviously don't want to re-introduce the regression that your commit solved. I think we should solve this by including the code for both behaviours in the kernel, and having either a flag in struct hdmi_spec, or some extra ops in struct hdmi_ops, select between the two paths. Does that seem reasonable? If so, the only remaining question is: which behaviour should be default a/k/a should I blacklist the problematic GM965 (mentioned in https://bugzilla.kernel.org/show_bug.cgi?id=51421) against having the dynamic PIN_OUT behaviour, or whitelist NVIDIA GPUs as requiring the dynamic PIN_OUT behaviour? I know that my patch worked fine on my wife's Intel chipset, but I have no idea if the dynamic PIN_OUT behaviour was actually needed there, or simply tolerated, and I have no way to test much aside from NVIDIA GPUs. That all said, the dynamic PIN_OUT behaviour seems to make most sense to me, or some systems may end up with audio being sent to multiple outputs when only one was expected.

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: imx-audmux: add RXFS/RXCLK defines for 6-wire connections
by Philipp Zabel 30 Jan '14

30 Jan '14

In asynchronous mode, a RxFS and RxClk connection needs to be made between two ports. Add a define for the bit to be set in the *SEL fields. Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> --- include/dt-bindings/sound/fsl-imx-audmux.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/include/dt-bindings/sound/fsl-imx-audmux.h b/include/dt-bindings/sound/fsl-imx-audmux.h index 50b09e9..63b48f8 100644 --- a/include/dt-bindings/sound/fsl-imx-audmux.h +++ b/include/dt-bindings/sound/fsl-imx-audmux.h @@ -24,6 +24,10 @@ #define MX51_AUDMUX_PORT6 5 #define MX51_AUDMUX_PORT7 6 +/* TFSEL/TCSEL/RFSEL/RCSEL can be sourced from Rx/Tx clocks on i.MX51/53/6Q */ +#define IMX_AUDMUX_RXFS 0x8 +#define IMX_AUDMUX_RXCLK 0x8 + /* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ #define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) #define IMX_AUDMUX_V1_PCR_INMEN (1 << 8) -- 1.8.5.3

2 2

[alsa-devel] [PATCH 1/2] ASoC: imx-audmux: add RXFS/RXCLK defines for 6-wire connections
by Philipp Zabel 30 Jan '14

30 Jan '14

In asynchronous mode, a RxFS and RxClk connection needs to be made between two ports. Add a define for the bit to be set in the *SEL fields. Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> --- include/dt-bindings/sound/fsl-imx-audmux.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/include/dt-bindings/sound/fsl-imx-audmux.h b/include/dt-bindings/sound/fsl-imx-audmux.h index 50b09e9..63b48f8 100644 --- a/include/dt-bindings/sound/fsl-imx-audmux.h +++ b/include/dt-bindings/sound/fsl-imx-audmux.h @@ -24,6 +24,10 @@ #define MX51_AUDMUX_PORT6 5 #define MX51_AUDMUX_PORT7 6 +/* TFSEL/TCSEL/RFSEL/RCSEL can be sourced from Rx/Tx clocks on i.MX51/53/6Q */ +#define IMX_AUDMUX_RXFS 0x8 +#define IMX_AUDMUX_RXCLK 0x8 + /* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ #define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) #define IMX_AUDMUX_V1_PCR_INMEN (1 << 8) -- 1.8.5.3

2 3

[alsa-devel] [PATCH 0/2] ASoC: core: volume control using signed register values
by Markus Pargmann 29 Jan '14

29 Jan '14

Hi, There are a lot of codecs using negative and positive volume control ranges as values in their registers. These two patches add support for arbitrary size signed integer values in registers. Regards, Markus Markus Pargmann (2): ASoC: core: Add signed register volume control logic ASoC: soc.h: Add SOC_DOUBLE_R_S_TLV include/sound/soc.h | 15 ++++++++++++ sound/soc/soc-core.c | 68 ++++++++++++++++++++++++++++++++++++++++++++++------ 2 files changed, 76 insertions(+), 7 deletions(-) -- 1.8.5.2

2 3

[alsa-devel] [RFC PATCH] ALSA: hda - Do not accept responses from non-existing codecs
by David Henningsson 29 Jan '14

29 Jan '14

While looking into some spurious responses, I found that the addr value was treated a bit inconsistent: values 8..0xf will be treated as codec 0 and values 0..7 will be treated as no error regardless of whether there is a codec there, or not. With this patch, all non-existing codecs will be treated equally. In addition, printing rp and wp could help figuring out if the wp value is reported wrongly from the controller or if something else is wrong. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 23 +++++++++-------------- 1 file changed, 9 insertions(+), 14 deletions(-) Note: this patch is untested (hence the RFC indication) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fa2879a..e87eac5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -834,18 +834,6 @@ static unsigned int azx_command_addr(u32 cmd) return addr; } -static unsigned int azx_response_addr(u32 res) -{ - unsigned int addr = res & 0xf; - - if (addr >= AZX_MAX_CODECS) { - snd_BUG(); - addr = 0; - } - - return addr; -} - /* send a command */ static int azx_corb_send_cmd(struct hda_bus *bus, u32 val) { @@ -907,8 +895,15 @@ static void azx_update_rirb(struct azx *chip) rp = chip->rirb.rp << 1; /* an RIRB entry is 8-bytes */ res_ex = le32_to_cpu(chip->rirb.buf[rp + 1]); res = le32_to_cpu(chip->rirb.buf[rp]); - addr = azx_response_addr(res_ex); - if (res_ex & ICH6_RIRB_EX_UNSOL_EV) + addr = res_ex & 0xf; + if ((addr >= AZX_MAX_CODECS) || !(chip->codec_mask & (1 << addr))) { + snd_printk(KERN_ERR SFX "%s: spurious response %#x:%#x, rp = %d, wp = %d", + pci_name(chip->pci), + res, res_ex, + chip->rirb.rp, wp); + snd_BUG(); + } + else if (res_ex & ICH6_RIRB_EX_UNSOL_EV) snd_hda_queue_unsol_event(chip->bus, res, res_ex); else if (chip->rirb.cmds[addr]) { chip->rirb.res[addr] = res; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Add parameter for dumping processing coefficients
by David Henningsson 29 Jan '14

29 Jan '14

Processing coefficients are often a vital part of the codec's configuration, so dumping them can be important. However, because they are undocumented and secret, we do not want to enable this for all codecs by default. Therefore instead add this as a debugging parameter. I have prepared for codecs that want to enable this by default by the extra dump_coef bitfield, but unsure if we want to do that as long as the (unlikely, but still) race remains. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_proc.c | 34 ++++++++++++++++++++++++++++++++-- 2 files changed, 33 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 2b5d19e..ab2a444 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -361,6 +361,7 @@ struct hda_codec { unsigned int epss:1; /* supporting EPSS? */ unsigned int cached_write:1; /* write only to caches */ unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ + unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ #ifdef CONFIG_PM unsigned int power_on :1; /* current (global) power-state */ unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index a8cb22e..ce5a6da 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -24,9 +24,14 @@ #include <linux/init.h> #include <linux/slab.h> #include <sound/core.h> +#include <linux/module.h> #include "hda_codec.h" #include "hda_local.h" +static int dump_coef = -1; +module_param(dump_coef, int, 0644); +MODULE_PARM_DESC(dump_coef, "Dump processing coefficients in codec proc file (-1=auto, 0=disable, 1=enable)"); + static char *bits_names(unsigned int bits, char *names[], int size) { int i, n; @@ -488,14 +493,39 @@ static void print_unsol_cap(struct snd_info_buffer *buffer, (unsol & AC_UNSOL_ENABLED) ? 1 : 0); } +static inline bool can_dump_coef(struct hda_codec *codec) +{ + switch (dump_coef) { + case 0: return false; + case 1: return true; + default: return codec->dump_coef; + } +} + static void print_proc_caps(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { + unsigned int i, ncoeff, oldindex; unsigned int proc_caps = snd_hda_param_read(codec, nid, AC_PAR_PROC_CAP); + ncoeff = (proc_caps & AC_PCAP_NUM_COEF) >> AC_PCAP_NUM_COEF_SHIFT; snd_iprintf(buffer, " Processing caps: benign=%d, ncoeff=%d\n", - proc_caps & AC_PCAP_BENIGN, - (proc_caps & AC_PCAP_NUM_COEF) >> AC_PCAP_NUM_COEF_SHIFT); + proc_caps & AC_PCAP_BENIGN, ncoeff); + + if (!can_dump_coef(codec)) + return; + + /* Note: This is racy - another process could run in parallel and change + the coef index too. */ + oldindex = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_COEF_INDEX, 0); + for (i = 0; i < ncoeff; i++) { + unsigned int val; + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_COEF_INDEX, i); + val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PROC_COEF, + 0); + snd_iprintf(buffer, " Coeff 0x%02x: 0x%04x\n", i, val); + } + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_COEF_INDEX, oldindex); } static void print_conn_list(struct snd_info_buffer *buffer, -- 1.7.9.5

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[alsa-devel] OSS virtual driver - dma buffer size problems
by Igor Kotrasiński 29 Jan '14

29 Jan '14

Resolved. Used snd_pcm_hw_constraint_minmax.

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[alsa-devel] BockW 4th sound doesn't work on latest linus/master
by Kuninori Morimoto 29 Jan '14

29 Jan '14

Hi Simon, Mark I noticed that bockw 4th sound device doesn't work on latest linus/master. My git bisect said the trigger is this patch e14ee5deab24200e4b70fe31a8c806f0acd3d37c (ARM: shmobile: bockw: Do not enable CONFIG_DEVTMPFS defconfig) This patch removed below from defconfig -CONFIG_DEVTMPFS=y -CONFIG_DEVTMPFS_MOUNT=y But, why this has relationship to sound ?? #0 - #3 sound device have no problem. But #4 sound device will be broken by this patch Do you have any idea ? --- my kernel log ------- ... ALSA device list: #0: SSI56-AK4643 #1: SSI3-AK4554(playback) #2: SSI4-AK4554(capture) #3: SSI7-AK4554(playback) #4: SSI8-AK4554(capture) --- test result of 4th device (bad case) --- # /home/arecord -c 2 -t wav -d 10 -r 48000 -f S16_LE -D hw:4 /tmp/hoge ALSA lib pcm_hw.c:1433:(_snd_pcm_hw_open) Invalid value for card arecord: main:608: audio open error: No such file or directory --- test result of 4th device (good case) --- # /home/arecord -c 2 -t wav -d 10 -r 48000 -f S16_LE -D hw:4 /tmp/hoge Recording WAVE '/tmp/hoge' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo

2 4

[alsa-devel] ASoC updates for v3.14
by Mark Brown 29 Jan '14

29 Jan '14

The following changes since commit 7552f34a790069a008bd3e2ab4c0954b30c2f63b: Merge tag 'asoc-v3.14-3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus (2014-01-20 15:48:55 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.14-rc1 for you to fetch changes up to e65523ca79fd087d9888598eadeb9554e2bd2102: Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/fsl', 'asoc/fix/omap', 'asoc/fix/samsung', 'asoc/fix/simple', 'asoc/fix/tlv320aic32x4' and 'asoc/fix/wm5100' into asoc-linus (2014-01-27 18:18:36 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.14 A few fixes, all in drivers. Nothing stands out particularly, the biggest set of fixes is some build coverage issues from Sachin. ---------------------------------------------------------------- Charles Keepax (1): ASoC: wm5110: Extend SYSCLK patch file for rev D Fabio Estevam (2): ASoC: fsl_ssi: We do support master mode now ASoC: fsl_ssi: Do not print 'baud clock' error message all the time Mark Brown (2): ASoC: omap: Make RX51 depend on GPIOLIB due to jack usage Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/fsl', 'asoc/fix/omap', 'asoc/fix/samsung', 'asoc/fix/simple', 'asoc/fix/tlv320aic32x4' and 'asoc/fix/wm5100' into asoc-linus Markus Pargmann (2): ASoC: tlv320aic32x4: Fix mono playback ASoC: tlv320aic32x4: Fix MICPGA input configuration Sachin Kamat (8): ASoC: samsung: Fix build regressions due to gpio re-org ASoC: Samsung: s3c-i2s-v2: Fix build error ASoC: samsung: smdk_wm8994: Fix build error ASoC: wm5100: Export wm5100_detect ASoC: samsung: Fix Kconfig dependency ASoC: Samsung: Fix build error due to missing dependency ASoC: samsung: Add NULL check in i2s.c ASoC: samsung: Remove dma.h inclusion Xiubo Li (1): ASoC: simple-card: fix simple card widgets routing property name usage .../devicetree/bindings/sound/simple-card.txt | 2 +- MAINTAINERS | 3 ++ Makefile | 2 +- arch/arm/kernel/devtree.c | 2 +- arch/arm/kernel/perf_event_cpu.c | 2 +- arch/arm/kernel/traps.c | 5 +- arch/arm/mach-highbank/highbank.c | 1 + arch/arm/mach-omap2/omap4-common.c | 1 + arch/arm/mm/init.c | 2 +- arch/arm/net/bpf_jit_32.c | 6 +-- arch/arm64/include/asm/io.h | 2 +- arch/mips/include/asm/cacheops.h | 2 +- arch/mips/include/asm/r4kcache.h | 51 +++++++++++---------- arch/mips/mm/c-r4k.c | 11 ++++- arch/parisc/include/uapi/asm/socket.h | 2 +- arch/powerpc/kernel/prom_init.c | 22 +++++---- arch/powerpc/net/bpf_jit_comp.c | 7 +-- arch/s390/net/bpf_jit_comp.c | 29 +++++++----- arch/sparc/net/bpf_jit_comp.c | 17 +++++-- arch/x86/include/asm/fpu-internal.h | 13 +++--- arch/x86/kernel/cpu/perf_event_amd_ibs.c | 53 ++++++++++++++++++---- arch/x86/kernel/entry_32.S | 4 +- arch/x86/kernel/entry_64.S | 2 +- arch/x86/kvm/lapic.c | 2 +- arch/x86/mm/fault.c | 18 ++++++++ arch/x86/net/bpf_jit_comp.c | 14 ++++-- arch/x86/vdso/vclock_gettime.c | 8 ++-- drivers/acpi/acpi_lpss.c | 1 - drivers/clocksource/cadence_ttc_timer.c | 21 +++++---- drivers/gpu/drm/drm_modes.c | 2 +- drivers/gpu/drm/i915/i915_irq.c | 2 + drivers/gpu/drm/i915/intel_ddi.c | 8 +++- drivers/gpu/drm/i915/intel_display.c | 4 +- drivers/gpu/drm/nouveau/core/include/subdev/i2c.h | 2 +- .../gpu/drm/nouveau/core/include/subdev/instmem.h | 7 +++ drivers/gpu/drm/nouveau/core/subdev/i2c/base.c | 4 +- drivers/gpu/drm/nouveau/core/subdev/mxm/nv50.c | 4 +- drivers/gpu/drm/nouveau/core/subdev/therm/ic.c | 10 ++-- drivers/gpu/drm/nouveau/dispnv04/dfp.c | 2 +- drivers/gpu/drm/nouveau/dispnv04/tvnv04.c | 2 +- drivers/hwmon/coretemp.c | 2 +- drivers/md/md.c | 18 ++++++-- drivers/md/md.h | 3 ++ drivers/md/raid1.c | 3 +- drivers/md/raid10.c | 12 ++--- drivers/md/raid5.c | 7 +-- drivers/net/ethernet/broadcom/bnx2x/bnx2x_main.c | 29 ++++++------ drivers/net/ethernet/chelsio/cxgb4/l2t.c | 2 +- drivers/net/ethernet/emulex/benet/be_main.c | 11 ++++- drivers/net/ethernet/intel/e1000e/netdev.c | 8 +--- drivers/net/ethernet/qlogic/qlge/qlge_main.c | 2 + drivers/net/ethernet/via/via-rhine.c | 1 + drivers/net/usb/dm9601.c | 12 +++++ drivers/net/usb/usbnet.c | 2 +- drivers/pinctrl/pinctrl-baytrail.c | 1 - fs/dcache.c | 7 ++- fs/fs-writeback.c | 15 +++--- fs/namespace.c | 2 +- fs/nilfs2/segment.c | 10 ++-- include/linux/crash_dump.h | 2 + include/linux/i2c.h | 2 +- include/linux/seqlock.h | 27 +++++++---- include/net/if_inet6.h | 1 - kernel/fork.c | 2 +- kernel/sched/fair.c | 2 +- kernel/time/sched_clock.c | 6 +-- lib/percpu_counter.c | 4 +- mm/util.c | 5 +- net/batman-adv/main.c | 2 +- net/core/filter.c | 30 +----------- net/ieee802154/nl-phy.c | 6 ++- net/ipv4/inet_diag.c | 5 +- net/ipv4/ipmr.c | 7 ++- net/ipv4/tcp_metrics.c | 51 +++++++++++++-------- net/ipv6/addrconf.c | 38 +++++++--------- net/ipv6/ip6mr.c | 7 ++- net/rds/ib_recv.c | 7 ++- sound/soc/codecs/tlv320aic32x4.c | 28 +++++++----- sound/soc/codecs/tlv320aic32x4.h | 3 ++ sound/soc/codecs/wm5100.c | 2 + sound/soc/codecs/wm5110.c | 48 ++++++++++++++++++++ sound/soc/fsl/fsl_ssi.c | 9 +--- sound/soc/omap/Kconfig | 1 + sound/soc/samsung/Kconfig | 4 +- sound/soc/samsung/h1940_uda1380.c | 1 + sound/soc/samsung/i2s.c | 5 +- sound/soc/samsung/neo1973_wm8753.c | 1 + sound/soc/samsung/rx1950_uda1380.c | 1 + sound/soc/samsung/s3c-i2s-v2.c | 6 +-- sound/soc/samsung/s3c2412-i2s.c | 2 + sound/soc/samsung/s3c24xx-i2s.c | 2 + sound/soc/samsung/smartq_wm8987.c | 1 + sound/soc/samsung/smdk_wm8994.c | 4 +- 93 files changed, 525 insertions(+), 296 deletions(-)

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[alsa-devel] OSS virtual driver - dma buffer size problems
by Igor Kotrasiński 28 Jan '14

28 Jan '14

Hello, I'm trying to get the "cuckoo" OSS module (virtual ALSA driver) to work, and have some trouble properly setting up the dma buffer size. Relevant code (old type names are properly typedeffed from new ones) is: static snd_pcm_hardware_t snd_cuckoo_playback = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START), .channels_min = 1, .channels_max = 2, .buffer_bytes_max = (64 * 1024), .period_bytes_min = 64, .period_bytes_max = (32 * 1024), .periods_min = 2, .periods_max = 1024, (...) }; static snd_pcm_ops_t snd_cuckoo_playback_ops = { .open = snd_cuckoo_playback_open, .hw_params = snd_cuckoo_playback_hw_params, .prepare = snd_cuckoo_playback_prepare, (...) }; static int snd_cuckoo_playback_hw_params (snd_pcm_substream_t * substream, snd_pcm_hw_params_t * hw_params) { cuckoo_t *chip = snd_pcm_substream_chip (substream); snd_pcm_runtime_t *runtime = substream->runtime; dmap_t *dmap; //relevant buffer data here snd_dma_buffer_t *dmab; (...) if (runtime->dma_buffer_p) dmab = runtime->dma_buffer_p; else dmab = &(substream->dma_buffer); dmab->area = dmap->dmabuf; dmab->addr = dmap->dmabuf_phys; dmab->bytes = dmap -> buffsize; //supposed to be 4096 snd_pcm_set_runtime_buffer(substream, dmab); memset (dmap->dmabuf, 0, dmap->buffsize); return 0; } Inspecting values in snd_cuckoo_playback_prepare shows that runtime->dma_bytes is correctly set to 4096, but runtime->buffer_size is 16384 (with frame_bits == 32) and snd_pcm_lib_buffer_bytes(substream) returns 65536. If not aborted, it indeed overwrites the kernel. What is the correct way to pass a manually initialized dma buffer?

1 0

[alsa-devel] [PATCH 0/27]: ASoC: rsnd: add Renesas Gen2 DMAEngine support
by Kuninori Morimoto 28 Jan '14

28 Jan '14

Hi Mark These are Renesas Gen2 DMAEngine support patches. #1 - #26 are cleanup/prepare patches. #27 adds Gen2 DMAEngine support. These patches are based on asoc-v3.14-3 tag Kuninori Morimoto (27): 1) ASoC: rsnd: tidyup register naming of BUSIF_MODE 2) ASoC: rsnd: cleanup debug information method 3) ASoC: rsnd: remove meaningless rsnd_ssi_non() 4) ASoC: rsnd: control SCU ops in probe timing 5) ASoC: rsnd: add rsnd_scu_init(), and separate init/start 6) ASoC: rsnd: remove meaningless function parameter 7) ASoC: rsnd: merge rsnd_scu_start/stop() and rsnd_scu_transfer_start/stop() 8) ASoC: rsnd: rsnd_dai_is_clk_master() can be shared 9) ASoC: rsnd: remove pin sync option 10) ASoC: rsnd: SSI_MODE0/1 settings goes to scu.c 11) ASoC: rsnd: remove ssiu from ssi.c 12) ASoC: rsnd: rename rsnd_scu_convert_rate_ctrl() 13) ASoC: rsnd: explain SRC bypass mode settings in comment 14) ASoC: rsnd: remove duplicate *priv from rsnd_dma 15) ASoC: rsnd: non 0 is error on probe 16) ASoC: rsnd: fixup Gen2 module naming 17) ASoC: rsnd: don't use schedule_work() when rsnd_dma_start() 18) ASoC: rsnd: SCU should be called before SSI 19) ASoC: rsnd: clarify scu.c area 20) ASoC: rsnd: Merge macros in scu.c 21) ASoC: rsnd: rsnd_scu_hpbif_is_enable() become macro 22) ASoC: rsnd: merge SRC clock timing/setting 23) ASoC: rsnd: add rsnd_ssi_is_play() 24) ASoC: rsnd: extracts Gen1/Gen2 common parts 25) ASoC: rsnd: remove SSI dependent DMAEngine callback 26) ASoC: rsnd: rsnd_ssi_probe() goes forwarder than rsnd_scu_probe() 27) ASoC: rsnd: add Gen2 SRC and DMAEngine support include/sound/rcar_snd.h | 7 +- sound/soc/sh/rcar/adg.c | 158 ++++++++++-- sound/soc/sh/rcar/core.c | 70 ++++-- sound/soc/sh/rcar/gen.c | 38 ++- sound/soc/sh/rcar/rsnd.h | 66 +++-- sound/soc/sh/rcar/scu.c | 605 ++++++++++++++++++++++++++++++++++------------ sound/soc/sh/rcar/ssi.c | 222 +++++------------ 7 files changed, 779 insertions(+), 387 deletions(-)) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 0/4] add a TDA998x CODEC
by Jean-Francois Moine 28 Jan '14

28 Jan '14

The TDA998x HDMI transmitter accepts audio input from either I2S or S/PDIF. Theses inputs have different intrinsic constraints and these constraints may be modified by the audio parameters of the connected video device. The choice of I2S or S/PDIF may be the done by the user or by automatic processing (DPCM?) at each audio starting time. This asks for a dynamic audio input switch in the HDMI driver. This patch series implements the TDA998x specific CODEC. A simple function call mechanism is used for exchanges between the CODEC and the HDMI driver. Jean-Francois Moine (4): drm/i2c: tda998x: add a function for dynamic audio input switch ASoC: tda998x: add a codec driver for TDA998x ASoC: tda998x: add DT documentation ASoC: tda998x: adjust the audio hw parameters from EDID .../devicetree/bindings/sound/tda998x.txt | 14 + drivers/gpu/drm/i2c/tda998x_drv.c | 58 +++- include/drm/i2c/tda998x.h | 8 +- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 304 +++++++++++++++++++++ 6 files changed, 390 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tda998x.txt create mode 100644 sound/soc/codecs/tda998x.c -- 1.8.5.3

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[alsa-devel] Fwd: [LAC-team] LAC 2014: the Linux Audio Conference - Deadline Extension
by Jörn Nettingsmeier 28 Jan '14

28 Jan '14

FYI -------- Original Message -------- Subject: [LAC-team] LAC 2014: the Linux Audio Conference - Deadline Extension Date: Mon, 27 Jan 2014 20:02:53 +0100 From: Götz Dipper <dipper(a)zkm.de> To: LAC-team(a)lists.linuxaudio.org sorry for >< please >> << Hi all, The Linux Audio Conference submissions deadline has been extended! It is now February 3rd, 2014 (23:59 HAST) So, if you were considering to submit a paper but couldn't make up your mind yet, here is your chance to become active! Never forget that this conference lives through the people participating in it. February 3rd is the new deadline for all submission types: papers, music, installations, workshop proposals. Check out the link below for more info: http://lac.linuxaudio.org/2014/participation Please spread this information to anyone who might be interested. If you have any questions, drop us a line at lac(a)linuxaudio.org We are looking forward to seeing you in Karlsruhe in May! Thanks, The LAC2014 organization team _______________________________________________ LAC-team mailing list LAC-team(a)lists.linuxaudio.org http://lists.linuxaudio.org/listinfo/lac-team

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[alsa-devel] [PATCH 0/3] ASoC: atmel_ssc_dai: add option to choose clock
by Bo Shen 27 Jan '14

27 Jan '14

When SSC work in slave mode, the clock can come from TK pin and also can come from RK pin, this is hardware design decided. So, make it available to choose where the clock from. Bo Shen (3): ASoC: atmel_ssc_dai: make option to choose clock ASoC: atmel_wm8904: make it available to choose clock Binding: atmel-wm8904: add option to choose clock Documentation/devicetree/bindings/sound/atmel-wm8904.txt | 2 ++ sound/soc/atmel/atmel_ssc_dai.c | 16 ++++++++++++---- sound/soc/atmel/atmel_ssc_dai.h | 1 + sound/soc/atmel/atmel_wm8904.c | 10 ++++++++++ 4 files changed, 25 insertions(+), 4 deletions(-) -- 1.8.5.2

4 12

[alsa-devel] [PATCH] ASoC: fsl-sai: convert to use regmap API for Freeacale SAI
by Xiubo Li 27 Jan '14

27 Jan '14

Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> Cc: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 243 +++++++++++++++++++++++++----------------------- sound/soc/fsl/fsl_sai.h | 47 +++++----- 2 files changed, 150 insertions(+), 140 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 2858a6c..a80d709 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -15,6 +15,7 @@ #include <linux/dmaengine.h> #include <linux/module.h> #include <linux/of_address.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/dmaengine_pcm.h> @@ -22,34 +23,6 @@ #include "fsl_sai.h" -static inline u32 sai_readl(struct fsl_sai *sai, - const void __iomem *addr) -{ - u32 val; - - val = __raw_readl(addr); - - if (likely(sai->big_endian_regs)) - val = be32_to_cpu(val); - else - val = le32_to_cpu(val); - rmb(); - - return val; -} - -static inline void sai_writel(struct fsl_sai *sai, - u32 val, void __iomem *addr) -{ - wmb(); - if (likely(sai->big_endian_regs)) - val = cpu_to_be32(val); - else - val = cpu_to_le32(val); - - __raw_writel(val, addr); -} - static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int fsl_dir) { @@ -61,7 +34,8 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, else reg_cr2 = FSL_SAI_RCR2; - val_cr2 = sai_readl(sai, sai->base + reg_cr2); + regmap_read(sai->regmap, reg_cr2, &val_cr2); + val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; switch (clk_id) { @@ -81,40 +55,28 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, return -EINVAL; } - sai_writel(sai, val_cr2, sai->base + reg_cr2); - - return 0; + return regmap_write(sai->regmap, reg_cr2, val_cr2); } static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { - struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); int ret; if (dir == SND_SOC_CLOCK_IN) return 0; - ret = clk_prepare_enable(sai->clk); - if (ret) - return ret; - ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, FSL_FMT_TRANSMITTER); if (ret) { dev_err(cpu_dai->dev, "Cannot set tx sysclk: %d\n", ret); - goto err_clk; + return ret; } ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, FSL_FMT_RECEIVER); - if (ret) { + if (ret) dev_err(cpu_dai->dev, "Cannot set rx sysclk: %d\n", ret); - goto err_clk; - } - -err_clk: - clk_disable_unprepare(sai->clk); return ret; } @@ -133,8 +95,8 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, reg_cr4 = FSL_SAI_RCR4; } - val_cr2 = sai_readl(sai, sai->base + reg_cr2); - val_cr4 = sai_readl(sai, sai->base + reg_cr4); + regmap_read(sai->regmap, reg_cr2, &val_cr2); + regmap_read(sai->regmap, reg_cr4, &val_cr4); if (sai->big_endian_data) val_cr4 &= ~FSL_SAI_CR4_MF; @@ -183,35 +145,25 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, return -EINVAL; } - sai_writel(sai, val_cr2, sai->base + reg_cr2); - sai_writel(sai, val_cr4, sai->base + reg_cr4); + regmap_write(sai->regmap, reg_cr2, val_cr2); + regmap_write(sai->regmap, reg_cr4, val_cr4); return 0; } static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); int ret; - ret = clk_prepare_enable(sai->clk); - if (ret) - return ret; - ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_TRANSMITTER); if (ret) { dev_err(cpu_dai->dev, "Cannot set tx format: %d\n", ret); - goto err_clk; + return ret; } ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_RECEIVER); - if (ret) { + if (ret) dev_err(cpu_dai->dev, "Cannot set rx format: %d\n", ret); - goto err_clk; - } - -err_clk: - clk_disable_unprepare(sai->clk); return ret; } @@ -235,11 +187,12 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, reg_mr = FSL_SAI_RMR; } - val_cr4 = sai_readl(sai, sai->base + reg_cr4); + regmap_read(sai->regmap, reg_cr4, &val_cr4); + regmap_read(sai->regmap, reg_cr4, &val_cr5); + val_cr4 &= ~FSL_SAI_CR4_SYWD_MASK; val_cr4 &= ~FSL_SAI_CR4_FRSZ_MASK; - val_cr5 = sai_readl(sai, sai->base + reg_cr5); val_cr5 &= ~FSL_SAI_CR5_WNW_MASK; val_cr5 &= ~FSL_SAI_CR5_W0W_MASK; val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; @@ -257,9 +210,9 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr4 |= FSL_SAI_CR4_FRSZ(channels); val_mr = ~0UL - ((1 << channels) - 1); - sai_writel(sai, val_cr4, sai->base + reg_cr4); - sai_writel(sai, val_cr5, sai->base + reg_cr5); - sai_writel(sai, val_mr, sai->base + reg_mr); + regmap_write(sai->regmap, reg_cr4, val_cr4); + regmap_write(sai->regmap, reg_cr5, val_cr5); + regmap_write(sai->regmap, reg_mr, val_mr); return 0; } @@ -268,18 +221,15 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 tcsr, rcsr, val_cr2; - - val_cr2 = sai_readl(sai, sai->base + FSL_SAI_TCR2); - val_cr2 &= ~FSL_SAI_CR2_SYNC; - sai_writel(sai, val_cr2, sai->base + FSL_SAI_TCR2); + u32 tcsr, rcsr; - val_cr2 = sai_readl(sai, sai->base + FSL_SAI_RCR2); - val_cr2 |= FSL_SAI_CR2_SYNC; - sai_writel(sai, val_cr2, sai->base + FSL_SAI_RCR2); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, + ~FSL_SAI_CR2_SYNC); + regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC, + FSL_SAI_CR2_SYNC); - tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR); - rcsr = sai_readl(sai, sai->base + FSL_SAI_RCSR); + regmap_read(sai->regmap, FSL_SAI_TCSR, &tcsr); + regmap_read(sai->regmap, FSL_SAI_RCSR, &rcsr); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { tcsr |= FSL_SAI_CSR_FRDE; @@ -296,10 +246,9 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, tcsr |= FSL_SAI_CSR_TERE; rcsr |= FSL_SAI_CSR_TERE; - sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); - sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); + regmap_write(sai->regmap, FSL_SAI_RCSR, rcsr); + regmap_write(sai->regmap, FSL_SAI_TCSR, tcsr); break; - case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: @@ -308,8 +257,8 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, rcsr &= ~FSL_SAI_CSR_TERE; } - sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); - sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); + regmap_write(sai->regmap, FSL_SAI_TCSR, tcsr); + regmap_write(sai->regmap, FSL_SAI_RCSR, rcsr); break; default: return -EINVAL; @@ -322,41 +271,32 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 val, reg; - int ret; - - ret = clk_prepare_enable(sai->clk); - if (ret) - return ret; + u32 reg; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = FSL_SAI_TCR3; else reg = FSL_SAI_RCR3; - val = sai_readl(sai, sai->base + reg); - val |= FSL_SAI_CR3_TRCE; - sai_writel(sai, val, sai->base + reg); + regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, + FSL_SAI_CR3_TRCE); - return clk_prepare_enable(sai->clk); + return 0; } static void fsl_sai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 val, reg; + u32 reg; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = FSL_SAI_TCR3; else reg = FSL_SAI_RCR3; - val = sai_readl(sai, sai->base + reg); - val &= ~FSL_SAI_CR3_TRCE; - sai_writel(sai, val, sai->base + reg); - - clk_disable_unprepare(sai->clk); + regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, + ~FSL_SAI_CR3_TRCE); } static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { @@ -371,18 +311,13 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); - int ret; - - ret = clk_prepare_enable(sai->clk); - if (ret) - return ret; - sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR); - sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR); - sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1); - sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1); - - clk_disable_unprepare(sai->clk); + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); + regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK, + FSL_SAI_MAXBURST_TX * 2); + regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK, + FSL_SAI_MAXBURST_RX - 1); snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx, &sai->dma_params_rx); @@ -413,11 +348,88 @@ static const struct snd_soc_component_driver fsl_component = { .name = "fsl-sai", }; +/* Freescale SAI regmap */ +static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_SAI_TCSR: + case FSL_SAI_TCR1: + case FSL_SAI_TCR2: + case FSL_SAI_TCR3: + case FSL_SAI_TCR4: + case FSL_SAI_TCR5: + case FSL_SAI_TFR: + case FSL_SAI_TMR: + case FSL_SAI_RCSR: + case FSL_SAI_RCR1: + case FSL_SAI_RCR2: + case FSL_SAI_RCR3: + case FSL_SAI_RCR4: + case FSL_SAI_RCR5: + case FSL_SAI_RDR: + case FSL_SAI_RFR: + case FSL_SAI_RMR: + return true; + default: + return false; + } +} + +static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_SAI_TFR: + case FSL_SAI_RFR: + case FSL_SAI_TDR: + case FSL_SAI_RDR: + return true; + default: + return false; + } + +} + +static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_SAI_TCSR: + case FSL_SAI_TCR1: + case FSL_SAI_TCR2: + case FSL_SAI_TCR3: + case FSL_SAI_TCR4: + case FSL_SAI_TCR5: + case FSL_SAI_TDR: + case FSL_SAI_TMR: + case FSL_SAI_RCSR: + case FSL_SAI_RCR1: + case FSL_SAI_RCR2: + case FSL_SAI_RCR3: + case FSL_SAI_RCR4: + case FSL_SAI_RCR5: + case FSL_SAI_RMR: + return true; + default: + return false; + } +} + +static const struct regmap_config fsl_sai_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + + .max_register = FSL_SAI_RMR, + .readable_reg = fsl_sai_readable_reg, + .volatile_reg = fsl_sai_volatile_reg, + .writeable_reg = fsl_sai_writeable_reg, +}; + static int fsl_sai_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct fsl_sai *sai; struct resource *res; + void __iomem *base; int ret; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); @@ -425,14 +437,15 @@ static int fsl_sai_probe(struct platform_device *pdev) return -ENOMEM; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - sai->base = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(sai->base)) - return PTR_ERR(sai->base); - - sai->clk = devm_clk_get(&pdev->dev, "sai"); - if (IS_ERR(sai->clk)) { - dev_err(&pdev->dev, "Cannot get SAI's clock\n"); - return PTR_ERR(sai->clk); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); + + sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, + "sai", base, &fsl_sai_regmap_config); + if (IS_ERR(sai->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + return PTR_ERR(sai->regmap); } sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 41bb62e..1571459 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -15,31 +15,36 @@ SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +/* SAI Register Map Register */ +#define FSL_SAI_TCSR 0x00 /* SAI Transmit Control */ +#define FSL_SAI_TCR1 0x04 /* SAI Transmit Configuration 1 */ +#define FSL_SAI_TCR2 0x08 /* SAI Transmit Configuration 2 */ +#define FSL_SAI_TCR3 0x0c /* SAI Transmit Configuration 3 */ +#define FSL_SAI_TCR4 0x10 /* SAI Transmit Configuration 4 */ +#define FSL_SAI_TCR5 0x14 /* SAI Transmit Configuration 5 */ +#define FSL_SAI_TDR 0x20 /* SAI Transmit Data */ +#define FSL_SAI_TFR 0x40 /* SAI Transmit FIFO */ +#define FSL_SAI_TMR 0x60 /* SAI Transmit Mask */ +#define FSL_SAI_RCSR 0x80 /* SAI Receive Control */ +#define FSL_SAI_RCR1 0x84 /* SAI Receive Configuration 1 */ +#define FSL_SAI_RCR2 0x88 /* SAI Receive Configuration 2 */ +#define FSL_SAI_RCR3 0x8c /* SAI Receive Configuration 3 */ +#define FSL_SAI_RCR4 0x90 /* SAI Receive Configuration 4 */ +#define FSL_SAI_RCR5 0x94 /* SAI Receive Configuration 5 */ +#define FSL_SAI_RDR 0xa0 /* SAI Receive Data */ +#define FSL_SAI_RFR 0xc0 /* SAI Receive FIFO */ +#define FSL_SAI_RMR 0xe0 /* SAI Receive Mask */ + /* SAI Transmit/Recieve Control Register */ -#define FSL_SAI_TCSR 0x00 -#define FSL_SAI_RCSR 0x80 #define FSL_SAI_CSR_TERE BIT(31) #define FSL_SAI_CSR_FWF BIT(17) #define FSL_SAI_CSR_FRIE BIT(8) #define FSL_SAI_CSR_FRDE BIT(0) -/* SAI Transmit Data/FIFO/MASK Register */ -#define FSL_SAI_TDR 0x20 -#define FSL_SAI_TFR 0x40 -#define FSL_SAI_TMR 0x60 - -/* SAI Recieve Data/FIFO/MASK Register */ -#define FSL_SAI_RDR 0xa0 -#define FSL_SAI_RFR 0xc0 -#define FSL_SAI_RMR 0xe0 - /* SAI Transmit and Recieve Configuration 1 Register */ -#define FSL_SAI_TCR1 0x04 -#define FSL_SAI_RCR1 0x84 +#define FSL_SAI_CR1_RFW_MASK 0x1f /* SAI Transmit and Recieve Configuration 2 Register */ -#define FSL_SAI_TCR2 0x08 -#define FSL_SAI_RCR2 0x88 #define FSL_SAI_CR2_SYNC BIT(30) #define FSL_SAI_CR2_MSEL_MASK (0xff << 26) #define FSL_SAI_CR2_MSEL_BUS 0 @@ -50,15 +55,11 @@ #define FSL_SAI_CR2_BCD_MSTR BIT(24) /* SAI Transmit and Recieve Configuration 3 Register */ -#define FSL_SAI_TCR3 0x0c -#define FSL_SAI_RCR3 0x8c #define FSL_SAI_CR3_TRCE BIT(16) #define FSL_SAI_CR3_WDFL(x) (x) #define FSL_SAI_CR3_WDFL_MASK 0x1f /* SAI Transmit and Recieve Configuration 4 Register */ -#define FSL_SAI_TCR4 0x10 -#define FSL_SAI_RCR4 0x90 #define FSL_SAI_CR4_FRSZ(x) (((x) - 1) << 16) #define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16) #define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8) @@ -69,8 +70,6 @@ #define FSL_SAI_CR4_FSD_MSTR BIT(0) /* SAI Transmit and Recieve Configuration 5 Register */ -#define FSL_SAI_TCR5 0x14 -#define FSL_SAI_RCR5 0x94 #define FSL_SAI_CR5_WNW(x) (((x) - 1) << 24) #define FSL_SAI_CR5_WNW_MASK (0x1f << 24) #define FSL_SAI_CR5_W0W(x) (((x) - 1) << 16) @@ -100,9 +99,7 @@ #define FSL_SAI_MAXBURST_RX 6 struct fsl_sai { - struct clk *clk; - - void __iomem *base; + struct regmap *regmap; bool big_endian_regs; bool big_endian_data; -- 1.8.4

4 4

[alsa-devel] [PATCH v2 0/3] ASoC: atmel_ssc_dai: add option to choose clock
by Bo Shen 27 Jan '14

27 Jan '14

When SSC work in slave mode, the clock can come from TK pin and also can come from RK pin, this is hardware design decided. So, make it available to choose where the clock from. Changes in v2: - using "-" replace "_" in binding document Bo Shen (3): ASoC: atmel_ssc_dai: make option to choose clock ASoC: atmel_wm8904: make it available to choose clock Binding: atmel-wm8904: add option to choose clock Documentation/devicetree/bindings/sound/atmel-wm8904.txt | 6 ++++++ sound/soc/atmel/atmel_ssc_dai.c | 16 ++++++++++++---- sound/soc/atmel/atmel_ssc_dai.h | 1 + sound/soc/atmel/atmel_wm8904.c | 10 ++++++++++ 4 files changed, 29 insertions(+), 4 deletions(-) -- 1.8.5.2

3 5

[alsa-devel] [PATCH] ASoC: arizona: Check clocking during hw_params rather than startup
by Charles Keepax 27 Jan '14

27 Jan '14

It is a fairly common usage to configure SYSCLK as part of the hw_params callback in the machine driver. The current driver applies PCM constraints based on the value of SYSCLK during the startup callback however because many systems will have not configured SYSCLK yet this will often use the SYSCLK value that was used for the last stream. The patch checks the clocking constraints as part of the hw_params callback, which will be after any clocking changes have been made by the machine driver. Change-Id: I48e16256e9ed4df2f4ef1350c356b8b331d288bf Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 79 +++++++------------------------------------ 1 files changed, 13 insertions(+), 66 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e4295fe..ae5031b 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -990,29 +990,6 @@ static const int arizona_48k_bclk_rates[] = { 24576000, }; -static const unsigned int arizona_48k_rates[] = { - 12000, - 24000, - 48000, - 96000, - 192000, - 384000, - 768000, - 4000, - 8000, - 16000, - 32000, - 64000, - 128000, - 256000, - 512000, -}; - -static const struct snd_pcm_hw_constraint_list arizona_48k_constraint = { - .count = ARRAY_SIZE(arizona_48k_rates), - .list = arizona_48k_rates, -}; - static const int arizona_44k1_bclk_rates[] = { -1, 44100, @@ -1035,21 +1012,6 @@ static const int arizona_44k1_bclk_rates[] = { 22579200, }; -static const unsigned int arizona_44k1_rates[] = { - 11025, - 22050, - 44100, - 88200, - 176400, - 352800, - 705600, -}; - -static const struct snd_pcm_hw_constraint_list arizona_44k1_constraint = { - .count = ARRAY_SIZE(arizona_44k1_rates), - .list = arizona_44k1_rates, -}; - static int arizona_sr_vals[] = { 0, 12000, @@ -1077,13 +1039,15 @@ static int arizona_sr_vals[] = { 512000, }; -static int arizona_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int arizona_hw_params_rate(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; - const struct snd_pcm_hw_constraint_list *constraint; + int base = dai->driver->base; + int i, sr_val; unsigned int base_rate; switch (dai_priv->clk) { @@ -1094,31 +1058,16 @@ static int arizona_startup(struct snd_pcm_substream *substream, base_rate = priv->asyncclk; break; default: - return 0; + arizona_aif_err(dai, "Unknown clock: %d\n", dai_priv->clk); + return -EINVAL; } - if (base_rate == 0) - return 0; - - if (base_rate % 8000) - constraint = &arizona_44k1_constraint; - else - constraint = &arizona_48k_constraint; - - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - constraint); -} - -static int arizona_hw_params_rate(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); - struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; - int base = dai->driver->base; - int i, sr_val; + if (!!(base_rate % 8000) != !!(params_rate(params) % 8000)) { + arizona_aif_err(dai, + "Rate %dHz not supported off %dHz clock\n", + params_rate(params), base_rate); + return -EINVAL; + } /* * We will need to be more flexible than this in future, @@ -1308,7 +1257,6 @@ static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate) } const struct snd_soc_dai_ops arizona_dai_ops = { - .startup = arizona_startup, .set_fmt = arizona_set_fmt, .hw_params = arizona_hw_params, .set_sysclk = arizona_dai_set_sysclk, @@ -1317,7 +1265,6 @@ const struct snd_soc_dai_ops arizona_dai_ops = { EXPORT_SYMBOL_GPL(arizona_dai_ops); const struct snd_soc_dai_ops arizona_simple_dai_ops = { - .startup = arizona_startup, .hw_params = arizona_hw_params_rate, .set_sysclk = arizona_dai_set_sysclk, }; -- 1.7.2.5

3 5

[alsa-devel] Fwd: [PATCH] fix alsa-tools/hdspmixer for RME 9632 A
by Daniel Turing 27 Jan '14

27 Jan '14

Sorry, this should've been CC'd to the list, forwarding here: ---------- Forwarded message ---------- From: Daniel Turing <mail(a)danielturing.com> Date: Mon, Jan 27, 2014 at 10:52 AM Subject: Re: [PATCH] fix alsa-tools/hdspmixer for RME 9632 A To: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> Adi, thanks for getting back to me on this. On Thu, Jan 23, 2014 at 8:05 PM, Adrian Knoth <adi(a)drcomp.erfurt.thur.de> wrote: > OTOH, I have to admit I don't fully understand the patch. What you're > doing is to increment the number of output channels in case you have an > additional input module. > > The input module is handled in HDSPMixerCard.cxx. Is it just a side > effect? Because semantically, it is wrong to create output destinations > just to get input channels rendered. > > Why is hdspmixer segfaulting? Do you happen to have a backtrace or > explanation? This should be part of the commit message. I admit i don't fully understand hdspmixer ;) Here's the backtrace of the segfault in the unpatched version: [dan@demo2 hdspmixer]$ gdb src/hdspmixer GNU gdb (GDB) 7.6.2 Copyright (C) 2013 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-unknown-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /home/dan/alsa-tools/alsa-tools-1.0.27/hdspmixer/src/hdspmixer...done. (gdb) r Starting program: /home/dan/alsa-tools/alsa-tools-1.0.27/hdspmixer/src/hdspmixer warning: no loadable sections found in added symbol-file system-supplied DSO at 0x7ffff7ffa000 warning: Could not load shared library symbols for linux-vdso.so.1. Do you need "set solib-search-path" or "set sysroot"? [Thread debugging using libthread_db enabled] Using host libthread_db library "/usr/lib/libthread_db.so.1". HDSPMixer 1.11 - Copyright (C) 2003 Thomas Charbonnel <thomas(a)undata.org> This program comes with ABSOLUTELY NO WARRANTY HDSPMixer is free software, see the file COPYING for details Looking for RME cards: Card 0: RME Hammerfall HDSP 9632 at 0xfebf0000, irq 17 HDSP 9632 found! 1 RME cards card found. Initializing default presets Program received signal SIGSEGV, Segmentation fault. 0x00007ffff71399eb in fl_draw(char const*, int, int, int, int, unsigned int, Fl_Image*, int) () from /usr/lib/libfltk.so.1.3 (gdb) bt #0 0x00007ffff71399eb in fl_draw(char const*, int, int, int, int, unsigned int, Fl_Image*, int) () from /usr/lib/libfltk.so.1.3 #1 0x000000000040aa73 in HDSPMixerSelector::draw (this=0x6a8a50) at HDSPMixerSelector.cxx:180 #2 0x00007ffff70f0709 in Fl_Group::draw_child(Fl_Widget&) const () from /usr/lib/libfltk.so.1.3 #3 0x000000000040a647 in HDSPMixerIOMixer::draw (this=0x6a2360) at HDSPMixerIOMixer.cxx:79 #4 0x00007ffff70f0709 in Fl_Group::draw_child(Fl_Widget&) const () from /usr/lib/libfltk.so.1.3 #5 0x00007ffff70f08fb in Fl_Group::draw_children() () from /usr/lib/libfltk.so.1.3 #6 0x00007ffff70f0709 in Fl_Group::draw_child(Fl_Widget&) const () from /usr/lib/libfltk.so.1.3 #7 0x00007ffff711108b in Fl_Scroll::draw_clip(void*, int, int, int, int) () from /usr/lib/libfltk.so.1.3 #8 0x00007ffff7111d29 in Fl_Scroll::draw() () from /usr/lib/libfltk.so.1.3 #9 0x00007ffff70f0709 in Fl_Group::draw_child(Fl_Widget&) const () from /usr/lib/libfltk.so.1.3 #10 0x00007ffff70f08fb in Fl_Group::draw_children() () from /usr/lib/libfltk.so.1.3 #11 0x00007ffff70e9707 in Fl_Double_Window::flush(int) () from /usr/lib/libfltk.so.1.3 #12 0x00007ffff70db362 in Fl::flush() () from /usr/lib/libfltk.so.1.3 #13 0x00007ffff70dc1c5 in Fl::wait(double) () from /usr/lib/libfltk.so.1.3 #14 0x00007ffff70dc2fd in Fl::run() () from /usr/lib/libfltk.so.1.3 #15 0x0000000000404433 in main (argc=1, argv=0x7fffffffeb88) at hdspmixer.cxx:122 (gdb) It crashes when it tries to render the mixer labels for the inputs (top row). Hdspmixer correctly shows 16 channels, but fails to initialize the right amount of labels (in HDSPMixerSelector::draw(), mvalue()->label() is NULL or some illegal low address value like 0x20 for the last four channels). So i dont think the labels initialized in HDSPMixerSelector::setLabels only relate to outputs, instead they seem to double for inputs and outputs. I'm sorry that i have only limited time and energy to work on this. If the patch would go thru as it is, i can try to jump thru the hoops for "Occasional Developers" though. Best, -daniel -- from the desk of Daniel Turing see also: http://danielturing.com -- from the desk of Daniel Turing see also: http://danielturing.com

1 0

Re: [alsa-devel] [Bug 1155202] [Intel DZ77SL-50K, Intel PantherPoint HDMI, Digital Out, HDMI] No sound at all
by Niraj Kulkarni 27 Jan '14

27 Jan '14

>This is doubtful. Here we see the behavior of the PCI controller, and >there shouldn't be difference by CPU instructions. >The difference in the code between azx_reset() and your code are: >- Use of azx_writeb() and azx_writel() >- Limited loop time for GCTL_RESET bit check I tried both things. It did not work. So, I tried adding a delay between while loop and reset writel in my code. A few experiments with it shows that delay between reset state and on state determines codec detection. Only a delay of less than 100 uSec results in successful codec detection. Here is table of my observations: usleep        result 5-10             ok 50-100         ok 75-100         ok 75-150         fail 100-200       fail 250-500       fail 500-1000     fail Since link_reset_enter and exit have a lot of code in between (atleast greater than 100us) it always results in failure.

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix silent output on MacBook Air 1, 1
by Adrien Vergé 25 Jan '14

25 Jan '14

Similarly to other Apple products, MBA 1,1 needs a specific quirk. Pin 0x18 must be set to VREF_50 to have sound output. This was no longer done since commit 1a97b7f, resulting in a mute built-in speaker. This patch corrects the regression by creating a fixup for the MBA 1,1. Tested-by: Adrien Vergé <adrienverge(a)gmail.com> Signed-off-by: Adrien Vergé <adrienverge(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c564694..723de28 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1780,6 +1780,7 @@ enum { ALC889_FIXUP_DAC_ROUTE, ALC889_FIXUP_MBP_VREF, ALC889_FIXUP_IMAC91_VREF, + ALC889_FIXUP_MBA11_VREF, ALC889_FIXUP_MBA21_VREF, ALC882_FIXUP_INV_DMIC, ALC882_FIXUP_NO_PRIMARY_HP, @@ -1910,6 +1911,16 @@ static void alc889_fixup_imac91_vref(struct hda_codec *codec, alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids)); } +/* Set VREF on speaker pins on mba11 */ +static void alc889_fixup_mba11_vref(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static hda_nid_t nids[1] = { 0x18 }; + + if (action == HDA_FIXUP_ACT_INIT) + alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids)); +} + /* Set VREF on speaker pins on mba21 */ static void alc889_fixup_mba21_vref(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -2119,6 +2130,12 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC882_FIXUP_GPIO1, }, + [ALC889_FIXUP_MBA11_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc889_fixup_mba11_vref, + .chained = true, + .chain_id = ALC889_FIXUP_MBP_VREF, + }, [ALC889_FIXUP_MBA21_VREF] = { .type = HDA_FIXUP_FUNC, .v.func = alc889_fixup_mba21_vref, @@ -2194,7 +2211,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_FIXUP_EAPD), - SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC889_FIXUP_MBP_VREF), + SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC889_FIXUP_MBA11_VREF), SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC889_FIXUP_MBA21_VREF), SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF), -- 1.8.5.2

4 7

[alsa-devel] [PATCH 1/2] ASoC: pcm1681: Convert to params_width()
by Mark Brown 25 Jan '14

25 Jan '14

From: Mark Brown <broonie(a)linaro.org> This will help support future enhancements in the way we negotiate parameters in the core. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm1681.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 73f9c3630e2c..e427544183d7 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -172,16 +172,21 @@ static int pcm1681_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); int val = 0, ret; - int pcm_format = params_format(params); priv->rate = params_rate(params); switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_RIGHT_J: - if (pcm_format == SNDRV_PCM_FORMAT_S24_LE) - val = 0x00; - else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) - val = 0x03; + switch (params_width(params)) { + case 24: + val = 0; + break; + case 16: + val = 3; + break; + default: + return -EINVAL; + } break; case SND_SOC_DAIFMT_I2S: val = 0x04; -- 1.8.5.3

2 2

[alsa-devel] Audio usb switch to alternate setting 0
by Konstantinos Georgantas 25 Jan '14

25 Jan '14

Hello, I am trying to control from the client side which Alternate Setting to use. Basically I would like to be able to switch from Alternate Setting 0 to 1 and vice versa so that I am not receiving any set interface requests in the meantime. From the audio class 2.0 documentation I understand that I should be able to do something like this by using the active alternate setting control, and here comes my main problem. When I generate an interrupt to the interrupt endpoint I should expect a GET request which I never see in the host side (hope I am right here). Here comes the interrupt: interrupt->bInfo = 0x00; interrupt->bAttribute = 0x01; interrupt->wValue = 0x0100; interrupt->wIndex = 0x0004; Could you please tell me if such a functionality is supported and how the interrupt should look like? Thank you, Konstantinos

2 3

[alsa-devel] [PATCH RFC 0/9] Beaglebone-Black HDMI audio
by Jyri Sarha 24 Jan '14

24 Jan '14

This set of patches implements PCM HDMI audio driver for Beaglebone-Black HDMI output. The patches have been rebased on top of: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound topic/davinci There is couple of earlier RFC versions of the clk patch floating around, but the one here is the latest version. I'll send the associated dts changes shortly as a separate patch. Best regards, Jyri Jyri Sarha (9): clk: add gpio controlled clock ASoC: davinci-evm: Add named clock reference to DT bindings ASoC: davinci-evm: HDMI audio support for TDA998x trough McASP I2S bus ASoC: hdmi-codec: Add devicetree binding with documentation ASoC: hdmi-codec: Add SNDRV_PCM_FMTBIT_32_LE playback format ASoC: davinci: HDMI audio build for AM33XX and TDA998x drm/tilcdc: Add I2C HDMI audio config for tda998x ARM: OMAP2+: omap2plus_defconfig: Enable tilcdc and TDA998X HDMI support ARM: OMAP2+: omap2plus_defconfig: Enable BeagleBone Black HDMI audio support .../devicetree/bindings/clock/gpio-clock.txt | 21 ++ .../bindings/sound/davinci-evm-audio.txt | 12 +- Documentation/devicetree/bindings/sound/hdmi.txt | 17 ++ arch/arm/configs/omap2plus_defconfig | 5 + drivers/clk/Makefile | 1 + drivers/clk/clk-gpio.c | 209 ++++++++++++++++++ drivers/gpu/drm/tilcdc/tilcdc_slave.c | 24 ++- include/linux/clk-provider.h | 25 +++ sound/soc/codecs/hdmi.c | 12 +- sound/soc/davinci/Kconfig | 12 ++ sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 227 +++++++++++++++++++- 12 files changed, 558 insertions(+), 8 deletions(-) create mode 100644 Documentation/devicetree/bindings/clock/gpio-clock.txt create mode 100644 Documentation/devicetree/bindings/sound/hdmi.txt create mode 100644 drivers/clk/clk-gpio.c -- 1.7.9.5

6 58

[alsa-devel] HD-audio runtime PM
by Takashi Iwai 24 Jan '14

24 Jan '14

Hi, after my previous fix, the runtime PM seems working stably finally. However, there seem still some glitches: 1. The wakeup via jack or HDMI/DP detection doesn't seem to work on my test machines. WAKEEN is set properly. And its value can be read correctly at the point of runtime resume, too. 2. We don't change the device to D3 in runtime suspend. So I guess we save little power as of now. Actually, setting to D3 and resuming to D0 works fine, as far as I tested -- except for HDMI/DP ELD read out on Haswell. Although ELD read doesn't give any errors, the received bytes are corrupted. In my case, the read starts from offset 0x1c. The HDMI can be still played even at that state, but then it falls back to the stereo 2ch mode, of course. The tests were performed on both Haswell (DP and LynxPoint-LP) and IvyBridge (PantherPoint) machines. The first problem has been present, and it's also the behavior of some old chips where the codec can't go to sleep with unsol wakeup. So, in this regard, it's no regression, per se. But certainly it's no good thing. I thought Mengdong once made WAKEEN working. Mengdong, do you remember what was the condition? Could you check whether the latest kernel (at best sound git tree for-linus branch) still works? About the second point: we should do D3 if we do care power, and that's the very right reason for runtime PM. But the ELD issue is a bad bug, and it makes me wonder whether it's a graphics driver side issue or a codec side. As a quick workaround, we can implement D3 on controllers but for Haswell HDMI/DP. In anyway, this is no urgent issue, clearly targeted for 3.14. The whole patch series are found in my sound-unstable git tree test/hda branch. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git thanks, Takashi

4 28

[alsa-devel] [PATCH 1/3] ASoC: Samsung: Add NULL check in i2s.c
by Sachin Kamat 24 Jan '14

24 Jan '14

'res' could be NULL from one of the operations above (line 1243). Thus check 'res' for NULL before releasing the region to avoid null pointer dereference. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/samsung/i2s.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 92f64363427d..a9da24f34834 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1268,7 +1268,8 @@ static int samsung_i2s_probe(struct platform_device *pdev) return 0; err: - release_mem_region(regs_base, resource_size(res)); + if (res) + release_mem_region(regs_base, resource_size(res)); return ret; } -- 1.7.9.5

2 7

[alsa-devel] [PATCH 1/2] ASoC: wm8580: Fix build error
by Sachin Kamat 24 Jan '14

24 Jan '14

Re-organize the code a bit to fix the following build errors and warnings: sound/soc/codecs/wm8580.c:932:21: error: variable ‘wm8580_regmap’ has initializer but incomplete type static const struct regmap_config wm8580_regmap = { sound/soc/codecs/wm8580.c:233:20: warning: ‘wm8580_supply_names’ defined but not used [-Wunused-variable] sound/soc/codecs/wm8580.c:840:34: warning: ‘wm8580_dai’ defined but not used [-Wunused-variable] sound/soc/codecs/wm8580.c:914:36: warning: ‘soc_codec_dev_wm8580’ defined but not used [-Wunused-variable] Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/wm8580.c | 38 ++++++++++++++++++++------------------ 1 file changed, 20 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 318989acbbe5..7a9d6fa9f3b3 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -214,28 +214,12 @@ static const struct reg_default wm8580_reg_defaults[] = { { 52, 0x0000 }, }; -static bool wm8580_volatile(struct device *dev, unsigned int reg) -{ - switch (reg) { - case WM8580_RESET: - return true; - default: - return false; - } -} - struct pll_state { unsigned int in; unsigned int out; }; #define WM8580_NUM_SUPPLIES 3 -static const char *wm8580_supply_names[WM8580_NUM_SUPPLIES] = { - "AVDD", - "DVDD", - "PVDD", -}; - /* codec private data */ struct wm8580_priv { struct regmap *regmap; @@ -837,6 +821,7 @@ static const struct snd_soc_dai_ops wm8580_dai_ops_capture = { .set_pll = wm8580_set_dai_pll, }; +#if IS_ENABLED(CONFIG_I2C) static struct snd_soc_dai_driver wm8580_dai[] = { { .name = "wm8580-hifi-playback", @@ -863,6 +848,7 @@ static struct snd_soc_dai_driver wm8580_dai[] = { .ops = &wm8580_dai_ops_capture, }, }; +#endif static int wm8580_probe(struct snd_soc_codec *codec) { @@ -911,7 +897,7 @@ static int wm8580_remove(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_wm8580 = { +static struct snd_soc_codec_driver soc_codec_dev_wm8580 __maybe_unused = { .probe = wm8580_probe, .remove = wm8580_remove, .set_bias_level = wm8580_set_bias_level, @@ -929,6 +915,23 @@ static const struct of_device_id wm8580_of_match[] = { { }, }; +#if IS_ENABLED(CONFIG_I2C) +static const char *wm8580_supply_names[WM8580_NUM_SUPPLIES] = { + "AVDD", + "DVDD", + "PVDD", +}; + +static bool wm8580_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8580_RESET: + return true; + default: + return false; + } +} + static const struct regmap_config wm8580_regmap = { .reg_bits = 7, .val_bits = 9, @@ -941,7 +944,6 @@ static const struct regmap_config wm8580_regmap = { .volatile_reg = wm8580_volatile, }; -#if IS_ENABLED(CONFIG_I2C) static int wm8580_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { -- 1.7.9.5

2 5

Re: [alsa-devel] Fw: [Bug 1155202] [Intel DZ77SL-50K, Intel PantherPoint HDMI, Digital Out, HDMI] No sound at all
by Takashi Iwai 24 Jan '14

24 Jan '14

At 22 Jan 2014 14:20:23 -0000, Niraj Kulkarni wrote: > > Let me try to explain my findings: > > > > > How does this magic sequence come? > > By dumping Win7 driver's init sequence and reproducing it through ALSA driver. > > > > >why does it need to set UNSOL flag at first? > > No such need, I've put it after regular init so that only affected chipsets are reset again. Putting this anywhere around init code works (tried it). OK. > >Is the while loop guaranteed to quit for a reasonable time? > > Being household user, I can only confirm on my chipset. Only a thorough testing may ensure sanity. Then you must put a timeout not to lock up the machine. > >Can't it be simply calling azx_reset() twice? > > It is interesting hardware bug which kicks in only during first init after power-on. So if Win7 driver in qemu inits the board, and I simply plugin "unchanged" hda_intel afterwards, it still detects codecs. But multiple normal resets do not have any effect. Being no hardware expert, I can only speculate that tight spin loop after reset has the crux. A normal delay call there does not detect codec. This is doubtful. Here we see the behavior of the PCI controller, and there shouldn't be difference by CPU instructions. The difference in the code between azx_reset() and your code are: - Use of azx_writeb() and azx_writel() - Limited loop time for GCTL_RESET bit check For the first one, you can try the code change like below: -- 8< -- diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fa2879a21a50..03525cff565a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1143,10 +1143,10 @@ static void azx_enter_link_reset(struct azx *chip) unsigned long timeout; /* reset controller */ - azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_RESET); + azx_writel(chip, GCTL, 0); timeout = jiffies + msecs_to_jiffies(100); - while ((azx_readb(chip, GCTL) & ICH6_GCTL_RESET) && + while (azx_readl(chip, GCTL) && time_before(jiffies, timeout)) usleep_range(500, 1000); } @@ -1156,10 +1156,10 @@ static void azx_exit_link_reset(struct azx *chip) { unsigned long timeout; - azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); + azx_writel(chip, GCTL, ICH6_GCTL_RESET); timeout = jiffies + msecs_to_jiffies(100); - while (!azx_readb(chip, GCTL) && + while (!azx_readl(chip, GCTL) && time_before(jiffies, timeout)) usleep_range(500, 1000); } -- 8< -- For the second point, you can increase the timeout value e.g. timeout = jiffies + msecs_to_jiffies(500); > >And, yet another question comes up: doesn't S4 need the similar workaround? If S4 works, what's the difference? > > Haven't tried with sleep-wake or hibernate-restore sequence yet. Please check it. I guess S4 is broken with your patch. Takashi

1 0

Re: [alsa-devel] [PATCH Resend 7/8] ASoC: add snd_soc_of_parse_audio_simple_widgets for DeviceTree
by Mark Brown 24 Jan '14

24 Jan '14

On Fri, Jan 24, 2014 at 02:46:33AM +0000, Li.Xiubo(a)freescale.com wrote: > > > +static struct snd_soc_dapm_widget simple_widgets[] = { > > > + SND_SOC_DAPM_MIC("Mic", NULL), > > > + SND_SOC_DAPM_LINE("Line", NULL), > > > + SND_SOC_DAPM_HP("Hp", NULL), > > > + SND_SOC_DAPM_SPK("Spk", NULL), > > > +}; > This is the templates for "Microphone XXX", "Line XXX", "Headphone XXX" > and "Spk XXX" . > > Does this mean we're restricted to a particular set of names? That > > seems sad and won't work if there's a desire for more than one of a > > given widget - the main use case I can see is multiple microphones with > > separate microphone biases. How about having some templates that we > > copy and then replace the name with the one the user supplied? > This code has already supplied this I think. OK, that's fine - this is one of the things that the lack of binding documentation doesn't help with, it says what the code is supposed to do. When reviewing DT code without bindings I tend to just skim the code.

1 0

[alsa-devel] [PATCH] alsa-libs: use CLOCK_REALTIME for non-monotonic ts
by Olivier Langlois 24 Jan '14

24 Jan '14

gettimeofday() is the same than clock_gettime(CLOCK_REALTIME) except for the loss of precision and the double unnecessary conversion timespec <-> timeval. Signed-off-by: Olivier Langlois <olivier(a)trillion01.com> --- src/pcm/pcm_local.h | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index e1c0baa..8a6c743 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -961,17 +961,21 @@ typedef union snd_tmp_double { /* get the current timestamp */ static inline void gettimestamp(snd_htimestamp_t *tstamp, int monotonic) { -#if defined(HAVE_CLOCK_GETTIME) && defined(CLOCK_MONOTONIC) +#if defined(HAVE_CLOCK_GETTIME) +#if defined(CLOCK_MONOTONIC) if (monotonic) { clock_gettime(CLOCK_MONOTONIC, tstamp); } else { #endif + clock_gettime(CLOCK_REALTIME, tstamp); +#else struct timeval tv; gettimeofday(&tv, 0); tstamp->tv_sec = tv.tv_sec; tstamp->tv_nsec = tv.tv_usec * 1000L; -#if defined(HAVE_CLOCK_GETTIME) && defined(CLOCK_MONOTONIC) +#endif +#if defined(HAVE_CLOCK_GETTIME) } #endif } -- 1.8.5.3

2 1

[alsa-devel] Kernel Patch: MOTU Micro Lite (5x5 MIDI USB Interface)
by Studio65 24 Jan '14

24 Jan '14

Hello Alsa Team, may I try to explain my/our problem as short as possible. We have been truing to hack a kernel patch to get a MOTU Micro Lite (5x5 MIDI USB Interface) to run within a Debian "wheezy" with RT Kernel. We did get the five channels appear in #qjackctl or #patchage. we tried to receive input from #vkeybd and get output to #gmidimonitor while on the Hardware we connected: MOTU Midi In 1 <--> Motu Midi Out 1 MOTU Midi In 2 <--> Motu Midi Out 2 MOTU Midi In 3 <--> Motu Midi Out 3 MOTU Midi In 4 <--> Motu Midi Out 4 MOTU Midi In 5 <--> Motu Midi Out 5 ######################################################################################################################## Here is some Logs: Output of #tail -f /var/log/messages http://pastebin.com/R6xppq7k ----------------------------------------------------------------------- Jan 5 02:31:40 debian kernel: [24513.648043] usb 2-2: new full-speed USB device number 8 using ohci_hcd Jan 5 02:31:40 debian kernel: [24513.869063] usb 2-2: New USB device found, idVendor=07fd, idProduct=0001 Jan 5 02:31:40 debian kernel: [24513.869068] usb 2-2: New USB device strings: Mfr=1, Product=2, SerialNumber=0 Jan 5 02:31:40 debian kernel: [24513.869072] usb 2-2: Product: micro lite Jan 5 02:31:40 debian kernel: [24513.869074] usb 2-2: Manufacturer: Mark of the Unicorn Jan 5 02:31:40 debian kernel: [24513.872194] created 16 output and 16 input ports Jan 5 02:38:04 debian kernel: [24897.727180] usb 2-2: USB disconnect, device number 8 Jan 5 02:38:14 debian kernel: [24907.724039] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:38:24 debian kernel: [24917.724024] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:38:34 debian kernel: [24927.724032] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:38:44 debian kernel: [24937.724038] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:38:54 debian kernel: [24947.724025] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:39:04 debian kernel: [24957.724031] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:39:14 debian kernel: [24967.724026] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:39:24 debian kernel: [24977.724021] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:39:34 debian kernel: [24987.724029] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:39:44 debian kernel: [24997.724030] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:39:54 debian kernel: [25007.724034] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:40:04 debian kernel: [25017.724034] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:40:14 debian kernel: [25027.724026] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:40:24 debian kernel: [25037.724037] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:40:34 debian kernel: [25047.724041] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) Jan 5 02:40:44 debian kernel: [25057.724021] ALSA rawmidi.c:185 >rawmidi drain error (avail = 1678, buffer_size = 4096) ------------------------------------------------------------------------------------------------------------------------ ######################################################################################################################## Output of #lsusb -vv http://pastebin.com/Du0N6hCa ------------------------------------------------------------------------------------------------------------------------ nathanael@debian:~$ lsusb -vv Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 0 Full speed (or root) hub bMaxPacketSize0 64 idVendor 0x1d6b Linux Foundation idProduct 0x0002 2.0 root hub bcdDevice 3.02 iManufacturer 3 iProduct 2 iSerial 1 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 Unused bInterfaceProtocol 0 Full speed (or root) hub iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0004 1x 4 bytes bInterval 12 Bus 002 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 0 Full speed (or root) hub bMaxPacketSize0 64 idVendor 0x1d6b Linux Foundation idProduct 0x0001 1.1 root hub bcdDevice 3.02 iManufacturer 3 iProduct 2 iSerial 1 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 Unused bInterfaceProtocol 0 Full speed (or root) hub iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0002 1x 2 bytes bInterval 255 Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 0 Full speed (or root) hub bMaxPacketSize0 64 idVendor 0x1d6b Linux Foundation idProduct 0x0002 2.0 root hub bcdDevice 3.02 iManufacturer 3 iProduct 2 iSerial 1 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 Unused bInterfaceProtocol 0 Full speed (or root) hub iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0004 1x 4 bytes bInterval 12 Bus 004 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 0 Full speed (or root) hub bMaxPacketSize0 64 idVendor 0x1d6b Linux Foundation idProduct 0x0001 1.1 root hub bcdDevice 3.02 iManufacturer 3 iProduct 2 iSerial 1 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 Unused bInterfaceProtocol 0 Full speed (or root) hub iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0002 1x 2 bytes bInterval 255 Bus 004 Device 002: ID 04b4:0033 Cypress Semiconductor Corp. Mouse Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x04b4 Cypress Semiconductor Corp. idProduct 0x0033 Mouse bcdDevice 1.00 iManufacturer 0 iProduct 0 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 34 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xa0 (Bus Powered) Remote Wakeup MaxPower 100mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 1 Boot Interface Subclass bInterfaceProtocol 2 Mouse iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 61 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 0x0008 1x 8 bytes bInterval 10 Bus 002 Device 009: ID 07fd:0001 Mark of the Unicorn FastLane Quad MIDI Interface Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 3 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x07fd Mark of the Unicorn idProduct 0x0001 FastLane Quad MIDI Interface bcdDevice 1.00 iManufacturer 1 iProduct 2 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 107 bNumInterfaces 2 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 100mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 255 iInterface 0 ** UNRECOGNIZED: 0b 24 01 01 05 00 02 05 00 01 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 255 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 255 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 2 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 255 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 3 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 255 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1 ------------------------------------------------------------------------------------------------------------------------ ######################################################################################################################## Greets, Regards and all the good stuff... Nathanael Studio65 _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

2 1

[alsa-devel] [PATCH] ASoC: simple-card: fix simple card widgets routing property name usage
by Xiubo Li 24 Jan '14

24 Jan '14

Fix the usage of simple card widgets routing property, and make it the same with simple card routing property name. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Documentation/devicetree/bindings/sound/simple-card.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index e9e20ec..19c84df 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -43,7 +43,7 @@ Example: sound { compatible = "simple-audio-card"; simple-audio-card,format = "left_j"; - simple-audio-routing = + simple-audio-card,routing = "MIC_IN", "Mic Jack", "Headphone Jack", "HP_OUT", "Ext Spk", "LINE_OUT"; -- 1.8.4

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Re: [alsa-devel] [ISSUE] rtl8192ce appears to interfere with ALSA playback
by Olivier Langlois 24 Jan '14

24 Jan '14

James, I think that I have found and fix the cause of the timing miscalculation [1] I have recompiled rtlwifi modules to have access to the debug option which I have set to 4. I have also compiled ALSA modules with the XRUN_DEBUG option that I have activated with echo 1 > /proc/asound/card0/pcm0p/xrun_debug I have reproduced the problems while collecting dmesg output and using 'watch -n1 cat /proc/interrupts' I have made a couple of observations. 1. there is a small burst of interrupts for rtl8192ce at a regular interval which seems to happen during rtl_op_sw_scan and the underruns appear to coincide with that. 2. I was expecting to see XRUN entries in dmesg but I didn't. The closest that I have is [ 1700.756017] PCM: Lost interrupts? [Q] (stream=0, delta=16108, new_hw_ptr=383756, old_hw_ptr=367648) 3. I didn't realise that the default ALSA device was a plugin using dmix. I have no underruns by using directly the ALSA HW device (hw:0,0) 4. I have seen this in dmesg logs: hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj. here is the aplay output with the dmesg logs around this time: $ aplay -v ~/Music/sine.wav Playing WAVE '/home/lano1106/Music/sine.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Rate conversion PCM (48000, sformat=S32_LE) Converter: linear-interpolation Protocol version: 10002 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 15052 period_size : 940 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 940 period_event : 0 start_threshold : 15052 stop_threshold : 15052 silence_threshold: 0 silence_size : 0 boundary : 986447872 Slave: Soft volume PCM Control: PCM Playback Volume min_dB: -51 max_dB: 0 resolution: 256 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 1073741824 silence_threshold: 0 silence_size : 1073741824 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 underrun!!! (at least 7098.589 ms long) Status: state : XRUN trigger_time: 1692.774265154 tstamp : 1700.760145135 delay : -301 avail : 15353 avail_max : 15341 [ 1692.761247] hda-intel 0000:00:1b.0: azx_pcm_prepare: bufsize=0x20000, format=0x31 [ 1692.761268] hda_codec_setup_stream: NID=0x13, stream=0x8, channel=0, format=0x31 [ 1692.761275] hda_codec_setup_stream: NID=0x14, stream=0x8, channel=0, format=0x31 [ 1694.303968] rtl8192ce:_rtl92ce_phy_set_rf_power_state():<0-1> IPS Set eRf nic enable [ 1694.304021] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304086] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304200] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304314] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304427] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304541] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304655] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304693] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304693] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304693] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304693] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1694.304693] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:98053f15:10 [ 1694.304693] rtl8192ce:rtl92ce_sw_led_on():<0-1> LedAddr:4E ledpin=1 [ 1694.304693] rtl8192c_common:rtl92c_download_fw():<0-1> Firmware Version(49), Signature(0x88c1),Size(32) [ 1694.318113] rtl8192ce:rtl92ce_enable_hw_security_config():<0-1> PairwiseEncAlgorithm = 0 GroupEncAlgorithm = 0 [ 1694.318113] rtl8192ce:rtl92ce_enable_hw_security_config():<0-1> The SECR-value cc [ 1694.318113] rtl8192c_common:rtl92c_dm_check_txpower_tracking_thermal_meter():<0-1> Schedule TxPowerTracking direct call!! [ 1694.318113] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> rtl92c_dm_txpower_tracking_callback_thermalmeter [ 1694.318113] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> Readback Thermal Meter = 0xd pre thermal meter 0xf eeprom_thermalmeter 0xf [ 1694.318113] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> Initial pathA ele_d reg0xc80 = 0x40000000, ofdm_index=0xc [ 1694.318113] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> Initial reg0xa24 = 0x90e1317, cck_index=0xc, ch14 0 [ 1694.318113] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> Readback Thermal Meter = 0xd pre thermal meter 0xf eeprom_thermalmeter 0xf delta 0x2 delta_lck 0x1 delta_iqk 0x1 [ 1694.318113] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> <=== [ 1694.318113] rtl8192c_common:rtl92c_dm_initialize_txpower_tracking_thermalmeter():<0-1> pMgntInfo->txpower_tracking = 1 [ 1694.318113] rtl8192ce:rtl92ce_led_control():<0-1> ledaction 3 [ 1694.318113] rtl8192ce:rtl92ce_sw_led_on():<0-1> LedAddr:4E ledpin=1 [ 1694.640920] PCM: Lost interrupts? [Q] (stream=0, delta=14444, new_hw_ptr=90248, old_hw_ptr=75804) [ 1694.641440] rtl8192c_common:rtl92c_dm_dynamic_bb_powersaving():<0-0> Not connected to any [ 1694.641454] rtl8192c_common:rtl92c_dm_dynamic_bb_powersaving():<0-0> AP Ext Port PWDB = 0x0 [ 1694.641472] rtl8192c_common:rtl92c_dm_check_txpower_tracking_thermal_meter():<0-0> Trigger 92S Thermal Meter!! [ 1694.641517] rtlwifi:rtl_op_sw_scan_start():<0-0> [ 1694.641530] rtl8192ce:rtl92ce_led_control():<0-0> ledaction 6 [ 1694.641545] rtl8192c_common:rtl92c_dm_write_dig():<0-0> cur_igvalue = 0x20, pre_igvalue = 0x0, back_val = 10 [ 1694.654779] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1694.654786] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1694.724173] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1694.724179] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1694.794176] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1694.794181] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1694.864217] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1694.864226] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1694.934184] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1694.934193] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1695.004187] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1695.004196] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1695.074187] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1695.074196] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1695.144185] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1695.144194] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1695.214184] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1695.214196] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1695.284182] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1695.284192] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1695.354186] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1695.354195] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1695.424207] rtl8192c_common:rtl92c_phy_sw_chnl():<0-0> sw_chnl_inprogress false schedule workitem [ 1695.424217] rtl8192ce:rtl92ce_set_hw_reg():<0-0> HW_VAR_SLOT_TIME 14 [ 1695.424275] rtlwifi:rtl_op_sw_scan_complete():<0-0> [ 1695.424282] rtl8192c_common:rtl92c_dm_write_dig():<0-0> cur_igvalue = 0x22, pre_igvalue = 0x22, back_val = 10 [ 1695.524091] rtl8192ce:_rtl92ce_phy_set_rf_power_state():<0-0> IPS Set eRf nic disable [ 1695.524141] rtl8192ce:rtl92ce_led_control():<0-0> ledaction 2 [ 1695.524147] rtl8192ce:rtl92ce_sw_led_on():<0-0> LedAddr:4E ledpin=1 [ 1700.419054] rtl8192ce:_rtl92ce_phy_set_rf_power_state():<0-1> IPS Set eRf nic enable [ 1700.419106] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.419172] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.419286] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.419399] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.419513] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.419627] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.419745] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.419866] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.419984] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.420015] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.420015] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:18051d59:11 [ 1700.420015] rtl8192ce:_rtl92ce_init_mac():<0-1> reg0xec:98053f15:10 [ 1700.420015] rtl8192ce:rtl92ce_sw_led_on():<0-1> LedAddr:4E ledpin=1 [ 1700.420015] rtl8192c_common:rtl92c_download_fw():<0-1> Firmware Version(49), Signature(0x88c1),Size(32) [ 1700.420015] rtl8192ce:rtl92ce_enable_hw_security_config():<0-1> PairwiseEncAlgorithm = 0 GroupEncAlgorithm = 0 [ 1700.420015] rtl8192ce:rtl92ce_enable_hw_security_config():<0-1> The SECR-value cc [ 1700.420015] rtl8192c_common:rtl92c_dm_check_txpower_tracking_thermal_meter():<0-1> Schedule TxPowerTracking direct call!! [ 1700.420015] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> rtl92c_dm_txpower_tracking_callback_thermalmeter [ 1700.420015] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> Readback Thermal Meter = 0xd pre thermal meter 0xf eeprom_thermalmeter 0xf [ 1700.420015] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> Initial pathA ele_d reg0xc80 = 0x40000000, ofdm_index=0xc [ 1700.420015] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> Initial reg0xa24 = 0x90e1317, cck_index=0xc, ch14 0 [ 1700.420015] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> Readback Thermal Meter = 0xd pre thermal meter 0xf eeprom_thermalmeter 0xf delta 0x2 delta_lck 0x1 delta_iqk 0x1 [ 1700.420015] rtl8192c_common:rtl92c_dm_txpower_tracking_callback_thermalmeter():<0-1> <=== [ 1700.420015] rtl8192c_common:rtl92c_dm_initialize_txpower_tracking_thermalmeter():<0-1> pMgntInfo->txpower_tracking = 1 [ 1700.420015] rtl8192ce:rtl92ce_led_control():<0-1> ledaction 3 [ 1700.420015] rtl8192ce:rtl92ce_sw_led_on():<0-1> LedAddr:4E ledpin=1 [ 1700.756017] PCM: Lost interrupts? [Q] (stream=0, delta=16108, new_hw_ptr=383756, old_hw_ptr=367648) [ 1700.756513] rtlwifi:rtl_op_sw_scan_start():<0-0> [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071521.ht… On Thu, 2014-01-16 at 16:55 +1100, James Cameron wrote: > On Thu, Jan 16, 2014 at 12:25:42AM -0500, Olivier Langlois wrote: > > On 01/15/2014 12:01 PM, Larry Finger wrote: > > >On 01/15/2014 12:37 AM, Olivier Langlois wrote: > > >>How to reproduce: > > >> > > >>1. Enable Wifi while not connecting to any AP. > > >>2. lano1106@hpmini ~/Music $ aplay -c1 sine.wav > > >>underrun!!! (at least 1856093977.967 ms long) > > Indeed, that timing is fishy. > > I've looked at the alsa-devel thread [1] and your problem description > there. > > At OLPC during development we found similar symptoms showing up in > ALSA playback that were contributed to other drivers, but the > underlying causes were in the ALSA driver for our codec chip, and the > other drivers were changing the appearance of the fault only because > of timing changes. > > But there's no need to conclude that interrupts were disabled or > delayed. There can be many other causes. > > The timing miscalculation by ALSA might be important. I can't figure > out how a wireless driver can easily cause that. > > 1. > http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071142.ht… >

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[alsa-devel] [PATCH] fix alsa-tools/hdspmixer for RME 9632 A
by Daniel Turing 23 Jan '14

23 Jan '14

Dear alsa-devel, Jaroslav and Adi, since i never turned up here, let me just quickly congratulate and thank y'all for your outstanding work on linux audio. My world wouldn't be as great without you! I run a HDSP 9632 with the "AI4s-192 AIO" 4x Analog input extension board. It works great so far, but hdspmixer segfaults because it tries to show 16 channels with fltk, but doesnt care to initialize the channel labels for the input extension board. Since one can only add either the output (AO4S), or the input (AI4S) extension board, and hdspmixer already prepares channel labels for the output case, the fix is quite easy, and attached. I'd be happy if this patch could find its way into mainline alsa-tools. Thanks, -daniel -- from the desk of Daniel Turing see also: http://danielturing.com

2 1

[alsa-devel] [PATCH 1/3] ASoC: wm5102: Correct typo in EQ coefficient sizes
by Charles Keepax 23 Jan '14

23 Jan '14

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Sorry seems there was a small typo in my patches from yesterday. Let me know if you would rather I respun the patches from yesterday but for now here they are as an incremental fix. Thanks, Charles sound/soc/codecs/wm5102.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 5e03a83..ebffe81 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -685,7 +685,7 @@ 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, 18), +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -698,7 +698,7 @@ 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, 18), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -711,7 +711,7 @@ 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, 18), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -724,7 +724,7 @@ 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, 18), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), -- 1.7.2.5

2 3

[alsa-devel] [TINYCOMPRESS PATCH 1/2 v3] crec: Initial version of a compressed capture utility
by Charles Keepax 23 Jan '14

23 Jan '14

This version only supports capture of PCM streams over a compressed device and saves these as WAV files. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v2: - Switched to using POSIX file IO to improve signal safety Thanks, Charles Android.mk | 10 ++ crec.c | 418 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ makefile.linux | 20 ++- 3 files changed, 441 insertions(+), 7 deletions(-) create mode 100644 crec.c diff --git a/Android.mk b/Android.mk index 104dfef..3cdd3d0 100644 --- a/Android.mk +++ b/Android.mk @@ -20,3 +20,13 @@ LOCAL_MODULE_TAGS := optional include $(BUILD_EXECUTABLE) +include $(CLEAR_VARS) + +LOCAL_C_INCLUDES:= $(LOCAL_PATH)/include +LOCAL_SRC_FILES:= crec.c +LOCAL_MODULE := crec +LOCAL_SHARED_LIBRARIES:= libcutils libutils libtinycompress +LOCAL_MODULE_TAGS := optional + +include $(BUILD_EXECUTABLE) + diff --git a/crec.c b/crec.c new file mode 100644 index 0000000..df53e4e --- /dev/null +++ b/crec.c @@ -0,0 +1,418 @@ +/* + * BSD LICENSE + * + * crec command line recorder for compress audio record in alsa + * Copyright (c) 2011-2012, Intel Corporation + * Copyright (c) 2013-2014, Wolfson Microelectronic Ltd. + * All rights reserved. + * + * Author: Vinod Koul <vinod.koul(a)linux.intel.com> + * Author: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions are met: + * + * Redistributions of source code must retain the above copyright notice, + * this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * Neither the name of Intel Corporation nor the names of its contributors + * may be used to endorse or promote products derived from this software + * without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + * LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF + * THE POSSIBILITY OF SUCH DAMAGE. + * + * LGPL LICENSE + * + * crec command line recorder for compress audio record in alsa + * Copyright (c) 2011-2012, Intel Corporation + * Copyright (c) 2013-2014, Wolfson Microelectronic Ltd. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU Lesser General Public License, + * version 2.1, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public + * License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to + * the Free Software Foundation, Inc., + * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include <stdint.h> +#include <linux/types.h> +#include <fcntl.h> +#include <errno.h> +#include <unistd.h> +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <signal.h> +#include <stdbool.h> +#include <getopt.h> +#include <sys/time.h> +#define __force +#define __bitwise +#define __user +#include "sound/compress_params.h" +#include "sound/compress_offload.h" +#include "tinycompress/tinycompress.h" + +static int verbose; + +static const unsigned int DEFAULT_CHANNELS = 1; +static const unsigned int DEFAULT_RATE = 44100; +static const unsigned int DEFAULT_FORMAT = SNDRV_PCM_FORMAT_S16_LE; + +struct riff_chunk { + char desc[4]; + uint32_t size; +} __attribute__((__packed__)); + +struct wave_header { + struct { + struct riff_chunk chunk; + char format[4]; + } __attribute__((__packed__)) riff; + + struct { + struct riff_chunk chunk; + uint16_t type; + uint16_t channels; + uint32_t rate; + uint32_t byterate; + uint16_t blockalign; + uint16_t samplebits; + } __attribute__((__packed__)) fmt; + + struct { + struct riff_chunk chunk; + } __attribute__((__packed__)) data; +} __attribute__((__packed__)); + +const struct wave_header blank_wave_header = { + .riff = { + .chunk = { + .desc = "RIFF", + }, + .format = "WAVE", + }, + .fmt = { + .chunk = { + .desc = "fmt ", /* Note the space is important here */ + .size = sizeof(blank_wave_header.fmt) - + sizeof(blank_wave_header.fmt.chunk), + }, + .type = 0x01, /* PCM */ + }, + .data = { + .chunk = { + .desc = "data", + }, + }, +}; + +static void init_wave_header(struct wave_header *header, uint16_t channels, + uint32_t rate, uint16_t samplebits) +{ + memcpy(header, &blank_wave_header, sizeof(blank_wave_header)); + + header->fmt.channels = channels; + header->fmt.rate = rate; + header->fmt.byterate = channels * rate * (samplebits / 8); + header->fmt.blockalign = channels * (samplebits / 8); + header->fmt.samplebits = samplebits; +} + +static void size_wave_header(struct wave_header *header, uint32_t size) +{ + header->riff.chunk.size = sizeof(*header) - + sizeof(header->riff.chunk) + size; + header->data.chunk.size = size; +} + +static void usage(void) +{ + fprintf(stderr, "usage: crec [OPTIONS] filename\n" + "-c\tcard number\n" + "-d\tdevice node\n" + "-b\tbuffer size\n" + "-f\tfragments\n" + "-v\tverbose mode\n" + "-l\tlength of record in seconds\n" + "-h\tPrints this help list\n\n" + "-C\tSpecify the number of channels (default %u)\n" + "-R\tSpecify the sample rate (default %u)\n" + "-F\tSpecify the format: S16_LE, S32_LE (default S16_LE)\n\n" + "Example:\n" + "\tcrec -c 1 -d 2 test.wav\n" + "\tcrec -f 5 test.wav\n", + DEFAULT_CHANNELS, DEFAULT_RATE); + + exit(EXIT_FAILURE); +} + +static int print_time(struct compress *compress) +{ + unsigned int avail; + struct timespec tstamp; + + if (compress_get_hpointer(compress, &avail, &tstamp) != 0) { + fprintf(stderr, "Error querying timestamp\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + return -1; + } else { + printf("DSP recorded %jd.%jd\n", + (intmax_t)tstamp.tv_sec, (intmax_t)tstamp.tv_nsec*1000); + } + return 0; +} + +void capture_samples(char *name, unsigned int card, unsigned int device, + unsigned long buffer_size, unsigned int frag, + unsigned int length, unsigned int rate, + unsigned int channels, unsigned int format) +{ + struct compr_config config; + struct snd_codec codec; + struct compress *compress; + struct wave_header header; + int file; + char *buffer; + size_t written; + int read, ret; + unsigned int size, total_read = 0; + unsigned int samplebits; + + switch (format) { + case SNDRV_PCM_FORMAT_S32_LE: + samplebits = 32; + break; + default: + samplebits = 16; + break; + } + + /* Convert length from seconds to bytes */ + length = length * rate * (samplebits / 8) * channels; + + if (verbose) + printf("%s: entry, reading %u bytes\n", __func__, length); + + file = open(name, O_WRONLY); + if (file == -1) { + fprintf(stderr, "Unable to open file '%s'\n", name); + exit(EXIT_FAILURE); + } + + /* Write a header, will update with size once record is complete */ + init_wave_header(&header, channels, rate, samplebits); + written = write(file, &header, sizeof(header)); + if (written != sizeof(header)) { + fprintf(stderr, "Error writing output file header: %s\n", + strerror(errno)); + goto file_exit; + } + + memset(&codec, 0, sizeof(codec)); + memset(&config, 0, sizeof(config)); + codec.id = SND_AUDIOCODEC_PCM; + codec.ch_in = channels; + codec.ch_out = channels; + codec.sample_rate = rate; + if (!codec.sample_rate) { + fprintf(stderr, "invalid sample rate %d\n", rate); + goto file_exit; + } + codec.format = format; + if ((buffer_size != 0) && (frag != 0)) { + config.fragment_size = buffer_size/frag; + config.fragments = frag; + } + config.codec = &codec; + + compress = compress_open(card, device, COMPRESS_OUT, &config); + if (!compress || !is_compress_ready(compress)) { + fprintf(stderr, "Unable to open Compress device %d:%d\n", + card, device); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto file_exit; + }; + + if (verbose) + printf("%s: Opened compress device\n", __func__); + + size = config.fragment_size; + buffer = malloc(size * config.fragments); + if (!buffer) { + fprintf(stderr, "Unable to allocate %d bytes\n", size); + goto comp_exit; + } + + printf("Recording file %s On Card %u device %u, with buffer of %lu bytes\n", + name, card, device, buffer_size); + printf("Format %u Channels %u, %u Hz\n", + codec.id, codec.ch_out, rate); + + compress_start(compress); + + if (verbose) + printf("%s: Capturing audio NOW!!!\n", __func__); + + do { + if (length && size > length - total_read) + size = length - total_read; + + read = compress_read(compress, buffer, size); + if (read < 0) { + fprintf(stderr, "Error reading sample\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto buf_exit; + } + if ((unsigned int)read != size) { + fprintf(stderr, "We read %d, DSP sent %d\n", + size, read); + } + + if (read > 0) { + total_read += read; + + written = write(file, buffer, read); + if (written != (size_t)read) { + fprintf(stderr, "Error writing output file: %s\n", + strerror(errno)); + goto buf_exit; + } + if (verbose) { + print_time(compress); + printf("%s: read %d\n", __func__, read); + } + } + } while (!length || total_read < length); + + ret = compress_stop(compress); + if (ret < 0) { + fprintf(stderr, "Error closing stream\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + } + + /* Update file header now we know file size */ + size_wave_header(&header, total_read); + ret = lseek(file, 0, SEEK_SET); + if (ret < 0) { + fprintf(stderr, "Error seeking: %s\n", strerror(errno)); + goto buf_exit; + } + written = write(file, &header, sizeof(header)); + if (written != sizeof(header)) { + fprintf(stderr, "Error updating output file header: %s\n", + strerror(errno)); + goto buf_exit; + } + + if (verbose) + printf("%s: exit success\n", __func__); + + free(buffer); + close(file); + + compress_close(compress); + + return; +buf_exit: + free(buffer); +comp_exit: + compress_close(compress); +file_exit: + close(file); + + if (verbose) + printf("%s: exit failure\n", __func__); + + exit(EXIT_FAILURE); +} + +int main(int argc, char **argv) +{ + char *file; + unsigned long buffer_size = 0; + int c; + unsigned int card = 0, device = 0, frag = 0, length = 0; + unsigned int rate = DEFAULT_RATE, channels = DEFAULT_CHANNELS; + unsigned int format = DEFAULT_FORMAT; + + if (argc < 2) + usage(); + + verbose = 0; + while ((c = getopt(argc, argv, "hvl:R:C:F:b:f:c:d:")) != -1) { + switch (c) { + case 'h': + usage(); + break; + case 'b': + buffer_size = strtol(optarg, NULL, 0); + break; + case 'f': + frag = strtol(optarg, NULL, 10); + break; + case 'c': + card = strtol(optarg, NULL, 10); + break; + case 'd': + device = strtol(optarg, NULL, 10); + break; + case 'v': + verbose = 1; + break; + case 'l': + length = strtol(optarg, NULL, 10); + break; + case 'R': + rate = strtol(optarg, NULL, 10); + break; + case 'C': + channels = strtol(optarg, NULL, 10); + break; + case 'F': + if (strcmp(optarg, "S16_LE") == 0) { + format = SNDRV_PCM_FORMAT_S16_LE; + } else if (strcmp(optarg, "S32_LE") == 0) { + format = SNDRV_PCM_FORMAT_S32_LE; + } else { + fprintf(stderr, "Unrecognised format: %s\n", + optarg); + usage(); + } + break; + default: + exit(EXIT_FAILURE); + } + } + if (optind >= argc) + usage(); + + file = argv[optind]; + + capture_samples(file, card, device, buffer_size, frag, length, + rate, channels, format); + + printf("Finish capturing... Close Normally\n"); + exit(EXIT_SUCCESS); +} + diff --git a/makefile.linux b/makefile.linux index 677878d..81782d0 100644 --- a/makefile.linux +++ b/makefile.linux @@ -1,10 +1,13 @@ LIB = libtinycompress -BIN = cplay VER = 0.1 LIBSRC = compress.c -SRC = cplay.c utils.c +PLAYBIN = cplay +RECBIN = crec +PLAYSRC = cplay.c utils.c +RECSRC = crec.c utils.c LIBOBJ = ${LIBSRC:.c=.o} -OBJ = ${SRC:.c=.o} +PLAYOBJ = ${PLAYSRC:.c=.o} +RECOBJ = ${RECSRC:.c=.o} CC = gcc CROSS_COMPILE = @@ -13,23 +16,26 @@ CFLAGS += -std=c99 -Wall -Wextra -Wunused -DVERSION=\"${VER}\" -I./include LDFLAGS += -L. -ltinycompress LIBLDFLAGS = -lasound -all: libtinycompress cplay +all: libtinycompress cplay crec $(LIB): ${LIBOBJ} @echo " LD "$@ @${CROSS_COMPILE}${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} -$(BIN): ${OBJ} ${LIB} +$(PLAYBIN): ${PLAYOBJ} ${LIB} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${OBJ} ${LDFLAGS} + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${PLAYOBJ} ${LDFLAGS} +$(RECBIN): ${RECOBJ} ${LIB} + @echo " LD "$@ + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${RECOBJ} ${LDFLAGS} %.o: %.c @echo " CC "$< @${CROSS_COMPILE}${CC} ${CFLAGS} -c -fPIC -o $@ $< clean: - @rm -rf ${BIN} ${OBJ} ${LIB}.so ${LIBOBJ} + @rm -rf ${BIN} ${PLAYOBJ} ${RECOBJ} ${LIB}.so ${LIBOBJ} .PHONY: all clean -- 1.7.2.5

1 1

[alsa-devel] abdul, please add me to your LinkedIn network
by Putinei Ionut via LinkedIn 23 Jan '14

23 Jan '14

LinkedIn ------------ Putinei Ionut requested to add you as a connection on LinkedIn: ------------------------------------------ abdul, I'd like to add you to my professional network on LinkedIn. - Putinei Accept invitation from Putinei Ionut http://www.linkedin.com/e/jpxkk3-hqs72qer-24/uii-rtptP1H4ViCg9YiBUOvto1r4_5… View profile of Putinei Ionut http://www.linkedin.com/e/jpxkk3-hqs72qer-24/rso/227244299/bCFj/name/138036… ------------------------------------------ You are receiving Invitation emails. This email was intended for abdul moiz. Learn why this is included: http://www.linkedin.com/e/jpxkk3-hqs72qer-24/plh/http%3A%2F%2Fhelp%2Elinked… (c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

1 0

[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Use gpio_is_valid()
by Fabio Estevam 23 Jan '14

23 Jan '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> We could use gpio_is_valid() for doing the sanity check on the GPIO number. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/tlv320aic32x4.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 385dec1..02d2d3c 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -687,7 +687,7 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, aic32x4->rstn_gpio = -1; } - if (aic32x4->rstn_gpio >= 0) { + if (gpio_is_valid(aic32x4->rstn_gpio)) { ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio, GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); if (ret != 0) -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: codecs: Make OF supported CODECs visible in Kconfig
by Mark Brown 23 Jan '14

