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[alsa-devel] [PATCH 1/1] ALSA: compress: no need to set metadata before next track
by Richard Fitzgerald 26 Feb '14

26 Feb '14

The metadata is mainly for MP3 gapless playback, since the MP3 audio stream does not contain enough information to enable gapless. Other audio formats do not necessarily require any additional metadata so we should allow next_track to be called without any metadata. Signed-off-by: Zhao Weijia <weijia.zhao(a)capelabs.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 6 ------ 1 files changed, 0 insertions(+), 6 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 7a20897..5a351c7 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -744,12 +744,6 @@ static int snd_compr_next_track(struct snd_compr_stream *stream) if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) return -EPERM; - /* you can signal next track isf this is intended to be a gapless stream - * and current track metadata is set - */ - if (stream->metadata_set == false) - return -EPERM; - retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_NEXT_TRACK); if (retval != 0) return retval; -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: Intel: Baytrail: Fix implicit declaration of function 'memcpy_fromio'
by Jarkko Nikula 26 Feb '14

26 Feb '14

Some kernel configurations can cause following build error: sound/soc/intel/sst-baytrail-ipc.c: In function ‘sst_byt_get_dsp_position’: sound/soc/intel/sst-baytrail-ipc.c:744:2: error: implicit declaration of function ‘memcpy_fromio’ [-Werror=implicit-function-declaration] memcpy_fromio(&fw_tstamp, ^ cc1: some warnings being treated as errors Fix this by including <asm/io.h> explicitly. Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- Also sst-dsp., sst-baytrail-dsp.c and sst-haswell-dsp.c are using memcpy_fromio/_toio but there <asm/io.h> seems to come via <linux/dma-mapping.h> or <linux/io.h> so I didn't touch them. --- sound/soc/intel/sst-baytrail-ipc.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index 8c91a68b90bc..8b067e34094e 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -27,6 +27,7 @@ #include <linux/kthread.h> #include <linux/firmware.h> #include <asm/div64.h> +#include <asm/io.h> #include "sst-baytrail-ipc.h" #include "sst-dsp.h" -- 1.8.5.3

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[alsa-devel] [alsa-plugins] [PATCH 2/2] Allow number of channels
by Corfu, ValentinX 26 Feb '14

26 Feb '14

Intel GmbH Dornacher Strasse 1 85622 Feldkirchen/Muenchen, Deutschland Sitz der Gesellschaft: Feldkirchen bei Muenchen Geschaeftsfuehrer: Christian Lamprechter, Hannes Schwaderer, Douglas Lusk Registergericht: Muenchen HRB 47456 Ust.-IdNr./VAT Registration No.: DE129385895 Citibank Frankfurt a.M. (BLZ 502 109 00) 600119052

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[alsa-devel] [PATCH] Bugfix: Fix resampling when client and slave both use format float
by Maarten Baert 26 Feb '14

26 Feb '14

I found a bug in libasound. When both the client and the slave (in my case, the JACK plugin) try to use the float format, but the sample rate or channel count does not match, libasound *should* insert linear conversion plugins to convert from float to linear, then resample/remap channels, and then convert back to float (because apparently the resamplers and channel remapper don't support floating point, only 'linear' i.e. integers). Currently this doesn't work, snd_pcm_plug_change_format doesn't know what to do and simply returns EINVAL. As a result, snd_pcm_hw_params fails even though the HW params were perfectly valid (it indicates that both the float format and any sample rate are supported). In my test, this broke audio for WINE (and any other application that tries to use float, such as aplay with the right settings) when I wanted to use the JACK plugin (which only supports the float format). This patch fixes this bug by doing a conversion to S16 and back when resampling or remapping is needed. And while I was at it, I also removed a check that had no effect because the exact same check is already being done at the start of the function. I still think it's a bit silly that libasound requires integers for resampling, because both libsamplerate and libspeex use float internally for resampling. So now ALSA is literally doing a float-to-s16-to-float-to-s16-to-float conversion. But changing that would have been a lot harder. Maarten Baert

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[alsa-devel] [RFC v3] [PATCH 00/52] Enhancement for support of firewire devices
by Takashi Sakamoto 26 Feb '14