23 Jan '14

From: Mark Brown <broonie(a)linaro.org> Now that we have a generic card driver we can't rely on the card driver selecting the CODECs for us so make the CODECs that can be enabled with OF directly selectable in Kconfig. For the platforms not using OF it's not clear that we don't still want to have some board specific selection since the kernel needs to contain code to register the devices; ACPI could provide this from firmware does not yet support any kind of generic card. It may also be desirable to hide these if OF is not enabled to reduce noise. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/Kconfig | 96 +++++++++++++++++++++++++++++++----------------- 1 file changed, 62 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index fa989cb77fe4..e0c2febf3bc0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -191,8 +191,9 @@ config SND_SOC_AD73311 tristate config SND_SOC_ADAU1701 + tristate "Analog Devices ADAU1701 CODEC" + depends on I2C select SND_SOC_SIGMADSP - tristate config SND_SOC_ADAU1373 tristate @@ -204,28 +205,31 @@ config SND_SOC_ADS117X tristate config SND_SOC_AK4104 - tristate + tristate "AKM AK4104 CODEC" + depends on SPI_MASTER config SND_SOC_AK4535 tristate config SND_SOC_AK4554 - tristate + tristate "AKM AK4554 CODEC" config SND_SOC_AK4641 tristate config SND_SOC_AK4642 - tristate + tristate "AKM AK4642 CODEC" + depends on I2C config SND_SOC_AK4671 tristate config SND_SOC_AK5386 - tristate + tristate "AKM AK5638 CODEC" config SND_SOC_ALC5623 tristate + config SND_SOC_ALC5632 tristate @@ -236,14 +240,17 @@ config SND_SOC_CS42L51 tristate config SND_SOC_CS42L52 - tristate + tristate "Cirrus Logic CS42L52 CODEC" + depends on I2C config SND_SOC_CS42L73 - tristate + tristate "Cirrus Logic CS42L73 CODEC" + depends on I2C # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 - tristate + tristate "Cirrus Logic CS4270 CODEC" + depends on I2C # Cirrus Logic CS4270 Codec VD = 3.3V Errata # Select if you are affected by the errata where the part will not function @@ -254,7 +261,7 @@ config SND_SOC_CS4270_VD33_ERRATA depends on SND_SOC_CS4270 config SND_SOC_CS4271 - tristate + tristate "Cirrus Logic CS4271 CODEC" config SND_SOC_CX20442 tristate @@ -285,6 +292,9 @@ config SND_SOC_BT_SCO config SND_SOC_DMIC tristate +config SND_SOC_HDMI_CODEC + tristate "HDMI stub CODEC" + config SND_SOC_ISABELLE tristate @@ -303,14 +313,13 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate -config SND_SOC_HDMI_CODEC - tristate - config SND_SOC_PCM1681 - tristate + tristate "Texas Instruments PCM1681 CODEC" + depends on I2C config SND_SOC_PCM1792A - tristate + tristate "Texas Instruments PCM1792A CODEC" + depends on SPI_MASTER config SND_SOC_PCM3008 tristate @@ -326,7 +335,8 @@ config SND_SOC_RT5640 #Freescale sgtl5000 codec config SND_SOC_SGTL5000 - tristate + tristate "Freescale SGTL5000 CODEC" + depends on I2C config SND_SOC_SI476X tristate @@ -339,7 +349,7 @@ config SND_SOC_SN95031 tristate config SND_SOC_SPDIF - tristate + tristate "S/PDIF CODEC" config SND_SOC_SSM2518 tristate @@ -357,7 +367,8 @@ config SND_SOC_STAC9766 tristate config SND_SOC_TAS5086 - tristate + tristate "Texas Instruments TAS5086 speaker amplifier" + depends on I2C config SND_SOC_TLV320AIC23 tristate @@ -370,7 +381,8 @@ config SND_SOC_TLV320AIC32X4 tristate config SND_SOC_TLV320AIC3X - tristate + tristate "Texas Instruments TLV320AIC3x CODECs" + depends on I2C config SND_SOC_TLV320DAC33 tristate @@ -419,55 +431,69 @@ config SND_SOC_WM8400 tristate config SND_SOC_WM8510 - tristate + tristate "Wolfson Microelectronics WM8510 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8523 - tristate + tristate "Wolfson Microelectronics WM8523 DAC" + depends on I2C config SND_SOC_WM8580 - tristate + tristate "Wolfson Microelectronics WM8523 CODEC" + depends on I2C config SND_SOC_WM8711 - tristate + tristate "Wolfson Microelectronics WM8711 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8727 tristate config SND_SOC_WM8728 - tristate + tristate "Wolfson Microelectronics WM8728 DAC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8731 - tristate + tristate "Wolfson Microelectronics WM8731 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8737 - tristate + tristate "Wolfson Microelectronics WM8737 ADC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8741 - tristate + tristate "Wolfson Microelectronics WM8737 DAC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8750 - tristate + tristate "Wolfson Microelectronics WM8750 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8753 - tristate + tristate "Wolfson Microelectronics WM8753 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8770 - tristate + tristate "Wolfson Microelectronics WM8770 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8776 - tristate + tristate "Wolfson Microelectronics WM8776 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8782 tristate config SND_SOC_WM8804 - tristate + tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8900 tristate config SND_SOC_WM8903 - tristate + tristate "Wolfson Microelectronics WM8903 CODEC" + depends on I2C config SND_SOC_WM8904 tristate @@ -485,7 +511,8 @@ config SND_SOC_WM8961 tristate config SND_SOC_WM8962 - tristate + tristate "Wolfson Microelectronics WM8962 CODEC" + depends on I2C config SND_SOC_WM8971 tristate @@ -558,4 +585,5 @@ config SND_SOC_ML26124 tristate config SND_SOC_TPA6130A2 - tristate + tristate "Texas Instruments TPA6130A2 headphone amplifier" + depends on I2C -- 1.8.5.3

2 2

[alsa-devel] [PATCH] ASoC: ak4554: Add to SND_SOC_ALL_CODECS
by Mark Brown 23 Jan '14

23 Jan '14

From: Mark Brown <broonie(a)linaro.org> For build coverage. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5f74fc3f672c..fa989cb77fe4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -25,6 +25,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C + select SND_SOC_AK4554 select SND_SOC_AK4641 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C -- 1.8.5.3

1 0

[alsa-devel] [PATCH 1/2] ASoC: Samsung: Fix Kconfig dependency
by Sachin Kamat 23 Jan '14

23 Jan '14

Select S3C24XX_DMA instead of S3C2410_DMA to avoid following dependency issues and build errors: warning: (CPU_S3C2410 && CPU_S3C2442 && SND_SOC_SAMSUNG && SND_S3C24XX_I2S && SND_S3C2412_SOC_I2S && SND_SOC_SAMSUNG_SMDK2443_WM9710 && SND_SOC_SAMSUNG_LN2440SBC_ALC650) selects S3C2410_DMA which has unmet direct dependencies (ARCH_S3C24XX && S3C24XX_DMA && (CPU_S3C2410 || CPU_S3C2442)) warning: (CPU_S3C2410 && CPU_S3C2442 && SND_SOC_SAMSUNG && SND_S3C24XX_I2S && SND_S3C2412_SOC_I2S && SND_SOC_SAMSUNG_SMDK2443_WM9710 && SND_SOC_SAMSUNG_LN2440SBC_ALC650) selects S3C2410_DMA which has unmet direct dependencies (ARCH_S3C24XX && S3C24XX_DMA && (CPU_S3C2410 || CPU_S3C2442)) arch/arm/mach-s3c24xx/built-in.o: In function `s3c2410_dma_add': arch/arm/mach-s3c24xx/dma-s3c2410.c:134: undefined reference to `s3c2410_dma_init' arch/arm/mach-s3c24xx/dma-s3c2410.c:135: undefined reference to `s3c24xx_dma_order_set' arch/arm/mach-s3c24xx/dma-s3c2410.c:136: undefined reference to `s3c24xx_dma_init_map' arch/arm/plat-samsung/include/plat/dma-ops.h:60: undefined reference to `s3c_dma_get_ops' sound/soc/samsung/s3c24xx-i2s.c:293: undefined reference to `s3c2410_dma_ctrl' arch/arm/plat-samsung/include/plat/dma-ops.h:60: undefined reference to `s3c_dma_get_ops' arch/arm/plat-samsung/include/plat/dma-ops.h:60: undefined reference to `s3c_dma_get_ops' sound/built-in.o: In function `s3c2412_i2s_trigger': sound/soc/samsung/s3c-i2s-v2.c:432: undefined reference to `s3c2410_dma_ctrl' Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/samsung/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 27930fc432dc..e8fbf4e5732f 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -19,7 +19,7 @@ config SND_S3C_DMA_LEGACY config SND_S3C24XX_I2S tristate - select S3C2410_DMA + select S3C24XX_DMA config SND_S3C_I2SV2_SOC tristate -- 1.7.9.5

2 1

[alsa-devel] [PATCH 2/2] ASoC: wm5100: Export wm5100_detect
by Sachin Kamat 23 Jan '14

23 Jan '14

Export the symbol so that it is accessible to modules. Fixes the following error: ERROR: "wm5100_detect" [sound/soc/samsung/snd-soc-lowland.ko] undefined! Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/wm5100.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 4cf91deabc02..4e3e31aaf509 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -14,6 +14,7 @@ #include <linux/moduleparam.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/export.h> #include <linux/pm.h> #include <linux/gcd.h> #include <linux/gpio.h> @@ -2141,6 +2142,7 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) return 0; } +EXPORT_SYMBOL_GPL(wm5100_detect); static irqreturn_t wm5100_irq(int irq, void *data) { -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ALSA: Refactor slot assignment code
by Takashi Iwai 23 Jan '14

23 Jan '14

There are two loops that are almost identical but only with different checks. Refactor them with a simple helper, and give a bit more comments what's doing there. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/init.c | 57 +++++++++++++++++++++++++++++-------------------------- 1 file changed, 30 insertions(+), 27 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index e3c93cd77ee7..0d42fcda0de2 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -131,6 +131,31 @@ static inline int init_info_for_card(struct snd_card *card) #define init_info_for_card(card) #endif +static int check_empty_slot(struct module *module, int slot) +{ + return !slots[slot] || !*slots[slot]; +} + +/* return an empty slot number (>= 0) found in the given bitmask @mask. + * @mask == -1 == 0xffffffff means: take any free slot up to 32 + * when no slot is available, return the original @mask as is. + */ +static int get_slot_from_bitmask(int mask, int (*check)(struct module *, int), + struct module *module) +{ + int slot; + + for (slot = 0; slot < SNDRV_CARDS; slot++) { + if (slot < 32 && !(mask & (1U << slot))) + continue; + if (!test_bit(slot, snd_cards_lock)) { + if (check(module, slot)) + return slot; /* found */ + } + } + return mask; /* unchanged */ +} + /** * snd_card_create - create and initialize a soundcard structure * @idx: card index (address) [0 ... (SNDRV_CARDS-1)] @@ -152,7 +177,7 @@ int snd_card_create(int idx, const char *xid, struct snd_card **card_ret) { struct snd_card *card; - int err, idx2; + int err; if (snd_BUG_ON(!card_ret)) return -EINVAL; @@ -167,32 +192,10 @@ int snd_card_create(int idx, const char *xid, strlcpy(card->id, xid, sizeof(card->id)); err = 0; mutex_lock(&snd_card_mutex); - if (idx < 0) { - for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++) { - /* idx == -1 == 0xffff means: take any free slot */ - if (idx2 < 32 && !(idx & (1U << idx2))) - continue; - if (!test_bit(idx2, snd_cards_lock)) { - if (module_slot_match(module, idx2)) { - idx = idx2; - break; - } - } - } - } - if (idx < 0) { - for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++) { - /* idx == -1 == 0xffff means: take any free slot */ - if (idx2 < 32 && !(idx & (1U << idx2))) - continue; - if (!test_bit(idx2, snd_cards_lock)) { - if (!slots[idx2] || !*slots[idx2]) { - idx = idx2; - break; - } - } - } - } + if (idx < 0) /* first check the matching module-name slot */ + idx = get_slot_from_bitmask(idx, module_slot_match, module); + if (idx < 0) /* if not matched, assign an empty slot */ + idx = get_slot_from_bitmask(idx, check_empty_slot, module); if (idx < 0) err = -ENODEV; else if (idx < snd_ecards_limit) { -- 1.8.5.2

1 0

[alsa-devel] [patch] ALSA: bits vs bytes bug in snd_card_create()
by Dan Carpenter 23 Jan '14

23 Jan '14

The test here is intended intended to prevent shift wrapping bugs when we do "1U << idx2". We should consider the number of bits in a u32 instead of the number of bytes. Fixes: 7bb2491b35a2 ('ALSA: Add kconfig to specify the max card numbers') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- WARNING: Please check this one carefully because I might be wrong. diff --git a/sound/core/init.c b/sound/core/init.c index 1351f22f651c..e9e3db321e64 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -183,7 +183,7 @@ int snd_card_create(int idx, const char *xid, if (idx < 0) { for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++) { /* idx == -1 == 0xffff means: take any free slot */ - if (idx2 < sizeof(int) && !(idx & (1U << idx2))) + if (idx2 < 32 && !(idx & (1U << idx2))) continue; if (!test_bit(idx2, snd_cards_lock)) { if (!slots[idx2] || !*slots[idx2]) {

3 5

[alsa-devel] [PATCH] alsa-libs: initialize pcm_dmix monotonic field
by Olivier Langlois 23 Jan '14

23 Jan '14

not doing so, leaves the pcm object in an inconsistent state since 'info' field is copied from the slave which is then used when snd_pcm_hw_params_is_monotonic() is called. For instance, when using dmix with aplay and an underrun is occuring, the following info is returned: underrun!!! (at least 1248687948.256 ms long) Status: state : XRUN trigger_time: 1390347762.628483000 tstamp : 1390347766.184350000 delay : -635 avail : 15687 avail_max : 15675 now is computed from CLOCK_MONOTONIC while pcm status tstamps are from gettimeofday(). After the fix, underruns are still occuring on my setup but at least the displayed info is correct: underrun!!! (at least 7630.409 ms long) Status: state : XRUN trigger_time: 7652.739201431 tstamp : 7660.369600636 delay : -624 avail : 15676 avail_max : 15664 Signed-off-by: Olivier Langlois <olivier(a)trillion01.com> --- src/pcm/pcm_direct.c | 2 ++ src/pcm/pcm_dmix.c | 2 +- 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 899c250..5416cf7 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -857,6 +857,7 @@ static void save_slave_setting(snd_pcm_direct_t *dmix, snd_pcm_t *spcm) COPY_SLAVE(buffer_time); COPY_SLAVE(sample_bits); COPY_SLAVE(frame_bits); + COPY_SLAVE(monotonic); } #undef COPY_SLAVE @@ -1220,6 +1221,7 @@ static void copy_slave_setting(snd_pcm_direct_t *dmix, snd_pcm_t *spcm) COPY_SLAVE(buffer_time); COPY_SLAVE(sample_bits); COPY_SLAVE(frame_bits); + COPY_SLAVE(monotonic); spcm->info &= ~SND_PCM_INFO_PAUSE; spcm->boundary = recalc_boundary_size(dmix->shmptr->s.boundary, spcm->buffer_size); diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 2bd5d39..4aa6d4e 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -1104,7 +1104,7 @@ int snd_pcm_dmix_open(snd_pcm_t **pcmp, const char *name, pcm->poll_fd = dmix->poll_fd; pcm->poll_events = POLLIN; /* it's different than other plugins */ - + pcm->monotonic = spcm->monotonic; pcm->mmap_rw = 1; snd_pcm_set_hw_ptr(pcm, &dmix->hw_ptr, -1, 0); snd_pcm_set_appl_ptr(pcm, &dmix->appl_ptr, -1, 0); -- 1.8.5.3

2 1

[alsa-devel] [PATCH v2 8/8] ASoC: simple-card: simplify code
by Jean-Francois Moine 22 Jan '14

22 Jan '14

Rename the pointer to the private data structure to 'priv' to avoid confusion with the platform data pointer. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/generic/simple-card.c | 32 +++++++++++----------- 1 file changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 0f11c28..6443c87 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -47,18 +47,18 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct simple_card_data *info = + struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; - unsigned int daifmt = info->daifmt; + unsigned int daifmt = priv->daifmt; int ret; - ret = __asoc_simple_card_dai_init(codec, &info->codec_dai, daifmt); + ret = __asoc_simple_card_dai_init(codec, &priv->codec_dai, daifmt); if (ret < 0) return ret; - ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai, daifmt); + ret = __asoc_simple_card_dai_init(cpu, &priv->cpu_dai, daifmt); if (ret < 0) return ret; @@ -129,21 +129,21 @@ parse_error: } static int asoc_simple_card_parse_of(struct device_node *node, - struct simple_card_data *info, + struct simple_card_data *priv, struct device *dev) { - struct snd_soc_dai_link *dai_link = info->snd_card.dai_link; + struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; struct device_node *np; char *name; int ret; /* get CPU/CODEC common format via simple-audio-card,format */ - info->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & + priv->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); /* DAPM routes */ if (of_property_read_bool(node, "simple-audio-card,routing")) { - ret = snd_soc_of_parse_audio_routing(&info->snd_card, + ret = snd_soc_of_parse_audio_routing(&priv->snd_card, "simple-audio-card,routing"); if (ret) return ret; @@ -154,7 +154,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,cpu"); if (np) ret = asoc_simple_card_sub_parse_of(np, - &info->cpu_dai, + &priv->cpu_dai, &dai_link->cpu_of_node, &dai_link->cpu_dai_name); if (ret < 0) @@ -165,7 +165,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,codec"); if (np) ret = asoc_simple_card_sub_parse_of(np, - &info->codec_dai, + &priv->codec_dai, &dai_link->codec_of_node, &dai_link->codec_dai_name); if (ret < 0) @@ -181,22 +181,22 @@ static int asoc_simple_card_parse_of(struct device_node *node, GFP_KERNEL); sprintf(name, "%s-%s", dai_link->cpu_dai_name, dai_link->codec_dai_name); - info->snd_card.name = name; + priv->snd_card.name = name; dai_link->name = dai_link->stream_name = name; /* simple-card assumes platform == cpu */ dai_link->platform_of_node = dai_link->cpu_of_node; dev_dbg(dev, "card-name : %s\n", name); - dev_dbg(dev, "platform : %04x\n", info->daifmt); + dev_dbg(dev, "platform : %04x\n", priv->daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", dai_link->cpu_dai_name, - info->cpu_dai.fmt, - info->cpu_dai.sysclk); + priv->cpu_dai.fmt, + priv->cpu_dai.sysclk); dev_dbg(dev, "codec : %s / %04x / %d\n", dai_link->codec_dai_name, - info->codec_dai.fmt, - info->codec_dai.sysclk); + priv->codec_dai.fmt, + priv->codec_dai.sysclk); return 0; } -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH v2 7/8] ASoC: simple-card: simplify code
by Jean-Francois Moine 22 Jan '14

22 Jan '14

The platform data structure contains information which is used only by the driver, and the driver allocates platform information fields which are of no use. Move the driver specific data to a new private structure and cleanup the platform data structure. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- include/sound/simple_card.h | 4 -- sound/soc/generic/simple-card.c | 14 +++++++++++--- 2 files changed, 11 insertions(+), 7 deletions(-) diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index 6c74527..e1ac996 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -29,10 +29,6 @@ struct asoc_simple_card_info { unsigned int daifmt; struct asoc_simple_dai cpu_dai; struct asoc_simple_dai codec_dai; - - /* used in simple-card.c */ - struct snd_soc_dai_link snd_link; - struct snd_soc_card snd_card; }; #endif /* __SIMPLE_CARD_H */ diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 90c6fd5..0f11c28 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -15,6 +15,14 @@ #include <linux/string.h> #include <sound/simple_card.h> +struct simple_card_data { + struct snd_soc_card snd_card; + unsigned int daifmt; + struct asoc_simple_dai cpu_dai; + struct asoc_simple_dai codec_dai; + struct snd_soc_dai_link snd_link; +}; + static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set, unsigned int daifmt) @@ -39,7 +47,7 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct asoc_simple_card_info *info = + struct simple_card_data *info = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; @@ -121,7 +129,7 @@ parse_error: } static int asoc_simple_card_parse_of(struct device_node *node, - struct asoc_simple_card_info *info, + struct simple_card_data *info, struct device *dev) { struct snd_soc_dai_link *dai_link = info->snd_card.dai_link; @@ -195,7 +203,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, static int asoc_simple_card_probe(struct platform_device *pdev) { - struct asoc_simple_card_info *priv; + struct simple_card_data *priv; struct snd_soc_dai_link *dai_link; struct device_node *np = pdev->dev.of_node; struct device *dev = &pdev->dev; -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH v2 6/8] ASoC: simple-card: simplify code
by Jean-Francois Moine 22 Jan '14

22 Jan '14

In the non-DT sequence, the platform data is copied as a whole to the dynamic card info and the same pointer 'cinfo' is used to copy the platform information to the card. Use 'priv' as the pointer to the dynamic card info and copy only the useful information from the platform data. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/generic/simple-card.c | 34 +++++++++++++--------- 1 file changed, 20 insertions(+), 14 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 9068ab4..90c6fd5 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -195,40 +195,42 @@ static int asoc_simple_card_parse_of(struct device_node *node, static int asoc_simple_card_probe(struct platform_device *pdev) { - struct asoc_simple_card_info *cinfo; + struct asoc_simple_card_info *priv; struct snd_soc_dai_link *dai_link; struct device_node *np = pdev->dev.of_node; struct device *dev = &pdev->dev; int ret; - cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); - if (!cinfo) + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) return -ENOMEM; /* * init snd_soc_card */ - cinfo->snd_card.owner = THIS_MODULE; - cinfo->snd_card.dev = dev; - dai_link = &cinfo->snd_link; - cinfo->snd_card.dai_link = dai_link; - cinfo->snd_card.num_links = 1; + priv->snd_card.owner = THIS_MODULE; + priv->snd_card.dev = dev; + dai_link = &priv->snd_link; + priv->snd_card.dai_link = dai_link; + priv->snd_card.num_links = 1; if (np && of_device_is_available(np)) { - ret = asoc_simple_card_parse_of(np, cinfo, dev); + ret = asoc_simple_card_parse_of(np, priv, dev); if (ret < 0) { if (ret != -EPROBE_DEFER) dev_err(dev, "parse error %d\n", ret); return ret; } } else { - if (!dev->platform_data) { + struct asoc_simple_card_info *cinfo; + + cinfo = dev->platform_data; + if (!cinfo) { dev_err(dev, "no info for asoc-simple-card\n"); return -EINVAL; } - memcpy(cinfo, dev->platform_data, sizeof(*cinfo)); if (!cinfo->name || !cinfo->card || !cinfo->codec_dai.name || @@ -239,13 +241,17 @@ static int asoc_simple_card_probe(struct platform_device *pdev) return -EINVAL; } - cinfo->snd_card.name = cinfo->card; + priv->snd_card.name = cinfo->card; dai_link->name = cinfo->name; dai_link->stream_name = cinfo->name; dai_link->platform_name = cinfo->platform; dai_link->codec_name = cinfo->codec; dai_link->cpu_dai_name = cinfo->cpu_dai.name; dai_link->codec_dai_name = cinfo->codec_dai.name; + memcpy(&priv->cpu_dai, &cinfo->cpu_dai, + sizeof(priv->cpu_dai)); + memcpy(&priv->codec_dai, &cinfo->codec_dai, + sizeof(priv->codec_dai)); } /* @@ -253,9 +259,9 @@ static int asoc_simple_card_probe(struct platform_device *pdev) */ dai_link->init = asoc_simple_card_dai_init; - snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo); + snd_soc_card_set_drvdata(&priv->snd_card, priv); - return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); + return devm_snd_soc_register_card(&pdev->dev, &priv->snd_card); } static const struct of_device_id asoc_simple_of_match[] = { -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH v2 5/8] ASoC: simple-card: simplify code
by Jean-Francois Moine 22 Jan '14

22 Jan '14

The CPU and CODEC DAI names are still copied to the user info structure. Put them directly in the DAI links. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/generic/simple-card.c | 28 ++++++++++++---------- 1 file changed, 16 insertions(+), 12 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 797696f..9068ab4 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -60,7 +60,8 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) static int asoc_simple_card_sub_parse_of(struct device_node *np, struct asoc_simple_dai *dai, - const struct device_node **p_node) + const struct device_node **p_node, + const char **name) { struct device_node *node; struct clk *clk; @@ -76,7 +77,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, *p_node = node; /* get dai->name */ - ret = snd_soc_of_get_dai_name(np, &dai->name); + ret = snd_soc_of_get_dai_name(np, name); if (ret < 0) goto parse_error; @@ -146,7 +147,8 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (np) ret = asoc_simple_card_sub_parse_of(np, &info->cpu_dai, - &dai_link->cpu_of_node); + &dai_link->cpu_of_node, + &dai_link->cpu_dai_name); if (ret < 0) return ret; @@ -156,19 +158,21 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (np) ret = asoc_simple_card_sub_parse_of(np, &info->codec_dai, - &dai_link->codec_of_node); + &dai_link->codec_of_node, + &dai_link->codec_dai_name); if (ret < 0) return ret; - if (!info->cpu_dai.name || !info->codec_dai.name) + if (!dai_link->cpu_dai_name || !dai_link->codec_dai_name) return -EINVAL; /* card name is created from CPU/CODEC dai name */ name = devm_kzalloc(dev, - strlen(info->cpu_dai.name) + - strlen(info->codec_dai.name) + 2, + strlen(dai_link->cpu_dai_name) + + strlen(dai_link->codec_dai_name) + 2, GFP_KERNEL); - sprintf(name, "%s-%s", info->cpu_dai.name, info->codec_dai.name); + sprintf(name, "%s-%s", dai_link->cpu_dai_name, + dai_link->codec_dai_name); info->snd_card.name = name; dai_link->name = dai_link->stream_name = name; @@ -178,11 +182,11 @@ static int asoc_simple_card_parse_of(struct device_node *node, dev_dbg(dev, "card-name : %s\n", name); dev_dbg(dev, "platform : %04x\n", info->daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", - info->cpu_dai.name, + dai_link->cpu_dai_name, info->cpu_dai.fmt, info->cpu_dai.sysclk); dev_dbg(dev, "codec : %s / %04x / %d\n", - info->codec_dai.name, + dai_link->codec_dai_name, info->codec_dai.fmt, info->codec_dai.sysclk); @@ -240,13 +244,13 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dai_link->stream_name = cinfo->name; dai_link->platform_name = cinfo->platform; dai_link->codec_name = cinfo->codec; + dai_link->cpu_dai_name = cinfo->cpu_dai.name; + dai_link->codec_dai_name = cinfo->codec_dai.name; } /* * init snd_soc_dai_link */ - dai_link->cpu_dai_name = cinfo->cpu_dai.name; - dai_link->codec_dai_name = cinfo->codec_dai.name; dai_link->init = asoc_simple_card_dai_init; snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH v2 4/8] ASoC: simple-card: simplify code
by Jean-Francois Moine 22 Jan '14

22 Jan '14

Have a cleaner code using a DAI link pointer. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/generic/simple-card.c | 29 ++++++++++++---------- 1 file changed, 16 insertions(+), 13 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 89f83b3..797696f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -123,6 +123,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, struct asoc_simple_card_info *info, struct device *dev) { + struct snd_soc_dai_link *dai_link = info->snd_card.dai_link; struct device_node *np; char *name; int ret; @@ -145,7 +146,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (np) ret = asoc_simple_card_sub_parse_of(np, &info->cpu_dai, - &info->snd_link.cpu_of_node); + &dai_link->cpu_of_node); if (ret < 0) return ret; @@ -155,7 +156,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (np) ret = asoc_simple_card_sub_parse_of(np, &info->codec_dai, - &info->snd_link.codec_of_node); + &dai_link->codec_of_node); if (ret < 0) return ret; @@ -169,10 +170,10 @@ static int asoc_simple_card_parse_of(struct device_node *node, GFP_KERNEL); sprintf(name, "%s-%s", info->cpu_dai.name, info->codec_dai.name); info->snd_card.name = name; - info->snd_link.name = info->snd_link.stream_name = name; + dai_link->name = dai_link->stream_name = name; /* simple-card assumes platform == cpu */ - info->snd_link.platform_of_node = info->snd_link.cpu_of_node; + dai_link->platform_of_node = dai_link->cpu_of_node; dev_dbg(dev, "card-name : %s\n", name); dev_dbg(dev, "platform : %04x\n", info->daifmt); @@ -191,6 +192,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, static int asoc_simple_card_probe(struct platform_device *pdev) { struct asoc_simple_card_info *cinfo; + struct snd_soc_dai_link *dai_link; struct device_node *np = pdev->dev.of_node; struct device *dev = &pdev->dev; int ret; @@ -204,8 +206,9 @@ static int asoc_simple_card_probe(struct platform_device *pdev) */ cinfo->snd_card.owner = THIS_MODULE; cinfo->snd_card.dev = dev; - cinfo->snd_card.dai_link = &cinfo->snd_link; - cinfo->snd_card.num_links = 1; + dai_link = &cinfo->snd_link; + cinfo->snd_card.dai_link = dai_link; + cinfo->snd_card.num_links = 1; if (np && of_device_is_available(np)) { @@ -233,18 +236,18 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } cinfo->snd_card.name = cinfo->card; - cinfo->snd_link.name = cinfo->name; - cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.platform_name = cinfo->platform; - cinfo->snd_link.codec_name = cinfo->codec; + dai_link->name = cinfo->name; + dai_link->stream_name = cinfo->name; + dai_link->platform_name = cinfo->platform; + dai_link->codec_name = cinfo->codec; } /* * init snd_soc_dai_link */ - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; - cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; - cinfo->snd_link.init = asoc_simple_card_dai_init; + dai_link->cpu_dai_name = cinfo->cpu_dai.name; + dai_link->codec_dai_name = cinfo->codec_dai.name; + dai_link->init = asoc_simple_card_dai_init; snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH v2 3/8] ASoC: simple-card: simplify code
by Jean-Francois Moine 22 Jan '14

22 Jan '14

The OF pointers are put in the stack and then copied to the card descriptor. Put them directly at their right place. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/generic/simple-card.c | 55 ++++++++-------------- 1 file changed, 20 insertions(+), 35 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7c5dc73..89f83b3 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -60,8 +60,9 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) static int asoc_simple_card_sub_parse_of(struct device_node *np, struct asoc_simple_dai *dai, - struct device_node **node) + const struct device_node **p_node) { + struct device_node *node; struct clk *clk; int ret; @@ -69,9 +70,10 @@ asoc_simple_card_sub_parse_of(struct device_node *np, * get node via "sound-dai = <&phandle port>" * it will be used as xxx_of_node on soc_bind_dai_link() */ - *node = of_parse_phandle(np, "sound-dai", 0); - if (!*node) + node = of_parse_phandle(np, "sound-dai", 0); + if (!node) return -ENODEV; + *p_node = node; /* get dai->name */ ret = snd_soc_of_get_dai_name(np, &dai->name); @@ -104,7 +106,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, "system-clock-frequency", &dai->sysclk); } else { - clk = of_clk_get(*node, 0); + clk = of_clk_get(node, 0); if (!IS_ERR(clk)) dai->sysclk = clk_get_rate(clk); } @@ -112,17 +114,14 @@ asoc_simple_card_sub_parse_of(struct device_node *np, ret = 0; parse_error: - of_node_put(*node); + of_node_put(node); return ret; } static int asoc_simple_card_parse_of(struct device_node *node, struct asoc_simple_card_info *info, - struct device *dev, - struct device_node **of_cpu, - struct device_node **of_codec, - struct device_node **of_platform) + struct device *dev) { struct device_node *np; char *name; @@ -146,7 +145,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (np) ret = asoc_simple_card_sub_parse_of(np, &info->cpu_dai, - of_cpu); + &info->snd_link.cpu_of_node); if (ret < 0) return ret; @@ -156,7 +155,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (np) ret = asoc_simple_card_sub_parse_of(np, &info->codec_dai, - of_codec); + &info->snd_link.codec_of_node); if (ret < 0) return ret; @@ -173,7 +172,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, info->snd_link.name = info->snd_link.stream_name = name; /* simple-card assumes platform == cpu */ - *of_platform = *of_cpu; + info->snd_link.platform_of_node = info->snd_link.cpu_of_node; dev_dbg(dev, "card-name : %s\n", name); dev_dbg(dev, "platform : %04x\n", info->daifmt); @@ -193,34 +192,29 @@ static int asoc_simple_card_probe(struct platform_device *pdev) { struct asoc_simple_card_info *cinfo; struct device_node *np = pdev->dev.of_node; - struct device_node *of_cpu, *of_codec, *of_platform; struct device *dev = &pdev->dev; int ret; - cinfo = NULL; - of_cpu = NULL; - of_codec = NULL; - of_platform = NULL; - cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); if (!cinfo) return -ENOMEM; + /* + * init snd_soc_card + */ + cinfo->snd_card.owner = THIS_MODULE; + cinfo->snd_card.dev = dev; + cinfo->snd_card.dai_link = &cinfo->snd_link; + cinfo->snd_card.num_links = 1; + if (np && of_device_is_available(np)) { - cinfo->snd_card.dev = dev; - ret = asoc_simple_card_parse_of(np, cinfo, dev, - &of_cpu, - &of_codec, - &of_platform); + ret = asoc_simple_card_parse_of(np, cinfo, dev); if (ret < 0) { if (ret != -EPROBE_DEFER) dev_err(dev, "parse error %d\n", ret); return ret; } - cinfo->snd_link.cpu_of_node = of_cpu; - cinfo->snd_link.codec_of_node = of_codec; - cinfo->snd_link.platform_of_node = of_platform; } else { if (!dev->platform_data) { dev_err(dev, "no info for asoc-simple-card\n"); @@ -228,8 +222,6 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } memcpy(cinfo, dev->platform_data, sizeof(*cinfo)); - cinfo->snd_card.dev = dev; - if (!cinfo->name || !cinfo->card || !cinfo->codec_dai.name || @@ -254,13 +246,6 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; cinfo->snd_link.init = asoc_simple_card_dai_init; - /* - * init snd_soc_card - */ - cinfo->snd_card.owner = THIS_MODULE; - cinfo->snd_card.dai_link = &cinfo->snd_link; - cinfo->snd_card.num_links = 1; - snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo); return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH v2 2/8] ASoC: simple-card: simplify code
by Jean-Francois Moine 22 Jan '14

22 Jan '14

The DT values are copied to the non-DT structure before being moved to the card structure. Set directly the DT values in the card and move the non-DT copy to the non-DT sequence. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/generic/simple-card.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index f0784ca..7c5dc73 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -169,12 +169,13 @@ static int asoc_simple_card_parse_of(struct device_node *node, strlen(info->codec_dai.name) + 2, GFP_KERNEL); sprintf(name, "%s-%s", info->cpu_dai.name, info->codec_dai.name); - info->name = info->card = name; + info->snd_card.name = name; + info->snd_link.name = info->snd_link.stream_name = name; /* simple-card assumes platform == cpu */ *of_platform = *of_cpu; - dev_dbg(dev, "card-name : %s\n", info->card); + dev_dbg(dev, "card-name : %s\n", name); dev_dbg(dev, "platform : %04x\n", info->daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", info->cpu_dai.name, @@ -217,6 +218,9 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dev_err(dev, "parse error %d\n", ret); return ret; } + cinfo->snd_link.cpu_of_node = of_cpu; + cinfo->snd_link.codec_of_node = of_codec; + cinfo->snd_link.platform_of_node = of_platform; } else { if (!dev->platform_data) { dev_err(dev, "no info for asoc-simple-card\n"); @@ -235,26 +239,24 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dev_err(dev, "insufficient asoc_simple_card_info settings\n"); return -EINVAL; } + + cinfo->snd_card.name = cinfo->card; + cinfo->snd_link.name = cinfo->name; + cinfo->snd_link.stream_name = cinfo->name; + cinfo->snd_link.platform_name = cinfo->platform; + cinfo->snd_link.codec_name = cinfo->codec; } /* * init snd_soc_dai_link */ - cinfo->snd_link.name = cinfo->name; - cinfo->snd_link.stream_name = cinfo->name; cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; - cinfo->snd_link.platform_name = cinfo->platform; - cinfo->snd_link.codec_name = cinfo->codec; cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; - cinfo->snd_link.cpu_of_node = of_cpu; - cinfo->snd_link.codec_of_node = of_codec; - cinfo->snd_link.platform_of_node = of_platform; cinfo->snd_link.init = asoc_simple_card_dai_init; /* * init snd_soc_card */ - cinfo->snd_card.name = cinfo->card; cinfo->snd_card.owner = THIS_MODULE; cinfo->snd_card.dai_link = &cinfo->snd_link; cinfo->snd_card.num_links = 1; -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH v2 1/8] ASoC: simple-card: simplify code
by Jean-Francois Moine 22 Jan '14

22 Jan '14

The check of the mandatory fields is done for DT in its specific sequence. Move the global check to the non-DT sequence. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/generic/simple-card.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 2a1b1b5..f0784ca 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -225,16 +225,16 @@ static int asoc_simple_card_probe(struct platform_device *pdev) memcpy(cinfo, dev->platform_data, sizeof(*cinfo)); cinfo->snd_card.dev = dev; - } - if (!cinfo->name || - !cinfo->card || - !cinfo->codec_dai.name || - !(cinfo->codec || of_codec) || - !(cinfo->platform || of_platform) || - !(cinfo->cpu_dai.name || of_cpu)) { - dev_err(dev, "insufficient asoc_simple_card_info settings\n"); - return -EINVAL; + if (!cinfo->name || + !cinfo->card || + !cinfo->codec_dai.name || + !cinfo->codec || + !cinfo->platform || + !cinfo->cpu_dai.name) { + dev_err(dev, "insufficient asoc_simple_card_info settings\n"); + return -EINVAL; + } } /* -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH v2 0/8] ASoC: simple-card: simplify code
by Jean-Francois Moine 22 Jan '14

22 Jan '14

This patch set simplifies the code and make easier a possible multi-DAI links extension. -- v2: - split the changes into 8 patches (Mark Brown request) - remove the dynamic DAI link array allocation - fix lack of init function when DT use -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH v2 0/6] ASoC: Samsung: Fix build errors and warnings
by Sachin Kamat 22 Jan '14

22 Jan '14

This series attempts to fix several build errors and warnings that came up while building for different Samsung defconfigs. Some of these could possibly be applied to stable kernels as well (like patch 2 & 6). This series is based on latest linux-next and is build tested on various Samsung configs. Only change in v2 is that CONFIG_OF has been dropped in patch 6 as suggested by reviewers. This made earlier patch 7 redundant and has been dropped in this version of the series. Sachin Kamat (6): ASoC: Samsung: Fix build regressions due to gpio re-org ASoC: Samsung: s3c-i2s-v2: Fix build error ASoC: Samsung: h1940_uda1380: Remove unused variable ASoC: Samsung: h1940_uda1380: Fix build warning ASoC: Samsung: rx1950_uda1380: Remove unused variable ASoC: Samsung: smdk_wm8994: Fix build error sound/soc/samsung/h1940_uda1380.c | 4 ++-- sound/soc/samsung/neo1973_wm8753.c | 1 + sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 6 +++--- sound/soc/samsung/s3c2412-i2s.c | 2 ++ sound/soc/samsung/s3c24xx-i2s.c | 2 ++ sound/soc/samsung/smartq_wm8987.c | 1 + sound/soc/samsung/smdk_wm8994.c | 4 +--- 8 files changed, 13 insertions(+), 9 deletions(-) -- 1.7.9.5

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[alsa-devel] Hello list! My first question!
by gianluca 22 Jan '14

22 Jan '14

Hello list! I am working on a custom embedded ARM based-board (iMX28) with a SGTL5000 audio codec (correctly seen by kernel 3.12.2) but the lineout lines (LINEOUT_L and LINEOUT_R) are inputs for a Texas Instruments' Volume Controller named LM48100Q i2c-bus based. I would like to know if there are any chances to add a ALSA Driver for this VOLUME CONTROLLER. I am looking at sounds/i2c/other/pt2258.c driver and it looks very similar to lm48100. Actually this lm48100 has 5 registers but only 3 are useful to me: MODE CONTROL (how many inputs it uses (1 or 2) or POWER ON/OFF), and one register for INPUT1 VOLUME SETTING and another one for INPUT2 VOLUME SETTING. Or I will be forced to use this in a userspace driver accessing the i2c, but with no integration in any linux-based audio mixer alsa tool. Can you help me on this? Regards, -- ,,, (o o) ======oOO==(_)==OOo====== Gianluca Renzi R&D phone: +39.0542.609120 fax: +39.0542.609212 .oooO Oooo. ======( )==( )======= \ ( ) / \_) (_/ =================== | I would like to | | fix the world, | | but they're not | | giving me the | \ source code! / --------------- ¯\_(".)_/¯

1 0

[alsa-devel] Fw: [Bug 1155202] [Intel DZ77SL-50K, Intel PantherPoint HDMI, Digital Out, HDMI] No sound at all
by Niraj Kulkarni 22 Jan '14

22 Jan '14

Let me try to explain my findings: > How does this magic sequence come? By dumping Win7 driver's init sequence and reproducing it through ALSA driver. >why does it need to set UNSOL flag at first? No such need, I've put it after regular init so that only affected chipsets are reset again. Putting this anywhere around init code works (tried it). >Is the while loop guaranteed to quit for a reasonable time? Being household user, I can only confirm on my chipset. Only a thorough testing may ensure sanity. >Can't it be simply calling azx_reset() twice? It is interesting hardware bug which kicks in only during first init after power-on. So if  Win7 driver in qemu inits the board, and I simply plugin "unchanged" hda_intel afterwards, it still detects codecs. But multiple normal resets do not have any effect. Being no hardware expert, I can only speculate that tight spin loop after reset has the crux. A normal delay call there does not detect codec. >And, yet another question comes up: doesn't S4 need the similar workaround? If S4 works, what's the difference? Haven't tried with sleep-wake or hibernate-restore sequence yet. -- You received this bug notification because you are subscribed to the bug report. https://bugs.launchpad.net/bugs/1155202 Title:  [Intel DZ77SL-50K, Intel PantherPoint HDMI, Digital Out, HDMI] No  sound at all Status in “alsa-driver” package in Ubuntu:  Incomplete Bug description:  I am using ubuntu 10.04  I expected sound  Got no sound at all  I am very new to ubuntu so I have no idea how to find the version of  the package I am using  ProblemType: Bug  DistroRelease: Ubuntu 12.04  Package: alsa-base 1.0.25+dfsg-0ubuntu1.1  ProcVersionSignature: Ubuntu 3.2.0-38.61-generic-pae 3.2.37  Uname: Linux 3.2.0-38-generic-pae i686  AlsaVersion: Advanced Linux Sound Architecture Driver Version 1.0.24.  AplayDevices:   **** List of PLAYBACK Hardware Devices ****   card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0]     Subdevices: 1/1     Subdevice #0: subdevice #0  ApportVersion: 2.0.1-0ubuntu17.1  Architecture: i386  ArecordDevices: **** List of CAPTURE Hardware Devices ****  AudioDevicesInUse:   USER        PID ACCESS COMMAND   /dev/snd/controlC0:  jan        1750 F.... pulseaudio  Card0.Amixer.info:   Card hw:0 'PCH'/'HDA Intel PCH at 0xf7c30000 irq 45'     Mixer name   : 'Intel PantherPoint HDMI'     Components   : 'HDA:80862806,80860101,00100000'     Controls      : 7     Simple ctrls  : 2  CurrentDmesg: [   24.291482] eth0: no IPv6 routers present  Date: Thu Mar 14 08:53:13 2013  InstallationMedia: Ubuntu 12.04.1 LTS "Precise Pangolin" - Release i386 (20120817.3)  MarkForUpload: True  PackageArchitecture: all  ProcEnviron:   TERM=xterm   PATH=(custom, no user)   LANG=en_US.UTF-8   SHELL=/bin/bash  SourcePackage: alsa-driver  Symptom: audio  Symptom_AlsaPlaybackTest: ALSA playback test through plughw:PCH failed  Symptom_Card: Built-in Audio - HDA Intel PCH  Symptom_DevicesInUse:   1750  jan       F.... pulseaudio       PID ACCESS COMMAND  Symptom_Jack: Digital Out, HDMI  Symptom_Type: No sound at all  Title: [, Intel PantherPoint HDMI, Digital Out, HDMI] No sound at all  UpgradeStatus: No upgrade log present (probably fresh install)  dmi.bios.date: 03/19/2012  dmi.bios.vendor: Intel Corporation  dmi.bios.version: SLZ7710H.86A.0055.2012.0319.2140  dmi.board.name: DZ77SL-50K  dmi.board.vendor: Intel Corporation  dmi.board.version: AAG55115-300  dmi.chassis.type: 3  dmi.modalias: dmi:bvnIntelCorporation:bvrSLZ7710H.86A.0055.2012.0319.2140:bd03/19/2012:svn:pn:pvr:rvnIntelCorporation:rnDZ77SL-50K:rvrAAG55115-300:cvn:ct3:cvr: To manage notifications about this bug go to: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1155202/+subscri…

1 0

[alsa-devel] [RFC PATCH] ALSA: hda - Detect codecs on Intel DZ77SL-50K
by David Henningsson 22 Jan '14

22 Jan '14

From: Niraj <kulkarniniraj(a)rediffmail.com> Apparently this board seems to need some extra initialization before a proper codec probe value can be read. BugLink: https://bugs.launchpad.net/bugs/1155202 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 6 ++++++ 1 file changed, 6 insertions(+) Hi Takashi, Here's a patch from Niraj, which has found this code below to make audio work on the Intel DZ77SL-50K board. Without it, no codecs will be detected. I'm not sure if you like it this way or if you want me or Niraj to change anything, or debug further. Or maybe add a new quirk for this particular device? diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fa2879a..2bf46e9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1306,6 +1306,12 @@ static void azx_init_chip(struct azx *chip, int full_reset) /* reset controller */ azx_reset(chip, full_reset); + if (!chip->codec_mask) { + azx_writel(chip, GCTL, 0x100); + while(azx_readl(chip, GCTL) != 0x0); + azx_writel(chip, GCTL, 0x1); + azx_reset(chip, full_reset); + } /* initialize interrupts */ azx_int_clear(chip); -- 1.7.9.5

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[alsa-devel] [PATCH v2] ASoC: wm_adsp: Add debug print to note that the DSP has shutdown
by Charles Keepax 22 Jan '14

22 Jan '14

It can be useful for debugging purposes to see at what point the DSP has powered down, so add a message to inform us of this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 4fbcab6..90d7baf 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1637,6 +1637,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, list_del(&alg_region->list); kfree(alg_region); } + + adsp_dbg(dsp, "Shutdown complete\n"); break; default: -- 1.7.2.5

2 1

[alsa-devel] [PATCH 0/7] ASoC: Samsung: Fix build errors and warnings
by Sachin Kamat 22 Jan '14

22 Jan '14

This series attempts to fix several build errors and warnings that came up while building for deifferent Samsung defconfigs. Some of these could possibly be applied to stable kernels as well (like patch 2 & 6). This series is based on latest linux-next and is build tested on various Samsung configs. Sachin Kamat (7): ASoC: Samsung: Fix build regressions due to gpio re-org ASoC: Samsung: s3c-i2s-v2: Fix build error ASoC: Samsung: h1940_uda1380: Remove unused variable ASoC: Samsung: h1940_uda1380: Fix build warning ASoC: Samsung: rx1950_uda1380: Remove unused variable ASoC: Samsung: smdk_wm8994: Fix build error ASoC: Samsung: smdk_wm8994: Fix unused variable warning sound/soc/samsung/h1940_uda1380.c | 4 ++-- sound/soc/samsung/neo1973_wm8753.c | 1 + sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 6 +++--- sound/soc/samsung/s3c2412-i2s.c | 2 ++ sound/soc/samsung/s3c24xx-i2s.c | 2 ++ sound/soc/samsung/smartq_wm8987.c | 1 + sound/soc/samsung/smdk_wm8994.c | 12 ++++++------ 8 files changed, 18 insertions(+), 12 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: wm8804: Allow control of master clock divider in PLL generation
by Daniel Matuschek 22 Jan '14

22 Jan '14

WM8804 can run with PLL frequencies of 256xfs and 128xfs for most sample rates. At 192kHz only 128xfs is supported. The existing driver selects 128xfs automatically for some lower samples rates. By using an additional mclk_div divider, it is now possible to control the behaviour. This allows using 256xfs PLL frequency on all sample rates up to 96kHz. It should allow lower jitter and better signal quality. The behavior has to be controlled by the sound card driver, because some sample frequency share the same setting. e.g. 192kHz and 96kHz use 24.576MHz master clock. The only difference is the MCLK divider. Signed-off-by: Daniel Matuschek <daniel(a)matuschek.net> --- sound/soc/codecs/wm8804.c | 17 ++++++++++++++--- sound/soc/codecs/wm8804.h | 4 ++++ 2 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 1704b1e..4619bf8 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -63,6 +63,7 @@ struct wm8804_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; + int mclk_div; }; static int txsrc_get(struct snd_kcontrol *kcontrol, @@ -318,7 +319,7 @@ static struct { #define FIXED_PLL_SIZE ((1ULL << 22) * 10) static int pll_factors(struct pll_div *pll_div, unsigned int target, - unsigned int source) + unsigned int source, unsigned int mclk_div) { u64 Kpart; unsigned long int K, Ndiv, Nmod, tmp; @@ -330,7 +331,8 @@ static int pll_factors(struct pll_div *pll_div, unsigned int target, */ for (i = 0; i < ARRAY_SIZE(post_table); i++) { tmp = target * post_table[i].div; - if (tmp >= 90000000 && tmp <= 100000000) { + if ((tmp >= 90000000 && tmp <= 100000000) && + (mclk_div == post_table[i].mclkdiv)) { pll_div->freqmode = post_table[i].freqmode; pll_div->mclkdiv = post_table[i].mclkdiv; target *= post_table[i].div; @@ -387,8 +389,12 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, } else { int ret; struct pll_div pll_div; + struct wm8804_priv *wm8804; - ret = pll_factors(&pll_div, freq_out, freq_in); + wm8804 = snd_soc_codec_get_drvdata(codec); + + ret = pll_factors(&pll_div, freq_out, freq_in, + wm8804->mclk_div); if (ret) return ret; @@ -452,6 +458,7 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { struct snd_soc_codec *codec; + struct wm8804_priv *wm8804; codec = dai->codec; switch (div_id) { @@ -459,6 +466,10 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, snd_soc_update_bits(codec, WM8804_PLL5, 0x30, (div & 0x3) << 4); break; + case WM8804_MCLK_DIV: + wm8804 = snd_soc_codec_get_drvdata(codec); + wm8804->mclk_div = div; + break; default: dev_err(dai->dev, "Unknown clock divider: %d\n", div_id); return -EINVAL; diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index 8ec14f5..e72d4f4 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -57,5 +57,9 @@ #define WM8804_CLKOUT_SRC_OSCCLK 4 #define WM8804_CLKOUT_DIV 1 +#define WM8804_MCLK_DIV 2 + +#define WM8804_MCLKDIV_256FS 0 +#define WM8804_MCLKDIV_128FS 1 #endif /* _WM8804_H */ -- 1.7.9.5

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[alsa-devel] Got headphone out working via hda-verb setting on MacPro1, 1
by Nicolai Beuermann 22 Jan '14

22 Jan '14

Hello, hda-analyzer told me to write a note to this list when settings are successful. My setup: MacPro 1,1 Kernel 3.10.17 (gentoo sources) ALSA k3.10.17 Had to set /usr/bin/hda-verb /dev/snd/hwC0D0 0x18 SET_PIN_WIDGET_CONTROL 0xc1 to get the headphone out at the front working. Wrote a bit (in german) about it: http://www.nico-beuermann.de/blogging/archives/117-Hier-gibt-es-was-auf-die… regards Nicolai Beuermann -- mailto: mail(a)nico-beuermann.de online: http://www.nico-beuermann.de GnuPG Fingerabdruck: 56DA 4E32 3A4A 52AC B769 DFC2 BF3E 9805 09BB 4259

2 1

[alsa-devel] [PATCH v2 0/4] ASoC: tlv320aic32x4: DT support
by Markus Pargmann 22 Jan '14

22 Jan '14

Hi, The series adds basic DT support for the TI tlv320aic32x4 codec. v2 adds a new patch to support regulators. The codec now decides on the given regulators if LDO for AVDD should be enabled. 'ti,ldo-enable' was removed. Regards, Markus Changes in v2: - Regulator support - Fix documentation for gpio property Markus Pargmann (4): ASoC: tlv320aic32x4: Use signed int mixer controls ASoC: tlv320aic32x4: DT support ASoC: tlv320aic32x4: Support for master clock ASoC: tlv320aic32x4: Support for regulators .../devicetree/bindings/sound/tlv320aic32x4.txt | 28 +++++ sound/soc/codecs/tlv320aic32x4.c | 126 +++++++++++++++++++-- 2 files changed, 144 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt -- 1.8.5.2

3 8

[alsa-devel] [PATCH] ALSA: cs46xx: Fix memory leak at destructor
by Takashi Iwai 22 Jan '14

22 Jan '14

The release of module object itself was forgotten. Spotted by COVERIY CID 1162828. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/cs46xx/cs46xx_lib.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index f18e5878f58b..062398ec5335 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -369,6 +369,7 @@ static void free_module_desc(struct dsp_module_desc *module) kfree(module->segments[i].data); kfree(module->segments); } + kfree(module); } /* firmware binary format: -- 1.8.5.2

1 0

[alsa-devel] [PATCH] alsa-utils: fix timespec to msec conversion
by Olivier Langlois 22 Jan '14

22 Jan '14

There is 1M nsec in a msec. Signed-off-by: Olivier Langlois <olivier(a)trillion01.com> --- aplay/aplay.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index 69e8bda..33a8ed5 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -1509,7 +1509,7 @@ static void xrun(void) timermsub(&now, &tstamp, &diff); fprintf(stderr, _("%s!!! (at least %.3f ms long)\n"), stream == SND_PCM_STREAM_PLAYBACK ? _("underrun") : _("overrun"), - diff.tv_sec * 1000 + diff.tv_nsec / 10000000.0); + diff.tv_sec * 1000 + diff.tv_nsec / 1000000.0); #else fprintf(stderr, "%s !!!\n", _("underrun")); #endif -- 1.8.5.3

2 1

[alsa-devel] [PATCH - my module 1/1] control_hw: fix potential memory leak
by vanyacpp＠gmail.com 22 Jan '14

22 Jan '14

From: Ivan Sorokin <vanyacpp(a)gmail.com> Signed-off-by: Ivan Sorokin <vanyacpp(a)gmail.com> diff --git a/src/control/control_hw.c b/src/control/control_hw.c index 148097f..dfc9dcd 100644 --- a/src/control/control_hw.c +++ b/src/control/control_hw.c @@ -240,8 +240,10 @@ static int snd_ctl_hw_elem_tlv(snd_ctl_t *handle, int op_flag, return -errno; } if (op_flag == 0) { - if (xtlv->tlv[1] + 2 * sizeof(unsigned int) > tlv_size) + if (xtlv->tlv[1] + 2 * sizeof(unsigned int) > tlv_size) { + free(xtlv); return -EFAULT; + } memcpy(tlv, xtlv->tlv, xtlv->tlv[1] + 2 * sizeof(unsigned int)); } free(xtlv); -- 1.8.1.2

2 1

[alsa-devel] [ISSUE] crackles and clicks
by Eric F 22 Jan '14

22 Jan '14

Hello, I tried my DAC (Tenor TE8802 chipset) in USB on Archlinux (LTS 3.10.26). I can hear regularly crackles ans clicks. I've tried all available USB ports, other kernels (3.12.6, 3.12.7), other distro (ubuntu), without luck... The DAC is Teac UD-H01. here is lsusb verbose : http://ix.io/9Q4 Feel free to ask for more precisions, I'll give you all you need. I thank you very much. Eric

4 6

[alsa-devel] HD audio problem (fwd)
by Russ Whitaker 22 Jan '14

22 Jan '14

Still not heard from you. WHY? ---------- Forwarded message ---------- Date: Mon, 30 Dec 2013 20:20:31 -0800 (PST) From: Russ Whitaker <russ(a)ashlandhome.net> To: alsa-devel(a)alsa-project.org Subject: HD audio problem (fwd) Have not heard back from you. What should I do? ---------- Forwarded message ---------- Date: Thu, 19 Dec 2013 23:48:19 -0800 (PST) From: Russ Whitaker <russ(a)ashlandhome.net> To: alsa-devel(a)alsa-project.org Subject: HD audio problem Hi Bought a new mini-itx mb: ECS KBN-I On board sound chip: VT1705CF Installed kernel linux-3.10.24 Otherwise distro is Slackware64 14.1 mplayer plays .mp3 files just fine. No sound from video clips that use Adobe flashplayer, and I get this error message: ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave I think the problem is: card0 and card1 are interchanged as card0 is listed as the HDMI video output. How to fixit, please. Thank you, Russ

2 2

[alsa-devel] [PATCH 2/2] ASoC: fsl_ssi: Do not print 'baud clock' error message all the time
by Fabio Estevam 21 Jan '14

21 Jan '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Currently everytime we get the following message on boot: fsl-ssi-dai 202c000.ssi: could not get baud clock: -2 This is not really useful information to get on every boot, so make it a debug message instead. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 7b1772d..3155611 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1407,7 +1407,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) */ ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); if (IS_ERR(ssi_private->baudclk)) - dev_warn(&pdev->dev, "could not get baud clock: %ld\n", + dev_dbg(&pdev->dev, "could not get baud clock: %ld\n", PTR_ERR(ssi_private->baudclk)); else clk_prepare_enable(ssi_private->baudclk); -- 1.8.1.2

2 3

[alsa-devel] [PATCH] ASoC: fsl: use snd_soc_dai_init_dma_data()
by Xiubo Li 21 Jan '14

21 Jan '14

Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 4d075f1..8f36f49 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -911,8 +911,8 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai) { struct fsl_spdif_priv *spdif_private = snd_soc_dai_get_drvdata(dai); - dai->playback_dma_data = &spdif_private->dma_params_tx; - dai->capture_dma_data = &spdif_private->dma_params_rx; + snd_soc_dai_init_dma_data(dai, &spdif_private->dma_params_tx, + &spdif_private->dma_params_rx); snd_soc_add_dai_controls(dai, fsl_spdif_ctrls, ARRAY_SIZE(fsl_spdif_ctrls)); -- 1.8.4

3 2

[alsa-devel] [PATCH 1/3] ASoC: wm5102: Improve EQ coefficient controls
by Charles Keepax 21 Jan '14

21 Jan '14

The EQ coefficient binary controls overlapped with the volume controls for the B4 and B5 volumes, which were controllable from either the coefficient control or the volume control itself. This patch adds controls for the mode and moves the coefficient control to only cover the coefficients. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 17 ++++++++--------- 1 files changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a08e8bf..ac19f62 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -685,15 +685,8 @@ 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_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21, - ARIZONA_EQ1_ENA_MASK), -SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21, - ARIZONA_EQ2_ENA_MASK), -SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21, - ARIZONA_EQ3_ENA_MASK), -SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21, - ARIZONA_EQ4_ENA_MASK), - +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 18), +SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), 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, @@ -705,6 +698,8 @@ 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, 18), +SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), 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, @@ -716,6 +711,8 @@ 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, 18), +SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), 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, @@ -727,6 +724,8 @@ 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, 18), +SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), 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

2 3

[alsa-devel] [PATCH] ASoC: wm5110: Extend SYSCLK patch file for rev D
by Charles Keepax 21 Jan '14

21 Jan '14

Latest evaluation of the the device has given some patch file additions for improved performance. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 48 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 48 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 34fbb03..92f1b40 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -43,6 +43,54 @@ static const struct reg_default wm5110_sysclk_revd_patch[] = { { 0x3133, 0x1201 }, { 0x3183, 0x1501 }, { 0x31D3, 0x1401 }, + { 0x0049, 0x01ea }, + { 0x004a, 0x01f2 }, + { 0x0057, 0x01e7 }, + { 0x0058, 0x01fb }, + { 0x33ce, 0xc4f5 }, + { 0x33cf, 0x1361 }, + { 0x33d0, 0x0402 }, + { 0x33d1, 0x4700 }, + { 0x33d2, 0x026d }, + { 0x33d3, 0xff00 }, + { 0x33d4, 0x026d }, + { 0x33d5, 0x0101 }, + { 0x33d6, 0xc4f5 }, + { 0x33d7, 0x0361 }, + { 0x33d8, 0x0402 }, + { 0x33d9, 0x6701 }, + { 0x33da, 0xc4f5 }, + { 0x33db, 0x136f }, + { 0x33dc, 0xc4f5 }, + { 0x33dd, 0x134f }, + { 0x33de, 0xc4f5 }, + { 0x33df, 0x131f }, + { 0x33e0, 0x026d }, + { 0x33e1, 0x4f01 }, + { 0x33e2, 0x026d }, + { 0x33e3, 0xf100 }, + { 0x33e4, 0x026d }, + { 0x33e5, 0x0001 }, + { 0x33e6, 0xc4f5 }, + { 0x33e7, 0x0361 }, + { 0x33e8, 0x0402 }, + { 0x33e9, 0x6601 }, + { 0x33ea, 0xc4f5 }, + { 0x33eb, 0x136f }, + { 0x33ec, 0xc4f5 }, + { 0x33ed, 0x134f }, + { 0x33ee, 0xc4f5 }, + { 0x33ef, 0x131f }, + { 0x33f0, 0x026d }, + { 0x33f1, 0x4e01 }, + { 0x33f2, 0x026d }, + { 0x33f3, 0xf000 }, + { 0x33f6, 0xc4f5 }, + { 0x33f7, 0x1361 }, + { 0x33f8, 0x0402 }, + { 0x33f9, 0x4600 }, + { 0x33fa, 0x026d }, + { 0x33fb, 0xfe00 }, }; static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w, -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Add info print to note that the DSP has shutdown
by Charles Keepax 21 Jan '14

21 Jan '14

It can be useful for debugging purposes to see at what point the DSP has powered down, so add a message to inform us of this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 4fbcab6..a15f75b 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1637,6 +1637,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, list_del(&alg_region->list); kfree(alg_region); } + + adsp_info(dsp, "Shutdown complete\n"); break; default: -- 1.7.2.5

2 1

[alsa-devel] [alsa-plugins] [PATCH 2/2] Allow to specify a client name as parameters
by Corfu, ValentinX 21 Jan '14

21 Jan '14

0001-Modify-jack-client-name.patch is spitted in: 0001-Allow-number-of-channels.patch and 0002-Allow-to-specify-a-client-name-as-parameters-in-the-.patch >From 8d6d397306bdd09c6c3fae83543e124c89443e94 Mon Sep 17 00:00:00 2001 From: Valentin Corfu <valentinx.corfu(a)intel.com> Date: Tue, 21 Jan 2014 11:32:51 -0500 Subject: [PATCH 2/2] Allow to specify a client name as parameters The current jack client name contains the process id of the application providing the audio samples. This leads to unpredictable jack client names which makes handling of the connections by a controlling application very hard. This modification now, allows to specify a client name as parameters in the configuration file. The implementation is backward compatible and simply adds a new configuration option to the plugin. Signed-off-by: Valentin Corfu <valentinx.corfu(a)intel.com> diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index 4306a13..61ac197 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -32,6 +32,11 @@ typedef enum _jack_format { SND_PCM_JACK_FORMAT_RAW } snd_pcm_jack_format_t; +typedef enum _naming_scheme { + SND_PCM_JACK_STD_NAME, + SND_PCM_JACK_USER_DEFINED_NAME +} snd_pcm_jack_naming_t; + typedef struct { snd_pcm_ioplug_t io; @@ -325,7 +330,8 @@ static int make_nonblock(int fd) static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, snd_config_t *playback_conf, snd_config_t *capture_conf, - snd_pcm_stream_t stream, int mode) + snd_pcm_stream_t stream, int mode, + snd_pcm_jack_naming_t naming) { snd_pcm_jack_t *jack; int err; @@ -356,11 +362,19 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, return -EINVAL; } - if (snprintf(jack_client_name, sizeof(jack_client_name), "alsa-jack.%s%s.%d.%d", name, - stream == SND_PCM_STREAM_PLAYBACK ? "P" : "C", getpid(), num++) - >= (int)sizeof(jack_client_name)) { - fprintf(stderr, "%s: WARNING: JACK client name '%s' truncated to %d characters, might not be unique\n", - __func__, jack_client_name, (int)strlen(jack_client_name)); + if (naming == SND_PCM_JACK_STD_NAME ) { + if (snprintf(jack_client_name, sizeof(jack_client_name), "alsa-jack.%s%s.%d.%d", name, + stream == SND_PCM_STREAM_PLAYBACK ? "P" : "C", getpid(), num++) + >= (int)sizeof(jack_client_name)) { + fprintf(stderr, "%s: WARNING: JACK client name '%s' truncated to %d characters, might not be unique\n", + __func__, jack_client_name, (int)strlen(jack_client_name)); + } + } else { + if (snprintf(jack_client_name, sizeof(jack_client_name), "%s", name) + >= (int)sizeof(jack_client_name)) { + fprintf(stderr, "%s: WARNING: JACK client name '%s' truncated to %d characters, might not be unique\n", + __func__, jack_client_name, (int)strlen(jack_client_name)); + } } jack->client = jack_client_new(jack_client_name); @@ -417,6 +431,8 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) int err; long num_channels = 0; long chan; + const char *local_name = name; + snd_pcm_jack_naming_t naming = SND_PCM_JACK_STD_NAME; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); @@ -425,6 +441,11 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0 || strcmp(id, "hint") == 0) continue; + if (strcmp(id, "name") == 0) { + snd_config_get_string(n, &local_name); + naming = SND_PCM_JACK_USER_DEFINED_NAME; + continue; + } if (strcmp(id, "nchan") == 0) { snd_config_get_integer(n, &num_channels); continue; @@ -449,7 +470,10 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) return -EINVAL; } - if ( (num_channels != 0) && (playback_conf == NULL) && (capture_conf == NULL) ) { + if ( (naming == SND_PCM_JACK_USER_DEFINED_NAME) + && (num_channels != 0) + && (playback_conf == NULL) + && (capture_conf == NULL) ) { err = snd_config_make_compound(&playback_conf, "playback_ports", 0); if (err) { SNDERR("Error during snd_config_make_compound(p): %s\n", strerror(err)); @@ -486,7 +510,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) return err; } } - err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + err = snd_pcm_jack_open(pcmp, local_name, playback_conf, capture_conf, stream, mode, naming); if (err) { SNDERR("Error during snd_pcm_jack_open: %s\n", strerror(err)); return err; @@ -513,7 +537,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) return err; } } else { - err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + err = snd_pcm_jack_open(pcmp, local_name, playback_conf, capture_conf, stream, mode, naming); } return err; -- 1.8.1.4 -------------------------------------------------------------- Intel Shannon Limited Registered in Ireland Registered Office: Collinstown Industrial Park, Leixlip, County Kildare Registered Number: 308263 Business address: Dromore House, East Park, Shannon, Co. Clare This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

2 1

[alsa-devel] [alsa-plugins] [PATCH 1/2] Allow number of channels
by Corfu, ValentinX 21 Jan '14

21 Jan '14

0001-Modify-jack-client-name.patch is spitted in: 0001-Allow-number-of-channels.patch and 0002-Allow-to-specify-a-client-name-as-parameters-in-the-.patch >From d247e4d675dd68c6f683a8c09ff792172396e870 Mon Sep 17 00:00:00 2001 From: Valentin Corfu <valentinx.corfu(a)intel.com> Date: Tue, 21 Jan 2014 10:10:52 -0500 Subject: [PATCH 1/2] Allow number of channels The modification allows to specify the number of channels as parameters in the configuration file The given number of channels are then created but not connected. Signed-off-by: Valentin Corfu <valentinx.corfu(a)intel.com> diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index c51fe84..4306a13 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -415,6 +415,8 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) snd_config_t *playback_conf = NULL; snd_config_t *capture_conf = NULL; int err; + long num_channels = 0; + long chan; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); @@ -423,6 +425,10 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0 || strcmp(id, "hint") == 0) continue; + if (strcmp(id, "nchan") == 0) { + snd_config_get_integer(n, &num_channels); + continue; + } if (strcmp(id, "playback_ports") == 0) { if (snd_config_get_type(n) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for %s", id); @@ -443,7 +449,72 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) return -EINVAL; } - err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + if ( (num_channels != 0) && (playback_conf == NULL) && (capture_conf == NULL) ) { + err = snd_config_make_compound(&playback_conf, "playback_ports", 0); + if (err) { + SNDERR("Error during snd_config_make_compound(p): %s\n", strerror(err)); + return err; + } + err = snd_config_make_compound(&capture_conf, "capture_ports", 0); + if (err) { + SNDERR("Error during snd_config_make_compound(c): %s\n", strerror(err)); + return err; + } + char value[4]; + for (chan=0; chan<num_channels; ++chan) { + snprintf(value, sizeof(value), "%d", chan); + snd_config_t *newLeafPlayback; + snd_config_t *newLeafCapture; + err = snd_config_imake_integer(&newLeafPlayback, value, chan); + if (err) { + SNDERR("Error during snd_config_imake_string(p): %s\n", strerror(err)); + return err; + } + err = snd_config_add(playback_conf, newLeafPlayback); + if (err) { + SNDERR("Error during snd_config_add(p): %s\n", strerror(err)); + return err; + } + err = snd_config_imake_integer(&newLeafCapture, value, chan); + if (err) { + SNDERR("Error during snd_config_imake_string(c): %s\n", strerror(err)); + return err; + } + err = snd_config_add(capture_conf, newLeafCapture); + if (err) { + SNDERR("Error during snd_config_add(c): %s\n", strerror(err)); + return err; + } + } + err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + if (err) { + SNDERR("Error during snd_pcm_jack_open: %s\n", strerror(err)); + return err; + } + + err = snd_config_delete_compound_members(playback_conf); + if (err) { + SNDERR("Error during snd_config_delete_compound_members(p): %s\n", strerror(err)); + return err; + } + err = snd_config_delete_compound_members(capture_conf); + if (err) { + SNDERR("Error during snd_config_delete_compound_members(c): %s\n", strerror(err)); + return err; + } + err = snd_config_delete(playback_conf); + if (err) { + SNDERR("Error during snd_config_delete(p): %s\n", strerror(err)); + return err; + } + err = snd_config_delete(capture_conf); + if (err) { + SNDERR("Error during snd_config_delete(c): %s\n", strerror(err)); + return err; + } + } else { + err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + } return err; } -- 1.8.1.4 -------------------------------------------------------------- Intel Shannon Limited Registered in Ireland Registered Office: Collinstown Industrial Park, Leixlip, County Kildare Registered Number: 308263 Business address: Dromore House, East Park, Shannon, Co. Clare This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Resume mixer values properly
by Takashi Iwai 21 Jan '14

21 Jan '14

Implement reset_resume callback so that the mixer values are properly restored. Still no boot quirks are called, so it might not work well on some devices. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 18 ++++++++-- sound/usb/mixer.c | 99 ++++++++++++++++++++++++++++++++++++++++++++----------- sound/usb/mixer.h | 7 ++-- 3 files changed, 99 insertions(+), 25 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index d979050e6a6a..025224136129 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -691,12 +691,12 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) } list_for_each_entry(mixer, &chip->mixer_list, list) - snd_usb_mixer_inactivate(mixer); + snd_usb_mixer_suspend(mixer); return 0; } -static int usb_audio_resume(struct usb_interface *intf) +static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) { struct snd_usb_audio *chip = usb_get_intfdata(intf); struct usb_mixer_interface *mixer; @@ -711,7 +711,7 @@ static int usb_audio_resume(struct usb_interface *intf) * we just notify and restart the mixers */ list_for_each_entry(mixer, &chip->mixer_list, list) { - err = snd_usb_mixer_activate(mixer); + err = snd_usb_mixer_resume(mixer, reset_resume); if (err < 0) goto err_out; } @@ -723,9 +723,20 @@ static int usb_audio_resume(struct usb_interface *intf) err_out: return err; } + +static int usb_audio_resume(struct usb_interface *intf) +{ + return __usb_audio_resume(intf, false); +} + +static int usb_audio_reset_resume(struct usb_interface *intf) +{ + return __usb_audio_resume(intf, true); +} #else #define usb_audio_suspend NULL #define usb_audio_resume NULL +#define usb_audio_reset_resume NULL #endif /* CONFIG_PM */ static struct usb_device_id usb_audio_ids [] = { @@ -747,6 +758,7 @@ static struct usb_driver usb_audio_driver = { .disconnect = usb_audio_disconnect, .suspend = usb_audio_suspend, .resume = usb_audio_resume, + .reset_resume = usb_audio_reset_resume, .id_table = usb_audio_ids, .supports_autosuspend = 1, }; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 44b0ba4feab3..aa9bc19aae68 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2299,26 +2299,6 @@ requeue: } } -/* stop any bus activity of a mixer */ -void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer) -{ - usb_kill_urb(mixer->urb); - usb_kill_urb(mixer->rc_urb); -} - -int snd_usb_mixer_activate(struct usb_mixer_interface *mixer) -{ - int err; - - if (mixer->urb) { - err = usb_submit_urb(mixer->urb, GFP_NOIO); - if (err < 0) - return err; - } - - return 0; -} - /* create the handler for the optional status interrupt endpoint */ static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer) { @@ -2417,3 +2397,82 @@ void snd_usb_mixer_disconnect(struct list_head *p) usb_kill_urb(mixer->urb); usb_kill_urb(mixer->rc_urb); } + +#ifdef CONFIG_PM +/* stop any bus activity of a mixer */ +static void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer) +{ + usb_kill_urb(mixer->urb); + usb_kill_urb(mixer->rc_urb); +} + +static int snd_usb_mixer_activate(struct usb_mixer_interface *mixer) +{ + int err; + + if (mixer->urb) { + err = usb_submit_urb(mixer->urb, GFP_NOIO); + if (err < 0) + return err; + } + + return 0; +} + +int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer) +{ + snd_usb_mixer_inactivate(mixer); + return 0; +} + +static int restore_mixer_value(struct usb_mixer_elem_info *cval) +{ + int c, err, idx; + + if (cval->cmask) { + idx = 0; + for (c = 0; c < MAX_CHANNELS; c++) { + if (!(cval->cmask & (1 << c))) + continue; + if (cval->cached & (1 << c)) { + err = set_cur_mix_value(cval, c + 1, idx, + cval->cache_val[idx]); + if (err < 0) + return err; + } + idx++; + } + } else { + /* master */ + if (cval->cached) { + err = set_cur_mix_value(cval, 0, 0, *cval->cache_val); + if (err < 0) + return err; + } + } + + return 0; +} + +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) +{ + struct usb_mixer_elem_info *cval; + int id, err; + + /* FIXME: any mixer quirks? */ + + if (reset_resume) { + /* restore cached mixer values */ + for (id = 0; id < MAX_ID_ELEMS; id++) { + for (cval = mixer->id_elems[id]; cval; + cval = cval->next_id_elem) { + err = restore_mixer_value(cval); + if (err < 0) + return err; + } + } + } + + return snd_usb_mixer_activate(mixer); +} +#endif diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index aab80df201bd..73b1f649447b 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -63,8 +63,6 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid); int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set); -void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer); -int snd_usb_mixer_activate(struct usb_mixer_interface *mixer); int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer, struct snd_kcontrol *kctl); @@ -72,4 +70,9 @@ int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer, int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *_tlv); +#ifdef CONFIG_PM +int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer); +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume); +#endif + #endif /* __USBMIXER_H */ -- 1.8.5.2

1 0

Re: [alsa-devel] [PATCH] ASoC: da9055: Partially fix device
by Opensource [Adam Thomson] 21 Jan '14

21 Jan '14

On Wed, 8 Jan 2014 20:38:55 +0000, Mark Brown wrote: > It has never been possible to even load the da9055 driver since both the > MFD and CODEC drivers register the I2C device "da9055". Fix this by using > the usual pattern for MFDs with multiple I2C addresses and having the CODEC > driver reference an I2C dummy registered by the MFD. > Since I don't know which I2C address to use for the CODEC a FIXME has been > left in the MFD, this doesn't make anything any worse since the device has > never been able to load in the first place. What you're saying here is incorrect, as previously discussed at length in this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-September/066027.… The two drivers did work together fine until someone in 3.12 changed the PMIC I2C Id back to just 'da9055' and broke it again. Also, I wanted to add a simple patch to tidy up the I2C Id names and avoid this but you would not agree. The codec part can be standalone and by doing what you've done here means that customers would need to now add additional, unnecessary code to be able to use just the codec driver. This is not right to me.

2 11

[alsa-devel] [PATCH] ASoC: omap: Make RX51 depend on GPIOLIB due to jack usage
by Mark Brown 21 Jan '14

21 Jan '14

From: Mark Brown <broonie(a)linaro.org> Since the GPIO jacks are only supported if gpiolib is built and fail to compile otherwise add a build depedency. This is unlikely to have any practical impact outside of coverage testing. Reported-by: Russell King <linux(a)arm.linux.org.uk> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/omap/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 4a07f7179690..22ad9c5654b5 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -30,6 +30,7 @@ config SND_OMAP_SOC_RX51 select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X select SND_SOC_TPA6130A2 + depends on GPIOLIB help Say Y if you want to add support for SoC audio on Nokia RX-51 hardware. This is also known as Nokia N900 product. -- 1.8.5.2

2 1

[alsa-devel] ALSA: hda - Fix possible races in HDMI driver - lockup on shutdown when radeon.audio=1 after using audacity
by Arthur Marsh 21 Jan '14

21 Jan '14

I have had reproducible lock-ups on shut-down (at the shutting down ALSA stage) of my AMD64 machine (Asus M3A78Pro motherboard, BIOS 1701 01/27/2011, CPU AMD Athlon(tm) II X4 640 Processor) running the 64 bit Linux kernel more recent than 3.12 when *both* radeon.audio=1 was set and I had been running audacity 2.0.5. (iommu=noaperture is also set). The problem was reproducible with the stock Debian kernel linux-image-3.13-rc6-amd64 version 3.13~rc6-1~exp1. The machine is using an ATI/AMD 3850HD video card with a DVI cable to a DVI input on my monitor, and the default audio device is the motherboard's on-board audio device: 00:14.2 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] SBx00 Azalia (Intel HDA) 01:00.0 VGA compatible controller: Advanced Micro Devices, Inc. [AMD/ATI] RV670 [Radeon HD 3690/3850] 01:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] RV670/680 HDMI Audio [Radeon HD 3690/3800 Series] $ git bisect bad cbbaa603a03cc46681e24d6b2804b62fde95a2af is the first bad commit commit cbbaa603a03cc46681e24d6b2804b62fde95a2af Author: Takashi Iwai <tiwai(a)suse.de> Date: Thu Oct 17 18:03:24 2013 +0200 ALSA: hda - Fix possible races in HDMI driver Some per_pin fields and ELD contents might be changed dynamically in multiple ways where the concurrent accesses are still opened in the current code. This patch fixes such possible races by using eld->lock in appropriate places. Reported-by: Anssi Hannula <anssi.hannula(a)iki.fi> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> :040000 040000 0c29281f82a3ebd9a704d481114f9cfcefea07c8 d71fd101125cd29a628cb5e66c7ee4f56e28329b M sound When running audacity from the command line there was the following output: ALSA lib pcm_dsnoop.c:618:(snd_pcm_dsnoop_open) unable to open slave ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave Expression 'stream->playback.pcm' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 Expression 'stream->playback.pcm' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 Expression 'stream->playback.pcm' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 I am happy to supply further information or run further tests to help in isolating the problem and verifying a solution. Arthur.

2 5

[alsa-devel] [PATCH] ASoC: wm8804: Allow control of master clock divider in PLLgeneration
by Daniel Matuschek 20 Jan '14

20 Jan '14

WM8804 can run with PLL frequencies of 256xfs and 128xfs for most sample rates. At 192kHz only 128xfs is supported. The existing driver selects 128xfs automatically for some lower samples rates. By using an additional mclk_div divider, it is now possible to control the behaviour. This allows using 256xfs PLL frequency on all sample rates up to 96kHz. It should allow lower jitter and better signal quality. The behavior has to be controlled by the sound card driver, because some sample frequency share the same setting. e.g. 192kHz and 96kHz use 24.576MHz master clock. The only difference is the MCLK divider. Signed-off-by: Daniel Matuschek <daniel(a)matuschek.net> Acked-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Tested-by: Florian Meier <florian.meier(a)koalo.de> --- sound/soc/codecs/wm8804.c | 17 ++++++++++++++--- sound/soc/codecs/wm8804.h | 4 ++++ 2 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 9bc8206..4163786 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -63,6 +63,7 @@ struct wm8804_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; + int mclk_div; }; static int txsrc_get(struct snd_kcontrol *kcontrol, @@ -318,7 +319,7 @@ static struct { #define FIXED_PLL_SIZE ((1ULL << 22) * 10) static int pll_factors(struct pll_div *pll_div, unsigned int target, - unsigned int source) + unsigned int source, unsigned int mclk_div) { u64 Kpart; unsigned long int K, Ndiv, Nmod, tmp; @@ -330,7 +331,8 @@ static int pll_factors(struct pll_div *pll_div, unsigned int target, */ for (i = 0; i < ARRAY_SIZE(post_table); i++) { tmp = target * post_table[i].div; - if (tmp >= 90000000 && tmp <= 100000000) { + if ((tmp >= 90000000 && tmp <= 100000000) && + (mclk_div == post_table[i].mclkdiv)) { pll_div->freqmode = post_table[i].freqmode; pll_div->mclkdiv = post_table[i].mclkdiv; target *= post_table[i].div; @@ -387,8 +389,12 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, } else { int ret; struct pll_div pll_div; + struct wm8804_priv *wm8804; - ret = pll_factors(&pll_div, freq_out, freq_in); + wm8804 = snd_soc_codec_get_drvdata(codec); + + ret = pll_factors(&pll_div, freq_out, freq_in, + wm8804->mclk_div); if (ret) return ret; @@ -452,6 +458,7 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { struct snd_soc_codec *codec; + struct wm8804_priv *wm8804; codec = dai->codec; switch (div_id) { @@ -459,6 +466,10 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, snd_soc_update_bits(codec, WM8804_PLL5, 0x30, (div & 0x3) << 4); break; + case WM8804_MCLK_DIV: + wm8804 = snd_soc_codec_get_drvdata(codec); + wm8804->mclk_div = div; + break; default: dev_err(dai->dev, "Unknown clock divider: %d\n", div_id); return -EINVAL; diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index 8ec14f5..e72d4f4 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -57,5 +57,9 @@ #define WM8804_CLKOUT_SRC_OSCCLK 4 #define WM8804_CLKOUT_DIV 1 +#define WM8804_MCLK_DIV 2 + +#define WM8804_MCLKDIV_256FS 0 +#define WM8804_MCLKDIV_128FS 1 #endif /* _WM8804_H */ -- 1.8.3.2

1 0

[alsa-devel] ASoC updates for v3.14
by Mark Brown 20 Jan '14

20 Jan '14

The following changes since commit 258753361534a40ad7180c742da813fc659e427b: Merge branch 'for-next' into for-linus (2014-01-20 10:20:14 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.14-3 for you to fetch changes up to ac6d7c48e3a5e82e876114b9ca0b489e96997743: Merge remote-tracking branch 'asoc/topic/compress' into asoc-next (2014-01-20 11:50:41 +0000) ---------------------------------------------------------------- ASoC: Last updates for the merge window A couple more fixes plus some extensions to DPCM for use with compressed audio from Liam which arrived just after my previous pull request. ---------------------------------------------------------------- Arun Shamanna Lakshmi (1): ASoC: dapm: Fix double prefix addition Liam Girdwood (2): ASoC: DPCM: make some DPCM API calls non static for compressed usage ASoC: compress: Add suport for DPCM into compressed audio Mark Brown (3): Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/compress' into asoc-next Xiubo Li (1): ASoC: core: Fix possible NULL pointer dereference of pcm->config include/sound/soc-dpcm.h | 22 +++ include/sound/soc.h | 1 + sound/soc/soc-compress.c | 301 +++++++++++++++++++++++++++++++++- sound/soc/soc-dapm.c | 11 +- sound/soc/soc-generic-dmaengine-pcm.c | 9 +- sound/soc/soc-pcm.c | 29 ++-- 6 files changed, 347 insertions(+), 26 deletions(-)

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[alsa-devel] [PATCH v3 01/16] ALSA: Oxygen: Add the separate SPI waiting function
by Roman Volkov 20 Jan '14

20 Jan '14

This function performs waiting when the SPI bus completes a transaction. Timeout error checking introduced and the timeout increased to 400 from 40. Signed-off-by: Roman Volkov <v1ron(a)mail.ru> --- sound/pci/oxygen/oxygen_io.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 521eae4..a917f37 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -194,6 +194,21 @@ void oxygen_write_ac97_masked(struct oxygen *chip, unsigned int codec, } EXPORT_SYMBOL(oxygen_write_ac97_masked); +static int oxygen_wait_spi(struct oxygen *chip) +{ + /* Higher timeout to be sure: 400 microseconds */ + unsigned int count; + for (count = 100; count > 0; count--) { + if ((oxygen_read8(chip, OXYGEN_SPI_CONTROL) & + OXYGEN_SPI_BUSY) == 0) + return 0; + udelay(4); + --count; + } + snd_printk(KERN_ERR "Oxygen: SPI wait timeout\n"); + return -EIO; +} + void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data) { unsigned int count; -- 1.7.10.4

2 23

[alsa-devel] Thinkpad T520 (Low Volume)
by Raymond Yau 20 Jan '14

20 Jan '14

> 2014/1/16 Axel Braun <axel.braun(a)gmx.de> > same problem on a ThinkPad T520, sound volume is way too low - watching movie via loudspeakers is fairly impossible. > Any hint you could give me? > > post the output of alsa-info.sh , there is a bug in alsa-driver which treat > dock headphone and headphone to support surround 4.0 you get low volume because Amp Out vals are 0x3d instead of 0x4a but speaker volumes are at 100% (0dB) Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 74 [100%] [0.00dB] [on] Front Right: Playback 74 [100%] [0.00dB] [on] Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CX20590 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x3d 0x3d] https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=ca2… thinkpad t520 was supported with CXT5066_AUTO which use cxt5066_update_speaker which check port a and port d Do this mean thinkpad don't support surround 4.0 by dock headphone and headphone ? https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… ALSA: hda/conexant - Set up the missing docking-station pins [ 9.956636] ALSA hda_auto_parser.c:393 autoconfig: line_outs=1 (0x1f/0x0/0x0/0x0/0x0) type:speaker [ 9.956639] ALSA hda_auto_parser.c:397 speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 9.956641] ALSA hda_auto_parser.c:401 hp_outs=2 (0x1c/0x19/0x0/0x0/0x0) [ 9.956643] ALSA hda_auto_parser.c:402 mono: mono_out=0x0 [ 9.956644] ALSA hda_auto_parser.c:406 inputs: [ 9.956646] ALSA hda_auto_parser.c:410 Mic=0x1b [ 9.956647] ALSA hda_auto_parser.c:410 Internal Mic=0x23 [ 9.956649] ALSA hda_auto_parser.c:410 Dock Mic=0x1a the driver select this as best config ==> Best config: lo_type=1, wired=1, mio=1 multi_outs = 1f/0/0/0 : 10/0/0/0 (type SP) out path: depth=2 :10:1f hp_outs = 19/1c/0/0 : 11/11/0/0 hp path: depth=2 :11:19 hp path: depth=2 :11:1c https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… ALSA: hda - Fix multi-io channel mode management this patch change the the channel_count from 2 to 4 since there are two headphone + if (cfg->line_out_type != AUTO_PIN_HP_OUT) + spec->const_channel_count = max(spec->const_channel_count, + cfg->hp_outs * 2); if you don't have docking station , the workaround is to remove those pin fix of node 0x1a and 0x1c

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[alsa-devel] valgrind error in snd_tlv_get_dB_range
by Ivan Sorokin 20 Jan '14

20 Jan '14

One program (qasmixer -- alsa mixer) on my system prints lots of valgrind errors. The first one is this: ==25716== Conditional jump or move depends on uninitialised value(s) ==25716== at 0x4E72307: snd_tlv_get_dB_range (tlv.c:170) ==25716== by 0x4E8A746: get_dB_range (simple_none.c:1162) ==25716== by 0x4E8A7E7: get_dB_range_ops (simple_none.c:1176) ==25716== by 0x4E85590: snd_mixer_selem_get_playback_dB_range (simple.c:298) ==25716== by 0x472B2B: QSnd::Mixer_Simple_Elem::set_snd_mixer_selem_id(_snd_mixer_selem_id*) (mixer_simple_elem.cpp:161) ==25716== by 0x4724B8: QSnd::Mixer_Simple_Elem::Mixer_Simple_Elem(QObject*, _snd_mixer*, _snd_mixer_selem_id*) (mixer_simple_elem.cpp:31) ==25716== by 0x479F17: QSnd::Mixer_Simple::create_mixer_elem(_snd_mixer_elem*) (mixer_simple.cpp:318) ==25716== by 0x4797B3: QSnd::Mixer_Simple::create_mixer_elems() (mixer_simple.cpp:217) ==25716== by 0x4795DF: QSnd::Mixer_Simple::open(QString const&) (mixer_simple.cpp:178) ==25716== by 0x4D4754: Tray_Mixer::load_mixer() (tray_mixer.cpp:222) ==25716== by 0x4D4505: Tray_Mixer::set_mdev_setup(Tray_Mixer_MDev_Setup*) (tray_mixer.cpp:137) ==25716== by 0x4DB863: Desktop_Items::tray_mixer_create() (desktop_items.cpp:504) ==25716== Uninitialised value was created by a heap allocation ==25716== at 0x4C2CD7B: malloc (in /usr/lib/valgrind/vgpreload_memcheck-amd64-linux.so) ==25716== by 0x4E8A572: init_db_range (simple_none.c:1114) ==25716== by 0x4E8A70D: get_dB_range (simple_none.c:1159) ==25716== by 0x4E8A7E7: get_dB_range_ops (simple_none.c:1176) ==25716== by 0x4E85590: snd_mixer_selem_get_playback_dB_range (simple.c:298) ==25716== by 0x472B2B: QSnd::Mixer_Simple_Elem::set_snd_mixer_selem_id(_snd_mixer_selem_id*) (mixer_simple_elem.cpp:161) ==25716== by 0x4724B8: QSnd::Mixer_Simple_Elem::Mixer_Simple_Elem(QObject*, _snd_mixer*, _snd_mixer_selem_id*) (mixer_simple_elem.cpp:31) ==25716== by 0x479F17: QSnd::Mixer_Simple::create_mixer_elem(_snd_mixer_elem*) (mixer_simple.cpp:318) ==25716== by 0x4797B3: QSnd::Mixer_Simple::create_mixer_elems() (mixer_simple.cpp:217) ==25716== by 0x4795DF: QSnd::Mixer_Simple::open(QString const&) (mixer_simple.cpp:178) ==25716== by 0x4D4754: Tray_Mixer::load_mixer() (tray_mixer.cpp:222) ==25716== by 0x4D4505: Tray_Mixer::set_mdev_setup(Tray_Mixer_MDev_Setup*) (tray_mixer.cpp:137) I compiled libasound2 from the latest git from git://gitorious.org/alsa/alsa-lib.git with -g -O0, so you can see line numbers. As I understand the problem is that init_db_range initialize invalid rec->db_info, because snd_hctl_elem_tlv_read failed to fully initialize tlv: > tlv = malloc(tlv_size); // line 1114 > if (! tlv) > return -ENOMEM; > if (snd_hctl_elem_tlv_read(ctl, tlv, tlv_size) < 0) > goto error; > db_size = snd_tlv_parse_dB_info(tlv, tlv_size, &dbrec); > if (db_size < 0) > goto error; > rec->db_info = malloc(db_size); > if (!rec->db_info) > goto error; > memcpy(rec->db_info, dbrec, db_size); > free(tlv); I tried to trace problem in snd_hctl_elem_tlv_read, but I failed to do so due to lack of experience with alsa-lib. Could someone more experienced with alsa-lib look at the problem?

3 3

[alsa-devel] Realtec ALC898 supported?
by Haug Bürger 20 Jan '14

20 Jan '14

Hi, I would like to know if the new Realtec 89x chips are supported. I found drivers on the Realtec site but the seem to be older than the alsa-project.org drivers and I'm not sure if they support the 89x chips. thanks Haug

2 1

[alsa-devel] [PATCH] ASoC: core: Use devm_kzalloc() instead kzalloc()
by Xiubo Li 20 Jan '14

20 Jan '14

Makes the code slightly shorter Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 5bace12..bfb012f 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -388,7 +388,7 @@ int snd_dmaengine_pcm_register(struct device *dev, struct dmaengine_pcm *pcm; int ret; - pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); + pcm = devm_kzalloc(dev, sizeof(*pcm), GFP_KERNEL); if (!pcm) return -ENOMEM; @@ -408,7 +408,6 @@ int snd_dmaengine_pcm_register(struct device *dev, err_free_dma: dmaengine_pcm_release_chan(pcm); - kfree(pcm); return ret; } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_register); @@ -433,7 +432,6 @@ void snd_dmaengine_pcm_unregister(struct device *dev) snd_soc_remove_platform(platform); dmaengine_pcm_release_chan(pcm); - kfree(pcm); } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_unregister); -- 1.8.4

4 3

[alsa-devel] [PATCH] ASoC: dapm: Fix double prefix addition
by Arun Shamanna Lakshmi 17 Jan '14

17 Jan '14

The prefix for the codec driver can be used during dual identical codec usecases. However, dapm adds prefix twice for codec DAI widget in snd_soc_dapm_add_route API. This change is to avoid double prefix addition for codec DAI widget and is needed while using identical dual codecs. Signed-off-by: Songhee Baek <sbaek(a)nvidia.com> Signed-off-by: Arun Shamanna Lakshmi <aruns(a)nvidia.com> --- sound/soc/soc-dapm.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2a44fe9..dc8ff13 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2476,7 +2476,8 @@ err: } static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, - const struct snd_soc_dapm_route *route) + const struct snd_soc_dapm_route *route, + unsigned int is_prefixed) { struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w; struct snd_soc_dapm_widget *wtsource = NULL, *wtsink = NULL; @@ -2486,7 +2487,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, char prefixed_source[80]; int ret; - if (dapm->codec && dapm->codec->name_prefix) { + if (dapm->codec && dapm->codec->name_prefix && !is_prefixed) { snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s", dapm->codec->name_prefix, route->sink); sink = prefixed_sink; @@ -2614,7 +2615,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { - r = snd_soc_dapm_add_route(dapm, route); + r = snd_soc_dapm_add_route(dapm, route, false); if (r < 0) { dev_err(dapm->dev, "ASoC: Failed to add route %s -> %s -> %s\n", route->source, @@ -3670,7 +3671,7 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) cpu_dai->codec->name, r.source, codec_dai->platform->name, r.sink); - snd_soc_dapm_add_route(&card->dapm, &r); + snd_soc_dapm_add_route(&card->dapm, &r, true); } /* connect BE DAI capture if widgets are valid */ @@ -3681,7 +3682,7 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) codec_dai->codec->name, r.source, cpu_dai->platform->name, r.sink); - snd_soc_dapm_add_route(&card->dapm, &r); + snd_soc_dapm_add_route(&card->dapm, &r, true); } } -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH] ASoC: wm8804: Allow control of master clock divider in PLL generation
by Florian Meier 17 Jan '14

17 Jan '14

On 01/17/2014 07:33 PM, Mark Brown wrote: > On Fri, Jan 17, 2014 at 07:06:24PM +0100, Florian Meier wrote: >> > Intentionally off-list? Oh no - I am sorry! >> If I remember correctly the error was >> "codec can not start from non-off bias with idle_bias_off==true" > >> I think the solution is just to set idle_bias_off = false and everything >> seems to be working with that. I just don't know if there might be any >> side effects. > > Setting it to false increases power consumption since the device is > kept more powered on when idle but reduces startup time from idle. For > digital only devices like the wm8804 there shouldn't be any reason to > keep it powered up when not in use, the startup time is generally > negligable anyway. > So a better solution would be to set SND_SOC_BIAS_OFF instead of SND_SOC_BIAS_STANDBY at the end of probe?