26 Feb '14

This series of patch is to update patches which I've posted below. [RFC v2][PATCH 00/38] Enhancement for support of some firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2013-December/070424.h… [RFC][PATCH v2 00/13] speakers: Add support for capture/playback of PCM/MIDI http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071071.ht… This series may be last RFC. Repository: https://github.com/takaswie/sound For backport to Linux kernel 3.11 or later: https://github.com/takaswie/snd-firewire-improve For maintainers: I have 4 items. 1.expected impacts to user-land Refer to my last message. 2. Address allocation for Fireworks drivers Related to this suggestion: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071120.ht… The alternative way is to implement: 1.'AV/C SIGNAL FORMAT' commanids defined in 'AV/C Digital Interface Command Set General Specification Version 4.2 (Sep 2004, 1394TA)' - 'snd-firewire-lib' has these commands 2.'AV/C DESCRIPTOR' mechanism defined in 'Enhancements to the AV/C General Specification 3.0 Version 1.1 (Oct 2000, 1394TA)' 3.'AV/C SIGNAL SOURCE' commands defined in 'AV/C Music Subunit 1.0' - I don't have this specification. They cost higher than actual effects. Especially, 2 is a bit complicated and need much codes. Actually FFADO has all of them. And snd-fireworks gives hwdep interface for application to command/response. There is currently several ways in user-land so I think it good to keep the address non-relocatable yet. 3.Support for devices with fatal quirk M-Audio 'Firewire 1814' and 'ProjectMix I/O' has BeBoB firmware which is heavily customized. See last patch. They have a time not to respond transaction. As long as I experienced, the time may be 300 msec per 1 sec. As a result, snd-bebob often fail to handle them. Especially, they can ignore transactions for resource management of connection/channel/bandwidth. This is a fatal for host system. But there is no user-land driver for these devices and I received some requests from FFADO users to support them. I hope to get your opinion about this support. 4.Rest of my work I've finished almost work for them. If I receive no big problems from developers and testers, I have a plan to post merge-request. Updates since previous patches: snd-firewire-lib: - remove wrong line to regain 'stream_running' member to 'struct fwspk' - allow drivers to indicate timeout value - remove 'amdtp_stream_midi_running' snd-bebob: - add reference counter for MIDI substreams - improve 'mutex_lock()/mutex_unlock()' for streaming - remove 'device' member fron 'struct snd_bebob' - update device entry snd-fireworks: - add reference counter for MIDI substreams - improve 'mutex_lock()/mutex_unlock()' for streaming - remove 'device' member from 'struct snd_fireworks' snd-firewire-speakers: - rename to snd-oxfw and obsolete snd-oxfw: - add reference counter for MIDI substreams - improve 'mutex_lock()/mutex_unlock()' for streaming - fix lock mechanism of streaming by hwdep - fix wrong formation for Lacie FWSpeaker and Griffin Firewave - remove 'fw_unit_get()/fw_unit_put()' because helper function do this Confirm to work with: snd-fireworks: - Echo Audio AudioFire4 - Echo Audio AudioFire8 (by tester) - Echo Audio AudioFirePre8 - Echo Audio AudioFire12 snd-bebob: - Edirol FA-66 (by tester) - M-Audio Ozonic - M-Audio Firewire Solo - M-Audio Firewire Audiophile - M-Audio Firewire 410 - M-Audio Firewire 1814 (by tester) - PreSonus FIREBOX - PreSonus FP10 (former known as FIREPOD, by tester) - Yamaha GO44 - Yamaha GO46 snd-oxfw: - Behringer F-Control Audio 202 Takashi Sakamoto (52): firewire-lib: Rename functions, structure, member for AMDTP firewire-lib: Add macros instead of fixed value for AMDTP firewire-lib: Add 'direction' member to 'amdtp_stream' structure firewire-lib: Split some codes into functions to reuse for both streams firewire-lib: Add support for AMDTP in-stream and PCM capture firewire-lib: Add support for MIDI capture/playback firewire-lib: Give syt value as parameter to handle_out_packet() firewire-lib: Add support for duplex streams synchronization in blocking mode firewire-lib: Add sort function for transmitted packet firewire-lib: Add transfer delay to synchronized duplex streams firewire-lib: Add support for channel mapping firewire-lib: Rename macros, variables and functions for CMP firewire-lib: Add 'direction' member to 'cmp_connection' structure firewire-lib: Add handling output connection by CMP firewire-lib: Add a new function to check others' connection firewire-lib: Add some AV/C general commands firewire-lib: Add quirks for Fireworks fireworks: Add skelton for Fireworks based devices fireworks: Add transaction and some commands fireworks: Add connection and stream management fireworks: Add proc interface for debugging purpose fireworks: Add MIDI interface fireworks: Add PCM interface fireworks: Add hwdep interface fireworks: Add command/response functionality into hwdep interface bebob: Add skelton for BeBoB based devices bebob: Add commands and connections/streams management bebob: Add proc interface for debugging purpose bebob: Add MIDI interface bebob: Add PCM interface bebob: Add hwdep interface bebob: Prepare for device specific operations bebob: Add support for Terratec PHASE, EWS series and Aureon bebob: Add support for Yamaha GO series bebob: Add support for Focusrite Saffire/SaffirePro series bebob: Add support for M-Audio usual Firewire series bebob: Send a cue to load firmware for M-Audio Firewire series speakers: Rename to oxfw oxfw: Move to its own directory oxfw: Split stream functionality to a new file and add a header file oxfw: Split PCM functionality to a new file oxfw: Split control functionality to a new file oxfw: Change the way to name card oxfw: Change the way to make PCM rules/constraints oxfw: Add proc interface for debugging purpose oxfw: Change the way to start stream oxfw: Add some AV/C commands to get stream formation and supported sample rate oxfw: Add support for Behringer/Mackie devices oxfw: Add support AMDTP in-stream and PCM capture oxfw: Add support for capture/playback MIDI messages oxfw: Add hwdep interface bebob: Add support for M-Audio special Firewire series include/uapi/sound/asound.h | 5 +- include/uapi/sound/firewire.h | 23 +- sound/firewire/Kconfig | 76 +- sound/firewire/Makefile | 5 +- sound/firewire/amdtp.c | 905 ++++++++++++++++------ sound/firewire/amdtp.h | 161 +++- sound/firewire/bebob/Makefile | 4 + sound/firewire/bebob/bebob.c | 453 +++++++++++ sound/firewire/bebob/bebob.h | 262 +++++++ sound/firewire/bebob/bebob_command.c | 354 +++++++++ sound/firewire/bebob/bebob_focusrite.c | 289 +++++++ sound/firewire/bebob/bebob_hwdep.c | 197 +++++ sound/firewire/bebob/bebob_maudio.c | 922 +++++++++++++++++++++++ sound/firewire/bebob/bebob_midi.c | 178 +++++ sound/firewire/bebob/bebob_pcm.c | 419 ++++++++++ sound/firewire/bebob/bebob_proc.c | 185 +++++ sound/firewire/bebob/bebob_stream.c | 873 +++++++++++++++++++++ sound/firewire/bebob/bebob_terratec.c | 68 ++ sound/firewire/bebob/bebob_yamaha.c | 50 ++ sound/firewire/cmp.c | 217 ++++-- sound/firewire/cmp.h | 14 +- sound/firewire/dice.c | 46 +- sound/firewire/fcp.c | 151 ++++ sound/firewire/fcp.h | 21 + sound/firewire/fireworks/Makefile | 4 + sound/firewire/fireworks/fireworks.c | 330 ++++++++ sound/firewire/fireworks/fireworks.h | 223 ++++++ sound/firewire/fireworks/fireworks_command.c | 397 ++++++++++ sound/firewire/fireworks/fireworks_hwdep.c | 299 ++++++++ sound/firewire/fireworks/fireworks_midi.c | 186 +++++ sound/firewire/fireworks/fireworks_pcm.c | 443 +++++++++++ sound/firewire/fireworks/fireworks_proc.c | 206 +++++ sound/firewire/fireworks/fireworks_stream.c | 356 +++++++++ sound/firewire/fireworks/fireworks_transaction.c | 327 ++++++++ sound/firewire/oxfw/Makefile | 3 + sound/firewire/oxfw/oxfw.c | 264 +++++++ sound/firewire/oxfw/oxfw.h | 146 ++++ sound/firewire/oxfw/oxfw_command.c | 162 ++++ sound/firewire/oxfw/oxfw_control.c | 283 +++++++ sound/firewire/oxfw/oxfw_hwdep.c | 195 +++++ sound/firewire/oxfw/oxfw_midi.c | 176 +++++ sound/firewire/oxfw/oxfw_pcm.c | 399 ++++++++++ sound/firewire/oxfw/oxfw_proc.c | 60 ++ sound/firewire/oxfw/oxfw_stream.c | 582 ++++++++++++++ sound/firewire/speakers.c | 830 -------------------- 45 files changed, 10576 insertions(+), 1173 deletions(-) create mode 100644 sound/firewire/bebob/Makefile create mode 100644 sound/firewire/bebob/bebob.c create mode 100644 sound/firewire/bebob/bebob.h create mode 100644 sound/firewire/bebob/bebob_command.c create mode 100644 sound/firewire/bebob/bebob_focusrite.c create mode 100644 sound/firewire/bebob/bebob_hwdep.c create mode 100644 sound/firewire/bebob/bebob_maudio.c create mode 100644 sound/firewire/bebob/bebob_midi.c create mode 100644 sound/firewire/bebob/bebob_pcm.c create mode 100644 sound/firewire/bebob/bebob_proc.c create mode 100644 sound/firewire/bebob/bebob_stream.c create mode 100644 sound/firewire/bebob/bebob_terratec.c create mode 100644 sound/firewire/bebob/bebob_yamaha.c create mode 100644 sound/firewire/fireworks/Makefile create mode 100644 sound/firewire/fireworks/fireworks.c create mode 100644 sound/firewire/fireworks/fireworks.h create mode 100644 sound/firewire/fireworks/fireworks_command.c create mode 100644 sound/firewire/fireworks/fireworks_hwdep.c create mode 100644 sound/firewire/fireworks/fireworks_midi.c create mode 100644 sound/firewire/fireworks/fireworks_pcm.c create mode 100644 sound/firewire/fireworks/fireworks_proc.c create mode 100644 sound/firewire/fireworks/fireworks_stream.c create mode 100644 sound/firewire/fireworks/fireworks_transaction.c 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_control.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 delete mode 100644 sound/firewire/speakers.c -- 1.8.3.2 Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