2 1

[alsa-devel] [PATCH 0/4] ASoC: tlv320aic32x4: DT support
by Markus Pargmann 17 Jan '14

17 Jan '14

Hi, Here are some more updates for the tlv320aic32x4 codec. This series fixes the mixer controls to operate on the full range instead of only the positive range. This depends on my other series: "ASoC: core: volume control using signed register values" Patch 3 adds DT support. Patch 4 adds support to control the master clock connected to the codec. Regards, Markus Markus Pargmann (4): ASoC: tlv320aic32x4: Use signed int mixer controls ASoC: tlv320aic32x4: Use gpio_is_valid ASoC: tlv320aic32x4: DT support ASoC: tlv320aic32x4: Support for master clock .../devicetree/bindings/sound/tlv320aic32x4.txt | 25 +++++++ sound/soc/codecs/tlv320aic32x4.c | 83 ++++++++++++++++++---- 2 files changed, 96 insertions(+), 12 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt -- 1.8.5.2

2 6

[alsa-devel] [PATCH v2 0/2] Dynamic routing for compressed streams.
by Liam Girdwood 17 Jan '14

17 Jan '14

Changes for V2 * We no longer copy the compressed params into the BE hw_params. This was bogus, but didn't show up on Haswell testing as the BE params were never used. The DSP ran the BE is a fixed configuration so all BE params were ignored. * The BE hw_params are now zeroed to force the machine driver to fixup the BE configuration. i.e. machine driver can query the DSP encodec/decoder for correct hw_params given the use case. * No longer need to allocate memory for hw_params. Regards Liam

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: DPCM: make some DPCM API calls non static for compressed usage
by Liam Girdwood 17 Jan '14

17 Jan '14

The ASoC compressed code needs to call the internal DPCM APIs in order to dynamically route compressed data to different DAIs. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dpcm.h | 22 ++++++++++++++++++++++ sound/soc/soc-pcm.c | 29 ++++++++++++----------------- 2 files changed, 34 insertions(+), 17 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 047d657..2883a7a 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -11,6 +11,7 @@ #ifndef __LINUX_SND_SOC_DPCM_H #define __LINUX_SND_SOC_DPCM_H +#include <linux/slab.h> #include <linux/list.h> #include <sound/pcm.h> @@ -135,4 +136,25 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute); int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd); int soc_dpcm_runtime_update(struct snd_soc_card *); +int dpcm_path_get(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list_); +int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list, int new); +int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream); +void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream); +void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int tream); +int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd); +int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, + int event); + +static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) +{ + kfree(*list); +} + + #endif diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1534115..7023f7e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -58,7 +58,7 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); /* DPCM stream event, send event to FE and all active BEs. */ -static int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, +int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event) { struct snd_soc_dpcm *dpcm; @@ -877,7 +877,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, } /* disconnect a BE and FE */ -static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) +void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm, *d; @@ -973,7 +973,7 @@ static int widget_in_list(struct snd_soc_dapm_widget_list *list, return 0; } -static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, +int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { struct snd_soc_dai *cpu_dai = fe->cpu_dai; @@ -995,11 +995,6 @@ static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, return paths; } -static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) -{ - kfree(*list); -} - static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { @@ -1069,7 +1064,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; /* don't connect if FE is not running */ - if (!fe->dpcm[stream].runtime) + if (!fe->dpcm[stream].runtime && !fe->fe_compr) continue; /* newly connected FE and BE */ @@ -1094,7 +1089,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, * Find the corresponding BE DAIs that source or sink audio to this * FE substream. */ -static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, +int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list, int new) { if (new) @@ -1103,7 +1098,7 @@ static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, return dpcm_prune_paths(fe, stream, list); } -static void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) +void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1141,7 +1136,7 @@ static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, } } -static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int err, count = 0; @@ -1293,7 +1288,7 @@ be_err: return ret; } -static int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1354,7 +1349,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) return 0; } -static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1419,7 +1414,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) return 0; } -static int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret; @@ -1549,7 +1544,7 @@ static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, return ret; } -static int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, +int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; @@ -1717,7 +1712,7 @@ out: return ret; } -static int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret = 0; -- 1.8.3.2

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[alsa-devel] ProVideo PV 918A - audi o device created but cannot capture sound
by siddharth 17 Jan '14

17 Jan '14

Hi, I have installed a ProVideo PV 981. I was able to get the sound device detected by putting modprobe snd-bt87x load_all=1 digital_rate=48000 enable=1 but am unable to capture sound. I am using Fedora 19 on an intel i5 machine. In dmesg i get the error: [ 235.145932] ALSA sound/pci/bt87x.c:860 unknown card 0x878-0xaa00:0x1460 [ 235.145935] ALSA sound/pci/bt87x.c:863 please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> [ 235.146061] ALSA sound/pci/bt87x.c:931 bt87x0: Using board 0, analog, digital (rate 48000 Hz) Any help will be appreciated. regards Siddharth

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Re: [alsa-devel] [pulseaudio-discuss] Need help with audio gain on an Intel HDA equipt ASUS M2N-SLI Deluxe
by Raymond Yau 17 Jan '14

17 Jan '14

2014/1/16 Axel Braun <axel.braun(a)gmx.de> > Am Donnerstag, 16. Januar 2014, 16:23:19 schrieb Raymond Yau: > > Any hint you could give me? > > > > post the output of alsa-info.sh , there is a bug in alsa-driver which > treat > > dock headphone and headphone to support surround 4.0 > > /sys/class/sound/hwC0D0/init_pin_configs: 0x19 0x04211040 0x1a 0x61a19050 0x1b 0x04a11060 0x1c 0x6121401f 0x1d 0x40f001f0 0x1e 0x40f001f0 0x1f 0x90170110 0x20 0x40f001f0 0x22 0x40f001f0 0x23 0x90a60170 /sys/class/sound/hwC0D0/driver_pin_configs: 0x19 0x042110ff 0x1a 0x21a190f0 0x1c 0x212140ff https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… it was using auto but now the driver modify the dock headphone , dock mic and headphone pin default. there are also changes in the thinkpad_acpi module too. https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… do this thinkpad support headset or headphone ? it is strange that speaker volume is at max but amp out vals at node 0x11 is 0x3d instead of 0x4a (0dB) Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 74 Mono: Playback 61 [82%] [-13.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 0 [0%] [-74.00dB] [off] Front Right: Playback 0 [0%] [-74.00dB] [off] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 74 [100%] [0.00dB] [on] Front Right: Playback 74 [100%] [0.00dB] [on] Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CX20590 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x3d 0x3d] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x80 0x80] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0

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[alsa-devel] [PATCH 1/2] ASoC: DPCM: make some DPCM API calls non static for compressed usage
by Liam Girdwood 17 Jan '14

17 Jan '14

The ASoC compressed code needs to call the internal DPCM APIs in order to dynamically route compressed data to different DAIs. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dpcm.h | 22 ++++++++++++++++++++++ sound/soc/soc-pcm.c | 29 ++++++++++++----------------- 2 files changed, 34 insertions(+), 17 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 047d657..2883a7a 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -11,6 +11,7 @@ #ifndef __LINUX_SND_SOC_DPCM_H #define __LINUX_SND_SOC_DPCM_H +#include <linux/slab.h> #include <linux/list.h> #include <sound/pcm.h> @@ -135,4 +136,25 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute); int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd); int soc_dpcm_runtime_update(struct snd_soc_card *); +int dpcm_path_get(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list_); +int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list, int new); +int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream); +void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream); +void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int tream); +int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd); +int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, + int event); + +static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) +{ + kfree(*list); +} + + #endif diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1534115..7023f7e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -58,7 +58,7 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); /* DPCM stream event, send event to FE and all active BEs. */ -static int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, +int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event) { struct snd_soc_dpcm *dpcm; @@ -877,7 +877,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, } /* disconnect a BE and FE */ -static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) +void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm, *d; @@ -973,7 +973,7 @@ static int widget_in_list(struct snd_soc_dapm_widget_list *list, return 0; } -static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, +int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { struct snd_soc_dai *cpu_dai = fe->cpu_dai; @@ -995,11 +995,6 @@ static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, return paths; } -static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) -{ - kfree(*list); -} - static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { @@ -1069,7 +1064,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; /* don't connect if FE is not running */ - if (!fe->dpcm[stream].runtime) + if (!fe->dpcm[stream].runtime && !fe->fe_compr) continue; /* newly connected FE and BE */ @@ -1094,7 +1089,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, * Find the corresponding BE DAIs that source or sink audio to this * FE substream. */ -static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, +int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list, int new) { if (new) @@ -1103,7 +1098,7 @@ static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, return dpcm_prune_paths(fe, stream, list); } -static void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) +void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1141,7 +1136,7 @@ static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, } } -static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int err, count = 0; @@ -1293,7 +1288,7 @@ be_err: return ret; } -static int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1354,7 +1349,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) return 0; } -static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1419,7 +1414,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) return 0; } -static int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret; @@ -1549,7 +1544,7 @@ static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, return ret; } -static int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, +int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; @@ -1717,7 +1712,7 @@ out: return ret; } -static int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret = 0; -- 1.8.3.2

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[alsa-devel] [PATCH v4 0/9] ASoC: fsl-ssi: offline/online configuration and cleanups
by Markus Pargmann 17 Jan '14

17 Jan '14

Hi, This patch series fixes some fsl-ssi code that does not act exactly as it is described in the reference manuals. The reference manuals before imx51 do mention that some register bits, including TDMAE/RDMAE, should not be changed while the SSI unit is enabled (SSIEN). At the same time the SDMA unit has a undefined request handling if there are DMA requests before the SDMA engine and its channel is configured. The reference manual states that SSI DMA requests are sent as soon as the FIFO and the DMAE bits are enabled. It does not mention a dependency to SCR's TE/RE bit. To avoid sending DMA requests before a channel is configured, we have to seperate the fsl-ssi behavior into offline and online configuration. SoCs before imx51 have to use offline configuration while later SoC versions can use online reconfiguration. This series adds support for online configuration and cleans up the configuration register code especially in fsl_ssi_trigger. At the end we have a seperation between enable/disable logic and the actual configuration register values used to enable/disable TX/RX. v4 is another rebase onto latest topic/fsl branch. It includes the DAI master mode changes from Nicolin. @Nicolin: Could you please test this series with your hardware to verify that your patch still works? I tested this series on mx53. The series is based on Mark Brown's topic/fsl branch. Regards, Markus Changes in v4: - Support for irqstats on all imx SoCs - Fix offline_config and use switch statement - Introduce ssi_private->irq_stats bool Changes in v3: - Rebase on latest topic/fsl - Added patch to cleanup probe function error handling - Fix interrupt stats for imx53 - Replace imx50-ssi by imx51-ssi Markus Pargmann (9): ASoC: fsl-ssi: Fix probe error handling ASoC: fsl-ssi: Move sysfs stats to debugfs ASoC: fsl-ssi: Add imx51-ssi and of_device_id matching ASoC: fsl-ssi: Fix interrupt stats for imx ASoC: fsl-ssi: Add offline_config flag ASoC: fsl-ssi: Add configuration helper functions ASoC: fsl-ssi: Move RX/TX configuration to seperate functions ASoC: fsl-ssi: Drop ac97 specific trigger function ARM: DTS: imx5* imx6*, use imx51-ssi arch/arm/boot/dts/imx53.dtsi | 10 +- arch/arm/boot/dts/imx6qdl.dtsi | 12 +- arch/arm/boot/dts/imx6sl.dtsi | 12 +- sound/soc/fsl/fsl_ssi.c | 567 ++++++++++++++++++++++++++++------------- 4 files changed, 411 insertions(+), 190 deletions(-) -- 1.8.5.1

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[alsa-devel] [PATCH] ASoC: core: Fix possible NULL pointer dereference of pcm->config
by Xiubo Li 16 Jan '14

16 Jan '14

Since the soc generic dmaengine pcm driver allows using the defualt settings, so the pcm->config maybe NULL. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 560a778..5bace12 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -176,17 +176,20 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); struct snd_dmaengine_dai_dma_data *dma_data; + dma_filter_fn fn = NULL; dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) return pcm->chan[0]; - if (pcm->config->compat_request_channel) + if (pcm->config && pcm->config->compat_request_channel) return pcm->config->compat_request_channel(rtd, substream); - return snd_dmaengine_pcm_request_channel(pcm->config->compat_filter_fn, - dma_data->filter_data); + if (pcm->config) + fn = pcm->config->compat_filter_fn; + + return snd_dmaengine_pcm_request_channel(fn, dma_data->filter_data); } static bool dmaengine_pcm_can_report_residue(struct dma_chan *chan) -- 1.8.4

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[alsa-devel] ASoC fixes for v3.13
by Mark Brown 16 Jan '14

16 Jan '14

The following changes since commit 7e22e91102c6b9df7c4ae2168910e19d2bb14cd6: Linux 3.13-rc8 (2014-01-12 17:04:18 +0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13-rc8 for you to fetch changes up to c5281227c90578c7b7beffca56e8c3a68d4607de: Merge remote-tracking branches 'asoc/fix/adau1701', 'asoc/fix/fsl' and 'asoc/fix/tlv320aic32x4' into asoc-linus (2014-01-15 23:17:15 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.13 A few small fixes in drivers, nothing too remarkable here but all good to have - mainly these are fixes for things that were introduced in the last merge window but only just got useful testing. ---------------------------------------------------------------- Lars-Peter Clausen (1): ASoC: adau1701: Fix ADAU1701_SEROCTL_WORD_LEN_16 constant Mark Brown (1): Merge remote-tracking branches 'asoc/fix/adau1701', 'asoc/fix/fsl' and 'asoc/fix/tlv320aic32x4' into asoc-linus Markus Pargmann (2): ASoC: codec: tlv320aic32x4: Fix regmap range config ASoC: tlv320aic32x4: Fix regmap range_min Nicolin Chen (1): ASoC: imx-spdif: Use snd-soc-dummy CODEC driver to link card sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 4 +- sound/soc/fsl/Kconfig | 1 - sound/soc/fsl/imx-spdif.c | 84 ++++++++++------------------------------ 4 files changed, 24 insertions(+), 67 deletions(-)

2 4

[alsa-devel] [GIT PULL] ASoC updates for v3.13
by Mark Brown 16 Jan '14

16 Jan '14

The following changes since commit 7e22e91102c6b9df7c4ae2168910e19d2bb14cd6: Linux 3.13-rc8 (2014-01-12 17:04:18 +0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13-rc8-2 for you to fetch changes up to c6affc0dba45bb2074ff56b21386d22eeaf682f8: Merge remote-tracking branches 'asoc/fix/adau1701' and 'asoc/fix/tlv320aic32x4' into asoc-linus (2014-01-16 14:02:52 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.13 A few small fixes in drivers, nothing too remarkable here but all good to have - mainly these are fixes for things that were introduced in the last merge window but only just got useful testing. ---------------------------------------------------------------- Lars-Peter Clausen (1): ASoC: adau1701: Fix ADAU1701_SEROCTL_WORD_LEN_16 constant Mark Brown (1): Merge remote-tracking branches 'asoc/fix/adau1701' and 'asoc/fix/tlv320aic32x4' into asoc-linus Markus Pargmann (2): ASoC: codec: tlv320aic32x4: Fix regmap range config ASoC: tlv320aic32x4: Fix regmap range_min sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-)

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[alsa-devel] [GIT PULL] ASoC updates for v3.14
by Mark Brown 16 Jan '14

16 Jan '14

The following changes since commit cdef2e5f358db5fd27eec4c72eb295c232287813: sound: oss: remove last sleep_on users (2014-01-14 16:12:07 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.14-2 for you to fetch changes up to 701caa51a2ce74182d39380ca11defeb163d98c1: Merge remote-tracking branches 'asoc/topic/adsp', 'asoc/topic/atmel', 'asoc/topic/bcm2835', 'asoc/topic/docs', 'asoc/topic/fsl', 'asoc/topic/generic', 'asoc/topic/kirkwood', 'asoc/topic/mc13783', 'asoc/topic/mxs', 'asoc/topic/nuc900', 'asoc/topic/sai', 'asoc/topic/sh', 'asoc/topic/ssm2602', 'asoc/topic/tlv320aic3x', 'asoc/topic/twl4030', 'asoc/topic/ux500', 'asoc/topic/width' and 'asoc/topic/x86' into for-tiwai (2014-01-16 12:44:01 +0000) ---------------------------------------------------------------- ASoC: More updates for v3.14 A few more updates for v3.14 since the last set, highlights include: - Lots of DMA updates from Lars-Peter - Improvements to the constraints handling code from Lars-Peter - A very helpful conversion of the TWL4030 driver to regmap from Peter - A new driver for the Freescale ESAI controller from Nicolin Chen - Conversion of some of the drivers to use params_width() ---------------------------------------------------------------- Alexander Shiyan (5): ASoC: fsl_ssi: Fix printing return code on clk error ASoC: mc13783: Use module_platform_driver_probe() ASoC: mc13783: Use core error messages if registration fails ASoC: mc13783: Drop fixed ADC & DAC ports usage ASoC: mc13783: trivial: Cleanup module Andreas Pretzsch (1): ASoC: ssm2602: add 16kHz sampling rate support Arun Shamanna Lakshmi (1): ASoC: dapm: Change prototype of soc_widget_read Charles Keepax (5): ASoC: wm_adsp: Factor out ADSP2 boot proceedure ASoC: wm_adsp: Start DSP booting earlier in the DAPM process ASoC: wm_adsp: Mark wm_adsp2_boot_work as static mfd: wm5110: Add registers for headphone short circuit control ASoC: wm5110: Add controls for headphone short circuit protection Fabio Estevam (1): ASoC: fsl: fsl_ssi: Use '%ld' to print 'long int' Jean-Francois Moine (1): ASoC: pcm: Fix lack of platform bespoke_trigger() call Lars-Peter Clausen (24): ASoC: atmel: Don't set unused struct snd_pcm_hardware fields ASoC: dpcm: Add helper function for initializing runtime pcm ASoC: dpcm: Allow PCMs to omit the set of supported formats ASoC: fsl: Don't set unused struct snd_pcm_hardware fields ASoC: mxs: Don't set unused struct snd_pcm_hardware fields ASoC: mxs: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag ASoC: adau1701: Fix ADAU1701_SEROCTL_WORD_LEN_16 constant ASoC: intel: Don't set unused struct snd_pcm_hardware fields ASoC: kirkwood: Don't set unused struct snd_pcm_hardware fields ASoC: nuc900: Don't set unused struct snd_pcm_hardware fields ASoC: sh: Don't set unused struct snd_pcm_hardware fields ASoC: ux500: Don't set unused struct snd_pcm_hardware fields ASoC: Allow PCMs to restrict the supported formats ASoC: pcm: Properly initialize hw->rate_max ASoC: fsl: Don't mix SNDRV_PCM_RATE_CONTINUOUS with specific rates ASoC: s6000: Don't mix SNDRV_PCM_RATE_CONTINUOUS with specific rates ALSA: Add helper function for intersecting two rate masks ASoC: pcm: Use snd_pcm_rate_mask_intersect() helper dma: Indicate residue granularity in dma_slave_caps dma: pl330: Set residue_granularity ASoC: generic-dmaengine-pcm: Check NO_RESIDUE flag at runtime ASoC: generic-dmaengine-pcm: Check DMA residue granularity ASoC: axi-{spdif,i2s}: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag ASoC: samsung: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag Lee Jones (10): ARM: ux500: Don't use enums for MSP IDs - for easy DT conversion ASoC: ux500: Provide better checking for Device Tree and/or Platform Data ASoC: Ux500: Match platform by device node when booting Device Tree ASoC: ux500_pcm: Stop pretending that we support varying address widths ASoC: ux500_pcm: Expect different saved DMA data when obtaining from DAI store ASoC: ux500_pcm: Extend Device Tree support to deal with DMA data ASoC: ux500: Store DMA data in the DAI differently in the pdata and DT case ASoC: ux500_pcm: Take out pointless dev_dbg() call ASoC: ux500_pcm: Differentiate between pdata and DT initialisation ASoC: ux500: Dramatically reduce the size of the DAI driver data struct Liam Girdwood (4): ASoC: docs: Update the Overview document ASoC: dpcm: Explicitly set BE DAI link supported stream directions ASoC: utils: Add internal call to determine if DAI is dummy. ASoC: sapm: Automatically connect DAI link widgets in DAPM graph. Mark Brown (30): Merge tag 'ib-asoc-3.14.2' of git://git.linaro.org/people/ljones/mfd into asoc-tlw4030 ASoC: ad1836: Use params_width() rather than explicit memory format ASoC: ad193x: Use params_width() rather than memory format ASoC: adau1373: Use params_width() rather than memory format ASoC: adau1701: Use params_width() rather than memory format ASoC: adav80x: Use params_width() rather than memory format Merge tag 'v3.13-rc3' into asoc-arizona ASoC: alc5623: Use params_width() rather than memory format ASoC: alc5632: Use params_width() rather than memory format ASoC: cs42l51: Use params_width() rather than memory format ASoC: da7210: Use params_width() rather than memory format ASoC: da7213: Use params_width() rather than memory format ASoC: da732x: Use params_width() rather than memory format ASoC: da9055: Use params_width() rather than memory format ASoC: isabelle: Use params_width() rather than memory format ASoC: max98088: Use params_width() rather than memory format ASoC: max98090: Use params_width() rather than memory format ASoC: max98095: Use params_width() rather than memory format ASoC: max9850: Use params_width() rather than memory format Merge tag 'v3.13-rc3' into asoc-pcm Merge branch 'topic/axi' of git://git.kernel.org/.../broonie/sound into asoc-dma Merge branch 'topic/samsung' of git://git.kernel.org/.../broonie/sound into asoc-dma ASoC: core: Return -ENOTSUPP from set_sysclk() if no operation provided Merge remote-tracking branches 'asoc/fix/adau1701' and 'asoc/fix/tlv320aic32x4' into for-tiwai Merge remote-tracking branch 'asoc/topic/core' into for-tiwai Merge remote-tracking branch 'asoc/topic/dapm' into for-tiwai Merge remote-tracking branch 'asoc/topic/dma' into for-tiwai Merge remote-tracking branch 'asoc/topic/pcm' into for-tiwai Merge remote-tracking branch 'asoc/topic/arizona' into for-tiwai Merge remote-tracking branches 'asoc/topic/adsp', 'asoc/topic/atmel', 'asoc/topic/bcm2835', 'asoc/topic/docs', 'asoc/topic/fsl', 'asoc/topic/generic', 'asoc/topic/kirkwood', 'asoc/topic/mc13783', 'asoc/topic/mxs', 'asoc/topic/nuc900', 'asoc/topic/sai', 'asoc/topic/sh', 'asoc/topic/ssm2602', 'asoc/topic/tlv320aic3x', 'asoc/topic/twl4030', 'asoc/topic/ux500', 'asoc/topic/width' and 'asoc/topic/x86' into for-tiwai Markus Pargmann (13): ASoC: fsl-ssi: Fix probe error handling ASoC: fsl-ssi: Move sysfs stats to debugfs ASoC: fsl-ssi: Add imx51-ssi and of_device_id matching ASoC: fsl-ssi: Fix interrupt stats for imx ASoC: fsl-ssi: Add offline_config flag ASoC: fsl-ssi: Add configuration helper functions ASoC: fsl-ssi: Move RX/TX configuration to seperate functions ASoC: fsl-ssi: Drop ac97 specific trigger function ASoC: fsl-ssi doc: Add list of supported compatibles ASoC: fsl-ssi: Fix stats compile warning ASoC: codec: tlv320aic32x4: Fix regmap range config ASoC: tlv320aic3x: Add tlv320aic32x4 as compatible ASoC: tlv320aic32x4: Fix regmap range_min Nicolin Chen (2): ASoC: fsl_ssi: Set default slot number for common cases ASoC: fsl_esai: Add ESAI CPU DAI driver Peter Ujfalusi (11): mfd: twl-core: Simplify IO wrapper functions by moving common code out mfd: twl-core: API to set the regcache bypass for a given regmap in twl mfd: twl-core: Enable regcache for audio registers ASoC: twl4030: Separate write condition checking from I/O function ASoC: twl4030: Remove check defaults functionality ASoC: twl4030: Remove reset registers functionality ASoC: twl4030: Introduce local ctl register cache ASoC: twl4030: Remove local reg cache ASoC: twl4030: Parameter alignment fixes (for code consistency) ASoC: twl4030: Move the ctl cache update local to twl4030_write() function ASoC: twl4030: Pass the twl4030_priv directly to twl4030_can_write_to_chip() Takashi Iwai (1): ASoC: bcm: Remove obsoleted Kconfig dependency Wei Yongjun (2): ASoC: ux500: Fix sparse non static symbol warning ASoC: fsl-ssi: Add missing clk_disable_unprepare() on error in fsl_ssi_probe() Xiubo Li (8): ASoC: simple-card: Fix the sysclk selection. ASoC: simple-card: fix the DAPM routes map parsing ASoC: simple-card: keep the property's name the same pattern ASoC: fsl-sai: Clean up the code ASoC: simple-card: fix a bug where cinfo will be NULL before using it ASoC: simple-card: fix the cinfo error check ASoC: simple-card: use snd_soc_card_set/get_drvdata ASoC: simple-card: fix one bug to writing to the platform data .../devicetree/bindings/sound/fsl,esai.txt | 50 ++ .../devicetree/bindings/sound/fsl,ssi.txt | 7 +- .../devicetree/bindings/sound/simple-card.txt | 2 +- .../devicetree/bindings/sound/tlv320aic3x.txt | 1 + Documentation/sound/alsa/soc/overview.txt | 27 +- arch/arm/mach-ux500/board-mop500-audio.c | 8 +- drivers/dma/pl330.c | 1 + drivers/mfd/twl-core.c | 190 ++++- drivers/mfd/wm5110-tables.c | 6 + include/linux/dmaengine.h | 28 + include/linux/i2c/twl.h | 5 +- include/linux/mfd/arizona/registers.h | 27 + include/linux/platform_data/asoc-ux500-msp.h | 9 +- include/sound/pcm.h | 2 + include/sound/soc-dai.h | 2 + include/sound/soc-dapm.h | 1 + include/sound/soc.h | 4 + sound/core/pcm_misc.c | 39 + sound/soc/adi/axi-i2s.c | 3 +- sound/soc/adi/axi-spdif.c | 3 +- sound/soc/atmel/atmel-pcm-dma.c | 1 - sound/soc/atmel/atmel-pcm-pdc.c | 1 - sound/soc/bcm/Kconfig | 1 - sound/soc/codecs/ad1836.c | 10 +- sound/soc/codecs/ad193x.c | 10 +- sound/soc/codecs/adau1373.c | 10 +- sound/soc/codecs/adau1701.c | 36 +- sound/soc/codecs/adav80x.c | 30 +- sound/soc/codecs/alc5623.c | 10 +- sound/soc/codecs/alc5632.c | 8 +- sound/soc/codecs/arizona.h | 17 +- sound/soc/codecs/cs42l51.c | 14 +- sound/soc/codecs/da7210.c | 10 +- sound/soc/codecs/da7213.c | 10 +- sound/soc/codecs/da732x.c | 10 +- sound/soc/codecs/da9055.c | 10 +- sound/soc/codecs/isabelle.c | 6 +- sound/soc/codecs/max98088.c | 6 +- sound/soc/codecs/max98090.c | 4 +- sound/soc/codecs/max98095.c | 6 +- sound/soc/codecs/max9850.c | 8 +- sound/soc/codecs/mc13783.c | 34 +- sound/soc/codecs/ssm2602.c | 14 +- sound/soc/codecs/tlv320aic32x4.c | 4 +- sound/soc/codecs/twl4030.c | 376 ++++------ sound/soc/codecs/wm5110.c | 7 + sound/soc/codecs/wm_adsp.c | 199 +++-- sound/soc/codecs/wm_adsp.h | 12 +- sound/soc/fsl/Kconfig | 3 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_dma.c | 7 - sound/soc/fsl/fsl_esai.c | 815 +++++++++++++++++++++ sound/soc/fsl/fsl_esai.h | 354 +++++++++ sound/soc/fsl/fsl_sai.c | 7 +- sound/soc/fsl/fsl_ssi.c | 586 ++++++++++----- sound/soc/fsl/imx-pcm-dma.c | 3 - sound/soc/fsl/imx-pcm-fiq.c | 3 - sound/soc/fsl/mpc5200_dma.c | 4 - sound/soc/fsl/mpc5200_psc_i2s.c | 3 +- sound/soc/generic/simple-card.c | 70 +- sound/soc/intel/sst_platform.c | 10 - sound/soc/intel/sst_platform.h | 4 - sound/soc/kirkwood/kirkwood-dma.c | 16 - sound/soc/mxs/mxs-pcm.c | 6 - sound/soc/nuc900/nuc900-pcm.c | 3 - sound/soc/s6000/s6000-i2s.c | 3 +- sound/soc/samsung/dmaengine.c | 1 - sound/soc/sh/dma-sh7760.c | 17 - sound/soc/sh/fsi.c | 6 - sound/soc/sh/rcar/core.c | 6 - sound/soc/soc-core.c | 5 +- sound/soc/soc-dapm.c | 75 +- sound/soc/soc-generic-dmaengine-pcm.c | 66 +- sound/soc/soc-pcm.c | 58 +- sound/soc/soc-utils.c | 7 + sound/soc/ux500/mop500.c | 2 + sound/soc/ux500/ux500_msp_dai.c | 146 ++-- sound/soc/ux500/ux500_msp_i2s.c | 56 +- sound/soc/ux500/ux500_msp_i2s.h | 2 +- sound/soc/ux500/ux500_pcm.c | 65 +- 80 files changed, 2656 insertions(+), 1034 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,esai.txt create mode 100644 sound/soc/fsl/fsl_esai.c create mode 100644 sound/soc/fsl/fsl_esai.h

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[alsa-devel] different sample rate in alsa.conf file, ALSA DMIX mode
by Pavel Botev 16 Jan '14

16 Jan '14

Hi ALSA support team, Can we set more sample rate for DMIX mode in the alsa.conf file than the default one (defaults.pcm.dmix.rate 48000)? Can we use one alsa.conf file, with many sample rates (8KHz, 11.025KHz, 16KHz, 22.05KHz, 32KHz, 44.1KHz, 48KHz, 64KHz, 88.2KHz and 96KHz), not with just one default sample rate? The problem is that when we use one sample rate in alsa.conf and other in the ALSA application, we have resampling executed which is very CPU intensive. Thus we search a way to use alsa.conf file with many sample rates, thus avoiding resampling. If not possible, can we change the algorithm of resampling, thus have low CPU usage when resampling is activated? Thanks for your time. Best regards, Pavel

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[alsa-devel] [PATCH] ALSA: hda - add headset mic detect quirks for some Dell machines
by Hui Wang 16 Jan '14

16 Jan '14

When we plug a 3-ring headset on some Dell machines, the headset mic can't be detected, after apply this patch, the headset mic can work well on all those machines. On the machine with the Subsytem ID 0x10280610, if we use ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, the headset mic can be detected and work well, but the sound can't be outputed via headphone anymore, use ALC269_FIXUP_DELL3_MIC_NO_PRESENCE can fix this problem. BugLink: https://bugs.launchpad.net/bugs/1260303 Cc: David Henningsson <david.henningsson(a)canonical.com> Tested-by: David Chen <david.chen(a)canonical.com> Tested-by: Cyrus Lien <cyrus.lien(a)canonical.com> Tested-by: Shawn Wang <shawn.wang(a)canonical.com> Tested-by: Chih-Hsyuan Ho <chih.ho(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a1c16457..f9b22fb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4253,7 +4253,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x060f, "Dell", ALC269_FIXUP_DELL3_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC269_FIXUP_DELL3_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0615, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), @@ -4264,6 +4265,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0640, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0651, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0652, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0653, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0658, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0662, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), @@ -5065,6 +5071,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_AUTO_MUTE), + SND_PCI_QUIRK(0x1028, 0x064e, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A_CHMAP), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP), -- 1.8.1.2

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[alsa-devel] [3.13-rc8] AM335x BeagleBone Buffer XRUN
by Jack Mitchell 16 Jan '14

16 Jan '14

I am currently trying to debug a buffer overrun when capturing with the BeagleBone Black and Audio Cape attached on 3.13-rc8, this is pretty much the same setup as the am335x-evm. The following is the traceback captured using the XRUN_BUFFER_DEBUG kernel config option. I think an interrupt is taking too long to return. Example Traceback 1 (Writing to uSD card) > [ 802.578728] ALSA sound/core/pcm_lib.c:177 XRUN: pcmC0D0c:0 > CPU: 0 PID: 0 Comm: swapper/0 Not tainted 3.13.0-rc8-00005-ga6da83f-dirty #12 > [ 802.593139] [<c00160e8>] (unwind_backtrace+0x0/0x12c) from [<c0012b08>] (show_stack+0x10/0x14) > [ 802.602174] [<c0012b08>] (show_stack+0x10/0x14) from [<c058c80c>] (dump_stack+0x7c/0x94) > [ 802.610659] [<c058c80c>] (dump_stack+0x7c/0x94) from [<c04862e8>] (xrun+0x80/0xb0) > [ 802.618596] [<c04862e8>] (xrun+0x80/0xb0) from [<c0486e1c>] (snd_pcm_update_state+0xf8/0x100) > [ 802.627531] [<c0486e1c>] (snd_pcm_update_state+0xf8/0x100) from [<c0487048>] (snd_pcm_update_hw_ptr0+0x224/0x82c) > [ 802.638281] [<c0487048>] (snd_pcm_update_hw_ptr0+0x224/0x82c) from [<c0487700>] (snd_pcm_period_elapsed+0xb0/0xf4) > [ 802.649131] [<c0487700>] (snd_pcm_period_elapsed+0xb0/0xf4) from [<c0021458>] (dma_irq_handler+0x130/0x1f4) > [ 802.659345] [<c0021458>] (dma_irq_handler+0x130/0x1f4) from [<c008e504>] (handle_irq_event_percpu+0x5c/0x21c) > [ 802.669735] [<c008e504>] (handle_irq_event_percpu+0x5c/0x21c) from [<c008e700>] (handle_irq_event+0x3c/0x5c) > [ 802.680035] [<c008e700>] (handle_irq_event+0x3c/0x5c) from [<c0091038>] (handle_level_irq+0xac/0x118) > [ 802.689697] [<c0091038>] (handle_level_irq+0xac/0x118) from [<c008dc8c>] (generic_handle_irq+0x28/0x3c) > [ 802.699542] [<c008dc8c>] (generic_handle_irq+0x28/0x3c) from [<c000f968>] (handle_IRQ+0x4c/0xb4) > [ 802.708750] [<c000f968>] (handle_IRQ+0x4c/0xb4) from [<c00085b4>] (omap3_intc_handle_irq+0x60/0x74) > [ 802.718237] [<c00085b4>] (omap3_intc_handle_irq+0x60/0x74) from [<c0592a64>] (__irq_svc+0x44/0x5c) > [ 802.727618] Exception stack(0xc088df58 to 0xc088dfa0) > [ 802.732912] df40: 00000001 00000001 > [ 802.741479] df60: 00000000 c0897cb0 c088c000 c088c000 00000000 c0933bd4 c08949a0 c0894880 > [ 802.750047] df80: c0894930 c059f4a8 c088c020 c088dfa0 c0082ef8 c000fd00 20070013 ffffffff > [ 802.758620] [<c0592a64>] (__irq_svc+0x44/0x5c) from [<c000fd00>] (arch_cpu_idle+0x20/0x3c) > [ 802.767283] [<c000fd00>] (arch_cpu_idle+0x20/0x3c) from [<c008db84>] (cpu_startup_entry+0x120/0x1f0) > [ 802.776858] [<c008db84>] (cpu_startup_entry+0x120/0x1f0) from [<c0826a9c>] (start_kernel+0x304/0x364) Example Traceback 2 (Writing to tmpfs) > [ 1310.025256] ALSA sound/core/pcm_lib.c:177 XRUN: pcmC0D0c:0 > [ 1310.031046] CPU: 0 PID: 998 Comm: avconv Not tainted 3.13.0-rc8-00005-ga6da83f-dirty #12 > [ 1310.039580] [<c00160e8>] (unwind_backtrace+0x0/0x12c) from [<c0012b08>] (show_stack+0x10/0x14) > [ 1310.048615] [<c0012b08>] (show_stack+0x10/0x14) from [<c058c80c>] (dump_stack+0x7c/0x94) > [ 1310.057102] [<c058c80c>] (dump_stack+0x7c/0x94) from [<c04862e8>] (xrun+0x80/0xb0) > 1310.065039] [<c04862e8>] (xrun+0x80/0xb0) from [<c0486e1c>] (snd_pcm_update_state+0xf8/0x100) > [ 1310.073974] [<c0486e1c>] (snd_pcm_update_state+0xf8/0x100) from [<c0487048>] (snd_pcm_update_hw_ptr0+0x224/0x82c) > [ 1310.084724] [<c0487048>] (snd_pcm_update_hw_ptr0+0x224/0x82c) from [<c0487700>] (snd_pcm_period_elapsed+0xb0/0xf4) > [ 1310.095574] [<c0487700>] (snd_pcm_period_elapsed+0xb0/0xf4) from [<c0021458>] (dma_irq_handler+0x130/0x1f4) > [ 1310.105788] [<c0021458>] (dma_irq_handler+0x130/0x1f4) from [<c008e504>] (handle_irq_event_percpu+0x5c/0x21c) > [ 1310.116178] [<c008e504>] (handle_irq_event_percpu+0x5c/0x21c) from [<c008e700>] (handle_irq_event+0x3c/0x5c) > [ 1310.126477] [<c008e700>] (handle_irq_event+0x3c/0x5c) from [<c0091038>] (handle_level_irq+0xac/0x118) > [ 1310.136138] [<c0091038>] (handle_level_irq+0xac/0x118) from [<c008dc8c>] (generic_handle_irq+0x28/0x3c) > [ 1310.145982] [<c008dc8c>] (generic_handle_irq+0x28/0x3c) from [<c000f968>] (handle_IRQ+0x4c/0xb4) > [ 1310.155190] [<c000f968>] (handle_IRQ+0x4c/0xb4) from [<c00085b4>] (omap3_intc_handle_irq+0x60/0x74) > [ 1310.164677] [<c00085b4>] (omap3_intc_handle_irq+0x60/0x74) from [<c0592a64>] (__irq_svc+0x44/0x5c) > [ 1310.174056] Exception stack(0xde0f7ee0 to 0xde0f7f28) > [ 1310.179354] 7ee0: 00000001 00000000 00000000 de3f8a80 00000000 00000002 c0047968 00000054 > [ 1310.187920] 7f00: 00000000 b5440c64 c0933670 beb92fb8 de0f6028 de0f7f28 c0082eb0 c0047348 > [ 1310.196480] 7f20: 60000113 ffffffff > [ 1310.200153] [<c0592a64>] (__irq_svc+0x44/0x5c) from [<c0047348>] (__do_softirq+0xa8/0x2f4) > [ 1310.208817] [<c0047348>] (__do_softirq+0xa8/0x2f4) from [<c0047968>] (irq_exit+0xac/0x104) > [ 1310.217479] [<c0047968>] (irq_exit+0xac/0x104) from [<c000f970>] (handle_IRQ+0x54/0xb4) > [ 1310.225868] [<c000f970>] (handle_IRQ+0x54/0xb4) from [<c00085b4>] (omap3_intc_handle_irq+0x60/0x74) > [ 1310.235347] [<c00085b4>] (omap3_intc_handle_irq+0x60/0x74) from [<c0592c3c>] (__irq_usr+0x3c/0x60) > [ 1310.244724] Exception stack(0xde0f7fb0 to 0xde0f7ff8) > [ 1310.250019] 7fa0: beb94588 beb92ab8 beb9301c beb93038 > [ 1310.258586] 7fc0: 00088ca0 00000140 00000160 b5440f18 00000080 b5440c64 beb92ab8 beb92fb8 > [ 1310.267150] 7fe0: 4b000001 beb9267c b5f4d6a0 b5f4c8c8 80000010 ffffffff I have done some tests and capturing to a uSD card is the easiest way to trigger XRUNS, where they occur often. Capturing to the eMMC is a little bit better. Capturing directly to tmpfs is nearly perfect, apart from very rare XRUN occurrences (10 minutes average maybe). This makes sense as EDMA is used to perform writes to the uSD and eMMC devices, so if they're being hogged with writes, then there is no time to transfer out the sound buffer and we get an overrun. So, does anybody have ideas where to start debugging this further? I have tried to increase the buffer size from 32K up to the supposed max of 128K but I can't find where to do it... if anyone could shed any light I would most grateful, do you think increasing the buffer would solve this, as it would give more time to trigger the irq and get the data out before overrun? I have tried multiple userspace programs to do the capture and the best so far is arecord initialized like this: nice -n -20 arecord -D hw:0,0 -f cd -v jack-test.wav > root@beaglebone:~# nice -n -20 arecord -D hw:0,0 -f dat -v jack-test.wav > Recording WAVE 'jack-test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo > DEBUG: buffer_time = 682667 > DEBUG: max_buffer_size = 32768 > Hardware PCM card 0 'AM335x-EVM' device 0 subdevice 0 > Its setup is: > stream : CAPTURE > access : RW_INTERLEAVED > format : S16_LE > subformat : STD > channels : 2 > rate : 48000 > exact rate : 48000 (48000/1) > msbits : 16 > buffer_size : 32768 > period_size : 2048 > period_time : 42666 > tstamp_mode : NONE > period_step : 1 > avail_min : 2048 > period_event : 0 > start_threshold : 1 > stop_threshold : 32768 > silence_threshold: 0 > silence_size : 0 > boundary : 1073741824 > appl_ptr : 0 > hw_ptr : 0 > overrun!!! (at least 221.598 ms long) > Status: > state : XRUN > trigger_time: 1389534120.970142840 > tstamp : 1389534121.191651631 > delay : 0 > avail : 32768 > avail_max : 32768 If you need any more information or debugging output please let me know. Regards, -- Jack Mitchell (jack(a)embed.me.uk) Embedded Systems Engineer http://www.embed.me.uk --

1 1

Re: [alsa-devel] [pulseaudio-discuss] Need help with audio gain on an Intel HDA equipt ASUS M2N-SLI Deluxe
by Raymond Yau 16 Jan '14

16 Jan '14

2014/1/16 Axel Braun <axel.braun(a)gmx.de> > Am Donnerstag, 16. Januar 2014, 09:03:25 schrieb Raymond Yau: > > Post the output of > > > > pactl list > > > > pulseaudio verbose log > > I jump in here as well, same problem on a ThinkPad T520, sound volume is > way > too low - watching movie via loudspeakers is fairly impossible. > > Any hint you could give me? > post the output of alsa-info.sh , there is a bug in alsa-driver which treat dock headphone and headphone to support surround 4.0 Profile: input:analog-stereo: Analog Stereo Input (sinks: 0, sources: 1, priority: 60, available: yes) output:analog-stereo: Analog Stereo Output (sinks: 1, sources: 0, priority: 6000, available: yes) output:analog-stereo+input:analog-stereo: Analog Stereo Duplex (sinks: 1, sources: 1, priority: 6060, available: yes) output:analog-surround-40: Analog Surround 4.0 Output (sinks: 1, sources: 0, priority: 700, available: yes) output:analog-surround-40+input:analog-stereo: Analog Surround 4.0 Output + Analog Stereo Input (sinks: 1, sources: 1, priority: 760, available: yes)

1 0

[alsa-devel] [PATCH] ASoC: simple-card: Add snd_card's name parsing from DT node support
by Xiubo Li 16 Jan '14

16 Jan '14

If the DT is used and the CPU DAI device has only one DAI, the card name will be like : ALSA device list: 0: 40031000.sai-sgtl5000 And this name maybe a little ugly to some customers, so here the card name parsing from DT node is supported. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card | 0 sound/soc/generic/simple-card. | 0 sound/soc/generic/simple-card.c | 7 ++++++- 3 files changed, 6 insertions(+), 1 deletion(-) create mode 100644 sound/soc/generic/simple-card create mode 100644 sound/soc/generic/simple-card. diff --git a/sound/soc/generic/simple-card b/sound/soc/generic/simple-card new file mode 100644 index 0000000..e69de29 diff --git a/sound/soc/generic/simple-card. b/sound/soc/generic/simple-card. new file mode 100644 index 0000000..e69de29 diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 0e2fbdd..b40b57c 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -143,6 +143,10 @@ static int asoc_simple_card_parse_of(struct device_node *node, return ret; } + /* parsing the card name from DT */ + snd_soc_of_parse_card_name(&info->snd_card, + "simple-audio-card,name"); + /* CPU sub-node */ ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,cpu"); @@ -257,7 +261,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) /* * init snd_soc_card */ - cinfo->snd_card.name = cinfo->card; + if (!cinfo->snd_card.name) + cinfo->snd_card.name = cinfo->card; cinfo->snd_card.owner = THIS_MODULE; cinfo->snd_card.dai_link = &cinfo->snd_link; cinfo->snd_card.num_links = 1; -- 1.8.4

3 2

[alsa-devel] [PATCHv2] ASoC: simple-card: Add snd_card's name parsing from DT node support
by Xiubo Li 16 Jan '14

16 Jan '14

If the DT is used and the CPU DAI device has only one DAI, the card name will be like : ALSA device list: 0: 40031000.sai-sgtl5000 And this name maybe a little ugly to some customers, so here the card name parsing from DT node is supported. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Resend this patch. sound/soc/generic/simple-card.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index d437c80..711d6a1 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -152,6 +152,10 @@ static int asoc_simple_card_parse_of(struct device_node *node, return ret; } + /* parsing the card name from DT */ + snd_soc_of_parse_card_name(&info->snd_card, + "simple-audio-card,name"); + /* CPU sub-node */ ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,cpu"); @@ -266,7 +270,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) /* * init snd_soc_card */ - cinfo->snd_card.name = cinfo->card; + if (!cinfo->snd_card.name) + cinfo->snd_card.name = cinfo->card; cinfo->snd_card.owner = THIS_MODULE; cinfo->snd_card.dai_link = &cinfo->snd_link; cinfo->snd_card.num_links = 1; -- 1.8.4

1 0

[alsa-devel] Problems with simple-card
by Jean-Francois Moine 16 Jan '14

16 Jan '14

Xiubo, I did a mistake in the v1 of my 'ASoC: simple-card: simplify code': I did not initialize the pointer to the asoc_simple_card_dai_init() function when DT. Then, I fixed that, and the simple card does not work for me. First, without any 'format' in the DT, I get a fmt for each CPU / CODEC DAI: SND_SOC_DAIFMT_CBS_CFS is always set. Well, some code is executed for nothing, but this is not critical. The main problem is about sysclk: when there is no clock in the DT, you get the clock of the CPU or CODEC DAI. In my system, two clocks are declared in the controller CPU DAI, but there is no .set_sysclk pointer. So, snd_soc_dai_set_sysclk() returns -EINVAL and the card is not created. As I don't know why you need these fmt and sysclk, may you have a look at these problems? -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

4 4

[alsa-devel] [PATCH] ASoC: simple-card: simplify the daifmt code
by Xiubo Li 16 Jan '14

16 Jan '14

In the asoc_simple_card_parse_of() will parse the device node's CPU/CODEC DAI commone fmts, and then in asoc_simple_card_sub_parse_of() will parse the CPU/CODEC DAI's sub-node fmts, so we can combine the info->daifmt and info->set.fmt in asoc_simple_card_sub_parse_of() not while just before set_fmt(). Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index b6014fe..0e2fbdd 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -16,15 +16,12 @@ #include <sound/simple_card.h> static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, - struct asoc_simple_dai *set, - unsigned int daifmt) + struct asoc_simple_dai *set) { int ret; - daifmt |= set->fmt; - - if (daifmt) { - ret = snd_soc_dai_set_fmt(dai, daifmt); + if (set->fmt) { + ret = snd_soc_dai_set_fmt(dai, set->fmt); if (ret && ret != -ENOTSUPP) { dev_err(dai->dev, "simple-card: set_fmt error\n"); return ret; @@ -48,14 +45,13 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; - unsigned int daifmt = info->daifmt; int ret; - ret = __asoc_simple_card_dai_init(codec, &info->codec_dai, daifmt); + ret = __asoc_simple_card_dai_init(codec, &info->codec_dai); if (ret < 0) return ret; - ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai, daifmt); + ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai); if (ret < 0) return ret; @@ -64,6 +60,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) static int asoc_simple_card_sub_parse_of(struct device_node *np, + unsigned int daifmt, struct asoc_simple_dai *dai, struct device_node **node) { @@ -89,6 +86,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, * and specific "format" if it has */ dai->fmt = snd_soc_of_parse_daifmt(np, NULL); + dai->fmt |= daifmt; /* * dai->sysclk come from @@ -149,7 +147,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,cpu"); if (np) - ret = asoc_simple_card_sub_parse_of(np, + ret = asoc_simple_card_sub_parse_of(np, info->daifmt, &info->cpu_dai, of_cpu); if (ret < 0) @@ -159,7 +157,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,codec"); if (np) - ret = asoc_simple_card_sub_parse_of(np, + ret = asoc_simple_card_sub_parse_of(np, info->daifmt, &info->codec_dai, of_codec); if (ret < 0) -- 1.8.4

1 0

[alsa-devel] [PATCH] ASoC: simple-card: fix __asoc_simple_card_dai_init
by Xiubo Li 16 Jan '14

16 Jan '14

If the CPU/CODEC DAI set_sysclk() is not support, the -ENOTSUPP will returnd. Here do the check like set_fmt(). Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 2a1b1b5..b6014fe 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -19,22 +19,27 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set, unsigned int daifmt) { - int ret = 0; + int ret; daifmt |= set->fmt; - if (daifmt) + if (daifmt) { ret = snd_soc_dai_set_fmt(dai, daifmt); - - if (ret == -ENOTSUPP) { - dev_dbg(dai->dev, "ASoC: set_fmt is not supported\n"); - ret = 0; + if (ret && ret != -ENOTSUPP) { + dev_err(dai->dev, "simple-card: set_fmt error\n"); + return ret; + } } - if (!ret && set->sysclk) + if (set->sysclk) { ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0); + if (ret && ret != -ENOTSUPP) { + dev_err(dai->dev, "simple-card: set_sysclk error\n"); + return ret; + } + } - return ret; + return 0; } static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) -- 1.8.4

1 0

[alsa-devel] [PATCH 0/3] ASoC: tlv320aic32x4 bugfixes
by Markus Pargmann 16 Jan '14

16 Jan '14

Hi, Another round of tlv320aic32x4 bugfixes. I will send some more non-bugfix patches for this codec next week including the DT compatibles we talked about. Regards, Markus Markus Pargmann (3): ASoC: tlv320aic32x4: Fix regmap range_min ASoC: tlv320aic32x4: Fix mono playback ASoC: tlv320aic32x4: Fix MICPGA input configuration sound/soc/codecs/tlv320aic32x4.c | 30 ++++++++++++++++++------------ sound/soc/codecs/tlv320aic32x4.h | 3 +++ 2 files changed, 21 insertions(+), 12 deletions(-) -- 1.8.5.2

2 4

[alsa-devel] [PATCH] ASoC: core: Return -ENOTSUPP from set_sysclk() if no operation provided
by Mark Brown 15 Jan '14

15 Jan '14

From: Mark Brown <broonie(a)linaro.org> Make it easier for generic code to work with set_sysclk() by distinguishing between the operation not being supported and an error as is done for other operations like set_dai_fmt() Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3a128f0cd752..fe1df50805a3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3485,7 +3485,7 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, return dai->codec->driver->set_sysclk(dai->codec, clk_id, 0, freq, dir); else - return -EINVAL; + return -ENOTSUPP; } EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); @@ -3506,7 +3506,7 @@ int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, return codec->driver->set_sysclk(codec, clk_id, source, freq, dir); else - return -EINVAL; + return -ENOTSUPP; } EXPORT_SYMBOL_GPL(snd_soc_codec_set_sysclk); -- 1.8.5.2

1 0

Re: [alsa-devel] Surround sound not working on ALC887 based motherboard
by Raymond Yau 15 Jan '14

15 Jan '14

2014/1/15 Olivier Crête <olivier.crete(a)ocrete.ca> > Hi, > > > > > I have a Asus P5G41T-M LX Plus motherboard which is ICH7 + > > > Realtek > > > ALC887-VD, alsa only lets me access the first 2 channels > > > (stereo), but > > > not the 6 other ones. It works fine in Windows. > > > > > > Trying to run this after killing pulseaudio fails: > > > > > > $ aplay -D surround40:0 -f S16_LE -r 44100 -c 4 /dev/zero -v > > > Playing raw data '/dev/zero' : Signed 16 bit Little Endian, > > > Rate 44100 Hz, Channels 4 > > > aplay: set_params:1239: Channels count non available > > > > > > > > > Output from alsa-info: > > > > http://www.alsa-project.org/db/?f=fe378d28f0dbb5a427b005d20866a762fd37d31a > > > > > > > > post output of pactl list or pulseaudio verbose log > > > > http://www.ocrete.ca/surround/pactl-list.txt > > http://www.ocrete.ca/surround/pulseaudio-vv.txt > > > > > you have to change channel mode from 2ch to 6ch to retask the input > > > jacks as output if your motherboard only have three audio jacks at > > > rear panel > > > > How would one do that ? Pulseaudio only offers Stereo profiles. > amixer -c0 set "Channel Mode" "6ch" > > > do you mean pulseaudio cannot probe the surroun40 or surround51 > > profile ? > > Yes, but I can't use it with aplay even without pulseaudio running. > > > Simple mixer control 'Channel Mode',0 > > Capabilities: enum > > Items: '2ch' '4ch' '6ch' > > Item0: '2ch' > > > For some reason didn't get your last email, but I saw I saw it on the > archive.. Thanks for your solution, it worked! Definitely something > PulseAudio should expose better. do it really work since rear mic become active port even when it had been retasked as output ? Ports: analog-input-front-mic: Front Microphone (priority: 8500, not available) analog-input-rear-mic: Rear Microphone (priority: 8200, available) analog-input-linein: Line In (priority: 8100, available) Active Port: analog-input-rear-mic

2 1

[alsa-devel] [PATCH] Change the prototype of soc_widget_read
by Arun Shamanna Lakshmi 15 Jan '14

15 Jan '14

soc_widget_read API returns the register data and it is possible that a register can contain 0xffffffff at any point of time. In such cases snd_soc_widget_write is not called after the read operation inside snd_soc_update_bits_locked API. Thus, change the prototype of soc_widget_read to return only the error code and pass the register data through the pointer argument. Arun Shamanna Lakshmi (1): ASoC: dapm: Change prototype of soc_widget_read sound/soc/soc-dapm.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) -- 1.7.9.5

2 2

[alsa-devel] [PATCH 0/2] ASoC: codec tlv320aic32x4 fixes
by Markus Pargmann 15 Jan '14

15 Jan '14

Hi, Here are two fixes for the tlv320aic32x4 codec. The first patch fixes a regmap bug introduced in 3.13. The second is not as important, it adds the codec to the compatible list in the binding documentation. Regards, Markus Markus Pargmann (2): ASoC: codec: tlv320aic32x4: Fix regmap range config ASoC: codec doc, tlv320aic3x: Add tlv320aic32x4 as compatible Documentation/devicetree/bindings/sound/tlv320aic3x.txt | 1 + sound/soc/codecs/tlv320aic32x4.c | 2 +- 2 files changed, 2 insertions(+), 1 deletion(-) -- 1.8.5.2

2 9

[alsa-devel] [PATCH v3 0/6] dma: Indicate residue granularity in dma_slave_caps
by Lars-Peter Clausen 14 Jan '14

14 Jan '14

New in v3: * Expanded the kernel doc documentation for the dma_residue_granularity struct a bit New in v2: * Dropped byte level granularity as discussed during v1 review * Added support for auto-detecting the NO_RESIDUE flag on the compat path in the generic ASoC dmaengine PCM driver * Updated a few ASoC drivers to take advantage of the new API Updated original cover text from v1 below: This patch series introduces a new field to the dma_slave_caps struct which exposes the granularity with which the reported residue is updated, updates the pl330 DMA driver to set this field and lets the generic dmaengine PCM audio driver check that field. The reason for introducing the field is that for audio we need to know the granularity with which the reported residue is updated. The uncertainty of the ALSA period pointer depends on the granularity of the reported residue. E.g. if residue is only updated once every period this means that the position in the buffer from/to which the hardware is currently reading/writing could be anywhere between the reported PCM pointer and the reported PCM pointer plus one period size (plus some extra margin for interrupt latency, etc). Certain algorithms (e.g. pulseaudio's timer based scheduling) only work correctly if the uncertainty of the PCM pointer is low enough. Exposing this information from the DMA driver makes it possible for applications to switch to alternative modes of operation which also work with a high granularity. The patch series introduces four levels of granularity, which I think will cover most DMA cores. I'd like some feedback though if anybody can think of a DMA controller for which none of the levels would be correct. The four levels are: * DESCRIPTOR: The DMA channel is only able to tell whether a descriptor has been completed or not, which means residue reporting is not supported by this channel. The residue field of the dma_tx_state field will always be 0. * SEGMENT: The DMA channel updates the residue field after each successfully completed segment of the transfer (For cyclic transfers this is after each period). This is typically implemented by having the hardware generate an interrupt after each transferred segment and then the driver updates the outstanding residue by the size of the segment. Another possibility is if the hardware supports SG and the segment descriptor has a field which gets set by the hardware after the segment has been completed. The driver then counts the number of segments which do not have the flag set to compute the residue. * BURST: The DMA channel updates the residue field after each transferred burst. This is typically only supported if the hardware has a progress register of some sort (E.g. a register with the current read/write address or a register with the amount of bursts/beats/bytes that have been transferred or still need to be transferred). - Lars Lars-Peter Clausen (6): dma: Indicate residue granularity in dma_slave_caps dma: pl330: Set residue_granularity ASoC: generic-dmaengine-pcm: Check NO_RESIDUE flag at runtime ASoC: generic-dmaengine-pcm: Check DMA residue granularity ASoC: axi-{spdif,i2s}: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag ASoC: samsung: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag drivers/dma/pl330.c | 1 + include/linux/dmaengine.h | 28 +++++++++++++++ sound/soc/adi/axi-i2s.c | 3 +- sound/soc/adi/axi-spdif.c | 3 +- sound/soc/samsung/dmaengine.c | 1 - sound/soc/soc-generic-dmaengine-pcm.c | 66 ++++++++++++++++++++++------------- 6 files changed, 73 insertions(+), 29 deletions(-) -- 1.8.0

2 7

[alsa-devel] [PATCH resend] ASoC: simple-card: fix one bug to writing to the platform data
by Xiubo Li 14 Jan '14

14 Jan '14

It's a bug that writing to the platform data directly, for it should be constant. So just copy it before writing. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Resend this patch. sound/soc/generic/simple-card.c | 40 +++++++++++++++++++++------------------- 1 file changed, 21 insertions(+), 19 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 5528dd6..53395f5 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -9,9 +9,10 @@ * published by the Free Software Foundation. */ #include <linux/clk.h> +#include <linux/module.h> #include <linux/of.h> #include <linux/platform_device.h> -#include <linux/module.h> +#include <linux/string.h> #include <sound/simple_card.h> static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, @@ -190,36 +191,37 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct device_node *of_cpu, *of_codec, *of_platform; struct device *dev = &pdev->dev; + int ret; cinfo = NULL; of_cpu = NULL; of_codec = NULL; of_platform = NULL; + + cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); + if (!cinfo) + return -ENOMEM; + if (np && of_device_is_available(np)) { - cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); - if (cinfo) { - int ret; - cinfo->snd_card.dev = &pdev->dev; - ret = asoc_simple_card_parse_of(np, cinfo, dev, - &of_cpu, - &of_codec, - &of_platform); - if (ret < 0) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "parse error %d\n", ret); - return ret; - } - } else { - return -ENOMEM; + cinfo->snd_card.dev = dev; + + ret = asoc_simple_card_parse_of(np, cinfo, dev, + &of_cpu, + &of_codec, + &of_platform); + if (ret < 0) { + if (ret != -EPROBE_DEFER) + dev_err(dev, "parse error %d\n", ret); + return ret; } } else { - cinfo = pdev->dev.platform_data; - if (!cinfo) { + if (!dev->platform_data) { dev_err(dev, "no info for asoc-simple-card\n"); return -EINVAL; } - cinfo->snd_card.dev = &pdev->dev; + memcpy(cinfo, dev->platform_data, sizeof(*cinfo)); + cinfo->snd_card.dev = dev; } if (!cinfo->name || -- 1.8.4

2 1

[alsa-devel] [PATCH v2 1/5] ASoC: pcm: Properly initialize hw->rate_max
by Lars-Peter Clausen 14 Jan '14

14 Jan '14

If none of the components (CODEC or CPU DAI) sets a maximum sample rate we'll end up with the rate_max field of the runtime hardware set to 0. (Note that it is still possible for the components to constrain the supported sample rates using other methods, e.g. setting a list constraint) If rate_max is 0 this means that the sound card doesn't support any rates at all, which is not the desired result. So initialize rate_max to UINT_MAX. For symmetry reasons also set rate_min to 0. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- No changes in v2 --- sound/soc/soc-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 313adba..b6e0c32 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -252,6 +252,9 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime, & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) hw->rates |= codec_stream->rates; + hw->rate_min = 0; + hw->rate_max = UINT_MAX; + snd_pcm_limit_hw_rates(runtime); hw->rate_min = max(hw->rate_min, cpu_stream->rate_min); -- 1.8.0

3 7

[alsa-devel] [PATCH] ASoC: simple-card: simplify code
by Jean-Francois Moine 14 Jan '14

14 Jan '14

This patch - removes the fields of the platform data which are of no use to the non-DT platform callers, - uses a new private structure to handle all the sound card information, - simplifies the code and make easier a possible multi-DAI links extension. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- Xiubo, I also removed 'of_device_is_available' which seems really useless: the module is not probed when the DT status is not "okay". --- include/sound/simple_card.h | 4 -- sound/soc/generic/simple-card.c | 178 ++++++++++----------- 2 files changed, 88 insertions(+), 94 deletions(-) diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index 6c74527..e1ac996 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -29,10 +29,6 @@ struct asoc_simple_card_info { unsigned int daifmt; struct asoc_simple_dai cpu_dai; struct asoc_simple_dai codec_dai; - - /* used in simple-card.c */ - struct snd_soc_dai_link snd_link; - struct snd_soc_card snd_card; }; #endif /* __SIMPLE_CARD_H */ diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 5528dd6..580900e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -14,6 +14,13 @@ #include <linux/module.h> #include <sound/simple_card.h> +struct simple_card_data { + struct snd_soc_card snd_card; + unsigned int daifmt; + struct asoc_simple_dai cpu_dai; + struct asoc_simple_dai codec_dai; +}; + static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set, unsigned int daifmt) @@ -38,18 +45,18 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct asoc_simple_card_info *info = + struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; - unsigned int daifmt = info->daifmt; + unsigned int daifmt = priv->daifmt; int ret; - ret = __asoc_simple_card_dai_init(codec, &info->codec_dai, daifmt); + ret = __asoc_simple_card_dai_init(codec, &priv->codec_dai, daifmt); if (ret < 0) return ret; - ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai, daifmt); + ret = __asoc_simple_card_dai_init(cpu, &priv->cpu_dai, daifmt); if (ret < 0) return ret; @@ -59,7 +66,8 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) static int asoc_simple_card_sub_parse_of(struct device_node *np, struct asoc_simple_dai *dai, - struct device_node **node) + struct device_node **node, + const char **name) { struct clk *clk; int ret; @@ -73,7 +81,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, return -ENODEV; /* get dai->name */ - ret = snd_soc_of_get_dai_name(np, &dai->name); + ret = snd_soc_of_get_dai_name(np, name); if (ret < 0) goto parse_error; @@ -117,23 +125,21 @@ parse_error: } static int asoc_simple_card_parse_of(struct device_node *node, - struct asoc_simple_card_info *info, - struct device *dev, - struct device_node **of_cpu, - struct device_node **of_codec, - struct device_node **of_platform) + struct simple_card_data *priv, + struct device *dev) { + struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; struct device_node *np; char *name; int ret; /* get CPU/CODEC common format via simple-audio-card,format */ - info->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & + priv->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); /* DAPM routes */ if (of_property_read_bool(node, "simple-audio-card,routing")) { - ret = snd_soc_of_parse_audio_routing(&info->snd_card, + ret = snd_soc_of_parse_audio_routing(&priv->snd_card, "simple-audio-card,routing"); if (ret) return ret; @@ -144,8 +150,10 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,cpu"); if (np) ret = asoc_simple_card_sub_parse_of(np, - &info->cpu_dai, - of_cpu); + &priv->cpu_dai, + (struct device_node **) + &dai_link->cpu_of_node, + &dai_link->cpu_dai_name); if (ret < 0) return ret; @@ -154,109 +162,99 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,codec"); if (np) ret = asoc_simple_card_sub_parse_of(np, - &info->codec_dai, - of_codec); + &priv->codec_dai, + (struct device_node **) + &dai_link->codec_of_node, + &dai_link->codec_dai_name); if (ret < 0) return ret; /* card name is created from CPU/CODEC dai name */ name = devm_kzalloc(dev, - strlen(info->cpu_dai.name) + - strlen(info->codec_dai.name) + 2, + strlen(dai_link->cpu_dai_name) + + strlen(dai_link->codec_dai_name) + 2, GFP_KERNEL); - sprintf(name, "%s-%s", info->cpu_dai.name, info->codec_dai.name); - info->name = info->card = name; + sprintf(name, "%s-%s", dai_link->cpu_dai_name, dai_link->codec_dai_name); + priv->snd_card.name = name; + dai_link->name = dai_link->stream_name = name; /* simple-card assumes platform == cpu */ - *of_platform = *of_cpu; + dai_link->platform_of_node = dai_link->cpu_of_node; - dev_dbg(dev, "card-name : %s\n", info->card); - dev_dbg(dev, "platform : %04x\n", info->daifmt); + dev_dbg(dev, "card-name : %s\n", name); + dev_dbg(dev, "platform : %04x\n", priv->daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", - info->cpu_dai.name, - info->cpu_dai.fmt, - info->cpu_dai.sysclk); + dai_link->cpu_dai_name, + priv->cpu_dai.fmt, + priv->cpu_dai.sysclk); dev_dbg(dev, "codec : %s / %04x / %d\n", - info->codec_dai.name, - info->codec_dai.fmt, - info->codec_dai.sysclk); + dai_link->codec_dai_name, + priv->codec_dai.fmt, + priv->codec_dai.sysclk); return 0; } static int asoc_simple_card_probe(struct platform_device *pdev) { + struct simple_card_data *priv; + struct snd_soc_dai_link *dai_link; struct asoc_simple_card_info *cinfo; struct device_node *np = pdev->dev.of_node; - struct device_node *of_cpu, *of_codec, *of_platform; struct device *dev = &pdev->dev; - cinfo = NULL; - of_cpu = NULL; - of_codec = NULL; - of_platform = NULL; - if (np && of_device_is_available(np)) { - cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); - if (cinfo) { - int ret; - cinfo->snd_card.dev = &pdev->dev; - ret = asoc_simple_card_parse_of(np, cinfo, dev, - &of_cpu, - &of_codec, - &of_platform); - if (ret < 0) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "parse error %d\n", ret); - return ret; - } - } else { - return -ENOMEM; - } - } else { - cinfo = pdev->dev.platform_data; - if (!cinfo) { - dev_err(dev, "no info for asoc-simple-card\n"); + priv = devm_kzalloc(dev, + sizeof(*priv) + sizeof(*dai_link), + GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->snd_card.owner = THIS_MODULE; + priv->snd_card.dev = &pdev->dev; + dai_link = (struct snd_soc_dai_link *) (priv + 1); + priv->snd_card.dai_link = dai_link; + priv->snd_card.num_links = 1; + + cinfo = pdev->dev.platform_data; + if (cinfo) { + if (!cinfo->name || + !cinfo->card || + !cinfo->codec_dai.name || + !cinfo->codec || + !cinfo->platform || + !cinfo->cpu_dai.name) { + dev_err(dev, "insufficient asoc_simple_card_info settings\n"); return -EINVAL; } - cinfo->snd_card.dev = &pdev->dev; - } - - if (!cinfo->name || - !cinfo->card || - !cinfo->codec_dai.name || - !(cinfo->codec || of_codec) || - !(cinfo->platform || of_platform) || - !(cinfo->cpu_dai.name || of_cpu)) { - dev_err(dev, "insufficient asoc_simple_card_info settings\n"); + dai_link->name = cinfo->name; + dai_link->stream_name = cinfo->name; + dai_link->cpu_dai_name = cinfo->cpu_dai.name; + dai_link->platform_name = cinfo->platform; + dai_link->codec_name = cinfo->codec; + dai_link->codec_dai_name = cinfo->codec_dai.name; + memcpy(&priv->cpu_dai, &cinfo->cpu_dai, + sizeof(priv->cpu_dai)); + memcpy(&priv->codec_dai, &cinfo->codec_dai, + sizeof(priv->codec_dai)); + dai_link->init = asoc_simple_card_dai_init; + } else if (np) { + int ret; + + ret = asoc_simple_card_parse_of(np, priv, dev); + if (ret < 0) { + if (ret != -EPROBE_DEFER) + dev_err(dev, "parse error %d\n", ret); + return ret; + } + } else { + dev_err(dev, "no info for asoc-simple-card\n"); return -EINVAL; } - /* - * init snd_soc_dai_link - */ - cinfo->snd_link.name = cinfo->name; - cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; - cinfo->snd_link.platform_name = cinfo->platform; - cinfo->snd_link.codec_name = cinfo->codec; - cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; - cinfo->snd_link.cpu_of_node = of_cpu; - cinfo->snd_link.codec_of_node = of_codec; - cinfo->snd_link.platform_of_node = of_platform; - cinfo->snd_link.init = asoc_simple_card_dai_init; - - /* - * init snd_soc_card - */ - cinfo->snd_card.name = cinfo->card; - cinfo->snd_card.owner = THIS_MODULE; - cinfo->snd_card.dai_link = &cinfo->snd_link; - cinfo->snd_card.num_links = 1; - - snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo); + snd_soc_card_set_drvdata(&priv->snd_card, priv); - return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); + return devm_snd_soc_register_card(&pdev->dev, &priv->snd_card); } static const struct of_device_id asoc_simple_of_match[] = { -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 3

[alsa-devel] [PATCH, RFC 00/30] sleep_on removal
by Arnd Bergmann 14 Jan '14

14 Jan '14

The functions sleep_on, sleep_on_timeout, interruptible_sleep_on and interruptible_sleep_on_timeout have been deprecated for as long as I can remember, and a number of people have contributed patches in the past to remove them from various drivers. This has recently popped up again and I decided to spend some of my time on tracking down the last users and kill it for good. The work is somewhat related to the BKL removal I did a couple of years ago, since most of these drivers were using the BKL in a way that actually made the sleep_on function safe to call. As the BKL was converted into mutexes, the semantics changed slightly and typically opened up a race between checking a condition and going to sleep. I don't have any of the hardware that I'm sending the patches for, so it would be great if someone could test them before they get applied. Otherwise please review very carefully. I'm definitely happy for any patches to go into maintainer trees right away. Obviously the final patch cannot go in until everything else gets merged first and I suspect there will be a series of patches for maintainerless drivers that will go along with it. Arnd Arnd Bergmann (30): ataflop: fix sleep_on races scsi: atari_scsi: fix sleep_on race DAC960: remove sleep_on usage swim3: fix interruptible_sleep_on race [media] omap_vout: avoid sleep_on race [media] usbvision: remove bogus sleep_on_timeout [media] radio-cadet: avoid interruptible_sleep_on race [media] arv: fix sleep_on race staging: serqt_usb2: don't use sleep_on staging: gdm72xx: fix interruptible_sleep_on race staging: panel: fix interruptible_sleep_on race parport: fix interruptible_sleep_on race cris: sync_serial: remove interruptible_sleep_on tty/amiserial: avoid interruptible_sleep_on usbserial: stop using interruptible_sleep_on tty: synclink: avoid sleep_on race atm: nicstar: remove interruptible_sleep_on_timeout atm: firestream: fix interruptible_sleep_on race isdn: pcbit: fix interruptible_sleep_on race isdn: hisax/elsa: fix sleep_on race in elsa FSM isdn: divert, hysdn: fix interruptible_sleep_on race isdn: fix multiple sleep_on races oss: msnd_pinnacle: avoid interruptible_sleep_on_timeout oss: midibuf: fix sleep_on races oss: vwsnd: avoid interruptible_sleep_on oss: dmasound: kill SLEEP() macro to avoid race oss: remove last sleep_on users sgi-xp: open-code interruptible_sleep_on_timeout char: nwbutton: open-code interruptible_sleep_on sched: remove sleep_on() and friends Documentation/DocBook/kernel-hacking.tmpl | 10 ------ arch/cris/arch-v10/drivers/sync_serial.c | 4 ++- arch/cris/arch-v32/drivers/sync_serial.c | 4 ++- drivers/atm/firestream.c | 4 +-- drivers/atm/nicstar.c | 13 ++++---- drivers/block/DAC960.c | 34 +++++++++---------- drivers/block/ataflop.c | 16 ++++----- drivers/block/swim3.c | 18 ++++++---- drivers/char/nwbutton.c | 5 ++- drivers/char/pcmcia/synclink_cs.c | 4 +-- drivers/isdn/divert/divert_procfs.c | 7 ++-- drivers/isdn/hisax/elsa.c | 9 +++-- drivers/isdn/hisax/elsa_ser.c | 3 +- drivers/isdn/hysdn/hysdn_proclog.c | 7 ++-- drivers/isdn/i4l/isdn_common.c | 13 +++++--- drivers/isdn/pcbit/drv.c | 6 ++-- drivers/media/platform/arv.c | 4 +-- drivers/media/platform/omap/omap_vout_vrfb.c | 3 +- drivers/media/radio/radio-cadet.c | 12 +++++-- drivers/media/usb/usbvision/usbvision.h | 4 +-- drivers/misc/sgi-xp/xpc_channel.c | 5 ++- drivers/parport/share.c | 3 +- drivers/scsi/atari_scsi.c | 12 +++++-- drivers/staging/gdm72xx/gdm_usb.c | 5 +-- drivers/staging/panel/panel.c | 4 +-- drivers/staging/serqt_usb2/serqt_usb2.c | 17 ++++------ drivers/tty/amiserial.c | 26 ++++++++++----- drivers/tty/synclink.c | 4 +-- drivers/tty/synclink_gt.c | 4 +-- drivers/tty/synclinkmp.c | 4 +-- drivers/usb/serial/ch341.c | 29 +++++++++++----- drivers/usb/serial/cypress_m8.c | 49 ++++++++++++++++++---------- drivers/usb/serial/f81232.c | 29 +++++++++++----- drivers/usb/serial/pl2303.c | 29 +++++++++++----- include/linux/wait.h | 11 ------- kernel/sched/core.c | 46 -------------------------- sound/oss/dmabuf.c | 14 ++++---- sound/oss/dmasound/dmasound.h | 1 - sound/oss/dmasound/dmasound_core.c | 28 +++++++++++----- sound/oss/midibuf.c | 18 +++++----- sound/oss/msnd_pinnacle.c | 31 ++++++++++-------- sound/oss/sequencer.c | 16 ++++----- sound/oss/sleep.h | 18 ++++++++++ sound/oss/swarm_cs4297a.c | 14 ++++---- sound/oss/vwsnd.c | 14 +++++--- 45 files changed, 334 insertions(+), 277 deletions(-) create mode 100644 sound/oss/sleep.h -- 1.8.3.2 Cc: akpm(a)osdl.org Cc: chas(a)cmf.nrl.navy.mil Cc: davem(a)davemloft.net Cc: geert(a)linux-m68k.org Cc: gregkh(a)linuxfoundation.org Cc: hverkuil(a)xs4all.nl Cc: mingo(a)kernel.org Cc: JBottomley(a)parallels.com Cc: perex(a)perex.cz Cc: axboe(a)kernel.dk Cc: jslaby(a)suse.cz Cc: isdn(a)linux-pingi.de Cc: m.chehab(a)samsung.com Cc: peterz(a)infradead.org Cc: robinmholt(a)gmail.com Cc: tiwai(a)suse.de Cc: alsa-devel(a)alsa-project.org Cc: devel(a)driverdev.osuosl.org Cc: linux-atm-general(a)lists.sourceforge.net Cc: linux-cris-kernel(a)axis.com Cc: linux-media(a)vger.kernel.org Cc: linux-scsi(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Cc: netdev(a)vger.kernel.org

2 6

[alsa-devel] [ISSUE] crackles and clicks
by Eric F 14 Jan '14

14 Jan '14

Please never do the following things on public mailing lists: 1. Take the mailing list address out of Cc. ALWAYS(!) use "reply to all" 2. Top-post. Please quote your reposonse properly, as everyone else does 3. Send HTML-Email. Please use plain-text only. As for your problem, please refer to the the reply I posted to you earlier mail. I'm really having trouble understanding your problem. Please follow the three basic rules above when replying :) I use Gmail, haven't seen a reply to all option... I'm not used to this, pplease excuse me... HERE IS WHAT I SAID : Hello, There is no distortion ! only crackles and clicks. When using it with Jack, no xruns at all. But for testing purpose, I have disabled Jack, mpd, I use only aplay. Dunno if there is a way for buffer stuff with it. Gonna buy a powered USB Hub. Thanks !

1 0

[alsa-devel] [RFC][PATCH] ASoC: simple-card: Add asoc_simple_card_data for the simple card driver data
by Xiubo Li 14 Jan '14

14 Jan '14

Whether the dt is used or not, almost all the simple card information for the DAI link and sound card are initialized in the simple card driver. And for the platform caller, the snd_link and snd_card are of no use, so move them from struct asoc_simple_card_info to struct asoc_simple_card_data. And now only one DAI link is supported for simple card. Suggested-by: Jean-Francois Moine <moinejf(a)free.fr> Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- include/sound/simple_card.h | 4 -- sound/soc/generic/simple-card.c | 115 +++++++++++++++++++++------------------- 2 files changed, 60 insertions(+), 59 deletions(-) diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index 6c74527..e1ac996 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -29,10 +29,6 @@ struct asoc_simple_card_info { unsigned int daifmt; struct asoc_simple_dai cpu_dai; struct asoc_simple_dai codec_dai; - - /* used in simple-card.c */ - struct snd_soc_dai_link snd_link; - struct snd_soc_card snd_card; }; #endif /* __SIMPLE_CARD_H */ diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 2a1b1b5..eab8f92 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -15,6 +15,12 @@ #include <linux/string.h> #include <sound/simple_card.h> +struct asoc_simple_card_data { + struct asoc_simple_card_info info; + struct snd_soc_dai_link snd_link; + struct snd_soc_card snd_card; +}; + static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set, unsigned int daifmt) @@ -39,18 +45,18 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct asoc_simple_card_info *info = + struct asoc_simple_card_data *sdata = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; - unsigned int daifmt = info->daifmt; + unsigned int daifmt = sdata->info.daifmt; int ret; - ret = __asoc_simple_card_dai_init(codec, &info->codec_dai, daifmt); + ret = __asoc_simple_card_dai_init(codec, &sdata->info.codec_dai, daifmt); if (ret < 0) return ret; - ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai, daifmt); + ret = __asoc_simple_card_dai_init(cpu, &sdata->info.cpu_dai, daifmt); if (ret < 0) return ret; @@ -118,23 +124,23 @@ parse_error: } static int asoc_simple_card_parse_of(struct device_node *node, - struct asoc_simple_card_info *info, - struct device *dev, + struct asoc_simple_card_data *sdata, struct device_node **of_cpu, struct device_node **of_codec, struct device_node **of_platform) { + struct device *dev = sdata->snd_card.dev; struct device_node *np; char *name; int ret; /* get CPU/CODEC common format via simple-audio-card,format */ - info->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & - (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); + sdata->info.daifmt = snd_soc_of_parse_daifmt(node, + "simple-audio-card,") & SND_SOC_DAIFMT_FORMAT_MASK; /* DAPM routes */ if (of_property_read_bool(node, "simple-audio-card,routing")) { - ret = snd_soc_of_parse_audio_routing(&info->snd_card, + ret = snd_soc_of_parse_audio_routing(&sdata->snd_card, "simple-audio-card,routing"); if (ret) return ret; @@ -145,7 +151,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,cpu"); if (np) ret = asoc_simple_card_sub_parse_of(np, - &info->cpu_dai, + &sdata->info.cpu_dai, of_cpu); if (ret < 0) return ret; @@ -155,60 +161,60 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,codec"); if (np) ret = asoc_simple_card_sub_parse_of(np, - &info->codec_dai, + &sdata->info.codec_dai, of_codec); if (ret < 0) return ret; - if (!info->cpu_dai.name || !info->codec_dai.name) + if (!sdata->info.cpu_dai.name || !sdata->info.codec_dai.name) return -EINVAL; /* card name is created from CPU/CODEC dai name */ name = devm_kzalloc(dev, - strlen(info->cpu_dai.name) + - strlen(info->codec_dai.name) + 2, + strlen(sdata->info.cpu_dai.name) + + strlen(sdata->info.codec_dai.name) + 2, GFP_KERNEL); - sprintf(name, "%s-%s", info->cpu_dai.name, info->codec_dai.name); - info->name = info->card = name; + sprintf(name, "%s-%s", sdata->info.cpu_dai.name, + sdata->info.codec_dai.name); + sdata->info.name = sdata->info.card = name; /* simple-card assumes platform == cpu */ *of_platform = *of_cpu; - dev_dbg(dev, "card-name : %s\n", info->card); - dev_dbg(dev, "platform : %04x\n", info->daifmt); + dev_dbg(dev, "card-name : %s\n", sdata->info.card); + dev_dbg(dev, "platform : %04x\n", sdata->info.daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", - info->cpu_dai.name, - info->cpu_dai.fmt, - info->cpu_dai.sysclk); + sdata->info.cpu_dai.name, + sdata->info.cpu_dai.fmt, + sdata->info.cpu_dai.sysclk); dev_dbg(dev, "codec : %s / %04x / %d\n", - info->codec_dai.name, - info->codec_dai.fmt, - info->codec_dai.sysclk); + sdata->info.codec_dai.name, + sdata->info.codec_dai.fmt, + sdata->info.codec_dai.sysclk); return 0; } static int asoc_simple_card_probe(struct platform_device *pdev) { - struct asoc_simple_card_info *cinfo; + struct asoc_simple_card_data *sdata; struct device_node *np = pdev->dev.of_node; struct device_node *of_cpu, *of_codec, *of_platform; struct device *dev = &pdev->dev; int ret; - cinfo = NULL; of_cpu = NULL; of_codec = NULL; of_platform = NULL; - cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); - if (!cinfo) + sdata = devm_kzalloc(dev, sizeof(*sdata), GFP_KERNEL); + if (!sdata) return -ENOMEM; - if (np && of_device_is_available(np)) { - cinfo->snd_card.dev = dev; + sdata->snd_card.dev = dev; - ret = asoc_simple_card_parse_of(np, cinfo, dev, + if (np && of_device_is_available(np)) { + ret = asoc_simple_card_parse_of(np, sdata, &of_cpu, &of_codec, &of_platform); @@ -223,16 +229,15 @@ static int asoc_simple_card_probe(struct platform_device *pdev) return -EINVAL; } - memcpy(cinfo, dev->platform_data, sizeof(*cinfo)); - cinfo->snd_card.dev = dev; + memcpy(&sdata->info, dev->platform_data, sizeof(sdata->info)); } - if (!cinfo->name || - !cinfo->card || - !cinfo->codec_dai.name || - !(cinfo->codec || of_codec) || - !(cinfo->platform || of_platform) || - !(cinfo->cpu_dai.name || of_cpu)) { + if (!sdata->info.name || + !sdata->info.card || + !sdata->info.codec_dai.name || + !(sdata->info.codec || of_codec) || + !(sdata->info.platform || of_platform) || + !(sdata->info.cpu_dai.name || of_cpu)) { dev_err(dev, "insufficient asoc_simple_card_info settings\n"); return -EINVAL; } @@ -240,28 +245,28 @@ static int asoc_simple_card_probe(struct platform_device *pdev) /* * init snd_soc_dai_link */ - cinfo->snd_link.name = cinfo->name; - cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; - cinfo->snd_link.platform_name = cinfo->platform; - cinfo->snd_link.codec_name = cinfo->codec; - cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; - cinfo->snd_link.cpu_of_node = of_cpu; - cinfo->snd_link.codec_of_node = of_codec; - cinfo->snd_link.platform_of_node = of_platform; - cinfo->snd_link.init = asoc_simple_card_dai_init; + sdata->snd_link.name = sdata->info.name; + sdata->snd_link.stream_name = sdata->info.name; + sdata->snd_link.cpu_dai_name = sdata->info.cpu_dai.name; + sdata->snd_link.platform_name = sdata->info.platform; + sdata->snd_link.codec_name = sdata->info.codec; + sdata->snd_link.codec_dai_name = sdata->info.codec_dai.name; + sdata->snd_link.cpu_of_node = of_cpu; + sdata->snd_link.codec_of_node = of_codec; + sdata->snd_link.platform_of_node = of_platform; + sdata->snd_link.init = asoc_simple_card_dai_init; /* * init snd_soc_card */ - cinfo->snd_card.name = cinfo->card; - cinfo->snd_card.owner = THIS_MODULE; - cinfo->snd_card.dai_link = &cinfo->snd_link; - cinfo->snd_card.num_links = 1; + sdata->snd_card.name = sdata->info.card; + sdata->snd_card.owner = THIS_MODULE; + sdata->snd_card.dai_link = &sdata->snd_link; + sdata->snd_card.num_links = 1; - snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo); + snd_soc_card_set_drvdata(&sdata->snd_card, sdata); - return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); + return devm_snd_soc_register_card(dev, &sdata->snd_card); } static const struct of_device_id asoc_simple_of_match[] = { -- 1.8.4

4 4

[alsa-devel] [PATCH] ALSA: hda - Fix endless vmaster hook call in thinkpad_helper.c
by Takashi Iwai 14 Jan '14

14 Jan '14

The new vmaster hook, update_tpacpi_mute_led(), calls the original vmaster hook, but I forgot to save the original hook function but keep calling the updated one, which of course results in a stupid endless loop. Fixed now. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/thinkpad_helper.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 8492b8a3ce47..5799fbc24c28 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -8,6 +8,7 @@ #include <linux/thinkpad_acpi.h> static int (*led_set_func)(int, bool); +static void (*old_vmaster_hook)(void *, int); static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context, void **rv) @@ -30,11 +31,8 @@ static bool is_thinkpad(struct hda_codec *codec) static void update_tpacpi_mute_led(void *private_data, int enabled) { - struct hda_codec *codec = private_data; - struct hda_gen_spec *spec = codec->spec; - - if (spec->vmaster_mute.hook) - spec->vmaster_mute.hook(private_data, enabled); + if (old_vmaster_hook) + old_vmaster_hook(private_data, enabled); if (led_set_func) led_set_func(TPACPI_LED_MUTE, !enabled); @@ -70,6 +68,7 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, removefunc = true; if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { + old_vmaster_hook = spec->vmaster_mute.hook; spec->vmaster_mute.hook = update_tpacpi_mute_led; removefunc = false; } @@ -86,6 +85,7 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) { symbol_put(tpacpi_led_set); led_set_func = NULL; + old_vmaster_hook = NULL; } } -- 1.8.5.2

1 0

[alsa-devel] [PATCH] ALSA: snd-usb: re-order some quirk entries
by Daniel Mack 14 Jan '14

14 Jan '14

No code change, just a cosmetic cleanup to keep entries ordered by the device ID within a block of unique vendor IDs. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/quirks-table.h | 170 +++++++++++++++++++++++------------------------ 1 file changed, 84 insertions(+), 86 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 3454780..f652b10 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -72,22 +72,21 @@ } }, -/* Creative/Toshiba Multimedia Center SB-0500 */ +/* Creative/E-Mu devices */ { - USB_DEVICE(0x041e, 0x3048), + USB_DEVICE(0x041e, 0x3010), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Toshiba", - .product_name = "SB-0500", + .vendor_name = "Creative Labs", + .product_name = "Sound Blaster MP3+", .ifnum = QUIRK_NO_INTERFACE } }, - -/* Creative/E-Mu devices */ +/* Creative/Toshiba Multimedia Center SB-0500 */ { - USB_DEVICE(0x041e, 0x3010), + USB_DEVICE(0x041e, 0x3048), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Creative Labs", - .product_name = "Sound Blaster MP3+", + .vendor_name = "Toshiba", + .product_name = "SB-0500", .ifnum = QUIRK_NO_INTERFACE } }, @@ -2521,17 +2520,17 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE(0x1235, 0x0018), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Novation", - .product_name = "Twitch", + USB_DEVICE(0x1235, 0x0010), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Focusrite", + .product_name = "Saffire 6 USB", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { + .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 4, .iface = 0, @@ -2561,26 +2560,17 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x1235, 0x4661), + USB_DEVICE(0x1235, 0x0018), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Novation", - .product_name = "ReMOTE25", - .ifnum = 0, - .type = QUIRK_MIDI_NOVATION - } -}, -{ - USB_DEVICE(0x1235, 0x0010), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Focusrite", - .product_name = "Saffire 6 USB", + .product_name = "Twitch", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + .data = & (const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 4, .iface = 0, @@ -2609,6 +2599,66 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE_VENDOR_SPEC(0x1235, 0x4661), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Novation", + .product_name = "ReMOTE25", + .ifnum = 0, + .type = QUIRK_MIDI_NOVATION + } +}, +{ + /* + * Focusrite Scarlett 18i6 + * + * Avoid mixer creation, which otherwise fails because some of + * the interface descriptor subtypes for interface 0 are + * unknown. That should be fixed or worked-around but this at + * least allows the device to be used successfully with a DAW + * and an external mixer. See comments below about other + * ignored interfaces. + */ + USB_DEVICE(0x1235, 0x8004), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Focusrite", + .product_name = "Scarlett 18i6", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = & (const struct snd_usb_audio_quirk[]) { + { + /* InterfaceSubClass 1 (Control Device) */ + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + /* InterfaceSubClass 1 (Control Device) */ + .ifnum = 3, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 4, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + /* InterfaceSubClass 1 (Device Firmware Update) */ + .ifnum = 5, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, /* Access Music devices */ { @@ -2711,7 +2761,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7240), + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7210), .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, @@ -2719,13 +2769,13 @@ YAMAHA_DEVICE(0x7010, "UB99"), .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", - .product_name = "HVR-850", + .product_name = "HVR-950Q", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_AUDIO_ALIGN_TRANSFER, } }, { - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7210), + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7217), .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, @@ -2739,7 +2789,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7217), + USB_DEVICE_VENDOR_SPEC(0x2040, 0x721b), .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, @@ -2753,7 +2803,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x2040, 0x721b), + USB_DEVICE_VENDOR_SPEC(0x2040, 0x721e), .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, @@ -2767,7 +2817,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x2040, 0x721e), + USB_DEVICE_VENDOR_SPEC(0x2040, 0x721f), .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, @@ -2781,7 +2831,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x2040, 0x721f), + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7240), .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, @@ -2789,7 +2839,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", + .product_name = "HVR-850", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_AUDIO_ALIGN_TRANSFER, } @@ -3094,58 +3144,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* - * Focusrite Scarlett 18i6 - * - * Avoid mixer creation, which otherwise fails because some of - * the interface descriptor subtypes for interface 0 are - * unknown. That should be fixed or worked-around but this at - * least allows the device to be used successfully with a DAW - * and an external mixer. See comments below about other - * ignored interfaces. - */ - USB_DEVICE(0x1235, 0x8004), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Focusrite", - .product_name = "Scarlett 18i6", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = & (const struct snd_usb_audio_quirk[]) { - { - /* InterfaceSubClass 1 (Control Device) */ - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - /* InterfaceSubClass 1 (Control Device) */ - .ifnum = 3, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 4, - .type = QUIRK_MIDI_STANDARD_INTERFACE - }, - { - /* InterfaceSubClass 1 (Device Firmware Update) */ - .ifnum = 5, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = -1 - } - } - } -}, - -{ - /* * Some USB MIDI devices don't have an audio control interface, * so we have to grab MIDI streaming interfaces here. */ -- 1.8.4.2

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Fix Creative VF0420 rate
by Pavel Hofman 14 Jan '14

14 Jan '14

>From db1dcaf0f3c2459b02b29c55639a9a6943cffec9 Mon Sep 17 00:00:00 2001 From: Pavel Hofman <pavel.hofman(a)ivitera.com> Date: Tue, 14 Jan 2014 13:42:57 +0100 Subject: [PATCH] ALSA: usb-audio: Fix Creative VF0420 rate Creative Live! Cam Vista IM (VF0420) reports rate of 16kHz while working at 8kHz. The patch adds its USB ID to the existing quirk. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> --- sound/usb/format.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index 3525231..d244fd3 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -189,8 +189,10 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof chip->usb_id == USB_ID(0x0ccd, 0x00b1)) && fp->altsetting == 5 && fp->maxpacksize == 392) rate = 96000; - /* Creative VF0470 Live Cam reports 16 kHz instead of 8kHz */ - if (rate == 16000 && chip->usb_id == USB_ID(0x041e, 0x4068)) + /* Creative VF0420/VF0470 Live Cams report 16 kHz instead of 8kHz */ + if (rate == 16000 && + (chip->usb_id == USB_ID(0x041e, 0x4064) || + chip->usb_id == USB_ID(0x041e, 0x4068))) rate = 8000; fp->rate_table[fp->nr_rates] = rate; -- 1.7.1

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Add support for Focusrite Saffire 6 USB
by Eduard Gilmutdinov 14 Jan '14

14 Jan '14

Signed-off-by: Eduard Gilmutdinov <edgilmutdinov(a)gmail.com> --- sound/usb/quirks-table.h | 40 ++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 9 +++++---- 2 files changed, 45 insertions(+), 4 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f5f0595..3454780 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2569,6 +2569,46 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_NOVATION } }, +{ + USB_DEVICE(0x1235, 0x0010), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Focusrite", + .product_name = "Saffire 6 USB", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = 1, + .type = QUIRK_MIDI_RAW_BYTES + }, + { + .ifnum = -1 + } + } + } +}, /* Access Music devices */ { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 0df9ede..e9a59e9 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -662,8 +662,9 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) /* * Novation Twitch DJ controller + * Focusrite Novation Saffire 6 USB audio card */ -static int snd_usb_twitch_boot_quirk(struct usb_device *dev) +static int snd_usb_novation_boot_quirk(struct usb_device *dev) { /* preemptively set up the device because otherwise the * raw MIDI endpoints are not active */ @@ -972,9 +973,9 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, /* Digidesign Mbox 2 */ return snd_usb_mbox2_boot_quirk(dev); - case USB_ID(0x1235, 0x0018): - /* Focusrite Novation Twitch */ - return snd_usb_twitch_boot_quirk(dev); + case USB_ID(0x1235, 0x0010): /* Focusrite Novation Saffire 6 USB */ + case USB_ID(0x1235, 0x0018): /* Focusrite Novation Twitch */ + return snd_usb_novation_boot_quirk(dev); case USB_ID(0x133e, 0x0815): /* Access Music VirusTI Desktop */ -- 1.8.5.2

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[alsa-devel] [PATCHv2 0/6] One simple card use case
by Xiubo Li 14 Jan '14

14 Jan '14

This patch series is one use case of simple card driver based on dts. For some audio cards they need to add some off-CODEC widgets. For the audio card, if it needs off-CODEC widgets, some explict code for a simple audio is needed, if not, just ignore of this. This can make the audio card driver simpler and easier to debug by using the simple card driver and some explict code if needed.

3 8

[alsa-devel] [PATCH v2] ALSA: hda - automute via amp instead of pinctl on some AIO models
by Hui Wang 14 Jan '14

14 Jan '14

On some AIO (All In One) models with the codec alc668 (Vendor ID: 0x10ec0668) on it, when we plug a headphone into the jack, the system will switch the output to headphone and set the speaker to automute as well as change the speaker Pin-ctls from 0x40 to 0x00, this will bring loud noise to the headphone. I tried to disable the corresponding EAPD, but it did not help to eliminate the noise. According to Takashi's suggestion, we use amp operation to replace the pinctl modification for the automute, this really eliminate the noise. BugLink: https://bugs.launchpad.net/bugs/1268468 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 22 +++++++++++++++++++--- 1 file changed, 19 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d35cf3d..a1c16457 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3580,6 +3580,15 @@ static void alc_fixup_headset_mode_alc255(struct hda_codec *codec, alc_fixup_headset_mode(codec, fix, action); } +static void alc_fixup_auto_mute_via_amp(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct alc_spec *spec = codec->spec; + spec->gen.auto_mute_via_amp = 1; + } +} + static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4839,6 +4848,7 @@ enum { ALC662_FIXUP_BASS_CHMAP, ALC662_FIXUP_BASS_1A, ALC662_FIXUP_BASS_1A_CHMAP, + ALC668_FIXUP_AUTO_MUTE, }; static const struct hda_fixup alc662_fixups[] = { @@ -4999,6 +5009,12 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, }, + [ALC668_FIXUP_AUTO_MUTE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_auto_mute_via_amp, + .chained = true, + .chain_id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE + }, [ALC668_FIXUP_DELL_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5044,11 +5060,11 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_AUTO_MUTE), + SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A_CHMAP), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP), -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: add snd_soc_of_parse_audio_simple_widgets for DeviceTree
by Xiubo Li 14 Jan '14

14 Jan '14

This patch adds snd_soc_of_parse_audio_simple_widgets() and supports below style of widgets name on DT. "wname-prefix[ individual name]" "wname-prefix" includes: "Mic", "Line", "Hp", "Spk"... For instance: simple-audio-widgets = "Mic Jack", "Line In Jack", "Hp Jack", "Spk Ext", "Line Out Jack"; Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 62 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 64 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 03ce45b..b2b5900 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1172,6 +1172,8 @@ void snd_soc_util_exit(void); int snd_soc_of_parse_card_name(struct snd_soc_card *card, const char *propname); +int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, + const char *propname); int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname); unsigned int snd_soc_of_parse_daifmt(struct device_node *np, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3a128f0..36cadd3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4417,6 +4417,68 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_card_name); +static struct snd_soc_dapm_widget simple_widgets[] = { + SND_SOC_DAPM_MIC("Mic", NULL), + SND_SOC_DAPM_LINE("Line", NULL), + SND_SOC_DAPM_HP("Hp", NULL), + SND_SOC_DAPM_SPK("Spk", NULL), +}; + +int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, + const char *propname) +{ + struct device_node *np = card->dev->of_node; + struct snd_soc_dapm_widget *widgets; + const char *wname; + int i, j, cnt, ret; + + cnt = of_property_count_strings(np, propname); + if (cnt <= 0) { + dev_err(card->dev, + "ASoC: Property '%s' does not exist or " + "length is zero\n", propname); + return -EINVAL; + } + + widgets = devm_kcalloc(card->dev, cnt, sizeof(*widgets), GFP_KERNEL); + if (!widgets) { + dev_err(card->dev, + "ASoC: Could not allocate DAPM widgets table\n"); + return -ENOMEM; + } + + for (i = 0; i < cnt; i++) { + ret = of_property_read_string_index(np, propname, i, &wname); + if (ret) { + dev_err(card->dev, + "ASoC: Property '%s' index %d could not be read: %d\n", + propname, i, ret); + return -EINVAL; + } + + for (j = 0; j < ARRAY_SIZE(simple_widgets); j++) { + if (!strncmp(wname, simple_widgets[j].name, + strlen(simple_widgets[j].name))) { + widgets[i] = simple_widgets[j]; + break; + } + } + + if (j >= ARRAY_SIZE(simple_widgets)) { + dev_err(card->dev, + "ASoC: DAPM widget : '%s' is not supported\n", + wname); + return -EINVAL; + } + } + + card->dapm_widgets = widgets; + card->num_dapm_widgets = cnt; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets); + int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname) { -- 1.8.4

1 0

[alsa-devel] Surround sound not working on ALC887 based motherboard
by Olivier Crête 14 Jan '14

14 Jan '14

Hello, I have a Asus P5G41T-M LX Plus motherboard which is ICH7 + Realtek ALC887-VD, alsa only lets me access the first 2 channels (stereo), but not the 6 other ones. It works fine in Windows. Trying to run this after killing pulseaudio fails: $ aplay -D surround40:0 -f S16_LE -r 44100 -c 4 /dev/zero -v Playing raw data '/dev/zero' : Signed 16 bit Little Endian, Rate 44100 Hz, Channels 4 aplay: set_params:1239: Channels count non available Output from alsa-info: http://www.alsa-project.org/db/?f=fe378d28f0dbb5a427b005d20866a762fd37d31a -- Olivier Crête olivier.crete(a)ocrete.ca

2 3

[alsa-devel] [PATCH] ALSA: hda - Prevent speaker from entering automute on some AIO models
by Hui Wang 14 Jan '14

14 Jan '14

On some AIO (All In One) models with the codec alc668 (Vendor ID: 0x10ec0668) on it, when we plug a headphone into the jack, the system will switch the output to headphone and set the speaker to automute as well as change the speaker Pin-ctls from 0x40 to 0x00, this will bring loud noise to the headphone. I tried to disable the corresponding EAPD, but it did not help to eliminate the noise. So far, the only way to eliminate the noise is preventing the speaker from entering automute. BugLink: https://bugs.launchpad.net/bugs/1268468 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- This problem has been reported on three Dell AIO models. sound/pci/hda/patch_realtek.c | 24 +++++++++++++++++++++--- 1 file changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6a705a8..27111d0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3589,6 +3589,17 @@ static void alc_fixup_headset_mode_alc255(struct hda_codec *codec, alc_fixup_headset_mode(codec, fix, action); } +static void alc_fixup_spk_automute_alc668(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct alc_spec *spec = codec->spec; + /* On some AIO (All In One) machines, setting the speaker + to audomute will bring the noise to the headphone. */ + spec->gen.suppress_auto_mute = 1; + } +} + static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4935,6 +4946,7 @@ enum { ALC662_FIXUP_BASS_CHMAP, ALC662_FIXUP_BASS_1A, ALC662_FIXUP_BASS_1A_CHMAP, + ALC668_FIXUP_AUTO_MUTE, }; static const struct hda_fixup alc662_fixups[] = { @@ -5095,6 +5107,12 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, }, + [ALC668_FIXUP_AUTO_MUTE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_spk_automute_alc668, + .chained = true, + .chain_id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE + }, [ALC668_FIXUP_DELL_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5140,11 +5158,11 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_AUTO_MUTE), + SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A_CHMAP), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP), -- 1.8.1.2

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[alsa-devel] Possible regression of Mackerel/ASoC Simple Card in next-20130107
by Simon Horman 14 Jan '14

14 Jan '14

Hi Morimoto-san, Hi all, I noticed that next-20130107 does not boot on the Mackerel board does not boot using its defconfig. The boot log is as follows. I have not investigated any further as typically when I investigate problems in next they turn out to be known problems. Linux version 3.13.0-rc7-next-20140107-00005-gad2dc8a (horms(a)ayumi.isobedori.kobe.vergenet.net) (gcc version 4.7.2 (Debian 4.7.2-4) ) #771 Wed Jan 8 12:16:31 JST 2014 CPU: ARMv7 Processor [413fc081] revision 1 (ARMv7), cr=10c53c7d CPU: PIPT / VIPT nonaliasing data cache, VIPT aliasing instruction cache Machine model: Mackerel (AP4 EVM 2nd) Memory policy: Data cache writeback CPU: All CPU(s) started in SVC mode. bootconsole [early_ttySC0] enabled Built 1 zonelists in Zone order, mobility grouping off. Total pages: 60960 Kernel command line: console=tty0, console=ttySC0,115200 earlyprintk=sh-sci.0,115200 root=/dev/nfs nfsroot=,tcp,v3 ip=dhcp mem=240m rw PID hash table entries: 1024 (order: 0, 4096 bytes) Dentry cache hash table entries: 32768 (order: 5, 131072 bytes) Inode-cache hash table entries: 16384 (order: 4, 65536 bytes) Memory: 238412K/245760K available (3419K kernel code, 200K rwdata, 1072K rodata, 148K init, 174K bss, 7348K reserved) Virtual kernel memory layout: vector : 0xffff0000 - 0xffff1000 ( 4 kB) fixmap : 0xfff00000 - 0xfffe0000 ( 896 kB) vmalloc : 0xcf800000 - 0xff000000 ( 760 MB) lowmem : 0xc0000000 - 0xcf000000 ( 240 MB) modules : 0xbf000000 - 0xc0000000 ( 16 MB) .text : 0xc0008000 - 0xc046bf6c (4496 kB) .init : 0xc046c000 - 0xc04910d0 ( 149 kB) .data : 0xc0492000 - 0xc04c41a0 ( 201 kB) .bss : 0xc04c41a0 - 0xc04efb08 ( 175 kB) NR_IRQS:16 nr_irqs:16 16 intc: Registered controller 'sh7372-intca' with 108 IRQs intc: Registered controller 'sh7372-intca-irq-lo' with 16 IRQs intc: Registered controller 'sh7372-intca-irq-hi' with 16 IRQs intc: Registered controller 'sh7372-intcs' with 61 IRQs sched_clock: 32 bits at 128 Hz, resolution 7812500ns, wraps every 16777216000000000ns Console: colour dummy device 80x30 console [tty0] enabled sh_tmu.0: used for clock events sh_tmu.0: used for periodic clock events sh_tmu.1: used as clock source Calibrating delay loop... 805.59 BogoMIPS (lpj=3145728) pid_max: default: 32768 minimum: 301 Mount-cache hash table entries: 512 CPU: Testing write buffer coherency: ok Setting up static identity map for 0x4035d778 - 0x4035d7d0 VFP support v0.3: implementor 41 architecture 3 part 30 variant c rev 3 pinctrl core: initialized pinctrl subsystem regulator-dummy: no parameters NET: Registered protocol family 16 DMA: preallocated 256 KiB pool for atomic coherent allocations sh-pfc pfc-sh7372: sh7372_pfc handling gpio 0 -> 190 sh-pfc pfc-sh7372: sh7372_pfc support registered No ATAGs? bio: create slab <bio-0> at 0 fixed-1.8V: at 1800 mV fixed-3.3V: at 3300 mV fixed-dummy: no parameters SCSI subsystem initialized usbcore: registered new interface driver usbfs usbcore: registered new interface driver hub usbcore: registered new device driver usb i2c-sh_mobile i2c-sh_mobile.0: I2C adapter 0 with bus speed 100000 Hz (L/H=f0/ce) i2c-sh_mobile i2c-sh_mobile.1: I2C adapter 1 with bus speed 100000 Hz (L/H=f0/ce) sh_cmt sh_cmt.2: used for clock events sh_cmt sh_cmt.2: used as clock source sh_tmu sh_tmu.0: kept as earlytimer sh_tmu sh_tmu.1: kept as earlytimer Advanced Linux Sound Architecture Driver Initialized. Switched to clocksource sh_tmu.1 NET: Registered protocol family 2 TCP established hash table entries: 2048 (order: 1, 8192 bytes) TCP bind hash table entries: 2048 (order: 1, 8192 bytes) TCP: Hash tables configured (established 2048 bind 2048) TCP: reno registered UDP hash table entries: 256 (order: 0, 4096 bytes) UDP-Lite hash table entries: 256 (order: 0, 4096 bytes) NET: Registered protocol family 1 RPC: Registered named UNIX socket transport module. RPC: Registered udp transport module. RPC: Registered tcp transport module. RPC: Registered tcp NFSv4.1 backchannel transport module. NFS: Registering the id_resolver key type Key type id_resolver registered Key type id_legacy registered nfs4filelayout_init: NFSv4 File Layout Driver Registering... msgmni has been set to 465 io scheduler noop registered (default) sh-mobile-hdmi sh-mobile-hdmi: Detected HDMI controller 0x0:0x0 Console: switching to colour frame buffer device 100x30 sh_mobile_lcdc_fb sh_mobile_lcdc_fb.0: registered sh_mobile_lcdc_fb.0/mainlcd as 800x480 16bpp. sh_mobile_lcdc_fb sh_mobile_lcdc_fb.1: registered sh_mobile_lcdc_fb.1/mainlcd as 1280x720 16bpp. SuperH (H)SCI(F) driver initialized sh-sci.0: ttySC0 at MMIO 0xe6c40000 (irq = 80, base_baud = 0) is a scifa console [ttySC0] enabled console [ttySC0] enabled bootconsole [early_ttySC0] disabled bootconsole [early_ttySC0] disabled sh-sci.1: ttySC1 at MMIO 0xe6c50000 (irq = 81, base_baud = 0) is a scifa sh-sci.2: ttySC2 at MMIO 0xe6c60000 (irq = 82, base_baud = 0) is a scifa sh-sci.3: ttySC3 at MMIO 0xe6c70000 (irq = 83, base_baud = 0) is a scifa sh-sci.4: ttySC4 at MMIO 0xe6c80000 (irq = 89, base_baud = 0) is a scifa sh-sci.5: ttySC5 at MMIO 0xe6cb0000 (irq = 90, base_baud = 0) is a scifa sh-sci.6: ttySC6 at MMIO 0xe6c30000 (irq = 91, base_baud = 0) is a scifb physmap platform flash device: 08000000 at 20000000 physmap-flash.0: Found 1 x16 devices at 0x0 in 16-bit bank. Manufacturer ID 0x000089 Chip ID 0x0088b1 Intel/Sharp Extended Query Table at 0x010A Intel/Sharp Extended Query Table at 0x010A Intel/Sharp Extended Query Table at 0x010A Intel/Sharp Extended Query Table at 0x010A Intel/Sharp Extended Query Table at 0x010A Using buffer write method Using auto-unlock on power-up/resume cfi_cmdset_0001: Erase suspend on write enabled Creating 5 MTD partitions on "physmap-flash.0": 0x000000000000-0x000000080000 : "loader" 0x000000080000-0x000000100000 : "bootenv" 0x000000100000-0x000000900000 : "kernel_ro" 0x000000900000-0x000001100000 : "kernel" 0x000001100000-0x000008000000 : "data" smsc911x: Driver version 2008-10-21 smsc911x smsc911x (unregistered net_device): couldn't get clock -2 libphy: smsc911x-mdio: probed smsc911x smsc911x eth0: attached PHY driver [Generic PHY] (mii_bus:phy_addr=smsc911x-fffffff:01, irq=-1) smsc911x smsc911x eth0: MAC Address: 74:90:50:00:00:ac usbcore: registered new interface driver usb-storage renesas_usbhs renesas_usbhs.0: host probed renesas_usbhs renesas_usbhs.0: gadget probed renesas_usbhs renesas_usbhs.0: probed renesas_usbhs renesas_usbhs.1: host probed renesas_usbhs renesas_usbhs.1: gadget probed renesas_usbhs renesas_usbhs.1: probed renesas_usbhs renesas_usbhs.1: USB Host Controller renesas_usbhs renesas_usbhs.1: new USB bus registered, assigned bus number 1 mousedev: PS/2 mouse device common for all mice hub 1-0:1.0: USB hub found hub 1-0:1.0: 1 port detected sh_mobile_sdhi sh_mobile_sdhi.0: mmc0 base at 0xe6850000 clock rate 99 MHz sh_mobile_sdhi sh_mobile_sdhi.2: mmc1 base at 0xe6870000 clock rate 99 MHz sh_mmcif sh_mmcif.0: driver version 2010-04-28 usbcore: registered new interface driver usbhid usbhid: USB HID core driver Unable to handle kernel NULL pointer dereference at virtual address 0000008c pgd = c0004000 [0000008c] *pgd=00000000 Internal error: Oops: 805 [#1] ARM Modules linked in: CPU: 0 PID: 1 Comm: swapper Not tainted 3.13.0-rc7-next-20140107-00005-gad2dc8a #771 task: c8041bc0 ti: c8042000 task.ti: c8042000 PC is at asoc_simple_card_probe+0x44/0x2a4 LR is at platform_drv_probe+0x20/0x50 pc : [<c02b1648>] lr : [<c01f33b0>] psr: 60000113 sp : c8043da0 ip : c8043dd0 fp : c8043dcc r10: c048a278 r9 : c8042000 r8 : 00000000 r7 : 00000000 r6 : c04a5a50 r5 : c04bf58c r4 : c04a5a40 r3 : c02b1604 r2 : 00000000 r1 : 00000000 r0 : c04a5a40 Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c5387d Table: 40004019 DAC: 00000015 Process swapper (pid: 1, stack limit = 0xc8042238) Stack: (0xc8043da0 to 0xc8044000) 3da0: 00000000 00000000 c04a5a50 c04bf58c 00000000 c04bf58c c0482c2c c048a278 3dc0: c8043de4 c8043dd0 c01f33b0 c02b1610 c04e4af0 c04a5a50 c8043e04 c8043de8 3de0: c01f1e28 c01f339c c04a5a50 c04bf58c c04a5a84 00000000 c8043e24 c8043e08 3e00: c01f1fd4 c01f1d20 c807f5bc 00000000 c04bf58c c01f1f40 c8043e4c c8043e28 3e20: c01f057c c01f1f4c c8030e4c c807f5b0 c839b6b4 c04bf58c c839b680 c04b6f30 3e40: c8043e5c c8043e50 c01f1934 c01f052c c8043e84 c8043e60 c01f1538 c01f191c 3e60: c0412b28 c8043e70 c04bf58c 00000006 c04c41c0 00000070 c8043e9c c8043e88 3e80: c01f265c c01f1468 c0491070 00000006 c8043eac c8043ea0 c01f3304 c01f25e8 3ea0: c8043ebc c8043eb0 c0482c44 c01f32c0 c8043f4c c8043ec0 c00094e8 c0482c38 3ec0: 00000264 c046ab4c 00000000 00000000 c8043efc c8043ee0 c8043f04 c8043ee8 3ee0: c8043f00 c8043ef0 c046c4d0 c06f0daf c0386f80 00000070 c8043f4c c8043f08 3f00: c00348bc c046c4dc c048a250 00000006 00000006 c046a44c 00000000 c043536c 3f20: c8043f4c c0491070 00000006 c04c41c0 00000070 c046c4d0 c048a26c c048a278 3f40: c8043f94 c8043f50 c046cb50 c0009400 00000006 00000006 c046c4d0 16a3cbde 3f60: a0080220 6244bd2d 21509030 00000000 c035765c 00000000 00000000 00000000 3f80: 00000000 00000000 c8043fac c8043f98 c035766c c046ca6c 00000000 00000000 3fa0: 00000000 c8043fb0 c000f478 c0357668 00000000 00000000 00000000 00000000 3fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 3fe0: 00000000 00000000 00000000 00000000 00000013 00000000 00000002 54f1c041 Backtrace: [<c02b1604>] (asoc_simple_card_probe) from [<c01f33b0>] (platform_drv_probe+0x20/0x50) r10:c048a278 r8:c0482c2c r7:c04bf58c r6:00000000 r5:c04bf58c r4:c04a5a50 [<c01f3390>] (platform_drv_probe) from [<c01f1e28>] (driver_probe_device+0x114/0x22c) r5:c04a5a50 r4:c04e4af0 [<c01f1d14>] (driver_probe_device) from [<c01f1fd4>] (__driver_attach+0x94/0x98) r7:00000000 r6:c04a5a84 r5:c04bf58c r4:c04a5a50 [<c01f1f40>] (__driver_attach) from [<c01f057c>] (bus_for_each_dev+0x5c/0x90) r6:c01f1f40 r5:c04bf58c r4:00000000 r3:c807f5bc [<c01f0520>] (bus_for_each_dev) from [<c01f1934>] (driver_attach+0x24/0x28) r6:c04b6f30 r5:c839b680 r4:c04bf58c [<c01f1910>] (driver_attach) from [<c01f1538>] (bus_add_driver+0xdc/0x1d8) [<c01f145c>] (bus_add_driver) from [<c01f265c>] (driver_register+0x80/0xfc) r7:00000070 r6:c04c41c0 r5:00000006 r4:c04bf58c [<c01f25dc>] (driver_register) from [<c01f3304>] (__platform_driver_register+0x50/0x64) r5:00000006 r4:c0491070 [<c01f32b4>] (__platform_driver_register) from [<c0482c44>] (asoc_simple_card_init+0x18/0x20) [<c0482c2c>] (asoc_simple_card_init) from [<c00094e8>] (do_one_initcall+0xf4/0x150) [<c00093f4>] (do_one_initcall) from [<c046cb50>] (kernel_init_freeable+0xf0/0x1b8) r10:c048a278 r9:c048a26c r8:c046c4d0 r7:00000070 r6:c04c41c0 r5:00000006 r4:c0491070 [<c046ca60>] (kernel_init_freeable) from [<c035766c>] (kernel_init+0x10/0x118) r10:00000000 r9:00000000 r8:00000000 r7:00000000 r6:00000000 r5:c035765c r4:00000000 [<c035765c>] (kernel_init) from [<c000f478>] (ret_from_fork+0x14/0x3c) r4:00000000 r3:00000000 Code: ebff5df2 e1500007 1a000031 e3a07000 (e587608c) ---[ end trace fa979cfc03b9a020 ]--- Kernel panic - not syncing: Attempted to kill init! exitcode=0x0000000b

4 3

[alsa-devel] On mixer controls naming
by R. Lemos 14 Jan '14

14 Jan '14

Hi, I'm trying to write a new driver for a virtual sound card. It isn't actually a sound card, but only a software solution to control volume and bass of an external device (using SPI and GPIO). So it only has a mixer (no pcm or capture subdevices). I could successfully configure two controls "Master Playback Switch" and "Master Playback Volume". Alsamixer shows them correctly under "Playback" view (and very nicely it joined them together to form a single composite control "Master") The other control I've named "Tone Control - Bass", as indicated in writing-an-alsa-driver.pdf. But alsamixer shows it under both "Playback" and "Capture" views. How can I restrict this latter control to just playback? -- []'s Rodrigo

2 4

[alsa-devel] [PATCH v3] ASoC: imx: Allow using sound with DT descriptions only
by Alexander Shiyan 13 Jan '14

13 Jan '14

This patch allow using generic sound cards/codecs based on DT description. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/Kconfig | 43 +++++++++++++++++-------------------------- 1 file changed, 17 insertions(+), 26 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 1c7b3b7..8249000 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -7,6 +7,7 @@ config SND_SOC_FSL_SSI config SND_SOC_FSL_SPDIF tristate + select REGMAP_MMIO config SND_SOC_FSL_UTILS tristate @@ -103,27 +104,32 @@ config SND_MPC52xx_SOC_EFIKA endif # SND_POWERPC_SOC -menuconfig SND_IMX_SOC - tristate "SoC Audio for Freescale i.MX CPUs" - depends on ARCH_MXC || COMPILE_TEST - help - Say Y or M if you want to add support for codecs attached to - the i.MX CPUs. - -if SND_IMX_SOC - -config SND_SOC_IMX_SSI +config SND_SOC_IMX_AUDMUX tristate config SND_SOC_IMX_PCM_FIQ tristate select FIQ + default y if SND_SOC_IMX_SSI=y || SND_SOC_FSL_SSI=y config SND_SOC_IMX_PCM_DMA tristate select SND_SOC_GENERIC_DMAENGINE_PCM + default y if SND_SOC_IMX_SSI=y || SND_SOC_FSL_SSI=y -config SND_SOC_IMX_AUDMUX +menuconfig SND_IMX_SOC + tristate "SoC Audio for Freescale i.MX CPUs" + depends on ARCH_MXC || COMPILE_TEST + select SND_SOC_FSL_SSI if OF + select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_PCM_DMA + help + Say Y or M if you want to add support for codecs attached to + the i.MX CPUs. + +if SND_IMX_SOC + +config SND_SOC_IMX_SSI tristate config SND_MXC_SOC_WM1133_EV1 @@ -131,7 +137,6 @@ config SND_MXC_SOC_WM1133_EV1 depends on MACH_MX31ADS_WM1133_EV1 select SND_SOC_WM8350 select SND_SOC_IMX_PCM_FIQ - select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help Enable support for audio on the i.MX31ADS with the WM1133-EV1 @@ -141,8 +146,6 @@ config SND_SOC_MX27VIS_AIC32X4 tristate "SoC audio support for Visstrim M10 boards" depends on MACH_IMX27_VISSTRIM_M10 && I2C select SND_SOC_TLV320AIC32X4 - select SND_SOC_IMX_PCM_DMA - select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help Say Y if you want to add support for SoC audio on Visstrim SM10 @@ -154,7 +157,6 @@ config SND_SOC_PHYCORE_AC97 select SND_SOC_AC97_BUS select SND_SOC_WM9712 select SND_SOC_IMX_PCM_FIQ - select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help Say Y if you want to add support for SoC audio on Phytec phyCORE @@ -169,7 +171,6 @@ config SND_SOC_EUKREA_TLV320 depends on I2C select SND_SOC_TLV320AIC23 select SND_SOC_IMX_PCM_FIQ - select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help Enable I2S based access to the TLV320AIC23B codec attached @@ -179,9 +180,6 @@ config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C select SND_SOC_WM8962 - select SND_SOC_IMX_PCM_DMA - select SND_SOC_IMX_AUDMUX - select SND_SOC_FSL_SSI help Say Y if you want to add support for SoC audio on an i.MX board with a wm8962 codec. @@ -190,18 +188,13 @@ config SND_SOC_IMX_SGTL5000 tristate "SoC Audio support for i.MX boards with sgtl5000" depends on OF && I2C select SND_SOC_SGTL5000 - select SND_SOC_IMX_PCM_DMA - select SND_SOC_IMX_AUDMUX - select SND_SOC_FSL_SSI help Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. config SND_SOC_IMX_SPDIF tristate "SoC Audio support for i.MX boards with S/PDIF" - select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_SPDIF - select REGMAP_MMIO help SoC Audio support for i.MX boards with S/PDIF Say Y if you want to add support for SoC audio on an i.MX board with @@ -211,8 +204,6 @@ config SND_SOC_IMX_MC13783 tristate "SoC Audio support for I.MX boards with mc13783" depends on MFD_MC13XXX && ARM select SND_SOC_IMX_SSI - select SND_SOC_IMX_AUDMUX select SND_SOC_MC13783 - select SND_SOC_IMX_PCM_DMA endif # SND_IMX_SOC -- 1.8.3.2

3 5

[alsa-devel] [PATCH] ASoC: simple-card: use snd_soc_card_set/get_drvdata
by Xiubo Li 13 Jan '14

13 Jan '14

Remove asoc_simple_get_card_info macro and use snd_soc_card_set_drvdata and snd_soc_card_get_drvdata instead. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index dbd93cc..c025b6d 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -14,9 +14,6 @@ #include <linux/module.h> #include <sound/simple_card.h> -#define asoc_simple_get_card_info(p) \ - container_of(p->dai_link, struct asoc_simple_card_info, snd_link) - static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set, unsigned int daifmt) @@ -41,7 +38,8 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct asoc_simple_card_info *info = asoc_simple_get_card_info(rtd); + struct asoc_simple_card_info *info = + snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; unsigned int daifmt = info->daifmt; @@ -268,6 +266,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo->snd_card.dai_link = &cinfo->snd_link; cinfo->snd_card.num_links = 1; + snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo); + return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); } -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: fsl-sai: Add device tree nodes and its availiable check
by Xiubo Li 13 Jan '14

13 Jan '14

SAIs that are not connected on the board should have a status = "disabled" property in their device tree nodes. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index cdd3fa8..6b2ab91 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -404,6 +404,9 @@ static int fsl_sai_probe(struct platform_device *pdev) struct resource *res; int ret; + if (!np || !of_device_is_available(np)) + return -ENODEV; + sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) return -ENOMEM; -- 1.8.4

3 3

[alsa-devel] [PATCH] ALSA: hda - Apply codec power_filter to FG nodes
by Takashi Iwai 13 Jan '14

13 Jan '14

Apply the codec->power_filter to the FG nodes in general for reducing hackish set_power_state ops override in patch_sigmatel.c. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 12 +++++++++--- sound/pci/hda/hda_generic.c | 5 +++-- sound/pci/hda/hda_generic.h | 3 +++ sound/pci/hda/patch_sigmatel.c | 36 ++++++++++++------------------------ 4 files changed, 27 insertions(+), 29 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ab2a991d916f..ec4536c8d8d4 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3927,6 +3927,8 @@ unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec, hda_nid_t nid, unsigned int power_state) { + if (nid == codec->afg || nid == codec->mfg) + return power_state; if (power_state == AC_PWRST_D3 && get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_PIN && (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)) { @@ -3965,9 +3967,13 @@ static unsigned int hda_set_power_state(struct hda_codec *codec, codec->patch_ops.set_power_state(codec, fg, power_state); else { - snd_hda_codec_read(codec, fg, flags, - AC_VERB_SET_POWER_STATE, - power_state); + state = power_state; + if (codec->power_filter) + state = codec->power_filter(codec, fg, state); + if (state == power_state || power_state != AC_PWRST_D3) + snd_hda_codec_read(codec, fg, flags, + AC_VERB_SET_POWER_STATE, + state); snd_hda_codec_set_power_to_all(codec, fg, power_state); } state = hda_sync_power_state(codec, fg, power_state); diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 7513089c8985..8321a97d5c05 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4312,11 +4312,11 @@ static int check_auto_mic_availability(struct hda_codec *codec) } /* power_filter hook; make inactive widgets into power down */ -static unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, +unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, hda_nid_t nid, unsigned int power_state) { - if (power_state != AC_PWRST_D0) + if (power_state != AC_PWRST_D0 || nid == codec->afg) return power_state; if (get_wcaps_type(get_wcaps(codec, nid)) >= AC_WID_POWER) return power_state; @@ -4324,6 +4324,7 @@ static unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, return power_state; return AC_PWRST_D3; } +EXPORT_SYMBOL_GPL(snd_hda_gen_path_power_filter); /* mute all aamix inputs initially; parse up to the first leaves */ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 0929a06df812..07f767231c9f 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -335,5 +335,8 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec); #ifdef CONFIG_PM int snd_hda_gen_check_power_status(struct hda_codec *codec, hda_nid_t nid); #endif +unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, + hda_nid_t nid, + unsigned int power_state); #endif /* __SOUND_HDA_GENERIC_H */ diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 088a5afbd1b9..6998cf29b9bc 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -368,6 +368,17 @@ static int stac_vrefout_set(struct hda_codec *codec, return 1; } +/* prevent codec AFG to D3 state when vref-out pin is used for mute LED */ +/* this hook is set in stac_setup_gpio() */ +static unsigned int stac_vref_led_power_filter(struct hda_codec *codec, + hda_nid_t nid, + unsigned int power_state) +{ + if (nid == codec->afg && power_state == AC_PWRST_D3) + return AC_PWRST_D1; + return snd_hda_gen_path_power_filter(codec, nid, power_state); +} + /* update mute-LED accoring to the master switch */ static void stac_update_led_status(struct hda_codec *codec, int enabled) { @@ -4260,30 +4271,8 @@ static int stac_suspend(struct hda_codec *codec) stac_shutup(codec); return 0; } - -static void stac_set_power_state(struct hda_codec *codec, hda_nid_t fg, - unsigned int power_state) -{ - unsigned int afg_power_state = power_state; - struct sigmatel_spec *spec = codec->spec; - - if (power_state == AC_PWRST_D3) { - if (spec->vref_mute_led_nid) { - /* with vref-out pin used for mute led control - * codec AFG is prevented from D3 state - */ - afg_power_state = AC_PWRST_D1; - } - /* this delay seems necessary to avoid click noise at power-down */ - msleep(100); - } - snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, - afg_power_state); - snd_hda_codec_set_power_to_all(codec, fg, power_state); -} #else #define stac_suspend NULL -#define stac_set_power_state NULL #endif /* CONFIG_PM */ static const struct hda_codec_ops stac_patch_ops = { @@ -4466,8 +4455,7 @@ static void stac_setup_gpio(struct hda_codec *codec) spec->gpio_dir |= spec->gpio_led; spec->gpio_data |= spec->gpio_led; } else { - codec->patch_ops.set_power_state = - stac_set_power_state; + codec->power_filter = stac_vref_led_power_filter; } } -- 1.8.5.2

1 0

[alsa-devel] [PATCH 1/1] PATCH - alsa-plugins: Modify jack client name
by Corfu, ValentinX 13 Jan '14

13 Jan '14

The current jack client name contains the process id of the application providing the audio samples. This leads to unpredictable jack client names which makes handling of the connections by a controlling application very hard. The modification now allows to specify a client name and the number of channels as parameters in the configuration file The given number of channels are then created but not connected. The implementation is backward compatible and simply adds a new configuration option to the plugin. -------------------------------------------------------------- Intel Shannon Limited Registered in Ireland Registered Office: Collinstown Industrial Park, Leixlip, County Kildare Registered Number: 308263 Business address: Dromore House, East Park, Shannon, Co. Clare This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

2 1

[alsa-devel] [PATCH] ALSA: hda - Don't set indep_hp flag for old AD codecs
by Takashi Iwai 13 Jan '14

13 Jan '14

Some old AD codecs don't like the independent HP handling, either it contains a single DAC (AD1981) or it mandates the mixer routing (AD1986A). This patch removes the indep_hp flag for such codecs. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=68081 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 819d4a98abaf..7a426ed491f2 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -185,7 +185,7 @@ static const struct hda_codec_ops ad198x_auto_patch_ops = { }; -static int ad198x_parse_auto_config(struct hda_codec *codec) +static int ad198x_parse_auto_config(struct hda_codec *codec, bool indep_hp) { struct ad198x_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->gen.autocfg; @@ -195,7 +195,7 @@ static int ad198x_parse_auto_config(struct hda_codec *codec) codec->no_trigger_sense = 1; codec->no_sticky_stream = 1; - spec->gen.indep_hp = 1; + spec->gen.indep_hp = indep_hp; spec->gen.add_stereo_mix_input = 1; err = snd_hda_parse_pin_defcfg(codec, cfg, NULL, 0); @@ -379,7 +379,7 @@ static int patch_ad1986a(struct hda_codec *codec) ad1986a_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = ad198x_parse_auto_config(codec); + err = ad198x_parse_auto_config(codec, false); if (err < 0) { snd_hda_gen_free(codec); return err; @@ -481,7 +481,7 @@ static int patch_ad1983(struct hda_codec *codec) spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); - err = ad198x_parse_auto_config(codec); + err = ad198x_parse_auto_config(codec, false); if (err < 0) goto error; err = ad1983_add_spdif_mux_ctl(codec); @@ -568,7 +568,7 @@ static int patch_ad1981(struct hda_codec *codec) snd_hda_pick_fixup(codec, NULL, ad1981_fixup_tbl, ad1981_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = ad198x_parse_auto_config(codec); + err = ad198x_parse_auto_config(codec, false); if (err < 0) goto error; err = ad1983_add_spdif_mux_ctl(codec); @@ -894,7 +894,7 @@ static int patch_ad1988(struct hda_codec *codec) snd_hda_pick_fixup(codec, ad1988_fixup_models, NULL, ad1988_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = ad198x_parse_auto_config(codec); + err = ad198x_parse_auto_config(codec, true); if (err < 0) goto error; err = ad1988_add_spdif_mux_ctl(codec); @@ -1071,7 +1071,7 @@ static int patch_ad1884(struct hda_codec *codec) snd_hda_pick_fixup(codec, NULL, ad1884_fixup_tbl, ad1884_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = ad198x_parse_auto_config(codec); + err = ad198x_parse_auto_config(codec, true); if (err < 0) goto error; err = ad1983_add_spdif_mux_ctl(codec); @@ -1113,7 +1113,7 @@ static int patch_ad1882(struct hda_codec *codec) spec->gen.mixer_merge_nid = 0x21; spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); - err = ad198x_parse_auto_config(codec); + err = ad198x_parse_auto_config(codec, true); if (err < 0) goto error; err = ad1988_add_spdif_mux_ctl(codec); -- 1.8.5.2

1 0

Re: [alsa-devel] [PATCH] ASoC: simple-card: fix one bug to writing to the platform data
by Jean-Francois Moine 13 Jan '14

13 Jan '14

Xiubo Li wrote: > It's a bug that writing to the platform data directly, for it should > be constant. So just copy it before writing. > > Signed-off-by: Xiubo Li <Li.Xiubo at freescale.com> > --- > sound/soc/generic/simple-card.c | 40 +++++++++++++++++++++------------------- > 1 file changed, 21 insertions(+), 19 deletions(-) > > diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c > index 406e4ea..5b65324 100644 > --- a/sound/soc/generic/simple-card.c > +++ b/sound/soc/generic/simple-card.c [snip] > @@ -204,36 +205,37 @@ static int asoc_simple_card_probe(struct platform_device *pdev) > struct device_node *np = pdev->dev.of_node; > struct device_node *of_cpu, *of_codec, *of_platform; > struct device *dev = &pdev->dev; > + int ret; > > cinfo = NULL; > of_cpu = NULL; > of_codec = NULL; > of_platform = NULL; > + > + cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); > + if (!cinfo) > + return -ENOMEM; > + > if (np && of_device_is_available(np)) { > - cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); > - if (cinfo) { > - int ret; > - cinfo->snd_card.dev = &pdev->dev; > - ret = asoc_simple_card_parse_of(np, cinfo, dev, > - &of_cpu, > - &of_codec, > - &of_platform); > - if (ret < 0) { > - if (ret != -EPROBE_DEFER) > - dev_err(dev, "parse error %d\n", ret); > - return ret; > - } > - } else { > - return -ENOMEM; > + cinfo->snd_card.dev = dev; > + > + ret = asoc_simple_card_parse_of(np, cinfo, dev, > + &of_cpu, > + &of_codec, > + &of_platform); > + if (ret < 0) { > + if (ret != -EPROBE_DEFER) > + dev_err(dev, "parse error %d\n", ret); > + return ret; > } > } else { > - cinfo = pdev->dev.platform_data; > - if (!cinfo) { > + if (!pdev->dev.platform_data) { > dev_err(dev, "no info for asoc-simple-card\n"); > return -EINVAL; > } > > - cinfo->snd_card.dev = &pdev->dev; > + memcpy(cinfo, pdev->dev.platform_data, sizeof(cinfo)); > + cinfo->snd_card.dev = dev; > } > > if (!cinfo->name || If the original cinfo is not used anymore, the use of its structure to handle the card information is not a good idea: - almost all cinfo information are in the struct snd_soc_card, - this cinfo structure cannot be extended to handle many DAI links, - it contains simple-card information which are of no use for the platform caller. So, I'd rather have seen: - the removal of 'snd_link' and 'snd_card' from the platform interface (struct asoc_simple_card_info), - the definition of a local struct simple_card_data containing the struct snd_soc_card and a pointer to an array of fmt/sysclk values (one per DAI link). -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

3 3

[alsa-devel] [PATCH] ASoC: wm8804: Allow control of master clock divider in PLL generation
by Daniel Matuschek 13 Jan '14

13 Jan '14

Signed-off-by: Daniel Matuschek <daniel(a)matuschek.net> After some discussions of the patch last week, here is a new version. Simply reducing the post_table did not work, as for some frequencies both settings (MCLKDIV=0 and MCLKDIV=1) are needed (e.g. 96 and 192kHz) WM8804 can run with PLL frequencies of 256xfs and 128xfs for most sample rates. At 192kHz only 128xfs is supported. The existing driver selects 128xfs automatically for some lower samples rates. By using an additional mclk_div divider, is is now possible to control the behaviour. This allows using 256xfs PLL frequency on all sample rates up to 96kHz. It should allow lower jitter and better signal quality. The behavior has to be controlled by the sound card driver, because some sample frequency share the same setting. e.g. 192kHz and 96kHz use 24.576MHz master clock. The only difference is the MCLK divider. --- sound/soc/codecs/wm8804.c | 17 ++++++++++++++--- sound/soc/codecs/wm8804.h | 4 ++++ 2 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 1704b1e..4619bf8 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -63,6 +63,7 @@ struct wm8804_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; + int mclk_div; }; static int txsrc_get(struct snd_kcontrol *kcontrol, @@ -318,7 +319,7 @@ static struct { #define FIXED_PLL_SIZE ((1ULL << 22) * 10) static int pll_factors(struct pll_div *pll_div, unsigned int target, - unsigned int source) + unsigned int source, unsigned int mclk_div) { u64 Kpart; unsigned long int K, Ndiv, Nmod, tmp; @@ -330,7 +331,8 @@ static int pll_factors(struct pll_div *pll_div, unsigned int target, */ for (i = 0; i < ARRAY_SIZE(post_table); i++) { tmp = target * post_table[i].div; - if (tmp >= 90000000 && tmp <= 100000000) { + if ((tmp >= 90000000 && tmp <= 100000000) && + (mclk_div == post_table[i].mclkdiv)) { pll_div->freqmode = post_table[i].freqmode; pll_div->mclkdiv = post_table[i].mclkdiv; target *= post_table[i].div; @@ -387,8 +389,12 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, } else { int ret; struct pll_div pll_div; + struct wm8804_priv *wm8804; - ret = pll_factors(&pll_div, freq_out, freq_in); + wm8804 = snd_soc_codec_get_drvdata(codec); + + ret = pll_factors(&pll_div, freq_out, freq_in, + wm8804->mclk_div); if (ret) return ret; @@ -452,6 +458,7 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { struct snd_soc_codec *codec; + struct wm8804_priv *wm8804; codec = dai->codec; switch (div_id) { @@ -459,6 +466,10 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, snd_soc_update_bits(codec, WM8804_PLL5, 0x30, (div & 0x3) << 4); break; + case WM8804_MCLK_DIV: + wm8804 = snd_soc_codec_get_drvdata(codec); + wm8804->mclk_div = div; + break; default: dev_err(dai->dev, "Unknown clock divider: %d\n", div_id); return -EINVAL; diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index 8ec14f5..e72d4f4 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -57,5 +57,9 @@ #define WM8804_CLKOUT_SRC_OSCCLK 4 #define WM8804_CLKOUT_DIV 1 +#define WM8804_MCLK_DIV 2 + +#define WM8804_MCLKDIV_256FS 0 +#define WM8804_MCLKDIV_128FS 1 #endif /* _WM8804_H */ -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: dpcm: don't do hw_param when BE has done hw_param
by Nenghua Cao 13 Jan '14

13 Jan '14

From: Nenghua Cao <nhcao(a)marvell.com> It fixes the following case: Two FEs connects the same BE; FE1 & BE path has been opened and hw_paramed. At this momment, FE2 & BE path is being opened and hw_paramed. The BE dai will do hw_param again even if it has done hw_param. It is not reasonable. FE1------------>BE FE2-------------^ Signed-off-by: Nenghua Cao <nhcao(a)marvell.com> --- sound/soc/soc-pcm.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 891b9a9..ec07e37 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1339,7 +1339,6 @@ static int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) continue; if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE)) continue; -- 1.7.0.4

4 15

[alsa-devel] [PATCH] alsa-restore.rules: refer to correct attr
by Dave Reisner 13 Jan '14

13 Jan '14

$attr{number} in the RUN rule is an empty expansion. This makes sense, because the path doesn't exist -- i.e., it refers to the path: /sys/devices/pci0000:00/foo/bar/sound/card0/controlC0/number Instead, refer to $attr{device/number}, which does exist. Signed-off-by: Dave Reisner <dreisner(a)archlinux.org> --- This seems to have been broken when it was first introduced a little over three years ago (de7c3eff0e). Please CC me on replies as I am not subscribed to the list. alsactl/90-alsa-restore.rules.in | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/alsactl/90-alsa-restore.rules.in b/alsactl/90-alsa-restore.rules.in index 88e12e0..c68119d 100644 --- a/alsactl/90-alsa-restore.rules.in +++ b/alsactl/90-alsa-restore.rules.in @@ -2,7 +2,7 @@ ACTION=="add", SUBSYSTEM=="sound", KERNEL=="controlC*", KERNELS!="card*", GOTO=" GOTO="alsa_restore_end" LABEL="alsa_restore_go" -TEST!="@daemonswitch@", RUN+="@sbindir@/alsactl restore $attr{number}" -TEST=="@daemonswitch@", RUN+="@sbindir@/alsactl nrestore $attr{number}" +TEST!="@daemonswitch@", RUN+="@sbindir@/alsactl restore $attr{device/number}" +TEST=="@daemonswitch@", RUN+="@sbindir@/alsactl nrestore $attr{device/number}" LABEL="alsa_restore_end" -- 1.8.5.2

3 2

[alsa-devel] [PATCH] ASoC: simple-card: remove snd_link and snd_card from struct asoc_xx_info
by Xiubo Li 13 Jan '14

13 Jan '14

Whether the dt is used or not, almost all the simple card information for the DAI link and sound card are initialized in the simple card driver. And for the platform caller, the snd_link and snd_card are of no use, so remove it from struct asoc_simple_card_info, and let them to be the simple card driver's global data. This is also fix one bug about writing to the platform data directly, for it should be constant. And now only one DAI link is supported for simple card. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- This patch maybe not very perfect, such as supporting many DAI links, etc. And there still need followed patches to fulfil these. include/sound/simple_card.h | 4 --- sound/soc/generic/simple-card.c | 54 +++++++++++++++++++++++------------------ 2 files changed, 30 insertions(+), 28 deletions(-) diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index 6c74527..e1ac996 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -29,10 +29,6 @@ struct asoc_simple_card_info { unsigned int daifmt; struct asoc_simple_dai cpu_dai; struct asoc_simple_dai codec_dai; - - /* used in simple-card.c */ - struct snd_soc_dai_link snd_link; - struct snd_soc_card snd_card; }; #endif /* __SIMPLE_CARD_H */ diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index dbd93cc..94dc743 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -14,9 +14,6 @@ #include <linux/module.h> #include <sound/simple_card.h> -#define asoc_simple_get_card_info(p) \ - container_of(p->dai_link, struct asoc_simple_card_info, snd_link) - static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set, unsigned int daifmt) @@ -41,7 +38,8 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct asoc_simple_card_info *info = asoc_simple_get_card_info(rtd); + struct asoc_simple_card_info *info = + snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; unsigned int daifmt = info->daifmt; @@ -58,6 +56,16 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static struct snd_soc_dai_link simple_dai_link = { + .init = asoc_simple_card_dai_init, +}; + +static struct snd_soc_card simple_snd_card = { + .owner = THIS_MODULE, + .dai_link = &simple_dai_link, + .num_links = 1, +}; + static int asoc_simple_card_sub_parse_of(struct device_node *np, struct asoc_simple_dai *dai, @@ -144,7 +152,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, /* DAPM routes */ if (of_property_read_bool(node, "simple-audio-card,routing")) { - ret = snd_soc_of_parse_audio_routing(&info->snd_card, + ret = snd_soc_of_parse_audio_routing(&simple_snd_card, "simple-audio-card,routing"); if (ret) return ret; @@ -213,7 +221,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); if (cinfo) { int ret; - cinfo->snd_card.dev = &pdev->dev; + simple_snd_card.dev = dev; ret = asoc_simple_card_parse_of(np, cinfo, dev, &of_cpu, &of_codec, @@ -233,7 +241,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) return -EINVAL; } - cinfo->snd_card.dev = &pdev->dev; + simple_snd_card.dev = dev; } if (!cinfo->name || @@ -247,28 +255,26 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } /* - * init snd_soc_dai_link + * init simple_dai_link */ - cinfo->snd_link.name = cinfo->name; - cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; - cinfo->snd_link.platform_name = cinfo->platform; - cinfo->snd_link.codec_name = cinfo->codec; - cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; - cinfo->snd_link.cpu_of_node = of_cpu; - cinfo->snd_link.codec_of_node = of_codec; - cinfo->snd_link.platform_of_node = of_platform; - cinfo->snd_link.init = asoc_simple_card_dai_init; + simple_dai_link.name = cinfo->name; + simple_dai_link.stream_name = cinfo->name; + simple_dai_link.cpu_dai_name = cinfo->cpu_dai.name; + simple_dai_link.platform_name = cinfo->platform; + simple_dai_link.codec_name = cinfo->codec; + simple_dai_link.codec_dai_name = cinfo->codec_dai.name; + simple_dai_link.cpu_of_node = of_cpu; + simple_dai_link.codec_of_node = of_codec; + simple_dai_link.platform_of_node = of_platform; /* - * init snd_soc_card + * init simple_snd_card */ - cinfo->snd_card.name = cinfo->card; - cinfo->snd_card.owner = THIS_MODULE; - cinfo->snd_card.dai_link = &cinfo->snd_link; - cinfo->snd_card.num_links = 1; + simple_snd_card.name = cinfo->card; + + snd_soc_card_set_drvdata(&simple_snd_card, cinfo); - return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); + return devm_snd_soc_register_card(dev, &simple_snd_card); } static const struct of_device_id asoc_simple_of_match[] = { -- 1.8.4

3 2

[alsa-devel] How to make Realtek HD-A codec alc282 create "Input Source" control to select a input for recording?
by Lin, Mengdong 13 Jan '14

13 Jan '14

Hi, How to make the driver create a "input source" control for ALC282 with 3 Mic pins? My platform uses a Realtek HD-A codec ALC282 card, it has 3 Mic pins: Mic = 0x18 (MIC1) Internal Mic = 0x19 (MIC2) Internal Mic = 0x12 (DMIC) Recording on Mic1 (0x18) cannot work because the mixer 0x23 always mutes the input from 0x18. And there is no "input source" control to let the mixer select and unmute 0x18. I've tried to disable pin sense on pin 0x18 as a board fix-up like below, but "input source" control is still not created: caps = snd_hda_query_pin_caps(codec, 0x18); caps &= ~AC_PINCAP_PRES_DETECT; snd_hda_override_pin_caps(codec, 0x18, caps); snd_hda_codec_set_pincfg(codec, 0x18, 0x03a19120); ... set 'Jack Detect Override' Thanks Mengdong Here is the alsa-info: upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.61 !!################################ !!Script ran on: Thu Jan 9 06:09:23 UTC 2014 !!Linux Distribution !!------------------ Ubuntu Saucy Salamander (development branch) \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu Saucy Salamander (development branch)" NAME="Ubuntu" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu Saucy Salamander (development branch)" HOME_URL="http://www.ubuntu.com/" SUPPORT_URL="http://help.ubuntu.com/" BUG_REPORT_URL="http://bugs.launchpad.net/ubuntu/" !!DMI Information !!--------------- Manufacturer: Intel Corporation Product Name: Broadwell Client platform Product Version: 0.1 Firmware Version: BDW-E1R1.86C.0052.R01.1311260206 !!Kernel Information !!------------------ Kernel release: 3.13.0-rc4-drm-bdw+ Operating System: GNU/Linux Architecture: x86_64 Processor: x86_64 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k3.13.0-rc4-drm-bdw+ Library version: 1.0.27.2 Utilities version: 1.0.27.1 !!Loaded ALSA modules !!------------------- snd_hda_intel snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [HDMI ]: HDA-Intel - HDA Intel HDMI HDA Intel HDMI at 0xd1030000 irq 61 1 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xd1034000 irq 59 !!PCI Soundcards installed in the system !!-------------------------------------- 00:03.0 Audio device: Intel Corporation Device 160c (rev 04) 00:1b.0 Audio device: Intel Corporation Device 9ca0 !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:03.0 0403: 8086:160c (rev 04) Subsystem: 8086:2010 -- 00:1b.0 0403: 8086:9ca0 Subsystem: 8086:7270 !!Modprobe options (Sound related) !!-------------------------------- snd_atiixp_modem: index=-2 snd_intel8x0m: index=-2 snd_via82xx_modem: index=-2 snd_usb_audio: index=-2 snd_usb_caiaq: index=-2 snd_usb_ua101: index=-2 snd_usb_us122l: index=-2 snd_usb_usx2y: index=-2 snd_cmipci: mpu_port=0x330 fm_port=0x388 snd_pcsp: index=-2 snd_usb_audio: index=-2 !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 32,1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 32,1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Intel Broadwell HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x80862808 Subsystem Id: 0x80860101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D3 CLKSTOP EPSS Power: setting=D0, actual=D0, Clock-stop-OK GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D3, actual=D3 Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled KAE Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D3, actual=D3 Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=3 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="ELD", index=0, device=3 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=01, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=7 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=1, device=0 Control: name="IEC958 Playback Pro Mask", index=1, device=0 Control: name="IEC958 Playback Default", index=1, device=0 Control: name="IEC958 Playback Switch", index=1, device=0 Control: name="ELD", index=0, device=7 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=02, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 3 0x02* 0x03 0x04 Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=8 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=2, device=0 Control: name="IEC958 Playback Pro Mask", index=2, device=0 Control: name="IEC958 Playback Default", index=2, device=0 Control: name="IEC958 Playback Switch", index=2, device=0 Control: name="ELD", index=0, device=8 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=03, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 0 In-driver Connection: 3 0x02 0x03 0x04 Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono Codec: Realtek ALC282 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0282 Subsystem Id: 0x10ec0282 Revision Id: 0x100002 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=3, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC282 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x54 0x54] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC282 Analog", type="Audio", device=0 Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 Amp-In vals: [0x29 0x29] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 Amp-In vals: [0x97 0x97] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x22 Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Internal Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Internal Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 5 0x18 0x19 0x1a 0x1b 0x1d Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x03 0x0b Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0f [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x0d Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 Amp-In vals: [0x29 0x29] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x12 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Internal Mic Phantom Jack", index=1, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x90a60930: [Fixed] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00010014: OUT EAPD Detect EAPD 0x2: EAPD Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x0c Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x17 [Pin Complex] wcaps 0x40050c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x0f Node 0x18 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x03 0x03] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x03a19820: [Jack] Mic at Ext Left Conn = 1/8, Color = Pink DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x19 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Internal Mic Phantom Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x90a7092f: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x2, Sequence = 0xf Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1a [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1d [Pin Complex] wcaps 0x400400: Mono Pincap 0x00000020: IN Pin Default 0x40189d2d: [N/A] Speaker at Ext N/A Conn = DIN, Color = Pink DefAssociation = 0x2, Sequence = 0xd Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1e [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=117 Node 0x21 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000001c: OUT HP Detect Pin Default 0x0321401f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c 0x0d* Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 6 0x18 0x19 0x1a 0x1b 0x1d 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 6 0x18 0x19 0x1a 0x1b 0x1d 0x0b --endcollapse--

2 5

[alsa-devel] [PATCH] ASoC: wm8804: Cleanup of post_table
by Daniel Matuschek 12 Jan '14

12 Jan '14

Signed-off-by: Daniel Matuschek <inof(a)crazy-audio.com> Existing post_table has duplicate settings for some PLL settings. Only the first setting was used with mclk_div=1 With the updated post_table, the driver uses mclk_div=0 whenever it is possible. This allows the WM8804 to work at 256xfs internal clock and only use 128xfs internal clock for high sample rates where 256xfs is not supported anymore. --- sound/soc/codecs/wm8804.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 1704b1e..f605b8b 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -307,9 +307,7 @@ static struct { unsigned int mclkdiv; } post_table[] = { { 2, 0, 0 }, - { 4, 0, 1 }, { 4, 1, 0 }, - { 8, 1, 1 }, { 8, 2, 0 }, { 16, 2, 1 }, { 12, 3, 0 }, -- 1.7.9.5

2 1

[alsa-devel] searching volunteer to test Hoontech DSP24 to check if the driver is realy working well
by YvanM 12 Jan '14

12 Jan '14

Hi everybody ! First please apologize if I'm in the wrong place. Secondly thanks for all the work you are all doing ! I have a Hoontech DSP24 (http://fr.audiofanzine.com/carte-son-rack/hoontech/DSP24-+-ADDA2000-DSP2000…), but I had problem to make it working. I don't use it so, if someone with knowledge wants to check if this soundcard is really recognized, I can send it to him. After that, feel free to play with it, improve the driver, or even sell it ! The problem I had : - spent 3 months to find on the Internet the following lines to had to /etc/modprobe.d/alsa-base to make the card recognized : stall snd-ice1712 /sbin/modprobe --ignore-install snd-ice1712 && { /sbin/modprobe -Qb snd-ice1712 ; } options snd-ice1712 model=dsp24 - impossible to have "normal" stereo : it seems that each logical canal (right and left) is redirected to both right and left speaker. I someone wants to have a try with it, do not hesitate to mail me. It is a pity that a so good soundcard can't be used anymore… And as I said, feel then free to do what you want with it. Thanks, Yvan

3 2

[alsa-devel] [PATCH] ALSA: Enable CONFIG_ZONE_DMA for smaller PCI DMA masks
by Takashi Iwai 11 Jan '14

11 Jan '14

The PCI devices with DMA masks smaller than 32bit should enable CONFIG_ZONE_DMA. Since the recent change of page allocator, page allocations via dma_alloc_coherent() with the limited DMA mask bits may fail more frequently, ended up with no available buffers, when CONFIG_ZONE_DMA isn't enabled. With CONFIG_ZONE_DMA, the system has much more chance to obtain such pages. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=68221 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/Kconfig | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 3a0504dfeb8a..8756c8e32922 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -25,6 +25,7 @@ config SND_ALS300 select SND_PCM select SND_AC97_CODEC select SND_OPL3_LIB + select ZONE_DMA help Say 'Y' or 'M' to include support for Avance Logic ALS300/ALS300+ @@ -49,6 +50,7 @@ config SND_ALI5451 tristate "ALi M5451 PCI Audio Controller" select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for the integrated AC97 sound device on motherboards using the ALi M5451 Audio Controller @@ -153,6 +155,7 @@ config SND_AZT3328 select SND_PCM select SND_RAWMIDI select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for Aztech AZF3328 (PCI168) soundcards. @@ -459,6 +462,7 @@ config SND_EMU10K1 select SND_HWDEP select SND_RAWMIDI select SND_AC97_CODEC + select ZONE_DMA help Say Y to include support for Sound Blaster PCI 512, Live!, Audigy and E-mu APS (partially supported) soundcards. @@ -474,6 +478,7 @@ config SND_EMU10K1X tristate "Emu10k1X (Dell OEM Version)" select SND_AC97_CODEC select SND_RAWMIDI + select ZONE_DMA help Say Y here to include support for the Dell OEM version of the Sound Blaster Live!. @@ -507,6 +512,7 @@ config SND_ES1938 select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on ESS Solo-1 (ES1938, ES1946, ES1969) chips. @@ -518,6 +524,7 @@ config SND_ES1968 tristate "ESS ES1968/1978 (Maestro-1/2/2E)" select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on ESS Maestro 1/2/2E chips. @@ -606,6 +613,7 @@ config SND_ICE1712 select SND_MPU401_UART select SND_AC97_CODEC select BITREVERSE + select ZONE_DMA help Say Y here to include support for soundcards based on the ICE1712 (Envy24) chip. @@ -693,6 +701,7 @@ config SND_LX6464ES config SND_MAESTRO3 tristate "ESS Allegro/Maestro3" select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on ESS Maestro 3 (Allegro) chips. @@ -789,6 +798,7 @@ config SND_SIS7019 tristate "SiS 7019 Audio Accelerator" depends on X86 && !X86_64 select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for the SiS 7019 Audio Accelerator. @@ -800,6 +810,7 @@ config SND_SONICVIBES select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on the S3 SonicVibes chip. @@ -811,6 +822,7 @@ config SND_TRIDENT tristate "Trident 4D-Wave DX/NX; SiS 7018" select SND_MPU401_UART select SND_AC97_CODEC + select ZONE_DMA help Say Y here to include support for soundcards based on Trident 4D-Wave DX/NX or SiS 7018 chips. -- 1.8.5.2

1 0

Re: [alsa-devel] [PATCH v2] ASoC: fsl_esai: Add ESAI CPU DAI driver
by Nicolin Chen 10 Jan '14

10 Jan '14

Resent this because of losing attached file. On Fri, Jan 10, 2014 at 10:32:52AM +0800, Nicolin Chen wrote: > On Thu, Jan 09, 2014 at 06:44:53PM +0000, Mark Brown wrote: > > On Thu, Jan 09, 2014 at 06:57:58PM +0800, Nicolin Chen wrote: > > > > > +/** > > > + * This function configures the ratio between MCLK (HCK) and BCLK (SCK) > > > + * (For DAI Master Mode only) > > > + * > > > + * Note: Machine driver should calculate the ratio to call this function. > > > + * Only effective after calling set_dai_sysclk() to set HCK direction. > > > + */ > > > +static int fsl_esai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) > > > > Why does the machine driver have to do this by hand, being able to > > override is fine but having sensible defaults is easier? Or does it > > actually do that and the comment just needs updating? > > The divider part of ESAI is pretty complicated due to caring about three > configure bits - ETO, ETI, HCKD. (I've attached a diagram to this mail.) > So setting sysclk() alone is not enough to take care all the configurations. > That's why I designed these two interfaces at the first place. And it should > be hard to find a default situation. > > But there is one approach to omit this calling for machine driver is to do > the set_bclk_ratio() at this CPU DAI driver. And this might be a good idea > because we can then separate the settings between PLAYBACK and CAPTURE, even > though we might then need to check the Master or Slave state to apply the > settings accordingly. > > I'll try this idea in v3 if you feel it's plausible. > > > > + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component, > > > + &fsl_esai_dai, 1); > > > + if (ret) { > > > + dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); > > > + return ret; > > > + } > > > + > > > + ret = imx_pcm_dma_init(pdev); > > > + if (ret) > > > + dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); > > > + > > > + /* Reset ESAI unit */ > > > + regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ERST); > > > + > > > + /* > > > + * We need to enable ESAI so as to access some of its registers. > > > + * Otherwise, we would fail to dump regmap from user space. > > > + */ > > > + regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ESAIEN); > > > > I would expect to see the hardware initialisation before we start > > registering with the core otherwise the core might start trying to run > > prior to the hardware being initialised. > > Will switch this initial code in v3 as your suggestion. > > Thank you, > Nicolin Chen

2 6

[alsa-devel] [PATCH 1/5] ASoC: pcm: Properly initialize hw->rate_max
by Lars-Peter Clausen 10 Jan '14

10 Jan '14

If none of the components (CODEC or CPU DAI) sets a maximum sample rate we'll end up with the rate_max field of the runtime hardware set to 0. (Note that it is still possible for the components to constrain the supported sample rates using other methods, e.g. setting a list constraint) If rate_max is 0 this means that the sound card doesn't support any rates at all, which is not the desired result. So initialize rate_max to UINT_MAX. For symmetry reasons also set rate_min to 0. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 313adba..b6e0c32 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -252,6 +252,9 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime, & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) hw->rates |= codec_stream->rates; + hw->rate_min = 0; + hw->rate_max = UINT_MAX; + snd_pcm_limit_hw_rates(runtime); hw->rate_min = max(hw->rate_min, cpu_stream->rate_min); -- 1.8.0

2 6

[alsa-devel] [PATCH v2 28/28] drm/i2c: tda998x: add a CODEC interface
by Jean-Francois Moine 10 Jan '14

10 Jan '14

The TDA998x chips may get audio input from I2S or S/PDIF and this input is defined by the audio subsystem. In the other way, the audio subsystem may need to know if audio output may be done through the HDMI connector. This patch adds a CODEC interface for exchanges between the HDMI transmitter and the audio subsystem. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- drivers/gpu/drm/i2c/tda998x_drv.c | 70 ++++++++++++++++++++-- 1 file changed, 65 insertions(+), 5 deletions(-) diff --git a/drivers/gpu/drm/i2c/tda998x_drv.c b/drivers/gpu/drm/i2c/tda998x_drv.c index 67d7450..5a186e4 100644 --- a/drivers/gpu/drm/i2c/tda998x_drv.c +++ b/drivers/gpu/drm/i2c/tda998x_drv.c @@ -42,7 +42,8 @@ struct tda998x_priv { u8 vip_cntrl_1; u8 vip_cntrl_2; - u8 audio_type; /* audio type */ + u8 audio_type; /* audio type and 'active' flag */ +#define AUDIO_ACTIVE 0x80 u8 audio_frame[6]; u32 audio_port; @@ -50,6 +51,9 @@ struct tda998x_priv { wait_queue_head_t wq_edid; volatile int wq_edid_wait; struct drm_encoder *encoder; + + struct snd_soc_codec *codec; + void (*event)(struct snd_soc_codec *codec, int activate); }; #define to_tda998x_priv(x) ((struct tda998x_priv *)to_encoder_slave(x)->slave_priv) @@ -731,14 +735,14 @@ tda998x_configure_audio(struct tda998x_priv *priv, /* Set audio input source */ switch (priv->audio_type) { - case AFMT_SPDIF: + case AFMT_SPDIF | AUDIO_ACTIVE: reg_write(priv, REG_MUX_AP, MUX_AP_SELECT_SPDIF); clksel_aip = AIP_CLKSEL_AIP(SEL_AIP_SPDIF); clksel_fs = AIP_CLKSEL_FS(CTSREF_FS64SPDIF); cts_n = CTS_N_M(3) | CTS_N_K(3); break; - case AFMT_I2S: + case AFMT_I2S | AUDIO_ACTIVE: reg_write(priv, REG_MUX_AP, MUX_AP_SELECT_I2S); clksel_aip = AIP_CLKSEL_AIP(SEL_AIP_I2S); clksel_fs = AIP_CLKSEL_FS(CTSREF_ACLK); @@ -813,6 +817,58 @@ tda998x_configure_audio(struct tda998x_priv *priv, tda998x_write_aif(priv); } +/* tda998x codec interface */ +void tda998x_audio_register(struct i2c_client *client, + struct snd_soc_codec *codec, + void (*event)(struct snd_soc_codec *codec, int activate)) +{ + struct tda998x_priv *priv = i2c_get_clientdata(client); + + /* memorize the codec and the callback function */ + priv->codec = codec; + priv->event = event; +} +EXPORT_SYMBOL_GPL(tda998x_audio_register); + +void tda998x_audio_update(struct i2c_client *client, + int type, + u32 port) +{ + struct tda998x_priv *priv = i2c_get_clientdata(client); + + /* if the audio output is active, it may be a second start or a stop */ + if (type == 0 || (priv->audio_type & AUDIO_ACTIVE)) { + if (type == 0) { + priv->audio_type &= ~AUDIO_ACTIVE; /* stop */ + reg_write(priv, REG_ENA_AP, 0); + } + return; + } + + priv->audio_port = port; + + /* don't restart audio if same input type */ + if (type == priv->audio_type) { + priv->audio_type |= AUDIO_ACTIVE; + reg_write(priv, REG_ENA_AP, priv->audio_port); + return; + } + + priv->audio_type = type | AUDIO_ACTIVE; + + tda998x_configure_audio(priv, &priv->encoder->crtc->hwmode); +} +EXPORT_SYMBOL_GPL(tda998x_audio_update); + +/* send an audio event to the tda codec */ +static void send_audio_event(struct tda998x_priv *priv, + int event) +{ + if (!priv->event) + return; + priv->event(priv->codec, event); +} + /* DRM encoder functions */ /* this function is not called when no info->platform (DT support) */ @@ -840,7 +896,7 @@ tda998x_encoder_set_config(struct drm_encoder *encoder, void *params) if (p->audio_cfg) { priv->audio_port = p->audio_cfg; - priv->audio_type = p->audio_format; + priv->audio_type = p->audio_format | AUDIO_ACTIVE; } } @@ -1105,9 +1161,10 @@ tda998x_encoder_mode_set(struct drm_encoder *encoder, tda998x_write_avi(priv, mode); - if (priv->audio_type) + if (priv->audio_type & AUDIO_ACTIVE) tda998x_configure_audio(priv, mode); } + send_audio_event(priv, priv->is_hdmi_sink); /* must be last register set: */ reg_write(priv, REG_TBG_CNTRL_0, 0); @@ -1263,6 +1320,7 @@ tda998x_encoder_get_modes(struct drm_encoder *encoder, drm_mode_connector_update_edid_property(connector, edid); n = drm_add_edid_modes(connector, edid); priv->is_hdmi_sink = drm_detect_hdmi_monitor(edid); + send_audio_event(priv, priv->is_hdmi_sink); kfree(edid); } @@ -1349,6 +1407,8 @@ tda998x_encoder_init(struct i2c_client *client, if (!priv) return -ENOMEM; + i2c_set_clientdata(client, priv); + priv->vip_cntrl_0 = VIP_CNTRL_0_SWAP_A(2) | VIP_CNTRL_0_SWAP_B(3); priv->vip_cntrl_1 = VIP_CNTRL_1_SWAP_C(0) | VIP_CNTRL_1_SWAP_D(1); priv->vip_cntrl_2 = VIP_CNTRL_2_SWAP_E(4) | VIP_CNTRL_2_SWAP_F(5); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH 1/8] ASoC: alc5623: Use params_width() rather than memory format
by Mark Brown 10 Jan '14

10 Jan '14

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/alc5623.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 256c364193a5..d3036283482a 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -714,17 +714,17 @@ static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream, iface &= ~ALC5623_DAI_I2S_DL_MASK; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: iface |= ALC5623_DAI_I2S_DL_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: iface |= ALC5623_DAI_I2S_DL_20; break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: iface |= ALC5623_DAI_I2S_DL_24; break; - case SNDRV_PCM_FORMAT_S32_LE: + case 32: iface |= ALC5623_DAI_I2S_DL_32; break; default: -- 1.8.5.2

4 12

[alsa-devel] [PATCH] ASoC: fsl: imx-wm8962: Grant hw_params() permission to reprogram FLL
by Nicolin Chen 10 Jan '14

10 Jan '14

From: Nicolin Chen <b42378(a)freescale.com> Previously, we couldn't use hw_params() to reclock FLL becuase there might be a race between two simmultaneous substreams so the FLL reprogramming would be changed and accordingly mulfunction. Also it wouldn't make sense for analogue bypass path feature of WM8962. So we adopted the DAPM way to control it. However, if we want to playback a different sample rate file, we need to wait for DAPM to change its bias_level and reconfigure FLL. After we introduced full symmetry protection in the soc-pcm, we don't need to worry about the race any more. And the instance by using hw_params() to reprogram FLL will allow us to support flexible use cases -- for example: 'aplay -Dhw:0 44k16bit.wav 48k24bit.wav 32k16bit.wav'. Thus this patch mainly adds FLL reprogramming code to hw_params() so as to enchance the sound card's capability. Meanwhile in order not to break the analogue bypass path feature, we make both set_bias_level() and hw_params() ways coexist by only reclocking FLL if requiring a different frequency and meanwhile not running any analogue bypass path in the background. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/imx-wm8962.c | 167 ++++++++++++++++++++++++++++++++------------- 1 file changed, 121 insertions(+), 46 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3fd76bc..9621909 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -17,6 +17,7 @@ #include <linux/of_platform.h> #include <linux/i2c.h> #include <linux/slab.h> +#include <linux/spinlock.h> #include <linux/clk.h> #include <sound/soc.h> #include <sound/pcm_params.h> @@ -35,6 +36,8 @@ struct imx_wm8962_data { char platform_name[DAI_NAME_SIZE]; struct clk *codec_clk; unsigned int clk_frequency; + unsigned int fll_frequency; + spinlock_t fll_lock; }; struct imx_priv { @@ -50,15 +53,125 @@ static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { }; static int sample_rate = 44100; +static unsigned int fll_frequency = 44100 * 256; static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; +static inline u32 imx_wm8962_calculate_fll(u32 rate, snd_pcm_format_t format) +{ + switch (format) { + case SNDRV_PCM_FORMAT_S24_LE: + return rate * 384; + default: + return rate * 256; + } +} + +static int imx_wm8962_enable_fll(struct snd_soc_dai *codec_dai) +{ + struct imx_priv *priv = &card_priv; + struct device *dev = &priv->pdev->dev; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + u32 ret = 0; + + spin_lock(&data->fll_lock); + + /* Protect FLL if it has not been disabled */ + if (data->fll_frequency) + goto err_unlock; + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, + data->clk_frequency, fll_frequency); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + goto err_unlock; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, + fll_frequency, SND_SOC_CLOCK_IN); + if (ret) + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + + data->fll_frequency = fll_frequency; + +err_unlock: + spin_unlock(&data->fll_lock); + + return ret; +} + +static int imx_wm8962_disable_fll(struct snd_soc_dai *codec_dai) +{ + struct imx_priv *priv = &card_priv; + struct device *dev = &priv->pdev->dev; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + int ret = 0; + + spin_lock(&data->fll_lock); + + if (!data->fll_frequency) + goto err_unlock; + + /* Switch to MCLK as sysclk once so as to disable FLL */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + 0, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to switch away from FLL: %d\n", ret); + goto err_unlock; + } + + /* Disable FLL so that we can reset its output freq later */ + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, 0, 0); + if (ret) + dev_err(dev, "failed to stop FLL: %d\n", ret); + + data->fll_frequency = 0; + +err_unlock: + spin_unlock(&data->fll_lock); + + return ret; +} + +static int imx_wm8962_reprogram_fll(struct snd_pcm_substream *substream) +{ + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + u32 mask, ret = 0; + + mask = WM8962_MIXINL_TO_HPMIXL_MASK | WM8962_MIXINR_TO_HPMIXL_MASK | + WM8962_IN4L_TO_HPMIXL_MASK | WM8962_IN4R_TO_HPMIXL_MASK; + ret |= snd_soc_read(codec, WM8962_HEADPHONE_MIXER_1) & mask; + ret |= snd_soc_read(codec, WM8962_HEADPHONE_MIXER_2) & mask; + ret |= snd_soc_read(codec, WM8962_SPEAKER_MIXER_1) & mask; + ret |= snd_soc_read(codec, WM8962_SPEAKER_MIXER_2) & mask; + + spin_lock(&data->fll_lock); + /* Do not reprogram FLL if not changed or running analogue bypass */ + if (fll_frequency == data->fll_frequency || ret) { + spin_unlock(&data->fll_lock); + return 0; + } + spin_unlock(&data->fll_lock); + + ret = imx_wm8962_disable_fll(codec_dai); + if (ret) + return ret; + + return imx_wm8962_enable_fll(codec_dai); +} + static int imx_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { sample_rate = params_rate(params); sample_format = params_format(params); + fll_frequency = imx_wm8962_calculate_fll(sample_rate, sample_format); - return 0; + return imx_wm8962_reprogram_fll(substream); } static struct snd_soc_ops imx_hifi_ops = { @@ -70,60 +183,19 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, enum snd_soc_bias_level level) { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - struct imx_priv *priv = &card_priv; - struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); - struct device *dev = &priv->pdev->dev; - unsigned int pll_out; - int ret; if (dapm->dev != codec_dai->dev) return 0; switch (level) { case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - if (sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = sample_rate * 384; - else - pll_out = sample_rate * 256; - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - WM8962_FLL_MCLK, data->clk_frequency, - pll_out); - if (ret < 0) { - dev_err(dev, "failed to start FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_FLL, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - return ret; - } - } + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) + return imx_wm8962_enable_fll(codec_dai); break; case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_MCLK, data->clk_frequency, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, - "failed to switch away from FLL: %d\n", - ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - 0, 0, 0); - if (ret < 0) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - } + if (dapm->bias_level == SND_SOC_BIAS_PREPARE) + return imx_wm8962_disable_fll(codec_dai); break; default: @@ -225,6 +297,9 @@ static int imx_wm8962_probe(struct platform_device *pdev) goto fail; } + data->fll_frequency = 0; + spin_lock_init(&data->fll_lock); + data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) { ret = PTR_ERR(data->codec_clk); -- 1.8.4

3 7

[alsa-devel] [PATCH v3] ASoC: fsl_esai: Add ESAI CPU DAI driver
by Nicolin Chen 10 Jan '14

10 Jan '14

This patch implements a device-tree-only CPU DAI driver for Freescale ESAI controller that supports: - 12 channels playback and 8 channels record. [ Some of the inner transmitters and receivers are sharing same group of pins. So the maxmium 12 output or 8 input channels are only valid if there is no pin conflict occurring to it. ] - Independent (asynchronous mode) or shared (synchronous mode) transmit and receive sections with separate or shared internal/external clocks and frame syncs, operating in Master or Slave mode. [ Current ALSA seems not to allow CPU DAI drivers to configure DAI format separately for PLAYBACK and CAPTURE. So this first version only supports the case that uses the same DAI format for both directions. ] - Various DAI formats: I2S, Left-Justified, Right-Justified, DSP-A and DSP-B. - Programmable word length (8, 16, 20 or 24bits) - Flexible selection between system clock or external oscillator as input clock source, programmable internal clock divider and frame sync generation. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v3: * Added monaural audio support by using Normal mode. And accordingly dropped the esai-network-mode property in DT -- We let driver handle it. * Separated divider settings for Tx and Rx so that we can drop set_clkdiv(). * Changed set_bclk_ratio() to set_bclk() and let the new one be called in CPU DAI driver to omit the calling in machine driver. * Moved the ESAI reset and starting before registering the driver. * Implemented full symmetry for synchronous mode. * Use correct maxburst size. v2: * Corrected multi-line comments. * Added missing spaces between number and operator. .../devicetree/bindings/sound/fsl,esai.txt | 50 ++ sound/soc/fsl/Kconfig | 3 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_esai.c | 815 +++++++++++++++++++++ sound/soc/fsl/fsl_esai.h | 354 +++++++++ 5 files changed, 1224 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,esai.txt create mode 100644 sound/soc/fsl/fsl_esai.c create mode 100644 sound/soc/fsl/fsl_esai.h diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt new file mode 100644 index 0000000..d7b99fa --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt @@ -0,0 +1,50 @@ +Freescale Enhanced Serial Audio Interface (ESAI) Controller + +The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port +for serial communication with a variety of serial devices, including industry +standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and +other DSPs. It has up to six transmitters and four receivers. + +Required properties: + + - compatible : Compatible list, must contain "fsl,imx35-esai". + + - reg : Offset and length of the register set for the device. + + - interrupts : Contains the spdif interrupt. + + - dmas : Generic dma devicetree binding as described in + Documentation/devicetree/bindings/dma/dma.txt. + + - dma-names : Two dmas have to be defined, "tx" and "rx". + + - clocks: Contains an entry for each entry in clock-names. + + - clock-names : Includes the following entries: + "core" The core clock used to access registers + "extal" The esai baud clock for esai controller used to derive + HCK, SCK and FS. + "fsys" The system clock derived from ahb clock used to derive + HCK, SCK and FS. + + - fsl,fifo-depth: The number of elements in the transmit and receive FIFOs. + This number is the maximum allowed value for TFCR[TFWM] or RFCR[RFWM]. + + - fsl,esai-synchronous: This is a boolean property. If present, indicating + that ESAI would work in the synchronous mode, which means all the settings + for Receiving would be duplicated from Transmition related registers. + +Example: + +esai: esai@02024000 { + compatible = "fsl,imx35-esai"; + reg = <0x02024000 0x4000>; + interrupts = <0 51 0x04>; + clocks = <&clks 208>, <&clks 118>, <&clks 208>; + clock-names = "core", "extal", "fsys"; + dmas = <&sdma 23 21 0>, <&sdma 24 21 0>; + dma-names = "rx", "tx"; + fsl,fifo-depth = <128>; + fsl,esai-synchronous; + status = "disabled"; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 514c275..324988d 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -8,6 +8,9 @@ config SND_SOC_FSL_SSI config SND_SOC_FSL_SPDIF tristate +config SND_SOC_FSL_ESAI + tristate + config SND_SOC_FSL_UTILS tristate diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index aaccbee..b12ad4b 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -14,11 +14,13 @@ obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o snd-soc-fsl-sai-objs := fsl_sai.o snd-soc-fsl-ssi-objs := fsl_ssi.o snd-soc-fsl-spdif-objs := fsl_spdif.o +snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o +obj-$(CONFIG_SND_SOC_FSL_ESAI) += snd-soc-fsl-esai.o obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c new file mode 100644 index 0000000..d0c72ed --- /dev/null +++ b/sound/soc/fsl/fsl_esai.c @@ -0,0 +1,815 @@ +/* + * Freescale ESAI ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/dmaengine.h> +#include <linux/module.h> +#include <linux/of_irq.h> +#include <linux/of_platform.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> + +#include "fsl_esai.h" +#include "imx-pcm.h" + +#define FSL_ESAI_RATES SNDRV_PCM_RATE_8000_192000 +#define FSL_ESAI_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +/** + * fsl_esai: ESAI private data + * + * @dma_params_rx: DMA parameters for receive channel + * @dma_params_tx: DMA parameters for transmit channel + * @pdev: platform device pointer + * @regmap: regmap handler + * @coreclk: clock source to access register + * @extalclk: esai clock source to derive HCK, SCK and FS + * @fsysclk: system clock source to derive HCK, SCK and FS + * @fifo_depth: depth of tx/rx FIFO + * @slot_width: width of each DAI slot + * @hck_rate: clock rate of desired HCKx clock + * @sck_div: if using PSR/PM dividers for SCKx clock + * @slave_mode: if fully using DAI slave mode + * @synchronous: if using tx/rx synchronous mode + * @name: driver name + */ +struct fsl_esai { + struct snd_dmaengine_dai_dma_data dma_params_rx; + struct snd_dmaengine_dai_dma_data dma_params_tx; + struct platform_device *pdev; + struct regmap *regmap; + struct clk *coreclk; + struct clk *extalclk; + struct clk *fsysclk; + u32 fifo_depth; + u32 slot_width; + u32 hck_rate[2]; + bool sck_div[2]; + bool slave_mode; + bool synchronous; + char name[32]; +}; + +static irqreturn_t esai_isr(int irq, void *devid) +{ + struct fsl_esai *esai_priv = (struct fsl_esai *)devid; + struct platform_device *pdev = esai_priv->pdev; + u32 esr; + + regmap_read(esai_priv->regmap, REG_ESAI_ESR, &esr); + + if (esr & ESAI_ESR_TINIT_MASK) + dev_dbg(&pdev->dev, "isr: Transmition Initialized\n"); + + if (esr & ESAI_ESR_RFF_MASK) + dev_warn(&pdev->dev, "isr: Receiving overrun\n"); + + if (esr & ESAI_ESR_TFE_MASK) + dev_warn(&pdev->dev, "isr: Transmition underrun\n"); + + if (esr & ESAI_ESR_TLS_MASK) + dev_dbg(&pdev->dev, "isr: Just transmitted the last slot\n"); + + if (esr & ESAI_ESR_TDE_MASK) + dev_dbg(&pdev->dev, "isr: Transmition data exception\n"); + + if (esr & ESAI_ESR_TED_MASK) + dev_dbg(&pdev->dev, "isr: Transmitting even slots\n"); + + if (esr & ESAI_ESR_TD_MASK) + dev_dbg(&pdev->dev, "isr: Transmitting data\n"); + + if (esr & ESAI_ESR_RLS_MASK) + dev_dbg(&pdev->dev, "isr: Just received the last slot\n"); + + if (esr & ESAI_ESR_RDE_MASK) + dev_dbg(&pdev->dev, "isr: Receiving data exception\n"); + + if (esr & ESAI_ESR_RED_MASK) + dev_dbg(&pdev->dev, "isr: Receiving even slots\n"); + + if (esr & ESAI_ESR_RD_MASK) + dev_dbg(&pdev->dev, "isr: Receiving data\n"); + + return IRQ_HANDLED; +} + +/** + * This function is used to calculate the divisors of psr, pm, fp and it is + * supposed to be called in set_dai_sysclk() and set_bclk(). + * + * @ratio: desired overall ratio for the paticipating dividers + * @usefp: for HCK setting, there is no need to set fp divider + * @fp: bypass other dividers by setting fp directly if fp != 0 + * @tx: current setting is for playback or capture + */ +static int fsl_esai_divisor_cal(struct snd_soc_dai *dai, bool tx, u32 ratio, + bool usefp, u32 fp) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + u32 psr, pm = 999, maxfp, prod, sub, savesub, i, j; + + maxfp = usefp ? 16 : 1; + + if (usefp && fp) + goto out_fp; + + if (ratio > 2 * 8 * 256 * maxfp || ratio < 2) { + dev_err(dai->dev, "the ratio is out of range (2 ~ %d)\n", + 2 * 8 * 256 * maxfp); + return -EINVAL; + } else if (ratio % 2) { + dev_err(dai->dev, "the raio must be even if using upper divider\n"); + return -EINVAL; + } + + ratio /= 2; + + psr = ratio <= 256 * maxfp ? ESAI_xCCR_xPSR_BYPASS : ESAI_xCCR_xPSR_DIV8; + + /* Set the max fluctuation -- 0.1% of the max devisor */ + savesub = (psr ? 1 : 8) * 256 * maxfp / 1000; + + /* Find the best value for PM */ + for (i = 1; i <= 256; i++) { + for (j = 1; j <= maxfp; j++) { + /* PSR (1 or 8) * PM (1 ~ 256) * FP (1 ~ 16) */ + prod = (psr ? 1 : 8) * i * j; + + if (prod == ratio) + sub = 0; + else if (prod / ratio == 1) + sub = prod - ratio; + else if (ratio / prod == 1) + sub = ratio - prod; + else + continue; + + /* Calculate the fraction */ + sub = sub * 1000 / ratio; + if (sub < savesub) { + savesub = sub; + pm = i; + fp = j; + } + + /* We are lucky */ + if (savesub == 0) + goto out; + } + } + + if (pm == 999) { + dev_err(dai->dev, "failed to calculate proper divisors\n"); + return -EINVAL; + } + +out: + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCCR(tx), + ESAI_xCCR_xPSR_MASK | ESAI_xCCR_xPM_MASK, + psr | ESAI_xCCR_xPM(pm)); + +out_fp: + /* Bypass fp if not being required */ + if (maxfp <= 1) + return 0; + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCCR(tx), + ESAI_xCCR_xFP_MASK, ESAI_xCCR_xFP(fp)); + + return 0; +} + +/** + * This function mainly configures the clock frequency of MCLK (HCKT/HCKR) + * + * @Parameters: + * clk_id: The clock source of HCKT/HCKR + * (Input from outside; output from inside, FSYS or EXTAL) + * freq: The required clock rate of HCKT/HCKR + * dir: The clock direction of HCKT/HCKR + * + * Note: If the direction is input, we do not care about clk_id. + */ +static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + struct clk *clksrc = esai_priv->extalclk; + bool tx = clk_id <= ESAI_HCKT_EXTAL; + bool in = dir == SND_SOC_CLOCK_IN; + u32 ret, ratio, ecr = 0; + unsigned long clk_rate; + + /* sck_div can be only bypassed if ETO/ERO=0 and SNC_SOC_CLOCK_OUT */ + esai_priv->sck_div[tx] = true; + + /* Set the direction of HCKT/HCKR pins */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCCR(tx), + ESAI_xCCR_xHCKD, in ? 0 : ESAI_xCCR_xHCKD); + + if (in) + goto out; + + switch (clk_id) { + case ESAI_HCKT_FSYS: + case ESAI_HCKR_FSYS: + clksrc = esai_priv->fsysclk; + break; + case ESAI_HCKT_EXTAL: + ecr |= ESAI_ECR_ETI; + case ESAI_HCKR_EXTAL: + ecr |= ESAI_ECR_ERI; + break; + default: + return -EINVAL; + } + + if (IS_ERR(clksrc)) { + dev_err(dai->dev, "no assigned %s clock\n", + clk_id % 2 ? "extal" : "fsys"); + return PTR_ERR(clksrc); + } + clk_rate = clk_get_rate(clksrc); + + ratio = clk_rate / freq; + if (ratio * freq > clk_rate) + ret = ratio * freq - clk_rate; + else if (ratio * freq < clk_rate) + ret = clk_rate - ratio * freq; + else + ret = 0; + + /* Block if clock source can not be divided into the required rate */ + if (ret != 0 && clk_rate / ret < 1000) { + dev_err(dai->dev, "failed to derive required HCK%c rate\n", + tx ? 'T' : 'R'); + return -EINVAL; + } + + if (ratio == 1) { + /* Bypass all the dividers if not being needed */ + ecr |= tx ? ESAI_ECR_ETO : ESAI_ECR_ERO; + goto out; + } + + ret = fsl_esai_divisor_cal(dai, tx, ratio, false, 0); + if (ret) + return ret; + + esai_priv->sck_div[tx] = false; + +out: + esai_priv->hck_rate[tx] = freq; + + regmap_update_bits(esai_priv->regmap, REG_ESAI_ECR, + tx ? ESAI_ECR_ETI | ESAI_ECR_ETO : + ESAI_ECR_ERI | ESAI_ECR_ERO, ecr); + + return 0; +} + +/** + * This function configures the related dividers according to the bclk rate + */ +static int fsl_esai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + u32 hck_rate = esai_priv->hck_rate[tx]; + u32 sub, ratio = hck_rate / freq; + + /* Don't apply for fully slave mode*/ + if (esai_priv->slave_mode) + return 0; + + if (ratio * freq > hck_rate) + sub = ratio * freq - hck_rate; + else if (ratio * freq < hck_rate) + sub = hck_rate - ratio * freq; + else + sub = 0; + + /* Block if clock source can not be divided into the required rate */ + if (sub != 0 && hck_rate / sub < 1000) { + dev_err(dai->dev, "failed to derive required SCK%c rate\n", + tx ? 'T' : 'R'); + return -EINVAL; + } + + if (esai_priv->sck_div[tx] && (ratio > 16 || ratio == 0)) { + dev_err(dai->dev, "the ratio is out of range (1 ~ 16)\n"); + return -EINVAL; + } + + return fsl_esai_divisor_cal(dai, tx, ratio, true, + esai_priv->sck_div[tx] ? 0 : ratio); +} + +static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, + u32 rx_mask, int slots, int slot_width) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, + ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, + ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(tx_mask)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, + ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, + ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(rx_mask)); + + esai_priv->slot_width = slot_width; + + return 0; +} + +static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + u32 xcr = 0, xccr = 0, mask; + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* Data on rising edge of bclk, frame low, 1clk before data */ + xcr |= ESAI_xCR_xFSR; + xccr |= ESAI_xCCR_xFSP | ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* Data on rising edge of bclk, frame high */ + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + /* Data on rising edge of bclk, frame high, right aligned */ + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCR_xWA; + break; + case SND_SOC_DAIFMT_DSP_A: + /* Data on rising edge of bclk, frame high, 1clk before data */ + xcr |= ESAI_xCR_xFSL | ESAI_xCR_xFSR; + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_DSP_B: + /* Data on rising edge of bclk, frame high */ + xcr |= ESAI_xCR_xFSL; + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Nothing to do for both normal cases */ + break; + case SND_SOC_DAIFMT_IB_NF: + /* Invert bit clock */ + xccr ^= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_NB_IF: + /* Invert frame clock */ + xccr ^= ESAI_xCCR_xFSP; + break; + case SND_SOC_DAIFMT_IB_IF: + /* Invert both clocks */ + xccr ^= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP; + break; + default: + return -EINVAL; + } + + esai_priv->slave_mode = false; + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + esai_priv->slave_mode = true; + break; + case SND_SOC_DAIFMT_CBS_CFM: + xccr |= ESAI_xCCR_xCKD; + break; + case SND_SOC_DAIFMT_CBM_CFS: + xccr |= ESAI_xCCR_xFSD; + break; + case SND_SOC_DAIFMT_CBS_CFS: + xccr |= ESAI_xCCR_xFSD | ESAI_xCCR_xCKD; + break; + default: + return -EINVAL; + } + + mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR; + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, xcr); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, mask, xcr); + + mask = ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP | + ESAI_xCCR_xFSD | ESAI_xCCR_xCKD | ESAI_xCR_xWA; + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, mask, xccr); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, mask, xccr); + + return 0; +} + +static int fsl_esai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + /* + * Some platforms might use the same bit to gate all three or two of + * clocks, so keep all clocks open/close at the same time for safety + */ + clk_prepare_enable(esai_priv->coreclk); + if (!IS_ERR(esai_priv->extalclk)) + clk_prepare_enable(esai_priv->extalclk); + if (!IS_ERR(esai_priv->fsysclk)) + clk_prepare_enable(esai_priv->fsysclk); + + if (!dai->active) { + /* Reset Port C */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC, + ESAI_PRRC_PDC_MASK, ESAI_PRRC_PDC(ESAI_GPIO)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC, + ESAI_PCRC_PC_MASK, ESAI_PCRC_PC(ESAI_GPIO)); + + /* Set synchronous mode */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_SAICR, + ESAI_SAICR_SYNC, esai_priv->synchronous ? + ESAI_SAICR_SYNC : 0); + + /* Set a default slot number -- 2 */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + } + + return 0; +} + +static int fsl_esai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 width = snd_pcm_format_width(params_format(params)); + u32 channels = params_channels(params); + u32 bclk, mask, val, ret; + + bclk = params_rate(params) * esai_priv->slot_width * 2; + + ret = fsl_esai_set_bclk(dai, tx, bclk); + if (ret) + return ret; + + /* Use Normal mode to support monaural audio */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), + ESAI_xCR_xMOD_MASK, params_channels(params) > 1 ? + ESAI_xCR_xMOD_NETWORK : 0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFR_MASK, ESAI_xFCR_xFR); + + mask = ESAI_xFCR_xFR_MASK | ESAI_xFCR_xWA_MASK | ESAI_xFCR_xFWM_MASK | + (tx ? ESAI_xFCR_TE_MASK | ESAI_xFCR_TIEN : ESAI_xFCR_RE_MASK); + val = ESAI_xFCR_xWA(width) | ESAI_xFCR_xFWM(esai_priv->fifo_depth) | + (tx ? ESAI_xFCR_TE(channels) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(channels)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val); + + mask = ESAI_xCR_xSWS_MASK | (tx ? ESAI_xCR_PADC : 0); + val = ESAI_xCR_xSWS(esai_priv->slot_width, width) | (tx ? ESAI_xCR_PADC : 0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val); + + return 0; +} + +static void fsl_esai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + if (!IS_ERR(esai_priv->fsysclk)) + clk_disable_unprepare(esai_priv->fsysclk); + if (!IS_ERR(esai_priv->extalclk)) + clk_disable_unprepare(esai_priv->extalclk); + clk_disable_unprepare(esai_priv->coreclk); +} + +static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u8 i, channels = substream->runtime->channels; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFEN_MASK, ESAI_xFCR_xFEN); + + /* Write initial words reqiured by ESAI as normal procedure */ + for (i = 0; tx && i < channels; i++) + regmap_write(esai_priv->regmap, REG_ESAI_ETDR, 0x0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), + tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, + tx ? ESAI_xCR_TE(channels) : ESAI_xCR_RE(channels)); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), + tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0); + + /* Disable and reset FIFO */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFR | ESAI_xFCR_xFEN, ESAI_xFCR_xFR); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFR, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_dai_ops fsl_esai_dai_ops = { + .startup = fsl_esai_startup, + .shutdown = fsl_esai_shutdown, + .trigger = fsl_esai_trigger, + .hw_params = fsl_esai_hw_params, + .set_sysclk = fsl_esai_set_dai_sysclk, + .set_fmt = fsl_esai_set_dai_fmt, + .set_tdm_slot = fsl_esai_set_dai_tdm_slot, +}; + +static int fsl_esai_dai_probe(struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &esai_priv->dma_params_tx, + &esai_priv->dma_params_rx); + + return 0; +} + +static struct snd_soc_dai_driver fsl_esai_dai = { + .probe = fsl_esai_dai_probe, + .playback = { + .channels_min = 1, + .channels_max = 12, + .rates = FSL_ESAI_RATES, + .formats = FSL_ESAI_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = FSL_ESAI_RATES, + .formats = FSL_ESAI_FORMATS, + }, + .ops = &fsl_esai_dai_ops, +}; + +static const struct snd_soc_component_driver fsl_esai_component = { + .name = "fsl-esai", +}; + +static bool fsl_esai_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ESAI_ERDR: + case REG_ESAI_ECR: + case REG_ESAI_ESR: + case REG_ESAI_TFCR: + case REG_ESAI_TFSR: + case REG_ESAI_RFCR: + case REG_ESAI_RFSR: + case REG_ESAI_RX0: + case REG_ESAI_RX1: + case REG_ESAI_RX2: + case REG_ESAI_RX3: + case REG_ESAI_SAISR: + case REG_ESAI_SAICR: + case REG_ESAI_TCR: + case REG_ESAI_TCCR: + case REG_ESAI_RCR: + case REG_ESAI_RCCR: + case REG_ESAI_TSMA: + case REG_ESAI_TSMB: + case REG_ESAI_RSMA: + case REG_ESAI_RSMB: + case REG_ESAI_PRRC: + case REG_ESAI_PCRC: + return true; + default: + return false; + } +} + +static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ESAI_ETDR: + case REG_ESAI_ECR: + case REG_ESAI_TFCR: + case REG_ESAI_RFCR: + case REG_ESAI_TX0: + case REG_ESAI_TX1: + case REG_ESAI_TX2: + case REG_ESAI_TX3: + case REG_ESAI_TX4: + case REG_ESAI_TX5: + case REG_ESAI_TSR: + case REG_ESAI_SAICR: + case REG_ESAI_TCR: + case REG_ESAI_TCCR: + case REG_ESAI_RCR: + case REG_ESAI_RCCR: + case REG_ESAI_TSMA: + case REG_ESAI_TSMB: + case REG_ESAI_RSMA: + case REG_ESAI_RSMB: + case REG_ESAI_PRRC: + case REG_ESAI_PCRC: + return true; + default: + return false; + } +} + +static const struct regmap_config fsl_esai_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + + .max_register = REG_ESAI_PCRC, + .readable_reg = fsl_esai_readable_reg, + .writeable_reg = fsl_esai_writeable_reg, +}; + +static int fsl_esai_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct fsl_esai *esai_priv; + struct resource *res; + const uint32_t *iprop; + void __iomem *regs; + int irq, ret; + + esai_priv = devm_kzalloc(&pdev->dev, sizeof(*esai_priv), GFP_KERNEL); + if (!esai_priv) + return -ENOMEM; + + esai_priv->pdev = pdev; + strcpy(esai_priv->name, np->name); + + /* Get the addresses and IRQ */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + esai_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, + "core", regs, &fsl_esai_regmap_config); + if (IS_ERR(esai_priv->regmap)) { + dev_err(&pdev->dev, "failed to init regmap: %ld\n", + PTR_ERR(esai_priv->regmap)); + return PTR_ERR(esai_priv->regmap); + } + + esai_priv->coreclk = devm_clk_get(&pdev->dev, "core"); + if (IS_ERR(esai_priv->coreclk)) { + dev_err(&pdev->dev, "failed to get core clock: %ld\n", + PTR_ERR(esai_priv->coreclk)); + return PTR_ERR(esai_priv->coreclk); + } + + esai_priv->extalclk = devm_clk_get(&pdev->dev, "extal"); + if (IS_ERR(esai_priv->extalclk)) + dev_warn(&pdev->dev, "failed to get extal clock: %ld\n", + PTR_ERR(esai_priv->extalclk)); + + esai_priv->fsysclk = devm_clk_get(&pdev->dev, "fsys"); + if (IS_ERR(esai_priv->fsysclk)) + dev_warn(&pdev->dev, "failed to get fsys clock: %ld\n", + PTR_ERR(esai_priv->fsysclk)); + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + return irq; + } + + ret = devm_request_irq(&pdev->dev, irq, esai_isr, 0, + esai_priv->name, esai_priv); + if (ret) { + dev_err(&pdev->dev, "failed to claim irq %u\n", irq); + return ret; + } + + /* Set a default slot size */ + esai_priv->slot_width = 32; + + /* Set a default master/slave state */ + esai_priv->slave_mode = true; + + /* Determine the FIFO depth */ + iprop = of_get_property(np, "fsl,fifo-depth", NULL); + if (iprop) + esai_priv->fifo_depth = be32_to_cpup(iprop); + else + esai_priv->fifo_depth = 64; + + esai_priv->dma_params_tx.maxburst = 16; + esai_priv->dma_params_rx.maxburst = 16; + esai_priv->dma_params_tx.addr = res->start + REG_ESAI_ETDR; + esai_priv->dma_params_rx.addr = res->start + REG_ESAI_ERDR; + + esai_priv->synchronous = + of_property_read_bool(np, "fsl,esai-synchronous"); + + /* Implement full symmetry for synchronous mode */ + if (esai_priv->synchronous) { + fsl_esai_dai.symmetric_rates = 1; + fsl_esai_dai.symmetric_channels = 1; + fsl_esai_dai.symmetric_samplebits = 1; + } + + dev_set_drvdata(&pdev->dev, esai_priv); + + /* Reset ESAI unit */ + ret = regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ERST); + if (ret) { + dev_err(&pdev->dev, "failed to reset ESAI: %d\n", ret); + return ret; + } + + /* + * We need to enable ESAI so as to access some of its registers. + * Otherwise, we would fail to dump regmap from user space. + */ + ret = regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ESAIEN); + if (ret) { + dev_err(&pdev->dev, "failed to enable ESAI: %d\n", ret); + return ret; + } + + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component, + &fsl_esai_dai, 1); + if (ret) { + dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); + return ret; + } + + ret = imx_pcm_dma_init(pdev); + if (ret) + dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); + + return ret; +} + +static const struct of_device_id fsl_esai_dt_ids[] = { + { .compatible = "fsl,imx35-esai", }, + {} +}; +MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids); + +static struct platform_driver fsl_esai_driver = { + .probe = fsl_esai_probe, + .driver = { + .name = "fsl-esai-dai", + .owner = THIS_MODULE, + .of_match_table = fsl_esai_dt_ids, + }, +}; + +module_platform_driver(fsl_esai_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale ESAI CPU DAI driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:fsl-esai-dai"); diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h new file mode 100644 index 0000000..9c9f957 --- /dev/null +++ b/sound/soc/fsl/fsl_esai.h @@ -0,0 +1,354 @@ +/* + * fsl_esai.h - ALSA ESAI interface for the Freescale i.MX SoC + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _FSL_ESAI_DAI_H +#define _FSL_ESAI_DAI_H + +/* ESAI Register Map */ +#define REG_ESAI_ETDR 0x00 +#define REG_ESAI_ERDR 0x04 +#define REG_ESAI_ECR 0x08 +#define REG_ESAI_ESR 0x0C +#define REG_ESAI_TFCR 0x10 +#define REG_ESAI_TFSR 0x14 +#define REG_ESAI_RFCR 0x18 +#define REG_ESAI_RFSR 0x1C +#define REG_ESAI_xFCR(tx) (tx ? REG_ESAI_TFCR : REG_ESAI_RFCR) +#define REG_ESAI_xFSR(tx) (tx ? REG_ESAI_TFSR : REG_ESAI_RFSR) +#define REG_ESAI_TX0 0x80 +#define REG_ESAI_TX1 0x84 +#define REG_ESAI_TX2 0x88 +#define REG_ESAI_TX3 0x8C +#define REG_ESAI_TX4 0x90 +#define REG_ESAI_TX5 0x94 +#define REG_ESAI_TSR 0x98 +#define REG_ESAI_RX0 0xA0 +#define REG_ESAI_RX1 0xA4 +#define REG_ESAI_RX2 0xA8 +#define REG_ESAI_RX3 0xAC +#define REG_ESAI_SAISR 0xCC +#define REG_ESAI_SAICR 0xD0 +#define REG_ESAI_TCR 0xD4 +#define REG_ESAI_TCCR 0xD8 +#define REG_ESAI_RCR 0xDC +#define REG_ESAI_RCCR 0xE0 +#define REG_ESAI_xCR(tx) (tx ? REG_ESAI_TCR : REG_ESAI_RCR) +#define REG_ESAI_xCCR(tx) (tx ? REG_ESAI_TCCR : REG_ESAI_RCCR) +#define REG_ESAI_TSMA 0xE4 +#define REG_ESAI_TSMB 0xE8 +#define REG_ESAI_RSMA 0xEC +#define REG_ESAI_RSMB 0xF0 +#define REG_ESAI_xSMA(tx) (tx ? REG_ESAI_TSMA : REG_ESAI_RSMA) +#define REG_ESAI_xSMB(tx) (tx ? REG_ESAI_TSMB : REG_ESAI_RSMB) +#define REG_ESAI_PRRC 0xF8 +#define REG_ESAI_PCRC 0xFC + +/* ESAI Control Register -- REG_ESAI_ECR 0x8 */ +#define ESAI_ECR_ETI_SHIFT 19 +#define ESAI_ECR_ETI_MASK (1 << ESAI_ECR_ETI_SHIFT) +#define ESAI_ECR_ETI (1 << ESAI_ECR_ETI_SHIFT) +#define ESAI_ECR_ETO_SHIFT 18 +#define ESAI_ECR_ETO_MASK (1 << ESAI_ECR_ETO_SHIFT) +#define ESAI_ECR_ETO (1 << ESAI_ECR_ETO_SHIFT) +#define ESAI_ECR_ERI_SHIFT 17 +#define ESAI_ECR_ERI_MASK (1 << ESAI_ECR_ERI_SHIFT) +#define ESAI_ECR_ERI (1 << ESAI_ECR_ERI_SHIFT) +#define ESAI_ECR_ERO_SHIFT 16 +#define ESAI_ECR_ERO_MASK (1 << ESAI_ECR_ERO_SHIFT) +#define ESAI_ECR_ERO (1 << ESAI_ECR_ERO_SHIFT) +#define ESAI_ECR_ERST_SHIFT 1 +#define ESAI_ECR_ERST_MASK (1 << ESAI_ECR_ERST_SHIFT) +#define ESAI_ECR_ERST (1 << ESAI_ECR_ERST_SHIFT) +#define ESAI_ECR_ESAIEN_SHIFT 0 +#define ESAI_ECR_ESAIEN_MASK (1 << ESAI_ECR_ESAIEN_SHIFT) +#define ESAI_ECR_ESAIEN (1 << ESAI_ECR_ESAIEN_SHIFT) + +/* ESAI Status Register -- REG_ESAI_ESR 0xC */ +#define ESAI_ESR_TINIT_SHIFT 10 +#define ESAI_ESR_TINIT_MASK (1 << ESAI_ESR_TINIT_SHIFT) +#define ESAI_ESR_TINIT (1 << ESAI_ESR_TINIT_SHIFT) +#define ESAI_ESR_RFF_SHIFT 9 +#define ESAI_ESR_RFF_MASK (1 << ESAI_ESR_RFF_SHIFT) +#define ESAI_ESR_RFF (1 << ESAI_ESR_RFF_SHIFT) +#define ESAI_ESR_TFE_SHIFT 8 +#define ESAI_ESR_TFE_MASK (1 << ESAI_ESR_TFE_SHIFT) +#define ESAI_ESR_TFE (1 << ESAI_ESR_TFE_SHIFT) +#define ESAI_ESR_TLS_SHIFT 7 +#define ESAI_ESR_TLS_MASK (1 << ESAI_ESR_TLS_SHIFT) +#define ESAI_ESR_TLS (1 << ESAI_ESR_TLS_SHIFT) +#define ESAI_ESR_TDE_SHIFT 6 +#define ESAI_ESR_TDE_MASK (1 << ESAI_ESR_TDE_SHIFT) +#define ESAI_ESR_TDE (1 << ESAI_ESR_TDE_SHIFT) +#define ESAI_ESR_TED_SHIFT 5 +#define ESAI_ESR_TED_MASK (1 << ESAI_ESR_TED_SHIFT) +#define ESAI_ESR_TED (1 << ESAI_ESR_TED_SHIFT) +#define ESAI_ESR_TD_SHIFT 4 +#define ESAI_ESR_TD_MASK (1 << ESAI_ESR_TD_SHIFT) +#define ESAI_ESR_TD (1 << ESAI_ESR_TD_SHIFT) +#define ESAI_ESR_RLS_SHIFT 3 +#define ESAI_ESR_RLS_MASK (1 << ESAI_ESR_RLS_SHIFT) +#define ESAI_ESR_RLS (1 << ESAI_ESR_RLS_SHIFT) +#define ESAI_ESR_RDE_SHIFT 2 +#define ESAI_ESR_RDE_MASK (1 << ESAI_ESR_RDE_SHIFT) +#define ESAI_ESR_RDE (1 << ESAI_ESR_RDE_SHIFT) +#define ESAI_ESR_RED_SHIFT 1 +#define ESAI_ESR_RED_MASK (1 << ESAI_ESR_RED_SHIFT) +#define ESAI_ESR_RED (1 << ESAI_ESR_RED_SHIFT) +#define ESAI_ESR_RD_SHIFT 0 +#define ESAI_ESR_RD_MASK (1 << ESAI_ESR_RD_SHIFT) +#define ESAI_ESR_RD (1 << ESAI_ESR_RD_SHIFT) + +/* + * Transmit FIFO Configuration Register -- REG_ESAI_TFCR 0x10 + * Receive FIFO Configuration Register -- REG_ESAI_RFCR 0x18 + */ +#define ESAI_xFCR_TIEN_SHIFT 19 +#define ESAI_xFCR_TIEN_MASK (1 << ESAI_xFCR_TIEN_SHIFT) +#define ESAI_xFCR_TIEN (1 << ESAI_xFCR_TIEN_SHIFT) +#define ESAI_xFCR_REXT_SHIFT 19 +#define ESAI_xFCR_REXT_MASK (1 << ESAI_xFCR_REXT_SHIFT) +#define ESAI_xFCR_REXT (1 << ESAI_xFCR_REXT_SHIFT) +#define ESAI_xFCR_xWA_SHIFT 16 +#define ESAI_xFCR_xWA_WIDTH 3 +#define ESAI_xFCR_xWA_MASK (((1 << ESAI_xFCR_xWA_WIDTH) - 1) << ESAI_xFCR_xWA_SHIFT) +#define ESAI_xFCR_xWA(v) (((8 - ((v) >> 2)) << ESAI_xFCR_xWA_SHIFT) & ESAI_xFCR_xWA_MASK) +#define ESAI_xFCR_xFWM_SHIFT 8 +#define ESAI_xFCR_xFWM_WIDTH 8 +#define ESAI_xFCR_xFWM_MASK (((1 << ESAI_xFCR_xFWM_WIDTH) - 1) << ESAI_xFCR_xFWM_SHIFT) +#define ESAI_xFCR_xFWM(v) ((((v) - 1) << ESAI_xFCR_xFWM_SHIFT) & ESAI_xFCR_xFWM_MASK) +#define ESAI_xFCR_xE_SHIFT 2 +#define ESAI_xFCR_TE_WIDTH 6 +#define ESAI_xFCR_RE_WIDTH 4 +#define ESAI_xFCR_TE_MASK (((1 << ESAI_xFCR_TE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) +#define ESAI_xFCR_RE_MASK (((1 << ESAI_xFCR_RE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) +#define ESAI_xFCR_TE(x) ((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_TE_MASK) +#define ESAI_xFCR_RE(x) ((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_RE_MASK) +#define ESAI_xFCR_xFR_SHIFT 1 +#define ESAI_xFCR_xFR_MASK (1 << ESAI_xFCR_xFR_SHIFT) +#define ESAI_xFCR_xFR (1 << ESAI_xFCR_xFR_SHIFT) +#define ESAI_xFCR_xFEN_SHIFT 0 +#define ESAI_xFCR_xFEN_MASK (1 << ESAI_xFCR_xFEN_SHIFT) +#define ESAI_xFCR_xFEN (1 << ESAI_xFCR_xFEN_SHIFT) + +/* + * Transmit FIFO Status Register -- REG_ESAI_TFSR 0x14 + * Receive FIFO Status Register --REG_ESAI_RFSR 0x1C + */ +#define ESAI_xFSR_NTFO_SHIFT 12 +#define ESAI_xFSR_NRFI_SHIFT 12 +#define ESAI_xFSR_NTFI_SHIFT 8 +#define ESAI_xFSR_NRFO_SHIFT 8 +#define ESAI_xFSR_NTFx_WIDTH 3 +#define ESAI_xFSR_NRFx_WIDTH 2 +#define ESAI_xFSR_NTFO_MASK (((1 << ESAI_xFSR_NTFx_WIDTH) - 1) << ESAI_xFSR_NTFO_SHIFT) +#define ESAI_xFSR_NTFI_MASK (((1 << ESAI_xFSR_NTFx_WIDTH) - 1) << ESAI_xFSR_NTFI_SHIFT) +#define ESAI_xFSR_NRFO_MASK (((1 << ESAI_xFSR_NRFx_WIDTH) - 1) << ESAI_xFSR_NRFO_SHIFT) +#define ESAI_xFSR_NRFI_MASK (((1 << ESAI_xFSR_NRFx_WIDTH) - 1) << ESAI_xFSR_NRFI_SHIFT) +#define ESAI_xFSR_xFCNT_SHIFT 0 +#define ESAI_xFSR_xFCNT_WIDTH 8 +#define ESAI_xFSR_xFCNT_MASK (((1 << ESAI_xFSR_xFCNT_WIDTH) - 1) << ESAI_xFSR_xFCNT_SHIFT) + +/* ESAI Transmit Slot Register -- REG_ESAI_TSR 0x98 */ +#define ESAI_TSR_SHIFT 0 +#define ESAI_TSR_WIDTH 24 +#define ESAI_TSR_MASK (((1 << ESAI_TSR_WIDTH) - 1) << ESAI_TSR_SHIFT) + +/* Serial Audio Interface Status Register -- REG_ESAI_SAISR 0xCC */ +#define ESAI_SAISR_TODFE_SHIFT 17 +#define ESAI_SAISR_TODFE_MASK (1 << ESAI_SAISR_TODFE_SHIFT) +#define ESAI_SAISR_TODFE (1 << ESAI_SAISR_TODFE_SHIFT) +#define ESAI_SAISR_TEDE_SHIFT 16 +#define ESAI_SAISR_TEDE_MASK (1 << ESAI_SAISR_TEDE_SHIFT) +#define ESAI_SAISR_TEDE (1 << ESAI_SAISR_TEDE_SHIFT) +#define ESAI_SAISR_TDE_SHIFT 15 +#define ESAI_SAISR_TDE_MASK (1 << ESAI_SAISR_TDE_SHIFT) +#define ESAI_SAISR_TDE (1 << ESAI_SAISR_TDE_SHIFT) +#define ESAI_SAISR_TUE_SHIFT 14 +#define ESAI_SAISR_TUE_MASK (1 << ESAI_SAISR_TUE_SHIFT) +#define ESAI_SAISR_TUE (1 << ESAI_SAISR_TUE_SHIFT) +#define ESAI_SAISR_TFS_SHIFT 13 +#define ESAI_SAISR_TFS_MASK (1 << ESAI_SAISR_TFS_SHIFT) +#define ESAI_SAISR_TFS (1 << ESAI_SAISR_TFS_SHIFT) +#define ESAI_SAISR_RODF_SHIFT 10 +#define ESAI_SAISR_RODF_MASK (1 << ESAI_SAISR_RODF_SHIFT) +#define ESAI_SAISR_RODF (1 << ESAI_SAISR_RODF_SHIFT) +#define ESAI_SAISR_REDF_SHIFT 9 +#define ESAI_SAISR_REDF_MASK (1 << ESAI_SAISR_REDF_SHIFT) +#define ESAI_SAISR_REDF (1 << ESAI_SAISR_REDF_SHIFT) +#define ESAI_SAISR_RDF_SHIFT 8 +#define ESAI_SAISR_RDF_MASK (1 << ESAI_SAISR_RDF_SHIFT) +#define ESAI_SAISR_RDF (1 << ESAI_SAISR_RDF_SHIFT) +#define ESAI_SAISR_ROE_SHIFT 7 +#define ESAI_SAISR_ROE_MASK (1 << ESAI_SAISR_ROE_SHIFT) +#define ESAI_SAISR_ROE (1 << ESAI_SAISR_ROE_SHIFT) +#define ESAI_SAISR_RFS_SHIFT 6 +#define ESAI_SAISR_RFS_MASK (1 << ESAI_SAISR_RFS_SHIFT) +#define ESAI_SAISR_RFS (1 << ESAI_SAISR_RFS_SHIFT) +#define ESAI_SAISR_IF2_SHIFT 2 +#define ESAI_SAISR_IF2_MASK (1 << ESAI_SAISR_IF2_SHIFT) +#define ESAI_SAISR_IF2 (1 << ESAI_SAISR_IF2_SHIFT) +#define ESAI_SAISR_IF1_SHIFT 1 +#define ESAI_SAISR_IF1_MASK (1 << ESAI_SAISR_IF1_SHIFT) +#define ESAI_SAISR_IF1 (1 << ESAI_SAISR_IF1_SHIFT) +#define ESAI_SAISR_IF0_SHIFT 0 +#define ESAI_SAISR_IF0_MASK (1 << ESAI_SAISR_IF0_SHIFT) +#define ESAI_SAISR_IF0 (1 << ESAI_SAISR_IF0_SHIFT) + +/* Serial Audio Interface Control Register -- REG_ESAI_SAICR 0xD0 */ +#define ESAI_SAICR_ALC_SHIFT 8 +#define ESAI_SAICR_ALC_MASK (1 << ESAI_SAICR_ALC_SHIFT) +#define ESAI_SAICR_ALC (1 << ESAI_SAICR_ALC_SHIFT) +#define ESAI_SAICR_TEBE_SHIFT 7 +#define ESAI_SAICR_TEBE_MASK (1 << ESAI_SAICR_TEBE_SHIFT) +#define ESAI_SAICR_TEBE (1 << ESAI_SAICR_TEBE_SHIFT) +#define ESAI_SAICR_SYNC_SHIFT 6 +#define ESAI_SAICR_SYNC_MASK (1 << ESAI_SAICR_SYNC_SHIFT) +#define ESAI_SAICR_SYNC (1 << ESAI_SAICR_SYNC_SHIFT) +#define ESAI_SAICR_OF2_SHIFT 2 +#define ESAI_SAICR_OF2_MASK (1 << ESAI_SAICR_OF2_SHIFT) +#define ESAI_SAICR_OF2 (1 << ESAI_SAICR_OF2_SHIFT) +#define ESAI_SAICR_OF1_SHIFT 1 +#define ESAI_SAICR_OF1_MASK (1 << ESAI_SAICR_OF1_SHIFT) +#define ESAI_SAICR_OF1 (1 << ESAI_SAICR_OF1_SHIFT) +#define ESAI_SAICR_OF0_SHIFT 0 +#define ESAI_SAICR_OF0_MASK (1 << ESAI_SAICR_OF0_SHIFT) +#define ESAI_SAICR_OF0 (1 << ESAI_SAICR_OF0_SHIFT) + +/* + * Transmit Control Register -- REG_ESAI_TCR 0xD4 + * Receive Control Register -- REG_ESAI_RCR 0xDC + */ +#define ESAI_xCR_xLIE_SHIFT 23 +#define ESAI_xCR_xLIE_MASK (1 << ESAI_xCR_xLIE_SHIFT) +#define ESAI_xCR_xLIE (1 << ESAI_xCR_xLIE_SHIFT) +#define ESAI_xCR_xIE_SHIFT 22 +#define ESAI_xCR_xIE_MASK (1 << ESAI_xCR_xIE_SHIFT) +#define ESAI_xCR_xIE (1 << ESAI_xCR_xIE_SHIFT) +#define ESAI_xCR_xEDIE_SHIFT 21 +#define ESAI_xCR_xEDIE_MASK (1 << ESAI_xCR_xEDIE_SHIFT) +#define ESAI_xCR_xEDIE (1 << ESAI_xCR_xEDIE_SHIFT) +#define ESAI_xCR_xEIE_SHIFT 20 +#define ESAI_xCR_xEIE_MASK (1 << ESAI_xCR_xEIE_SHIFT) +#define ESAI_xCR_xEIE (1 << ESAI_xCR_xEIE_SHIFT) +#define ESAI_xCR_xPR_SHIFT 19 +#define ESAI_xCR_xPR_MASK (1 << ESAI_xCR_xPR_SHIFT) +#define ESAI_xCR_xPR (1 << ESAI_xCR_xPR_SHIFT) +#define ESAI_xCR_PADC_SHIFT 17 +#define ESAI_xCR_PADC_MASK (1 << ESAI_xCR_PADC_SHIFT) +#define ESAI_xCR_PADC (1 << ESAI_xCR_PADC_SHIFT) +#define ESAI_xCR_xFSR_SHIFT 16 +#define ESAI_xCR_xFSR_MASK (1 << ESAI_xCR_xFSR_SHIFT) +#define ESAI_xCR_xFSR (1 << ESAI_xCR_xFSR_SHIFT) +#define ESAI_xCR_xFSL_SHIFT 15 +#define ESAI_xCR_xFSL_MASK (1 << ESAI_xCR_xFSL_SHIFT) +#define ESAI_xCR_xFSL (1 << ESAI_xCR_xFSL_SHIFT) +#define ESAI_xCR_xSWS_SHIFT 10 +#define ESAI_xCR_xSWS_WIDTH 5 +#define ESAI_xCR_xSWS_MASK (((1 << ESAI_xCR_xSWS_WIDTH) - 1) << ESAI_xCR_xSWS_SHIFT) +#define ESAI_xCR_xSWS(s, w) ((w < 24 ? (s - w + ((w - 8) >> 2)) : (s < 32 ? 0x1e : 0x1f)) << ESAI_xCR_xSWS_SHIFT) +#define ESAI_xCR_xMOD_SHIFT 8 +#define ESAI_xCR_xMOD_WIDTH 2 +#define ESAI_xCR_xMOD_MASK (((1 << ESAI_xCR_xMOD_WIDTH) - 1) << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xMOD_ONDEMAND (0x1 << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xMOD_NETWORK (0x1 << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xMOD_AC97 (0x3 << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xWA_SHIFT 7 +#define ESAI_xCR_xWA_MASK (1 << ESAI_xCR_xWA_SHIFT) +#define ESAI_xCR_xWA (1 << ESAI_xCR_xWA_SHIFT) +#define ESAI_xCR_xSHFD_SHIFT 6 +#define ESAI_xCR_xSHFD_MASK (1 << ESAI_xCR_xSHFD_SHIFT) +#define ESAI_xCR_xSHFD (1 << ESAI_xCR_xSHFD_SHIFT) +#define ESAI_xCR_xE_SHIFT 0 +#define ESAI_xCR_TE_WIDTH 6 +#define ESAI_xCR_RE_WIDTH 4 +#define ESAI_xCR_TE_MASK (((1 << ESAI_xCR_TE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) +#define ESAI_xCR_RE_MASK (((1 << ESAI_xCR_RE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) +#define ESAI_xCR_TE(x) ((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_TE_MASK) +#define ESAI_xCR_RE(x) ((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_RE_MASK) + +/* + * Transmit Clock Control Register -- REG_ESAI_TCCR 0xD8 + * Receive Clock Control Register -- REG_ESAI_RCCR 0xE0 + */ +#define ESAI_xCCR_xHCKD_SHIFT 23 +#define ESAI_xCCR_xHCKD_MASK (1 << ESAI_xCCR_xHCKD_SHIFT) +#define ESAI_xCCR_xHCKD (1 << ESAI_xCCR_xHCKD_SHIFT) +#define ESAI_xCCR_xFSD_SHIFT 22 +#define ESAI_xCCR_xFSD_MASK (1 << ESAI_xCCR_xFSD_SHIFT) +#define ESAI_xCCR_xFSD (1 << ESAI_xCCR_xFSD_SHIFT) +#define ESAI_xCCR_xCKD_SHIFT 21 +#define ESAI_xCCR_xCKD_MASK (1 << ESAI_xCCR_xCKD_SHIFT) +#define ESAI_xCCR_xCKD (1 << ESAI_xCCR_xCKD_SHIFT) +#define ESAI_xCCR_xHCKP_SHIFT 20 +#define ESAI_xCCR_xHCKP_MASK (1 << ESAI_xCCR_xHCKP_SHIFT) +#define ESAI_xCCR_xHCKP (1 << ESAI_xCCR_xHCKP_SHIFT) +#define ESAI_xCCR_xFSP_SHIFT 19 +#define ESAI_xCCR_xFSP_MASK (1 << ESAI_xCCR_xFSP_SHIFT) +#define ESAI_xCCR_xFSP (1 << ESAI_xCCR_xFSP_SHIFT) +#define ESAI_xCCR_xCKP_SHIFT 18 +#define ESAI_xCCR_xCKP_MASK (1 << ESAI_xCCR_xCKP_SHIFT) +#define ESAI_xCCR_xCKP (1 << ESAI_xCCR_xCKP_SHIFT) +#define ESAI_xCCR_xFP_SHIFT 14 +#define ESAI_xCCR_xFP_WIDTH 4 +#define ESAI_xCCR_xFP_MASK (((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT) +#define ESAI_xCCR_xFP(v) ((((v) - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK) +#define ESAI_xCCR_xDC_SHIFT 9 +#define ESAI_xCCR_xDC_WIDTH 4 +#define ESAI_xCCR_xDC_MASK (((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT) +#define ESAI_xCCR_xDC(v) ((((v) - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK) +#define ESAI_xCCR_xPSR_SHIFT 8 +#define ESAI_xCCR_xPSR_MASK (1 << ESAI_xCCR_xPSR_SHIFT) +#define ESAI_xCCR_xPSR_BYPASS (1 << ESAI_xCCR_xPSR_SHIFT) +#define ESAI_xCCR_xPSR_DIV8 (0 << ESAI_xCCR_xPSR_SHIFT) +#define ESAI_xCCR_xPM_SHIFT 0 +#define ESAI_xCCR_xPM_WIDTH 8 +#define ESAI_xCCR_xPM_MASK (((1 << ESAI_xCCR_xPM_WIDTH) - 1) << ESAI_xCCR_xPM_SHIFT) +#define ESAI_xCCR_xPM(v) ((((v) - 1) << ESAI_xCCR_xPM_SHIFT) & ESAI_xCCR_xPM_MASK) + +/* Transmit Slot Mask Register A/B -- REG_ESAI_TSMA/B 0xE4 ~ 0xF0 */ +#define ESAI_xSMA_xS_SHIFT 0 +#define ESAI_xSMA_xS_WIDTH 16 +#define ESAI_xSMA_xS_MASK (((1 << ESAI_xSMA_xS_WIDTH) - 1) << ESAI_xSMA_xS_SHIFT) +#define ESAI_xSMA_xS(v) ((v) & ESAI_xSMA_xS_MASK) +#define ESAI_xSMB_xS_SHIFT 0 +#define ESAI_xSMB_xS_WIDTH 16 +#define ESAI_xSMB_xS_MASK (((1 << ESAI_xSMB_xS_WIDTH) - 1) << ESAI_xSMB_xS_SHIFT) +#define ESAI_xSMB_xS(v) (((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMA_xS_MASK) + +/* Port C Direction Register -- REG_ESAI_PRRC 0xF8 */ +#define ESAI_PRRC_PDC_SHIFT 0 +#define ESAI_PRRC_PDC_WIDTH 12 +#define ESAI_PRRC_PDC_MASK (((1 << ESAI_PRRC_PDC_WIDTH) - 1) << ESAI_PRRC_PDC_SHIFT) +#define ESAI_PRRC_PDC(v) ((v) & ESAI_PRRC_PDC_MASK) + +/* Port C Control Register -- REG_ESAI_PCRC 0xFC */ +#define ESAI_PCRC_PC_SHIFT 0 +#define ESAI_PCRC_PC_WIDTH 12 +#define ESAI_PCRC_PC_MASK (((1 << ESAI_PCRC_PC_WIDTH) - 1) << ESAI_PCRC_PC_SHIFT) +#define ESAI_PCRC_PC(v) ((v) & ESAI_PCRC_PC_MASK) + +#define ESAI_GPIO 0xfff + +/* ESAI clock source */ +#define ESAI_HCKT_FSYS 0 +#define ESAI_HCKT_EXTAL 1 +#define ESAI_HCKR_FSYS 2 +#define ESAI_HCKR_EXTAL 3 + +/* ESAI clock divider */ +#define ESAI_TX_DIV_PSR 0 +#define ESAI_TX_DIV_PM 1 +#define ESAI_TX_DIV_FP 2 +#define ESAI_RX_DIV_PSR 3 +#define ESAI_RX_DIV_PM 4 +#define ESAI_RX_DIV_FP 5 +#endif /* _FSL_ESAI_DAI_H */ -- 1.8.4