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[alsa-devel] [PATCH] ASoC: da732x: Mark DC offset control registers volatile
by Mark Brown 26 Feb '14

26 Feb '14

From: Mark Brown <broonie(a)linaro.org> The driver reads from the DC offset control registers during callibration but since the registers are marked as volatile and there is a register cache the values will not be read from the hardware after the first reading rendering the callibration ineffective. It appears that the driver was originally written for the ASoC level register I/O code but converted to regmap prior to merge and this issue was missed during the conversion as the framework level volatile register functionality was not being used. Signed-off-by: Mark Brown <broonie(a)linaro.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/da732x.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 8e2c74d..798eb43 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1268,11 +1268,23 @@ static struct snd_soc_dai_driver da732x_dai[] = { }, }; +static bool da732x_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case DA732X_REG_HPL_DAC_OFF_CNTL: + case DA732X_REG_HPR_DAC_OFF_CNTL: + return true; + default: + return false; + } +} + static const struct regmap_config da732x_regmap = { .reg_bits = 8, .val_bits = 8, .max_register = DA732X_MAX_REG, + .volatile_reg = da732x_volatile, .reg_defaults = da732x_reg_cache, .num_reg_defaults = ARRAY_SIZE(da732x_reg_cache), .cache_type = REGCACHE_RBTREE, -- 1.9.0.rc3

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[alsa-devel] Fwd: i.MX audio support
by jwoodward 26 Feb '14

26 Feb '14

-------- Original Message -------- Subject: i.MX audio support Date: Wed, 26 Feb 2014 09:33:06 +0000 From: jwoodward <jwoodward(a)tensor.co.uk> To: s.hauer(a)pengutronix.de Hi Sascha, I am having trouble implimenting the wm8731 codec with the i.MX21. I have read your article https://lwn.net/Articles/363406/ which does not fill me with confidence on this. It is at the stage of finding the i2c device, and setting up the transmit line for AC97, I am not very experienced with audio and would appreciate any advise possible? Here are a few traces I have added: [ 2.935589] wm8731_i2c_probe: i2c_set_clientdata [ 2.989318] imx_ssi_init [ 2.997084] imx_ssi_probe [ 3.007482] imx_ssi_probe [ 3.015846] setup_channel_to_ac97 [ 3.029115] imx-fiq-pcm-audio imx-fiq-pcm-audio.1: failed to claim fiq: -16 [ 3.049930] imx-fiq-pcm-audio: probe of imx-fiq-pcm-audio.1 failed with error -16 [ 3.075173] ALSA device list: [ 3.084401] No soundcards found. Attached is a schematic of our interface. Many thanks, Jon -- Regards, __| __| __| Jonathan Woodward __| __| __| Embedded Developer __| __| __| Tensor plc jwoodward(a)tensor.co.uk Tel: +44 (0)1480 215530 http://www.tensor.co.uk Fax: +44 (0)1480 223966 ---------------------------------------------------------------------------------- Important Email Information The information in this email is confidential and may be legally privileged. It is intended solely for the addressee. Access to this email by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and unlawful. If you are not the intended addressee please contact the sender and dispose of this email. Any opinions expressed in this e-mail are those of the individual and not necessarily those of Tensor plc or any companies within the Tensor group. Any prices for the supply of goods or services are only valid if supported by a formal written quotation. This communication is for information only and shall not create or change any contractual relationship. Every effort has been made to ensure that any attachment to this mail does not contain a virus. Whilst we have taken every reasonable precaution to minimise this risk, neither it nor the sender can accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checks before opening the attachment. ---------------------------------------------------------------------------------- Tensor plc, Hail Weston House, Hail Weston, St. Neots, Cambs, PE19 5JY, UK Registered in England number 03670909 - VAT Number 761 546 815

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[alsa-devel] I lost one SSID for HP mute led
by Kailang 26 Feb '14

26 Feb '14

Hi Takashi, I lost one SSID. I add it as attach patch. Thanks. BR, Kailang

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[alsa-devel] alsa on non-linux
by Patrick Welche 26 Feb '14

26 Feb '14

I am currently successfully running alsa-lib 1.0.22 with oss plugins on a non-linux box. The advantage is that programs written to use libasound work. I just tried to update alsa-lib, and see that now alsa-lib directly includes headers like linux/types.h, and all protection, such as #if defined(LINUX) || defined(__LINUX__) || defined(__linux__) has been removed. This means that compilation on non-linux is essentially impossible. Is it that there is now a different way of obtaining that alsa lib front end / oss back end layer? Cheers, Patrick