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[alsa-devel] [PATCH v2] ASoC: imx: Allow using sound with DT descriptions only
by Alexander Shiyan 10 Jan '14

10 Jan '14

This patch allow using generic sound cards/codecs based on DT description. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/Kconfig | 25 +++---------------------- sound/soc/fsl/Makefile | 5 ++--- sound/soc/fsl/imx-pcm.h | 7 ------- 3 files changed, 5 insertions(+), 32 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 1c7b3b7..cf0d5af 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -7,6 +7,7 @@ config SND_SOC_FSL_SSI config SND_SOC_FSL_SPDIF tristate + select REGMAP_MMIO config SND_SOC_FSL_UTILS tristate @@ -106,6 +107,8 @@ endif # SND_POWERPC_SOC menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" depends on ARCH_MXC || COMPILE_TEST + select SND_SOC_FSL_SSI if OF + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the i.MX CPUs. @@ -119,19 +122,11 @@ config SND_SOC_IMX_PCM_FIQ tristate select FIQ -config SND_SOC_IMX_PCM_DMA - tristate - select SND_SOC_GENERIC_DMAENGINE_PCM - -config SND_SOC_IMX_AUDMUX - tristate - config SND_MXC_SOC_WM1133_EV1 tristate "Audio on the i.MX31ADS with WM1133-EV1 fitted" depends on MACH_MX31ADS_WM1133_EV1 select SND_SOC_WM8350 select SND_SOC_IMX_PCM_FIQ - select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help Enable support for audio on the i.MX31ADS with the WM1133-EV1 @@ -141,8 +136,6 @@ config SND_SOC_MX27VIS_AIC32X4 tristate "SoC audio support for Visstrim M10 boards" depends on MACH_IMX27_VISSTRIM_M10 && I2C select SND_SOC_TLV320AIC32X4 - select SND_SOC_IMX_PCM_DMA - select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help Say Y if you want to add support for SoC audio on Visstrim SM10 @@ -154,7 +147,6 @@ config SND_SOC_PHYCORE_AC97 select SND_SOC_AC97_BUS select SND_SOC_WM9712 select SND_SOC_IMX_PCM_FIQ - select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help Say Y if you want to add support for SoC audio on Phytec phyCORE @@ -169,7 +161,6 @@ config SND_SOC_EUKREA_TLV320 depends on I2C select SND_SOC_TLV320AIC23 select SND_SOC_IMX_PCM_FIQ - select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI help Enable I2S based access to the TLV320AIC23B codec attached @@ -179,9 +170,6 @@ config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C select SND_SOC_WM8962 - select SND_SOC_IMX_PCM_DMA - select SND_SOC_IMX_AUDMUX - select SND_SOC_FSL_SSI help Say Y if you want to add support for SoC audio on an i.MX board with a wm8962 codec. @@ -190,18 +178,13 @@ config SND_SOC_IMX_SGTL5000 tristate "SoC Audio support for i.MX boards with sgtl5000" depends on OF && I2C select SND_SOC_SGTL5000 - select SND_SOC_IMX_PCM_DMA - select SND_SOC_IMX_AUDMUX - select SND_SOC_FSL_SSI help Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. config SND_SOC_IMX_SPDIF tristate "SoC Audio support for i.MX boards with S/PDIF" - select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_SPDIF - select REGMAP_MMIO help SoC Audio support for i.MX boards with S/PDIF Say Y if you want to add support for SoC audio on an i.MX board with @@ -211,8 +194,6 @@ config SND_SOC_IMX_MC13783 tristate "SoC Audio support for I.MX boards with mc13783" depends on MFD_MC13XXX && ARM select SND_SOC_IMX_SSI - select SND_SOC_IMX_AUDMUX select SND_SOC_MC13783 - select SND_SOC_IMX_PCM_DMA endif # SND_IMX_SOC diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index aaccbee..5474a0c 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -34,11 +34,10 @@ obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o # i.MX Platform Support snd-soc-imx-ssi-objs := imx-ssi.o snd-soc-imx-audmux-objs := imx-audmux.o +obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm-fiq.o obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o -obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o -obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm-fiq.o -obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += imx-pcm-dma.o +obj-$(CONFIG_SND_IMX_SOC) += imx-pcm-dma.o snd-soc-imx-audmux.o # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index c79cb27..a1364cb 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -38,14 +38,7 @@ struct imx_pcm_fiq_params { struct snd_dmaengine_dai_dma_data *dma_params_tx; }; -#if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_DMA) int imx_pcm_dma_init(struct platform_device *pdev); -#else -static inline int imx_pcm_dma_init(struct platform_device *pdev) -{ - return -ENODEV; -} -#endif #if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_FIQ) int imx_pcm_fiq_init(struct platform_device *pdev, -- 1.8.3.2

2 2

[alsa-devel] [PATCH] ALSA: PCM: Warn when buffer preallocation fails
by Takashi Iwai 10 Jan '14

10 Jan '14

The failures of buffer preallocations at driver initializations aren't critical but it's still helpful to inform, so that user can know that something doesn't work as expected. For example, the recent page allocator change triggered regressions, but developers didn't notice until recently because the driver didn't complain. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_memory.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 01f8eafebda6..54debc07f5cb 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -51,6 +51,7 @@ static const size_t snd_minimum_buffer = 16384; static int preallocate_pcm_pages(struct snd_pcm_substream *substream, size_t size) { struct snd_dma_buffer *dmab = &substream->dma_buffer; + size_t orig_size = size; int err; do { @@ -63,6 +64,10 @@ static int preallocate_pcm_pages(struct snd_pcm_substream *substream, size_t siz size >>= 1; } while (size >= snd_minimum_buffer); dmab->bytes = 0; /* tell error */ + pr_warn("ALSA pcmC%dD%d%c,%d:%s: cannot preallocate for size %zu\n", + substream->pcm->card->number, substream->pcm->device, + substream->stream ? 'c' : 'p', substream->number, + substream->pcm->name, orig_size); return 0; } -- 1.8.5.2

1 0

[alsa-devel] [PATCH v2 1/2] mfd: wm5110: Add registers for headphone short circuit control
by Charles Keepax 10 Jan '14

10 Jan '14

Add the registers necessary to enable/disable the headphone short circuit protection. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm5110-tables.c | 6 ++++++ include/linux/mfd/arizona/registers.h | 27 +++++++++++++++++++++++++++ 2 files changed, 33 insertions(+), 0 deletions(-) Changes since v1: - Add switch to the control names in the second patch in the series. Thanks, Charles diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index 24ec464..30f8163 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -637,6 +637,9 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000491, 0x0000 }, /* R1169 - PDM SPK1 CTRL 2 */ { 0x00000492, 0x0069 }, /* R1170 - PDM SPK2 CTRL 1 */ { 0x00000493, 0x0000 }, /* R1171 - PDM SPK2 CTRL 2 */ + { 0x000004A0, 0x3480 }, /* R1184 - HP1 Short Circuit Ctrl */ + { 0x000004A1, 0x3480 }, /* R1185 - HP2 Short Circuit Ctrl */ + { 0x000004A2, 0x3480 }, /* R1186 - HP3 Short Circuit Ctrl */ { 0x00000500, 0x000C }, /* R1280 - AIF1 BCLK Ctrl */ { 0x00000501, 0x0008 }, /* R1281 - AIF1 Tx Pin Ctrl */ { 0x00000502, 0x0000 }, /* R1282 - AIF1 Rx Pin Ctrl */ @@ -1633,6 +1636,9 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_PDM_SPK1_CTRL_2: case ARIZONA_PDM_SPK2_CTRL_1: case ARIZONA_PDM_SPK2_CTRL_2: + case ARIZONA_HP1_SHORT_CIRCUIT_CTRL: + case ARIZONA_HP2_SHORT_CIRCUIT_CTRL: + case ARIZONA_HP3_SHORT_CIRCUIT_CTRL: case ARIZONA_AIF1_BCLK_CTRL: case ARIZONA_AIF1_TX_PIN_CTRL: case ARIZONA_AIF1_RX_PIN_CTRL: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index b40960d..2d3ac50 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -226,6 +226,9 @@ #define ARIZONA_PDM_SPK1_CTRL_2 0x491 #define ARIZONA_PDM_SPK2_CTRL_1 0x492 #define ARIZONA_PDM_SPK2_CTRL_2 0x493 +#define ARIZONA_HP1_SHORT_CIRCUIT_CTRL 0x4A0 +#define ARIZONA_HP2_SHORT_CIRCUIT_CTRL 0x4A1 +#define ARIZONA_HP3_SHORT_CIRCUIT_CTRL 0x4A2 #define ARIZONA_SPK_CTRL_2 0x4B5 #define ARIZONA_SPK_CTRL_3 0x4B6 #define ARIZONA_DAC_COMP_1 0x4DC @@ -3297,6 +3300,30 @@ #define ARIZONA_SPK2_FMT_WIDTH 1 /* SPK2_FMT */ /* + * R1184 (0x4A0) - HP1 Short Circuit Ctrl + */ +#define ARIZONA_HP1_SC_ENA 0x1000 /* HP1_SC_ENA */ +#define ARIZONA_HP1_SC_ENA_MASK 0x1000 /* HP1_SC_ENA */ +#define ARIZONA_HP1_SC_ENA_SHIFT 12 /* HP1_SC_ENA */ +#define ARIZONA_HP1_SC_ENA_WIDTH 1 /* HP1_SC_ENA */ + +/* + * R1185 (0x4A1) - HP2 Short Circuit Ctrl + */ +#define ARIZONA_HP2_SC_ENA 0x1000 /* HP2_SC_ENA */ +#define ARIZONA_HP2_SC_ENA_MASK 0x1000 /* HP2_SC_ENA */ +#define ARIZONA_HP2_SC_ENA_SHIFT 12 /* HP2_SC_ENA */ +#define ARIZONA_HP2_SC_ENA_WIDTH 1 /* HP2_SC_ENA */ + +/* + * R1186 (0x4A2) - HP3 Short Circuit Ctrl + */ +#define ARIZONA_HP3_SC_ENA 0x1000 /* HP3_SC_ENA */ +#define ARIZONA_HP3_SC_ENA_MASK 0x1000 /* HP3_SC_ENA */ +#define ARIZONA_HP3_SC_ENA_SHIFT 12 /* HP3_SC_ENA */ +#define ARIZONA_HP3_SC_ENA_WIDTH 1 /* HP3_SC_ENA */ + +/* * R1244 (0x4DC) - DAC comp 1 */ #define ARIZONA_OUT_COMP_COEFF_MASK 0xFFFF /* OUT_COMP_COEFF - [15:0] */ -- 1.7.2.5

3 3

[alsa-devel] [PATCH v2 0/6] dma: Indicate residue granularity in dma_slave_caps
by Lars-Peter Clausen 10 Jan '14

10 Jan '14

Here is an updated version of the DMA residue granularity patchset. As discuessed last time it would be best if this could go through ASoC tree. It needs some feedback from the DMA folks first though. New in v2: * Dropped byte level granularity as discussed during v1 review * Added support for auto-detecting the NO_RESIDUE flag on the compat path in the generic ASoC dmaengine PCM driver * Updated a few ASoC drivers to take advantage of the new API Updated original cover text from v1 below: This patch series introduces a new field to the dma_slave_caps struct which exposes the granularity with which the reported residue is updated, updates the pl330 DMA driver to set this field and lets the generic dmaengine PCM audio driver check that field. The reason for introducing the field is that for audio we need to know the granularity with which the reported residue is updated. The uncertainty of the ALSA period pointer depends on the granularity of the reported residue. E.g. if residue is only updated once every period this means that the position in the buffer from/to which the hardware is currently reading/writing could be anywhere between the reported PCM pointer and the reported PCM pointer plus one period size (plus some extra margin for interrupt latency, etc). Certain algorithms (e.g. pulseaudio's timer based scheduling) only work correctly if the uncertainty of the PCM pointer is low enough. Exposing this information from the DMA driver makes it possible for applications to switch to alternative modes of operation which also work with a high granularity. The patch series introduces four levels of granularity, which I think will cover most DMA cores. I'd like some feedback though if anybody can think of a DMA controller for which none of the levels would be correct. The four levels are: * DESCRIPTOR: The DMA channel is only able to tell whether a descriptor has been completed or not, which means residue reporting is not supported by this channel. The residue field of the dma_tx_state field will always be 0. * SEGMENT: The DMA channel updates the residue field after each successfully completed segment of the transfer (For cyclic transfers this is after each period). This is typically implemented by having the hardware generate an interrupt after each transferred segment and then the driver updates the outstanding residue by the size of the segment. Another possibility is if the hardware supports SG and the segment descriptor has a field which gets set by the hardware after the segment has been completed. The driver then counts the number of segments which do not have the flag set to compute the residue. * BURST: The DMA channel updates the residue field after each transferred burst. This is typically only supported if the hardware has a progress register of some sort (E.g. a register with the current read/write address or a register with the amount of bursts/beats/bytes that have been transferred or still need to be transferred). - Lars Lars-Peter Clausen (6): dma: Indicate residue granularity in dma_slave_caps dma: pl330: Set residue_granularity ASoC: generic-dmaengine-pcm: Check NO_RESIDUE flag at runtime ASoC: generic-dmaengine-pcm: Check DMA residue granularity ASoC: axi-{spdif,i2s}: Drop NO_RESIDUE flag ASoC: samsung: Remove NO_RESIDUE flag drivers/dma/pl330.c | 1 + include/linux/dmaengine.h | 15 ++++++++ sound/soc/adi/axi-i2s.c | 3 +- sound/soc/adi/axi-spdif.c | 3 +- sound/soc/samsung/dmaengine.c | 1 - sound/soc/soc-generic-dmaengine-pcm.c | 66 ++++++++++++++++++++++------------- 6 files changed, 60 insertions(+), 29 deletions(-) -- 1.8.0

3 10

[alsa-devel] [PATCH] a52: switch to AV_CODEC_ID identifiers
by Anton Khirnov 10 Jan '14

10 Jan '14

Fixes build with latest libavcodec versions. --- a52/pcm_a52.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/a52/pcm_a52.c b/a52/pcm_a52.c index 359608f..b467ec8 100644 --- a/a52/pcm_a52.c +++ b/a52/pcm_a52.c @@ -58,6 +58,9 @@ #endif #endif +#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(54, 25, 0) +#define AV_CODEC_ID_AC3 CODEC_ID_AC3 +#endif struct a52_ctx { snd_pcm_ioplug_t io; @@ -916,7 +919,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(a52) if (rec->codec == NULL) rec->codec = avcodec_find_encoder_by_name("ac3"); if (rec->codec == NULL) - rec->codec = avcodec_find_encoder(CODEC_ID_AC3); + rec->codec = avcodec_find_encoder(AV_CODEC_ID_AC3); if (rec->codec == NULL) { SNDERR("Cannot find codec engine"); err = -EINVAL; -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ALSA: ctxfi - Work around emu20k1 glitch to prevent buffered sound data loss
by Sarah Bessmer 10 Jan '14

10 Jan '14

Occasionally, on playback stream ringbuffer wraparound, the EMU20K1 hardware will momentarily return 0 instead of the proper current(loop) address. This patch handles that case, fixing the problem of playback position corruption and subsequent loss of buffered sound data, that occurs with some common buffering layout patterns(e.g. multiple simultaneous output streams with differently-sized or non-power-of-2-sized buffers). An alternate means of fixing the problem would be to read the ca register continuously, until two sequential reads return the same value; however, that would be a more invasive change, has performance implications, and isn't necessary unless there are also issues with the value not being updated atomically in regards to individual bits or something similar(which I have not encountered through light testing). I have no EMU20K2 hardware to confirm if the issue is present there, but even if it's not, this change shouldn't break anything that's not already broken. Signed-off-by: Sarah Bessmer <aotos(a)fastmail.fm> Cc: Takashi Iwai <tiwai(a)suse.de> --- --- linux-3.10.25/sound/pci/ctxfi/ctatc.c.orig 2014-01-09 10:12:00.000000000 -0800 +++ linux-3.10.25/sound/pci/ctxfi/ctatc.c 2014-01-09 10:32:42.000000000 -0800 @@ -433,10 +433,15 @@ atc_pcm_playback_position(struct ct_atc if (!src) return 0; position = src->ops->get_ca(src); + if(position < apcm->vm_block->addr) { + snd_printdd("ctxfi: bad ca - ca=0x%08x, vba=0x%08x, vbs=0x%08x\n", position, apcm->vm_block->addr, apcm->vm_block->size); + position = apcm->vm_block->addr; + } + size = apcm->vm_block->size; max_cisz = src->multi * src->rsc.msr; max_cisz = 128 * (max_cisz < 8 ? max_cisz : 8); return (position + size - max_cisz - apcm->vm_block->addr) % size;

2 1

[alsa-devel] [PATCH] ASoC: simple-card: fix one bug to writing to the platform data
by Xiubo Li 10 Jan '14

10 Jan '14

It's a bug that writing to the platform data directly, for it should be constant. So just copy it before writing. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 40 +++++++++++++++++++++------------------- 1 file changed, 21 insertions(+), 19 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 406e4ea..5b65324 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -9,9 +9,10 @@ * published by the Free Software Foundation. */ #include <linux/clk.h> +#include <linux/module.h> #include <linux/of.h> #include <linux/platform_device.h> -#include <linux/module.h> +#include <linux/string.h> #include <sound/simple_card.h> #define asoc_simple_get_card_info(p) \ @@ -204,36 +205,37 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct device_node *of_cpu, *of_codec, *of_platform; struct device *dev = &pdev->dev; + int ret; cinfo = NULL; of_cpu = NULL; of_codec = NULL; of_platform = NULL; + + cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); + if (!cinfo) + return -ENOMEM; + if (np && of_device_is_available(np)) { - cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); - if (cinfo) { - int ret; - cinfo->snd_card.dev = &pdev->dev; - ret = asoc_simple_card_parse_of(np, cinfo, dev, - &of_cpu, - &of_codec, - &of_platform); - if (ret < 0) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "parse error %d\n", ret); - return ret; - } - } else { - return -ENOMEM; + cinfo->snd_card.dev = dev; + + ret = asoc_simple_card_parse_of(np, cinfo, dev, + &of_cpu, + &of_codec, + &of_platform); + if (ret < 0) { + if (ret != -EPROBE_DEFER) + dev_err(dev, "parse error %d\n", ret); + return ret; } } else { - cinfo = pdev->dev.platform_data; - if (!cinfo) { + if (!pdev->dev.platform_data) { dev_err(dev, "no info for asoc-simple-card\n"); return -EINVAL; } - cinfo->snd_card.dev = &pdev->dev; + memcpy(cinfo, pdev->dev.platform_data, sizeof(cinfo)); + cinfo->snd_card.dev = dev; } if (!cinfo->name || -- 1.8.4

1 0

[alsa-devel] [PATCH v2] ASoC: fsl_esai: Add ESAI CPU DAI driver
by Nicolin Chen 09 Jan '14

09 Jan '14

This patch implements a device-tree-only CPU DAI driver for Freescale ESAI controller that supports: - 12 channels playback and 8 channels record. [ Some of the inner transmitters and receivers are sharing same group of pins. So the maxmium 12 output or 8 input channels are only valid if there is no pin conflict occurring to it. ] - Independent (asynchronous mode) or shared (synchronous mode) transmit and receive sections with separate or shared internal/external clocks and frame syncs, operating in Master or Slave mode. [ Current ALSA seems not to allow CPU DAI drivers to set DAI format and clks separately for PLAYBACK and CAPTURE. So this first version only supports the case that uses the same DAI format for both directions. ] - Various DAI formats: I2S, Left-Justified, Right-Justified, DSP-A and DSP-B. - Programmable word length (8, 16, 20 or 24bits) - Flexible selection between system clock or external oscillator as input clock source, programmable internal clock divider and frame sync generation. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v2: * Correct multi-line comments. * Add missing spaces between number and operator. .../devicetree/bindings/sound/fsl,esai.txt | 54 ++ sound/soc/fsl/Kconfig | 3 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_esai.c | 815 +++++++++++++++++++++ sound/soc/fsl/fsl_esai.h | 352 +++++++++ 5 files changed, 1226 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,esai.txt create mode 100644 sound/soc/fsl/fsl_esai.c create mode 100644 sound/soc/fsl/fsl_esai.h diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt new file mode 100644 index 0000000..4372833 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt @@ -0,0 +1,54 @@ +Freescale Enhanced Serial Audio Interface (ESAI) Controller + +The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port +for serial communication with a variety of serial devices, including industry +standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and +other DSPs. It has up to six transmitters and four receivers. + +Required properties: + + - compatible : Compatible list, must contain "fsl,imx35-esai". + + - reg : Offset and length of the register set for the device. + + - interrupts : Contains the spdif interrupt. + + - dmas : Generic dma devicetree binding as described in + Documentation/devicetree/bindings/dma/dma.txt. + + - dma-names : Two dmas have to be defined, "tx" and "rx". + + - clocks: Contains an entry for each entry in clock-names. + + - clock-names : Includes the following entries: + "core" The core clock used to access registers + "extal" The esai baud clock for esai controller used to derive + HCK, SCK and FS. + "fsys" The system clock derived from ahb clock used to derive + HCK, SCK and FS. + + - fsl,fifo-depth: The number of elements in the transmit and receive FIFOs. + This number is the maximum allowed value for TFCR[TFWM] or RFCR[RFWM]. + + - fsl,esai-synchronous: This is a boolean property. If present, indicating + that ESAI would work in the synchronous mode, which means all the settings + for Receiving would be duplicated from Transmition related registers. + + - fsl,esai-network-mode: This is a boolean property. If present, indicating + that ESAI controller would work in its network mode. + +Example: + +esai: esai@02024000 { + compatible = "fsl,imx35-esai"; + reg = <0x02024000 0x4000>; + interrupts = <0 51 0x04>; + clocks = <&clks 208>, <&clks 118>, <&clks 208>; + clock-names = "core", "extal", "fsys"; + dmas = <&sdma 23 21 0>, <&sdma 24 21 0>; + dma-names = "rx", "tx"; + fsl,fifo-depth = <128>; + fsl,esai-synchronous; + fsl,esai-network-mode; + status = "disabled"; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 514c275..324988d 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -8,6 +8,9 @@ config SND_SOC_FSL_SSI config SND_SOC_FSL_SPDIF tristate +config SND_SOC_FSL_ESAI + tristate + config SND_SOC_FSL_UTILS tristate diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index aaccbee..b12ad4b 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -14,11 +14,13 @@ obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o snd-soc-fsl-sai-objs := fsl_sai.o snd-soc-fsl-ssi-objs := fsl_ssi.o snd-soc-fsl-spdif-objs := fsl_spdif.o +snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o +obj-$(CONFIG_SND_SOC_FSL_ESAI) += snd-soc-fsl-esai.o obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c new file mode 100644 index 0000000..a30b1fd --- /dev/null +++ b/sound/soc/fsl/fsl_esai.c @@ -0,0 +1,815 @@ +/* + * Freescale ESAI ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/dmaengine.h> +#include <linux/module.h> +#include <linux/of_irq.h> +#include <linux/of_platform.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> + +#include "fsl_esai.h" +#include "imx-pcm.h" + +#define FSL_ESAI_RATES SNDRV_PCM_RATE_8000_192000 +#define FSL_ESAI_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +/** + * fsl_esai: ESAI private data + * + * @dma_params_rx: DMA parameters for receive channel + * @dma_params_tx: DMA parameters for transmit channel + * @pdev: platform device pointer + * @regmap: regmap handler + * @coreclk: clock source to access register + * @extalclk: esai clock source to derive HCK, SCK and FS + * @fsysclk: system clock source to derive HCK, SCK and FS + * @fifo_depth: depth of tx/rx FIFO + * @sck_div: if using PSR/PM dividers for SCK clock + * @network_mode: if using network mode + * @synchronous: if using tx/rx synchronous mode + * @name: driver name + */ +struct fsl_esai { + struct snd_dmaengine_dai_dma_data dma_params_rx; + struct snd_dmaengine_dai_dma_data dma_params_tx; + struct platform_device *pdev; + struct regmap *regmap; + struct clk *coreclk; + struct clk *extalclk; + struct clk *fsysclk; + u32 fifo_depth; + u32 slot_width; + bool sck_div; + bool network_mode; + bool synchronous; + char name[32]; +}; + +static irqreturn_t esai_isr(int irq, void *devid) +{ + struct fsl_esai *esai_priv = (struct fsl_esai *)devid; + struct platform_device *pdev = esai_priv->pdev; + u32 esr; + + regmap_read(esai_priv->regmap, REG_ESAI_ESR, &esr); + + if (esr & ESAI_ESR_TINIT_MASK) + dev_dbg(&pdev->dev, "isr: Transmition Initialized\n"); + + if (esr & ESAI_ESR_RFF_MASK) + dev_warn(&pdev->dev, "isr: Receiving overrun\n"); + + if (esr & ESAI_ESR_TFE_MASK) + dev_warn(&pdev->dev, "isr: Transmition underrun\n"); + + if (esr & ESAI_ESR_TLS_MASK) + dev_dbg(&pdev->dev, "isr: Just transmitted the last slot\n"); + + if (esr & ESAI_ESR_TDE_MASK) + dev_dbg(&pdev->dev, "isr: Transmition data exception\n"); + + if (esr & ESAI_ESR_TED_MASK) + dev_dbg(&pdev->dev, "isr: Transmitting even slots\n"); + + if (esr & ESAI_ESR_TD_MASK) + dev_dbg(&pdev->dev, "isr: Transmitting data\n"); + + if (esr & ESAI_ESR_RLS_MASK) + dev_dbg(&pdev->dev, "isr: Just received the last slot\n"); + + if (esr & ESAI_ESR_RDE_MASK) + dev_dbg(&pdev->dev, "isr: Receiving data exception\n"); + + if (esr & ESAI_ESR_RED_MASK) + dev_dbg(&pdev->dev, "isr: Receiving even slots\n"); + + if (esr & ESAI_ESR_RD_MASK) + dev_dbg(&pdev->dev, "isr: Receiving data\n"); + + return IRQ_HANDLED; +} + +/** + * This function is used to calculate the divisors of psr, pm, fp and it is + * supposed to be called in set_dai_sysclk() and set_bclk_ratio(). + * + * @ratio: desired overall ratio for the paticipating dividers + * @usefp: for HCK setting, there is no need to set fp divider + * @fp: bypass other dividers by setting fp directly if fp != 0 + */ +static int fsl_esai_divisor_cal(struct snd_soc_dai *dai, u32 ratio, + bool usefp, u32 fp) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + u32 psr, pm = 999, maxfp, prod, sub, savesub, i, j; + + maxfp = usefp ? 16 : 1; + + if (usefp && fp) + goto out_fp; + + if (ratio > 2 * 8 * 256 * maxfp || ratio < 2) { + dev_err(dai->dev, "the ratio is out of range (2 ~ %d)\n", + 2 * 8 * 256 * maxfp); + return -EINVAL; + } else if (ratio % 2) { + dev_err(dai->dev, "the raio must be even if using upper divider\n"); + return -EINVAL; + } + + ratio /= 2; + + psr = ratio <= 256 * maxfp ? ESAI_xCCR_xPSR_BYPASS : ESAI_xCCR_xPSR_DIV8; + + /* Set the max fluctuation -- 0.1% of the max devisor */ + savesub = (psr ? 1 : 8) * 256 * maxfp / 1000; + + /* Find the best value for PM */ + for (i = 1; i <= 256; i++) { + for (j = 1; j <= maxfp; j++) { + /* PSR (1 or 8) * PM (1 ~ 256) * FP (1 ~ 16) */ + prod = (psr ? 1 : 8) * i * j; + + if (prod == ratio) + sub = 0; + else if (prod / ratio == 1) + sub = prod - ratio; + else if (ratio / prod == 1) + sub = ratio - prod; + else + continue; + + /* Calculate the fraction */ + sub = sub * 1000 / ratio; + if (sub < savesub) { + savesub = sub; + pm = i; + fp = j; + } + + /* We are lucky */ + if (savesub == 0) + goto out; + } + } + + if (pm == 999) { + dev_err(dai->dev, "failed to calculate proper divisors\n"); + return -EINVAL; + } + +out: + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, + ESAI_xCCR_xPSR_MASK | ESAI_xCCR_xPM_MASK, + psr | ESAI_xCCR_xPM(pm)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, + ESAI_xCCR_xPSR_MASK | ESAI_xCCR_xPM_MASK, + psr | ESAI_xCCR_xPM(pm)); + +out_fp: + /* Bypass fp if not being required */ + if (maxfp <= 1) + return 0; + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, + ESAI_xCCR_xFP_MASK, ESAI_xCCR_xFP(fp)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, + ESAI_xCCR_xFP_MASK, ESAI_xCCR_xFP(fp)); + + return 0; +} + +/** + * This function mainly configures the clock frequency of MCLK (HCK) + * + * @Parameters: + * clk_id: The clock source of HCK + * (Input from outside; output from inside, FSYS or EXTAL) + * freq: The required clock rate of HCK + * dir: The clock direction of HCK + * + * Note: If the direction is input, we do not care about clk_id. + */ +static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + bool in = dir == SND_SOC_CLOCK_IN; + u32 ret, ratio, ecr = 0; + unsigned long clk_rate; + + /* sck_div can be only bypassed if ETO/ERO=0 and SNC_SOC_CLOCK_OUT */ + esai_priv->sck_div = true; + + /* Set the direction of HCK pins */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, + ESAI_xCCR_xHCKD, in ? 0 : ESAI_xCCR_xHCKD); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, + ESAI_xCCR_xHCKD, in ? 0 : ESAI_xCCR_xHCKD); + + if (in) + goto out; + + switch (clk_id) { + case ESAI_HCK_FSYS: + if (IS_ERR(esai_priv->fsysclk)) { + dev_err(dai->dev, "no assigned fsys clock\n"); + return -EINVAL; + } + clk_rate = clk_get_rate(esai_priv->fsysclk); + break; + case ESAI_HCK_EXTAL: + if (IS_ERR(esai_priv->extalclk)) { + dev_err(dai->dev, "no assigned extal clock\n"); + return -EINVAL; + } + clk_rate = clk_get_rate(esai_priv->extalclk); + + ecr |= ESAI_ECR_ETI | ESAI_ECR_ERI; + break; + default: + return -EINVAL; + } + + ratio = clk_rate / freq; + if (ratio * freq > clk_rate) + ret = ratio * freq - clk_rate; + else if (ratio * freq < clk_rate) + ret = clk_rate - ratio * freq; + else + ret = 0; + + /* Block if clock source can not be divided into the required rate */ + if (ret != 0 && clk_rate / ret < 1000) { + dev_err(dai->dev, "failed to derive the required frequency\n"); + return -EINVAL; + } + + if (ratio == 1) { + /* Bypass all the dividers if not being needed */ + ecr |= ESAI_ECR_ETO | ESAI_ECR_ERO; + goto out; + } + + ret = fsl_esai_divisor_cal(dai, ratio, false, 0); + if (ret) + return ret; + + esai_priv->sck_div = false; + +out: + regmap_update_bits(esai_priv->regmap, REG_ESAI_ECR, + ESAI_ECR_ETI | ESAI_ECR_ETO | + ESAI_ECR_ERI | ESAI_ECR_ERO, ecr); + + return 0; +} + +/** + * This function configures the ratio between MCLK (HCK) and BCLK (SCK) + * (For DAI Master Mode only) + * + * Note: Machine driver should calculate the ratio to call this function. + * Only effective after calling set_dai_sysclk() to set HCK direction. + */ +static int fsl_esai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + u32 fp = esai_priv->sck_div ? 0 : ratio; + + if (!esai_priv->sck_div && (ratio > 16 || ratio == 0)) { + dev_err(dai->dev, "the ratio is out of range (1 ~ 16)\n"); + return -EINVAL; + } + + return fsl_esai_divisor_cal(dai, ratio, true, fp); +} + +/** + * This function provides a manual way to configure all the dividers directly + * (For DAI Master Mode only) + * + * By using this function, the driver would allow user to configure a flexible + * combination of divisors for both TX and RX route separately + * + * Note: Only effective after calling set_dai_sysclk() to set HCK direction + */ +static int fsl_esai_set_dai_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + bool tx = div_id <= ESAI_TX_DIV_FP; + u32 mask, val; + + switch (div_id) { + case ESAI_TX_DIV_PSR: + case ESAI_RX_DIV_PSR: + mask = ESAI_xCCR_xPSR_MASK; + val = div ? ESAI_xCCR_xPSR_BYPASS : 0; + break; + case ESAI_TX_DIV_PM: + case ESAI_RX_DIV_PM: + mask = ESAI_xCCR_xPM_MASK; + val = ESAI_xCCR_xPM(div); + break; + case ESAI_TX_DIV_FP: + case ESAI_RX_DIV_FP: + mask = ESAI_xCCR_xFP_MASK; + val = ESAI_xCCR_xFP(div); + break; + default: + return -EINVAL; + } + + return regmap_update_bits(esai_priv->regmap, REG_ESAI_xCCR(tx), mask, val); +} + +static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, + u32 rx_mask, int slots, int slot_width) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, + ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, + ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(tx_mask)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, + ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, + ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(rx_mask)); + + esai_priv->slot_width = slot_width; + + return 0; +} + +static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + u32 xcr = 0, xccr = 0, mask; + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* Data on rising edge of bclk, frame low, 1clk before data */ + xcr |= ESAI_xCR_xFSR; + xccr |= ESAI_xCCR_xFSP | ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* Data on rising edge of bclk, frame high */ + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + /* Data on rising edge of bclk, frame high, right aligned */ + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCR_xWA; + break; + case SND_SOC_DAIFMT_DSP_A: + /* Data on rising edge of bclk, frame high, 1clk before data */ + xcr |= ESAI_xCR_xFSL | ESAI_xCR_xFSR; + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_DSP_B: + /* Data on rising edge of bclk, frame high */ + xcr |= ESAI_xCR_xFSL; + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Nothing to do for both normal cases */ + break; + case SND_SOC_DAIFMT_IB_NF: + /* Invert bit clock */ + xccr ^= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + break; + case SND_SOC_DAIFMT_NB_IF: + /* Invert frame clock */ + xccr ^= ESAI_xCCR_xFSP; + break; + case SND_SOC_DAIFMT_IB_IF: + /* Invert both clocks */ + xccr ^= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP; + break; + default: + return -EINVAL; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFM: + xccr |= ESAI_xCCR_xCKD; + break; + case SND_SOC_DAIFMT_CBM_CFS: + xccr |= ESAI_xCCR_xFSD; + break; + case SND_SOC_DAIFMT_CBS_CFS: + xccr |= ESAI_xCCR_xFSD | ESAI_xCCR_xCKD; + break; + default: + return -EINVAL; + } + + mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR; + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, xcr); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, mask, xcr); + + mask = ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP | + ESAI_xCCR_xFSD | ESAI_xCCR_xCKD | ESAI_xCR_xWA; + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, mask, xccr); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, mask, xccr); + + return 0; +} + +static int fsl_esai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + /* + * Some platforms might use the same bit to gate all three or two of + * clocks, so keep all clocks open/close at the same time for safety + */ + clk_prepare_enable(esai_priv->coreclk); + if (!IS_ERR(esai_priv->extalclk)) + clk_prepare_enable(esai_priv->extalclk); + if (!IS_ERR(esai_priv->fsysclk)) + clk_prepare_enable(esai_priv->fsysclk); + + if (!dai->active) { + /* Reset Port C */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC, + ESAI_PRRC_PDC_MASK, ESAI_PRRC_PDC(ESAI_GPIO)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC, + ESAI_PCRC_PC_MASK, ESAI_PCRC_PC(ESAI_GPIO)); + + /* Set synchronous mode */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_SAICR, + ESAI_SAICR_SYNC, esai_priv->synchronous ? + ESAI_SAICR_SYNC : 0); + + /* Set network mode */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_xMOD_MASK, esai_priv->network_mode ? + ESAI_xCR_xMOD_NETWORK : 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, + ESAI_xCR_xMOD_MASK, esai_priv->network_mode ? + ESAI_xCR_xMOD_NETWORK : 0); + + /* Set a default slot number -- 2 */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, + ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + } + + return 0; +} + +static int fsl_esai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 width = snd_pcm_format_width(params_format(params)); + u32 channels = params_channels(params); + u32 mask, val; + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFR_MASK, ESAI_xFCR_xFR); + + mask = ESAI_xFCR_xFR_MASK | ESAI_xFCR_xWA_MASK | ESAI_xFCR_xFWM_MASK | + (tx ? ESAI_xFCR_TE_MASK | ESAI_xFCR_TIEN : ESAI_xFCR_RE_MASK); + val = ESAI_xFCR_xWA(width) | ESAI_xFCR_xFWM(esai_priv->fifo_depth / 2) | + (tx ? ESAI_xFCR_TE(channels) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(channels)); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val); + + mask = ESAI_xCR_xSWS_MASK | (tx ? ESAI_xCR_PADC : 0); + val = ESAI_xCR_xSWS(esai_priv->slot_width, width) | (tx ? ESAI_xCR_PADC : 0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val); + + return 0; +} + +static void fsl_esai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + if (!IS_ERR(esai_priv->fsysclk)) + clk_disable_unprepare(esai_priv->fsysclk); + if (!IS_ERR(esai_priv->extalclk)) + clk_disable_unprepare(esai_priv->extalclk); + clk_disable_unprepare(esai_priv->coreclk); +} + +static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u8 i, channels = substream->runtime->channels; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFEN_MASK, ESAI_xFCR_xFEN); + + /* Write initial words reqiured by ESAI as normal procedure */ + for (i = 0; tx && i < channels; i++) + regmap_write(esai_priv->regmap, REG_ESAI_ETDR, 0x0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), + tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, + tx ? ESAI_xCR_TE(channels) : ESAI_xCR_RE(channels)); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), + tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0); + + /* Disable and reset FIFO */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFR | ESAI_xFCR_xFEN, ESAI_xFCR_xFR); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), + ESAI_xFCR_xFR, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_dai_ops fsl_esai_dai_ops = { + .startup = fsl_esai_startup, + .shutdown = fsl_esai_shutdown, + .trigger = fsl_esai_trigger, + .hw_params = fsl_esai_hw_params, + .set_sysclk = fsl_esai_set_dai_sysclk, + .set_bclk_ratio = fsl_esai_set_bclk_ratio, + .set_clkdiv = fsl_esai_set_dai_clkdiv, + .set_fmt = fsl_esai_set_dai_fmt, + .set_tdm_slot = fsl_esai_set_dai_tdm_slot, +}; + +static int fsl_esai_dai_probe(struct snd_soc_dai *dai) +{ + struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &esai_priv->dma_params_tx, + &esai_priv->dma_params_rx); + + return 0; +} + +static struct snd_soc_dai_driver fsl_esai_dai = { + .probe = fsl_esai_dai_probe, + .playback = { + .channels_min = 1, + .channels_max = 12, + .rates = FSL_ESAI_RATES, + .formats = FSL_ESAI_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = FSL_ESAI_RATES, + .formats = FSL_ESAI_FORMATS, + }, + .ops = &fsl_esai_dai_ops, +}; + +static const struct snd_soc_component_driver fsl_esai_component = { + .name = "fsl-esai", +}; + +static bool fsl_esai_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ESAI_ERDR: + case REG_ESAI_ECR: + case REG_ESAI_ESR: + case REG_ESAI_TFCR: + case REG_ESAI_TFSR: + case REG_ESAI_RFCR: + case REG_ESAI_RFSR: + case REG_ESAI_RX0: + case REG_ESAI_RX1: + case REG_ESAI_RX2: + case REG_ESAI_RX3: + case REG_ESAI_SAISR: + case REG_ESAI_SAICR: + case REG_ESAI_TCR: + case REG_ESAI_TCCR: + case REG_ESAI_RCR: + case REG_ESAI_RCCR: + case REG_ESAI_TSMA: + case REG_ESAI_TSMB: + case REG_ESAI_RSMA: + case REG_ESAI_RSMB: + case REG_ESAI_PRRC: + case REG_ESAI_PCRC: + return true; + default: + return false; + } +} + +static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case REG_ESAI_ETDR: + case REG_ESAI_ECR: + case REG_ESAI_TFCR: + case REG_ESAI_RFCR: + case REG_ESAI_TX0: + case REG_ESAI_TX1: + case REG_ESAI_TX2: + case REG_ESAI_TX3: + case REG_ESAI_TX4: + case REG_ESAI_TX5: + case REG_ESAI_TSR: + case REG_ESAI_SAICR: + case REG_ESAI_TCR: + case REG_ESAI_TCCR: + case REG_ESAI_RCR: + case REG_ESAI_RCCR: + case REG_ESAI_TSMA: + case REG_ESAI_TSMB: + case REG_ESAI_RSMA: + case REG_ESAI_RSMB: + case REG_ESAI_PRRC: + case REG_ESAI_PCRC: + return true; + default: + return false; + } +} + +static const struct regmap_config fsl_esai_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + + .max_register = REG_ESAI_PCRC, + .readable_reg = fsl_esai_readable_reg, + .writeable_reg = fsl_esai_writeable_reg, +}; + +static int fsl_esai_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct fsl_esai *esai_priv; + struct resource *res; + const uint32_t *iprop; + void __iomem *regs; + int irq, ret; + + esai_priv = devm_kzalloc(&pdev->dev, sizeof(*esai_priv), GFP_KERNEL); + if (!esai_priv) + return -ENOMEM; + + esai_priv->pdev = pdev; + strcpy(esai_priv->name, np->name); + + /* Get the addresses and IRQ */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + esai_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, + "core", regs, &fsl_esai_regmap_config); + if (IS_ERR(esai_priv->regmap)) { + dev_err(&pdev->dev, "failed to init regmap: %ld\n", + PTR_ERR(esai_priv->regmap)); + return PTR_ERR(esai_priv->regmap); + } + + esai_priv->coreclk = devm_clk_get(&pdev->dev, "core"); + if (IS_ERR(esai_priv->coreclk)) { + dev_err(&pdev->dev, "failed to get core clock: %ld\n", + PTR_ERR(esai_priv->coreclk)); + return PTR_ERR(esai_priv->coreclk); + } + + esai_priv->extalclk = devm_clk_get(&pdev->dev, "extal"); + if (IS_ERR(esai_priv->extalclk)) + dev_warn(&pdev->dev, "failed to get extal clock: %ld\n", + PTR_ERR(esai_priv->extalclk)); + + esai_priv->fsysclk = devm_clk_get(&pdev->dev, "fsys"); + if (IS_ERR(esai_priv->fsysclk)) + dev_warn(&pdev->dev, "failed to get fsys clock: %ld\n", + PTR_ERR(esai_priv->fsysclk)); + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + return irq; + } + + ret = devm_request_irq(&pdev->dev, irq, esai_isr, 0, + esai_priv->name, esai_priv); + if (ret) { + dev_err(&pdev->dev, "failed to claim irq %u\n", irq); + return ret; + } + + /* Set a default slot size */ + esai_priv->slot_width = 32; + + /* Determine the FIFO depth */ + iprop = of_get_property(np, "fsl,fifo-depth", NULL); + if (iprop) + esai_priv->fifo_depth = be32_to_cpup(iprop); + else + esai_priv->fifo_depth = 64; + + esai_priv->dma_params_tx.maxburst = esai_priv->fifo_depth > 1; + esai_priv->dma_params_rx.maxburst = esai_priv->fifo_depth > 1; + esai_priv->dma_params_tx.addr = res->start + REG_ESAI_ETDR; + esai_priv->dma_params_rx.addr = res->start + REG_ESAI_ERDR; + + esai_priv->synchronous = + of_property_read_bool(np, "fsl,esai-synchronous"); + esai_priv->network_mode = + of_property_read_bool(np, "fsl,esai-network-mode"); + + dev_set_drvdata(&pdev->dev, esai_priv); + + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component, + &fsl_esai_dai, 1); + if (ret) { + dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); + return ret; + } + + ret = imx_pcm_dma_init(pdev); + if (ret) + dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); + + /* Reset ESAI unit */ + regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ERST); + + /* + * We need to enable ESAI so as to access some of its registers. + * Otherwise, we would fail to dump regmap from user space. + */ + regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ESAIEN); + + return ret; +} + +static const struct of_device_id fsl_esai_dt_ids[] = { + { .compatible = "fsl,imx35-esai", }, + {} +}; +MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids); + +static struct platform_driver fsl_esai_driver = { + .probe = fsl_esai_probe, + .driver = { + .name = "fsl-esai-dai", + .owner = THIS_MODULE, + .of_match_table = fsl_esai_dt_ids, + }, +}; + +module_platform_driver(fsl_esai_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale ESAI CPU DAI driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:fsl-esai-dai"); diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h new file mode 100644 index 0000000..680aac0 --- /dev/null +++ b/sound/soc/fsl/fsl_esai.h @@ -0,0 +1,352 @@ +/* + * fsl_esai.h - ALSA ESAI interface for the Freescale i.MX SoC + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen(a)freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _FSL_ESAI_DAI_H +#define _FSL_ESAI_DAI_H + +/* ESAI Register Map */ +#define REG_ESAI_ETDR 0x00 +#define REG_ESAI_ERDR 0x04 +#define REG_ESAI_ECR 0x08 +#define REG_ESAI_ESR 0x0C +#define REG_ESAI_TFCR 0x10 +#define REG_ESAI_TFSR 0x14 +#define REG_ESAI_RFCR 0x18 +#define REG_ESAI_RFSR 0x1C +#define REG_ESAI_xFCR(tx) (tx ? REG_ESAI_TFCR : REG_ESAI_RFCR) +#define REG_ESAI_xFSR(tx) (tx ? REG_ESAI_TFSR : REG_ESAI_RFSR) +#define REG_ESAI_TX0 0x80 +#define REG_ESAI_TX1 0x84 +#define REG_ESAI_TX2 0x88 +#define REG_ESAI_TX3 0x8C +#define REG_ESAI_TX4 0x90 +#define REG_ESAI_TX5 0x94 +#define REG_ESAI_TSR 0x98 +#define REG_ESAI_RX0 0xA0 +#define REG_ESAI_RX1 0xA4 +#define REG_ESAI_RX2 0xA8 +#define REG_ESAI_RX3 0xAC +#define REG_ESAI_SAISR 0xCC +#define REG_ESAI_SAICR 0xD0 +#define REG_ESAI_TCR 0xD4 +#define REG_ESAI_TCCR 0xD8 +#define REG_ESAI_RCR 0xDC +#define REG_ESAI_RCCR 0xE0 +#define REG_ESAI_xCR(tx) (tx ? REG_ESAI_TCR : REG_ESAI_RCR) +#define REG_ESAI_xCCR(tx) (tx ? REG_ESAI_TCCR : REG_ESAI_RCCR) +#define REG_ESAI_TSMA 0xE4 +#define REG_ESAI_TSMB 0xE8 +#define REG_ESAI_RSMA 0xEC +#define REG_ESAI_RSMB 0xF0 +#define REG_ESAI_xSMA(tx) (tx ? REG_ESAI_TSMA : REG_ESAI_RSMA) +#define REG_ESAI_xSMB(tx) (tx ? REG_ESAI_TSMB : REG_ESAI_RSMB) +#define REG_ESAI_PRRC 0xF8 +#define REG_ESAI_PCRC 0xFC + +/* ESAI Control Register -- REG_ESAI_ECR 0x8 */ +#define ESAI_ECR_ETI_SHIFT 19 +#define ESAI_ECR_ETI_MASK (1 << ESAI_ECR_ETI_SHIFT) +#define ESAI_ECR_ETI (1 << ESAI_ECR_ETI_SHIFT) +#define ESAI_ECR_ETO_SHIFT 18 +#define ESAI_ECR_ETO_MASK (1 << ESAI_ECR_ETO_SHIFT) +#define ESAI_ECR_ETO (1 << ESAI_ECR_ETO_SHIFT) +#define ESAI_ECR_ERI_SHIFT 17 +#define ESAI_ECR_ERI_MASK (1 << ESAI_ECR_ERI_SHIFT) +#define ESAI_ECR_ERI (1 << ESAI_ECR_ERI_SHIFT) +#define ESAI_ECR_ERO_SHIFT 16 +#define ESAI_ECR_ERO_MASK (1 << ESAI_ECR_ERO_SHIFT) +#define ESAI_ECR_ERO (1 << ESAI_ECR_ERO_SHIFT) +#define ESAI_ECR_ERST_SHIFT 1 +#define ESAI_ECR_ERST_MASK (1 << ESAI_ECR_ERST_SHIFT) +#define ESAI_ECR_ERST (1 << ESAI_ECR_ERST_SHIFT) +#define ESAI_ECR_ESAIEN_SHIFT 0 +#define ESAI_ECR_ESAIEN_MASK (1 << ESAI_ECR_ESAIEN_SHIFT) +#define ESAI_ECR_ESAIEN (1 << ESAI_ECR_ESAIEN_SHIFT) + +/* ESAI Status Register -- REG_ESAI_ESR 0xC */ +#define ESAI_ESR_TINIT_SHIFT 10 +#define ESAI_ESR_TINIT_MASK (1 << ESAI_ESR_TINIT_SHIFT) +#define ESAI_ESR_TINIT (1 << ESAI_ESR_TINIT_SHIFT) +#define ESAI_ESR_RFF_SHIFT 9 +#define ESAI_ESR_RFF_MASK (1 << ESAI_ESR_RFF_SHIFT) +#define ESAI_ESR_RFF (1 << ESAI_ESR_RFF_SHIFT) +#define ESAI_ESR_TFE_SHIFT 8 +#define ESAI_ESR_TFE_MASK (1 << ESAI_ESR_TFE_SHIFT) +#define ESAI_ESR_TFE (1 << ESAI_ESR_TFE_SHIFT) +#define ESAI_ESR_TLS_SHIFT 7 +#define ESAI_ESR_TLS_MASK (1 << ESAI_ESR_TLS_SHIFT) +#define ESAI_ESR_TLS (1 << ESAI_ESR_TLS_SHIFT) +#define ESAI_ESR_TDE_SHIFT 6 +#define ESAI_ESR_TDE_MASK (1 << ESAI_ESR_TDE_SHIFT) +#define ESAI_ESR_TDE (1 << ESAI_ESR_TDE_SHIFT) +#define ESAI_ESR_TED_SHIFT 5 +#define ESAI_ESR_TED_MASK (1 << ESAI_ESR_TED_SHIFT) +#define ESAI_ESR_TED (1 << ESAI_ESR_TED_SHIFT) +#define ESAI_ESR_TD_SHIFT 4 +#define ESAI_ESR_TD_MASK (1 << ESAI_ESR_TD_SHIFT) +#define ESAI_ESR_TD (1 << ESAI_ESR_TD_SHIFT) +#define ESAI_ESR_RLS_SHIFT 3 +#define ESAI_ESR_RLS_MASK (1 << ESAI_ESR_RLS_SHIFT) +#define ESAI_ESR_RLS (1 << ESAI_ESR_RLS_SHIFT) +#define ESAI_ESR_RDE_SHIFT 2 +#define ESAI_ESR_RDE_MASK (1 << ESAI_ESR_RDE_SHIFT) +#define ESAI_ESR_RDE (1 << ESAI_ESR_RDE_SHIFT) +#define ESAI_ESR_RED_SHIFT 1 +#define ESAI_ESR_RED_MASK (1 << ESAI_ESR_RED_SHIFT) +#define ESAI_ESR_RED (1 << ESAI_ESR_RED_SHIFT) +#define ESAI_ESR_RD_SHIFT 0 +#define ESAI_ESR_RD_MASK (1 << ESAI_ESR_RD_SHIFT) +#define ESAI_ESR_RD (1 << ESAI_ESR_RD_SHIFT) + +/* + * Transmit FIFO Configuration Register -- REG_ESAI_TFCR 0x10 + * Receive FIFO Configuration Register -- REG_ESAI_RFCR 0x18 + */ +#define ESAI_xFCR_TIEN_SHIFT 19 +#define ESAI_xFCR_TIEN_MASK (1 << ESAI_xFCR_TIEN_SHIFT) +#define ESAI_xFCR_TIEN (1 << ESAI_xFCR_TIEN_SHIFT) +#define ESAI_xFCR_REXT_SHIFT 19 +#define ESAI_xFCR_REXT_MASK (1 << ESAI_xFCR_REXT_SHIFT) +#define ESAI_xFCR_REXT (1 << ESAI_xFCR_REXT_SHIFT) +#define ESAI_xFCR_xWA_SHIFT 16 +#define ESAI_xFCR_xWA_WIDTH 3 +#define ESAI_xFCR_xWA_MASK (((1 << ESAI_xFCR_xWA_WIDTH) - 1) << ESAI_xFCR_xWA_SHIFT) +#define ESAI_xFCR_xWA(v) (((8 - (v >> 2)) << ESAI_xFCR_xWA_SHIFT) & ESAI_xFCR_xWA_MASK) +#define ESAI_xFCR_xFWM_SHIFT 8 +#define ESAI_xFCR_xFWM_WIDTH 8 +#define ESAI_xFCR_xFWM_MASK (((1 << ESAI_xFCR_xFWM_WIDTH) - 1) << ESAI_xFCR_xFWM_SHIFT) +#define ESAI_xFCR_xFWM(v) (((v - 1) << ESAI_xFCR_xFWM_SHIFT) & ESAI_xFCR_xFWM_MASK) +#define ESAI_xFCR_xE_SHIFT 2 +#define ESAI_xFCR_TE_WIDTH 6 +#define ESAI_xFCR_RE_WIDTH 4 +#define ESAI_xFCR_TE_MASK (((1 << ESAI_xFCR_TE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) +#define ESAI_xFCR_RE_MASK (((1 << ESAI_xFCR_RE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT) +#define ESAI_xFCR_TE(x) ((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_TE_MASK) +#define ESAI_xFCR_RE(x) ((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_RE_MASK) +#define ESAI_xFCR_xFR_SHIFT 1 +#define ESAI_xFCR_xFR_MASK (1 << ESAI_xFCR_xFR_SHIFT) +#define ESAI_xFCR_xFR (1 << ESAI_xFCR_xFR_SHIFT) +#define ESAI_xFCR_xFEN_SHIFT 0 +#define ESAI_xFCR_xFEN_MASK (1 << ESAI_xFCR_xFEN_SHIFT) +#define ESAI_xFCR_xFEN (1 << ESAI_xFCR_xFEN_SHIFT) + +/* + * Transmit FIFO Status Register -- REG_ESAI_TFSR 0x14 + * Receive FIFO Status Register --REG_ESAI_RFSR 0x1C + */ +#define ESAI_xFSR_NTFO_SHIFT 12 +#define ESAI_xFSR_NRFI_SHIFT 12 +#define ESAI_xFSR_NTFI_SHIFT 8 +#define ESAI_xFSR_NRFO_SHIFT 8 +#define ESAI_xFSR_NTFx_WIDTH 3 +#define ESAI_xFSR_NRFx_WIDTH 2 +#define ESAI_xFSR_NTFO_MASK (((1 << ESAI_xFSR_NTFx_WIDTH) - 1) << ESAI_xFSR_NTFO_SHIFT) +#define ESAI_xFSR_NTFI_MASK (((1 << ESAI_xFSR_NTFx_WIDTH) - 1) << ESAI_xFSR_NTFI_SHIFT) +#define ESAI_xFSR_NRFO_MASK (((1 << ESAI_xFSR_NRFx_WIDTH) - 1) << ESAI_xFSR_NRFO_SHIFT) +#define ESAI_xFSR_NRFI_MASK (((1 << ESAI_xFSR_NRFx_WIDTH) - 1) << ESAI_xFSR_NRFI_SHIFT) +#define ESAI_xFSR_xFCNT_SHIFT 0 +#define ESAI_xFSR_xFCNT_WIDTH 8 +#define ESAI_xFSR_xFCNT_MASK (((1 << ESAI_xFSR_xFCNT_WIDTH) - 1) << ESAI_xFSR_xFCNT_SHIFT) + +/* ESAI Transmit Slot Register -- REG_ESAI_TSR 0x98 */ +#define ESAI_TSR_SHIFT 0 +#define ESAI_TSR_WIDTH 24 +#define ESAI_TSR_MASK (((1 << ESAI_TSR_WIDTH) - 1) << ESAI_TSR_SHIFT) + +/* Serial Audio Interface Status Register -- REG_ESAI_SAISR 0xCC */ +#define ESAI_SAISR_TODFE_SHIFT 17 +#define ESAI_SAISR_TODFE_MASK (1 << ESAI_SAISR_TODFE_SHIFT) +#define ESAI_SAISR_TODFE (1 << ESAI_SAISR_TODFE_SHIFT) +#define ESAI_SAISR_TEDE_SHIFT 16 +#define ESAI_SAISR_TEDE_MASK (1 << ESAI_SAISR_TEDE_SHIFT) +#define ESAI_SAISR_TEDE (1 << ESAI_SAISR_TEDE_SHIFT) +#define ESAI_SAISR_TDE_SHIFT 15 +#define ESAI_SAISR_TDE_MASK (1 << ESAI_SAISR_TDE_SHIFT) +#define ESAI_SAISR_TDE (1 << ESAI_SAISR_TDE_SHIFT) +#define ESAI_SAISR_TUE_SHIFT 14 +#define ESAI_SAISR_TUE_MASK (1 << ESAI_SAISR_TUE_SHIFT) +#define ESAI_SAISR_TUE (1 << ESAI_SAISR_TUE_SHIFT) +#define ESAI_SAISR_TFS_SHIFT 13 +#define ESAI_SAISR_TFS_MASK (1 << ESAI_SAISR_TFS_SHIFT) +#define ESAI_SAISR_TFS (1 << ESAI_SAISR_TFS_SHIFT) +#define ESAI_SAISR_RODF_SHIFT 10 +#define ESAI_SAISR_RODF_MASK (1 << ESAI_SAISR_RODF_SHIFT) +#define ESAI_SAISR_RODF (1 << ESAI_SAISR_RODF_SHIFT) +#define ESAI_SAISR_REDF_SHIFT 9 +#define ESAI_SAISR_REDF_MASK (1 << ESAI_SAISR_REDF_SHIFT) +#define ESAI_SAISR_REDF (1 << ESAI_SAISR_REDF_SHIFT) +#define ESAI_SAISR_RDF_SHIFT 8 +#define ESAI_SAISR_RDF_MASK (1 << ESAI_SAISR_RDF_SHIFT) +#define ESAI_SAISR_RDF (1 << ESAI_SAISR_RDF_SHIFT) +#define ESAI_SAISR_ROE_SHIFT 7 +#define ESAI_SAISR_ROE_MASK (1 << ESAI_SAISR_ROE_SHIFT) +#define ESAI_SAISR_ROE (1 << ESAI_SAISR_ROE_SHIFT) +#define ESAI_SAISR_RFS_SHIFT 6 +#define ESAI_SAISR_RFS_MASK (1 << ESAI_SAISR_RFS_SHIFT) +#define ESAI_SAISR_RFS (1 << ESAI_SAISR_RFS_SHIFT) +#define ESAI_SAISR_IF2_SHIFT 2 +#define ESAI_SAISR_IF2_MASK (1 << ESAI_SAISR_IF2_SHIFT) +#define ESAI_SAISR_IF2 (1 << ESAI_SAISR_IF2_SHIFT) +#define ESAI_SAISR_IF1_SHIFT 1 +#define ESAI_SAISR_IF1_MASK (1 << ESAI_SAISR_IF1_SHIFT) +#define ESAI_SAISR_IF1 (1 << ESAI_SAISR_IF1_SHIFT) +#define ESAI_SAISR_IF0_SHIFT 0 +#define ESAI_SAISR_IF0_MASK (1 << ESAI_SAISR_IF0_SHIFT) +#define ESAI_SAISR_IF0 (1 << ESAI_SAISR_IF0_SHIFT) + +/* Serial Audio Interface Control Register -- REG_ESAI_SAICR 0xD0 */ +#define ESAI_SAICR_ALC_SHIFT 8 +#define ESAI_SAICR_ALC_MASK (1 << ESAI_SAICR_ALC_SHIFT) +#define ESAI_SAICR_ALC (1 << ESAI_SAICR_ALC_SHIFT) +#define ESAI_SAICR_TEBE_SHIFT 7 +#define ESAI_SAICR_TEBE_MASK (1 << ESAI_SAICR_TEBE_SHIFT) +#define ESAI_SAICR_TEBE (1 << ESAI_SAICR_TEBE_SHIFT) +#define ESAI_SAICR_SYNC_SHIFT 6 +#define ESAI_SAICR_SYNC_MASK (1 << ESAI_SAICR_SYNC_SHIFT) +#define ESAI_SAICR_SYNC (1 << ESAI_SAICR_SYNC_SHIFT) +#define ESAI_SAICR_OF2_SHIFT 2 +#define ESAI_SAICR_OF2_MASK (1 << ESAI_SAICR_OF2_SHIFT) +#define ESAI_SAICR_OF2 (1 << ESAI_SAICR_OF2_SHIFT) +#define ESAI_SAICR_OF1_SHIFT 1 +#define ESAI_SAICR_OF1_MASK (1 << ESAI_SAICR_OF1_SHIFT) +#define ESAI_SAICR_OF1 (1 << ESAI_SAICR_OF1_SHIFT) +#define ESAI_SAICR_OF0_SHIFT 0 +#define ESAI_SAICR_OF0_MASK (1 << ESAI_SAICR_OF0_SHIFT) +#define ESAI_SAICR_OF0 (1 << ESAI_SAICR_OF0_SHIFT) + +/* + * Transmit Control Register -- REG_ESAI_TCR 0xD4 + * Receive Control Register -- REG_ESAI_RCR 0xDC + */ +#define ESAI_xCR_xLIE_SHIFT 23 +#define ESAI_xCR_xLIE_MASK (1 << ESAI_xCR_xLIE_SHIFT) +#define ESAI_xCR_xLIE (1 << ESAI_xCR_xLIE_SHIFT) +#define ESAI_xCR_xIE_SHIFT 22 +#define ESAI_xCR_xIE_MASK (1 << ESAI_xCR_xIE_SHIFT) +#define ESAI_xCR_xIE (1 << ESAI_xCR_xIE_SHIFT) +#define ESAI_xCR_xEDIE_SHIFT 21 +#define ESAI_xCR_xEDIE_MASK (1 << ESAI_xCR_xEDIE_SHIFT) +#define ESAI_xCR_xEDIE (1 << ESAI_xCR_xEDIE_SHIFT) +#define ESAI_xCR_xEIE_SHIFT 20 +#define ESAI_xCR_xEIE_MASK (1 << ESAI_xCR_xEIE_SHIFT) +#define ESAI_xCR_xEIE (1 << ESAI_xCR_xEIE_SHIFT) +#define ESAI_xCR_xPR_SHIFT 19 +#define ESAI_xCR_xPR_MASK (1 << ESAI_xCR_xPR_SHIFT) +#define ESAI_xCR_xPR (1 << ESAI_xCR_xPR_SHIFT) +#define ESAI_xCR_PADC_SHIFT 17 +#define ESAI_xCR_PADC_MASK (1 << ESAI_xCR_PADC_SHIFT) +#define ESAI_xCR_PADC (1 << ESAI_xCR_PADC_SHIFT) +#define ESAI_xCR_xFSR_SHIFT 16 +#define ESAI_xCR_xFSR_MASK (1 << ESAI_xCR_xFSR_SHIFT) +#define ESAI_xCR_xFSR (1 << ESAI_xCR_xFSR_SHIFT) +#define ESAI_xCR_xFSL_SHIFT 15 +#define ESAI_xCR_xFSL_MASK (1 << ESAI_xCR_xFSL_SHIFT) +#define ESAI_xCR_xFSL (1 << ESAI_xCR_xFSL_SHIFT) +#define ESAI_xCR_xSWS_SHIFT 10 +#define ESAI_xCR_xSWS_WIDTH 5 +#define ESAI_xCR_xSWS_MASK (((1 << ESAI_xCR_xSWS_WIDTH) - 1) << ESAI_xCR_xSWS_SHIFT) +#define ESAI_xCR_xSWS(s, w) ((w < 24 ? (s - w + ((w - 8) >> 2)) : (s < 32 ? 0x1e : 0x1f)) << ESAI_xCR_xSWS_SHIFT) +#define ESAI_xCR_xMOD_SHIFT 8 +#define ESAI_xCR_xMOD_WIDTH 2 +#define ESAI_xCR_xMOD_MASK (((1 << ESAI_xCR_xMOD_WIDTH) - 1) << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xMOD_ONDEMAND (0x1 << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xMOD_NETWORK (0x1 << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xMOD_AC97 (0x3 << ESAI_xCR_xMOD_SHIFT) +#define ESAI_xCR_xWA_SHIFT 7 +#define ESAI_xCR_xWA_MASK (1 << ESAI_xCR_xWA_SHIFT) +#define ESAI_xCR_xWA (1 << ESAI_xCR_xWA_SHIFT) +#define ESAI_xCR_xSHFD_SHIFT 6 +#define ESAI_xCR_xSHFD_MASK (1 << ESAI_xCR_xSHFD_SHIFT) +#define ESAI_xCR_xSHFD (1 << ESAI_xCR_xSHFD_SHIFT) +#define ESAI_xCR_xE_SHIFT 0 +#define ESAI_xCR_TE_WIDTH 6 +#define ESAI_xCR_RE_WIDTH 4 +#define ESAI_xCR_TE_MASK (((1 << ESAI_xCR_TE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) +#define ESAI_xCR_RE_MASK (((1 << ESAI_xCR_RE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT) +#define ESAI_xCR_TE(x) ((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_TE_MASK) +#define ESAI_xCR_RE(x) ((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_RE_MASK) + +/* + * Transmit Clock Control Register -- REG_ESAI_TCCR 0xD8 + * Receive Clock Control Register -- REG_ESAI_RCCR 0xE0 + */ +#define ESAI_xCCR_xHCKD_SHIFT 23 +#define ESAI_xCCR_xHCKD_MASK (1 << ESAI_xCCR_xHCKD_SHIFT) +#define ESAI_xCCR_xHCKD (1 << ESAI_xCCR_xHCKD_SHIFT) +#define ESAI_xCCR_xFSD_SHIFT 22 +#define ESAI_xCCR_xFSD_MASK (1 << ESAI_xCCR_xFSD_SHIFT) +#define ESAI_xCCR_xFSD (1 << ESAI_xCCR_xFSD_SHIFT) +#define ESAI_xCCR_xCKD_SHIFT 21 +#define ESAI_xCCR_xCKD_MASK (1 << ESAI_xCCR_xCKD_SHIFT) +#define ESAI_xCCR_xCKD (1 << ESAI_xCCR_xCKD_SHIFT) +#define ESAI_xCCR_xHCKP_SHIFT 20 +#define ESAI_xCCR_xHCKP_MASK (1 << ESAI_xCCR_xHCKP_SHIFT) +#define ESAI_xCCR_xHCKP (1 << ESAI_xCCR_xHCKP_SHIFT) +#define ESAI_xCCR_xFSP_SHIFT 19 +#define ESAI_xCCR_xFSP_MASK (1 << ESAI_xCCR_xFSP_SHIFT) +#define ESAI_xCCR_xFSP (1 << ESAI_xCCR_xFSP_SHIFT) +#define ESAI_xCCR_xCKP_SHIFT 18 +#define ESAI_xCCR_xCKP_MASK (1 << ESAI_xCCR_xCKP_SHIFT) +#define ESAI_xCCR_xCKP (1 << ESAI_xCCR_xCKP_SHIFT) +#define ESAI_xCCR_xFP_SHIFT 14 +#define ESAI_xCCR_xFP_WIDTH 4 +#define ESAI_xCCR_xFP_MASK (((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT) +#define ESAI_xCCR_xFP(v) (((v - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK) +#define ESAI_xCCR_xDC_SHIFT 9 +#define ESAI_xCCR_xDC_WIDTH 4 +#define ESAI_xCCR_xDC_MASK (((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT) +#define ESAI_xCCR_xDC(v) (((v - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK) +#define ESAI_xCCR_xPSR_SHIFT 8 +#define ESAI_xCCR_xPSR_MASK (1 << ESAI_xCCR_xPSR_SHIFT) +#define ESAI_xCCR_xPSR_BYPASS (1 << ESAI_xCCR_xPSR_SHIFT) +#define ESAI_xCCR_xPSR_DIV8 (0 << ESAI_xCCR_xPSR_SHIFT) +#define ESAI_xCCR_xPM_SHIFT 0 +#define ESAI_xCCR_xPM_WIDTH 8 +#define ESAI_xCCR_xPM_MASK (((1 << ESAI_xCCR_xPM_WIDTH) - 1) << ESAI_xCCR_xPM_SHIFT) +#define ESAI_xCCR_xPM(v) (((v - 1) << ESAI_xCCR_xPM_SHIFT) & ESAI_xCCR_xPM_MASK) + +/* Transmit Slot Mask Register A/B -- REG_ESAI_TSMA/B 0xE4 ~ 0xF0 */ +#define ESAI_xSMA_xS_SHIFT 0 +#define ESAI_xSMA_xS_WIDTH 16 +#define ESAI_xSMA_xS_MASK (((1 << ESAI_xSMA_xS_WIDTH) - 1) << ESAI_xSMA_xS_SHIFT) +#define ESAI_xSMA_xS(v) (v & ESAI_xSMA_xS_MASK) +#define ESAI_xSMB_xS_SHIFT 0 +#define ESAI_xSMB_xS_WIDTH 16 +#define ESAI_xSMB_xS_MASK (((1 << ESAI_xSMB_xS_WIDTH) - 1) << ESAI_xSMB_xS_SHIFT) +#define ESAI_xSMB_xS(v) ((v >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMA_xS_MASK) + +/* Port C Direction Register -- REG_ESAI_PRRC 0xF8 */ +#define ESAI_PRRC_PDC_SHIFT 0 +#define ESAI_PRRC_PDC_WIDTH 12 +#define ESAI_PRRC_PDC_MASK (((1 << ESAI_PRRC_PDC_WIDTH) - 1) << ESAI_PRRC_PDC_SHIFT) +#define ESAI_PRRC_PDC(v) (v & ESAI_PRRC_PDC_MASK) + +/* Port C Control Register -- REG_ESAI_PCRC 0xFC */ +#define ESAI_PCRC_PC_SHIFT 0 +#define ESAI_PCRC_PC_WIDTH 12 +#define ESAI_PCRC_PC_MASK (((1 << ESAI_PCRC_PC_WIDTH) - 1) << ESAI_PCRC_PC_SHIFT) +#define ESAI_PCRC_PC(v) (v & ESAI_PCRC_PC_MASK) + +#define ESAI_GPIO 0xfff + +/* ESAI clock source */ +#define ESAI_HCK_FSYS 0 +#define ESAI_HCK_EXTAL 1 + +/* ESAI clock divider */ +#define ESAI_TX_DIV_PSR 0 +#define ESAI_TX_DIV_PM 1 +#define ESAI_TX_DIV_FP 2 +#define ESAI_RX_DIV_PSR 3 +#define ESAI_RX_DIV_PM 4 +#define ESAI_RX_DIV_FP 5 +#endif /* _FSL_ESAI_DAI_H */ -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: simple-card: fix the cinfo error check
by Xiubo Li 09 Jan '14

09 Jan '14

If the dt is used and the cinfo is NULL, the -ENOMEM should be return. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 51a4784..406e4ea 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -223,15 +223,17 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dev_err(dev, "parse error %d\n", ret); return ret; } + } else { + return -ENOMEM; } } else { cinfo = pdev->dev.platform_data; - cinfo->snd_card.dev = &pdev->dev; - } + if (!cinfo) { + dev_err(dev, "no info for asoc-simple-card\n"); + return -EINVAL; + } - if (!cinfo) { - dev_err(dev, "no info for asoc-simple-card\n"); - return -EINVAL; + cinfo->snd_card.dev = &pdev->dev; } if (!cinfo->name || -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: simple-card: fix a bug where cinfo will be NULL before using it
by Xiubo Li 09 Jan '14

09 Jan '14

If the dt is not used, the cinfo will be always NULL before using it. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index c67d2db..51a4784 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -225,8 +225,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } } } else { - cinfo->snd_card.dev = &pdev->dev; cinfo = pdev->dev.platform_data; + cinfo->snd_card.dev = &pdev->dev; } if (!cinfo) { -- 1.8.4

2 1

[alsa-devel] [PATCH v2] ASoC: fsl_ssi: Set default slot number for common cases
by Nicolin Chen 09 Jan '14

09 Jan '14

For those platforms using DAI master mode like I2S, it's better to pre-set a default slot number so that there's no need for these common cases to set the slot number from its machine driver any more. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v2: * Correct coding style for multi-line comment. sound/soc/fsl/fsl_ssi.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 94dedcb..aad2a1f 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -711,6 +711,17 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) if (ssi_private->imx_ac97) fsl_ssi_setup_ac97(ssi_private); + /* + * Set a default slot number so that there is no need for those common + * cases like I2S mode to call the extra set_tdm_slot() any more. + */ + if (!ssi_private->imx_ac97) { + write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(2)); + write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(2)); + } + return 0; } -- 1.8.4

2 1

[alsa-devel] [PATCH -next] ASoC: fsl-ssi: Add missing clk_disable_unprepare() on error in fsl_ssi_probe()
by Wei Yongjun 09 Jan '14

09 Jan '14

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Add the missing clk_disable_unprepare() before return from fsl_ssi_probe() in the request irq error handling case. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index e18b4b3..4c6818d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1464,7 +1464,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (ret < 0) { dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq); - goto error_irqmap; + goto error_clk; } }

2 1

[alsa-devel] [PATCH] ASoC: wm8804: Allow fine-grained control of the PLL generation
by Daniel Matuschek 09 Jan '14

09 Jan '14

Signed-off-by: Daniel Matuschek <daniel(a)matuschek.net> WM8804 can run with PLL frequencies of 256xfs and 128xfs for most sample rates. At 192kHz only 128xfs is supported. The existing driver selects 128xfs automatically for some lower samples rates. By using the "pllid" argument of the "set_pll" function is is now possible to control the behaviour. This allows using 256xfs PLL frequency on all sample rates up to 96kHz. It should allow lower jitter and better signal quality. When pllid=0, the behaviour of the driver does not change. --- sound/soc/codecs/wm8804.c | 18 +++++++++++++----- sound/soc/codecs/wm8804.h | 7 +++++++ 2 files changed, 20 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 1704b1e..e79c408 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -2,6 +2,8 @@ * wm8804.c -- WM8804 S/PDIF transceiver driver * * Copyright 2010-11 Wolfson Microelectronics plc + * patched by Daniel Matuschek <info(a)crazyaudio.com> to allow + * fine-grained control of PLL * * Author: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> * @@ -318,7 +320,7 @@ static struct { #define FIXED_PLL_SIZE ((1ULL << 22) * 10) static int pll_factors(struct pll_div *pll_div, unsigned int target, - unsigned int source) + unsigned int source, int mclk_div) { u64 Kpart; unsigned long int K, Ndiv, Nmod, tmp; @@ -332,9 +334,15 @@ static int pll_factors(struct pll_div *pll_div, unsigned int target, tmp = target * post_table[i].div; if (tmp >= 90000000 && tmp <= 100000000) { pll_div->freqmode = post_table[i].freqmode; - pll_div->mclkdiv = post_table[i].mclkdiv; - target *= post_table[i].div; - break; + if ((mclk_div == WM8804_MCLKDIV_DONTCARE) || + ((post_table[i].mclkdiv == 1) && + (mclk_div == WM8804_MCLKDIV_1)) || + ((post_table[i].mclkdiv == 0) && + (mclk_div == WM8804_MCLKDIV_0))) { + pll_div->mclkdiv = post_table[i].mclkdiv; + target *= post_table[i].div; + break; + } } } @@ -388,7 +396,7 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, int ret; struct pll_div pll_div; - ret = pll_factors(&pll_div, freq_out, freq_in); + ret = pll_factors(&pll_div, freq_out, freq_in, pll_id); if (ret) return ret; diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index 8ec14f5..0365177 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -58,4 +58,11 @@ #define WM8804_CLKOUT_DIV 1 +#define WM8804_MCLKDIV_DONTCARE 0 +#define WM8804_MCLKDIV_0 1 +#define WM8804_MCLKDIV_1 2 +#define WM8804_PLL_MCLKDIV_DONTCARE WM8804_MCLKDIV_DONTCARE +#define WM8804_PLL_MCLKDIV_0 WM8804_MCLKDIV_0 +#define WM8804_PLL_MCLKDIV_1 WM8804_MCLKDIV_1 + #endif /* _WM8804_H */ -- 1.7.9.5

4 5

[alsa-devel] [PATCH -next] ASoC: wm_adsp: Fix sparse non static symbol warning
by Wei Yongjun 09 Jan '14

09 Jan '14

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Fixes the following sparse warning: sound/soc/codecs/wm_adsp.c:1500:6: warning: symbol 'wm_adsp2_boot_work' was not declared. Should it be static? Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 41669ad..444626f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1497,7 +1497,7 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) return 0; } -void wm_adsp2_boot_work(struct work_struct *work) +static void wm_adsp2_boot_work(struct work_struct *work) { struct wm_adsp *dsp = container_of(work, struct wm_adsp,