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[alsa-devel] [PATCH 00/10] A few more ALSA and HD-audio core updates
by Takashi Iwai 26 Feb '14

26 Feb '14

Hi, this is a series of patches to cleanup / enhance the ALSA core and HD-audio driver codes. A part of them are the patches I already submitted as RFC shortly ago (patch 4 & patch 5), another part are for better sysfs managment (part 1 & 2). The rest half are about HD-audio, for creating the device struct for each codec, followed by a cleanup patch to replace printk with dev_err() & co. Takashi

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[alsa-devel] CA0132 6 Channel Output
by Sean McCully 26 Feb '14

26 Feb '14

I have one of these sound cards that came with my motherboard, /http://www.asrock.com/microsite/intelx79/GameBlaster.html/[1] Two Channel Output from one audio Jack works perfectly, but there seems to be a problem with using the additional audio jacks for 6 channel "surround sound". http://www.alsa-project.org/db/?f=a8ae0aa3c0da4d697c1df5f403001dff50bb6e51 pasuspender -- speaker-test --channels=6 --device=surround51:0 speaker-test 1.0.27.2 Playback device is surround51:0 Stream parameters are 48000Hz, S16_LE, 6 channels Using 16 octaves of pink noise Channels count (6) not available for playbacks: Invalid argument Setting of hwparams failed: Invalid argument $ aplay -L null Discard all samples (playback) or generate zero samples (capture) oss Open Sound System pulse PulseAudio Sound Server sysdefault:CARD=PCH HDA Intel PCH, CA0132 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Analog Front speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Digital IEC958 (S/PDIF) Digital Audio Output hdmi:CARD=NVidia,DEV=0 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=1 HDA NVidia, HDMI 1 HDMI Audio Output hdmi:CARD=NVidia,DEV=2 HDA NVidia, HDMI 2 HDMI Audio Output -- -- Sean McCully -------- [1] http://www.asrock.com/microsite/intelx79/GameBlaster.html

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[alsa-devel] [PATCH] ASoC: cirrus: Remove excess dependencies on SND_SOC
by Alexander Shiyan 26 Feb '14

26 Feb '14

Configuration for Cirrus Logic audio support is included only if SND_SOC symbol selected, so no reason to check it once more. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/cirrus/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 06f938d..c872dac 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -1,6 +1,6 @@ config SND_EP93XX_SOC tristate "SoC Audio support for the Cirrus Logic EP93xx series" - depends on (ARCH_EP93XX || COMPILE_TEST) && SND_SOC + depends on ARCH_EP93XX || COMPILE_TEST select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to -- 1.8.3.2

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[alsa-devel] [PATCH] ASoC: fsl-sai: fix Freescale SAI DAI format setting.
by Xiubo Li 26 Feb '14

26 Feb '14

o Fix some bugs of fsl_sai_set_dai_fmt_tr(). o Add SND_SOC_DAIFMT_LEFT_J support. o Add SND_SOC_DAIFMT_CBS_CFM support. o Add SND_SOC_DAIFMT_CBM_CFS support. o And SND_SOC_DAIFMT_RIGHT_J need to be done in the future. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 38 +++++++++++++++++++++++++++++--------- 1 file changed, 29 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index faa65af..26d9f5e 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -105,35 +105,47 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, else val_cr4 |= FSL_SAI_CR4_MF; + /* DAI mode */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - val_cr4 |= FSL_SAI_CR4_FSE; + /* Data on rising edge of bclk, frame low, 1clk before data */ + val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr4 |= FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* Data on rising edge of bclk, frame high, 0clk before data */ + val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP); break; + case SND_SOC_DAIFMT_RIGHT_J: + /* To be done */ default: return -EINVAL; } + /* DAI clock inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_IB_IF: - val_cr4 |= FSL_SAI_CR4_FSP; - val_cr2 &= ~FSL_SAI_CR2_BCP; + /* Invert both clocks */ + val_cr2 ^= FSL_SAI_CR2_BCP; + val_cr4 ^= FSL_SAI_CR4_FSP; break; case SND_SOC_DAIFMT_IB_NF: - val_cr4 &= ~FSL_SAI_CR4_FSP; - val_cr2 &= ~FSL_SAI_CR2_BCP; + /* Invert bit clock */ + val_cr2 ^= FSL_SAI_CR2_BCP; break; case SND_SOC_DAIFMT_NB_IF: - val_cr4 |= FSL_SAI_CR4_FSP; - val_cr2 |= FSL_SAI_CR2_BCP; + /* Invert frame clock */ + val_cr4 ^= FSL_SAI_CR4_FSP; break; case SND_SOC_DAIFMT_NB_NF: - val_cr4 &= ~FSL_SAI_CR4_FSP; - val_cr2 |= FSL_SAI_CR2_BCP; + /* Nothing to do for both normal cases */ break; default: return -EINVAL; } + /* DAI clock master masks */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; @@ -143,6 +155,14 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR; val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR; break; + case SND_SOC_DAIFMT_CBS_CFM: + val_cr2 |= FSL_SAI_CR2_BCD_MSTR; + val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR; + val_cr4 |= FSL_SAI_CR4_FSD_MSTR; + break; default: return -EINVAL; } -- 1.8.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: Intel: byt-rt5640: Update internal mic and speaker kcontrol names
by Jarkko Nikula 26 Feb '14

26 Feb '14

Use more sensible kcontrol names than "Int Mic" and "Ext Spk". Speakers especially are usually integrated devices in sales models. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/byt-rt5640.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 038547b48b2d..672fedbfa680 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -29,27 +29,27 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("Internal Mic", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), }; static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"IN2P", NULL, "Headset Mic"}, {"IN2N", NULL, "Headset Mic"}, - {"DMIC1", NULL, "Int Mic"}, + {"DMIC1", NULL, "Internal Mic"}, {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, - {"Ext Spk", NULL, "SPOLP"}, - {"Ext Spk", NULL, "SPOLN"}, - {"Ext Spk", NULL, "SPORP"}, - {"Ext Spk", NULL, "SPORN"}, + {"Speaker", NULL, "SPOLP"}, + {"Speaker", NULL, "SPOLN"}, + {"Speaker", NULL, "SPORP"}, + {"Speaker", NULL, "SPORN"}, }; static const struct snd_kcontrol_new byt_rt5640_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Int Mic"), - SOC_DAPM_PIN_SWITCH("Ext Spk"), + SOC_DAPM_PIN_SWITCH("Internal Mic"), + SOC_DAPM_PIN_SWITCH("Speaker"), }; static int byt_rt5640_hw_params(struct snd_pcm_substream *substream, @@ -113,8 +113,8 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_enable_pin(dapm, "Headset Mic"); snd_soc_dapm_enable_pin(dapm, "Headphone"); - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - snd_soc_dapm_enable_pin(dapm, "Int Mic"); + snd_soc_dapm_enable_pin(dapm, "Speaker"); + snd_soc_dapm_enable_pin(dapm, "Internal Mic"); snd_soc_dapm_sync(dapm); return ret; -- 1.8.5.3