2 1

[alsa-devel] [PATCH v2 01/11] ASoC: dpcm: Add helper function for initializing runtime pcm
by Lars-Peter Clausen 09 Jan '14

09 Jan '14

We have the same code for initializing the runtime pcm on both the playback and the capture path. Factor this out into a common helper function. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Jean-Francois Moine <moinejf(a)free.fr> --- New in v2 --- sound/soc/soc-pcm.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 604e7e9..898149e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1235,6 +1235,17 @@ unwind: return err; } +static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime, + struct snd_soc_pcm_stream *stream) +{ + runtime->hw.rate_min = stream->rate_min; + runtime->hw.rate_max = stream->rate_max; + runtime->hw.channels_min = stream->channels_min; + runtime->hw.channels_max = stream->channels_max; + runtime->hw.formats &= stream->formats; + runtime->hw.rates = stream->rates; +} + static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -1242,21 +1253,10 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - runtime->hw.rate_min = cpu_dai_drv->playback.rate_min; - runtime->hw.rate_max = cpu_dai_drv->playback.rate_max; - runtime->hw.channels_min = cpu_dai_drv->playback.channels_min; - runtime->hw.channels_max = cpu_dai_drv->playback.channels_max; - runtime->hw.formats &= cpu_dai_drv->playback.formats; - runtime->hw.rates = cpu_dai_drv->playback.rates; - } else { - runtime->hw.rate_min = cpu_dai_drv->capture.rate_min; - runtime->hw.rate_max = cpu_dai_drv->capture.rate_max; - runtime->hw.channels_min = cpu_dai_drv->capture.channels_min; - runtime->hw.channels_max = cpu_dai_drv->capture.channels_max; - runtime->hw.formats &= cpu_dai_drv->capture.formats; - runtime->hw.rates = cpu_dai_drv->capture.rates; - } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback); + else + dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture); } static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) -- 1.8.0

3 21

[alsa-devel] [PATCH 1/5] ASoC: ad1836: Use params_width() rather than explicit memory format
by Mark Brown 09 Jan '14

09 Jan '14

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ad1836.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index d7c983862cf0..77f459868579 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -168,15 +168,15 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, int word_len = 0; /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: word_len = AD1836_WORD_LEN_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: word_len = AD1836_WORD_LEN_20; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + case 24: + case 32: word_len = AD1836_WORD_LEN_24; break; default: -- 1.8.5.2

2 6

[alsa-devel] [PATCH] ASoC: wm_adsp: Mark wm_adsp2_boot_work as static
by Charles Keepax 09 Jan '14

09 Jan '14

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 41669ad..444626f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1497,7 +1497,7 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) return 0; } -void wm_adsp2_boot_work(struct work_struct *work) +static void wm_adsp2_boot_work(struct work_struct *work) { struct wm_adsp *dsp = container_of(work, struct wm_adsp, -- 1.7.2.5

2 1

[alsa-devel] [PATCH 1/2] mfd: wm5110: Add registers for headphone short circuit control
by Charles Keepax 09 Jan '14

09 Jan '14

Add the registers necessary to enable/disable the headphone short circuit protection. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm5110-tables.c | 6 ++++++ include/linux/mfd/arizona/registers.h | 27 +++++++++++++++++++++++++++ 2 files changed, 33 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index 24ec464..30f8163 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -637,6 +637,9 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000491, 0x0000 }, /* R1169 - PDM SPK1 CTRL 2 */ { 0x00000492, 0x0069 }, /* R1170 - PDM SPK2 CTRL 1 */ { 0x00000493, 0x0000 }, /* R1171 - PDM SPK2 CTRL 2 */ + { 0x000004A0, 0x3480 }, /* R1184 - HP1 Short Circuit Ctrl */ + { 0x000004A1, 0x3480 }, /* R1185 - HP2 Short Circuit Ctrl */ + { 0x000004A2, 0x3480 }, /* R1186 - HP3 Short Circuit Ctrl */ { 0x00000500, 0x000C }, /* R1280 - AIF1 BCLK Ctrl */ { 0x00000501, 0x0008 }, /* R1281 - AIF1 Tx Pin Ctrl */ { 0x00000502, 0x0000 }, /* R1282 - AIF1 Rx Pin Ctrl */ @@ -1633,6 +1636,9 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_PDM_SPK1_CTRL_2: case ARIZONA_PDM_SPK2_CTRL_1: case ARIZONA_PDM_SPK2_CTRL_2: + case ARIZONA_HP1_SHORT_CIRCUIT_CTRL: + case ARIZONA_HP2_SHORT_CIRCUIT_CTRL: + case ARIZONA_HP3_SHORT_CIRCUIT_CTRL: case ARIZONA_AIF1_BCLK_CTRL: case ARIZONA_AIF1_TX_PIN_CTRL: case ARIZONA_AIF1_RX_PIN_CTRL: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index b40960d..2d3ac50 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -226,6 +226,9 @@ #define ARIZONA_PDM_SPK1_CTRL_2 0x491 #define ARIZONA_PDM_SPK2_CTRL_1 0x492 #define ARIZONA_PDM_SPK2_CTRL_2 0x493 +#define ARIZONA_HP1_SHORT_CIRCUIT_CTRL 0x4A0 +#define ARIZONA_HP2_SHORT_CIRCUIT_CTRL 0x4A1 +#define ARIZONA_HP3_SHORT_CIRCUIT_CTRL 0x4A2 #define ARIZONA_SPK_CTRL_2 0x4B5 #define ARIZONA_SPK_CTRL_3 0x4B6 #define ARIZONA_DAC_COMP_1 0x4DC @@ -3297,6 +3300,30 @@ #define ARIZONA_SPK2_FMT_WIDTH 1 /* SPK2_FMT */ /* + * R1184 (0x4A0) - HP1 Short Circuit Ctrl + */ +#define ARIZONA_HP1_SC_ENA 0x1000 /* HP1_SC_ENA */ +#define ARIZONA_HP1_SC_ENA_MASK 0x1000 /* HP1_SC_ENA */ +#define ARIZONA_HP1_SC_ENA_SHIFT 12 /* HP1_SC_ENA */ +#define ARIZONA_HP1_SC_ENA_WIDTH 1 /* HP1_SC_ENA */ + +/* + * R1185 (0x4A1) - HP2 Short Circuit Ctrl + */ +#define ARIZONA_HP2_SC_ENA 0x1000 /* HP2_SC_ENA */ +#define ARIZONA_HP2_SC_ENA_MASK 0x1000 /* HP2_SC_ENA */ +#define ARIZONA_HP2_SC_ENA_SHIFT 12 /* HP2_SC_ENA */ +#define ARIZONA_HP2_SC_ENA_WIDTH 1 /* HP2_SC_ENA */ + +/* + * R1186 (0x4A2) - HP3 Short Circuit Ctrl + */ +#define ARIZONA_HP3_SC_ENA 0x1000 /* HP3_SC_ENA */ +#define ARIZONA_HP3_SC_ENA_MASK 0x1000 /* HP3_SC_ENA */ +#define ARIZONA_HP3_SC_ENA_SHIFT 12 /* HP3_SC_ENA */ +#define ARIZONA_HP3_SC_ENA_WIDTH 1 /* HP3_SC_ENA */ + +/* * R1244 (0x4DC) - DAC comp 1 */ #define ARIZONA_OUT_COMP_COEFF_MASK 0xFFFF /* OUT_COMP_COEFF - [15:0] */ -- 1.7.2.5

2 2

[alsa-devel] [PATCH] ASoC: bcm: Remove obsoleted Kconfig dependency
by Takashi Iwai 09 Jan '14

09 Jan '14

CONFIG_SND_SOC_DMAENGINE_PCM was renamed to CONFIG_SND_DMAENGINE_PCM recently. And yet we don't have to select it since CONFIG_SND_GENERIC_DMAENGINE_PCM selects the dependency by itself, so just rip it off. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/bcm/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/bcm/Kconfig b/sound/soc/bcm/Kconfig index 3d82a29ce3a8..6a834e109f1d 100644 --- a/sound/soc/bcm/Kconfig +++ b/sound/soc/bcm/Kconfig @@ -1,7 +1,6 @@ config SND_BCM2835_SOC_I2S tristate "SoC Audio support for the Broadcom BCM2835 I2S module" depends on ARCH_BCM2835 || COMPILE_TEST - select SND_SOC_DMAENGINE_PCM select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help -- 1.8.5.2

3 8

[alsa-devel] [PATCH] ASoC: add RT5670 CODEC driver
by bardliao＠realtek.com 09 Jan '14