3 3

[alsa-devel] [PATCH] ASoC: Intel: sst-acpi: Fix Oops in case of missing firmware
by Jarkko Nikula 26 Feb '14

26 Feb '14

I swear I tested missing firmware in commit e5161d7987f1 ("ASoC: Intel: sst-acpi: Request firmware before SST platform driver probing"). Unfortunately same wasn't done in commit 6dda27cbbd1d ("ASoC: Intel: sst-acpi: Add support for multiple machine drivers per platform") which will cause NULL pointer dereference in sst_acpi_fw_cb() when printing the error since sst_acpi->mach is not set. Fix this obvious error by setting the sst_acpi->mach in sst_acpi_probe(). Reported-by: Mika Westerberg <mika.westerberg(a)linux.intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst-acpi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index 0bb43169e146..5d06eecb6198 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -139,6 +139,7 @@ static int sst_acpi_probe(struct platform_device *pdev) sst_pdata = &sst_acpi->sst_pdata; sst_pdata->id = desc->sst_id; sst_acpi->desc = desc; + sst_acpi->mach = mach; if (desc->resindex_dma_base >= 0) { sst_pdata->dma_engine = desc->dma_engine; -- 1.8.5.3

4 3

[alsa-devel] [patch] ASoC: Intel: sst-firmware: missing curly braces (harmless)
by Dan Carpenter 26 Feb '14

26 Feb '14

There were some curly braces intended here, but the code actually works the same either way so it's not a bug. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index b6f9b5ecb66a..51b56a473bc5 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -293,7 +293,7 @@ static int block_module_prepare(struct sst_module *module) /* enable each block so that's it'e ready for module P/S data */ list_for_each_entry(block, &module->block_list, module_list) { - if (block->ops && block->ops->enable) + if (block->ops && block->ops->enable) { ret = block->ops->enable(block); if (ret < 0) { dev_err(module->dsp->dev, @@ -301,6 +301,7 @@ static int block_module_prepare(struct sst_module *module) block->type, block->index); goto err; } + } } return ret;

2 1

[alsa-devel] [patch 1/2] ASoC: sst-haswell-ipc: restore IRQs on error
by Dan Carpenter 26 Feb '14

26 Feb '14

This should be spin_unlock_irqrestore() instead of spin_unlock() Fixes: 22981243589c ('ASoC: Intel: Add Haswell/Broadwell IPC') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 668d486520ae..552aebf2c8f5 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -527,7 +527,7 @@ static int ipc_tx_message(struct sst_hsw *hsw, u32 header, void *tx_data, msg = msg_get_empty(hsw); if (msg == NULL) { - spin_unlock(&hsw->dsp->spinlock); + spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); return -EBUSY; }

3 2

[alsa-devel] [PATCH v4] ASoC: add RT286 CODEC driver
by bardliao＠realtek.com 25 Feb '14