09 Jan '14

From: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- This is the initial codec driver of ALC5670 I have tested playback to Headphone, Line out and PDM interface. But not any capture path. --- include/sound/rt5670.h | 20 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5670-dsp.c | 817 +++++++++++ sound/soc/codecs/rt5670-dsp.h | 69 + sound/soc/codecs/rt5670.c | 3053 +++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5670.h | 1966 ++++++++++++++++++++++++++ 7 files changed, 5931 insertions(+) create mode 100644 include/sound/rt5670.h create mode 100644 sound/soc/codecs/rt5670-dsp.c create mode 100644 sound/soc/codecs/rt5670-dsp.h create mode 100644 sound/soc/codecs/rt5670.c create mode 100644 sound/soc/codecs/rt5670.h diff --git a/include/sound/rt5670.h b/include/sound/rt5670.h new file mode 100644 index 0000000..2397fb8 --- /dev/null +++ b/include/sound/rt5670.h @@ -0,0 +1,20 @@ +/* + * linux/sound/rt5670.h -- Platform data for RT5670 + * + * Copyright 2011 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_RT5670_H +#define __LINUX_SND_RT5670_H + +struct rt5670_platform_data { + bool asrc_en; + bool jd1_en; + bool in2_diff; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087a..4400433 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -61,6 +61,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C + select SND_SOC_RT5670 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SN95031 if INTEL_SCU_IPC @@ -319,6 +320,9 @@ config SND_SOC_RT5631 config SND_SOC_RT5640 tristate +config SND_SOC_RT5670 + tristate + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc12676..f95e65a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -48,6 +48,7 @@ snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o +snd-soc-rt5670-objs := rt5670.o rt5670-dsp.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -181,6 +182,7 @@ obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o +obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o diff --git a/sound/soc/codecs/rt5670-dsp.c b/sound/soc/codecs/rt5670-dsp.c new file mode 100644 index 0000000..ed01fa6 --- /dev/null +++ b/sound/soc/codecs/rt5670-dsp.c @@ -0,0 +1,817 @@ +/* + * rt5670.c -- RT5670 ALSA SoC DSP driver + * + * Copyright 2011 Realtek Semiconductor Corp. + * 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. + */ + +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "rt5670.h" +#include "rt5670-dsp.h" + +#define DSP_CLK_RATE RT5670_DSP_CLK_96K + +/* DSP init */ +static unsigned short rt5670_dsp_init[][2] = { + {0x2260, 0x30d9}, {0x2261, 0x30d9}, {0x2289, 0x7fff}, {0x2290, 0x7fff}, + {0x2288, 0x0002}, {0x22b2, 0x0002}, {0x2295, 0x0001}, {0x22b3, 0x0001}, + {0x22d7, 0x0008}, {0x22d8, 0x0009}, {0x22d9, 0x0000}, {0x22da, 0x0001}, + {0x22fd, 0x009e}, {0x22c1, 0x1006}, {0x22c2, 0x1006}, {0x22c3, 0x1007}, + {0x22c4, 0x1007} +}; +#define RT5670_DSP_INIT_NUM \ + (sizeof(rt5670_dsp_init) / sizeof(rt5670_dsp_init[0])) + +/* MCLK rate */ +static unsigned short rt5670_dsp_4096000[][2] = { + {0x226c, 0x000c}, {0x226d, 0x000c}, {0x226e, 0x0022}, +}; +#define RT5670_DSP_4096000_NUM \ + (sizeof(rt5670_dsp_4096000) / sizeof(rt5670_dsp_4096000[0])) + +static unsigned short rt5670_dsp_12288000[][2] = { + {0x226c, 0x000c}, {0x226d, 0x000c}, {0x226e, 0x0026}, +}; +#define RT5670_DSP_12288000_NUM \ + (sizeof(rt5670_dsp_12288000) / sizeof(rt5670_dsp_12288000[0])) + +static unsigned short rt5670_dsp_11289600[][2] = { + {0x226c, 0x0031}, {0x226d, 0x0050}, {0x226e, 0x0009}, +}; +#define RT5670_DSP_11289600_NUM \ + (sizeof(rt5670_dsp_11289600) / sizeof(rt5670_dsp_11289600[0])) + +static unsigned short rt5670_dsp_24576000[][2] = { + {0x226c, 0x000c}, {0x226d, 0x000c}, {0x226e, 0x002c}, +}; +#define RT5670_DSP_24576000_NUM \ + (sizeof(rt5670_dsp_24576000) / sizeof(rt5670_dsp_24576000[0])) + +/* sample rate */ +static unsigned short rt5670_dsp_48_441[][2] = { + {0x22f2, 0x0058}, {0x2301, 0x0016} +}; +#define RT5670_DSP_48_441_NUM \ + (sizeof(rt5670_dsp_48_441) / sizeof(rt5670_dsp_48_441[0])) + +static unsigned short rt5670_dsp_24[][2] = { + {0x22f2, 0x0058}, {0x2301, 0x0004} +}; +#define RT5670_DSP_24_NUM (sizeof(rt5670_dsp_24) / sizeof(rt5670_dsp_24[0])) + +static unsigned short rt5670_dsp_16[][2] = { + {0x22f2, 0x004c}, {0x2301, 0x0002} +}; +#define RT5670_DSP_16_NUM (sizeof(rt5670_dsp_16) / sizeof(rt5670_dsp_16[0])) + +static unsigned short rt5670_dsp_8[][2] = { + {0x22f2, 0x004c}, {0x2301, 0x0000} +}; +#define RT5670_DSP_8_NUM (sizeof(rt5670_dsp_8) / sizeof(rt5670_dsp_8[0])) + +/* DSP mode */ +static unsigned short rt5670_dsp_ns[][2] = { + {0x22f8, 0x8005}, {0x2375, 0x7ff0}, {0x2376, 0x7990}, {0x2377, 0x7332}, + {0x2388, 0x7fff}, {0x2389, 0x6000}, {0x238a, 0x0000}, {0x238b, 0x1000}, + {0x238c, 0x1000}, {0x23a1, 0x2000}, {0x2303, 0x0200}, {0x2304, 0x0032}, + {0x2305, 0x0000}, {0x230c, 0x0200}, {0x22fb, 0x0000} +}; +#define RT5670_DSP_NS_NUM \ + (sizeof(rt5670_dsp_ns) / sizeof(rt5670_dsp_ns[0])) + +static unsigned short rt5670_dsp_aec[][2] = { + {0x22f8, 0x8003}, {0x232f, 0x00d0}, {0x2355, 0x2666}, {0x2356, 0x2666}, + {0x2357, 0x2666}, {0x2358, 0x6666}, {0x2359, 0x6666}, {0x235a, 0x6666}, + {0x235b, 0x7fff}, {0x235c, 0x7fff}, {0x235d, 0x7fff}, {0x235e, 0x7fff}, + {0x235f, 0x7fff}, {0x2360, 0x7fff}, {0x2361, 0x7fff}, {0x2362, 0x1000}, + {0x2367, 0x0007}, {0x2368, 0x4000}, {0x2369, 0x0008}, {0x236a, 0x2000}, + {0x236b, 0x0009}, {0x236c, 0x003c}, {0x236d, 0x0000}, {0x236f, 0x2000}, + {0x2370, 0x0008}, {0x2371, 0x000a}, {0x2373, 0x0000}, {0x2374, 0x7fff}, + {0x2375, 0x7ff0}, {0x2376, 0x7990}, {0x2377, 0x7332}, {0x2379, 0x1000}, + {0x237a, 0x1000}, {0x2388, 0x7fff}, {0x2389, 0x6000}, {0x238a, 0x4000}, + {0x238b, 0x1000}, {0x238c, 0x1000}, {0x2398, 0x4668}, {0x23a1, 0x2000}, + {0x23a3, 0x4000}, {0x23ad, 0x2000}, {0x23ae, 0x2000}, {0x23af, 0x2000}, + {0x23b4, 0x2000}, {0x23b5, 0x2000}, {0x23b6, 0x2000}, {0x23bb, 0x6000}, + {0x2303, 0x0710}, {0x2304, 0x0332}, {0x230c, 0x0200}, {0x230d, 0x0080}, + {0x2310, 0x0010}, {0x22fb, 0x0000} +}; +#define RT5670_DSP_AEC_NUM \ + (sizeof(rt5670_dsp_aec) / sizeof(rt5670_dsp_aec[0])) + +static unsigned short rt5670_dsp_vt[][2] = { + {0x22f8, 0x8003}, {0x2371, 0x000a}, {0x2373, 0x0000}, {0x2374, 0x7fff}, + {0x2375, 0x7ff0}, {0x2376, 0x7990}, {0x2377, 0x7332}, {0x2379, 0x1000}, + {0x237a, 0x1000}, {0x2388, 0x7fff}, {0x2389, 0x6000}, {0x238b, 0x1000}, + {0x238c, 0x1000}, {0x23a1, 0x2000}, {0x2304, 0x0332}, {0x230c, 0x0200}, + {0x22fb, 0x0000} +}; +#define RT5670_DSP_VT_NUM (sizeof(rt5670_dsp_vt) / sizeof(rt5670_dsp_vt[0])) + +static unsigned short rt5670_dsp_vr[][2] = { + {0x22f8, 0x8003}, {0x2304, 0x0332}, {0x2305, 0x0000}, {0x2309, 0x0400}, + {0x230c, 0x0200}, {0x230d, 0x0080}, {0x2310, 0x0004}, {0x232f, 0x0100}, + {0x2371, 0x000a}, {0x2373, 0x3000}, {0x2374, 0x5000}, {0x2375, 0x7ff0}, + {0x2376, 0x7990}, {0x2377, 0x7332}, {0x2379, 0x1000}, {0x237a, 0x1000}, + {0x2386, 0x0200}, {0x2388, 0x4000}, {0x2389, 0x4000}, {0x238a, 0x0000}, + {0x238b, 0x2000}, {0x238c, 0x1800}, {0x239b, 0x0000}, {0x239c, 0x0a00}, + {0x239d, 0x0000}, {0x239e, 0x7fff}, {0x239f, 0x0001}, {0x23a1, 0x3000}, + {0x23a2, 0x1000}, {0x23ad, 0x0000}, {0x23ae, 0x0000}, {0x23af, 0x0000}, + {0x23c4, 0x2000}, {0x23b0, 0x0000}, {0x23b1, 0x0000}, {0x23b2, 0x0000}, + {0x23b3, 0x0000}, {0x23b4, 0x0000}, {0x23b5, 0x0000}, {0x23b6, 0x0000}, + {0x23b7, 0x0000}, {0x23b8, 0x0000}, {0x23b9, 0x0000}, {0x23ba, 0x0000}, + {0x23bb, 0x0000}, {0x22fb, 0x0000} +}; +#define RT5670_DSP_VR_NUM (sizeof(rt5670_dsp_vr) / sizeof(rt5670_dsp_vr[0])) + +static unsigned short rt5670_dsp_ffp_ns[][2] = { + {0x22f8, 0x8003}, {0x2304, 0x8332}, {0x2371, 0x000a}, {0x2373, 0x0000}, + {0x2374, 0x7fff}, {0x2379, 0x1800}, {0x237a, 0x1800}, {0x230c, 0x0200}, + {0x23a2, 0x1000}, {0x2388, 0x7000}, {0x238b, 0x2000}, {0x238c, 0x2000}, + {0x23a8, 0x2000}, {0x23a9, 0x4000}, {0x23aa, 0x0100}, {0x23ab, 0x7800}, + {0x22fb, 0x0000} +}; +#define RT5670_DSP_FFP_NS_NUM \ + (sizeof(rt5670_dsp_ffp_ns) / sizeof(rt5670_dsp_ffp_ns[0])) + +static unsigned short rt5670_dsp_48k_sto_ffp[][2] = { + {0x22c1, 0x1025}, {0x22c2, 0x1026}, {0x22f8, 0x8004}, {0x22ea, 0x0001}, + {0x230c, 0x0100}, {0x230d, 0x0100}, {0x2301, 0x0010}, {0x2303, 0x0200}, + {0x2304, 0x8000}, {0x2305, 0x0000}, {0x2388, 0x6500}, {0x238b, 0x4000}, + {0x238c, 0x4000}, {0x23a9, 0x2000}, {0x23aa, 0x0200}, {0x23ab, 0x7c00}, + {0x22fb, 0x0000} +}; +#define RT5670_DSP_48K_STO_FFP_NUM \ + (sizeof(rt5670_dsp_48k_sto_ffp) / sizeof(rt5670_dsp_48k_sto_ffp[0])) + +static unsigned short rt5670_dsp_2mic_handset[][2] = { + {0x22f8, 0x8002}, {0x2301, 0x0002}, {0x2302, 0x0002}, {0x2303, 0x0710}, + {0x2304, 0x4332}, {0x2305, 0x206c}, {0x236e, 0x0000}, {0x236f, 0x0001}, + {0x237e, 0x0001}, {0x237f, 0x3800}, {0x2380, 0x3000}, {0x2381, 0x0005}, + {0x2382, 0x0040}, {0x2383, 0x7fff}, {0x2388, 0x2c00}, {0x2389, 0x2800}, + {0x238b, 0x1800}, {0x238c, 0x1800}, {0x238f, 0x2000}, {0x239b, 0x0002}, + {0x239c, 0x0a00}, {0x239f, 0x0001}, {0x230c, 0x0200}, {0x22fb, 0x0000} +}; +#define RT5670_DSP_2MIC_HANDSET_NUM \ + (sizeof(rt5670_dsp_2mic_handset) / sizeof(rt5670_dsp_2mic_handset[0])) + +static unsigned short rt5670_dsp_2mic_handsfree[][2] = { + {0x22f8, 0x8003}, {0x2371, 0x000a}, {0x2373, 0x0000}, {0x2374, 0x7fff}, + {0x2375, 0x7ff0}, {0x2376, 0x7990}, {0x2377, 0x7332}, {0x2379, 0x1000}, + {0x237a, 0x1000}, {0x2388, 0x7fff}, {0x2389, 0x6000}, {0x238b, 0x1000}, + {0x238c, 0x1000}, {0x23a1, 0x2000}, {0x2304, 0x0332}, {0x230c, 0x0200}, + {0x22fb, 0x0000} +}; +#define RT5670_DSP_2MIC_HANDSFREE_NUM \ + (sizeof(rt5670_dsp_2mic_handsfree) / \ + sizeof(rt5670_dsp_2mic_handsfree[0])) + +static unsigned short rt5670_dsp_aec_handsfree[][2] = { + {0x22f8, 0x8003}, {0x232f, 0x00d0}, {0x2355, 0x2666}, {0x2356, 0x2666}, + {0x2357, 0x2666}, {0x2358, 0x6666}, {0x2359, 0x6666}, {0x235a, 0x6666}, + {0x235b, 0x7fff}, {0x235c, 0x7fff}, {0x235d, 0x7fff}, {0x235e, 0x7fff}, + {0x235f, 0x7fff}, {0x2360, 0x7fff}, {0x2361, 0x7fff}, {0x2362, 0x1000}, + {0x2367, 0x0007}, {0x2368, 0x4000}, {0x2369, 0x0008}, {0x236a, 0x2000}, + {0x236b, 0x0009}, {0x236c, 0x003c}, {0x236d, 0x0000}, {0x236f, 0x2000}, + {0x2370, 0x0008}, {0x2371, 0x000a}, {0x2373, 0x0000}, {0x2374, 0x7fff}, + {0x2375, 0x7ff0}, {0x2376, 0x7990}, {0x2377, 0x7332}, {0x2379, 0x1000}, + {0x237a, 0x1000}, {0x2388, 0x7fff}, {0x2389, 0x6000}, {0x238a, 0x4000}, + {0x238b, 0x1000}, {0x238c, 0x1000}, {0x2398, 0x4668}, {0x23a1, 0x2000}, + {0x23a3, 0x4000}, {0x23ad, 0x2000}, {0x23ae, 0x2000}, {0x23af, 0x2000}, + {0x23b4, 0x2000}, {0x23b5, 0x2000}, {0x23b6, 0x2000}, {0x23bb, 0x6000}, + {0x2303, 0x0710}, {0x2304, 0x0332}, {0x230c, 0x0200}, {0x230d, 0x0080}, + {0x2310, 0x0010}, {0x22fb, 0x0000} +}; +#define RT5670_DSP_AEC_HANDSFREE_NUM \ + (sizeof(rt5670_dsp_aec_handsfree) / sizeof(rt5670_dsp_aec_handsfree[0])) + +/** + * rt5670_dsp_done - Wait until DSP is ready. + * @codec: SoC Audio Codec device. + * + * To check voice DSP status and confirm it's ready for next work. + * + * Returns 0 for success or negative error code. + */ +static int rt5670_dsp_done(struct snd_soc_codec *codec) +{ + unsigned int count = 0, dsp_val; + + dsp_val = snd_soc_read(codec, RT5670_DSP_CTRL1); + while (dsp_val & RT5670_DSP_BUSY_MASK) { + if (count > 10) + return -EBUSY; + dsp_val = snd_soc_read(codec, RT5670_DSP_CTRL1); + count++; + } + + return 0; +} + +/** + * rt5670_dsp_write - Write DSP register. + * @codec: SoC audio codec device. + * @param: DSP parameters. + * + * Modify voice DSP register for sound effect. The DSP can be controlled + * through DSP command format (0xfc), addr (0xc4), data (0xc5) and cmd (0xc6) + * register. It has to wait until the DSP is ready. + * + * Returns 0 for success or negative error code. + */ +static int rt5670_dsp_write(struct snd_soc_codec *codec, + unsigned int addr, unsigned int data) +{ + unsigned int dsp_val; + int ret; + + ret = snd_soc_write(codec, RT5670_DSP_CTRL2, addr); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP addr reg: %d\n", ret); + goto err; + } + ret = snd_soc_write(codec, RT5670_DSP_CTRL3, data); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP data reg: %d\n", ret); + goto err; + } + dsp_val = RT5670_DSP_I2C_AL_16 | RT5670_DSP_DL_2 | + RT5670_DSP_CMD_MW | DSP_CLK_RATE | RT5670_DSP_CMD_EN; + + ret = snd_soc_write(codec, RT5670_DSP_CTRL1, dsp_val); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP cmd reg: %d\n", ret); + goto err; + } + ret = rt5670_dsp_done(codec); + if (ret < 0) { + dev_err(codec->dev, "DSP is busy: %d\n", ret); + goto err; + } + + return 0; + +err: + return ret; +} + +/** + * rt5670_dsp_read - Read DSP register. + * @codec: SoC audio codec device. + * @reg: DSP register index. + * + * Read DSP setting value from voice DSP. The DSP can be controlled + * through DSP addr (0xc4), data (0xc5) and cmd (0xc6) register. Each + * command has to wait until the DSP is ready. + * + * Returns DSP register value or negative error code. + */ +static unsigned int rt5670_dsp_read( + struct snd_soc_codec *codec, unsigned int reg) +{ + unsigned int value; + unsigned int dsp_val; + int ret = 0; + + ret = rt5670_dsp_done(codec); + if (ret < 0) { + dev_err(codec->dev, "DSP is busy: %d\n", ret); + goto err; + } + + ret = snd_soc_write(codec, RT5670_DSP_CTRL2, reg); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP addr reg: %d\n", ret); + goto err; + } + dsp_val = RT5670_DSP_I2C_AL_16 | RT5670_DSP_DL_0 | RT5670_DSP_RW_MASK | + RT5670_DSP_CMD_MR | DSP_CLK_RATE | RT5670_DSP_CMD_EN; + + ret = snd_soc_write(codec, RT5670_DSP_CTRL1, dsp_val); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP cmd reg: %d\n", ret); + goto err; + } + + ret = rt5670_dsp_done(codec); + if (ret < 0) { + dev_err(codec->dev, "DSP is busy: %d\n", ret); + goto err; + } + + ret = snd_soc_write(codec, RT5670_DSP_CTRL2, 0x26); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP addr reg: %d\n", ret); + goto err; + } + dsp_val = RT5670_DSP_DL_1 | RT5670_DSP_CMD_RR | RT5670_DSP_RW_MASK | + DSP_CLK_RATE | RT5670_DSP_CMD_EN; + + ret = snd_soc_write(codec, RT5670_DSP_CTRL1, dsp_val); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP cmd reg: %d\n", ret); + goto err; + } + + ret = rt5670_dsp_done(codec); + if (ret < 0) { + dev_err(codec->dev, "DSP is busy: %d\n", ret); + goto err; + } + + ret = snd_soc_write(codec, RT5670_DSP_CTRL2, 0x25); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP addr reg: %d\n", ret); + goto err; + } + + dsp_val = RT5670_DSP_DL_1 | RT5670_DSP_CMD_RR | RT5670_DSP_RW_MASK | + DSP_CLK_RATE | RT5670_DSP_CMD_EN; + + ret = snd_soc_write(codec, RT5670_DSP_CTRL1, dsp_val); + if (ret < 0) { + dev_err(codec->dev, "Failed to write DSP cmd reg: %d\n", ret); + goto err; + } + + ret = rt5670_dsp_done(codec); + if (ret < 0) { + dev_err(codec->dev, "DSP is busy: %d\n", ret); + goto err; + } + + ret = snd_soc_read(codec, RT5670_DSP_CTRL5); + if (ret < 0) { + dev_err(codec->dev, "Failed to read DSP data reg: %d\n", ret); + goto err; + } + + value = ret; + return value; + +err: + return ret; +} + +static int rt5670_dsp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = rt5670->dsp_sw; + + return 0; +} + +static int rt5670_dsp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + if (rt5670->dsp_sw != ucontrol->value.integer.value[0]) + rt5670->dsp_sw = ucontrol->value.integer.value[0]; + + return 0; +} + +static int rt5670_dsp_rate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + switch (rt5670->dsp_rate) { + case 8000: + ucontrol->value.integer.value[0] = 0; + break; + case 16000: + ucontrol->value.integer.value[0] = 1; + break; + case 24000: + ucontrol->value.integer.value[0] = 2; + break; + case 44100: + ucontrol->value.integer.value[0] = 3; + break; + case 48000: + ucontrol->value.integer.value[0] = 4; + break; + default: + ucontrol->value.integer.value[0] = 1; + break; + } + + return 0; +} + +static int rt5670_dsp_rate_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + switch (ucontrol->value.integer.value[0]) { + case 0: + rt5670->dsp_rate = 8000; + break; + case 1: + rt5670->dsp_rate = 16000; + break; + case 2: + rt5670->dsp_rate = 24000; + break; + case 3: + rt5670->dsp_rate = 44100; + break; + case 4: + rt5670->dsp_rate = 48000; + break; + default: + pr_err("unsupported dsp rate, assume it is 16 KHz\n"); + rt5670->dsp_rate = 16000; + break; + } + + return 0; +} + +/* DSP Path Control 1 */ +static const char * const rt5670_src_rxdp_mode[] = { + "Normal", "Divided by 2", "Divided by 3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_src_rxdp_enum, RT5670_DSP_PATH1, + RT5670_RXDP_SRC_SFT, rt5670_src_rxdp_mode); + +static const char * const rt5670_src_txdp_mode[] = { + "Normal", "Multiplied by 2", "Multiplied by 3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_src_txdp_enum, RT5670_DSP_PATH1, + RT5670_TXDP_SRC_SFT, rt5670_src_txdp_mode); + +/* Sound Effect */ +static const char * const rt5670_dsp_mode[] = { + "Disable", "NS", "AEC", "VT", "VR", "FFP+NS", "48K-stereo+FFP", + "2MIC Handset", "2MIC Handsfree", "AEC Handsfree" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5670_dsp_enum, SND_SOC_NOPM, 0, + rt5670_dsp_mode); + +static const char * const rt5670_dsp_rate_mode[] = { + "8K", "16K", "24K", "44.1K", "48K" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5670_dsp_rate_enum, SND_SOC_NOPM, 0, + rt5670_dsp_rate_mode); + +static const struct snd_kcontrol_new rt5670_dsp_snd_controls[] = { + SOC_ENUM("SRC for RxDP", rt5670_src_rxdp_enum), + SOC_ENUM("SRC for TxDP", rt5670_src_txdp_enum), + SOC_ENUM_EXT("DSP Function Switch", rt5670_dsp_enum, + rt5670_dsp_get, rt5670_dsp_put), + SOC_ENUM_EXT("DSP Rate", rt5670_dsp_rate_enum, + rt5670_dsp_rate_get, rt5670_dsp_rate_put), +}; + +/** + * rt5670_dsp_conf - Set DSP basic setting. + * + * @codec: SoC audio codec device. + * + * Set parameters of basic setting to DSP. + * + * Returns 0 for success or negative error code. + */ +static int rt5670_dsp_conf(struct snd_soc_codec *codec) +{ + int ret, i; + + for (i = 0; i < RT5670_DSP_INIT_NUM; i++) { + ret = rt5670_dsp_write(codec, rt5670_dsp_init[i][0], + rt5670_dsp_init[i][1]); + if (ret < 0) { + dev_err(codec->dev, "Fail to config Dsp: %d\n", ret); + goto conf_err; + } + } + + return 0; + +conf_err: + + return ret; +} + +/** + * rt5670_dsp_rate - Set DSP rate setting. + * + * @codec: SoC audio codec device. + * @sys_clk: System clock. + * @srate: Sampling rate. + * + * Set parameters of system clock and sampling rate to DSP. + * + * Returns 0 for success or negative error code. + */ +static int rt5670_dsp_rate(struct snd_soc_codec *codec, int sys_clk, + int srate) +{ + int ret, i, tab_num; + unsigned short (*rate_tab)[2]; + + dev_dbg(codec->dev, "%s sys:%d srate:%d\n", __func__, sys_clk, srate); + + switch (sys_clk) { + case 4096000: + rate_tab = rt5670_dsp_4096000; + tab_num = RT5670_DSP_4096000_NUM; + break; + case 11289600: + rate_tab = rt5670_dsp_11289600; + tab_num = RT5670_DSP_11289600_NUM; + break; + case 12288000: + rate_tab = rt5670_dsp_12288000; + tab_num = RT5670_DSP_12288000_NUM; + break; + case 24576000: + rate_tab = rt5670_dsp_24576000; + tab_num = RT5670_DSP_24576000_NUM; + break; + default: + return -EINVAL; + break; + } + + for (i = 0; i < tab_num; i++) { + ret = rt5670_dsp_write(codec, rate_tab[i][0], rate_tab[i][1]); + if (ret < 0) + goto rate_err; + } + + switch (srate) { + case 8000: + rate_tab = rt5670_dsp_8; + tab_num = RT5670_DSP_8_NUM; + break; + case 16000: + rate_tab = rt5670_dsp_16; + tab_num = RT5670_DSP_16_NUM; + break; + case 24000: + rate_tab = rt5670_dsp_24; + tab_num = RT5670_DSP_24_NUM; + break; + case 44100: + case 48000: + rate_tab = rt5670_dsp_48_441; + tab_num = RT5670_DSP_48_441_NUM; + break; + default: + return -EINVAL; + break; + } + + for (i = 0; i < tab_num; i++) { + ret = rt5670_dsp_write(codec, rate_tab[i][0], rate_tab[i][1]); + if (ret < 0) + goto rate_err; + } + + return 0; + +rate_err: + + dev_err(codec->dev, "Fail to set rate parameters: %d\n", ret); + return ret; +} + +/** + * rt5670_dsp_set_mode - Set DSP mode parameters. + * + * @codec: SoC audio codec device. + * @mode: DSP mode. + * + * Set parameters of mode to DSP. + * There are three modes which includes " mic AEC + NS + FENS", + * "HFBF" and "Far-field pickup". + * + * Returns 0 for success or negative error code. + */ +static int rt5670_dsp_set_mode(struct snd_soc_codec *codec, int mode) +{ + int ret, i, tab_num; + unsigned short (*mode_tab)[2]; + + switch (mode) { + case RT5670_DSP_NS: + dev_info(codec->dev, "NS\n"); + mode_tab = rt5670_dsp_ns; + tab_num = RT5670_DSP_NS_NUM; + break; + + case RT5670_DSP_AEC: + dev_info(codec->dev, "AEC\n"); + mode_tab = rt5670_dsp_aec; + tab_num = RT5670_DSP_AEC_NUM; + break; + + case RT5670_DSP_VT: + dev_info(codec->dev, "VT\n"); + mode_tab = rt5670_dsp_vt; + tab_num = RT5670_DSP_VT_NUM; + break; + + case RT5670_DSP_VR: + dev_info(codec->dev, "VR\n"); + mode_tab = rt5670_dsp_vr; + tab_num = RT5670_DSP_VR_NUM; + break; + + case RT5670_DSP_FFP_NS: + dev_info(codec->dev, "FFP_NS\n"); + mode_tab = rt5670_dsp_ffp_ns; + tab_num = RT5670_DSP_FFP_NS_NUM; + break; + + case RT5670_DSP_48K_STO_FFP: + dev_info(codec->dev, "48K_STO_FFP\n"); + mode_tab = rt5670_dsp_48k_sto_ffp; + tab_num = RT5670_DSP_48K_STO_FFP_NUM; + break; + + case RT5670_DSP_2MIC_HANDSET: + dev_info(codec->dev, "2MIC_HANDSET\n"); + mode_tab = rt5670_dsp_2mic_handset; + tab_num = RT5670_DSP_2MIC_HANDSET_NUM; + break; + + case RT5670_DSP_2MIC_HANDSFREE: + dev_info(codec->dev, "2MIC_HANDSFREE\n"); + mode_tab = rt5670_dsp_2mic_handsfree; + tab_num = RT5670_DSP_2MIC_HANDSFREE_NUM; + break; + + case RT5670_DSP_AEC_HANDSFREE: + dev_info(codec->dev, "AEC_HANDSFREE\n"); + mode_tab = rt5670_dsp_aec_handsfree; + tab_num = RT5670_DSP_AEC_HANDSFREE_NUM; + break; + + case RT5670_DSP_DIS: + default: + dev_info(codec->dev, "Disable\n"); + return 0; + } + + for (i = 0; i < tab_num; i++) { + ret = rt5670_dsp_write(codec, mode_tab[i][0], mode_tab[i][1]); + if (ret < 0) + goto mode_err; + } + + return 0; + +mode_err: + + dev_err(codec->dev, "Fail to set mode %d parameters: %d\n", mode, ret); + return ret; +} + +/** + * rt5670_dsp_snd_effect - Set DSP sound effect. + * + * Set parameters of sound effect to DSP. + * + * Returns 0 for success or negative error code. + */ +static int rt5670_dsp_snd_effect(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + int ret; + + snd_soc_update_bits(codec, RT5670_DIG_MISC, RT5670_RST_DSP, + RT5670_RST_DSP); + snd_soc_update_bits(codec, RT5670_DIG_MISC, RT5670_RST_DSP, 0); + + mdelay(10); + + ret = rt5670_dsp_rate(codec, + rt5670->sysclk ? + rt5670->sysclk : 11289600, + rt5670->dsp_rate ? + rt5670->dsp_rate : 16000); + if (ret < 0) + goto effect_err; + + ret = rt5670_dsp_conf(codec); + if (ret < 0) + goto effect_err; + + ret = rt5670_dsp_set_mode(codec, rt5670->dsp_sw); + if (ret < 0) + goto effect_err; + + return 0; + +effect_err: + + return ret; +} + +static int rt5670_dsp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + dev_dbg(codec->dev, "%s(): PMD\n", __func__); + rt5670_dsp_write(codec, 0x22f9, 1); + break; + + case SND_SOC_DAPM_POST_PMU: + dev_dbg(codec->dev, "%s(): PMU\n", __func__); + rt5670_dsp_snd_effect(codec); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5670_dsp_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY_S("Voice DSP", 1, SND_SOC_NOPM, + 0, 0, rt5670_dsp_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA("DSP Downstream", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DSP Upstream", SND_SOC_NOPM, + 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route rt5670_dsp_dapm_routes[] = { + {"DSP Downstream", NULL, "Voice DSP"}, + {"DSP Downstream", NULL, "RxDP Mux"}, + {"DSP Upstream", NULL, "Voice DSP"}, + {"DSP Upstream", NULL, "8CH TDM Data"}, + {"DSP DL Mux", "DSP", "DSP Downstream"}, + {"DSP UL Mux", "DSP", "DSP Upstream"}, +}; + +/** + * rt5670_dsp_probe - register DSP for rt5670 + * @codec: audio codec + * + * To register DSP function for rt5670. + * + * Returns 0 for success or negative error code. + */ +int rt5670_dsp_probe(struct snd_soc_codec *codec) +{ + if (codec == NULL) + return -EINVAL; + + snd_soc_update_bits(codec, RT5670_PWR_DIG2, + RT5670_PWR_I2S_DSP, RT5670_PWR_I2S_DSP); + + snd_soc_update_bits(codec, RT5670_DIG_MISC, RT5670_RST_DSP, + RT5670_RST_DSP); + snd_soc_update_bits(codec, RT5670_DIG_MISC, RT5670_RST_DSP, 0); + + mdelay(10); + + rt5670_dsp_write(codec, 0x22fb, 0); + /* power down DSP*/ + rt5670_dsp_write(codec, 0x22f9, 1); + + snd_soc_update_bits(codec, RT5670_PWR_DIG2, + RT5670_PWR_I2S_DSP, 0); + + snd_soc_add_codec_controls(codec, rt5670_dsp_snd_controls, + ARRAY_SIZE(rt5670_dsp_snd_controls)); + snd_soc_dapm_new_controls(&codec->dapm, rt5670_dsp_dapm_widgets, + ARRAY_SIZE(rt5670_dsp_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, rt5670_dsp_dapm_routes, + ARRAY_SIZE(rt5670_dsp_dapm_routes)); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5670_dsp_probe); + diff --git a/sound/soc/codecs/rt5670-dsp.h b/sound/soc/codecs/rt5670-dsp.h new file mode 100644 index 0000000..784544c --- /dev/null +++ b/sound/soc/codecs/rt5670-dsp.h @@ -0,0 +1,69 @@ +/* + * rt5670-dsp.h -- RT5670 ALSA SoC DSP 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 __RT5670_DSP_H__ +#define __RT5670_DSP_H__ + +#define RT5670_DSP_CTRL1 0xe0 +#define RT5670_DSP_CTRL2 0xe1 +#define RT5670_DSP_CTRL3 0xe2 +#define RT5670_DSP_CTRL4 0xe3 +#define RT5670_DSP_CTRL5 0xe4 + +/* DSP Control 1 (0xe0) */ +#define RT5670_DSP_CMD_MASK (0xff << 8) +#define RT5670_DSP_CMD_PE (0x0d << 8) /* Patch Entry */ +#define RT5670_DSP_CMD_MW (0x3b << 8) /* Memory Write */ +#define RT5670_DSP_CMD_MR (0x37 << 8) /* Memory Read */ +#define RT5670_DSP_CMD_RR (0x60 << 8) /* Register Read */ +#define RT5670_DSP_CMD_RW (0x68 << 8) /* Register Write */ +#define RT5670_DSP_REG_DATHI (0x26 << 8) /* High Data Addr */ +#define RT5670_DSP_REG_DATLO (0x25 << 8) /* Low Data Addr */ +#define RT5670_DSP_CLK_MASK (0x3 << 6) +#define RT5670_DSP_CLK_SFT 6 +#define RT5670_DSP_CLK_768K (0x0 << 6) +#define RT5670_DSP_CLK_384K (0x1 << 6) +#define RT5670_DSP_CLK_192K (0x2 << 6) +#define RT5670_DSP_CLK_96K (0x3 << 6) +#define RT5670_DSP_BUSY_MASK (0x1 << 5) +#define RT5670_DSP_RW_MASK (0x1 << 4) +#define RT5670_DSP_DL_MASK (0x3 << 2) +#define RT5670_DSP_DL_0 (0x0 << 2) +#define RT5670_DSP_DL_1 (0x1 << 2) +#define RT5670_DSP_DL_2 (0x2 << 2) +#define RT5670_DSP_DL_3 (0x3 << 2) +#define RT5670_DSP_I2C_AL_16 (0x1 << 1) +#define RT5670_DSP_CMD_EN (0x1) + +enum { + RT5670_DSP_DIS, + RT5670_DSP_NS, + RT5670_DSP_AEC, + RT5670_DSP_VT, + RT5670_DSP_VR, + RT5670_DSP_FFP_NS, + RT5670_DSP_48K_STO_FFP, + RT5670_DSP_2MIC_HANDSET, + RT5670_DSP_2MIC_HANDSFREE, + RT5670_DSP_AEC_HANDSFREE, +}; + +struct rt5670_dsp_param { + u16 cmd_fmt; + u16 addr; + u16 data; + u8 cmd; +}; + +int rt5670_dsp_probe(struct snd_soc_codec *codec); + +#endif /* __RT5670_DSP_H__ */ + diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c new file mode 100644 index 0000000..b1a8517 --- /dev/null +++ b/sound/soc/codecs/rt5670.c @@ -0,0 +1,3053 @@ +/* + * rt5670.c -- RT5670 ALSA SoC audio codec driver + * + * Copyright 2012 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 <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rt5670.h" +#include "rt5670-dsp.h" + +#define RT5670_DEVICE_ID 0x6271 + +#define RT5670_PR_RANGE_BASE (0xff + 1) +#define RT5670_PR_SPACING 0x100 + +#define RT5670_PR_BASE (RT5670_PR_RANGE_BASE + (0 * RT5670_PR_SPACING)) + +static const struct regmap_range_cfg rt5670_ranges[] = { + { .name = "PR", .range_min = RT5670_PR_BASE, + .range_max = RT5670_PR_BASE + 0xf8, + .selector_reg = RT5670_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT5670_PRIV_DATA, + .window_len = 0x1, }, +}; + +static struct reg_default init_list[] = { + { RT5670_PR_BASE + 0x14, 0x9a8a }, + { RT5670_PR_BASE + 0x3d, 0x3640 }, +}; +#define RT5670_INIT_REG_LEN ARRAY_SIZE(init_list) + +static const struct reg_default rt5670_reg[] = { + { 0x00, 0x0000 }, + { 0x02, 0x8888 }, + { 0x03, 0x8888 }, + { 0x0a, 0x0001 }, + { 0x0b, 0x0827 }, + { 0x0c, 0x0000 }, + { 0x0d, 0x0008 }, + { 0x0e, 0x0000 }, + { 0x0f, 0x0808 }, + { 0x19, 0xafaf }, + { 0x1a, 0xafaf }, + { 0x1b, 0x0011 }, + { 0x1c, 0x2f2f }, + { 0x1d, 0x2f2f }, + { 0x1e, 0x0000 }, + { 0x1f, 0x2f2f }, + { 0x20, 0x0000 }, + { 0x26, 0x7860 }, + { 0x27, 0x7860 }, + { 0x28, 0x7871 }, + { 0x29, 0x8080 }, + { 0x2a, 0x5656 }, + { 0x2b, 0x5454 }, + { 0x2c, 0xaaa0 }, + { 0x2d, 0x0000 }, + { 0x2e, 0x2f2f }, + { 0x2f, 0x1002 }, + { 0x30, 0x0000 }, + { 0x31, 0x5f00 }, + { 0x32, 0x0000 }, + { 0x33, 0x0000 }, + { 0x34, 0x0000 }, + { 0x35, 0x0000 }, + { 0x36, 0x0000 }, + { 0x37, 0x0000 }, + { 0x38, 0x0000 }, + { 0x3b, 0x0000 }, + { 0x3c, 0x007f }, + { 0x3d, 0x0000 }, + { 0x3e, 0x007f }, + { 0x45, 0xe00f }, + { 0x4c, 0x5380 }, + { 0x4f, 0x0073 }, + { 0x52, 0x00d3 }, + { 0x53, 0xf0f0 }, + { 0x61, 0x0000 }, + { 0x62, 0x0001 }, + { 0x63, 0x00c3 }, + { 0x64, 0x0000 }, + { 0x65, 0x0000 }, + { 0x66, 0x0000 }, + { 0x6f, 0x8000 }, + { 0x70, 0x8000 }, + { 0x71, 0x8000 }, + { 0x72, 0x8000 }, + { 0x73, 0x1110 }, + { 0x74, 0x0e00 }, + { 0x75, 0x1505 }, + { 0x76, 0x0015 }, + { 0x77, 0x0c00 }, + { 0x78, 0x4000 }, + { 0x79, 0x0123 }, + { 0x7f, 0x1100 }, + { 0x80, 0x0000 }, + { 0x81, 0x0000 }, + { 0x82, 0x0000 }, + { 0x83, 0x0000 }, + { 0x84, 0x0000 }, + { 0x85, 0x0000 }, + { 0x86, 0x0008 }, + { 0x87, 0x0000 }, + { 0x88, 0x0000 }, + { 0x89, 0x0000 }, + { 0x8a, 0x0000 }, + { 0x8b, 0x0000 }, + { 0x8c, 0x0007 }, + { 0x8d, 0x0000 }, + { 0x8e, 0x0004 }, + { 0x8f, 0x1100 }, + { 0x90, 0x0646 }, + { 0x91, 0x0c06 }, + { 0x93, 0x0000 }, + { 0x94, 0x0000 }, + { 0x95, 0x0000 }, + { 0x97, 0x0000 }, + { 0x98, 0x0000 }, + { 0x99, 0x0000 }, + { 0x9a, 0x2184 }, + { 0x9b, 0x010a }, + { 0x9c, 0x0aea }, + { 0x9d, 0x000c }, + { 0x9e, 0x0400 }, + { 0xae, 0x7000 }, + { 0xaf, 0x0000 }, + { 0xb0, 0x6000 }, + { 0xb1, 0x0000 }, + { 0xb2, 0x0000 }, + { 0xb3, 0x001f }, + { 0xb4, 0x2206 }, + { 0xb5, 0x1f00 }, + { 0xb6, 0x0000 }, + { 0xb7, 0x0000 }, + { 0xbb, 0x0000 }, + { 0xbc, 0x0000 }, + { 0xbd, 0x0000 }, + { 0xbe, 0x0000 }, + { 0xbf, 0x0000 }, + { 0xc0, 0x0000 }, + { 0xc1, 0x0000 }, + { 0xc2, 0x0000 }, + { 0xcd, 0x0000 }, + { 0xce, 0x0000 }, + { 0xcf, 0x1813 }, + { 0xd0, 0x0690 }, + { 0xd1, 0x1c17 }, + { 0xd3, 0xb320 }, + { 0xd4, 0x0000 }, + { 0xd6, 0x0400 }, + { 0xd9, 0x0809 }, + { 0xda, 0x0000 }, + { 0xdb, 0x0001 }, + { 0xdc, 0x0049 }, + { 0xdd, 0x0009 }, + { 0xe6, 0x8000 }, + { 0xe7, 0x0000 }, + { 0xec, 0xb300 }, + { 0xed, 0x0000 }, + { 0xee, 0xb300 }, + { 0xef, 0x0000 }, + { 0xf8, 0x0000 }, + { 0xf9, 0x0000 }, + { 0xfa, 0x8010 }, + { 0xfb, 0x0033 }, + { 0xfc, 0x0080 }, +}; + +unsigned rt5670_pdm1_read(struct snd_soc_codec *codec, unsigned int reg) +{ + int ret; + + ret = snd_soc_write(codec, RT5670_PDM1_DATA_CTRL2, reg<<8); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + ret = snd_soc_write(codec, RT5670_PDM_DATA_CTRL1, 0x0200); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + do + ret = snd_soc_read(codec, RT5670_PDM_DATA_CTRL1); + while (ret & 0x0100); + return snd_soc_read(codec, RT5670_PDM1_DATA_CTRL4); +err: + return ret; +} +EXPORT_SYMBOL(rt5670_pdm1_read); + +int rt5670_pdm1_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + int ret; + + ret = snd_soc_write(codec, RT5670_PDM1_DATA_CTRL3, value); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + goto err; + } + ret = snd_soc_write(codec, RT5670_PDM1_DATA_CTRL2, reg<<8); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + ret = snd_soc_write(codec, RT5670_PDM_DATA_CTRL1, 0x0600); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + ret = snd_soc_write(codec, RT5670_PDM_DATA_CTRL1, 0x3600); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + do + ret = snd_soc_read(codec, RT5670_PDM_DATA_CTRL1); + while (ret & 0x0100); + return 0; + +err: + return ret; +} +EXPORT_SYMBOL(rt5670_pdm1_write); + +unsigned rt5670_pdm2_read(struct snd_soc_codec *codec, unsigned int reg) +{ + int ret; + + ret = snd_soc_write(codec, RT5670_PDM2_DATA_CTRL2, reg<<8); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + ret = snd_soc_write(codec, RT5670_PDM_DATA_CTRL1, 0x0002); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + do + ret = snd_soc_read(codec, RT5670_PDM_DATA_CTRL1); + while (ret & 0x0001); + return snd_soc_read(codec, RT5670_PDM2_DATA_CTRL4); +err: + return ret; +} +EXPORT_SYMBOL(rt5670_pdm2_read); + +int rt5670_pdm2_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + int ret; + + ret = snd_soc_write(codec, RT5670_PDM2_DATA_CTRL3, value); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + goto err; + } + ret = snd_soc_write(codec, RT5670_PDM2_DATA_CTRL2, reg<<8); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + ret = snd_soc_write(codec, RT5670_PDM_DATA_CTRL1, 0x0006); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + ret = snd_soc_write(codec, RT5670_PDM_DATA_CTRL1, 0x0036); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + do + ret = snd_soc_read(codec, RT5670_PDM_DATA_CTRL1); + while (ret & 0x0001); + + return 0; + +err: + return ret; +} +EXPORT_SYMBOL(rt5670_pdm2_write); + +static bool rt5670_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5670_ranges); i++) + if ((reg >= rt5670_ranges[i].window_start && + reg <= rt5670_ranges[i].window_start + + rt5670_ranges[i].window_len) || + (reg >= rt5670_ranges[i].range_min && + reg <= rt5670_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5670_RESET: + case RT5670_PDM_DATA_CTRL1: + case RT5670_PDM1_DATA_CTRL4: + case RT5670_PDM2_DATA_CTRL4: + case RT5670_PRIV_DATA: + case RT5670_ASRC_5: + case RT5670_CJ_CTRL1: + case RT5670_CJ_CTRL2: + case RT5670_CJ_CTRL3: + case RT5670_A_JD_CTRL1: + case RT5670_A_JD_CTRL2: + case RT5670_VAD_CTRL5: + case RT5670_ADC_EQ_CTRL1: + case RT5670_EQ_CTRL1: + case RT5670_ALC_CTRL_1: + case RT5670_IRQ_CTRL2: + case RT5670_IRQ_CTRL3: + case RT5670_INT_IRQ_ST: + case RT5670_IL_CMD: + case RT5670_DSP_CTRL1: + case RT5670_DSP_CTRL2: + case RT5670_DSP_CTRL3: + case RT5670_DSP_CTRL4: + case RT5670_DSP_CTRL5: + case RT5670_VENDOR_ID: + case RT5670_VENDOR_ID1: + case RT5670_VENDOR_ID2: + return true; + default: + return false; + } +} + +static bool rt5670_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5670_ranges); i++) + if ((reg >= rt5670_ranges[i].window_start && + reg <= rt5670_ranges[i].window_start + + rt5670_ranges[i].window_len) || + (reg >= rt5670_ranges[i].range_min && + reg <= rt5670_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5670_RESET: + case RT5670_HP_VOL: + case RT5670_LOUT1: + case RT5670_CJ_CTRL1: + case RT5670_CJ_CTRL2: + case RT5670_CJ_CTRL3: + case RT5670_IN2: + case RT5670_INL1_INR1_VOL: + case RT5670_DAC1_DIG_VOL: + case RT5670_DAC2_DIG_VOL: + case RT5670_DAC_CTRL: + case RT5670_STO1_ADC_DIG_VOL: + case RT5670_MONO_ADC_DIG_VOL: + case RT5670_STO2_ADC_DIG_VOL: + case RT5670_ADC_BST_VOL1: + case RT5670_ADC_BST_VOL2: + case RT5670_STO2_ADC_MIXER: + case RT5670_STO1_ADC_MIXER: + case RT5670_MONO_ADC_MIXER: + case RT5670_AD_DA_MIXER: + case RT5670_STO_DAC_MIXER: + case RT5670_DD_MIXER: + case RT5670_DIG_MIXER: + case RT5670_DSP_PATH1: + case RT5670_DSP_PATH2: + case RT5670_DIG_INF1_DATA: + case RT5670_DIG_INF2_DATA: + case RT5670_PDM_OUT_CTRL: + case RT5670_PDM_DATA_CTRL1: + case RT5670_PDM1_DATA_CTRL2: + case RT5670_PDM1_DATA_CTRL3: + case RT5670_PDM1_DATA_CTRL4: + case RT5670_PDM2_DATA_CTRL2: + case RT5670_PDM2_DATA_CTRL3: + case RT5670_PDM2_DATA_CTRL4: + case RT5670_REC_L1_MIXER: + case RT5670_REC_L2_MIXER: + case RT5670_REC_R1_MIXER: + case RT5670_REC_R2_MIXER: + case RT5670_HPO_MIXER: + case RT5670_MONO_MIXER: + case RT5670_OUT_L1_MIXER: + case RT5670_OUT_R1_MIXER: + case RT5670_LOUT_MIXER: + case RT5670_PWR_DIG1: + case RT5670_PWR_DIG2: + case RT5670_PWR_ANLG1: + case RT5670_PWR_ANLG2: + case RT5670_PWR_MIXER: + case RT5670_PWR_VOL: + case RT5670_PRIV_INDEX: + case RT5670_PRIV_DATA: + case RT5670_I2S4_SDP: + case RT5670_I2S1_SDP: + case RT5670_I2S2_SDP: + case RT5670_I2S3_SDP: + case RT5670_ADDA_CLK1: + case RT5670_ADDA_CLK2: + case RT5670_DMIC_CTRL1: + case RT5670_DMIC_CTRL2: + case RT5670_TDM_CTRL_1: + case RT5670_TDM_CTRL_2: + case RT5670_TDM_CTRL_3: + case RT5670_DSP_CLK: + case RT5670_GLB_CLK: + case RT5670_PLL_CTRL1: + case RT5670_PLL_CTRL2: + case RT5670_ASRC_1: + case RT5670_ASRC_2: + case RT5670_ASRC_3: + case RT5670_ASRC_4: + case RT5670_ASRC_5: + case RT5670_ASRC_7: + case RT5670_ASRC_8: + case RT5670_ASRC_9: + case RT5670_ASRC_10: + case RT5670_ASRC_11: + case RT5670_ASRC_12: + case RT5670_ASRC_13: + case RT5670_ASRC_14: + case RT5670_DEPOP_M1: + case RT5670_DEPOP_M2: + case RT5670_DEPOP_M3: + case RT5670_CHARGE_PUMP: + case RT5670_MICBIAS: + case RT5670_A_JD_CTRL1: + case RT5670_A_JD_CTRL2: + case RT5670_VAD_CTRL1: + case RT5670_VAD_CTRL2: + case RT5670_VAD_CTRL3: + case RT5670_VAD_CTRL4: + case RT5670_VAD_CTRL5: + case RT5670_ADC_EQ_CTRL1: + case RT5670_ADC_EQ_CTRL2: + case RT5670_EQ_CTRL1: + case RT5670_EQ_CTRL2: + case RT5670_ALC_DRC_CTRL1: + case RT5670_ALC_DRC_CTRL2: + case RT5670_ALC_CTRL_1: + case RT5670_ALC_CTRL_2: + case RT5670_ALC_CTRL_3: + case RT5670_JD_CTRL: + case RT5670_IRQ_CTRL1: + case RT5670_IRQ_CTRL2: + case RT5670_IRQ_CTRL3: + case RT5670_INT_IRQ_ST: + case RT5670_GPIO_CTRL1: + case RT5670_GPIO_CTRL2: + case RT5670_GPIO_CTRL3: + case RT5670_SCRABBLE_FUN: + case RT5670_SCRABBLE_CTRL: + case RT5670_BASE_BACK: + case RT5670_MP3_PLUS1: + case RT5670_MP3_PLUS2: + case RT5670_ADJ_HPF1: + case RT5670_ADJ_HPF2: + case RT5670_HP_CALIB_AMP_DET: + case RT5670_SV_ZCD1: + case RT5670_SV_ZCD2: + case RT5670_IL_CMD: + case RT5670_IL_CMD2: + case RT5670_IL_CMD3: + case RT5670_DRC_HL_CTRL1: + case RT5670_DRC_HL_CTRL2: + case RT5670_ADC_MONO_HP_CTRL1: + case RT5670_ADC_MONO_HP_CTRL2: + case RT5670_ADC_STO2_HP_CTRL1: + case RT5670_ADC_STO2_HP_CTRL2: + case RT5670_JD_CTRL3: + case RT5670_JD_CTRL4: + case RT5670_DIG_MISC: + case RT5670_DSP_CTRL1: + case RT5670_DSP_CTRL2: + case RT5670_DSP_CTRL3: + case RT5670_DSP_CTRL4: + case RT5670_DSP_CTRL5: + case RT5670_GEN_CTRL2: + case RT5670_GEN_CTRL3: + case RT5670_VENDOR_ID: + case RT5670_VENDOR_ID1: + case RT5670_VENDOR_ID2: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static unsigned int bst_tlv[] = { + TLV_DB_RANGE_HEAD(7), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), +}; + +/* Interface data select */ +static const char * const rt5670_data_select[] = { + "Normal", "Swap", "left copy to right", "right copy to left" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5670_if2_dac_enum, RT5670_DIG_INF1_DATA, + RT5670_IF2_DAC_SEL_SFT, rt5670_data_select); + +static const SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_enum, RT5670_DIG_INF1_DATA, + RT5670_IF2_ADC_SEL_SFT, rt5670_data_select); + +static const struct snd_kcontrol_new rt5670_snd_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE("HP Playback Switch", RT5670_HP_VOL, + RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("HP Playback Volume", RT5670_HP_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 39, 0, out_vol_tlv), + /* OUTPUT Control */ + SOC_DOUBLE("OUT Playback Switch", RT5670_LOUT1, + RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("OUT Channel Switch", RT5670_LOUT1, + RT5670_VOL_L_SFT, RT5670_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("OUT Playback Volume", RT5670_LOUT1, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, 39, 1, out_vol_tlv), + /* DAC Digital Volume */ + SOC_DOUBLE("DAC2 Playback Switch", RT5670_DAC_CTRL, + RT5670_M_DAC_L2_VOL_SFT, RT5670_M_DAC_R2_VOL_SFT, 1, 1), + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5670_DAC1_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 175, 0, dac_vol_tlv), + SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5670_DAC2_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 175, 0, dac_vol_tlv), + /* IN1/IN2 Control */ + SOC_SINGLE_TLV("IN1 Boost", RT5670_CJ_CTRL1, + RT5670_BST_SFT1, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN2 Boost", RT5670_IN2, + RT5670_BST_SFT1, 8, 0, bst_tlv), + /* INL/INR Volume Control */ + SOC_DOUBLE_TLV("IN Capture Volume", RT5670_INL1_INR1_VOL, + RT5670_INL_VOL_SFT, RT5670_INR_VOL_SFT, + 31, 1, in_vol_tlv), + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5670_STO1_ADC_DIG_VOL, + RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5670_STO1_ADC_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 127, 0, adc_vol_tlv), + + SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5670_MONO_ADC_DIG_VOL, + RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, + 127, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain", RT5670_ADC_BST_VOL1, + RT5670_STO1_ADC_L_BST_SFT, RT5670_STO1_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), + + SOC_DOUBLE_TLV("STO2 ADC Boost Gain", RT5670_ADC_BST_VOL1, + RT5670_STO2_ADC_L_BST_SFT, RT5670_STO2_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), + + SOC_ENUM("ADC IF2 Data Switch", rt5670_if2_adc_enum), + SOC_ENUM("DAC IF2 Data Switch", rt5670_if2_dac_enum), +}; + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + int div[] = {2, 3, 4, 6, 8, 12}; + int idx = -EINVAL, i; + int rate, red, bound, temp; + + rate = rt5670->lrck[RT5670_AIF1] << 8; + red = 2000000 * 12; + for (i = 0; i < ARRAY_SIZE(div); i++) { + bound = div[i] * 2000000; + if (rate > bound) + continue; + temp = bound - rate; + if (temp < red) { + red = temp; + idx = i; + } + } + if (rt5670->pdata.asrc_en) + idx = 5; + + if (idx < 0) + dev_err(codec->dev, "Failed to set DMIC clock\n"); + else + snd_soc_update_bits(codec, RT5670_DMIC_CTRL1, + RT5670_DMIC_CLK_MASK, idx << RT5670_DMIC_CLK_SFT); + return idx; +} + +static int check_sysclk1_source(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + + val = snd_soc_read(source->codec, RT5670_GLB_CLK); + val &= RT5670_SCLK_SRC_MASK; + if (val == RT5670_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5670_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO1_ADC_MIXER, + RT5670_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto2_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto2_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_STO2_ADC_MIXER, + RT5670_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5670_MONO_ADC_MIXER, + RT5670_M_MONO_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER, + RT5670_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER, + RT5670_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5670_AD_DA_MIXER, + RT5670_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_AD_DA_MIXER, + RT5670_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_L2_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_R2_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_STO_DAC_MIXER, + RT5670_M_DAC_L1_STO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_L1_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_L2_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_R2_MONO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_mono_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_R1_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_R2_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DD_MIXER, + RT5670_M_DAC_L2_MONO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dig_l_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5670_DIG_MIXER, + RT5670_M_STO_L_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_L2_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_R2_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_dig_r_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5670_DIG_MIXER, + RT5670_M_STO_R_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_R2_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_DIG_MIXER, + RT5670_M_DAC_L2_DAC_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5670_rec_l_mix[] = { + SOC_DAPM_SINGLE("INL Switch", RT5670_REC_L2_MIXER, + RT5670_M_IN_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5670_REC_L2_MIXER, + RT5670_M_BST2_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5670_REC_L2_MIXER, + RT5670_M_BST1_RM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_rec_r_mix[] = { + SOC_DAPM_SINGLE("INR Switch", RT5670_REC_R2_MIXER, + RT5670_M_IN_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5670_REC_R2_MIXER, + RT5670_M_BST2_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5670_REC_R2_MIXER, + RT5670_M_BST1_RM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_out_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", RT5670_OUT_L1_MIXER, + RT5670_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5670_OUT_L1_MIXER, + RT5670_M_IN_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5670_OUT_L1_MIXER, + RT5670_M_DAC_L2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_OUT_L1_MIXER, + RT5670_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5670_OUT_R1_MIXER, + RT5670_M_BST2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5670_OUT_R1_MIXER, + RT5670_M_IN_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5670_OUT_R1_MIXER, + RT5670_M_DAC_R2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_OUT_R1_MIXER, + RT5670_M_DAC_R1_OM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpo_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER, + RT5670_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPVOL Switch", RT5670_HPO_MIXER, + RT5670_M_HPVOL_HM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpvoll_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACL1_HML_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5670_HPO_MIXER, + RT5670_M_INL1_HML_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpvolr_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACR1_HMR_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5670_HPO_MIXER, + RT5670_M_INR1_HMR_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_lout_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_LOUT_MIXER, + RT5670_M_DAC_L1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_LOUT_MIXER, + RT5670_M_DAC_R1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX L Switch", RT5670_LOUT_MIXER, + RT5670_M_OV_L_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX R Switch", RT5670_LOUT_MIXER, + RT5670_M_OV_R_LM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpl_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACL1_HML_SFT, 1, 1), + SOC_DAPM_SINGLE("INL1 Switch", RT5670_HPO_MIXER, + RT5670_M_INL1_HML_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5670_hpr_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5670_HPO_MIXER, + RT5670_M_DACR1_HMR_SFT, 1, 1), + SOC_DAPM_SINGLE("INR1 Switch", RT5670_HPO_MIXER, + RT5670_M_INR1_HMR_SFT, 1, 1), +}; + +/* DAC1 L/R source */ /* MX-29 [9:8] [11:10] */ +static const char * const rt5670_dac1_src[] = { + "IF1 DAC", "IF2 DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac1l_enum, RT5670_AD_DA_MIXER, + RT5670_DAC1_L_SEL_SFT, rt5670_dac1_src); + +static const struct snd_kcontrol_new rt5670_dac1l_mux = + SOC_DAPM_ENUM("DAC1 L source", rt5670_dac1l_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac1r_enum, RT5670_AD_DA_MIXER, + RT5670_DAC1_R_SEL_SFT, rt5670_dac1_src); + +static const struct snd_kcontrol_new rt5670_dac1r_mux = + SOC_DAPM_ENUM("DAC1 R source", rt5670_dac1r_enum); + +/*DAC2 L/R source*/ /* MX-1B [6:4] [2:0] */ +/* TODO Use SOC_VALUE_ENUM_SINGLE_DECL */ +static const char * const rt5670_dac12_src[] = { + "IF1 DAC", "IF2 DAC", "IF3 DAC", "TxDC DAC", + "Bass", "VAD_ADC", "IF4 DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac2l_enum, RT5670_DAC_CTRL, + RT5670_DAC2_L_SEL_SFT, rt5670_dac12_src); + +static const struct snd_kcontrol_new rt5670_dac_l2_mux = + SOC_DAPM_ENUM("DAC2 L source", rt5670_dac2l_enum); + +static const char * const rt5670_dacr2_src[] = { + "IF1 DAC", "IF2 DAC", "IF3 DAC", "TxDC DAC", "TxDP ADC", "IF4 DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dac2r_enum, RT5670_DAC_CTRL, + RT5670_DAC2_R_SEL_SFT, rt5670_dacr2_src); + +static const struct snd_kcontrol_new rt5670_dac_r2_mux = + SOC_DAPM_ENUM("DAC2 R source", rt5670_dac2r_enum); + +/*RxDP source*/ /* MX-2D [15:13] */ +static const char * const rt5670_rxdp_src[] = { + "IF2 DAC", "IF1 DAC", "STO1 ADC Mixer", "STO2 ADC Mixer", + "Mono ADC Mixer L", "Mono ADC Mixer R", "DAC1" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_rxdp_enum, RT5670_DSP_PATH1, + RT5670_RXDP_SEL_SFT, rt5670_rxdp_src); + +static const struct snd_kcontrol_new rt5670_rxdp_mux = + SOC_DAPM_ENUM("DAC2 L source", rt5670_rxdp_enum); + +/* MX-2D [1] [0] */ +static const char * const rt5670_dsp_bypass_src[] = { + "DSP", "Bypass" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dsp_ul_enum, RT5670_DSP_PATH1, + RT5670_DSP_UL_SFT, rt5670_dsp_bypass_src); + +static const struct snd_kcontrol_new rt5670_dsp_ul_mux = + SOC_DAPM_ENUM("DSP UL source", rt5670_dsp_ul_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_dsp_dl_enum, RT5670_DSP_PATH1, + RT5670_DSP_DL_SFT, rt5670_dsp_bypass_src); + +static const struct snd_kcontrol_new rt5670_dsp_dl_mux = + SOC_DAPM_ENUM("DSP DL source", rt5670_dsp_dl_enum); + +/* Stereo2 ADC source */ +/* MX-26 [15] */ +static const char * const rt5670_stereo2_adc_lr_src[] = { + "L", "LR" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc_lr_enum, RT5670_STO2_ADC_MIXER, + RT5670_STO2_ADC_SRC_SFT, rt5670_stereo2_adc_lr_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_lr_mux = + SOC_DAPM_ENUM("Stereo2 ADC LR source", rt5670_stereo2_adc_lr_enum); + +/* Stereo1 ADC source */ +/* MX-27 MX-26 [12] */ +static const char * const rt5670_stereo_adc1_src[] = { + "DAC MIX", "ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_adc1_enum, RT5670_STO1_ADC_MIXER, + RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5670_sto_adc_l1_mux = + SOC_DAPM_ENUM("Stereo1 ADC L1 source", rt5670_stereo1_adc1_enum); + +static const struct snd_kcontrol_new rt5670_sto_adc_r1_mux = + SOC_DAPM_ENUM("Stereo1 ADC R1 source", rt5670_stereo1_adc1_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc1_enum, RT5670_STO2_ADC_MIXER, + RT5670_ADC_1_SRC_SFT, rt5670_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_l1_mux = + SOC_DAPM_ENUM("Stereo2 ADC L1 source", rt5670_stereo2_adc1_enum); + +static const struct snd_kcontrol_new rt5670_sto2_adc_r1_mux = + SOC_DAPM_ENUM("Stereo2 ADC R1 source", rt5670_stereo2_adc1_enum); + +/* MX-27 MX-26 [11] */ +static const char * const rt5670_stereo_adc2_src[] = { + "DAC MIX", "DMIC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_adc2_enum, RT5670_STO1_ADC_MIXER, + RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5670_sto_adc_l2_mux = + SOC_DAPM_ENUM("Stereo1 ADC L2 source", rt5670_stereo1_adc2_enum); + +static const struct snd_kcontrol_new rt5670_sto_adc_r2_mux = + SOC_DAPM_ENUM("Stereo1 ADC R2 source", rt5670_stereo1_adc2_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc2_enum, RT5670_STO2_ADC_MIXER, + RT5670_ADC_2_SRC_SFT, rt5670_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_l2_mux = + SOC_DAPM_ENUM("Stereo2 ADC L2 source", rt5670_stereo2_adc2_enum); + +static const struct snd_kcontrol_new rt5670_sto2_adc_r2_mux = + SOC_DAPM_ENUM("Stereo2 ADC R2 source", rt5670_stereo2_adc2_enum); + +/* MX-27 MX26 [10] */ +static const char * const rt5670_stereo_adc_src[] = { + "ADC1L ADC2R", "ADC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_adc_enum, RT5670_STO1_ADC_MIXER, + RT5670_ADC_SRC_SFT, rt5670_stereo_adc_src); + +static const struct snd_kcontrol_new rt5670_sto_adc_mux = + SOC_DAPM_ENUM("Stereo1 ADC source", rt5670_stereo1_adc_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_adc_enum, RT5670_STO2_ADC_MIXER, + RT5670_ADC_SRC_SFT, rt5670_stereo_adc_src); + +static const struct snd_kcontrol_new rt5670_sto2_adc_mux = + SOC_DAPM_ENUM("Stereo2 ADC source", rt5670_stereo2_adc_enum); + +/* MX-27 MX-26 [9:8] */ +static const char * const rt5670_stereo_dmic_src[] = { + "DMIC1", "DMIC2", "DMIC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo1_dmic_enum, RT5670_STO1_ADC_MIXER, + RT5670_DMIC_SRC_SFT, rt5670_stereo_dmic_src); + +static const struct snd_kcontrol_new rt5670_sto1_dmic_mux = + SOC_DAPM_ENUM("Stereo1 DMIC source", rt5670_stereo1_dmic_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo2_dmic_enum, RT5670_STO2_ADC_MIXER, + RT5670_DMIC_SRC_SFT, rt5670_stereo_dmic_src); + +static const struct snd_kcontrol_new rt5670_sto2_dmic_mux = + SOC_DAPM_ENUM("Stereo2 DMIC source", rt5670_stereo2_dmic_enum); + +/* MX-27 [0] */ +static const char * const rt5670_stereo_dmic3_src[] = { + "DMIC3", "PDM ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_stereo_dmic3_enum, RT5670_STO1_ADC_MIXER, + RT5670_DMIC3_SRC_SFT, rt5670_stereo_dmic3_src); + +static const struct snd_kcontrol_new rt5670_sto_dmic3_mux = + SOC_DAPM_ENUM("Stereo DMIC3 source", rt5670_stereo_dmic3_enum); + +/* Mono ADC source */ +/* MX-28 [12] */ +static const char * const rt5670_mono_adc_l1_src[] = { + "Mono DAC MIXL", "ADC1" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_l1_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_L1_SRC_SFT, rt5670_mono_adc_l1_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_l1_mux = + SOC_DAPM_ENUM("Mono ADC1 left source", rt5670_mono_adc_l1_enum); +/* MX-28 [11] */ +static const char * const rt5670_mono_adc_l2_src[] = { + "Mono DAC MIXL", "DMIC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_l2_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_L2_SRC_SFT, rt5670_mono_adc_l2_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_l2_mux = + SOC_DAPM_ENUM("Mono ADC2 left source", rt5670_mono_adc_l2_enum); + +/* MX-28 [9:8] */ +static const char * const rt5670_mono_dmic_src[] = { + "DMIC1", "DMIC2", "DMIC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_dmic_l_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_DMIC_L_SRC_SFT, rt5670_mono_dmic_src); + +static const struct snd_kcontrol_new rt5670_mono_dmic_l_mux = + SOC_DAPM_ENUM("Mono DMIC left source", rt5670_mono_dmic_l_enum); +/* MX-28 [1:0] */ +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_dmic_r_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_DMIC_R_SRC_SFT, rt5670_mono_dmic_src); + +static const struct snd_kcontrol_new rt5670_mono_dmic_r_mux = + SOC_DAPM_ENUM("Mono DMIC Right source", rt5670_mono_dmic_r_enum); +/* MX-28 [4] */ +static const char * const rt5670_mono_adc_r1_src[] = { + "Mono DAC MIXR", "ADC2" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_r1_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_R1_SRC_SFT, rt5670_mono_adc_r1_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_r1_mux = + SOC_DAPM_ENUM("Mono ADC1 right source", rt5670_mono_adc_r1_enum); +/* MX-28 [3] */ +static const char * const rt5670_mono_adc_r2_src[] = { + "Mono DAC MIXR", "DMIC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_mono_adc_r2_enum, RT5670_MONO_ADC_MIXER, + RT5670_MONO_ADC_R2_SRC_SFT, rt5670_mono_adc_r2_src); + +static const struct snd_kcontrol_new rt5670_mono_adc_r2_mux = + SOC_DAPM_ENUM("Mono ADC2 right source", rt5670_mono_adc_r2_enum); + +/* MX-2F [15] */ +static const char * const rt5670_if1_adc2_in_src[] = { + "IF_ADC2", "VAD_ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc2_in_enum, RT5670_DIG_INF1_DATA, + RT5670_IF1_ADC2_IN_SFT, rt5670_if1_adc2_in_src); + +static const struct snd_kcontrol_new rt5670_if1_adc2_in_mux = + SOC_DAPM_ENUM("IF1 ADC2 IN source", rt5670_if1_adc2_in_enum); + +/* MX-2F [14:12] */ +static const char * const rt5670_if2_adc_in_src[] = { + "IF_ADC1", "IF_ADC2", "IF_ADC3", "TxDC_DAC", "TxDP_ADC", "VAD_ADC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if2_adc_in_enum, RT5670_DIG_INF1_DATA, + RT5670_IF2_ADC_IN_SFT, rt5670_if2_adc_in_src); + +static const struct snd_kcontrol_new rt5670_if2_adc_in_mux = + SOC_DAPM_ENUM("IF2 ADC IN source", rt5670_if2_adc_in_enum); + +/* MX-30 [5:4] */ +static const char * const rt5670_if4_adc_in_src[] = { + "IF_ADC1", "IF_ADC2", "IF_ADC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if4_adc_in_enum, RT5670_DIG_INF2_DATA, + RT5670_IF4_ADC_IN_SFT, rt5670_if4_adc_in_src); + +static const struct snd_kcontrol_new rt5670_if4_adc_in_mux = + SOC_DAPM_ENUM("IF4 ADC IN source", rt5670_if4_adc_in_enum); + +/* MX-31 [15] [13] [11] [9] */ +static const char * const rt5670_pdm_src[] = { + "Mono DAC", "Stereo DAC" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm1_l_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM1_L_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm1_l_mux = + SOC_DAPM_ENUM("PDM1 L source", rt5670_pdm1_l_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm1_r_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM1_R_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm1_r_mux = + SOC_DAPM_ENUM("PDM1 R source", rt5670_pdm1_r_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm2_l_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM2_L_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm2_l_mux = + SOC_DAPM_ENUM("PDM2 L source", rt5670_pdm2_l_enum); + +static const SOC_ENUM_SINGLE_DECL( + rt5670_pdm2_r_enum, RT5670_PDM_OUT_CTRL, + RT5670_PDM2_R_SFT, rt5670_pdm_src); + +static const struct snd_kcontrol_new rt5670_pdm2_r_mux = + SOC_DAPM_ENUM("PDM2 R source", rt5670_pdm2_r_enum); + +/* MX-FA [12] */ +static const char * const rt5670_if1_adc1_in1_src[] = { + "IF_ADC1", "IF1_ADC3" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc1_in1_enum, RT5670_DIG_MISC, + RT5670_IF1_ADC1_IN1_SFT, rt5670_if1_adc1_in1_src); + +static const struct snd_kcontrol_new rt5670_if1_adc1_in1_mux = + SOC_DAPM_ENUM("IF1 ADC1 IN1 source", rt5670_if1_adc1_in1_enum); + +/* MX-FA [11] */ +static const char * const rt5670_if1_adc1_in2_src[] = { + "IF1_ADC1_IN1", "IF1_ADC4" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc1_in2_enum, RT5670_DIG_MISC, + RT5670_IF1_ADC1_IN2_SFT, rt5670_if1_adc1_in2_src); + +static const struct snd_kcontrol_new rt5670_if1_adc1_in2_mux = + SOC_DAPM_ENUM("IF1 ADC1 IN2 source", rt5670_if1_adc1_in2_enum); + +/* MX-FA [10] */ +static const char * const rt5670_if1_adc2_in1_src[] = { + "IF1_ADC2_IN", "IF1_ADC4" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_if1_adc2_in1_enum, RT5670_DIG_MISC, + RT5670_IF1_ADC2_IN1_SFT, rt5670_if1_adc2_in1_src); + +static const struct snd_kcontrol_new rt5670_if1_adc2_in1_mux = + SOC_DAPM_ENUM("IF1 ADC2 IN1 source", rt5670_if1_adc2_in1_enum); + +/* MX-9D [9:8] */ +static const char * const rt5670_vad_adc_src[] = { + "Sto1 ADC L", "Mono ADC L", "Mono ADC R", "Sto2 ADC L" +}; + +static const SOC_ENUM_SINGLE_DECL( + rt5670_vad_adc_enum, RT5670_VAD_CTRL4, + RT5670_VAD_SEL_SFT, rt5670_vad_adc_src); + +static const struct snd_kcontrol_new rt5670_vad_adc_mux = + SOC_DAPM_ENUM("VAD ADC source", rt5670_vad_adc_enum); + +static int rt5670_adc_clk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(rt5670->regmap, RT5670_PR_BASE + + RT5670_CHOP_DAC_ADC, 0x1000, 0x1000); + break; + + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(rt5670->regmap, RT5670_PR_BASE + + RT5670_CHOP_DAC_ADC, 0x1000, 0x0000); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_sto1_adcl_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_STO1_ADC_DIG_VOL, + RT5670_L_MUTE, 0); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_STO1_ADC_DIG_VOL, + RT5670_L_MUTE, + RT5670_L_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_sto1_adcr_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_STO1_ADC_DIG_VOL, + RT5670_R_MUTE, 0); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_STO1_ADC_DIG_VOL, + RT5670_R_MUTE, + RT5670_R_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_mono_adcl_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_MONO_ADC_DIG_VOL, + RT5670_L_MUTE, 0); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_MONO_ADC_DIG_VOL, + RT5670_L_MUTE, + RT5670_L_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_mono_adcr_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_MONO_ADC_DIG_VOL, + RT5670_R_MUTE, 0); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_MONO_ADC_DIG_VOL, + RT5670_R_MUTE, + RT5670_R_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +void hp_amp_power(struct snd_soc_codec *codec, int on) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + if (on) { + regmap_update_bits(rt5670->regmap, RT5670_CHARGE_PUMP, + RT5670_PM_HP_MASK, RT5670_PM_HP_HV); + regmap_update_bits(rt5670->regmap, RT5670_GEN_CTRL2, + 0x0400, 0x0400); + /* headphone amp power on */ + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, + RT5670_PWR_HA | RT5670_PWR_FV1 | + RT5670_PWR_FV2, RT5670_PWR_HA | + RT5670_PWR_FV1 | RT5670_PWR_FV2); + /* depop parameters */ + regmap_write(rt5670->regmap, RT5670_DEPOP_M2, 0x3100); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8009); + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_HP_DCC_INT1, 0x9f00); + mdelay(20); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019); + } else { + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x0004); + msleep(30); + } +} + +static void rt5670_pmu_depop(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + /* headphone unmute sequence */ + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_MAMP_INT_REG2, 0xb400); + regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0772); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x805d); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x831d); + regmap_update_bits(rt5670->regmap, RT5670_GEN_CTRL2, + 0x0300, 0x0300); + regmap_update_bits(rt5670->regmap, RT5670_HP_VOL, + RT5670_L_MUTE | RT5670_R_MUTE, 0); + msleep(80); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019); +} + +static void rt5670_pmd_depop(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + /* headphone mute sequence */ + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_MAMP_INT_REG2, 0xb400); + regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0772); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x803d); + mdelay(10); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x831d); + mdelay(10); + regmap_update_bits(rt5670->regmap, RT5670_HP_VOL, + RT5670_L_MUTE | RT5670_R_MUTE, RT5670_L_MUTE | RT5670_R_MUTE); + msleep(20); + regmap_update_bits(rt5670->regmap, RT5670_GEN_CTRL2, 0x0300, 0x0); + regmap_write(rt5670->regmap, RT5670_DEPOP_M1, 0x8019); + regmap_write(rt5670->regmap, RT5670_DEPOP_M3, 0x0707); + regmap_write(rt5670->regmap, RT5670_PR_BASE + + RT5670_MAMP_INT_REG2, 0xfc00); +} + +static int rt5670_hp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + hp_amp_power(codec, 1); + break; + case SND_SOC_DAPM_PRE_PMD: + hp_amp_power(codec, 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt5670_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt5670_pmu_depop(codec); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt5670_pmd_depop(codec); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_lout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_LOUT1, + RT5670_L_MUTE | RT5670_R_MUTE, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_LOUT1, + RT5670_L_MUTE | RT5670_R_MUTE, + RT5670_L_MUTE | RT5670_R_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_bst1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x0); + snd_soc_update_bits(codec, RT5670_CHARGE_PUMP, + RT5670_OSW_L_MASK | RT5670_OSW_R_MASK, + RT5670_OSW_L_DIS | RT5670_OSW_R_DIS); + + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST1_P, RT5670_PWR_BST1_P); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x4); + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST1_P, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_bst2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST2_P, RT5670_PWR_BST2_P); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_PWR_ANLG2, + RT5670_PWR_BST2_P, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_pdm1_l_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_PDM_OUT_CTRL, + RT5670_M_PDM1_L, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_PDM_OUT_CTRL, + RT5670_M_PDM1_L, RT5670_M_PDM1_L); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_pdm1_r_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_PDM_OUT_CTRL, + RT5670_M_PDM1_R, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_PDM_OUT_CTRL, + RT5670_M_PDM1_R, RT5670_M_PDM1_R); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_pdm2_l_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_L, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_L, RT5670_M_PDM2_L); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_pdm2_r_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_R, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_R, RT5670_M_PDM2_R); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5670_asrc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(w->codec, RT5670_ASRC_1, 0xffff); + snd_soc_write(w->codec, RT5670_ASRC_2, 0x1221); + snd_soc_write(w->codec, RT5670_ASRC_3, 0x0022); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(w->codec, RT5670_ASRC_1, 0); + snd_soc_write(w->codec, RT5670_ASRC_2, 0); + snd_soc_write(w->codec, RT5670_ASRC_3, 0); + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL1", RT5670_PWR_ANLG2, + RT5670_PWR_PLL_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S DSP", RT5670_PWR_DIG2, + RT5670_PWR_I2S_DSP_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("JD Power", RT5670_PWR_ANLG2, + RT5670_PWR_JD1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5670_PWR_VOL, + RT5670_PWR_MIC_DET_BIT, 0, NULL, 0), + + /* Input Side */ + /* micbias */ + SND_SOC_DAPM_MICBIAS("micbias1", RT5670_PWR_ANLG2, + RT5670_PWR_MB1_BIT, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + SND_SOC_DAPM_INPUT("DMIC L2"), + SND_SOC_DAPM_INPUT("DMIC R2"), + SND_SOC_DAPM_INPUT("DMIC L3"), + SND_SOC_DAPM_INPUT("DMIC R3"), + + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + + SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC3", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5670_DMIC_CTRL1, + RT5670_DMIC_1_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC2 Power", RT5670_DMIC_CTRL1, + RT5670_DMIC_2_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC3 Power", RT5670_DMIC_CTRL1, + RT5670_DMIC_3_EN_SFT, 0, NULL, 0), + /* Boost */ + SND_SOC_DAPM_PGA_E("BST1", RT5670_PWR_ANLG2, + RT5670_PWR_BST1_BIT, 0, NULL, 0, rt5670_bst1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("BST2", RT5670_PWR_ANLG2, + RT5670_PWR_BST2_BIT, 0, NULL, 0, rt5670_bst2_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + /* Input Volume */ + SND_SOC_DAPM_PGA("INL VOL", RT5670_PWR_VOL, + RT5670_PWR_IN_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR VOL", RT5670_PWR_VOL, + RT5670_PWR_IN_R_BIT, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", RT5670_PWR_MIXER, RT5670_PWR_RM_L_BIT, 0, + rt5670_rec_l_mix, ARRAY_SIZE(rt5670_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", RT5670_PWR_MIXER, RT5670_PWR_RM_R_BIT, 0, + rt5670_rec_r_mix, ARRAY_SIZE(rt5670_rec_r_mix)), + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, + 0, 0), + SND_SOC_DAPM_ADC("ADC 2", NULL, SND_SOC_NOPM, + 0, 0), + + SND_SOC_DAPM_PGA("ADC 1_2", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("ADC 1 power", RT5670_PWR_DIG1, + RT5670_PWR_ADC_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC 2 power", RT5670_PWR_DIG1, + RT5670_PWR_ADC_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC clock", SND_SOC_NOPM, 0, 0, + rt5670_adc_clk_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU), + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto1_dmic_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_l2_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_r2_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_l1_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto_adc_r1_mux), + SND_SOC_DAPM_MUX("Stereo2 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_dmic_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_l2_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_r2_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_l1_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_r1_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC LR Mux", SND_SOC_NOPM, 0, 0, + &rt5670_sto2_adc_lr_mux), + SND_SOC_DAPM_MUX("Mono DMIC L Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_dmic_l_mux), + SND_SOC_DAPM_MUX("Mono DMIC R Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_dmic_r_mux), + SND_SOC_DAPM_MUX("Mono ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_l2_mux), + SND_SOC_DAPM_MUX("Mono ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_l1_mux), + SND_SOC_DAPM_MUX("Mono ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_r1_mux), + SND_SOC_DAPM_MUX("Mono ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_mono_adc_r2_mux), + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("adc stereo1 filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("adc stereo2 filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_S2F_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER_E("Sto1 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_sto1_adc_l_mix, ARRAY_SIZE(rt5670_sto1_adc_l_mix), + rt5670_sto1_adcl_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MIXER_E("Sto1 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_sto1_adc_r_mix, ARRAY_SIZE(rt5670_sto1_adc_r_mix), + rt5670_sto1_adcr_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MIXER("Sto2 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_sto2_adc_l_mix, ARRAY_SIZE(rt5670_sto2_adc_l_mix)), + SND_SOC_DAPM_MIXER("Sto2 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_sto2_adc_r_mix, ARRAY_SIZE(rt5670_sto2_adc_r_mix)), + SND_SOC_DAPM_SUPPLY("adc mono left filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER_E("Mono ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_mono_adc_l_mix, ARRAY_SIZE(rt5670_mono_adc_l_mix), + rt5670_mono_adcl_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("adc mono right filter", RT5670_PWR_DIG2, + RT5670_PWR_ADC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER_E("Mono ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_mono_adc_r_mix, ARRAY_SIZE(rt5670_mono_adc_r_mix), + rt5670_mono_adcr_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + /* ADC PGA */ + SND_SOC_DAPM_PGA("Stereo1 ADC MIXL", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo1 ADC MIXR", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIXL", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIXR", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Sto2 ADC LR MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo2 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("VAD_ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DSP */ + SND_SOC_DAPM_PGA("TxDP_ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TxDP_ADC_L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TxDP_ADC_R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TxDC_DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_PGA("8CH TDM Data", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DSP UL Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dsp_ul_mux), + SND_SOC_DAPM_MUX("DSP DL Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dsp_dl_mux), + + SND_SOC_DAPM_MUX("RxDP Mux", SND_SOC_NOPM, 0, 0, + &rt5670_rxdp_mux), + + /* IF2 Mux */ + SND_SOC_DAPM_MUX("IF2 ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if2_adc_in_mux), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5670_PWR_DIG1, + RT5670_PWR_I2S1_BIT, 0, NULL, 0), + 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 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2 R", SND_SOC_NOPM, 0, 0, NULL, 0), + 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), + SND_SOC_DAPM_SUPPLY("I2S2", RT5670_PWR_DIG1, + RT5670_PWR_I2S2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("IF1 ADC1 IN1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc1_in1_mux), + SND_SOC_DAPM_MUX("IF1 ADC1 IN2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc1_in2_mux), + SND_SOC_DAPM_MUX("IF1 ADC2 IN Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc2_in_mux), + SND_SOC_DAPM_MUX("IF1 ADC2 IN1 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_if1_adc2_in1_mux), + SND_SOC_DAPM_MUX("VAD ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5670_vad_adc_mux), + + /* 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), + + /* Audio DSP */ + SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, + rt5670_dac_l_mix, ARRAY_SIZE(rt5670_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, + rt5670_dac_r_mix, ARRAY_SIZE(rt5670_dac_r_mix)), + SND_SOC_DAPM_PGA("DAC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DAC2 channel Mux */ + SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dac_l2_mux), + SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dac_r2_mux), + SND_SOC_DAPM_PGA("DAC L2 Volume", RT5670_PWR_DIG1, + RT5670_PWR_DAC_L2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC R2 Volume", RT5670_PWR_DIG1, + RT5670_PWR_DAC_R2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DAC1 L Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dac1l_mux), + SND_SOC_DAPM_MUX("DAC1 R Mux", SND_SOC_NOPM, 0, 0, + &rt5670_dac1r_mux), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("dac stereo1 filter", RT5670_PWR_DIG2, + RT5670_PWR_DAC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono left filter", RT5670_PWR_DIG2, + RT5670_PWR_DAC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono right filter", RT5670_PWR_DIG2, + RT5670_PWR_DAC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_sto_dac_l_mix, ARRAY_SIZE(rt5670_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_sto_dac_r_mix, ARRAY_SIZE(rt5670_sto_dac_r_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_mono_dac_l_mix, ARRAY_SIZE(rt5670_mono_dac_l_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_mono_dac_r_mix, ARRAY_SIZE(rt5670_mono_dac_r_mix)), + SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5670_dig_l_mix, ARRAY_SIZE(rt5670_dig_l_mix)), + SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5670_dig_r_mix, ARRAY_SIZE(rt5670_dig_r_mix)), + + /* DACs */ + SND_SOC_DAPM_SUPPLY("DAC L1 Power", RT5670_PWR_DIG1, + RT5670_PWR_DAC_L1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R1 Power", RT5670_PWR_DIG1, + RT5670_PWR_DAC_R1_BIT, 0, NULL, 0), + SND_SOC_DAPM_DAC("DAC L1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC L2", NULL, RT5670_PWR_DIG1, + RT5670_PWR_DAC_L2_BIT, 0), + + SND_SOC_DAPM_DAC("DAC R2", NULL, RT5670_PWR_DIG1, + RT5670_PWR_DAC_R2_BIT, 0), + /* OUT Mixer */ + + SND_SOC_DAPM_MIXER("OUT MIXL", RT5670_PWR_MIXER, RT5670_PWR_OM_L_BIT, + 0, rt5670_out_l_mix, ARRAY_SIZE(rt5670_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5670_PWR_MIXER, RT5670_PWR_OM_R_BIT, + 0, rt5670_out_r_mix, ARRAY_SIZE(rt5670_out_r_mix)), + /* Ouput Volume */ + SND_SOC_DAPM_MIXER("HPOVOL MIXL", RT5670_PWR_VOL, RT5670_PWR_HV_L_BIT, + 0, rt5670_hpvoll_mix, ARRAY_SIZE(rt5670_hpvoll_mix)), + SND_SOC_DAPM_MIXER("HPOVOL MIXR", RT5670_PWR_VOL, RT5670_PWR_HV_R_BIT, + 0, rt5670_hpvolr_mix, ARRAY_SIZE(rt5670_hpvolr_mix)), + SND_SOC_DAPM_PGA("DAC 1", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC 2", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPOVOL", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("HPO MIX", SND_SOC_NOPM, 0, 0, + rt5670_hpo_mix, ARRAY_SIZE(rt5670_hpo_mix)), + SND_SOC_DAPM_MIXER("LOUT MIX", RT5670_PWR_ANLG1, RT5670_PWR_LM_BIT, + 0, rt5670_lout_mix, ARRAY_SIZE(rt5670_lout_mix)), + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, + 0, 0, rt5670_hp_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5670_PWR_ANLG1, + RT5670_PWR_HP_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5670_PWR_ANLG1, + RT5670_PWR_HP_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, + rt5670_hp_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("LOUT Amp", 1, SND_SOC_NOPM, 0, 0, + rt5670_lout_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + /* PDM */ + SND_SOC_DAPM_SUPPLY("PDM1 Power", RT5670_PWR_DIG2, + RT5670_PWR_PDM1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PDM2 Power", RT5670_PWR_DIG2, + RT5670_PWR_PDM2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MUX_E("PDM1 L Mux", SND_SOC_NOPM, 0, 0, + &rt5670_pdm1_l_mux, rt5670_pdm1_l_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX_E("PDM1 R Mux", SND_SOC_NOPM, 0, 0, + &rt5670_pdm1_r_mux, rt5670_pdm1_r_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX_E("PDM2 L Mux", SND_SOC_NOPM, 0, 0, + &rt5670_pdm2_l_mux, rt5670_pdm2_l_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX_E("PDM2 R Mux", SND_SOC_NOPM, 0, 0, + &rt5670_pdm2_r_mux, rt5670_pdm2_r_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("LOUTL"), + SND_SOC_DAPM_OUTPUT("LOUTR"), + SND_SOC_DAPM_OUTPUT("PDM1L"), + SND_SOC_DAPM_OUTPUT("PDM1R"), + SND_SOC_DAPM_OUTPUT("PDM2L"), + SND_SOC_DAPM_OUTPUT("PDM2R"), +}; + +static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { + { "micbias1", NULL, "DAC L1 Power" }, + { "micbias1", NULL, "DAC R1 Power" }, + + { "DMIC1", NULL, "DMIC L1" }, + { "DMIC1", NULL, "DMIC R1" }, + { "DMIC2", NULL, "DMIC L2" }, + { "DMIC2", NULL, "DMIC R2" }, + { "DMIC3", NULL, "DMIC L3" }, + { "DMIC3", NULL, "DMIC R3" }, + + { "BST1", NULL, "IN1P" }, + { "BST1", NULL, "IN1N" }, + { "BST1", NULL, "JD Power" }, + { "BST1", NULL, "Mic Det Power" }, + { "BST2", NULL, "IN2P" }, + { "BST2", NULL, "IN2N" }, + + { "INL VOL", NULL, "IN2P" }, + { "INR VOL", NULL, "IN2N" }, + + { "RECMIXL", "INL Switch", "INL VOL" }, + { "RECMIXL", "BST2 Switch", "BST2" }, + { "RECMIXL", "BST1 Switch", "BST1" }, + + { "RECMIXR", "INR Switch", "INR VOL" }, + { "RECMIXR", "BST2 Switch", "BST2" }, + { "RECMIXR", "BST1 Switch", "BST1" }, + + { "ADC 1", NULL, "RECMIXL" }, + { "ADC 1", NULL, "ADC 1 power" }, + { "ADC 1", NULL, "ADC clock" }, + { "ADC 2", NULL, "RECMIXR" }, + { "ADC 2", NULL, "ADC 2 power" }, + { "ADC 2", NULL, "ADC clock" }, + + { "DMIC L1", NULL, "DMIC CLK" }, + { "DMIC L1", NULL, "DMIC1 Power" }, + { "DMIC R1", NULL, "DMIC CLK" }, + { "DMIC R1", NULL, "DMIC1 Power" }, + { "DMIC L2", NULL, "DMIC CLK" }, + { "DMIC L2", NULL, "DMIC2 Power" }, + { "DMIC R2", NULL, "DMIC CLK" }, + { "DMIC R2", NULL, "DMIC2 Power" }, + { "DMIC L3", NULL, "DMIC CLK" }, + { "DMIC L3", NULL, "DMIC3 Power" }, + { "DMIC R3", NULL, "DMIC CLK" }, + { "DMIC R3", NULL, "DMIC3 Power" }, + + { "Stereo1 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo1 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo1 DMIC Mux", "DMIC3", "DMIC3" }, + + { "Stereo2 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo2 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo2 DMIC Mux", "DMIC3", "DMIC3" }, + + { "Mono DMIC L Mux", "DMIC1", "DMIC L1" }, + { "Mono DMIC L Mux", "DMIC2", "DMIC L2" }, + { "Mono DMIC L Mux", "DMIC3", "DMIC L3" }, + + { "Mono DMIC R Mux", "DMIC1", "DMIC R1" }, + { "Mono DMIC R Mux", "DMIC2", "DMIC R2" }, + { "Mono DMIC R Mux", "DMIC3", "DMIC R3" }, + + { "ADC 1_2", NULL, "ADC 1" }, + { "ADC 1_2", NULL, "ADC 2" }, + + { "Stereo1 ADC L2 Mux", "DMIC", "Stereo1 DMIC Mux" }, + { "Stereo1 ADC L2 Mux", "DAC MIX", "DAC MIXL" }, + { "Stereo1 ADC L1 Mux", "ADC", "ADC 1_2" }, + { "Stereo1 ADC L1 Mux", "DAC MIX", "DAC MIXL" }, + + { "Stereo1 ADC R1 Mux", "ADC", "ADC 1_2" }, + { "Stereo1 ADC R1 Mux", "DAC MIX", "DAC MIXR" }, + { "Stereo1 ADC R2 Mux", "DMIC", "Stereo1 DMIC Mux" }, + { "Stereo1 ADC R2 Mux", "DAC MIX", "DAC MIXR" }, + + { "Mono ADC L2 Mux", "DMIC", "Mono DMIC L Mux" }, + { "Mono ADC L2 Mux", "Mono DAC MIXL", "Mono DAC MIXL" }, + { "Mono ADC L1 Mux", "Mono DAC MIXL", "Mono DAC MIXL" }, + { "Mono ADC L1 Mux", "ADC1", "ADC 1" }, + + { "Mono ADC R1 Mux", "Mono DAC MIXR", "Mono DAC MIXR" }, + { "Mono ADC R1 Mux", "ADC2", "ADC 2" }, + { "Mono ADC R2 Mux", "DMIC", "Mono DMIC R Mux" }, + { "Mono ADC R2 Mux", "Mono DAC MIXR", "Mono DAC MIXR" }, + + { "Sto1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux" }, + { "Sto1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux" }, + { "Sto1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux" }, + { "Sto1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux" }, + + { "Stereo1 ADC MIXL", NULL, "Sto1 ADC MIXL" }, + { "Stereo1 ADC MIXL", NULL, "adc stereo1 filter" }, + { "adc stereo1 filter", NULL, "PLL1", check_sysclk1_source }, + + { "Stereo1 ADC MIXR", NULL, "Sto1 ADC MIXR" }, + { "Stereo1 ADC MIXR", NULL, "adc stereo1 filter" }, + { "adc stereo1 filter", NULL, "PLL1", check_sysclk1_source }, + + { "Mono ADC MIXL", "ADC1 Switch", "Mono ADC L1 Mux" }, + { "Mono ADC MIXL", "ADC2 Switch", "Mono ADC L2 Mux" }, + { "Mono ADC MIXL", NULL, "adc mono left filter" }, + { "adc mono left filter", NULL, "PLL1", check_sysclk1_source }, + + { "Mono ADC MIXR", "ADC1 Switch", "Mono ADC R1 Mux" }, + { "Mono ADC MIXR", "ADC2 Switch", "Mono ADC R2 Mux" }, + { "Mono ADC MIXR", NULL, "adc mono right filter" }, + { "adc mono right filter", NULL, "PLL1", check_sysclk1_source }, + + { "Stereo2 ADC L2 Mux", "DMIC", "Stereo2 DMIC Mux" }, + { "Stereo2 ADC L2 Mux", "DAC MIX", "DAC MIXL" }, + { "Stereo2 ADC L1 Mux", "ADC", "ADC 1_2" }, + { "Stereo2 ADC L1 Mux", "DAC MIX", "DAC MIXL" }, + + { "Stereo2 ADC R1 Mux", "ADC", "ADC 1_2" }, + { "Stereo2 ADC R1 Mux", "DAC MIX", "DAC MIXR" }, + { "Stereo2 ADC R2 Mux", "DMIC", "Stereo2 DMIC Mux" }, + { "Stereo2 ADC R2 Mux", "DAC MIX", "DAC MIXR" }, + + { "Sto2 ADC MIXL", "ADC1 Switch", "Stereo2 ADC L1 Mux" }, + { "Sto2 ADC MIXL", "ADC2 Switch", "Stereo2 ADC L2 Mux" }, + { "Sto2 ADC MIXR", "ADC1 Switch", "Stereo2 ADC R1 Mux" }, + { "Sto2 ADC MIXR", "ADC2 Switch", "Stereo2 ADC R2 Mux" }, + + { "Sto2 ADC LR MIX", NULL, "Sto2 ADC MIXL" }, + { "Sto2 ADC LR MIX", NULL, "Sto2 ADC MIXR" }, + + { "Stereo2 ADC LR Mux", "L", "Sto2 ADC MIXL" }, + { "Stereo2 ADC LR Mux", "LR", "Sto2 ADC LR MIX" }, + + { "Stereo2 ADC MIXL", NULL, "Stereo2 ADC LR Mux" }, + { "Stereo2 ADC MIXL", NULL, "adc stereo2 filter" }, + { "adc stereo2 filter", NULL, "PLL1", check_sysclk1_source }, + + { "Stereo2 ADC MIXR", NULL, "Sto2 ADC MIXR" }, + { "Stereo2 ADC MIXR", NULL, "adc stereo2 filter" }, + { "adc stereo2 filter", NULL, "PLL1", check_sysclk1_source }, + + { "VAD ADC Mux", "Sto1 ADC L", "Stereo1 ADC MIXL" }, + { "VAD ADC Mux", "Mono ADC L", "Mono ADC MIXL" }, + { "VAD ADC Mux", "Mono ADC R", "Mono ADC MIXR" }, + { "VAD ADC Mux", "Sto2 ADC L", "Sto2 ADC MIXL" }, + + { "VAD_ADC", NULL, "VAD ADC Mux" }, + + { "IF_ADC1", NULL, "Stereo1 ADC MIXL" }, + { "IF_ADC1", NULL, "Stereo1 ADC MIXR" }, + { "IF_ADC2", NULL, "Mono ADC MIXL" }, + { "IF_ADC2", NULL, "Mono ADC MIXR" }, + { "IF_ADC3", NULL, "Stereo2 ADC MIXL" }, + { "IF_ADC3", NULL, "Stereo2 ADC MIXR" }, + + { "IF1 ADC1 IN1 Mux", "IF_ADC1", "IF_ADC1" }, + { "IF1 ADC1 IN1 Mux", "IF1_ADC3", "IF1_ADC3" }, + + { "IF1 ADC1 IN2 Mux", "IF1_ADC1_IN1", "IF1 ADC1 IN1 Mux" }, + { "IF1 ADC1 IN2 Mux", "IF1_ADC4", "IF1_ADC4" }, + + { "IF1 ADC2 IN Mux", "IF_ADC2", "IF_ADC2" }, + { "IF1 ADC2 IN Mux", "VAD_ADC", "VAD_ADC" }, + + { "IF1 ADC2 IN1 Mux", "IF1_ADC2_IN", "IF1 ADC2 IN Mux" }, + { "IF1 ADC2 IN1 Mux", "IF1_ADC4", "IF1_ADC4" }, + + { "IF1_ADC1" , NULL, "IF1 ADC1 IN2 Mux" }, + { "IF1_ADC2" , NULL, "IF1 ADC2 IN1 Mux" }, + + { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL" }, + { "Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR" }, + { "Stereo2 ADC MIX", NULL, "Sto2 ADC MIXL" }, + { "Stereo2 ADC MIX", NULL, "Sto2 ADC MIXR" }, + + { "RxDP Mux", "IF2 DAC", "IF2 DAC" }, + { "RxDP Mux", "IF1 DAC", "IF1 DAC2" }, + { "RxDP Mux", "STO1 ADC Mixer", "Stereo1 ADC MIX" }, + { "RxDP Mux", "STO2 ADC Mixer", "Stereo2 ADC MIX" }, + { "RxDP Mux", "Mono ADC Mixer L", "Mono ADC MIXL" }, + { "RxDP Mux", "Mono ADC Mixer R", "Mono ADC MIXR" }, + { "RxDP Mux", "DAC1", "DAC MIX" }, + + { "8CH TDM Data", NULL, "Stereo1 ADC MIXL" }, + { "8CH TDM Data", NULL, "Stereo1 ADC MIXR" }, + { "8CH TDM Data", NULL, "Mono ADC MIXL" }, + { "8CH TDM Data", NULL, "Mono ADC MIXR" }, + { "8CH TDM Data", NULL, "Sto2 ADC MIXL" }, + { "8CH TDM Data", NULL, "Sto2 ADC MIXR" }, + { "8CH TDM Data", NULL, "IF2 DAC L" }, + { "8CH TDM Data", NULL, "IF2 DAC R" }, + + { "DSP UL Mux", "Bypass", "8CH TDM Data" }, + { "DSP UL Mux", NULL, "I2S DSP" }, + { "DSP DL Mux", "Bypass", "RxDP Mux" }, + { "DSP DL Mux", NULL, "I2S DSP" }, + + { "TxDP_ADC_L", NULL, "DSP UL Mux" }, + { "TxDP_ADC_R", NULL, "DSP UL Mux" }, + { "TxDC_DAC", NULL, "DSP DL Mux" }, + + { "TxDP_ADC", NULL, "TxDP_ADC_L" }, + { "TxDP_ADC", NULL, "TxDP_ADC_R" }, + + { "IF1 ADC", NULL, "I2S1" }, + { "IF1 ADC", NULL, "IF1_ADC1" }, + { "IF1 ADC", NULL, "IF1_ADC2" }, + { "IF1 ADC", NULL, "IF_ADC3" }, + { "IF1 ADC", NULL, "TxDP_ADC" }, + + { "IF2 ADC Mux", "IF_ADC1", "IF_ADC1" }, + { "IF2 ADC Mux", "IF_ADC2", "IF_ADC2" }, + { "IF2 ADC Mux", "IF_ADC3", "IF_ADC3" }, + { "IF2 ADC Mux", "TxDC_DAC", "TxDC_DAC" }, + { "IF2 ADC Mux", "TxDP_ADC", "TxDP_ADC" }, + { "IF2 ADC Mux", "VAD_ADC", "VAD_ADC" }, + + { "IF2 ADC L", NULL, "IF2 ADC Mux" }, + { "IF2 ADC R", NULL, "IF2 ADC Mux" }, + + { "IF2 ADC", NULL, "I2S2" }, + { "IF2 ADC", NULL, "IF2 ADC L" }, + { "IF2 ADC", NULL, "IF2 ADC R" }, + + { "AIF1TX", NULL, "IF1 ADC" }, + { "AIF2TX", NULL, "IF2 ADC" }, + + { "IF1 DAC1", NULL, "AIF1RX" }, + { "IF1 DAC2", NULL, "AIF1RX" }, + { "IF2 DAC", NULL, "AIF2RX" }, + + { "IF1 DAC1", NULL, "I2S1" }, + { "IF1 DAC2", NULL, "I2S1" }, + { "IF2 DAC", NULL, "I2S2" }, + + { "IF1 DAC2 L", NULL, "IF1 DAC2" }, + { "IF1 DAC2 R", NULL, "IF1 DAC2" }, + { "IF1 DAC1 L", NULL, "IF1 DAC1" }, + { "IF1 DAC1 R", NULL, "IF1 DAC1" }, + { "IF2 DAC L", NULL, "IF2 DAC" }, + { "IF2 DAC R", NULL, "IF2 DAC" }, + + { "DAC1 L Mux", "IF1 DAC", "IF1 DAC1 L" }, + { "DAC1 L Mux", "IF2 DAC", "IF2 DAC L" }, + + { "DAC1 R Mux", "IF1 DAC", "IF1 DAC1 R" }, + { "DAC1 R Mux", "IF2 DAC", "IF2 DAC R" }, + + { "DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL" }, + { "DAC1 MIXL", "DAC1 Switch", "DAC1 L Mux" }, + { "DAC1 MIXL", NULL, "dac stereo1 filter" }, + { "DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR" }, + { "DAC1 MIXR", "DAC1 Switch", "DAC1 R Mux" }, + { "DAC1 MIXR", NULL, "dac stereo1 filter" }, + + { "DAC MIX", NULL, "DAC1 MIXL" }, + { "DAC MIX", NULL, "DAC1 MIXR" }, + + { "Audio DSP", NULL, "DAC1 MIXL" }, + { "Audio DSP", NULL, "DAC1 MIXR" }, + + { "DAC L2 Mux", "IF1 DAC", "IF1 DAC2 L" }, + { "DAC L2 Mux", "IF2 DAC", "IF2 DAC L" }, + { "DAC L2 Mux", "TxDC DAC", "TxDC_DAC" }, + { "DAC L2 Mux", "VAD_ADC", "VAD_ADC" }, + { "DAC L2 Volume", NULL, "DAC L2 Mux" }, + { "DAC L2 Volume", NULL, "dac mono left filter" }, + + { "DAC R2 Mux", "IF1 DAC", "IF1 DAC2 R" }, + { "DAC R2 Mux", "IF2 DAC", "IF2 DAC R" }, + { "DAC R2 Mux", "TxDC DAC", "TxDC_DAC" }, + { "DAC R2 Mux", "TxDP ADC", "TxDP_ADC" }, + { "DAC R2 Volume", NULL, "DAC R2 Mux" }, + { "DAC R2 Volume", NULL, "dac mono right filter" }, + + { "Stereo DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXL", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "Stereo DAC MIXL", NULL, "dac stereo1 filter" }, + { "Stereo DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXR", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "Stereo DAC MIXR", NULL, "dac stereo1 filter" }, + + { "Mono DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" }, + { "Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Volume" }, + { "Mono DAC MIXL", NULL, "dac mono left filter" }, + { "Mono DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" }, + { "Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" }, + { "Mono DAC MIXR", NULL, "dac mono right filter" }, + + { "DAC MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" }, + { "DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "DAC MIXL", "DAC R2 Switch", "DAC R2 Volume" }, + { "DAC MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" }, + { "DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" }, + + { "DAC L1", NULL, "DAC L1 Power" }, + { "DAC L1", NULL, "Stereo DAC MIXL" }, + { "DAC L1", NULL, "PLL1", check_sysclk1_source }, + { "DAC R1", NULL, "DAC R1 Power" }, + { "DAC R1", NULL, "Stereo DAC MIXR" }, + { "DAC R1", NULL, "PLL1", check_sysclk1_source }, + { "DAC L2", NULL, "Mono DAC MIXL" }, + { "DAC L2", NULL, "PLL1", check_sysclk1_source }, + { "DAC R2", NULL, "Mono DAC MIXR" }, + { "DAC R2", NULL, "PLL1", check_sysclk1_source }, + + { "OUT MIXL", "BST1 Switch", "BST1" }, + { "OUT MIXL", "INL Switch", "INL VOL" }, + { "OUT MIXL", "DAC L2 Switch", "DAC L2" }, + { "OUT MIXL", "DAC L1 Switch", "DAC L1" }, + + { "OUT MIXR", "BST2 Switch", "BST2" }, + { "OUT MIXR", "INR Switch", "INR VOL" }, + { "OUT MIXR", "DAC R2 Switch", "DAC R2" }, + { "OUT MIXR", "DAC R1 Switch", "DAC R1" }, + + { "HPOVOL MIXL", "DAC1 Switch", "DAC L1" }, + { "HPOVOL MIXL", "INL Switch", "INL VOL" }, + { "HPOVOL MIXR", "DAC1 Switch", "DAC R1" }, + { "HPOVOL MIXR", "INR Switch", "INR VOL" }, + + { "DAC 2", NULL, "DAC L2" }, + { "DAC 2", NULL, "DAC R2" }, + { "DAC 1", NULL, "DAC L1" }, + { "DAC 1", NULL, "DAC R1" }, + { "HPOVOL", NULL, "HPOVOL MIXL" }, + { "HPOVOL", NULL, "HPOVOL MIXR" }, + { "HPO MIX", "DAC1 Switch", "DAC 1" }, + { "HPO MIX", "HPVOL Switch", "HPOVOL" }, + + { "LOUT MIX", "DAC L1 Switch", "DAC L1" }, + { "LOUT MIX", "DAC R1 Switch", "DAC R1" }, + { "LOUT MIX", "OUTMIX L Switch", "OUT MIXL" }, + { "LOUT MIX", "OUTMIX R Switch", "OUT MIXR" }, + + { "PDM1 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, + { "PDM1 L Mux", "Mono DAC", "Mono DAC MIXL" }, + { "PDM1 L Mux", NULL, "PDM1 Power" }, + { "PDM1 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, + { "PDM1 R Mux", "Mono DAC", "Mono DAC MIXR" }, + { "PDM1 R Mux", NULL, "PDM1 Power" }, + { "PDM2 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, + { "PDM2 L Mux", "Mono DAC", "Mono DAC MIXL" }, + { "PDM2 L Mux", NULL, "PDM2 Power" }, + { "PDM2 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, + { "PDM2 R Mux", "Mono DAC", "Mono DAC MIXR" }, + { "PDM2 R Mux", NULL, "PDM2 Power" }, + + { "HP Amp", NULL, "HPO MIX" }, + { "HP Amp", NULL, "JD Power" }, + { "HP Amp", NULL, "Mic Det Power" }, + { "HPOL", NULL, "HP Amp" }, + { "HPOL", NULL, "HP L Amp" }, + { "HPOL", NULL, "Improve HP Amp Drv" }, + { "HPOR", NULL, "HP Amp" }, + { "HPOR", NULL, "HP R Amp" }, + { "HPOR", NULL, "Improve HP Amp Drv" }, + + { "LOUT Amp", NULL, "LOUT MIX" }, + { "LOUTL", NULL, "LOUT Amp" }, + { "LOUTR", NULL, "LOUT Amp" }, + { "LOUTL", NULL, "Improve HP Amp Drv" }, + { "LOUTR", NULL, "Improve HP Amp Drv" }, + + { "PDM1L", NULL, "PDM1 L Mux" }, + { "PDM1R", NULL, "PDM1 R Mux" }, + { "PDM2L", NULL, "PDM2 L Mux" }, + { "PDM2R", NULL, "PDM2 R Mux" }, +}; + +static const struct snd_soc_dapm_widget rt5670_asrc_widgets[] = { + SND_SOC_DAPM_SUPPLY("ASRC enable", SND_SOC_NOPM, 0, 0, + rt5670_asrc_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), +}; + +static const struct snd_soc_dapm_route rt5670_asrc_routes[] = { + {"I2S1", NULL, "ASRC enable"}, + {"I2S2", NULL, "ASRC enable"}, +}; + +static int get_clk_info(int sclk, int rate, bool asrc_en) +{ + int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + + if (asrc_en) + return 0; + + if (sclk <= 0 || rate <= 0) + return -EINVAL; + + rate = rate << 8; + for (i = 0; i < ARRAY_SIZE(pd); i++) + if (sclk == rate * pd[i]) + return i; + + return -EINVAL; +} + +static int rt5670_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 rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk; + int pre_div, bclk_ms, frame_size; + + if (RT5670_AIF2 == dai->id) { + snd_soc_update_bits(codec, RT5670_GPIO_CTRL1, + RT5670_I2S2_PIN_MASK, RT5670_I2S2_PIN_I2S); + } + rt5670->lrck[dai->id] = params_rate(params); + pre_div = get_clk_info(rt5670->sysclk, rt5670->lrck[dai->id], + rt5670->pdata.asrc_en); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting\n"); + return -EINVAL; + } + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return -EINVAL; + } + bclk_ms = frame_size > 32 ? 1 : 0; + rt5670->bclk[dai->id] = rt5670->lrck[dai->id] * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5670->bclk[dai->id], rt5670->lrck[dai->id]); + dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len |= RT5670_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= RT5670_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S8: + val_len |= RT5670_I2S_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5670_AIF1: + mask_clk = RT5670_I2S_BCLK_MS1_MASK | RT5670_I2S_PD1_MASK; + val_clk = bclk_ms << RT5670_I2S_BCLK_MS1_SFT | + pre_div << RT5670_I2S_PD1_SFT; + snd_soc_update_bits(codec, RT5670_I2S1_SDP, + RT5670_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5670_ADDA_CLK1, mask_clk, val_clk); + break; + case RT5670_AIF2: + mask_clk = RT5670_I2S_BCLK_MS2_MASK | RT5670_I2S_PD2_MASK; + val_clk = bclk_ms << RT5670_I2S_BCLK_MS2_SFT | + pre_div << RT5670_I2S_PD2_SFT; + snd_soc_update_bits(codec, RT5670_I2S2_SDP, + RT5670_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5670_ADDA_CLK1, mask_clk, val_clk); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5670_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + + if (RT5670_AIF2 == dai->id) { + snd_soc_update_bits(codec, RT5670_GPIO_CTRL1, + RT5670_I2S2_PIN_MASK, RT5670_I2S2_PIN_GPIO); + } + + return 0; +} + +static int rt5670_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5670->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5670_I2S_MS_S; + rt5670->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5670_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5670_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5670_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5670_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5670_AIF1: + snd_soc_update_bits(codec, RT5670_I2S1_SDP, + RT5670_I2S_MS_MASK | RT5670_I2S_BP_MASK | + RT5670_I2S_DF_MASK, reg_val); + break; + case RT5670_AIF2: + snd_soc_update_bits(codec, RT5670_I2S2_SDP, + RT5670_I2S_MS_MASK | RT5670_I2S_BP_MASK | + RT5670_I2S_DF_MASK, reg_val); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5670->sysclk && clk_id == rt5670->sysclk_src) + return 0; + + switch (clk_id) { + case RT5670_SCLK_S_MCLK: + reg_val |= RT5670_SCLK_SRC_MCLK; + break; + case RT5670_SCLK_S_PLL1: + reg_val |= RT5670_SCLK_SRC_PLL1; + break; + case RT5670_SCLK_S_RCCLK: + reg_val |= RT5670_SCLK_SRC_RCCLK; + break; + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_SCLK_SRC_MASK, reg_val); + rt5670->sysclk = freq; + rt5670->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + + return 0; +} + +/** + * rt5670_pll_calc - Calcualte PLL M/N/K code. + * @freq_in: external clock provided to codec. + * @freq_out: target clock which codec works on. + * @pll_code: Pointer to structure with M, N, K and bypass flag. + * + * Calcualte M/N/K code to configure PLL for codec. And K is assigned to 2 + * which make calculation more efficiently. + * + * Returns 0 for success or negative error code. + */ +static int rt5670_pll_calc(const unsigned int freq_in, + const unsigned int freq_out, struct rt5670_pll_code *pll_code) +{ + int max_n = RT5670_PLL_N_MAX, max_m = RT5670_PLL_M_MAX; + int k, n = 0, m = 0, red, n_t, m_t, pll_out, in_t; + int out_t, red_t = abs(freq_out - freq_in); + bool bypass = false; + + if (RT5670_PLL_INP_MAX < freq_in || RT5670_PLL_INP_MIN > freq_in) + return -EINVAL; + + k = 100000000 / freq_out - 2; + if (k > RT5670_PLL_K_MAX) + k = RT5670_PLL_K_MAX; + for (n_t = 0; n_t <= max_n; n_t++) { + in_t = freq_in / (k + 2); + pll_out = freq_out / (n_t + 2); + if (in_t < 0) + continue; + if (in_t == pll_out) { + bypass = true; + n = n_t; + goto code_find; + } + red = abs(in_t - pll_out); + if (red < red_t) { + bypass = true; + n = n_t; + m = m_t; + if (red == 0) + goto code_find; + red_t = red; + } + for (m_t = 0; m_t <= max_m; m_t++) { + out_t = in_t / (m_t + 2); + red = abs(out_t - pll_out); + if (red < red_t) { + bypass = false; + n = n_t; + m = m_t; + if (red == 0) + goto code_find; + red_t = red; + } + } + } + pr_debug("Only get approximation about PLL\n"); + +code_find: + + pll_code->m_bp = bypass; + pll_code->m_code = m; + pll_code->n_code = n; + pll_code->k_code = k; + return 0; +} + +static int rt5670_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + struct rt5670_pll_code *pll_code = &rt5670->pll_code; + int ret; + + if (source == rt5670->pll_src && freq_in == rt5670->pll_in && + freq_out == rt5670->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5670->pll_in = 0; + rt5670->pll_out = 0; + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5670_PLL1_S_MCLK: + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_MCLK); + break; + case RT5670_PLL1_S_BCLK1: + case RT5670_PLL1_S_BCLK2: + case RT5670_PLL1_S_BCLK3: + case RT5670_PLL1_S_BCLK4: + switch (dai->id) { + case RT5670_AIF1: + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_BCLK1); + break; + case RT5670_AIF2: + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_PLL1_SRC_MASK, RT5670_PLL1_SRC_BCLK2); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + break; + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rt5670_pll_calc(freq_in, freq_out, pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=%d\n", pll_code->m_bp, + (pll_code->m_bp ? 0 : pll_code->m_code), pll_code->n_code, + pll_code->k_code); + + snd_soc_write(codec, RT5670_PLL_CTRL1, + pll_code->n_code << RT5670_PLL_N_SFT | pll_code->k_code); + snd_soc_write(codec, RT5670_PLL_CTRL2, + (pll_code->m_bp ? 0 : pll_code->m_code) << RT5670_PLL_M_SFT | + pll_code->m_bp << RT5670_PLL_M_BP_SFT); + + rt5670->pll_in = freq_in; + rt5670->pll_out = freq_out; + rt5670->pll_src = source; + + return 0; +} + +static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + + if (rx_mask || tx_mask) + val |= (1 << 14); + + switch (slots) { + case 4: + val |= (1 << 12); + break; + case 6: + val |= (2 << 12); + break; + case 8: + val |= (3 << 12); + break; + case 2: + default: + break; + } + + switch (slot_width) { + case 20: + val |= (1 << 10); + break; + case 24: + val |= (2 << 10); + break; + case 32: + val |= (3 << 10); + break; + case 16: + default: + break; + } + + snd_soc_update_bits(codec, RT5670_TDM_CTRL_1, 0x7c00, val); + + return 0; +} + +static int rt5670_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 == codec->dapm.bias_level) { + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2); + mdelay(10); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_FV1 | RT5670_PWR_FV2, + RT5670_PWR_FV1 | RT5670_PWR_FV2); + snd_soc_update_bits(codec, RT5670_CHARGE_PUMP, + RT5670_OSW_L_MASK | RT5670_OSW_R_MASK, + RT5670_OSW_L_DIS | RT5670_OSW_R_DIS); + snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x1); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_LDO_SEL_MASK, 0x3); + } + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, RT5670_PWR_DIG1, 0x0000); + snd_soc_write(codec, RT5670_PWR_DIG2, 0x0001); + snd_soc_write(codec, RT5670_PWR_VOL, 0x0000); + snd_soc_write(codec, RT5670_PWR_MIXER, 0x0001); + snd_soc_write(codec, RT5670_PWR_ANLG1, 0x2800); + snd_soc_write(codec, RT5670_PWR_ANLG2, 0x0004); + snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_LDO_SEL_MASK, 0x0); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt5670_probe(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + int ret; + + rt5670->codec = codec; + codec->control_data = rt5670->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + if (rt5670->pdata.asrc_en) { + snd_soc_dapm_new_controls(&codec->dapm, rt5670_asrc_widgets, + ARRAY_SIZE(rt5670_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, rt5670_asrc_routes, + ARRAY_SIZE(rt5670_dapm_routes)); + } + + regmap_write(rt5670->regmap, RT5670_DIG_MISC, 0xc019); + regmap_write(rt5670->regmap, RT5670_ADDA_CLK1, 0x0000); + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL2, 0x0004); + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL1, 0xc900); + regmap_write(rt5670->regmap, RT5670_JD_CTRL3, 0x0088); + + rt5670->dsp_sw = RT5670_DSP_NS; + rt5670_dsp_probe(codec); + + if (rt5670->pdata.jd1_en) { + regmap_update_bits(rt5670->regmap, + RT5670_IRQ_CTRL2, 0x200, 0x200); + snd_soc_dapm_force_enable_pin(&codec->dapm, "JD Power"); + snd_soc_dapm_sync(&codec->dapm); + } + + return 0; +} + +static int rt5670_remove(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + regmap_write(rt5670->regmap, RT5670_RESET, 0); + return 0; +} + +#ifdef CONFIG_PM +static int rt5670_suspend(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5670->regmap, true); + regcache_mark_dirty(rt5670->regmap); + return 0; +} + +static int rt5670_resume(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5670->regmap, false); + regcache_sync(rt5670->regmap); + + return 0; +} +#else +#define rt5670_suspend NULL +#define rt5670_resume NULL +#endif + +#define RT5670_STEREO_RATES SNDRV_PCM_RATE_8000_96000 +#define RT5670_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_ops rt5670_aif_dai_ops = { + .hw_params = rt5670_hw_params, + .hw_free = rt5670_hw_free, + .set_fmt = rt5670_set_dai_fmt, + .set_sysclk = rt5670_set_dai_sysclk, + .set_tdm_slot = rt5670_set_tdm_slot, + .set_pll = rt5670_set_dai_pll, +}; + +struct snd_soc_dai_driver rt5670_dai[] = { + { + .name = "rt5670-aif1", + .id = RT5670_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .ops = &rt5670_aif_dai_ops, + }, + { + .name = "rt5670-aif2", + .id = RT5670_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5670_STEREO_RATES, + .formats = RT5670_FORMATS, + }, + .ops = &rt5670_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5670 = { + .probe = rt5670_probe, + .remove = rt5670_remove, + .suspend = rt5670_suspend, + .resume = rt5670_resume, + .set_bias_level = rt5670_set_bias_level, + .idle_bias_off = true, + .controls = rt5670_snd_controls, + .num_controls = ARRAY_SIZE(rt5670_snd_controls), + .dapm_widgets = rt5670_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5670_dapm_widgets), + .dapm_routes = rt5670_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5670_dapm_routes), +}; + +static const struct regmap_config rt5670_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = RT5670_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5670_ranges) * + RT5670_PR_SPACING), + .volatile_reg = rt5670_volatile_register, + .readable_reg = rt5670_readable_register, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5670_reg, + .num_reg_defaults = ARRAY_SIZE(rt5670_reg), + .ranges = rt5670_ranges, + .num_ranges = ARRAY_SIZE(rt5670_ranges), +}; + +static const struct i2c_device_id rt5670_i2c_id[] = { + { "rt5670", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5670_i2c_id); + +static int rt5670_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5670_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5670_priv *rt5670; + int ret; + unsigned int val; + + rt5670 = devm_kzalloc(&i2c->dev, + sizeof(struct rt5670_priv), + GFP_KERNEL); + if (NULL == rt5670) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5670); + + if (pdata) + rt5670->pdata = *pdata; + + rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap); + if (IS_ERR(rt5670->regmap)) { + ret = PTR_ERR(rt5670->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt5670->regmap, RT5670_VENDOR_ID2, &val); + if ((val != RT5670_DEVICE_ID)) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt5670/72\n", val); + return -ENODEV; + } + + regmap_write(rt5670->regmap, RT5670_RESET, 0); + regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, + RT5670_PWR_HP_L | RT5670_PWR_HP_R | + RT5670_PWR_VREF2, RT5670_PWR_VREF2); + msleep(1000); + + regmap_write(rt5670->regmap, RT5670_RESET, 0); + + ret = regmap_register_patch(rt5670->regmap, init_list, + ARRAY_SIZE(init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + if (rt5670->pdata.in2_diff) + regmap_update_bits(rt5670->regmap, RT5670_IN2, + RT5670_IN_DF2, RT5670_IN_DF2); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5670, + rt5670_dai, ARRAY_SIZE(rt5670_dai)); + if (ret < 0) + goto err; + + return 0; +err: + return ret; +} + +static int rt5670_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +void rt5670_i2c_shutdown(struct i2c_client *client) +{ + struct rt5670_priv *rt5670 = i2c_get_clientdata(client); + struct snd_soc_codec *codec = rt5670->codec; + + if (codec != NULL) + rt5670_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +struct i2c_driver rt5670_i2c_driver = { + .driver = { + .name = "rt5670", + .owner = THIS_MODULE, + }, + .probe = rt5670_i2c_probe, + .remove = rt5670_i2c_remove, + .shutdown = rt5670_i2c_shutdown, + .id_table = rt5670_i2c_id, +}; + +static int __init rt5670_modinit(void) +{ + return i2c_add_driver(&rt5670_i2c_driver); +} +module_init(rt5670_modinit); + +static void __exit rt5670_modexit(void) +{ + i2c_del_driver(&rt5670_i2c_driver); +} +module_exit(rt5670_modexit); + +MODULE_DESCRIPTION("ASoC RT5670 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h new file mode 100644 index 0000000..3e9b79b --- /dev/null +++ b/sound/soc/codecs/rt5670.h @@ -0,0 +1,1966 @@ +/* + * rt5670.h -- RT5670 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 __RT5670_H__ +#define __RT5670_H__ + +#include <sound/rt5670.h> + +/* Info */ +#define RT5670_RESET 0x00 +#define RT5670_VENDOR_ID 0xfd +#define RT5670_VENDOR_ID1 0xfe +#define RT5670_VENDOR_ID2 0xff +/* I/O - Output */ +#define RT5670_HP_VOL 0x02 +#define RT5670_LOUT1 0x03 +/* I/O - Input */ +#define RT5670_CJ_CTRL1 0x0a +#define RT5670_CJ_CTRL2 0x0b +#define RT5670_CJ_CTRL3 0x0c +#define RT5670_IN2 0x0e +#define RT5670_INL1_INR1_VOL 0x0f +/* I/O - ADC/DAC/DMIC */ +#define RT5670_DAC1_DIG_VOL 0x19 +#define RT5670_DAC2_DIG_VOL 0x1a +#define RT5670_DAC_CTRL 0x1b +#define RT5670_STO1_ADC_DIG_VOL 0x1c +#define RT5670_MONO_ADC_DIG_VOL 0x1d +#define RT5670_ADC_BST_VOL1 0x1e +#define RT5670_STO2_ADC_DIG_VOL 0x1f +/* Mixer - D-D */ +#define RT5670_ADC_BST_VOL2 0x20 +#define RT5670_STO2_ADC_MIXER 0x26 +#define RT5670_STO1_ADC_MIXER 0x27 +#define RT5670_MONO_ADC_MIXER 0x28 +#define RT5670_AD_DA_MIXER 0x29 +#define RT5670_STO_DAC_MIXER 0x2a +#define RT5670_DD_MIXER 0x2b +#define RT5670_DIG_MIXER 0x2c +#define RT5670_DSP_PATH1 0x2d +#define RT5670_DSP_PATH2 0x2e +#define RT5670_DIG_INF1_DATA 0x2f +#define RT5670_DIG_INF2_DATA 0x30 +/* Mixer - PDM */ +#define RT5670_PDM_OUT_CTRL 0x31 +#define RT5670_PDM_DATA_CTRL1 0x32 +#define RT5670_PDM1_DATA_CTRL2 0x33 +#define RT5670_PDM1_DATA_CTRL3 0x34 +#define RT5670_PDM1_DATA_CTRL4 0x35 +#define RT5670_PDM2_DATA_CTRL2 0x36 +#define RT5670_PDM2_DATA_CTRL3 0x37 +#define RT5670_PDM2_DATA_CTRL4 0x38 +/* Mixer - ADC */ +#define RT5670_REC_L1_MIXER 0x3b +#define RT5670_REC_L2_MIXER 0x3c +#define RT5670_REC_R1_MIXER 0x3d +#define RT5670_REC_R2_MIXER 0x3e +/* Mixer - DAC */ +#define RT5670_HPO_MIXER 0x45 +#define RT5670_MONO_MIXER 0x4c +#define RT5670_OUT_L1_MIXER 0x4f +#define RT5670_OUT_R1_MIXER 0x52 +#define RT5670_LOUT_MIXER 0x53 +/* Power */ +#define RT5670_PWR_DIG1 0x61 +#define RT5670_PWR_DIG2 0x62 +#define RT5670_PWR_ANLG1 0x63 +#define RT5670_PWR_ANLG2 0x64 +#define RT5670_PWR_MIXER 0x65 +#define RT5670_PWR_VOL 0x66 +/* Private Register Control */ +#define RT5670_PRIV_INDEX 0x6a +#define RT5670_PRIV_DATA 0x6c +/* Format - ADC/DAC */ +#define RT5670_I2S4_SDP 0x6f +#define RT5670_I2S1_SDP 0x70 +#define RT5670_I2S2_SDP 0x71 +#define RT5670_I2S3_SDP 0x72 +#define RT5670_ADDA_CLK1 0x73 +#define RT5670_ADDA_CLK2 0x74 +#define RT5670_DMIC_CTRL1 0x75 +#define RT5670_DMIC_CTRL2 0x76 +/* Format - TDM Control */ +#define RT5670_TDM_CTRL_1 0x77 +#define RT5670_TDM_CTRL_2 0x78 +#define RT5670_TDM_CTRL_3 0x79 + +/* Function - Analog */ +#define RT5670_DSP_CLK 0x7f +#define RT5670_GLB_CLK 0x80 +#define RT5670_PLL_CTRL1 0x81 +#define RT5670_PLL_CTRL2 0x82 +#define RT5670_ASRC_1 0x83 +#define RT5670_ASRC_2 0x84 +#define RT5670_ASRC_3 0x85 +#define RT5670_ASRC_4 0x86 +#define RT5670_ASRC_5 0x87 +#define RT5670_ASRC_7 0x89 +#define RT5670_ASRC_8 0x8a +#define RT5670_ASRC_9 0x8b +#define RT5670_ASRC_10 0x8c +#define RT5670_ASRC_11 0x8d +#define RT5670_DEPOP_M1 0x8e +#define RT5670_DEPOP_M2 0x8f +#define RT5670_DEPOP_M3 0x90 +#define RT5670_CHARGE_PUMP 0x91 +#define RT5670_MICBIAS 0x93 +#define RT5670_A_JD_CTRL1 0x94 +#define RT5670_A_JD_CTRL2 0x95 +#define RT5670_ASRC_12 0x97 +#define RT5670_ASRC_13 0x98 +#define RT5670_ASRC_14 0x99 +#define RT5670_VAD_CTRL1 0x9a +#define RT5670_VAD_CTRL2 0x9b +#define RT5670_VAD_CTRL3 0x9c +#define RT5670_VAD_CTRL4 0x9d +#define RT5670_VAD_CTRL5 0x9e +/* Function - Digital */ +#define RT5670_ADC_EQ_CTRL1 0xae +#define RT5670_ADC_EQ_CTRL2 0xaf +#define RT5670_EQ_CTRL1 0xb0 +#define RT5670_EQ_CTRL2 0xb1 +#define RT5670_ALC_DRC_CTRL1 0xb2 +#define RT5670_ALC_DRC_CTRL2 0xb3 +#define RT5670_ALC_CTRL_1 0xb4 +#define RT5670_ALC_CTRL_2 0xb5 +#define RT5670_ALC_CTRL_3 0xb6 +#define RT5670_ALC_CTRL_4 0xb7 +#define RT5670_JD_CTRL 0xbb +#define RT5670_IRQ_CTRL1 0xbc +#define RT5670_IRQ_CTRL2 0xbd +#define RT5670_IRQ_CTRL3 0xbe +#define RT5670_INT_IRQ_ST 0xbf +#define RT5670_GPIO_CTRL1 0xc0 +#define RT5670_GPIO_CTRL2 0xc1 +#define RT5670_GPIO_CTRL3 0xc2 +#define RT5670_SCRABBLE_FUN 0xcd +#define RT5670_SCRABBLE_CTRL 0xce +#define RT5670_BASE_BACK 0xcf +#define RT5670_MP3_PLUS1 0xd0 +#define RT5670_MP3_PLUS2 0xd1 +#define RT5670_ADJ_HPF1 0xd3 +#define RT5670_ADJ_HPF2 0xd4 +#define RT5670_HP_CALIB_AMP_DET 0xd6 +#define RT5670_SV_ZCD1 0xd9 +#define RT5670_SV_ZCD2 0xda +#define RT5670_IL_CMD 0xdb +#define RT5670_IL_CMD2 0xdc +#define RT5670_IL_CMD3 0xdd +#define RT5670_DRC_HL_CTRL1 0xe6 +#define RT5670_DRC_HL_CTRL2 0xe7 +#define RT5670_ADC_MONO_HP_CTRL1 0xec +#define RT5670_ADC_MONO_HP_CTRL2 0xed +#define RT5670_ADC_STO2_HP_CTRL1 0xee +#define RT5670_ADC_STO2_HP_CTRL2 0xef +#define RT5670_JD_CTRL3 0xf8 +#define RT5670_JD_CTRL4 0xf9 +/* General Control */ +#define RT5670_DIG_MISC 0xfa +#define RT5670_GEN_CTRL2 0xfb +#define RT5670_GEN_CTRL3 0xfc + + +/* Index of Codec Private Register definition */ +#define RT5670_DIG_VOL 0x00 +#define RT5670_PR_ALC_CTRL_1 0x01 +#define RT5670_PR_ALC_CTRL_2 0x02 +#define RT5670_PR_ALC_CTRL_3 0x03 +#define RT5670_PR_ALC_CTRL_4 0x04 +#define RT5670_PR_ALC_CTRL_5 0x05 +#define RT5670_PR_ALC_CTRL_6 0x06 +#define RT5670_BIAS_CUR1 0x12 +#define RT5670_BIAS_CUR3 0x14 +#define RT5670_CLSD_INT_REG1 0x1c +#define RT5670_MAMP_INT_REG2 0x37 +#define RT5670_CHOP_DAC_ADC 0x3d +#define RT5670_MIXER_INT_REG 0x3f +#define RT5670_3D_SPK 0x63 +#define RT5670_WND_1 0x6c +#define RT5670_WND_2 0x6d +#define RT5670_WND_3 0x6e +#define RT5670_WND_4 0x6f +#define RT5670_WND_5 0x70 +#define RT5670_WND_8 0x73 +#define RT5670_DIP_SPK_INF 0x75 +#define RT5670_HP_DCC_INT1 0x77 +#define RT5670_EQ_BW_LOP 0xa0 +#define RT5670_EQ_GN_LOP 0xa1 +#define RT5670_EQ_FC_BP1 0xa2 +#define RT5670_EQ_BW_BP1 0xa3 +#define RT5670_EQ_GN_BP1 0xa4 +#define RT5670_EQ_FC_BP2 0xa5 +#define RT5670_EQ_BW_BP2 0xa6 +#define RT5670_EQ_GN_BP2 0xa7 +#define RT5670_EQ_FC_BP3 0xa8 +#define RT5670_EQ_BW_BP3 0xa9 +#define RT5670_EQ_GN_BP3 0xaa +#define RT5670_EQ_FC_BP4 0xab +#define RT5670_EQ_BW_BP4 0xac +#define RT5670_EQ_GN_BP4 0xad +#define RT5670_EQ_FC_HIP1 0xae +#define RT5670_EQ_GN_HIP1 0xaf +#define RT5670_EQ_FC_HIP2 0xb0 +#define RT5670_EQ_BW_HIP2 0xb1 +#define RT5670_EQ_GN_HIP2 0xb2 +#define RT5670_EQ_PRE_VOL 0xb3 +#define RT5670_EQ_PST_VOL 0xb4 + + +/* global definition */ +#define RT5670_L_MUTE (0x1 << 15) +#define RT5670_L_MUTE_SFT 15 +#define RT5670_VOL_L_MUTE (0x1 << 14) +#define RT5670_VOL_L_SFT 14 +#define RT5670_R_MUTE (0x1 << 7) +#define RT5670_R_MUTE_SFT 7 +#define RT5670_VOL_R_MUTE (0x1 << 6) +#define RT5670_VOL_R_SFT 6 +#define RT5670_L_VOL_MASK (0x3f << 8) +#define RT5670_L_VOL_SFT 8 +#define RT5670_R_VOL_MASK (0x3f) +#define RT5670_R_VOL_SFT 0 + +/* Combo Jack Control 1 (0x0a) */ +#define RT5670_CBJ_BST1_MASK (0xf << 12) +#define RT5670_CBJ_BST1_SFT (12) +#define RT5670_CBJ_JD_HP_EN (0x1 << 9) +#define RT5670_CBJ_JD_MIC_EN (0x1 << 8) +#define RT5670_CBJ_BST1_EN (0x1 << 2) + +/* Combo Jack Control 1 (0x0b) */ +#define RT5670_CBJ_MN_JD (0x1 << 12) +#define RT5670_CAPLESS_EN (0x1 << 11) +#define RT5670_CBJ_DET_MODE (0x1 << 7) + +/* IN2 Control (0x0e) */ +#define RT5670_BST_MASK1 (0xf<<12) +#define RT5670_BST_SFT1 12 +#define RT5670_BST_MASK2 (0xf<<8) +#define RT5670_BST_SFT2 8 +#define RT5670_IN_DF1 (0x1 << 7) +#define RT5670_IN_SFT1 7 +#define RT5670_IN_DF2 (0x1 << 6) +#define RT5670_IN_SFT2 6 + +/* INL and INR Volume Control (0x0f) */ +#define RT5670_INL_SEL_MASK (0x1 << 15) +#define RT5670_INL_SEL_SFT 15 +#define RT5670_INL_SEL_IN4P (0x0 << 15) +#define RT5670_INL_SEL_MONOP (0x1 << 15) +#define RT5670_INL_VOL_MASK (0x1f << 8) +#define RT5670_INL_VOL_SFT 8 +#define RT5670_INR_SEL_MASK (0x1 << 7) +#define RT5670_INR_SEL_SFT 7 +#define RT5670_INR_SEL_IN4N (0x0 << 7) +#define RT5670_INR_SEL_MONON (0x1 << 7) +#define RT5670_INR_VOL_MASK (0x1f) +#define RT5670_INR_VOL_SFT 0 + +/* Sidetone Control (0x18) */ +#define RT5670_ST_SEL_MASK (0x7 << 9) +#define RT5670_ST_SEL_SFT 9 +#define RT5670_M_ST_DACR2 (0x1 << 8) +#define RT5670_M_ST_DACR2_SFT 8 +#define RT5670_M_ST_DACL2 (0x1 << 7) +#define RT5670_M_ST_DACL2_SFT 7 +#define RT5670_ST_EN (0x1 << 6) +#define RT5670_ST_EN_SFT 6 + +/* DAC1 Digital Volume (0x19) */ +#define RT5670_DAC_L1_VOL_MASK (0xff << 8) +#define RT5670_DAC_L1_VOL_SFT 8 +#define RT5670_DAC_R1_VOL_MASK (0xff) +#define RT5670_DAC_R1_VOL_SFT 0 + +/* DAC2 Digital Volume (0x1a) */ +#define RT5670_DAC_L2_VOL_MASK (0xff << 8) +#define RT5670_DAC_L2_VOL_SFT 8 +#define RT5670_DAC_R2_VOL_MASK (0xff) +#define RT5670_DAC_R2_VOL_SFT 0 + +/* DAC2 Control (0x1b) */ +#define RT5670_M_DAC_L2_VOL (0x1 << 13) +#define RT5670_M_DAC_L2_VOL_SFT 13 +#define RT5670_M_DAC_R2_VOL (0x1 << 12) +#define RT5670_M_DAC_R2_VOL_SFT 12 +#define RT5670_DAC2_L_SEL_MASK (0x7 << 4) +#define RT5670_DAC2_L_SEL_SFT 4 +#define RT5670_DAC2_R_SEL_MASK (0x7 << 0) +#define RT5670_DAC2_R_SEL_SFT 0 + +/* ADC Digital Volume Control (0x1c) */ +#define RT5670_ADC_L_VOL_MASK (0x7f << 8) +#define RT5670_ADC_L_VOL_SFT 8 +#define RT5670_ADC_R_VOL_MASK (0x7f) +#define RT5670_ADC_R_VOL_SFT 0 + +/* Mono ADC Digital Volume Control (0x1d) */ +#define RT5670_MONO_ADC_L_VOL_MASK (0x7f << 8) +#define RT5670_MONO_ADC_L_VOL_SFT 8 +#define RT5670_MONO_ADC_R_VOL_MASK (0x7f) +#define RT5670_MONO_ADC_R_VOL_SFT 0 + +/* ADC Boost Volume Control (0x1e) */ +#define RT5670_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5670_STO1_ADC_L_BST_SFT 14 +#define RT5670_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5670_STO1_ADC_R_BST_SFT 12 +#define RT5670_STO1_ADC_COMP_MASK (0x3 << 10) +#define RT5670_STO1_ADC_COMP_SFT 10 +#define RT5670_STO2_ADC_L_BST_MASK (0x3 << 8) +#define RT5670_STO2_ADC_L_BST_SFT 8 +#define RT5670_STO2_ADC_R_BST_MASK (0x3 << 6) +#define RT5670_STO2_ADC_R_BST_SFT 6 +#define RT5670_STO2_ADC_COMP_MASK (0x3 << 4) +#define RT5670_STO2_ADC_COMP_SFT 4 + +/* Stereo2 ADC Mixer Control (0x26) */ +#define RT5670_STO2_ADC_SRC_MASK (0x1 << 15) +#define RT5670_STO2_ADC_SRC_SFT 15 + +/* Stereo ADC Mixer Control (0x26 0x27) */ +#define RT5670_M_ADC_L1 (0x1 << 14) +#define RT5670_M_ADC_L1_SFT 14 +#define RT5670_M_ADC_L2 (0x1 << 13) +#define RT5670_M_ADC_L2_SFT 13 +#define RT5670_ADC_1_SRC_MASK (0x1 << 12) +#define RT5670_ADC_1_SRC_SFT 12 +#define RT5670_ADC_1_SRC_ADC (0x1 << 12) +#define RT5670_ADC_1_SRC_DACMIX (0x0 << 12) +#define RT5670_ADC_2_SRC_MASK (0x1 << 11) +#define RT5670_ADC_2_SRC_SFT 11 +#define RT5670_ADC_SRC_MASK (0x1 << 10) +#define RT5670_ADC_SRC_SFT 10 +#define RT5670_DMIC_SRC_MASK (0x3 << 8) +#define RT5670_DMIC_SRC_SFT 8 +#define RT5670_M_ADC_R1 (0x1 << 6) +#define RT5670_M_ADC_R1_SFT 6 +#define RT5670_M_ADC_R2 (0x1 << 5) +#define RT5670_M_ADC_R2_SFT 5 +#define RT5670_DMIC3_SRC_MASK (0x1 << 1) +#define RT5670_DMIC3_SRC_SFT 0 + +/* Mono ADC Mixer Control (0x28) */ +#define RT5670_M_MONO_ADC_L1 (0x1 << 14) +#define RT5670_M_MONO_ADC_L1_SFT 14 +#define RT5670_M_MONO_ADC_L2 (0x1 << 13) +#define RT5670_M_MONO_ADC_L2_SFT 13 +#define RT5670_MONO_ADC_L1_SRC_MASK (0x1 << 12) +#define RT5670_MONO_ADC_L1_SRC_SFT 12 +#define RT5670_MONO_ADC_L1_SRC_DACMIXL (0x0 << 12) +#define RT5670_MONO_ADC_L1_SRC_ADCL (0x1 << 12) +#define RT5670_MONO_ADC_L2_SRC_MASK (0x1 << 11) +#define RT5670_MONO_ADC_L2_SRC_SFT 11 +#define RT5670_MONO_ADC_L_SRC_MASK (0x1 << 10) +#define RT5670_MONO_ADC_L_SRC_SFT 10 +#define RT5670_MONO_DMIC_L_SRC_MASK (0x3 << 8) +#define RT5670_MONO_DMIC_L_SRC_SFT 8 +#define RT5670_M_MONO_ADC_R1 (0x1 << 6) +#define RT5670_M_MONO_ADC_R1_SFT 6 +#define RT5670_M_MONO_ADC_R2 (0x1 << 5) +#define RT5670_M_MONO_ADC_R2_SFT 5 +#define RT5670_MONO_ADC_R1_SRC_MASK (0x1 << 4) +#define RT5670_MONO_ADC_R1_SRC_SFT 4 +#define RT5670_MONO_ADC_R1_SRC_ADCR (0x1 << 4) +#define RT5670_MONO_ADC_R1_SRC_DACMIXR (0x0 << 4) +#define RT5670_MONO_ADC_R2_SRC_MASK (0x1 << 3) +#define RT5670_MONO_ADC_R2_SRC_SFT 3 +#define RT5670_MONO_DMIC_R_SRC_MASK (0x3) +#define RT5670_MONO_DMIC_R_SRC_SFT 0 + +/* ADC Mixer to DAC Mixer Control (0x29) */ +#define RT5670_M_ADCMIX_L (0x1 << 15) +#define RT5670_M_ADCMIX_L_SFT 15 +#define RT5670_M_DAC1_L (0x1 << 14) +#define RT5670_M_DAC1_L_SFT 14 +#define RT5670_DAC1_R_SEL_MASK (0x3 << 10) +#define RT5670_DAC1_R_SEL_SFT 10 +#define RT5670_DAC1_R_SEL_IF1 (0x0 << 10) +#define RT5670_DAC1_R_SEL_IF2 (0x1 << 10) +#define RT5670_DAC1_R_SEL_IF3 (0x2 << 10) +#define RT5670_DAC1_R_SEL_IF4 (0x3 << 10) +#define RT5670_DAC1_L_SEL_MASK (0x3 << 8) +#define RT5670_DAC1_L_SEL_SFT 8 +#define RT5670_DAC1_L_SEL_IF1 (0x0 << 8) +#define RT5670_DAC1_L_SEL_IF2 (0x1 << 8) +#define RT5670_DAC1_L_SEL_IF3 (0x2 << 8) +#define RT5670_DAC1_L_SEL_IF4 (0x3 << 8) +#define RT5670_M_ADCMIX_R (0x1 << 7) +#define RT5670_M_ADCMIX_R_SFT 7 +#define RT5670_M_DAC1_R (0x1 << 6) +#define RT5670_M_DAC1_R_SFT 6 + +/* Stereo DAC Mixer Control (0x2a) */ +#define RT5670_M_DAC_L1 (0x1 << 14) +#define RT5670_M_DAC_L1_SFT 14 +#define RT5670_DAC_L1_STO_L_VOL_MASK (0x1 << 13) +#define RT5670_DAC_L1_STO_L_VOL_SFT 13 +#define RT5670_M_DAC_L2 (0x1 << 12) +#define RT5670_M_DAC_L2_SFT 12 +#define RT5670_DAC_L2_STO_L_VOL_MASK (0x1 << 11) +#define RT5670_DAC_L2_STO_L_VOL_SFT 11 +#define RT5670_M_DAC_R1_STO_L (0x1 << 9) +#define RT5670_M_DAC_R1_STO_L_SFT 9 +#define RT5670_DAC_R1_STO_L_VOL_MASK (0x1 << 8) +#define RT5670_DAC_R1_STO_L_VOL_SFT 8 +#define RT5670_M_DAC_R1 (0x1 << 6) +#define RT5670_M_DAC_R1_SFT 6 +#define RT5670_DAC_R1_STO_R_VOL_MASK (0x1 << 5) +#define RT5670_DAC_R1_STO_R_VOL_SFT 5 +#define RT5670_M_DAC_R2 (0x1 << 4) +#define RT5670_M_DAC_R2_SFT 4 +#define RT5670_DAC_R2_STO_R_VOL_MASK (0x1 << 3) +#define RT5670_DAC_R2_STO_R_VOL_SFT 3 +#define RT5670_M_DAC_L1_STO_R (0x1 << 1) +#define RT5670_M_DAC_L1_STO_R_SFT 1 +#define RT5670_DAC_L1_STO_R_VOL_MASK (0x1) +#define RT5670_DAC_L1_STO_R_VOL_SFT 0 + +/* Mono DAC Mixer Control (0x2b) */ +#define RT5670_M_DAC_L1_MONO_L (0x1 << 14) +#define RT5670_M_DAC_L1_MONO_L_SFT 14 +#define RT5670_DAC_L1_MONO_L_VOL_MASK (0x1 << 13) +#define RT5670_DAC_L1_MONO_L_VOL_SFT 13 +#define RT5670_M_DAC_L2_MONO_L (0x1 << 12) +#define RT5670_M_DAC_L2_MONO_L_SFT 12 +#define RT5670_DAC_L2_MONO_L_VOL_MASK (0x1 << 11) +#define RT5670_DAC_L2_MONO_L_VOL_SFT 11 +#define RT5670_M_DAC_R2_MONO_L (0x1 << 10) +#define RT5670_M_DAC_R2_MONO_L_SFT 10 +#define RT5670_DAC_R2_MONO_L_VOL_MASK (0x1 << 9) +#define RT5670_DAC_R2_MONO_L_VOL_SFT 9 +#define RT5670_M_DAC_R1_MONO_R (0x1 << 6) +#define RT5670_M_DAC_R1_MONO_R_SFT 6 +#define RT5670_DAC_R1_MONO_R_VOL_MASK (0x1 << 5) +#define RT5670_DAC_R1_MONO_R_VOL_SFT 5 +#define RT5670_M_DAC_R2_MONO_R (0x1 << 4) +#define RT5670_M_DAC_R2_MONO_R_SFT 4 +#define RT5670_DAC_R2_MONO_R_VOL_MASK (0x1 << 3) +#define RT5670_DAC_R2_MONO_R_VOL_SFT 3 +#define RT5670_M_DAC_L2_MONO_R (0x1 << 2) +#define RT5670_M_DAC_L2_MONO_R_SFT 2 +#define RT5670_DAC_L2_MONO_R_VOL_MASK (0x1 << 1) +#define RT5670_DAC_L2_MONO_R_VOL_SFT 1 + +/* Digital Mixer Control (0x2c) */ +#define RT5670_M_STO_L_DAC_L (0x1 << 15) +#define RT5670_M_STO_L_DAC_L_SFT 15 +#define RT5670_STO_L_DAC_L_VOL_MASK (0x1 << 14) +#define RT5670_STO_L_DAC_L_VOL_SFT 14 +#define RT5670_M_DAC_L2_DAC_L (0x1 << 13) +#define RT5670_M_DAC_L2_DAC_L_SFT 13 +#define RT5670_DAC_L2_DAC_L_VOL_MASK (0x1 << 12) +#define RT5670_DAC_L2_DAC_L_VOL_SFT 12 +#define RT5670_M_STO_R_DAC_R (0x1 << 11) +#define RT5670_M_STO_R_DAC_R_SFT 11 +#define RT5670_STO_R_DAC_R_VOL_MASK (0x1 << 10) +#define RT5670_STO_R_DAC_R_VOL_SFT 10 +#define RT5670_M_DAC_R2_DAC_R (0x1 << 9) +#define RT5670_M_DAC_R2_DAC_R_SFT 9 +#define RT5670_DAC_R2_DAC_R_VOL_MASK (0x1 << 8) +#define RT5670_DAC_R2_DAC_R_VOL_SFT 8 +#define RT5670_M_DAC_R2_DAC_L (0x1 << 7) +#define RT5670_M_DAC_R2_DAC_L_SFT 7 +#define RT5670_DAC_R2_DAC_L_VOL_MASK (0x1 << 6) +#define RT5670_DAC_R2_DAC_L_VOL_SFT 6 +#define RT5670_M_DAC_L2_DAC_R (0x1 << 5) +#define RT5670_M_DAC_L2_DAC_R_SFT 5 +#define RT5670_DAC_L2_DAC_R_VOL_MASK (0x1 << 4) +#define RT5670_DAC_L2_DAC_R_VOL_SFT 4 + +/* DSP Path Control 1 (0x2d) */ +#define RT5670_RXDP_SEL_MASK (0x7 << 13) +#define RT5670_RXDP_SEL_SFT 13 +#define RT5670_RXDP_SRC_MASK (0x3 << 11) +#define RT5670_RXDP_SRC_SFT 11 +#define RT5670_RXDP_SRC_NOR (0x0 << 11) +#define RT5670_RXDP_SRC_DIV2 (0x1 << 11) +#define RT5670_RXDP_SRC_DIV3 (0x2 << 11) +#define RT5670_TXDP_SRC_MASK (0x3 << 4) +#define RT5670_TXDP_SRC_SFT 4 +#define RT5670_TXDP_SRC_NOR (0x0 << 4) +#define RT5670_TXDP_SRC_DIV2 (0x1 << 4) +#define RT5670_TXDP_SRC_DIV3 (0x2 << 4) +#define RT5670_DSP_UL_SEL (0x1 << 1) +#define RT5670_DSP_UL_SFT 1 +#define RT5670_DSP_DL_SEL 0x1 +#define RT5670_DSP_DL_SFT 0 + +/* DSP Path Control 2 (0x2e) */ +#define RT5670_TXDP_L_VOL_MASK (0x7f << 8) +#define RT5670_TXDP_L_VOL_SFT 8 +#define RT5670_TXDP_R_VOL_MASK (0x7f) +#define RT5670_TXDP_R_VOL_SFT 0 + +/* Digital Interface Data Control (0x2f) */ +#define RT5670_IF1_ADC2_IN_SEL (0x1 << 15) +#define RT5670_IF1_ADC2_IN_SFT 15 +#define RT5670_IF2_ADC_IN_MASK (0x7 << 12) +#define RT5670_IF2_ADC_IN_SFT 12 +#define RT5670_IF2_DAC_SEL_MASK (0x3 << 10) +#define RT5670_IF2_DAC_SEL_SFT 10 +#define RT5670_IF2_ADC_SEL_MASK (0x3 << 8) +#define RT5670_IF2_ADC_SEL_SFT 8 + +/* Digital Interface Data Control (0x30) */ +#define RT5670_IF4_ADC_IN_MASK (0x3 << 4) +#define RT5670_IF4_ADC_IN_SFT 4 + +/* PDM Output Control (0x31) */ +#define RT5670_PDM1_L_MASK (0x1 << 15) +#define RT5670_PDM1_L_SFT 15 +#define RT5670_M_PDM1_L (0x1 << 14) +#define RT5670_M_PDM1_L_SFT 14 +#define RT5670_PDM1_R_MASK (0x1 << 13) +#define RT5670_PDM1_R_SFT 13 +#define RT5670_M_PDM1_R (0x1 << 12) +#define RT5670_M_PDM1_R_SFT 12 +#define RT5670_PDM2_L_MASK (0x1 << 11) +#define RT5670_PDM2_L_SFT 11 +#define RT5670_M_PDM2_L (0x1 << 10) +#define RT5670_M_PDM2_L_SFT 10 +#define RT5670_PDM2_R_MASK (0x1 << 9) +#define RT5670_PDM2_R_SFT 9 +#define RT5670_M_PDM2_R (0x1 << 8) +#define RT5670_M_PDM2_R_SFT 8 +#define RT5670_PDM2_BUSY (0x1 << 7) +#define RT5670_PDM1_BUSY (0x1 << 6) +#define RT5670_PDM_PATTERN (0x1 << 5) +#define RT5670_PDM_GAIN (0x1 << 4) +#define RT5670_PDM_DIV_MASK (0x3) + +/* REC Left Mixer Control 1 (0x3b) */ +#define RT5670_G_HP_L_RM_L_MASK (0x7 << 13) +#define RT5670_G_HP_L_RM_L_SFT 13 +#define RT5670_G_IN_L_RM_L_MASK (0x7 << 10) +#define RT5670_G_IN_L_RM_L_SFT 10 +#define RT5670_G_BST4_RM_L_MASK (0x7 << 7) +#define RT5670_G_BST4_RM_L_SFT 7 +#define RT5670_G_BST3_RM_L_MASK (0x7 << 4) +#define RT5670_G_BST3_RM_L_SFT 4 +#define RT5670_G_BST2_RM_L_MASK (0x7 << 1) +#define RT5670_G_BST2_RM_L_SFT 1 + +/* REC Left Mixer Control 2 (0x3c) */ +#define RT5670_G_BST1_RM_L_MASK (0x7 << 13) +#define RT5670_G_BST1_RM_L_SFT 13 +#define RT5670_M_IN_L_RM_L (0x1 << 5) +#define RT5670_M_IN_L_RM_L_SFT 5 +#define RT5670_M_BST2_RM_L (0x1 << 3) +#define RT5670_M_BST2_RM_L_SFT 3 +#define RT5670_M_BST1_RM_L (0x1 << 1) +#define RT5670_M_BST1_RM_L_SFT 1 + +/* REC Right Mixer Control 1 (0x3d) */ +#define RT5670_G_HP_R_RM_R_MASK (0x7 << 13) +#define RT5670_G_HP_R_RM_R_SFT 13 +#define RT5670_G_IN_R_RM_R_MASK (0x7 << 10) +#define RT5670_G_IN_R_RM_R_SFT 10 +#define RT5670_G_BST4_RM_R_MASK (0x7 << 7) +#define RT5670_G_BST4_RM_R_SFT 7 +#define RT5670_G_BST3_RM_R_MASK (0x7 << 4) +#define RT5670_G_BST3_RM_R_SFT 4 +#define RT5670_G_BST2_RM_R_MASK (0x7 << 1) +#define RT5670_G_BST2_RM_R_SFT 1 + +/* REC Right Mixer Control 2 (0x3e) */ +#define RT5670_G_BST1_RM_R_MASK (0x7 << 13) +#define RT5670_G_BST1_RM_R_SFT 13 +#define RT5670_M_IN_R_RM_R (0x1 << 5) +#define RT5670_M_IN_R_RM_R_SFT 5 +#define RT5670_M_BST2_RM_R (0x1 << 3) +#define RT5670_M_BST2_RM_R_SFT 3 +#define RT5670_M_BST1_RM_R (0x1 << 1) +#define RT5670_M_BST1_RM_R_SFT 1 + +/* HPMIX Control (0x45) */ +#define RT5670_M_DAC2_HM (0x1 << 15) +#define RT5670_M_DAC2_HM_SFT 15 +#define RT5670_M_HPVOL_HM (0x1 << 14) +#define RT5670_M_HPVOL_HM_SFT 14 +#define RT5670_M_DAC1_HM (0x1 << 13) +#define RT5670_M_DAC1_HM_SFT 13 +#define RT5670_G_HPOMIX_MASK (0x1 << 12) +#define RT5670_G_HPOMIX_SFT 12 +#define RT5670_M_INR1_HMR (0x1 << 3) +#define RT5670_M_INR1_HMR_SFT 3 +#define RT5670_M_DACR1_HMR (0x1 << 2) +#define RT5670_M_DACR1_HMR_SFT 2 +#define RT5670_M_INL1_HML (0x1 << 1) +#define RT5670_M_INL1_HML_SFT 1 +#define RT5670_M_DACL1_HML (0x1) +#define RT5670_M_DACL1_HML_SFT 0 + +/* Mono Output Mixer Control (0x4c) */ +#define RT5670_M_DAC_R2_MA (0x1 << 15) +#define RT5670_M_DAC_R2_MA_SFT 15 +#define RT5670_M_DAC_L2_MA (0x1 << 14) +#define RT5670_M_DAC_L2_MA_SFT 14 +#define RT5670_M_OV_R_MM (0x1 << 13) +#define RT5670_M_OV_R_MM_SFT 13 +#define RT5670_M_OV_L_MM (0x1 << 12) +#define RT5670_M_OV_L_MM_SFT 12 +#define RT5670_G_MONOMIX_MASK (0x1 << 10) +#define RT5670_G_MONOMIX_SFT 10 +#define RT5670_M_DAC_R2_MM (0x1 << 9) +#define RT5670_M_DAC_R2_MM_SFT 9 +#define RT5670_M_DAC_L2_MM (0x1 << 8) +#define RT5670_M_DAC_L2_MM_SFT 8 +#define RT5670_M_BST4_MM (0x1 << 7) +#define RT5670_M_BST4_MM_SFT 7 + +/* Output Left Mixer Control 1 (0x4d) */ +#define RT5670_G_BST3_OM_L_MASK (0x7 << 13) +#define RT5670_G_BST3_OM_L_SFT 13 +#define RT5670_G_BST2_OM_L_MASK (0x7 << 10) +#define RT5670_G_BST2_OM_L_SFT 10 +#define RT5670_G_BST1_OM_L_MASK (0x7 << 7) +#define RT5670_G_BST1_OM_L_SFT 7 +#define RT5670_G_IN_L_OM_L_MASK (0x7 << 4) +#define RT5670_G_IN_L_OM_L_SFT 4 +#define RT5670_G_RM_L_OM_L_MASK (0x7 << 1) +#define RT5670_G_RM_L_OM_L_SFT 1 + +/* Output Left Mixer Control 2 (0x4e) */ +#define RT5670_G_DAC_R2_OM_L_MASK (0x7 << 13) +#define RT5670_G_DAC_R2_OM_L_SFT 13 +#define RT5670_G_DAC_L2_OM_L_MASK (0x7 << 10) +#define RT5670_G_DAC_L2_OM_L_SFT 10 +#define RT5670_G_DAC_L1_OM_L_MASK (0x7 << 7) +#define RT5670_G_DAC_L1_OM_L_SFT 7 + +/* Output Left Mixer Control 3 (0x4f) */ +#define RT5670_M_BST1_OM_L (0x1 << 5) +#define RT5670_M_BST1_OM_L_SFT 5 +#define RT5670_M_IN_L_OM_L (0x1 << 4) +#define RT5670_M_IN_L_OM_L_SFT 4 +#define RT5670_M_DAC_L2_OM_L (0x1 << 1) +#define RT5670_M_DAC_L2_OM_L_SFT 1 +#define RT5670_M_DAC_L1_OM_L (0x1) +#define RT5670_M_DAC_L1_OM_L_SFT 0 + +/* Output Right Mixer Control 1 (0x50) */ +#define RT5670_G_BST4_OM_R_MASK (0x7 << 13) +#define RT5670_G_BST4_OM_R_SFT 13 +#define RT5670_G_BST2_OM_R_MASK (0x7 << 10) +#define RT5670_G_BST2_OM_R_SFT 10 +#define RT5670_G_BST1_OM_R_MASK (0x7 << 7) +#define RT5670_G_BST1_OM_R_SFT 7 +#define RT5670_G_IN_R_OM_R_MASK (0x7 << 4) +#define RT5670_G_IN_R_OM_R_SFT 4 +#define RT5670_G_RM_R_OM_R_MASK (0x7 << 1) +#define RT5670_G_RM_R_OM_R_SFT 1 + +/* Output Right Mixer Control 2 (0x51) */ +#define RT5670_G_DAC_L2_OM_R_MASK (0x7 << 13) +#define RT5670_G_DAC_L2_OM_R_SFT 13 +#define RT5670_G_DAC_R2_OM_R_MASK (0x7 << 10) +#define RT5670_G_DAC_R2_OM_R_SFT 10 +#define RT5670_G_DAC_R1_OM_R_MASK (0x7 << 7) +#define RT5670_G_DAC_R1_OM_R_SFT 7 + +/* Output Right Mixer Control 3 (0x52) */ +#define RT5670_M_BST2_OM_R (0x1 << 6) +#define RT5670_M_BST2_OM_R_SFT 6 +#define RT5670_M_IN_R_OM_R (0x1 << 4) +#define RT5670_M_IN_R_OM_R_SFT 4 +#define RT5670_M_DAC_R2_OM_R (0x1 << 1) +#define RT5670_M_DAC_R2_OM_R_SFT 1 +#define RT5670_M_DAC_R1_OM_R (0x1) +#define RT5670_M_DAC_R1_OM_R_SFT 0 + +/* LOUT Mixer Control (0x53) */ +#define RT5670_M_DAC_L1_LM (0x1 << 15) +#define RT5670_M_DAC_L1_LM_SFT 15 +#define RT5670_M_DAC_R1_LM (0x1 << 14) +#define RT5670_M_DAC_R1_LM_SFT 14 +#define RT5670_M_OV_L_LM (0x1 << 13) +#define RT5670_M_OV_L_LM_SFT 13 +#define RT5670_M_OV_R_LM (0x1 << 12) +#define RT5670_M_OV_R_LM_SFT 12 +#define RT5670_G_LOUTMIX_MASK (0x1 << 11) +#define RT5670_G_LOUTMIX_SFT 11 + +/* Power Management for Digital 1 (0x61) */ +#define RT5670_PWR_I2S1 (0x1 << 15) +#define RT5670_PWR_I2S1_BIT 15 +#define RT5670_PWR_I2S2 (0x1 << 14) +#define RT5670_PWR_I2S2_BIT 14 +#define RT5670_PWR_DAC_L1 (0x1 << 12) +#define RT5670_PWR_DAC_L1_BIT 12 +#define RT5670_PWR_DAC_R1 (0x1 << 11) +#define RT5670_PWR_DAC_R1_BIT 11 +#define RT5670_PWR_DAC_L2 (0x1 << 7) +#define RT5670_PWR_DAC_L2_BIT 7 +#define RT5670_PWR_DAC_R2 (0x1 << 6) +#define RT5670_PWR_DAC_R2_BIT 6 +#define RT5670_PWR_ADC_L (0x1 << 2) +#define RT5670_PWR_ADC_L_BIT 2 +#define RT5670_PWR_ADC_R (0x1 << 1) +#define RT5670_PWR_ADC_R_BIT 1 +#define RT5670_PWR_CLS_D (0x1) +#define RT5670_PWR_CLS_D_BIT 0 + +/* Power Management for Digital 2 (0x62) */ +#define RT5670_PWR_ADC_S1F (0x1 << 15) +#define RT5670_PWR_ADC_S1F_BIT 15 +#define RT5670_PWR_ADC_MF_L (0x1 << 14) +#define RT5670_PWR_ADC_MF_L_BIT 14 +#define RT5670_PWR_ADC_MF_R (0x1 << 13) +#define RT5670_PWR_ADC_MF_R_BIT 13 +#define RT5670_PWR_I2S_DSP (0x1 << 12) +#define RT5670_PWR_I2S_DSP_BIT 12 +#define RT5670_PWR_DAC_S1F (0x1 << 11) +#define RT5670_PWR_DAC_S1F_BIT 11 +#define RT5670_PWR_DAC_MF_L (0x1 << 10) +#define RT5670_PWR_DAC_MF_L_BIT 10 +#define RT5670_PWR_DAC_MF_R (0x1 << 9) +#define RT5670_PWR_DAC_MF_R_BIT 9 +#define RT5670_PWR_ADC_S2F (0x1 << 8) +#define RT5670_PWR_ADC_S2F_BIT 8 +#define RT5670_PWR_PDM1 (0x1 << 7) +#define RT5670_PWR_PDM1_BIT 7 +#define RT5670_PWR_PDM2 (0x1 << 6) +#define RT5670_PWR_PDM2_BIT 6 + +/* Power Management for Analog 1 (0x63) */ +#define RT5670_PWR_VREF1 (0x1 << 15) +#define RT5670_PWR_VREF1_BIT 15 +#define RT5670_PWR_FV1 (0x1 << 14) +#define RT5670_PWR_FV1_BIT 14 +#define RT5670_PWR_MB (0x1 << 13) +#define RT5670_PWR_MB_BIT 13 +#define RT5670_PWR_LM (0x1 << 12) +#define RT5670_PWR_LM_BIT 12 +#define RT5670_PWR_BG (0x1 << 11) +#define RT5670_PWR_BG_BIT 11 +#define RT5670_PWR_HP_L (0x1 << 7) +#define RT5670_PWR_HP_L_BIT 7 +#define RT5670_PWR_HP_R (0x1 << 6) +#define RT5670_PWR_HP_R_BIT 6 +#define RT5670_PWR_HA (0x1 << 5) +#define RT5670_PWR_HA_BIT 5 +#define RT5670_PWR_VREF2 (0x1 << 4) +#define RT5670_PWR_VREF2_BIT 4 +#define RT5670_PWR_FV2 (0x1 << 3) +#define RT5670_PWR_FV2_BIT 3 +#define RT5670_LDO_SEL_MASK (0x3) +#define RT5670_LDO_SEL_SFT 0 + +/* Power Management for Analog 2 (0x64) */ +#define RT5670_PWR_BST1 (0x1 << 15) +#define RT5670_PWR_BST1_BIT 15 +#define RT5670_PWR_BST2 (0x1 << 13) +#define RT5670_PWR_BST2_BIT 13 +#define RT5670_PWR_MB1 (0x1 << 11) +#define RT5670_PWR_MB1_BIT 11 +#define RT5670_PWR_MB2 (0x1 << 10) +#define RT5670_PWR_MB2_BIT 10 +#define RT5670_PWR_PLL (0x1 << 9) +#define RT5670_PWR_PLL_BIT 9 +#define RT5670_PWR_BST1_P (0x1 << 6) +#define RT5670_PWR_BST1_P_BIT 6 +#define RT5670_PWR_BST2_P (0x1 << 4) +#define RT5670_PWR_BST2_P_BIT 4 +#define RT5670_PWR_JD1 (0x1 << 2) +#define RT5670_PWR_JD1_BIT 2 +#define RT5670_PWR_JD (0x1 << 1) +#define RT5670_PWR_JD_BIT 1 + +/* Power Management for Mixer (0x65) */ +#define RT5670_PWR_OM_L (0x1 << 15) +#define RT5670_PWR_OM_L_BIT 15 +#define RT5670_PWR_OM_R (0x1 << 14) +#define RT5670_PWR_OM_R_BIT 14 +#define RT5670_PWR_RM_L (0x1 << 11) +#define RT5670_PWR_RM_L_BIT 11 +#define RT5670_PWR_RM_R (0x1 << 10) +#define RT5670_PWR_RM_R_BIT 10 + +/* Power Management for Volume (0x66) */ +#define RT5670_PWR_HV_L (0x1 << 11) +#define RT5670_PWR_HV_L_BIT 11 +#define RT5670_PWR_HV_R (0x1 << 10) +#define RT5670_PWR_HV_R_BIT 10 +#define RT5670_PWR_IN_L (0x1 << 9) +#define RT5670_PWR_IN_L_BIT 9 +#define RT5670_PWR_IN_R (0x1 << 8) +#define RT5670_PWR_IN_R_BIT 8 +#define RT5670_PWR_MIC_DET (0x1 << 5) +#define RT5670_PWR_MIC_DET_BIT 5 + +/* I2S1/2/3 Audio Serial Data Port Control (0x70 0x71 0x72) */ +#define RT5670_I2S_MS_MASK (0x1 << 15) +#define RT5670_I2S_MS_SFT 15 +#define RT5670_I2S_MS_M (0x0 << 15) +#define RT5670_I2S_MS_S (0x1 << 15) +#define RT5670_I2S_IF_MASK (0x7 << 12) +#define RT5670_I2S_IF_SFT 12 +#define RT5670_I2S_O_CP_MASK (0x3 << 10) +#define RT5670_I2S_O_CP_SFT 10 +#define RT5670_I2S_O_CP_OFF (0x0 << 10) +#define RT5670_I2S_O_CP_U_LAW (0x1 << 10) +#define RT5670_I2S_O_CP_A_LAW (0x2 << 10) +#define RT5670_I2S_I_CP_MASK (0x3 << 8) +#define RT5670_I2S_I_CP_SFT 8 +#define RT5670_I2S_I_CP_OFF (0x0 << 8) +#define RT5670_I2S_I_CP_U_LAW (0x1 << 8) +#define RT5670_I2S_I_CP_A_LAW (0x2 << 8) +#define RT5670_I2S_BP_MASK (0x1 << 7) +#define RT5670_I2S_BP_SFT 7 +#define RT5670_I2S_BP_NOR (0x0 << 7) +#define RT5670_I2S_BP_INV (0x1 << 7) +#define RT5670_I2S_DL_MASK (0x3 << 2) +#define RT5670_I2S_DL_SFT 2 +#define RT5670_I2S_DL_16 (0x0 << 2) +#define RT5670_I2S_DL_20 (0x1 << 2) +#define RT5670_I2S_DL_24 (0x2 << 2) +#define RT5670_I2S_DL_8 (0x3 << 2) +#define RT5670_I2S_DF_MASK (0x3) +#define RT5670_I2S_DF_SFT 0 +#define RT5670_I2S_DF_I2S (0x0) +#define RT5670_I2S_DF_LEFT (0x1) +#define RT5670_I2S_DF_PCM_A (0x2) +#define RT5670_I2S_DF_PCM_B (0x3) + +/* I2S2 Audio Serial Data Port Control (0x71) */ +#define RT5670_I2S2_SDI_MASK (0x1 << 6) +#define RT5670_I2S2_SDI_SFT 6 +#define RT5670_I2S2_SDI_I2S1 (0x0 << 6) +#define RT5670_I2S2_SDI_I2S2 (0x1 << 6) + +/* ADC/DAC Clock Control 1 (0x73) */ +#define RT5670_I2S_BCLK_MS1_MASK (0x1 << 15) +#define RT5670_I2S_BCLK_MS1_SFT 15 +#define RT5670_I2S_BCLK_MS1_32 (0x0 << 15) +#define RT5670_I2S_BCLK_MS1_64 (0x1 << 15) +#define RT5670_I2S_PD1_MASK (0x7 << 12) +#define RT5670_I2S_PD1_SFT 12 +#define RT5670_I2S_PD1_1 (0x0 << 12) +#define RT5670_I2S_PD1_2 (0x1 << 12) +#define RT5670_I2S_PD1_3 (0x2 << 12) +#define RT5670_I2S_PD1_4 (0x3 << 12) +#define RT5670_I2S_PD1_6 (0x4 << 12) +#define RT5670_I2S_PD1_8 (0x5 << 12) +#define RT5670_I2S_PD1_12 (0x6 << 12) +#define RT5670_I2S_PD1_16 (0x7 << 12) +#define RT5670_I2S_BCLK_MS2_MASK (0x1 << 11) +#define RT5670_I2S_BCLK_MS2_SFT 11 +#define RT5670_I2S_BCLK_MS2_32 (0x0 << 11) +#define RT5670_I2S_BCLK_MS2_64 (0x1 << 11) +#define RT5670_I2S_PD2_MASK (0x7 << 8) +#define RT5670_I2S_PD2_SFT 8 +#define RT5670_I2S_PD2_1 (0x0 << 8) +#define RT5670_I2S_PD2_2 (0x1 << 8) +#define RT5670_I2S_PD2_3 (0x2 << 8) +#define RT5670_I2S_PD2_4 (0x3 << 8) +#define RT5670_I2S_PD2_6 (0x4 << 8) +#define RT5670_I2S_PD2_8 (0x5 << 8) +#define RT5670_I2S_PD2_12 (0x6 << 8) +#define RT5670_I2S_PD2_16 (0x7 << 8) +#define RT5670_I2S_BCLK_MS3_MASK (0x1 << 7) +#define RT5670_I2S_BCLK_MS3_SFT 7 +#define RT5670_I2S_BCLK_MS3_32 (0x0 << 7) +#define RT5670_I2S_BCLK_MS3_64 (0x1 << 7) +#define RT5670_I2S_PD3_MASK (0x7 << 4) +#define RT5670_I2S_PD3_SFT 4 +#define RT5670_I2S_PD3_1 (0x0 << 4) +#define RT5670_I2S_PD3_2 (0x1 << 4) +#define RT5670_I2S_PD3_3 (0x2 << 4) +#define RT5670_I2S_PD3_4 (0x3 << 4) +#define RT5670_I2S_PD3_6 (0x4 << 4) +#define RT5670_I2S_PD3_8 (0x5 << 4) +#define RT5670_I2S_PD3_12 (0x6 << 4) +#define RT5670_I2S_PD3_16 (0x7 << 4) +#define RT5670_DAC_OSR_MASK (0x3 << 2) +#define RT5670_DAC_OSR_SFT 2 +#define RT5670_DAC_OSR_128 (0x0 << 2) +#define RT5670_DAC_OSR_64 (0x1 << 2) +#define RT5670_DAC_OSR_32 (0x2 << 2) +#define RT5670_DAC_OSR_16 (0x3 << 2) +#define RT5670_ADC_OSR_MASK (0x3) +#define RT5670_ADC_OSR_SFT 0 +#define RT5670_ADC_OSR_128 (0x0) +#define RT5670_ADC_OSR_64 (0x1) +#define RT5670_ADC_OSR_32 (0x2) +#define RT5670_ADC_OSR_16 (0x3) + +/* ADC/DAC Clock Control 2 (0x74) */ +#define RT5670_DAC_L_OSR_MASK (0x3 << 14) +#define RT5670_DAC_L_OSR_SFT 14 +#define RT5670_DAC_L_OSR_128 (0x0 << 14) +#define RT5670_DAC_L_OSR_64 (0x1 << 14) +#define RT5670_DAC_L_OSR_32 (0x2 << 14) +#define RT5670_DAC_L_OSR_16 (0x3 << 14) +#define RT5670_ADC_R_OSR_MASK (0x3 << 12) +#define RT5670_ADC_R_OSR_SFT 12 +#define RT5670_ADC_R_OSR_128 (0x0 << 12) +#define RT5670_ADC_R_OSR_64 (0x1 << 12) +#define RT5670_ADC_R_OSR_32 (0x2 << 12) +#define RT5670_ADC_R_OSR_16 (0x3 << 12) +#define RT5670_DAHPF_EN (0x1 << 11) +#define RT5670_DAHPF_EN_SFT 11 +#define RT5670_ADHPF_EN (0x1 << 10) +#define RT5670_ADHPF_EN_SFT 10 + +/* Digital Microphone Control (0x75) */ +#define RT5670_DMIC_1_EN_MASK (0x1 << 15) +#define RT5670_DMIC_1_EN_SFT 15 +#define RT5670_DMIC_1_DIS (0x0 << 15) +#define RT5670_DMIC_1_EN (0x1 << 15) +#define RT5670_DMIC_2_EN_MASK (0x1 << 14) +#define RT5670_DMIC_2_EN_SFT 14 +#define RT5670_DMIC_2_DIS (0x0 << 14) +#define RT5670_DMIC_2_EN (0x1 << 14) +#define RT5670_DMIC_1L_LH_MASK (0x1 << 13) +#define RT5670_DMIC_1L_LH_SFT 13 +#define RT5670_DMIC_1L_LH_FALLING (0x0 << 13) +#define RT5670_DMIC_1L_LH_RISING (0x1 << 13) +#define RT5670_DMIC_1R_LH_MASK (0x1 << 12) +#define RT5670_DMIC_1R_LH_SFT 12 +#define RT5670_DMIC_1R_LH_FALLING (0x0 << 12) +#define RT5670_DMIC_1R_LH_RISING (0x1 << 12) +#define RT5670_DMIC_2_DP_MASK (0x1 << 10) +#define RT5670_DMIC_2_DP_SFT 10 +#define RT5670_DMIC_2_DP_GPIO4 (0x0 << 10) +#define RT5670_DMIC_2_DP_IN1N (0x1 << 10) +#define RT5670_DMIC_2L_LH_MASK (0x1 << 9) +#define RT5670_DMIC_2L_LH_SFT 9 +#define RT5670_DMIC_2L_LH_FALLING (0x0 << 9) +#define RT5670_DMIC_2L_LH_RISING (0x1 << 9) +#define RT5670_DMIC_2R_LH_MASK (0x1 << 8) +#define RT5670_DMIC_2R_LH_SFT 8 +#define RT5670_DMIC_2R_LH_FALLING (0x0 << 8) +#define RT5670_DMIC_2R_LH_RISING (0x1 << 8) +#define RT5670_DMIC_CLK_MASK (0x7 << 5) +#define RT5670_DMIC_CLK_SFT 5 +#define RT5670_DMIC_3_EN_MASK (0x1 << 4) +#define RT5670_DMIC_3_EN_SFT 4 +#define RT5670_DMIC_3_DIS (0x0 << 4) +#define RT5670_DMIC_3_EN (0x1 << 4) +#define RT5670_DMIC_1_DP_MASK (0x3 << 0) +#define RT5670_DMIC_1_DP_SFT 0 +#define RT5670_DMIC_1_DP_GPIO6 (0x0 << 0) +#define RT5670_DMIC_1_DP_IN2P (0x1 << 0) +#define RT5670_DMIC_1_DP_GPIO7 (0x2 << 0) + +/* Global Clock Control (0x80) */ +#define RT5670_SCLK_SRC_MASK (0x3 << 14) +#define RT5670_SCLK_SRC_SFT 14 +#define RT5670_SCLK_SRC_MCLK (0x0 << 14) +#define RT5670_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5670_SCLK_SRC_RCCLK (0x2 << 14) /* 15MHz */ +#define RT5670_PLL1_SRC_MASK (0x3 << 12) +#define RT5670_PLL1_SRC_SFT 12 +#define RT5670_PLL1_SRC_MCLK (0x0 << 12) +#define RT5670_PLL1_SRC_BCLK1 (0x1 << 12) +#define RT5670_PLL1_SRC_BCLK2 (0x2 << 12) +#define RT5670_PLL1_SRC_BCLK3 (0x3 << 12) +#define RT5670_PLL1_PD_MASK (0x1 << 3) +#define RT5670_PLL1_PD_SFT 3 +#define RT5670_PLL1_PD_1 (0x0 << 3) +#define RT5670_PLL1_PD_2 (0x1 << 3) + +#define RT5670_PLL_INP_MAX 40000000 +#define RT5670_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x81) */ +#define RT5670_PLL_N_MAX 0x1ff +#define RT5670_PLL_N_MASK (RT5670_PLL_N_MAX << 7) +#define RT5670_PLL_N_SFT 7 +#define RT5670_PLL_K_MAX 0x1f +#define RT5670_PLL_K_MASK (RT5670_PLL_K_MAX) +#define RT5670_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x82) */ +#define RT5670_PLL_M_MAX 0xf +#define RT5670_PLL_M_MASK (RT5670_PLL_M_MAX << 12) +#define RT5670_PLL_M_SFT 12 +#define RT5670_PLL_M_BP (0x1 << 11) +#define RT5670_PLL_M_BP_SFT 11 + +/* ASRC Control 1 (0x83) */ +#define RT5670_STO_T_MASK (0x1 << 15) +#define RT5670_STO_T_SFT 15 +#define RT5670_STO_T_SCLK (0x0 << 15) +#define RT5670_STO_T_LRCK1 (0x1 << 15) +#define RT5670_M1_T_MASK (0x1 << 14) +#define RT5670_M1_T_SFT 14 +#define RT5670_M1_T_I2S2 (0x0 << 14) +#define RT5670_M1_T_I2S2_D3 (0x1 << 14) +#define RT5670_I2S2_F_MASK (0x1 << 12) +#define RT5670_I2S2_F_SFT 12 +#define RT5670_I2S2_F_I2S2_D2 (0x0 << 12) +#define RT5670_I2S2_F_I2S1_TCLK (0x1 << 12) +#define RT5670_DMIC_1_M_MASK (0x1 << 9) +#define RT5670_DMIC_1_M_SFT 9 +#define RT5670_DMIC_1_M_NOR (0x0 << 9) +#define RT5670_DMIC_1_M_ASYN (0x1 << 9) +#define RT5670_DMIC_2_M_MASK (0x1 << 8) +#define RT5670_DMIC_2_M_SFT 8 +#define RT5670_DMIC_2_M_NOR (0x0 << 8) +#define RT5670_DMIC_2_M_ASYN (0x1 << 8) + +/* ASRC Control 2 (0x84) */ +#define RT5670_MDA_L_M_MASK (0x1 << 15) +#define RT5670_MDA_L_M_SFT 15 +#define RT5670_MDA_L_M_NOR (0x0 << 15) +#define RT5670_MDA_L_M_ASYN (0x1 << 15) +#define RT5670_MDA_R_M_MASK (0x1 << 14) +#define RT5670_MDA_R_M_SFT 14 +#define RT5670_MDA_R_M_NOR (0x0 << 14) +#define RT5670_MDA_R_M_ASYN (0x1 << 14) +#define RT5670_MAD_L_M_MASK (0x1 << 13) +#define RT5670_MAD_L_M_SFT 13 +#define RT5670_MAD_L_M_NOR (0x0 << 13) +#define RT5670_MAD_L_M_ASYN (0x1 << 13) +#define RT5670_MAD_R_M_MASK (0x1 << 12) +#define RT5670_MAD_R_M_SFT 12 +#define RT5670_MAD_R_M_NOR (0x0 << 12) +#define RT5670_MAD_R_M_ASYN (0x1 << 12) +#define RT5670_ADC_M_MASK (0x1 << 11) +#define RT5670_ADC_M_SFT 11 +#define RT5670_ADC_M_NOR (0x0 << 11) +#define RT5670_ADC_M_ASYN (0x1 << 11) +#define RT5670_STO_DAC_M_MASK (0x1 << 5) +#define RT5670_STO_DAC_M_SFT 5 +#define RT5670_STO_DAC_M_NOR (0x0 << 5) +#define RT5670_STO_DAC_M_ASYN (0x1 << 5) +#define RT5670_I2S1_R_D_MASK (0x1 << 4) +#define RT5670_I2S1_R_D_SFT 4 +#define RT5670_I2S1_R_D_DIS (0x0 << 4) +#define RT5670_I2S1_R_D_EN (0x1 << 4) +#define RT5670_I2S2_R_D_MASK (0x1 << 3) +#define RT5670_I2S2_R_D_SFT 3 +#define RT5670_I2S2_R_D_DIS (0x0 << 3) +#define RT5670_I2S2_R_D_EN (0x1 << 3) +#define RT5670_PRE_SCLK_MASK (0x3) +#define RT5670_PRE_SCLK_SFT 0 +#define RT5670_PRE_SCLK_512 (0x0) +#define RT5670_PRE_SCLK_1024 (0x1) +#define RT5670_PRE_SCLK_2048 (0x2) + +/* ASRC Control 3 (0x85) */ +#define RT5670_I2S1_RATE_MASK (0xf << 12) +#define RT5670_I2S1_RATE_SFT 12 +#define RT5670_I2S2_RATE_MASK (0xf << 8) +#define RT5670_I2S2_RATE_SFT 8 + +/* ASRC Control 4 (0x89) */ +#define RT5670_I2S1_PD_MASK (0x7 << 12) +#define RT5670_I2S1_PD_SFT 12 +#define RT5670_I2S2_PD_MASK (0x7 << 8) +#define RT5670_I2S2_PD_SFT 8 + +/* HPOUT Over Current Detection (0x8b) */ +#define RT5670_HP_OVCD_MASK (0x1 << 10) +#define RT5670_HP_OVCD_SFT 10 +#define RT5670_HP_OVCD_DIS (0x0 << 10) +#define RT5670_HP_OVCD_EN (0x1 << 10) +#define RT5670_HP_OC_TH_MASK (0x3 << 8) +#define RT5670_HP_OC_TH_SFT 8 +#define RT5670_HP_OC_TH_90 (0x0 << 8) +#define RT5670_HP_OC_TH_105 (0x1 << 8) +#define RT5670_HP_OC_TH_120 (0x2 << 8) +#define RT5670_HP_OC_TH_135 (0x3 << 8) + +/* Class D Over Current Control (0x8c) */ +#define RT5670_CLSD_OC_MASK (0x1 << 9) +#define RT5670_CLSD_OC_SFT 9 +#define RT5670_CLSD_OC_PU (0x0 << 9) +#define RT5670_CLSD_OC_PD (0x1 << 9) +#define RT5670_AUTO_PD_MASK (0x1 << 8) +#define RT5670_AUTO_PD_SFT 8 +#define RT5670_AUTO_PD_DIS (0x0 << 8) +#define RT5670_AUTO_PD_EN (0x1 << 8) +#define RT5670_CLSD_OC_TH_MASK (0x3f) +#define RT5670_CLSD_OC_TH_SFT 0 + +/* Class D Output Control (0x8d) */ +#define RT5670_CLSD_RATIO_MASK (0xf << 12) +#define RT5670_CLSD_RATIO_SFT 12 +#define RT5670_CLSD_OM_MASK (0x1 << 11) +#define RT5670_CLSD_OM_SFT 11 +#define RT5670_CLSD_OM_MONO (0x0 << 11) +#define RT5670_CLSD_OM_STO (0x1 << 11) +#define RT5670_CLSD_SCH_MASK (0x1 << 10) +#define RT5670_CLSD_SCH_SFT 10 +#define RT5670_CLSD_SCH_L (0x0 << 10) +#define RT5670_CLSD_SCH_S (0x1 << 10) + +/* Depop Mode Control 1 (0x8e) */ +#define RT5670_SMT_TRIG_MASK (0x1 << 15) +#define RT5670_SMT_TRIG_SFT 15 +#define RT5670_SMT_TRIG_DIS (0x0 << 15) +#define RT5670_SMT_TRIG_EN (0x1 << 15) +#define RT5670_HP_L_SMT_MASK (0x1 << 9) +#define RT5670_HP_L_SMT_SFT 9 +#define RT5670_HP_L_SMT_DIS (0x0 << 9) +#define RT5670_HP_L_SMT_EN (0x1 << 9) +#define RT5670_HP_R_SMT_MASK (0x1 << 8) +#define RT5670_HP_R_SMT_SFT 8 +#define RT5670_HP_R_SMT_DIS (0x0 << 8) +#define RT5670_HP_R_SMT_EN (0x1 << 8) +#define RT5670_HP_CD_PD_MASK (0x1 << 7) +#define RT5670_HP_CD_PD_SFT 7 +#define RT5670_HP_CD_PD_DIS (0x0 << 7) +#define RT5670_HP_CD_PD_EN (0x1 << 7) +#define RT5670_RSTN_MASK (0x1 << 6) +#define RT5670_RSTN_SFT 6 +#define RT5670_RSTN_DIS (0x0 << 6) +#define RT5670_RSTN_EN (0x1 << 6) +#define RT5670_RSTP_MASK (0x1 << 5) +#define RT5670_RSTP_SFT 5 +#define RT5670_RSTP_DIS (0x0 << 5) +#define RT5670_RSTP_EN (0x1 << 5) +#define RT5670_HP_CO_MASK (0x1 << 4) +#define RT5670_HP_CO_SFT 4 +#define RT5670_HP_CO_DIS (0x0 << 4) +#define RT5670_HP_CO_EN (0x1 << 4) +#define RT5670_HP_CP_MASK (0x1 << 3) +#define RT5670_HP_CP_SFT 3 +#define RT5670_HP_CP_PD (0x0 << 3) +#define RT5670_HP_CP_PU (0x1 << 3) +#define RT5670_HP_SG_MASK (0x1 << 2) +#define RT5670_HP_SG_SFT 2 +#define RT5670_HP_SG_DIS (0x0 << 2) +#define RT5670_HP_SG_EN (0x1 << 2) +#define RT5670_HP_DP_MASK (0x1 << 1) +#define RT5670_HP_DP_SFT 1 +#define RT5670_HP_DP_PD (0x0 << 1) +#define RT5670_HP_DP_PU (0x1 << 1) +#define RT5670_HP_CB_MASK (0x1) +#define RT5670_HP_CB_SFT 0 +#define RT5670_HP_CB_PD (0x0) +#define RT5670_HP_CB_PU (0x1) + +/* Depop Mode Control 2 (0x8f) */ +#define RT5670_DEPOP_MASK (0x1 << 13) +#define RT5670_DEPOP_SFT 13 +#define RT5670_DEPOP_AUTO (0x0 << 13) +#define RT5670_DEPOP_MAN (0x1 << 13) +#define RT5670_RAMP_MASK (0x1 << 12) +#define RT5670_RAMP_SFT 12 +#define RT5670_RAMP_DIS (0x0 << 12) +#define RT5670_RAMP_EN (0x1 << 12) +#define RT5670_BPS_MASK (0x1 << 11) +#define RT5670_BPS_SFT 11 +#define RT5670_BPS_DIS (0x0 << 11) +#define RT5670_BPS_EN (0x1 << 11) +#define RT5670_FAST_UPDN_MASK (0x1 << 10) +#define RT5670_FAST_UPDN_SFT 10 +#define RT5670_FAST_UPDN_DIS (0x0 << 10) +#define RT5670_FAST_UPDN_EN (0x1 << 10) +#define RT5670_MRES_MASK (0x3 << 8) +#define RT5670_MRES_SFT 8 +#define RT5670_MRES_15MO (0x0 << 8) +#define RT5670_MRES_25MO (0x1 << 8) +#define RT5670_MRES_35MO (0x2 << 8) +#define RT5670_MRES_45MO (0x3 << 8) +#define RT5670_VLO_MASK (0x1 << 7) +#define RT5670_VLO_SFT 7 +#define RT5670_VLO_3V (0x0 << 7) +#define RT5670_VLO_32V (0x1 << 7) +#define RT5670_DIG_DP_MASK (0x1 << 6) +#define RT5670_DIG_DP_SFT 6 +#define RT5670_DIG_DP_DIS (0x0 << 6) +#define RT5670_DIG_DP_EN (0x1 << 6) +#define RT5670_DP_TH_MASK (0x3 << 4) +#define RT5670_DP_TH_SFT 4 + +/* Depop Mode Control 3 (0x90) */ +#define RT5670_CP_SYS_MASK (0x7 << 12) +#define RT5670_CP_SYS_SFT 12 +#define RT5670_CP_FQ1_MASK (0x7 << 8) +#define RT5670_CP_FQ1_SFT 8 +#define RT5670_CP_FQ2_MASK (0x7 << 4) +#define RT5670_CP_FQ2_SFT 4 +#define RT5670_CP_FQ3_MASK (0x7) +#define RT5670_CP_FQ3_SFT 0 +#define RT5670_CP_FQ_1_5_KHZ 0 +#define RT5670_CP_FQ_3_KHZ 1 +#define RT5670_CP_FQ_6_KHZ 2 +#define RT5670_CP_FQ_12_KHZ 3 +#define RT5670_CP_FQ_24_KHZ 4 +#define RT5670_CP_FQ_48_KHZ 5 +#define RT5670_CP_FQ_96_KHZ 6 +#define RT5670_CP_FQ_192_KHZ 7 + +/* HPOUT charge pump (0x91) */ +#define RT5670_OSW_L_MASK (0x1 << 11) +#define RT5670_OSW_L_SFT 11 +#define RT5670_OSW_L_DIS (0x0 << 11) +#define RT5670_OSW_L_EN (0x1 << 11) +#define RT5670_OSW_R_MASK (0x1 << 10) +#define RT5670_OSW_R_SFT 10 +#define RT5670_OSW_R_DIS (0x0 << 10) +#define RT5670_OSW_R_EN (0x1 << 10) +#define RT5670_PM_HP_MASK (0x3 << 8) +#define RT5670_PM_HP_SFT 8 +#define RT5670_PM_HP_LV (0x0 << 8) +#define RT5670_PM_HP_MV (0x1 << 8) +#define RT5670_PM_HP_HV (0x2 << 8) +#define RT5670_IB_HP_MASK (0x3 << 6) +#define RT5670_IB_HP_SFT 6 +#define RT5670_IB_HP_125IL (0x0 << 6) +#define RT5670_IB_HP_25IL (0x1 << 6) +#define RT5670_IB_HP_5IL (0x2 << 6) +#define RT5670_IB_HP_1IL (0x3 << 6) + +/* PV detection and SPK gain control (0x92) */ +#define RT5670_PVDD_DET_MASK (0x1 << 15) +#define RT5670_PVDD_DET_SFT 15 +#define RT5670_PVDD_DET_DIS (0x0 << 15) +#define RT5670_PVDD_DET_EN (0x1 << 15) +#define RT5670_SPK_AG_MASK (0x1 << 14) +#define RT5670_SPK_AG_SFT 14 +#define RT5670_SPK_AG_DIS (0x0 << 14) +#define RT5670_SPK_AG_EN (0x1 << 14) + +/* Micbias Control (0x93) */ +#define RT5670_MIC1_BS_MASK (0x1 << 15) +#define RT5670_MIC1_BS_SFT 15 +#define RT5670_MIC1_BS_9AV (0x0 << 15) +#define RT5670_MIC1_BS_75AV (0x1 << 15) +#define RT5670_MIC2_BS_MASK (0x1 << 14) +#define RT5670_MIC2_BS_SFT 14 +#define RT5670_MIC2_BS_9AV (0x0 << 14) +#define RT5670_MIC2_BS_75AV (0x1 << 14) +#define RT5670_MIC1_CLK_MASK (0x1 << 13) +#define RT5670_MIC1_CLK_SFT 13 +#define RT5670_MIC1_CLK_DIS (0x0 << 13) +#define RT5670_MIC1_CLK_EN (0x1 << 13) +#define RT5670_MIC2_CLK_MASK (0x1 << 12) +#define RT5670_MIC2_CLK_SFT 12 +#define RT5670_MIC2_CLK_DIS (0x0 << 12) +#define RT5670_MIC2_CLK_EN (0x1 << 12) +#define RT5670_MIC1_OVCD_MASK (0x1 << 11) +#define RT5670_MIC1_OVCD_SFT 11 +#define RT5670_MIC1_OVCD_DIS (0x0 << 11) +#define RT5670_MIC1_OVCD_EN (0x1 << 11) +#define RT5670_MIC1_OVTH_MASK (0x3 << 9) +#define RT5670_MIC1_OVTH_SFT 9 +#define RT5670_MIC1_OVTH_600UA (0x0 << 9) +#define RT5670_MIC1_OVTH_1500UA (0x1 << 9) +#define RT5670_MIC1_OVTH_2000UA (0x2 << 9) +#define RT5670_MIC2_OVCD_MASK (0x1 << 8) +#define RT5670_MIC2_OVCD_SFT 8 +#define RT5670_MIC2_OVCD_DIS (0x0 << 8) +#define RT5670_MIC2_OVCD_EN (0x1 << 8) +#define RT5670_MIC2_OVTH_MASK (0x3 << 6) +#define RT5670_MIC2_OVTH_SFT 6 +#define RT5670_MIC2_OVTH_600UA (0x0 << 6) +#define RT5670_MIC2_OVTH_1500UA (0x1 << 6) +#define RT5670_MIC2_OVTH_2000UA (0x2 << 6) +#define RT5670_PWR_MB_MASK (0x1 << 5) +#define RT5670_PWR_MB_SFT 5 +#define RT5670_PWR_MB_PD (0x0 << 5) +#define RT5670_PWR_MB_PU (0x1 << 5) +#define RT5670_PWR_CLK25M_MASK (0x1 << 4) +#define RT5670_PWR_CLK25M_SFT 4 +#define RT5670_PWR_CLK25M_PD (0x0 << 4) +#define RT5670_PWR_CLK25M_PU (0x1 << 4) + +/* VAD Control 4 (0x9d) */ +#define RT5670_VAD_SEL_MASK (0x3 << 8) +#define RT5670_VAD_SEL_SFT 8 + +/* EQ Control 1 (0xb0) */ +#define RT5670_EQ_SRC_MASK (0x1 << 15) +#define RT5670_EQ_SRC_SFT 15 +#define RT5670_EQ_SRC_DAC (0x0 << 15) +#define RT5670_EQ_SRC_ADC (0x1 << 15) +#define RT5670_EQ_UPD (0x1 << 14) +#define RT5670_EQ_UPD_BIT 14 +#define RT5670_EQ_CD_MASK (0x1 << 13) +#define RT5670_EQ_CD_SFT 13 +#define RT5670_EQ_CD_DIS (0x0 << 13) +#define RT5670_EQ_CD_EN (0x1 << 13) +#define RT5670_EQ_DITH_MASK (0x3 << 8) +#define RT5670_EQ_DITH_SFT 8 +#define RT5670_EQ_DITH_NOR (0x0 << 8) +#define RT5670_EQ_DITH_LSB (0x1 << 8) +#define RT5670_EQ_DITH_LSB_1 (0x2 << 8) +#define RT5670_EQ_DITH_LSB_2 (0x3 << 8) + +/* EQ Control 2 (0xb1) */ +#define RT5670_EQ_HPF1_M_MASK (0x1 << 8) +#define RT5670_EQ_HPF1_M_SFT 8 +#define RT5670_EQ_HPF1_M_HI (0x0 << 8) +#define RT5670_EQ_HPF1_M_1ST (0x1 << 8) +#define RT5670_EQ_LPF1_M_MASK (0x1 << 7) +#define RT5670_EQ_LPF1_M_SFT 7 +#define RT5670_EQ_LPF1_M_LO (0x0 << 7) +#define RT5670_EQ_LPF1_M_1ST (0x1 << 7) +#define RT5670_EQ_HPF2_MASK (0x1 << 6) +#define RT5670_EQ_HPF2_SFT 6 +#define RT5670_EQ_HPF2_DIS (0x0 << 6) +#define RT5670_EQ_HPF2_EN (0x1 << 6) +#define RT5670_EQ_HPF1_MASK (0x1 << 5) +#define RT5670_EQ_HPF1_SFT 5 +#define RT5670_EQ_HPF1_DIS (0x0 << 5) +#define RT5670_EQ_HPF1_EN (0x1 << 5) +#define RT5670_EQ_BPF4_MASK (0x1 << 4) +#define RT5670_EQ_BPF4_SFT 4 +#define RT5670_EQ_BPF4_DIS (0x0 << 4) +#define RT5670_EQ_BPF4_EN (0x1 << 4) +#define RT5670_EQ_BPF3_MASK (0x1 << 3) +#define RT5670_EQ_BPF3_SFT 3 +#define RT5670_EQ_BPF3_DIS (0x0 << 3) +#define RT5670_EQ_BPF3_EN (0x1 << 3) +#define RT5670_EQ_BPF2_MASK (0x1 << 2) +#define RT5670_EQ_BPF2_SFT 2 +#define RT5670_EQ_BPF2_DIS (0x0 << 2) +#define RT5670_EQ_BPF2_EN (0x1 << 2) +#define RT5670_EQ_BPF1_MASK (0x1 << 1) +#define RT5670_EQ_BPF1_SFT 1 +#define RT5670_EQ_BPF1_DIS (0x0 << 1) +#define RT5670_EQ_BPF1_EN (0x1 << 1) +#define RT5670_EQ_LPF_MASK (0x1) +#define RT5670_EQ_LPF_SFT 0 +#define RT5670_EQ_LPF_DIS (0x0) +#define RT5670_EQ_LPF_EN (0x1) +#define RT5670_EQ_CTRL_MASK (0x7f) + +/* Memory Test (0xb2) */ +#define RT5670_MT_MASK (0x1 << 15) +#define RT5670_MT_SFT 15 +#define RT5670_MT_DIS (0x0 << 15) +#define RT5670_MT_EN (0x1 << 15) + +/* DRC/AGC Control 1 (0xb4) */ +#define RT5670_DRC_AGC_P_MASK (0x1 << 15) +#define RT5670_DRC_AGC_P_SFT 15 +#define RT5670_DRC_AGC_P_DAC (0x0 << 15) +#define RT5670_DRC_AGC_P_ADC (0x1 << 15) +#define RT5670_DRC_AGC_MASK (0x1 << 14) +#define RT5670_DRC_AGC_SFT 14 +#define RT5670_DRC_AGC_DIS (0x0 << 14) +#define RT5670_DRC_AGC_EN (0x1 << 14) +#define RT5670_DRC_AGC_UPD (0x1 << 13) +#define RT5670_DRC_AGC_UPD_BIT 13 +#define RT5670_DRC_AGC_AR_MASK (0x1f << 8) +#define RT5670_DRC_AGC_AR_SFT 8 +#define RT5670_DRC_AGC_R_MASK (0x7 << 5) +#define RT5670_DRC_AGC_R_SFT 5 +#define RT5670_DRC_AGC_R_48K (0x1 << 5) +#define RT5670_DRC_AGC_R_96K (0x2 << 5) +#define RT5670_DRC_AGC_R_192K (0x3 << 5) +#define RT5670_DRC_AGC_R_441K (0x5 << 5) +#define RT5670_DRC_AGC_R_882K (0x6 << 5) +#define RT5670_DRC_AGC_R_1764K (0x7 << 5) +#define RT5670_DRC_AGC_RC_MASK (0x1f) +#define RT5670_DRC_AGC_RC_SFT 0 + +/* DRC/AGC Control 2 (0xb5) */ +#define RT5670_DRC_AGC_POB_MASK (0x3f << 8) +#define RT5670_DRC_AGC_POB_SFT 8 +#define RT5670_DRC_AGC_CP_MASK (0x1 << 7) +#define RT5670_DRC_AGC_CP_SFT 7 +#define RT5670_DRC_AGC_CP_DIS (0x0 << 7) +#define RT5670_DRC_AGC_CP_EN (0x1 << 7) +#define RT5670_DRC_AGC_CPR_MASK (0x3 << 5) +#define RT5670_DRC_AGC_CPR_SFT 5 +#define RT5670_DRC_AGC_CPR_1_1 (0x0 << 5) +#define RT5670_DRC_AGC_CPR_1_2 (0x1 << 5) +#define RT5670_DRC_AGC_CPR_1_3 (0x2 << 5) +#define RT5670_DRC_AGC_CPR_1_4 (0x3 << 5) +#define RT5670_DRC_AGC_PRB_MASK (0x1f) +#define RT5670_DRC_AGC_PRB_SFT 0 + +/* DRC/AGC Control 3 (0xb6) */ +#define RT5670_DRC_AGC_NGB_MASK (0xf << 12) +#define RT5670_DRC_AGC_NGB_SFT 12 +#define RT5670_DRC_AGC_TAR_MASK (0x1f << 7) +#define RT5670_DRC_AGC_TAR_SFT 7 +#define RT5670_DRC_AGC_NG_MASK (0x1 << 6) +#define RT5670_DRC_AGC_NG_SFT 6 +#define RT5670_DRC_AGC_NG_DIS (0x0 << 6) +#define RT5670_DRC_AGC_NG_EN (0x1 << 6) +#define RT5670_DRC_AGC_NGH_MASK (0x1 << 5) +#define RT5670_DRC_AGC_NGH_SFT 5 +#define RT5670_DRC_AGC_NGH_DIS (0x0 << 5) +#define RT5670_DRC_AGC_NGH_EN (0x1 << 5) +#define RT5670_DRC_AGC_NGT_MASK (0x1f) +#define RT5670_DRC_AGC_NGT_SFT 0 + +/* Jack Detect Control (0xbb) */ +#define RT5670_JD_MASK (0x7 << 13) +#define RT5670_JD_SFT 13 +#define RT5670_JD_DIS (0x0 << 13) +#define RT5670_JD_GPIO1 (0x1 << 13) +#define RT5670_JD_JD1_IN4P (0x2 << 13) +#define RT5670_JD_JD2_IN4N (0x3 << 13) +#define RT5670_JD_GPIO2 (0x4 << 13) +#define RT5670_JD_GPIO3 (0x5 << 13) +#define RT5670_JD_GPIO4 (0x6 << 13) +#define RT5670_JD_HP_MASK (0x1 << 11) +#define RT5670_JD_HP_SFT 11 +#define RT5670_JD_HP_DIS (0x0 << 11) +#define RT5670_JD_HP_EN (0x1 << 11) +#define RT5670_JD_HP_TRG_MASK (0x1 << 10) +#define RT5670_JD_HP_TRG_SFT 10 +#define RT5670_JD_HP_TRG_LO (0x0 << 10) +#define RT5670_JD_HP_TRG_HI (0x1 << 10) +#define RT5670_JD_SPL_MASK (0x1 << 9) +#define RT5670_JD_SPL_SFT 9 +#define RT5670_JD_SPL_DIS (0x0 << 9) +#define RT5670_JD_SPL_EN (0x1 << 9) +#define RT5670_JD_SPL_TRG_MASK (0x1 << 8) +#define RT5670_JD_SPL_TRG_SFT 8 +#define RT5670_JD_SPL_TRG_LO (0x0 << 8) +#define RT5670_JD_SPL_TRG_HI (0x1 << 8) +#define RT5670_JD_SPR_MASK (0x1 << 7) +#define RT5670_JD_SPR_SFT 7 +#define RT5670_JD_SPR_DIS (0x0 << 7) +#define RT5670_JD_SPR_EN (0x1 << 7) +#define RT5670_JD_SPR_TRG_MASK (0x1 << 6) +#define RT5670_JD_SPR_TRG_SFT 6 +#define RT5670_JD_SPR_TRG_LO (0x0 << 6) +#define RT5670_JD_SPR_TRG_HI (0x1 << 6) +#define RT5670_JD_MO_MASK (0x1 << 5) +#define RT5670_JD_MO_SFT 5 +#define RT5670_JD_MO_DIS (0x0 << 5) +#define RT5670_JD_MO_EN (0x1 << 5) +#define RT5670_JD_MO_TRG_MASK (0x1 << 4) +#define RT5670_JD_MO_TRG_SFT 4 +#define RT5670_JD_MO_TRG_LO (0x0 << 4) +#define RT5670_JD_MO_TRG_HI (0x1 << 4) +#define RT5670_JD_LO_MASK (0x1 << 3) +#define RT5670_JD_LO_SFT 3 +#define RT5670_JD_LO_DIS (0x0 << 3) +#define RT5670_JD_LO_EN (0x1 << 3) +#define RT5670_JD_LO_TRG_MASK (0x1 << 2) +#define RT5670_JD_LO_TRG_SFT 2 +#define RT5670_JD_LO_TRG_LO (0x0 << 2) +#define RT5670_JD_LO_TRG_HI (0x1 << 2) +#define RT5670_JD1_IN4P_MASK (0x1 << 1) +#define RT5670_JD1_IN4P_SFT 1 +#define RT5670_JD1_IN4P_DIS (0x0 << 1) +#define RT5670_JD1_IN4P_EN (0x1 << 1) +#define RT5670_JD2_IN4N_MASK (0x1) +#define RT5670_JD2_IN4N_SFT 0 +#define RT5670_JD2_IN4N_DIS (0x0) +#define RT5670_JD2_IN4N_EN (0x1) + +/* IRQ Control 1 (0xbd) */ +#define RT5670_IRQ_JD_MASK (0x1 << 15) +#define RT5670_IRQ_JD_SFT 15 +#define RT5670_IRQ_JD_BP (0x0 << 15) +#define RT5670_IRQ_JD_NOR (0x1 << 15) +#define RT5670_IRQ_OT_MASK (0x1 << 14) +#define RT5670_IRQ_OT_SFT 14 +#define RT5670_IRQ_OT_BP (0x0 << 14) +#define RT5670_IRQ_OT_NOR (0x1 << 14) +#define RT5670_JD_STKY_MASK (0x1 << 13) +#define RT5670_JD_STKY_SFT 13 +#define RT5670_JD_STKY_DIS (0x0 << 13) +#define RT5670_JD_STKY_EN (0x1 << 13) +#define RT5670_OT_STKY_MASK (0x1 << 12) +#define RT5670_OT_STKY_SFT 12 +#define RT5670_OT_STKY_DIS (0x0 << 12) +#define RT5670_OT_STKY_EN (0x1 << 12) +#define RT5670_JD_P_MASK (0x1 << 11) +#define RT5670_JD_P_SFT 11 +#define RT5670_JD_P_NOR (0x0 << 11) +#define RT5670_JD_P_INV (0x1 << 11) +#define RT5670_OT_P_MASK (0x1 << 10) +#define RT5670_OT_P_SFT 10 +#define RT5670_OT_P_NOR (0x0 << 10) +#define RT5670_OT_P_INV (0x1 << 10) + +/* IRQ Control 2 (0xbe) */ +#define RT5670_IRQ_MB1_OC_MASK (0x1 << 15) +#define RT5670_IRQ_MB1_OC_SFT 15 +#define RT5670_IRQ_MB1_OC_BP (0x0 << 15) +#define RT5670_IRQ_MB1_OC_NOR (0x1 << 15) +#define RT5670_IRQ_MB2_OC_MASK (0x1 << 14) +#define RT5670_IRQ_MB2_OC_SFT 14 +#define RT5670_IRQ_MB2_OC_BP (0x0 << 14) +#define RT5670_IRQ_MB2_OC_NOR (0x1 << 14) +#define RT5670_MB1_OC_STKY_MASK (0x1 << 11) +#define RT5670_MB1_OC_STKY_SFT 11 +#define RT5670_MB1_OC_STKY_DIS (0x0 << 11) +#define RT5670_MB1_OC_STKY_EN (0x1 << 11) +#define RT5670_MB2_OC_STKY_MASK (0x1 << 10) +#define RT5670_MB2_OC_STKY_SFT 10 +#define RT5670_MB2_OC_STKY_DIS (0x0 << 10) +#define RT5670_MB2_OC_STKY_EN (0x1 << 10) +#define RT5670_MB1_OC_P_MASK (0x1 << 7) +#define RT5670_MB1_OC_P_SFT 7 +#define RT5670_MB1_OC_P_NOR (0x0 << 7) +#define RT5670_MB1_OC_P_INV (0x1 << 7) +#define RT5670_MB2_OC_P_MASK (0x1 << 6) +#define RT5670_MB2_OC_P_SFT 6 +#define RT5670_MB2_OC_P_NOR (0x0 << 6) +#define RT5670_MB2_OC_P_INV (0x1 << 6) +#define RT5670_MB1_OC_CLR (0x1 << 3) +#define RT5670_MB1_OC_CLR_SFT 3 +#define RT5670_MB2_OC_CLR (0x1 << 2) +#define RT5670_MB2_OC_CLR_SFT 2 + +/* GPIO Control 1 (0xc0) */ +#define RT5670_GP1_PIN_MASK (0x1 << 15) +#define RT5670_GP1_PIN_SFT 15 +#define RT5670_GP1_PIN_GPIO1 (0x0 << 15) +#define RT5670_GP1_PIN_IRQ (0x1 << 15) +#define RT5670_GP2_PIN_MASK (0x1 << 14) +#define RT5670_GP2_PIN_SFT 14 +#define RT5670_GP2_PIN_GPIO2 (0x0 << 14) +#define RT5670_GP2_PIN_DMIC1_SCL (0x1 << 14) +#define RT5670_GP3_PIN_MASK (0x3 << 12) +#define RT5670_GP3_PIN_SFT 12 +#define RT5670_GP3_PIN_GPIO3 (0x0 << 12) +#define RT5670_GP3_PIN_DMIC1_SDA (0x1 << 12) +#define RT5670_GP3_PIN_IRQ (0x2 << 12) +#define RT5670_GP4_PIN_MASK (0x1 << 11) +#define RT5670_GP4_PIN_SFT 11 +#define RT5670_GP4_PIN_GPIO4 (0x0 << 11) +#define RT5670_GP4_PIN_DMIC2_SDA (0x1 << 11) +#define RT5670_DP_SIG_MASK (0x1 << 10) +#define RT5670_DP_SIG_SFT 10 +#define RT5670_DP_SIG_TEST (0x0 << 10) +#define RT5670_DP_SIG_AP (0x1 << 10) +#define RT5670_GPIO_M_MASK (0x1 << 9) +#define RT5670_GPIO_M_SFT 9 +#define RT5670_GPIO_M_FLT (0x0 << 9) +#define RT5670_GPIO_M_PH (0x1 << 9) +#define RT5670_I2S2_PIN_MASK (0x1 << 8) +#define RT5670_I2S2_PIN_SFT 8 +#define RT5670_I2S2_PIN_I2S (0x0 << 8) +#define RT5670_I2S2_PIN_GPIO (0x1 << 8) +#define RT5670_GP5_PIN_MASK (0x1 << 7) +#define RT5670_GP5_PIN_SFT 7 +#define RT5670_GP5_PIN_GPIO5 (0x0 << 7) +#define RT5670_GP5_PIN_DMIC3_SCL (0x1 << 7) +#define RT5670_GP6_PIN_MASK (0x1 << 6) +#define RT5670_GP6_PIN_SFT 6 +#define RT5670_GP6_PIN_GPIO6 (0x0 << 6) +#define RT5670_GP6_PIN_DMIC1_SDA (0x1 << 6) +#define RT5670_GP7_PIN_MASK (0x3 << 4) +#define RT5670_GP7_PIN_SFT 4 +#define RT5670_GP7_PIN_GPIO7 (0x0 << 4) +#define RT5670_GP7_PIN_DMIC1_SDA (0x1 << 4) +#define RT5670_GP7_PIN_PDM_SCL2 (0x2 << 4) +#define RT5670_GP8_PIN_MASK (0x1 << 3) +#define RT5670_GP8_PIN_SFT 3 +#define RT5670_GP8_PIN_GPIO8 (0x0 << 3) +#define RT5670_GP8_PIN_DMIC2_SDA (0x1 << 3) +#define RT5670_GP9_PIN_MASK (0x1 << 2) +#define RT5670_GP9_PIN_SFT 2 +#define RT5670_GP9_PIN_GPIO9 (0x0 << 2) +#define RT5670_GP9_PIN_DMIC3_SDA (0x1 << 2) +#define RT5670_GP10_PIN_MASK (0x3) +#define RT5670_GP10_PIN_SFT 0 +#define RT5670_GP10_PIN_GPIO9 (0x0) +#define RT5670_GP10_PIN_DMIC3_SDA (0x1) +#define RT5670_GP10_PIN_PDM_ADT2 (0x2) + +/* GPIO Control 2 (0xc1) */ +#define RT5670_GP4_PF_MASK (0x1 << 11) +#define RT5670_GP4_PF_SFT 11 +#define RT5670_GP4_PF_IN (0x0 << 11) +#define RT5670_GP4_PF_OUT (0x1 << 11) +#define RT5670_GP4_OUT_MASK (0x1 << 10) +#define RT5670_GP4_OUT_SFT 10 +#define RT5670_GP4_OUT_LO (0x0 << 10) +#define RT5670_GP4_OUT_HI (0x1 << 10) +#define RT5670_GP4_P_MASK (0x1 << 9) +#define RT5670_GP4_P_SFT 9 +#define RT5670_GP4_P_NOR (0x0 << 9) +#define RT5670_GP4_P_INV (0x1 << 9) +#define RT5670_GP3_PF_MASK (0x1 << 8) +#define RT5670_GP3_PF_SFT 8 +#define RT5670_GP3_PF_IN (0x0 << 8) +#define RT5670_GP3_PF_OUT (0x1 << 8) +#define RT5670_GP3_OUT_MASK (0x1 << 7) +#define RT5670_GP3_OUT_SFT 7 +#define RT5670_GP3_OUT_LO (0x0 << 7) +#define RT5670_GP3_OUT_HI (0x1 << 7) +#define RT5670_GP3_P_MASK (0x1 << 6) +#define RT5670_GP3_P_SFT 6 +#define RT5670_GP3_P_NOR (0x0 << 6) +#define RT5670_GP3_P_INV (0x1 << 6) +#define RT5670_GP2_PF_MASK (0x1 << 5) +#define RT5670_GP2_PF_SFT 5 +#define RT5670_GP2_PF_IN (0x0 << 5) +#define RT5670_GP2_PF_OUT (0x1 << 5) +#define RT5670_GP2_OUT_MASK (0x1 << 4) +#define RT5670_GP2_OUT_SFT 4 +#define RT5670_GP2_OUT_LO (0x0 << 4) +#define RT5670_GP2_OUT_HI (0x1 << 4) +#define RT5670_GP2_P_MASK (0x1 << 3) +#define RT5670_GP2_P_SFT 3 +#define RT5670_GP2_P_NOR (0x0 << 3) +#define RT5670_GP2_P_INV (0x1 << 3) +#define RT5670_GP1_PF_MASK (0x1 << 2) +#define RT5670_GP1_PF_SFT 2 +#define RT5670_GP1_PF_IN (0x0 << 2) +#define RT5670_GP1_PF_OUT (0x1 << 2) +#define RT5670_GP1_OUT_MASK (0x1 << 1) +#define RT5670_GP1_OUT_SFT 1 +#define RT5670_GP1_OUT_LO (0x0 << 1) +#define RT5670_GP1_OUT_HI (0x1 << 1) +#define RT5670_GP1_P_MASK (0x1) +#define RT5670_GP1_P_SFT 0 +#define RT5670_GP1_P_NOR (0x0) +#define RT5670_GP1_P_INV (0x1) + +/* Scramble Function (0xcd) */ +#define RT5670_SCB_KEY_MASK (0xff) +#define RT5670_SCB_KEY_SFT 0 + +/* Scramble Control (0xce) */ +#define RT5670_SCB_SWAP_MASK (0x1 << 15) +#define RT5670_SCB_SWAP_SFT 15 +#define RT5670_SCB_SWAP_DIS (0x0 << 15) +#define RT5670_SCB_SWAP_EN (0x1 << 15) +#define RT5670_SCB_MASK (0x1 << 14) +#define RT5670_SCB_SFT 14 +#define RT5670_SCB_DIS (0x0 << 14) +#define RT5670_SCB_EN (0x1 << 14) + +/* Baseback Control (0xcf) */ +#define RT5670_BB_MASK (0x1 << 15) +#define RT5670_BB_SFT 15 +#define RT5670_BB_DIS (0x0 << 15) +#define RT5670_BB_EN (0x1 << 15) +#define RT5670_BB_CT_MASK (0x7 << 12) +#define RT5670_BB_CT_SFT 12 +#define RT5670_BB_CT_A (0x0 << 12) +#define RT5670_BB_CT_B (0x1 << 12) +#define RT5670_BB_CT_C (0x2 << 12) +#define RT5670_BB_CT_D (0x3 << 12) +#define RT5670_M_BB_L_MASK (0x1 << 9) +#define RT5670_M_BB_L_SFT 9 +#define RT5670_M_BB_R_MASK (0x1 << 8) +#define RT5670_M_BB_R_SFT 8 +#define RT5670_M_BB_HPF_L_MASK (0x1 << 7) +#define RT5670_M_BB_HPF_L_SFT 7 +#define RT5670_M_BB_HPF_R_MASK (0x1 << 6) +#define RT5670_M_BB_HPF_R_SFT 6 +#define RT5670_G_BB_BST_MASK (0x3f) +#define RT5670_G_BB_BST_SFT 0 + +/* MP3 Plus Control 1 (0xd0) */ +#define RT5670_M_MP3_L_MASK (0x1 << 15) +#define RT5670_M_MP3_L_SFT 15 +#define RT5670_M_MP3_R_MASK (0x1 << 14) +#define RT5670_M_MP3_R_SFT 14 +#define RT5670_M_MP3_MASK (0x1 << 13) +#define RT5670_M_MP3_SFT 13 +#define RT5670_M_MP3_DIS (0x0 << 13) +#define RT5670_M_MP3_EN (0x1 << 13) +#define RT5670_EG_MP3_MASK (0x1f << 8) +#define RT5670_EG_MP3_SFT 8 +#define RT5670_MP3_HLP_MASK (0x1 << 7) +#define RT5670_MP3_HLP_SFT 7 +#define RT5670_MP3_HLP_DIS (0x0 << 7) +#define RT5670_MP3_HLP_EN (0x1 << 7) +#define RT5670_M_MP3_ORG_L_MASK (0x1 << 6) +#define RT5670_M_MP3_ORG_L_SFT 6 +#define RT5670_M_MP3_ORG_R_MASK (0x1 << 5) +#define RT5670_M_MP3_ORG_R_SFT 5 + +/* MP3 Plus Control 2 (0xd1) */ +#define RT5670_MP3_WT_MASK (0x1 << 13) +#define RT5670_MP3_WT_SFT 13 +#define RT5670_MP3_WT_1_4 (0x0 << 13) +#define RT5670_MP3_WT_1_2 (0x1 << 13) +#define RT5670_OG_MP3_MASK (0x1f << 8) +#define RT5670_OG_MP3_SFT 8 +#define RT5670_HG_MP3_MASK (0x3f) +#define RT5670_HG_MP3_SFT 0 + +/* 3D HP Control 1 (0xd2) */ +#define RT5670_3D_CF_MASK (0x1 << 15) +#define RT5670_3D_CF_SFT 15 +#define RT5670_3D_CF_DIS (0x0 << 15) +#define RT5670_3D_CF_EN (0x1 << 15) +#define RT5670_3D_HP_MASK (0x1 << 14) +#define RT5670_3D_HP_SFT 14 +#define RT5670_3D_HP_DIS (0x0 << 14) +#define RT5670_3D_HP_EN (0x1 << 14) +#define RT5670_3D_BT_MASK (0x1 << 13) +#define RT5670_3D_BT_SFT 13 +#define RT5670_3D_BT_DIS (0x0 << 13) +#define RT5670_3D_BT_EN (0x1 << 13) +#define RT5670_3D_1F_MIX_MASK (0x3 << 11) +#define RT5670_3D_1F_MIX_SFT 11 +#define RT5670_3D_HP_M_MASK (0x1 << 10) +#define RT5670_3D_HP_M_SFT 10 +#define RT5670_3D_HP_M_SUR (0x0 << 10) +#define RT5670_3D_HP_M_FRO (0x1 << 10) +#define RT5670_M_3D_HRTF_MASK (0x1 << 9) +#define RT5670_M_3D_HRTF_SFT 9 +#define RT5670_M_3D_D2H_MASK (0x1 << 8) +#define RT5670_M_3D_D2H_SFT 8 +#define RT5670_M_3D_D2R_MASK (0x1 << 7) +#define RT5670_M_3D_D2R_SFT 7 +#define RT5670_M_3D_REVB_MASK (0x1 << 6) +#define RT5670_M_3D_REVB_SFT 6 + +/* Adjustable high pass filter control 1 (0xd3) */ +#define RT5670_2ND_HPF_MASK (0x1 << 15) +#define RT5670_2ND_HPF_SFT 15 +#define RT5670_2ND_HPF_DIS (0x0 << 15) +#define RT5670_2ND_HPF_EN (0x1 << 15) +#define RT5670_HPF_CF_L_MASK (0x7 << 12) +#define RT5670_HPF_CF_L_SFT 12 +#define RT5670_1ST_HPF_MASK (0x1 << 11) +#define RT5670_1ST_HPF_SFT 11 +#define RT5670_1ST_HPF_DIS (0x0 << 11) +#define RT5670_1ST_HPF_EN (0x1 << 11) +#define RT5670_HPF_CF_R_MASK (0x7 << 8) +#define RT5670_HPF_CF_R_SFT 8 +#define RT5670_ZD_T_MASK (0x3 << 6) +#define RT5670_ZD_T_SFT 6 +#define RT5670_ZD_F_MASK (0x3 << 4) +#define RT5670_ZD_F_SFT 4 +#define RT5670_ZD_F_IM (0x0 << 4) +#define RT5670_ZD_F_ZC_IM (0x1 << 4) +#define RT5670_ZD_F_ZC_IOD (0x2 << 4) +#define RT5670_ZD_F_UN (0x3 << 4) + +/* HP calibration control and Amp detection (0xd6) */ +#define RT5670_SI_DAC_MASK (0x1 << 11) +#define RT5670_SI_DAC_SFT 11 +#define RT5670_SI_DAC_AUTO (0x0 << 11) +#define RT5670_SI_DAC_TEST (0x1 << 11) +#define RT5670_DC_CAL_M_MASK (0x1 << 10) +#define RT5670_DC_CAL_M_SFT 10 +#define RT5670_DC_CAL_M_CAL (0x0 << 10) +#define RT5670_DC_CAL_M_NOR (0x1 << 10) +#define RT5670_DC_CAL_MASK (0x1 << 9) +#define RT5670_DC_CAL_SFT 9 +#define RT5670_DC_CAL_DIS (0x0 << 9) +#define RT5670_DC_CAL_EN (0x1 << 9) +#define RT5670_HPD_RCV_MASK (0x7 << 6) +#define RT5670_HPD_RCV_SFT 6 +#define RT5670_HPD_PS_MASK (0x1 << 5) +#define RT5670_HPD_PS_SFT 5 +#define RT5670_HPD_PS_DIS (0x0 << 5) +#define RT5670_HPD_PS_EN (0x1 << 5) +#define RT5670_CAL_M_MASK (0x1 << 4) +#define RT5670_CAL_M_SFT 4 +#define RT5670_CAL_M_DEP (0x0 << 4) +#define RT5670_CAL_M_CAL (0x1 << 4) +#define RT5670_CAL_MASK (0x1 << 3) +#define RT5670_CAL_SFT 3 +#define RT5670_CAL_DIS (0x0 << 3) +#define RT5670_CAL_EN (0x1 << 3) +#define RT5670_CAL_TEST_MASK (0x1 << 2) +#define RT5670_CAL_TEST_SFT 2 +#define RT5670_CAL_TEST_DIS (0x0 << 2) +#define RT5670_CAL_TEST_EN (0x1 << 2) +#define RT5670_CAL_P_MASK (0x3) +#define RT5670_CAL_P_SFT 0 +#define RT5670_CAL_P_NONE (0x0) +#define RT5670_CAL_P_CAL (0x1) +#define RT5670_CAL_P_DAC_CAL (0x2) + +/* Soft volume and zero cross control 1 (0xd9) */ +#define RT5670_SV_MASK (0x1 << 15) +#define RT5670_SV_SFT 15 +#define RT5670_SV_DIS (0x0 << 15) +#define RT5670_SV_EN (0x1 << 15) +#define RT5670_SPO_SV_MASK (0x1 << 14) +#define RT5670_SPO_SV_SFT 14 +#define RT5670_SPO_SV_DIS (0x0 << 14) +#define RT5670_SPO_SV_EN (0x1 << 14) +#define RT5670_OUT_SV_MASK (0x1 << 13) +#define RT5670_OUT_SV_SFT 13 +#define RT5670_OUT_SV_DIS (0x0 << 13) +#define RT5670_OUT_SV_EN (0x1 << 13) +#define RT5670_HP_SV_MASK (0x1 << 12) +#define RT5670_HP_SV_SFT 12 +#define RT5670_HP_SV_DIS (0x0 << 12) +#define RT5670_HP_SV_EN (0x1 << 12) +#define RT5670_ZCD_DIG_MASK (0x1 << 11) +#define RT5670_ZCD_DIG_SFT 11 +#define RT5670_ZCD_DIG_DIS (0x0 << 11) +#define RT5670_ZCD_DIG_EN (0x1 << 11) +#define RT5670_ZCD_MASK (0x1 << 10) +#define RT5670_ZCD_SFT 10 +#define RT5670_ZCD_PD (0x0 << 10) +#define RT5670_ZCD_PU (0x1 << 10) +#define RT5670_M_ZCD_MASK (0x3f << 4) +#define RT5670_M_ZCD_SFT 4 +#define RT5670_M_ZCD_RM_L (0x1 << 9) +#define RT5670_M_ZCD_RM_R (0x1 << 8) +#define RT5670_M_ZCD_SM_L (0x1 << 7) +#define RT5670_M_ZCD_SM_R (0x1 << 6) +#define RT5670_M_ZCD_OM_L (0x1 << 5) +#define RT5670_M_ZCD_OM_R (0x1 << 4) +#define RT5670_SV_DLY_MASK (0xf) +#define RT5670_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0xda) */ +#define RT5670_ZCD_HP_MASK (0x1 << 15) +#define RT5670_ZCD_HP_SFT 15 +#define RT5670_ZCD_HP_DIS (0x0 << 15) +#define RT5670_ZCD_HP_EN (0x1 << 15) + + +/* Codec Private Register definition */ +/* 3D Speaker Control (0x63) */ +#define RT5670_3D_SPK_MASK (0x1 << 15) +#define RT5670_3D_SPK_SFT 15 +#define RT5670_3D_SPK_DIS (0x0 << 15) +#define RT5670_3D_SPK_EN (0x1 << 15) +#define RT5670_3D_SPK_M_MASK (0x3 << 13) +#define RT5670_3D_SPK_M_SFT 13 +#define RT5670_3D_SPK_CG_MASK (0x1f << 8) +#define RT5670_3D_SPK_CG_SFT 8 +#define RT5670_3D_SPK_SG_MASK (0x1f) +#define RT5670_3D_SPK_SG_SFT 0 + +/* Wind Noise Detection Control 1 (0x6c) */ +#define RT5670_WND_MASK (0x1 << 15) +#define RT5670_WND_SFT 15 +#define RT5670_WND_DIS (0x0 << 15) +#define RT5670_WND_EN (0x1 << 15) + +/* Wind Noise Detection Control 2 (0x6d) */ +#define RT5670_WND_FC_NW_MASK (0x3f << 10) +#define RT5670_WND_FC_NW_SFT 10 +#define RT5670_WND_FC_WK_MASK (0x3f << 4) +#define RT5670_WND_FC_WK_SFT 4 + +/* Wind Noise Detection Control 3 (0x6e) */ +#define RT5670_HPF_FC_MASK (0x3f << 6) +#define RT5670_HPF_FC_SFT 6 +#define RT5670_WND_FC_ST_MASK (0x3f) +#define RT5670_WND_FC_ST_SFT 0 + +/* Wind Noise Detection Control 4 (0x6f) */ +#define RT5670_WND_TH_LO_MASK (0x3ff) +#define RT5670_WND_TH_LO_SFT 0 + +/* Wind Noise Detection Control 5 (0x70) */ +#define RT5670_WND_TH_HI_MASK (0x3ff) +#define RT5670_WND_TH_HI_SFT 0 + +/* Wind Noise Detection Control 8 (0x73) */ +#define RT5670_WND_WIND_MASK (0x1 << 13) /* Read-Only */ +#define RT5670_WND_WIND_SFT 13 +#define RT5670_WND_STRONG_MASK (0x1 << 12) /* Read-Only */ +#define RT5670_WND_STRONG_SFT 12 +enum { + RT5670_NO_WIND, + RT5670_BREEZE, + RT5670_STORM, +}; + +/* Dipole Speaker Interface (0x75) */ +#define RT5670_DP_ATT_MASK (0x3 << 14) +#define RT5670_DP_ATT_SFT 14 +#define RT5670_DP_SPK_MASK (0x1 << 10) +#define RT5670_DP_SPK_SFT 10 +#define RT5670_DP_SPK_DIS (0x0 << 10) +#define RT5670_DP_SPK_EN (0x1 << 10) + +/* EQ Pre Volume Control (0xb3) */ +#define RT5670_EQ_PRE_VOL_MASK (0xffff) +#define RT5670_EQ_PRE_VOL_SFT 0 + +/* EQ Post Volume Control (0xb4) */ +#define RT5670_EQ_PST_VOL_MASK (0xffff) +#define RT5670_EQ_PST_VOL_SFT 0 + +/* Jack Detect Control 3 (0xf8) */ +#define RT5670_CMP_MIC_IN_DET_MASK (0x7 << 12) +#define RT5670_JD_CBJ_EN (0x1 << 7) +#define RT5670_JD_CBJ_POL (0x1 << 6) +#define RT5670_JD_TRI_CBJ_SEL_MASK (0x7 << 3) +#define RT5670_JD_TRI_CBJ_SEL_SFT (3) +#define RT5670_JD_TRI_HPO_SEL_MASK (0x7) +#define RT5670_JD_TRI_HPO_SEL_SFT (0) +#define RT5670_JD_F_GPIO_JD1 (0x0) +#define RT5670_JD_F_JD1_1 (0x1) +#define RT5670_JD_F_JD1_2 (0x2) +#define RT5670_JD_F_JD2 (0x3) +#define RT5670_JD_F_JD3 (0x4) +#define RT5670_JD_F_GPIO_JD2 (0x5) +#define RT5670_JD_F_MX0B_12 (0x6) + +/* Digital Misc Control (0xfa) */ +#define RT5670_RST_DSP (0x1 << 13) +#define RT5670_IF1_ADC1_IN1_SEL (0x1 << 12) +#define RT5670_IF1_ADC1_IN1_SFT 12 +#define RT5670_IF1_ADC1_IN2_SEL (0x1 << 11) +#define RT5670_IF1_ADC1_IN2_SFT 11 +#define RT5670_IF1_ADC2_IN1_SEL (0x1 << 10) +#define RT5670_IF1_ADC2_IN1_SFT 10 + +/* General Control2 (0xfb) */ +#define RT5670_RXDC_SRC_MASK (0x1 << 7) +#define RT5670_RXDC_SRC_STO (0x0 << 7) +#define RT5670_RXDC_SRC_MONO (0x1 << 7) +#define RT5670_RXDC_SRC_SFT (7) +#define RT5670_RXDP2_SEL_MASK (0x1 << 3) +#define RT5670_RXDP2_SEL_IF2 (0x0 << 3) +#define RT5670_RXDP2_SEL_ADC (0x1 << 3) +#define RT5670_RXDP2_SEL_SFT (3) + +/* System Clock Source */ +enum { + RT5670_SCLK_S_MCLK, + RT5670_SCLK_S_PLL1, + RT5670_SCLK_S_RCCLK, +}; + +/* PLL1 Source */ +enum { + RT5670_PLL1_S_MCLK, + RT5670_PLL1_S_BCLK1, + RT5670_PLL1_S_BCLK2, + RT5670_PLL1_S_BCLK3, + RT5670_PLL1_S_BCLK4, +}; + +enum { + RT5670_AIF1, + RT5670_AIF2, + RT5670_AIF3, + RT5670_AIF4, + RT5670_AIFS, +}; + +struct rt5670_pll_code { + bool m_bp; /* Indicates bypass m code or not. */ + int m_code; + int n_code; + int k_code; +}; + +struct rt5670_priv { + struct snd_soc_codec *codec; + struct rt5670_platform_data pdata; + struct regmap *regmap; + + int sysclk; + int sysclk_src; + int lrck[RT5670_AIFS]; + int bclk[RT5670_AIFS]; + int master[RT5670_AIFS]; + + struct rt5670_pll_code pll_code; + int pll_src; + int pll_in; + int pll_out; + + int dsp_sw; /* expected parameter setting */ + int dsp_rate; + int jack_type; +}; + +#endif /* __RT5670_H__ */ -- 1.8.1.1.439.g50a6b54