25 Feb '14

From: Bard Liao <bardliao(a)realtek.com> This patch adds the ALC286 codec driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Gustaw Lewandowski <gustaw.lewandowski(a)intel.com> --- ALC286 is a dual mode codec, which can run as HD-A or I2S mode. It is controlled by HD-A verb commands via I2C protocol. The following is the I/O difference between ALC286 and general I2S codecs. 1. A HD-A verb command contains three parts, NID, VID, and PID. And an I2S command contains only two parts: address and data. 2. Not only the register address is written, but the read command also includes the entire write command. 3. rt286 uses different registers for read and write the same bits. As a result, standard regmap is difficult to be used on ALC286. We don't request a standard I/O by snd_soc_codec_set_cache_io anymore. Now we have ,reg_write and .reg_read functions for ALC286's I/O. And we don't use cache due to item 3 above. Some dummy registers (address <= 0xff) are defined for dapm routing. Thhe dummy registers are cache only. Due to item 2 above, HD-A verb commands are put into the address part of regmap. When we issue HD-A verb write commands, the data part of regmap is zero. The jack detection function is done by Lewandowski Gustaw. So I add Gustaw's Signed-off-by line in this patch The difference between this version and previous version is listed below. * Use regmap_read/write instead of rt286_read/write to read write codec. * Use the standard jack detection APIs for jack detection. * Power off on bias level STANDBY. * Add .symmetric_rates = 1. * Add gpio2_en and remove irq_en in platform data. * Other minor changes. --- include/sound/rt286.h | 19 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt286.c | 1369 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 159 ++++++ 5 files changed, 1553 insertions(+) create mode 100644 include/sound/rt286.h create mode 100644 sound/soc/codecs/rt286.c create mode 100644 sound/soc/codecs/rt286.h diff --git a/include/sound/rt286.h b/include/sound/rt286.h new file mode 100644 index 0000000..eb773d1 --- /dev/null +++ b/include/sound/rt286.h @@ -0,0 +1,19 @@ +/* + * linux/sound/rt286.h -- Platform data for RT286 + * + * Copyright 2013 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT286_H +#define __LINUX_SND_RT286_H + +struct rt286_platform_data { + bool cbj_en; /*combo jack enable*/ + bool gpio2_en; /*GPIO2 enable*/ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f2383eb..889b69f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -67,6 +67,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 select SND_SOC_PCM512x if SND_SOC_I2C_AND_SPI + select SND_SOC_RT286 if I2C select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C @@ -365,6 +366,9 @@ config SND_SOC_PCM512x tristate "Texas Instruments PCM512x CODECs" select REGMAP +config SND_SOC_RT286 + tristate + config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6af7a55..2c42f88 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -54,6 +54,7 @@ snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-pcm512x-objs := pcm512x.o +snd-soc-rt286-objs := rt286.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -197,6 +198,7 @@ obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o +obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c new file mode 100644 index 0000000..715aee1 --- /dev/null +++ b/sound/soc/codecs/rt286.c @@ -0,0 +1,1369 @@ +/* + * rt286.c -- RT286 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <sound/rt286.h> + +#include "../../pci/hda/hda_codec.h" +#include "rt286.h" + +struct rt286_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct rt286_platform_data pdata; + struct snd_soc_jack *jack; + struct i2c_client *i2c; + int sys_clk; +}; + +static unsigned int rt286_reg_cache[] = { + [RT286_AUDIO_FUNCTION_GROUP] = 0x0000, + [RT286_DAC_OUT1] = 0x7f7f, + [RT286_DAC_OUT2] = 0x7f7f, + [RT286_SPDIF] = 0x0000, + [RT286_ADC_IN1] = 0x4343, + [RT286_ADC_IN2] = 0x4343, + [RT286_MIC1] = 0x0000, + [RT286_MIXER_IN] = 0x000b, + [RT286_MIXER_OUT1] = 0x0002, + [RT286_MIXER_OUT2] = 0x0000, + [RT286_DMIC1] = 0x0000, + [RT286_DMIC2] = 0x0000, + [RT286_LINE1] = 0x0000, + [RT286_BEEP] = 0x0000, + [RT286_VENDOR_REGISTERS] = 0x0000, + [RT286_SPK_OUT] = 0x8080, + [RT286_HP_OUT] = 0x8080, + [RT286_MIXER_IN1] = 0x0000, + [RT286_MIXER_IN2] = 0x0000, +}; + +static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + int ret; + unsigned int buf = 0x0; + + if (reg <= 0xff) { /*read cache*/ + *value = rt286_reg_cache[reg]; + return 0; + } + reg = cpu_to_be32(reg); + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 4; + xfer[0].buf = (u8 *)® + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 4; + xfer[1].buf = (u8 *)&buf; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret < 0) + return ret; + else if (ret != 2) + return -EIO; + + *value = be32_to_cpu(buf); + + return 0; +} + +static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) +{ + struct i2c_client *client = context; + u8 data[4]; + int ret; + + if (reg <= 0xff) { /*write cache*/ + rt286_reg_cache[reg] = value; + return 0; + } + data[0] = (reg >> 24) & 0xff; + data[1] = (reg >> 16) & 0xff; + data[2] = (reg >> 8) & 0xff; + data[3] = reg & 0xff; + + ret = i2c_master_send(client, data, 4); + if (ret == 4) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +static int rt286_update_bits(struct snd_soc_codec *codec, unsigned int vid, + unsigned int nid, unsigned int data, + unsigned int mask, unsigned int value) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int old, new, verb; + int change, ret; + + verb = VERB_CMD((vid | 0x800), nid, data); + regmap_read(rt286->regmap, verb, &old); + new = (old & ~mask) | (value & mask); + change = old != new; + + if (change) { + ret = regmap_write(rt286->regmap, verb, 0); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write private reg: %d\n", ret); + goto err; + } + } + return change; + +err: + return ret; +} + +static int rt286_index_write(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index, unsigned int data) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int ret; + unsigned int verb; + + verb = VERB_CMD(AC_VERB_SET_COEF_INDEX, wid, index); + ret = regmap_write(rt286->regmap, verb, 0); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + goto err; + } + verb = VERB_CMD(AC_VERB_SET_PROC_COEF, wid, data); + ret = regmap_write(rt286->regmap, verb, 0); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + + return 0; +err: + return ret; +} + +static unsigned int rt286_index_read(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int ret; + unsigned int verb, val; + + verb = VERB_CMD(AC_VERB_SET_COEF_INDEX, wid, index); + ret = regmap_write(rt286->regmap, verb, 0); + + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + return ret; + } + + verb = VERB_CMD(AC_VERB_GET_PROC_COEF, wid, index); + regmap_read(rt286->regmap, verb, &val); + return val; +} + +static int rt286_index_update_bits(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index, + unsigned int mask, unsigned int data) +{ + unsigned int old, new; + int