3 3

[alsa-devel] Fw: Isochronous transfer error on USB3
by Mauro Carvalho Chehab 09 Jan '14

09 Jan '14

Hi Hans/Takashi, I'm getting an weird behavior with em28xx, especially when the device is connected into an audio port. I'm using, on my tests, an em28xx HVR-950 device, using this tree: http://git.linuxtv.org/mchehab/experimental.git/shortlog/refs/heads/em28xx-… Where the alsa driver is at: http://git.linuxtv.org/mchehab/experimental.git/blob/refs/heads/em28xx-v4l2… I'm testing it with xawtv3 (http://git.linuxtv.org/xawtv3.git). The ALSA userspace code there is at: http://git.linuxtv.org/xawtv3.git/blob/HEAD:/common/alsa_stream.c What happens is that, when I require xawtv3 to use any latency lower than 65 ms, the audio doesn't work, as it gets lots of underruns per second. FYI, em28xx works at a 48000 KHz sampling rate, and its PM capture Hw is described as: static struct snd_pcm_hardware snd_em28xx_hw_capture = { .info = SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_MMAP_VALID, .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_KNOT, .rate_min = 48000, .rate_max = 48000, .channels_min = 2, .channels_max = 2, .buffer_bytes_max = 62720 * 8, /* just about the value in usbaudio.c */ .period_bytes_min = 64, /* 12544/2, */ .period_bytes_max = 12544, .periods_min = 2, .periods_max = 98, /* 12544, */ }; On my tests, I experimentally discovered that the minimal latency to avoid ALSA library underruns is: - 65ms when using xHCI; - 25ms when using EHCI. Any latency lower than that causes lots of overruns. Very high latency also causes overruns (but on a lower rate, as the period is bigger). I'm wandering if is there anything that could be done either at Kernel side or at userspace side to automatically get some configuration that works as-is, without requiring the user to play with the latency parameter by hand. The alsa-info data is enclosed. Thank you! Mauro PS.: I'm still trying to understand why the minimal allowed latency is different when using xHCI, but I suspect that it is because it uses a different urb->interval than EHCI. Forwarded message: Date: Wed, 08 Jan 2014 16:03:51 -0200 From: Mauro Carvalho Chehab <m.chehab(a)samsung.com> To: Sarah Sharp <sarah.a.sharp(a)linux.intel.com> Cc: jean-philippe francois <jp.francois(a)cynove.com>, linux-usb(a)vger.kernel.org, LMML <linux-media(a)vger.kernel.org>, Shuah Khan <shuah.kh(a)samsung.com> Subject: Re: Isochronous transfer error on USB3 Em Wed, 08 Jan 2014 15:05:12 -0200 Mauro Carvalho Chehab <m.chehab(a)samsung.com> escreveu: > Em Wed, 08 Jan 2014 14:31:28 -0200 > Mauro Carvalho Chehab <m.chehab(a)samsung.com> escreveu: > > > Em Thu, 02 Jan 2014 14:07:22 -0800 > > Sarah Sharp <sarah.a.sharp(a)linux.intel.com> escreveu: > > > > > On Sun, Dec 29, 2013 at 02:54:40AM -0200, Mauro Carvalho Chehab wrote: > > > > It seems that usb_unlink_urb() is causing troubles with xHCI: the > > > > endpoint stops streaming, but, after that, it doesn't start again, > > > > and lots of debug messages are produced. I emailed you the full log > > > > after start streaming in priv (too big for vger), but basically, > > > > it produces: > > > > > > > > [ 1635.754546] xhci_hcd 0000:00:14.0: Endpoint 0x81 not halted, refusing to reset. > > > > [ 1635.754562] xhci_hcd 0000:00:14.0: Endpoint 0x82 not halted, refusing to reset. > > > > [ 1635.754577] xhci_hcd 0000:00:14.0: Endpoint 0x83 not halted, refusing to reset. > > > > [ 1635.754594] xhci_hcd 0000:00:14.0: Endpoint 0x84 not halted, refusing to reset. > > > > > > I think that's due to the driver (or userspace) attempting to reset the > > > endpoint when it didn't actually receive a stall (-EPIPE) status from an > > > URB. When that happens, the xHCI host controller endpoint "toggle" bits > > > get out of sync with the device toggle bits, and the result is that all > > > transfers will fail to the endpoint from then on until you switch > > > alternate interface settings or unplug/replug the device. > > > > > > Try this patch: > > > > > > http://marc.info/?l=linux-usb&m=138116117104619&w=2 > > > > > > It's still under RFC, and I know it has race conditions, but it will let > > > you quickly test whether this fixes your issue. > > > > Didn't work fine, or at least it didn't solve all the problems. Also, it > > started to cause OOPSes due to the race conditions. > > > > > > > > This has been a long-standing xHCI driver bug. I asked my OPW intern to > > > work on the patch to fix it, but she may be a bit busy with her new job > > > to finish up the RFC. I'll probably have to take over finishing the > > > patch, if this turns out to be your issue. > > > > > > > (Not sure why it is trying to stop all endpoints - as just one endpoint was > > > > requested to restart). > > > > > > Something is calling into usb_clear_halt() with all the endpoints. > > > Userspace, perhaps? > > > > No, userspace is not doing it. The userspace doesn't even know that this > > device is USB (and were written at the time that all media drivers were > > PCI only - so it doesn't have any USB specific call on it). > > > > > You could add WARN() calls to usb_clear_halt() to > > > see what code is resetting the endpoints. In any case, it's not part of > > > the USB core code to change configuration or alt settings, since I don't > > > see any xHCI driver output from the endpoint bandwidth code in this > > > chunk of the dmesg you sent: > > > > The em28xx-audio.c driver may need to call usb_set_interface() while > > the video is still streaming, in order to unmute the audio. That happens > > when the audio device is opened. > > > > With EHCI, this works properly. > > > > > [ 1649.640783] xhci_hcd 0000:00:14.0: Removing canceled TD starting at 0xb41e8580 (dma). > > > [ 1649.640784] xhci_hcd 0000:00:14.0: TRB to noop at offset 0xb41e8580 > > > [ 1649.643159] xhci_hcd 0000:00:14.0: Endpoint 0x81 not halted, refusing to reset. > > > [ 1649.643188] xhci_hcd 0000:00:14.0: Endpoint 0x82 not halted, refusing to reset. > > > [ 1649.643215] xhci_hcd 0000:00:14.0: Endpoint 0x83 not halted, refusing to reset. > > > [ 1649.643239] xhci_hcd 0000:00:14.0: Endpoint 0x84 not halted, refusing to reset. > > > [ 1649.735539] xhci_hcd 0000:00:14.0: ERROR no room on ep ring, try ring expansion > > > > > > Sarah Sharp > > > > Btw, sometimes, I get such logs: > > > > [ 646.192273] xhci_hcd 0000:00:14.0: Miss service interval error, set skip flag > > [ 646.192292] xhci_hcd 0000:00:14.0: Miss service interval error, set skip flag > > [ 646.192311] xhci_hcd 0000:00:14.0: Miss service interval error, set skip flag > > [ 646.192329] xhci_hcd 0000:00:14.0: Miss service interval error, set skip flag > > [ 646.192351] xhci_hcd 0000:00:14.0: Miss service interval error, set skip flag > > [ 646.192376] xhci_hcd 0000:00:14.0: Miss service interval error, set skip flag > > > > After adding some debug at em28xx-audio, triggering alsa trigger start > > events, I'm getting those: > > > > [ 3078.971224] snd_em28xx_capture_trigger: start capture > > [ 3078.971284] xhci_hcd 0000:00:14.0: ERROR no room on ep ring, try ring expansion > > [ 3078.971311] xhci_hcd 0000:00:14.0: ring expansion succeed, now has 4 segments > > [ 3078.971350] xhci_hcd 0000:00:14.0: ERROR no room on ep ring, try ring expansion > > [ 3078.971387] xhci_hcd 0000:00:14.0: ring expansion succeed, now has 8 segments > > [ 3079.034626] em28xx_audio_isocirq, 64 packets (first one with size 12) > > > > Here, some audio data arrives. > > > > [ 3079.034665] snd_em28xx_capture_trigger: stop capture > > > > It seems, however, that this didn't arrive in time, and causes an alsa > > buffer underrun. So, it cancels the existing URBs. > > > > PS.: Even with EHCI, it causes a few ALSA underruns before it gets steady. > > I suspect that this is due to em28xx time to synchronize audio and video > > streams. > > > > [ 3079.034736] xhci_hcd 0000:00:14.0: Cancel URB ffff880207900000, dev 4, ep 0x83, starting at offset 0x1ffb13850 > > [ 3079.034755] xhci_hcd 0000:00:14.0: // Ding dong! > > [ 3079.034783] xhci_hcd 0000:00:14.0: Stopped on Transfer TRB > > [ 3079.034790] snd_em28xx_capture_trigger: start capture > > > > While xHCI is still canceling the URBs, a new trigger happens, and it > > calls usb_submit_urb(). > > > > [ 3079.034819] xhci_hcd 0000:00:14.0: Removing canceled TD starting at 0x1ffb13850 (dma). > > [ 3079.034835] xhci_hcd 0000:00:14.0: TRB to noop at offset 0x1ffb13850 > > ... > > [ 3079.036341] xhci_hcd 0000:00:14.0: Removing canceled TD starting at 0xb624b850 (dma). > > [ 3079.036352] xhci_hcd 0000:00:14.0: TRB to noop at offset 0xb624b850 > > [ 3079.036365] em28xx_audio_isocirq, 64 packets (first one with size 0) > > > > But xHCI only finishes cancelling the first URB here... > > > > [ 3079.036382] xhci_hcd 0000:00:14.0: Cancel URB ffff880207900800, dev 4, ep 0x83, starting at offset 0x1ff937010 > > ... > > [ 3079.043158] xhci_hcd 0000:00:14.0: TRB to noop at offset 0x1ffb13840 > > [ 3079.043170] em28xx_audio_isocirq, 64 packets (first one with size 0) > > > > And only here, it finishes to cancel the entire operation. > > > > [ 3079.043231] xhci_hcd 0000:00:14.0: ERROR no room on ep ring, try ring expansion > > [ 3079.043299] xhci_hcd 0000:00:14.0: ring expansion succeed, now has 16 segments > > [ 3079.428996] em28xx_audio_isocirq, 64 packets (first one with size 64) > > > > Finally, after ~400ms after the new usb_submit_urb(), the first audio packet > > appears... > > > > [ 3079.429069] snd_em28xx_capture_trigger: stop capture > > > > However, this is not fast enough to avoid ALSA buffer underrun. So, > > the driver cancels the existing URBs... > > > > [ 3079.429204] xhci_hcd 0000:00:14.0: Cancel URB ffff880207900000, dev 4, ep 0x83, starting at offset 0xc5a7f4b0 > > [ 3079.429241] snd_em28xx_capture_trigger: start capture > > > > And submits a new set. > > > > Not sure how to fix it. > > Hmm... calling xawtv with a very high-latency alsa buffer works (the > default latency is 30ms, with works fine with an EHCI port): > > $ xawtv --alsa-latency 500 > > Of course, a half-second latency means that audio and video won't be > properly synchronized, but at least audio works. > > I'll turn off the USB logs and do more experiences with the latency, > in order to have an idea on how faster is EHCI to handle the > ISOC requests, when compared with xHCI. Ok, at lest on my quad-core 3rd gen i7core notebook, a latency of 65ms is enough for audio to work on em28xx with xHCI. However, using such latency on EHCI causes underruns. A latency of 90ms seems to work fine on both drivers. I'm starting to wander that maybe xHCI is not using the same urb->interval than EHCI, and if this could explain those issues. Regards, -- Cheers, Mauro -- To unsubscribe from this list: send the line "unsubscribe linux-media" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html -- Cheers, Mauro -- upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.62 !!################################ !!Script ran on: Wed Jan 8 18:29:18 UTC 2014 !!Linux Distribution !!------------------ Fedora release 20 (Heisenbug) Fedora release 20 (Heisenbug) NAME=Fedora ID=fedora PRETTY_NAME="Fedora 20 (Heisenbug)" CPE_NAME="cpe:/o:fedoraproject:fedora:20" HOME_URL="https://fedoraproject.org/" BUG_REPORT_URL="https://bugzilla.redhat.com/" REDHAT_BUGZILLA_PRODUCT="Fedora" REDHAT_BUGZILLA_PRODUCT_VERSION=20 REDHAT_SUPPORT_PRODUCT="Fedora" REDHAT_SUPPORT_PRODUCT_VERSION=20 Fedora release 20 (Heisenbug) Fedora release 20 (Heisenbug) !!DMI Information !!--------------- Manufacturer: SAMSUNG ELECTRONICS CO., LTD. Product Name: 550P5C/550P7C Product Version: P04ABI Firmware Version: P04ABI.013.130220.dg !!Kernel Information !!------------------ Kernel release: 3.13.0-rc1+ Operating System: GNU/Linux Architecture: x86_64 Processor: x86_64 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k3.13.0-rc1+ Library version: 1.0.27.2 Utilities version: 1.0.27.2 !!Loaded ALSA modules !!------------------- em28xx_alsa snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/bin/pulseaudio) Running - Yes !!Soundcards recognised by ALSA !!----------------------------- 0 [Em28xxAudio ]: Em28xx-Audio - Em28xx Audio Empia Em28xx Audio 1 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xf7910000 irq 47 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 7 Series/C210 Series Chipset Family High Definition Audio Controller (rev 04) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:1b.0 0403: 8086:1e20 (rev 04) Subsystem: 144d:c0d1 !!Loaded sound module options !!--------------------------- !!Module: em28xx_alsa debug : 0 !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC269VC Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0269 Subsystem Id: 0x144dc0d1 Revision Id: 0x100202 No Modem Function Group found Default PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=1, dir=1, wake=0, sticky=0, data=1, unsol=0 Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x4b 0x4b] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC269VC Analog", type="Audio", device=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC269VC Analog", type="Audio", device=0 Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 Amp-In vals: [0x27 0x27] Converter: stream=4, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 Amp-In vals: [0x97 0x97] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x22 Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Internal Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Internal Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 5 0x18 0x19 0x1a 0x1b 0x1d Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x03 0x0b Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0f [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] [0x00] Connection: 2 0x02 0x0b Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00010014: OUT EAPD Detect EAPD 0x2: EAPD Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x15 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x0321101f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c 0x0d* Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x17 [Pin Complex] wcaps 0x40050c: Mono Amp-Out Control: name="Bass Speaker Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Speaker Surround Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00] Pincap 0x00000010: OUT Pin Default 0x90170120: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x0f Node 0x18 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x03a11830: [Jack] Mic at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x19 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Internal Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Internal Mic Phantom Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x90a70940: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x1a [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Front Phantom Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1d [Pin Complex] wcaps 0x400400: Mono Pincap 0x00000020: IN Pin Default 0x4006a21d: [N/A] Line Out at Ext N/A Conn = Digital, Color = UNKNOWN DefAssociation = 0x1, Sequence = 0xd Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1e [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=37 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 7 0x18 0x19 0x1a 0x1b 0x1d 0x0b 0x12 Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 6 0x18 0x19 0x1a 0x1b 0x1d 0x0b Codec: Intel PantherPoint HDMI Address: 3 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x80862806 Subsystem Id: 0x80860101 Revision Id: 0x100000 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D3 CLKSTOP EPSS Power: setting=D0, actual=D0, Clock-stop-OK GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x5]: PCM AC3 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Control: name="HDMI/DP,pcm=3 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="ELD", index=0, device=3 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=01, enabled=1 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x02 Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x80] Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x58560020: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x03 Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x80] Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x58560030: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x04 Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 3 Jan 8 16:21 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 9 Jan 8 15:41 /dev/snd/controlC1 crw-rw----+ 1 root audio 116, 8 Jan 8 15:41 /dev/snd/hwC1D0 crw-rw----+ 1 root audio 116, 7 Jan 8 15:41 /dev/snd/hwC1D3 crw-rw----+ 1 root audio 116, 2 Jan 8 16:21 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 6 Jan 8 15:51 /dev/snd/pcmC1D0c crw-rw----+ 1 root audio 116, 5 Jan 8 16:21 /dev/snd/pcmC1D0p crw-rw----+ 1 root audio 116, 4 Jan 8 15:51 /dev/snd/pcmC1D3p crw-rw----+ 1 root audio 116, 1 Jan 8 15:41 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Jan 8 15:41 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x. 2 root root 80 Jan 8 16:21 . drwxr-xr-x. 3 root root 260 Jan 8 16:21 .. lrwxrwxrwx. 1 root root 12 Jan 8 16:21 pci-0000:00:1a.0-usb-0:1.2 -> ../controlC0 lrwxrwxrwx. 1 root root 12 Jan 8 15:41 pci-0000:00:1b.0 -> ../controlC1 !!ALSA configuration files !!------------------------ !!System wide config file (/etc/asound.conf) # # Place your global alsa-lib configuration here... # !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 1: PCH [HDA Intel PCH], device 0: ALC269VC Analog [ALC269VC Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Em28xxAudio [Em28xx Audio], device 0: Em28xx Audio [Empia 28xx Capture] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 0: ALC269VC Analog [ALC269VC Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Em28xxAudio] Card hw:0 'Em28xxAudio'/'Empia Em28xx Audio' Mixer name : '' Components : '' Controls : 24 Simple ctrls : 12 Simple mixer control 'Master',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [off] Front Right: 31 [100%] Playback [off] Simple mixer control 'PCM',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [off] Front Right: 31 [100%] Playback [off] Simple mixer control 'Surround',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [on] Front Right: 31 [100%] Playback [on] Simple mixer control 'LFE',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [on] Front Right: 31 [100%] Playback [on] Simple mixer control 'Line',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [off] Front Right: 31 [100%] Playback [off] Simple mixer control 'Line In',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [off] Front Right: 31 [100%] Playback [off] Simple mixer control 'CD',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [off] Front Right: 31 [100%] Playback [off] Simple mixer control 'Microphone',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [off] Front Right: 31 [100%] Playback [off] Simple mixer control 'Video',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 23 [74%] Playback [on] Front Right: 31 [100%] Playback [on] Simple mixer control 'Phone',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [off] Front Right: 31 [100%] Playback [off] Simple mixer control 'Mono',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [off] Front Right: 31 [100%] Playback [off] Simple mixer control 'AUX',0 Capabilities: volume pswitch pswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 31 Front Left: 31 [100%] Playback [off] Front Right: 31 [100%] Playback [off] !!-------Mixer controls for card 1 [PCH] Card hw:1 'PCH'/'HDA Intel PCH at 0xf7910000 irq 47' Mixer name : 'Intel PantherPoint HDMI' Components : 'HDA:10ec0269,144dc0d1,00100202 HDA:80862806,80860101,00100000' Controls : 31 Simple ctrls : 12 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 75 [86%] [-9.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 0 [0%] [-65.25dB] [off] Front Right: Playback 0 [0%] [-65.25dB] [off] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 87 [100%] [0.00dB] [on] Front Right: Playback 87 [100%] [0.00dB] [on] Simple mixer control 'Bass Speaker',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 254 [100%] [-0.20dB] Front Right: Playback 254 [100%] [-0.20dB] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 63 Front Left: Capture 39 [62%] [12.00dB] [on] Front Right: Capture 39 [62%] [12.00dB] [on] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Enabled' Simple mixer control 'Internal Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Internal Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] !!Alsactl output !!-------------- --startcollapse-- state.Em28xxAudio { control.1 { iface MIXER name 'Video Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.2 { iface MIXER name 'Video Volume' value.0 23 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.3 { iface MIXER name 'Line In Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.4 { iface MIXER name 'Line In Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.5 { iface MIXER name 'Phone Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.6 { iface MIXER name 'Phone Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.7 { iface MIXER name 'Microphone Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.8 { iface MIXER name 'Microphone Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.9 { iface MIXER name 'CD Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.10 { iface MIXER name 'CD Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.11 { iface MIXER name 'AUX Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.12 { iface MIXER name 'AUX Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.13 { iface MIXER name 'PCM Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.14 { iface MIXER name 'PCM Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.15 { iface MIXER name 'Master Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.16 { iface MIXER name 'Master Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.17 { iface MIXER name 'Line Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.18 { iface MIXER name 'Line Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.19 { iface MIXER name 'Mono Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.20 { iface MIXER name 'Mono Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.21 { iface MIXER name 'LFE Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.22 { iface MIXER name 'LFE Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } control.23 { iface MIXER name 'Surround Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.24 { iface MIXER name 'Surround Volume' value.0 31 value.1 31 comment { access 'read write' type INTEGER count 2 range '0 - 31' } } } state.PCH { control.1 { iface MIXER name 'Headphone Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 87' dbmin -6525 dbmax 0 dbvalue.0 -6525 dbvalue.1 -6525 } } control.2 { iface MIXER name 'Headphone Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.3 { iface MIXER name 'Speaker Playback Volume' value.0 87 value.1 87 comment { access 'read write' type INTEGER count 2 range '0 - 87' dbmin -6525 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.4 { iface MIXER name 'Speaker Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.5 { iface MIXER name 'Bass Speaker Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.6 { iface MIXER name 'Internal Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.7 { iface MIXER name 'Internal Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.8 { iface MIXER name 'Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.9 { iface MIXER name 'Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.10 { iface MIXER name 'Auto-Mute Mode' value Enabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.11 { iface MIXER name 'Capture Volume' value.0 39 value.1 39 comment { access 'read write' type INTEGER count 2 range '0 - 63' dbmin -1725 dbmax 3000 dbvalue.0 1200 dbvalue.1 1200 } } control.12 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.13 { iface MIXER name 'Internal Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.14 { iface MIXER name 'Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.15 { iface MIXER name 'Master Playback Volume' value 75 comment { access 'read write' type INTEGER count 1 range '0 - 87' dbmin -6525 dbmax 0 dbvalue.0 -900 } } control.16 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.17 { iface CARD name 'Internal Mic Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.18 { iface CARD name 'Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.19 { iface CARD name 'Headphone Jack' value false comment { access read type BOOLEAN count 1 } } control.20 { iface CARD name 'Speaker Front Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.21 { iface CARD name 'Speaker Surround Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.22 { iface PCM name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 comment { access read type INTEGER count 4 range '0 - 36' } } control.23 { iface PCM name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.24 { iface CARD name 'HDMI/DP,pcm=3 Jack' value false comment { access read type BOOLEAN count 1 } } control.25 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.26 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.27 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.28 { iface MIXER name 'IEC958 Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.29 { iface PCM device 3 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.30 { iface PCM device 3 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.31 { iface MIXER name 'PCM Playback Volume' value.0 254 value.1 254 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 -20 dbvalue.1 -20 } } } --endcollapse-- !!All Loaded Modules !!------------------ Module fuse ccm nf_conntrack_netbios_ns nf_conntrack_broadcast ipt_MASQUERADE ip6t_REJECT xt_conntrack ebtable_nat ebtable_broute bridge stp llc ebtable_filter ebtables ip6table_nat nf_conntrack_ipv6 nf_defrag_ipv6 nf_nat_ipv6 ip6table_mangle ip6table_security ip6table_raw ip6table_filter ip6_tables iptable_nat nf_conntrack_ipv4 nf_defrag_ipv4 nf_nat_ipv4 nf_nat nf_conntrack iptable_mangle iptable_security iptable_raw bnep vfat fat arc4 iwldvm mac80211 iwlwifi cfg80211 x86_pkg_temp_thermal coretemp kvm_intel kvm crc32_pclmul crc32c_intel ghash_clmulni_intel netconsole rc_hauppauge em28xx_rc rc_core lgdt330x snd_hda_codec_hdmi snd_hda_codec_realtek em28xx_dvb dvb_core snd_hda_intel snd_hda_codec em28xx_alsa snd_hwdep snd_seq snd_seq_device snd_pcm snd_page_alloc snd_timer snd nfsd auth_rpcgss tuner_xc2028 tuner tvp5150 em28xx_v4l uvcvideo em28xx videobuf2_vmalloc tveeprom videobuf2_memops v4l2_common videobuf2_core videodev media btusb bluetooth joydev serio_raw iTCO_wdt iTCO_vendor_support rfkill soundcore r8169 mii shpchp mei_me mei microcode i2c_i801 lpc_ich mfd_core nfs_acl lockd sunrpc nouveau i915 ttm i2c_algo_bit drm_kms_helper drm i2c_core mxm_wmi wmi video !!Sysfs Files !!----------- /sys/class/sound/hwC1D0/init_pin_configs: 0x12 0x411111f0 0x14 0x411111f0 0x15 0x0321101f 0x17 0x90170120 0x18 0x03a11830 0x19 0x90a70940 0x1a 0x411111f0 0x1b 0x90170110 0x1d 0x4006a21d 0x1e 0x411111f0 /sys/class/sound/hwC1D0/driver_pin_configs: /sys/class/sound/hwC1D0/user_pin_configs: /sys/class/sound/hwC1D0/init_verbs: /sys/class/sound/hwC1D0/hints: /sys/class/sound/hwC1D3/init_pin_configs: 0x05 0x18560010 0x06 0x58560020 0x07 0x58560030 /sys/class/sound/hwC1D3/driver_pin_configs: /sys/class/sound/hwC1D3/user_pin_configs: /sys/class/sound/hwC1D3/init_verbs: /sys/class/sound/hwC1D3/hints: !!ALSA/HDA dmesg !!-------------- [ 24.204127] em28xx: Registered (Em28xx Audio Extension) extension [ 24.207614] snd_hda_intel 0000:00:1b.0: enabling device (0000 -> 0002) [ 24.209216] snd_hda_intel 0000:00:1b.0: irq 47 for MSI/MSI-X [ 24.213022] em2882/3 #0: Binding DVB extension [ 24.222898] ALSA sound/pci/hda/patch_realtek.c:419 SKU: Nid=0x1d sku_cfg=0x4006a21d [ 24.222901] ALSA sound/pci/hda/patch_realtek.c:421 SKU: port_connectivity=0x1 [ 24.222902] ALSA sound/pci/hda/patch_realtek.c:422 SKU: enable_pcbeep=0x0 [ 24.222903] ALSA sound/pci/hda/patch_realtek.c:423 SKU: check_sum=0x00000006 [ 24.222904] ALSA sound/pci/hda/patch_realtek.c:424 SKU: customization=0x000000a2 [ 24.222905] ALSA sound/pci/hda/patch_realtek.c:425 SKU: external_amp=0x3 [ 24.222906] ALSA sound/pci/hda/patch_realtek.c:426 SKU: platform_type=0x1 [ 24.222908] ALSA sound/pci/hda/patch_realtek.c:427 SKU: swap=0x0 [ 24.222909] ALSA sound/pci/hda/patch_realtek.c:428 SKU: override=0x1 [ 24.223221] ALSA sound/pci/hda/hda_auto_parser.c:393 autoconfig: line_outs=2 (0x1b/0x17/0x0/0x0/0x0) type:speaker [ 24.223223] ALSA sound/pci/hda/hda_auto_parser.c:397 speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 24.223224] ALSA sound/pci/hda/hda_auto_parser.c:401 hp_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 24.223225] ALSA sound/pci/hda/hda_auto_parser.c:402 mono: mono_out=0x0 [ 24.223226] ALSA sound/pci/hda/hda_auto_parser.c:406 inputs: [ 24.223228] ALSA sound/pci/hda/hda_auto_parser.c:410 Internal Mic=0x19 [ 24.223230] ALSA sound/pci/hda/hda_auto_parser.c:410 Mic=0x18 [ 24.223232] ALSA sound/pci/hda/patch_realtek.c:491 realtek: No valid SSID, checking pincfg 0x4006a21d for NID 0x1d [ 24.223234] ALSA sound/pci/hda/patch_realtek.c:507 realtek: Enabling init ASM_ID=0xa21d CODEC_ID=10ec0269 [ 24.236439] input: HDA Intel PCH HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1b.0/sound/card1/input11 [ 24.237093] input: HDA Intel PCH Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card1/input10 [ 24.237688] input: HDA Intel PCH Mic as /devices/pci0000:00/0000:00:1b.0/sound/card1/input9 [ 24.257693] xc2028: Xcv2028/3028 init called! -- [ 873.141600] URB ffff88022219f800 submitted while active [ 873.141619] Modules linked in: fuse ccm nf_conntrack_netbios_ns nf_conntrack_broadcast ipt_MASQUERADE ip6t_REJECT xt_conntrack ebtable_nat ebtable_broute bridge stp llc ebtable_filter ebtables ip6table_nat nf_conntrack_ipv6 nf_defrag_ipv6 nf_nat_ipv6 ip6table_mangle ip6table_security ip6table_raw ip6table_filter ip6_tables iptable_nat nf_conntrack_ipv4 nf_defrag_ipv4 nf_nat_ipv4 nf_nat nf_conntrack iptable_mangle iptable_security iptable_raw bnep vfat fat arc4 iwldvm mac80211 iwlwifi cfg80211 x86_pkg_temp_thermal coretemp kvm_intel kvm crc32_pclmul crc32c_intel ghash_clmulni_intel netconsole rc_hauppauge em28xx_rc rc_core lgdt330x snd_hda_codec_hdmi snd_hda_codec_realtek em28xx_dvb dvb_core snd_hda_intel snd_hda_codec em28xx_alsa snd_hwdep snd_seq snd_seq_device snd_pcm snd_page_alloc snd_timer snd [ 873.143618] nfsd auth_rpcgss tuner_xc2028 tuner tvp5150 em28xx_v4l uvcvideo em28xx videobuf2_vmalloc tveeprom videobuf2_memops v4l2_common videobuf2_core videodev media btusb bluetooth joydev serio_raw iTCO_wdt iTCO_vendor_support rfkill soundcore r8169 mii shpchp mei_me mei microcode i2c_i801 lpc_ich mfd_core nfs_acl lockd sunrpc nouveau i915 ttm i2c_algo_bit drm_kms_helper drm i2c_core mxm_wmi wmi video [ 873.145388] CPU: 4 PID: 2103 Comm: kworker/4:0 Not tainted 3.13.0-rc1+ #22

3 5

[alsa-devel] [PATCH] ASoC: fsl_ssi: Set default slot number for common cases
by Nicolin Chen 09 Jan '14

09 Jan '14

For those platforms using DAI master mode like I2S, it's better to pre-set a default slot number so that there's no need for these common cases to set the slot number from its machine driver any more. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 94dedcb..57ab45b 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -711,6 +711,16 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) if (ssi_private->imx_ac97) fsl_ssi_setup_ac97(ssi_private); + /* Set a default slot number so that there is no need for those common + * cases like I2S mode to call the extra set_tdm_slot() any more. + */ + if (!ssi_private->imx_ac97) { + write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(2)); + write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(2)); + } + return 0; } -- 1.8.4

2 3

[alsa-devel] [PATCH 1/2] ALSA: Remove superfluous header inclusions in memalloc.c
by Takashi Iwai 09 Jan '14

09 Jan '14

After cutting off the proc and page reservation codes, we don't need many headers any longer. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index be1544ac4613..e568c14ca88d 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -22,17 +22,13 @@ */ #include <linux/module.h> -#include <linux/proc_fs.h> #include <linux/init.h> -#include <linux/pci.h> #include <linux/slab.h> #include <linux/mm.h> -#include <linux/seq_file.h> #include <asm/uaccess.h> #include <linux/dma-mapping.h> #include <linux/genalloc.h> #include <linux/moduleparam.h> -#include <linux/mutex.h> #include <sound/memalloc.h> -- 1.8.5.2

1 1

[alsa-devel] [PATCH] Support for Yamaha MOXF
by Michael Slade 09 Jan '14

09 Jan '14

My apologies if I'm doing this wrong. I have managed to get a Yamaha MOXF8 working with the attached patch. Presumably also works with a MOXF6. The new Yamaha MOXF6/8 are successors to the MOX6/8. Apart from things like new voices, a flash expansion port, and a different colour at the ends, they are pretty much identical in operation. The MOXF has a USB ID of 0499:150f. So I needed to add a new quirks-table.h stanza for it, by copy-pasting the MOX one and changing he ID. I haven't tested it to death or anything, but I am having no trouble recording and playing audio and midi. Let me know if this patch is acceptable or of there is more that is needed from me. Note, this patch is against Ubuntu Raring's 3.8.0 kernel, but it's a trivial patch. Mick.

3 3

[alsa-devel] [PATCH 1/4] ASoC: simple-card: add simple audio card widgets support
by Xiubo Li 09 Jan '14

09 Jan '14

For many audio cards there need to add some off-CODEC widgets, and this is hard to be supported by dts only, because maybe some widgets need one or more call backs, such as wevent. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- .../devicetree/bindings/sound/simple-card.txt | 1 + include/sound/simple_card.h | 14 ++++ sound/soc/generic/Makefile | 3 +- sound/soc/generic/simple-card-widgets.c | 79 ++++++++++++++++++++++ 4 files changed, 96 insertions(+), 1 deletion(-) create mode 100644 sound/soc/generic/simple-card-widgets.c diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index e9e20ec..e626302 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -15,6 +15,7 @@ Optional properties: Each entry is a pair of strings, the first being the connection's sink, the second being the connection's source. +- simple-sudio-card,widgets : The name of the audio card off-CODEC widgets. Required subnodes: diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index 6c74527..75574f8 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -12,8 +12,16 @@ #ifndef __SIMPLE_CARD_H #define __SIMPLE_CARD_H +#include <linux/of.h> #include <sound/soc.h> +struct asoc_simple_card_widgets { + const char *name; + struct list_head list; + const struct snd_soc_dapm_widget *widgets; + unsigned int widgets_cnt; +}; + struct asoc_simple_dai { const char *name; unsigned int fmt; @@ -35,4 +43,10 @@ struct asoc_simple_card_info { struct snd_soc_card snd_card; }; +struct asoc_simple_card_widgets * +asoc_simple_card_get_widgets(struct device_node *); +int asoc_simple_card_widgets_register(struct asoc_simple_card_widgets *); +void +asoc_simple_card_widgets_unregister(struct asoc_simple_card_widgets *); + #endif /* __SIMPLE_CARD_H */ diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 9c3b246..dbacd12 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -1,3 +1,4 @@ -snd-soc-simple-card-objs := simple-card.o +snd-soc-simple-card-objs := simple-card-widgets.o +snd-soc-simple-card-objs += simple-card.o obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o diff --git a/sound/soc/generic/simple-card-widgets.c b/sound/soc/generic/simple-card-widgets.c new file mode 100644 index 0000000..6eb0c8c --- /dev/null +++ b/sound/soc/generic/simple-card-widgets.c @@ -0,0 +1,79 @@ +/* + * ASoC simple sound card widgets support + * + * Copyright 2013 Freescale Semiconductor, Inc. + * Xiubo Li <Li.Xiubo(a)freescale.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/of.h> +#include <linux/platform_device.h> +#include <sound/simple_card.h> + +static DEFINE_MUTEX(simple_widgets_mutex); +static LIST_HEAD(simple_widgets_list); + +void +asoc_simple_card_widgets_unregister(struct asoc_simple_card_widgets *comp) +{ + mutex_lock(&simple_widgets_mutex); + list_del(&comp->list); + mutex_unlock(&simple_widgets_mutex); +} +EXPORT_SYMBOL_GPL(asoc_simple_card_widgets_unregister); + +int +asoc_simple_card_widgets_register(struct asoc_simple_card_widgets *comp) +{ + if (!comp) { + pr_err("Simple Card: Failed to register widgets\n"); + return -EINVAL; + } + + mutex_lock(&simple_widgets_mutex); + list_add(&comp->list, &simple_widgets_list); + mutex_unlock(&simple_widgets_mutex); + + return 0; +} +EXPORT_SYMBOL_GPL(asoc_simple_card_widgets_register); + +struct asoc_simple_card_widgets * +asoc_simple_card_get_widgets(struct device_node *np) +{ + struct asoc_simple_card_widgets *comp; + const char *string; + int ret; + + if (!np) + return ERR_PTR(-EINVAL); + + if (!of_property_read_bool(np, "simple-audio-card,widgets")) + return NULL; + + ret = of_property_read_string(np, "simple-audio-card,widgets", + &string); + if (ret < 0) + return ERR_PTR(ret); + + mutex_lock(&simple_widgets_mutex); + list_for_each_entry(comp, &simple_widgets_list, list) { + if (!strcmp(string, comp->name)) { + mutex_unlock(&simple_widgets_mutex); + return comp; + } + + } + mutex_unlock(&simple_widgets_mutex); + + return ERR_PTR(-EPROBE_DEFER); +} +EXPORT_SYMBOL_GPL(asoc_simple_card_get_widgets); + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:asoc-simple-card-widgets"); +MODULE_DESCRIPTION("ASoC Simple Sound Card Widgets"); +MODULE_AUTHOR("Xiubo Li <Li.Xiubo(a)freescale.com>"); -- 1.8.4

3 7

[alsa-devel] [PATCH 0/3] ALSA: hda - enable display audio for Intel Broadwell
by mengdong.lin＠intel.com 09 Jan '14

09 Jan '14

From: Mengdong Lin <mengdong.lin(a)intel.com> The following patches enable HDMI/DP audio for Intel Broadwell platform from audio driver side. Gfx driver just reuses Haswell audio enabling code. Mengdong Lin (3): ALSA: hda - add device ID for Broadwell display audio controller ALSA: hda - add codec ID for Broadwell display audio codec ALSA: hda/hdmi - apply all Haswell fix-ups to Broadwell display codec sound/pci/hda/hda_intel.c | 3 +++ sound/pci/hda/patch_hdmi.c | 17 +++++++++++------ 2 files changed, 14 insertions(+), 6 deletions(-) -- 1.8.1.2

2 4

[alsa-devel] [PATCH] ASoC: da9055: Partially fix device registration
by Mark Brown 08 Jan '14

08 Jan '14

From: Mark Brown <broonie(a)linaro.org> It has never been possible to even load the da9055 driver since both the MFD and CODEC drivers register the I2C device "da9055". Fix this by using the usual pattern for MFDs with multiple I2C addresses and having the CODEC driver reference an I2C dummy registered by the MFD. Since I don't know which I2C address to use for the CODEC a FIXME has been left in the MFD, this doesn't make anything any worse since the device has never been able to load in the first place. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- drivers/mfd/da9055-core.c | 3 +++ drivers/mfd/da9055-i2c.c | 2 ++ include/linux/mfd/da9055/core.h | 2 +- sound/soc/codecs/da9055.c | 49 ++++++++++++++++++++--------------------- 4 files changed, 30 insertions(+), 26 deletions(-) diff --git a/drivers/mfd/da9055-core.c b/drivers/mfd/da9055-core.c index caf8dcffd0ad..fb0284cda4a5 100644 --- a/drivers/mfd/da9055-core.c +++ b/drivers/mfd/da9055-core.c @@ -365,6 +365,9 @@ static const struct mfd_cell da9055_devs[] = { .of_compatible = "dialog,da9055-watchdog", .name = "da9055-watchdog", }, + { + .name = "da9055-codec", + }, }; static struct regmap_irq_chip da9055_regmap_irq_chip = { diff --git a/drivers/mfd/da9055-i2c.c b/drivers/mfd/da9055-i2c.c index 13af7e50021e..4f5e18750e1a 100644 --- a/drivers/mfd/da9055-i2c.c +++ b/drivers/mfd/da9055-i2c.c @@ -39,6 +39,8 @@ static int da9055_i2c_probe(struct i2c_client *i2c, da9055->dev = &i2c->dev; da9055->chip_irq = i2c->irq; + /* FIXME: Should use i2c_add_dummy() for the CODEC */ + i2c_set_clientdata(i2c, da9055); return da9055_device_init(da9055); diff --git a/include/linux/mfd/da9055/core.h b/include/linux/mfd/da9055/core.h index 956afa445998..e293797ded66 100644 --- a/include/linux/mfd/da9055/core.h +++ b/include/linux/mfd/da9055/core.h @@ -42,7 +42,7 @@ struct da9055 { struct regmap *regmap; struct regmap_irq_chip_data *irq_data; struct device *dev; - struct i2c_client *i2c_client; + struct i2c_client *i2c_client, *codec; int irq_base; int chip_irq; diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 52b79a487ac7..88c1a884aed8 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -14,10 +14,10 @@ */ #include <linux/delay.h> -#include <linux/i2c.h> #include <linux/regmap.h> #include <linux/slab.h> #include <linux/module.h> +#include <linux/mfd/da9055/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -1474,7 +1474,7 @@ static struct snd_soc_codec_driver soc_codec_dev_da9055 = { .num_dapm_routes = ARRAY_SIZE(da9055_audio_map), }; -static const struct regmap_config da9055_regmap_config = { +static const struct regmap_config da9055_codec_regmap_config = { .reg_bits = 8, .val_bits = 8, @@ -1484,14 +1484,19 @@ static const struct regmap_config da9055_regmap_config = { .cache_type = REGCACHE_RBTREE, }; -static int da9055_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +static int da9055_dev_probe(struct platform_device *pdev) { + struct da9055 *core = dev_get_drvdata(pdev->dev.parent); struct da9055_priv *da9055; - struct da9055_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct da9055_platform_data *pdata = dev_get_platdata(&pdev->dev); int ret; - da9055 = devm_kzalloc(&i2c->dev, sizeof(struct da9055_priv), + if (!core->codec) { + dev_err(&pdev->dev, "No CODEC device registered\n"); + return -ENODEV; + } + + da9055 = devm_kzalloc(&pdev->dev, sizeof(struct da9055_priv), GFP_KERNEL); if (!da9055) return -ENOMEM; @@ -1499,48 +1504,42 @@ static int da9055_i2c_probe(struct i2c_client *i2c, if (pdata) da9055->pdata = pdata; - i2c_set_clientdata(i2c, da9055); + platform_set_drvdata(pdev, da9055); - da9055->regmap = devm_regmap_init_i2c(i2c, &da9055_regmap_config); + da9055->regmap = devm_regmap_init_i2c(core->codec, + &da9055_codec_regmap_config); if (IS_ERR(da9055->regmap)) { ret = PTR_ERR(da9055->regmap); - dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + dev_err(&pdev->dev, "regmap_init() failed: %d\n", ret); return ret; } - ret = snd_soc_register_codec(&i2c->dev, + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_da9055, &da9055_dai, 1); if (ret < 0) { - dev_err(&i2c->dev, "Failed to register da9055 codec: %d\n", + dev_err(&pdev->dev, "Failed to register da9055 codec: %d\n", ret); } return ret; } -static int da9055_remove(struct i2c_client *client) +static int da9055_dev_remove(struct platform_device *pdev) { - snd_soc_unregister_codec(&client->dev); + snd_soc_unregister_codec(&pdev->dev); return 0; } -static const struct i2c_device_id da9055_i2c_id[] = { - { "da9055", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); - /* I2C codec control layer */ -static struct i2c_driver da9055_i2c_driver = { +static struct platform_driver da9055_codec_driver = { .driver = { - .name = "da9055", + .name = "da9055-codec", .owner = THIS_MODULE, }, - .probe = da9055_i2c_probe, - .remove = da9055_remove, - .id_table = da9055_i2c_id, + .probe = da9055_dev_probe, + .remove = da9055_dev_remove, }; -module_i2c_driver(da9055_i2c_driver); +module_platform_driver(da9055_codec_driver); MODULE_DESCRIPTION("ASoC DA9055 Codec driver"); MODULE_AUTHOR("David Chen, Ashish Chavan"); -- 1.8.5.2

2 1

Re: [alsa-devel] snd-usb: "delay: estimated 0, actual 352"
by Takashi Iwai 08 Jan '14

08 Jan '14

At Thu, 6 Sep 2012 15:17:58 +0200, Markus Trippelsdorf wrote: > > On 2012.09.06 at 15:08 +0200, Takashi Iwai wrote: > > At Thu, 6 Sep 2012 11:43:48 +0200, > > Markus Trippelsdorf wrote: > > > > > > On 2012.09.06 at 10:21 +0200, Takashi Iwai wrote: > > > > At Thu, 06 Sep 2012 09:35:26 +0200, > > > > Takashi Iwai wrote: > > > > > > > > In short, a patch like below may fix the issue (note: completely > > > > untested!) > > > > > > No it doesn't, unfortunately... > > > > OK, I start tracking down the problem a bit more deeply now. > > > > The issue happens when the first two URBs are passed to > > retire_playback_urb(). These are URBs filled before start_endpoints() > > are set, so they contain actually zero size. Even though these are > > a sort of dummy packets, the driver still tries to check with the > > queued delay account, and gives bogus errors. > > > > So, essentially the messages are harmless and nothing to worry too > > much, but surely it doesn't look sexy. > > > > The patch below should fix the problem. Please give it a try. > > Yes, your patch finally fixes the problem. > Thank you Takashi-san. Thanks for your quick test! If Daniel has no objection with that patch, I'm going to merge it. Takashi

3 2

[alsa-devel] [PATCH 1/4] ASoC: max98088: Use params_width() rather than memory format
by Mark Brown 08 Jan '14

08 Jan '14

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/max98088.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 53d7dab4e054..ee660e2d3df3 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1233,12 +1233,12 @@ static int max98088_dai1_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (params_width(params)) { + case 16: snd_soc_update_bits(codec, M98088_REG_14_DAI1_FORMAT, M98088_DAI_WS, 0); break; - case SNDRV_PCM_FORMAT_S24_LE: + case 24: snd_soc_update_bits(codec, M98088_REG_14_DAI1_FORMAT, M98088_DAI_WS, M98088_DAI_WS); break; -- 1.8.5.2

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[alsa-devel] [PATCH 1/2 v2] ASoC: wm_adsp: Factor out ADSP2 boot proceedure
by Charles Keepax 08 Jan '14

08 Jan '14

Move the ADSP2 boot proceedure into a work structure in preparation for running it asynchronously with the reset of the audio path bring up. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 181 +++++++++++++++++++++++++------------------- sound/soc/codecs/wm_adsp.h | 2 + 2 files changed, 105 insertions(+), 78 deletions(-) Changes since v1: - Schedule to the system unbound work queue since it is likely to be a fairly long task. - Move final start signal to the DSP to be an out_drv widget such that the download is overlapped with the output enables. Thanks, Charles diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index fb0c678..41bbac9 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1497,107 +1497,131 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) return 0; } -int wm_adsp2_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +void wm_adsp2_boot_work(struct work_struct *work) { - struct snd_soc_codec *codec = w->codec; - struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); - struct wm_adsp *dsp = &dsps[w->shift]; - struct wm_adsp_alg_region *alg_region; - struct wm_coeff_ctl *ctl; - unsigned int val; + struct wm_adsp *dsp = container_of(work, + struct wm_adsp, + boot_work); int ret; + unsigned int val; - dsp->card = codec->card; + /* + * For simplicity set the DSP clock rate to be the + * SYSCLK rate rather than making it configurable. + */ + ret = regmap_read(dsp->regmap, ARIZONA_SYSTEM_CLOCK_1, &val); + if (ret != 0) { + adsp_err(dsp, "Failed to read SYSCLK state: %d\n", ret); + return; + } + val = (val & ARIZONA_SYSCLK_FREQ_MASK) + >> ARIZONA_SYSCLK_FREQ_SHIFT; - switch (event) { - case SND_SOC_DAPM_POST_PMU: - /* - * For simplicity set the DSP clock rate to be the - * SYSCLK rate rather than making it configurable. - */ - ret = regmap_read(dsp->regmap, ARIZONA_SYSTEM_CLOCK_1, &val); - if (ret != 0) { - adsp_err(dsp, "Failed to read SYSCLK state: %d\n", - ret); - return ret; - } - val = (val & ARIZONA_SYSCLK_FREQ_MASK) - >> ARIZONA_SYSCLK_FREQ_SHIFT; + ret = regmap_update_bits_async(dsp->regmap, + dsp->base + ADSP2_CLOCKING, + ADSP2_CLK_SEL_MASK, val); + if (ret != 0) { + adsp_err(dsp, "Failed to set clock rate: %d\n", ret); + return; + } - ret = regmap_update_bits_async(dsp->regmap, - dsp->base + ADSP2_CLOCKING, - ADSP2_CLK_SEL_MASK, val); + if (dsp->dvfs) { + ret = regmap_read(dsp->regmap, + dsp->base + ADSP2_CLOCKING, &val); if (ret != 0) { - adsp_err(dsp, "Failed to set clock rate: %d\n", - ret); - return ret; + dev_err(dsp->dev, "Failed to read clocking: %d\n", ret); + return; } - if (dsp->dvfs) { - ret = regmap_read(dsp->regmap, - dsp->base + ADSP2_CLOCKING, &val); + if ((val & ADSP2_CLK_SEL_MASK) >= 3) { + ret = regulator_enable(dsp->dvfs); if (ret != 0) { dev_err(dsp->dev, - "Failed to read clocking: %d\n", ret); - return ret; + "Failed to enable supply: %d\n", + ret); + return; } - if ((val & ADSP2_CLK_SEL_MASK) >= 3) { - ret = regulator_enable(dsp->dvfs); - if (ret != 0) { - dev_err(dsp->dev, - "Failed to enable supply: %d\n", - ret); - return ret; - } - - ret = regulator_set_voltage(dsp->dvfs, - 1800000, - 1800000); - if (ret != 0) { - dev_err(dsp->dev, - "Failed to raise supply: %d\n", - ret); - return ret; - } + ret = regulator_set_voltage(dsp->dvfs, + 1800000, + 1800000); + if (ret != 0) { + dev_err(dsp->dev, + "Failed to raise supply: %d\n", + ret); + return; } } + } - ret = wm_adsp2_ena(dsp); - if (ret != 0) - return ret; + ret = wm_adsp2_ena(dsp); + if (ret != 0) + return; - ret = wm_adsp_load(dsp); - if (ret != 0) - goto err; + ret = wm_adsp_load(dsp); + if (ret != 0) + goto err; - ret = wm_adsp_setup_algs(dsp); - if (ret != 0) - goto err; + ret = wm_adsp_setup_algs(dsp); + if (ret != 0) + goto err; - ret = wm_adsp_load_coeff(dsp); - if (ret != 0) - goto err; + ret = wm_adsp_load_coeff(dsp); + if (ret != 0) + goto err; - /* Initialize caches for enabled and unset controls */ - ret = wm_coeff_init_control_caches(dsp); - if (ret != 0) - goto err; + /* Initialize caches for enabled and unset controls */ + ret = wm_coeff_init_control_caches(dsp); + if (ret != 0) + goto err; - /* Sync set controls */ - ret = wm_coeff_sync_controls(dsp); - if (ret != 0) - goto err; + /* Sync set controls */ + ret = wm_coeff_sync_controls(dsp); + if (ret != 0) + goto err; + + ret = regmap_update_bits_async(dsp->regmap, + dsp->base + ADSP2_CONTROL, + ADSP2_CORE_ENA, + ADSP2_CORE_ENA); + if (ret != 0) + goto err; + + dsp->running = true; + + return; - ret = regmap_update_bits_async(dsp->regmap, - dsp->base + ADSP2_CONTROL, - ADSP2_CORE_ENA | ADSP2_START, - ADSP2_CORE_ENA | ADSP2_START); +err: + regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, + ADSP2_SYS_ENA | ADSP2_CORE_ENA | ADSP2_START, 0); +} + +int wm_adsp2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); + struct wm_adsp *dsp = &dsps[w->shift]; + struct wm_adsp_alg_region *alg_region; + struct wm_coeff_ctl *ctl; + int ret; + + dsp->card = codec->card; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + queue_work(system_unbound_wq, &dsp->boot_work); + flush_work(&dsp->boot_work); + + if (!dsp->running) + return -EIO; + + ret = regmap_update_bits(dsp->regmap, + dsp->base + ADSP2_CONTROL, + ADSP2_START, + ADSP2_START); if (ret != 0) goto err; - - dsp->running = true; break; case SND_SOC_DAPM_PRE_PMD: @@ -1668,6 +1692,7 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) INIT_LIST_HEAD(&adsp->alg_regions); INIT_LIST_HEAD(&adsp->ctl_list); + INIT_WORK(&adsp->boot_work, wm_adsp2_boot_work); if (dvfs) { adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD"); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index d018dea..b172c1d 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -59,6 +59,8 @@ struct wm_adsp { struct regulator *dvfs; struct list_head ctl_list; + + struct work_struct boot_work; }; #define WM_ADSP1(wname, num) \ -- 1.7.2.5

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