change, ret; + + old = rt286_index_read(codec, wid, index); + new = (old & ~mask) | (data & mask); + change = old != new; + + if (change) { + ret = rt286_index_write(codec, wid, index, new); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write private reg: %d\n", ret); + goto err; + } + } + return change; + +err: + return ret; +} + +static int rt286_support_power_controls[] = { + RT286_DAC_OUT1, + RT286_DAC_OUT2, + RT286_ADC_IN1, + RT286_ADC_IN2, + RT286_MIC1, + RT286_DMIC1, + RT286_DMIC2, + RT286_SPK_OUT, + RT286_HP_OUT, +}; +#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls) + +static const struct reg_default rt286_reg[] = { + { RT286_DAC_OUT1, 0x7f7f }, + { RT286_DAC_OUT2, 0x7f7f }, + { RT286_SPDIF, 0x0000 }, + { RT286_ADC_IN1, 0x4343 }, + { RT286_ADC_IN2, 0x4343 }, + { RT286_MIC1, 0x0000 }, + { RT286_MIXER_IN, 0x000b }, + { RT286_MIXER_OUT1, 0x0002 }, + { RT286_MIXER_OUT2, 0x0000 }, + { RT286_SPK_OUT, 0x0000 }, + { RT286_HP_OUT, 0x0000 }, + { RT286_MIXER_IN1, 0x0005 }, + { RT286_MIXER_IN2, 0x0005 }, +}; + +static bool rt286_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT286_DAC_OUT1: + case RT286_DAC_OUT2: + case RT286_ADC_IN1: + case RT286_ADC_IN2: + case RT286_MIC1: + case RT286_SPDIF: + case RT286_MIXER_IN: + case RT286_MIXER_OUT1: + case RT286_MIXER_OUT2: + case RT286_DMIC1: + case RT286_DMIC2: + case RT286_SPK_OUT: + case RT286_HP_OUT: + case RT286_MIXER_IN1: + case RT286_MIXER_IN2: + return true; + default: + return false; + } +} + +static int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val, buf; + int i; + + *hp = false; + *mic = false; + + if (rt286->pdata.cbj_en) { + regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf); + + *hp = buf & 0x80000000; + if (*hp) { + /* power on HV,VERF */ + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_POWER_CTRL1, 0x1001, 0x0); + /* power LDO1 */ + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_POWER_CTRL2, 0x4, 0x4); + regmap_write(rt286->regmap, RT286_SET_MIC1_24, 0); + val = rt286_index_read(codec, + RT286_VENDOR_REGISTERS, RT286_CBJ_CTRL2); + + msleep(200); + i = 40; + while (((val & 0x0800) == 0) && (i > 0)) { + val = rt286_index_read(codec, + RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL2); + i--; + msleep(20); + } + + if (0x0400 == (val & 0x0700)) { + *mic = false; + + regmap_write(rt286->regmap, + RT286_SET_MIC1_20, 0); + /* power off HV,VERF */ + rt286_index_update_bits(codec, + RT286_VENDOR_REGISTERS, + RT286_POWER_CTRL1, 0x1001, 0x1001); + rt286_index_update_bits(codec, + RT286_VENDOR_REGISTERS, + RT286_A_BIAS_CTRL3, 0xc000, 0x0000); + rt286_index_update_bits(codec, + RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL1, 0x0030, 0x0000); + rt286_index_update_bits(codec, + RT286_VENDOR_REGISTERS, + RT286_A_BIAS_CTRL2, 0xc000, 0x0000); + } else if ((0x0200 == (val & 0x0700)) || + (0x0100 == (val & 0x0700))) { + *mic = true; + } else { + *mic = false; + } + + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_MISC_CTRL1, + 0x0060, 0x0000); + } else { + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_MISC_CTRL1, + 0x0060, 0x0020); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_A_BIAS_CTRL3, + 0xc000, 0x8000); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL1, + 0x0030, 0x0020); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_A_BIAS_CTRL2, + 0xc000, 0x8000); + + *mic = false; + } + } else { + regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf); + *hp = buf & 0x80000000; + regmap_read(rt286->regmap, RT286_GET_MIC1_SENSE, &buf); + *mic = buf & 0x80000000; + } + + + /* Clear IRQ */ + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_IRQ_CTRL, 0x1, 0x1); + return 0; +} + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + rt286->jack = jack; + + /* Send an initial empty report */ + snd_soc_jack_report(rt286->jack, 0, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + return 0; +} +EXPORT_SYMBOL_GPL(rt286_mic_detect); + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static int rt286_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb, buf, io = 0x8000; + + if (RT286_ADC_IN1 == mc->reg) + io = 0; + verb = VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, mc->reg, (0x2000 | io)); + regmap_read(rt286->regmap, verb, &buf); + ucontrol->value.integer.value[0] = buf & 0x7f; + verb = VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, mc->reg, io); + regmap_read(rt286->regmap, verb, &buf); + ucontrol->value.integer.value[1] = buf & 0x7f; + + return 0; +} + +static int rt286_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb, vall, valr, io; + + if (RT286_ADC_IN1 == mc->reg) + io = 0x4000; + else + io = 0x8000; + vall = ucontrol->value.integer.value[0]; + valr = ucontrol->value.integer.value[1]; + if (vall == valr) { + vall = vall | 0x3000 | io; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + regmap_write(rt286->regmap, verb, 0); + } else { + vall = vall | 0x2000 | io; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + regmap_write(rt286->regmap, verb, 0); + valr = valr | 0x1000 | io; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, valr); + regmap_write(rt286->regmap, verb, 0); + } + + return 0; +} + +static int rt286_mic_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb, buf; + + verb = VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, mc->reg, 0x0); + regmap_read(rt286->regmap, verb, &buf); + ucontrol->value.integer.value[0] = buf & 0x3; + + return 0; +} + +static int rt286_mic_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb, val; + + val = ucontrol->value.integer.value[0]; + val = val | 0x7000; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, val); + regmap_write(rt286->regmap, verb, 0); + + return 0; +} + +static int rt286_playback_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb, buf; + + verb = VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, mc->reg, 0xa000); + regmap_read(rt286->regmap, verb, &buf); + ucontrol->value.integer.value[0] = !(buf & 0x80); + verb = VERB_CMD(AC_VERB_GET_AMP_GAIN_MUTE, mc->reg, 0x8000); + regmap_read(rt286->regmap, verb, &buf); + ucontrol->value.integer.value[1] = !(buf & 0x80); + + return 0; +} + +static int rt286_playback_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb, vall, valr, val = 0; + + vall = (!ucontrol->value.integer.value[0] << 7); + valr = (!ucontrol->value.integer.value[1] << 7); + if (vall) + val |= 0x8000; + if (valr) + val |= 0x0080; + + if (vall == valr) { + vall = vall | 0xb000; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + regmap_write(rt286->regmap, verb, 0); + } else { + vall = vall | 0xa000; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + regmap_write(rt286->regmap, verb, 0); + valr = valr | 0x9000; + verb = VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, valr); + regmap_write(rt286->regmap, verb, 0); + } + snd_soc_update_bits(codec, mc->reg, 0x8080, val); + + return 0; +} + +static const struct snd_kcontrol_new rt286_snd_controls[] = { + SOC_DOUBLE_EXT_TLV("DAC0 Playback Volume", RT286_DAC_OUT1, + 8, 0, 0x7f, 0, + rt286_vol_get, rt286_vol_put, + out_vol_tlv), + SOC_DOUBLE_EXT_TLV("ADC0 Capture Volume", RT286_ADC_IN1, + 8, 0, 0x7f, 0, + rt286_vol_get, rt286_vol_put, + out_vol_tlv), + SOC_SINGLE_EXT_TLV("AMIC Volume", RT286_MIC1, 0, 0x3, 0, + rt286_mic_gain_get, rt286_mic_gain_put, + mic_vol_tlv), + SOC_DOUBLE_EXT("Headphone Playback Switch", RT286_HP_OUT, + 15, 8, 1, 1, rt286_playback_switch_get, + rt286_playback_switch_put), + SOC_DOUBLE_EXT("Speaker Playback Switch", RT286_SPK_OUT, + 15, 8, 1, 1, rt286_playback_switch_get, + rt286_playback_switch_put), +}; + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt286_front_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", RT286_MIXER_OUT1, + RT286_M_FRONT_DAC_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIX Switch", RT286_MIXER_OUT1, + RT286_M_FRONT_REC_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt286_rec_mix[] = { + SOC_DAPM_SINGLE("Beep Switch", RT286_MIXER_IN, + RT286_M_REC_BEEP_SFT, 1, 1), + SOC_DAPM_SINGLE("Line1 Switch", RT286_MIXER_IN, + RT286_M_REC_LINE1_SFT, 1, 1), + SOC_DAPM_SINGLE("Mic1 Switch", RT286_MIXER_IN, + RT286_M_REC_MIC1_SFT, 1, 1), + SOC_DAPM_SINGLE("I2S Switch", RT286_MIXER_IN, + RT286_M_REC_I2S_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spo_enable_control = + SOC_DAPM_SINGLE("Switch", RT286_SPK_OUT, + RT286_M_SPK_MUX_SFT, 1, 1); + +static const struct snd_kcontrol_new hpo_enable_control = + SOC_DAPM_SINGLE("Switch", RT286_HP_OUT, + RT286_M_HP_MUX_SFT, 1, 1); + +/* ADC0 source */ +static const char * const rt286_adc_src[] = { + "Mic", "Dmic" +}; + +static int rt286_adc_values[] = { + 0, 5, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc0_enum, RT286_MIXER_IN1, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc0_mux = + SOC_DAPM_VALUE_ENUM("ADC 0 source", rt286_adc0_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc1_enum, RT286_MIXER_IN2, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc1_mux = + SOC_DAPM_VALUE_ENUM("ADC 1 source", rt286_adc1_enum); + +/* HP-OUT source */ +static const char * const rt286_hpo_src[] = { + "Front", "Surround" +}; + +static SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, RT286_HP_OUT, + RT286_HP_SEL_SFT, rt286_hpo_src); + +static const struct snd_kcontrol_new rt286_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt286_hpo_enum); + +/* SPK-OUT source */ +static const char * const rt286_spo_src[] = { + "Front", "Surround" +}; + +static SOC_ENUM_SINGLE_DECL(rt286_spo_enum, RT286_SPK_OUT, + RT286_SPK_SEL_SFT, rt286_spo_src); + +static const struct snd_kcontrol_new rt286_spo_mux = +SOC_DAPM_ENUM("SPO source", rt286_spo_enum); + +/* SPDIF source */ +static const char * const rt286_spdif_src[] = { + "PCM-IN 0", "PCM-IN 1", "SP-OUT", "PP" +}; + +static SOC_ENUM_SINGLE_DECL(rt286_spdif_enum, RT286_SPDIF, + RT286_SPDIF_SEL_SFT, rt286_spdif_src); + +static const struct snd_kcontrol_new rt286_spdif_mux = +SOC_DAPM_ENUM("SPDIF source", rt286_spdif_enum); + + +static int rt286_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt286->regmap, RT286_SET_SPK_D0, 0); + regmap_write(rt286->regmap, RT286_ENABLE_SPK, 0); + regmap_write(rt286->regmap, RT286_SPK_EAPD_HIGH, 0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt286->regmap, RT286_SPK_EAPD_LOW, 0); + regmap_write(rt286->regmap, RT286_DISABLE_SPK, 0); + regmap_write(rt286->regmap, RT286_SET_SPK_D3, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val, buf; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_read(rt286->regmap, RT286_HP_OUT, &buf); + val = buf & 0x8080; + switch (val) { + case 0x0: + regmap_write(rt286->regmap, RT286_UNMUTE_HPO, 0); + break; + case 0x8000: + regmap_write(rt286->regmap, RT286_UNMUTE_HPO_R, 0); + break; + case 0x0080: + regmap_write(rt286->regmap, RT286_UNMUTE_HPO_L, 0); + break; + } + if (val != 0x8080) + mdelay(100); + break; + + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt286->regmap, RT286_MUTE_HPO, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_set_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt286->regmap, RT286_SET_D0(w->reg), 0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt286->regmap, RT286_SET_D3(w->reg), 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt286->regmap, RT286_SET_DMIC1_D0, 0); + regmap_write(rt286->regmap, RT286_ENABLE_DMIC1, 0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt286->regmap, RT286_DISABLE_DMIC1, 0); + regmap_write(rt286->regmap, RT286_SET_DMIC1_D3, 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_hp_pow_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt286->regmap, RT286_SET_HPO_D0, 0); + regmap_write(rt286->regmap, RT286_ENABLE_HPO, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt286->regmap, RT286_DISABLE_HPO, 0); + regmap_write(rt286->regmap, RT286_SET_HPO_D3, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e; + unsigned int verb, val, buf; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (rt286->pdata.cbj_en) { + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_A_BIAS_CTRL3, 0xc000, 0x8000); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL1, 0x0030, 0x0020); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_A_BIAS_CTRL2, 0xc000, 0x8000); + } + regmap_write(rt286->regmap, RT286_SET_D0(w->reg), 0); + rt286_update_bits(codec, AC_VERB_SET_AMP_GAIN_MUTE, + w->reg, 0, 0x7080, 0x7000); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_update_bits(codec, AC_VERB_SET_AMP_GAIN_MUTE, + w->reg, 0, 0x7080, 0x7080); + regmap_write(rt286->regmap, RT286_SET_D3(w->reg), 0); + break; + case SND_SOC_DAPM_POST_REG: + e = (struct soc_enum *)kcontrol->private_value; + regmap_read(rt286->regmap, e->reg, &buf); + val = buf & RT286_ADC_SEL_MASK; + verb = VERB_CMD(AC_VERB_SET_CONNECT_SEL, e->reg, val); + regmap_write(rt286->regmap, verb, 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_out_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int verb, val, buf; + + switch (event) { + case SND_SOC_DAPM_POST_REG: + regmap_read(rt286->regmap, e->reg, &buf); + val = buf & 0x1; + verb = VERB_CMD(AC_VERB_SET_CONNECT_SEL, e->reg, val); + regmap_write(rt286->regmap, verb, 0); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("Beep"), + + /* DMIC */ + SND_SOC_DAPM_PGA_E("DMIC1", RT286_DMIC1, 15, 0, + NULL, 0, rt286_set_dmic1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("DMIC2", RT286_DMIC2, 15, 0, + NULL, 0, rt286_set_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0, + rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_ADC_IN1, 15, 0, + &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_ADC_IN2, 15, 0, + &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_REG), + + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + + /* Output Side */ + + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC 0", NULL, RT286_DAC_OUT1, + 15, 0, rt286_set_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("DAC 1", NULL, RT286_DAC_OUT2, + 15, 0, rt286_set_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Output Mux */ + SND_SOC_DAPM_MUX_E("SPK Mux", SND_SOC_NOPM, 0, 0, + &rt286_spo_mux, + rt286_out_mux_event, + SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_MUX_E("HPO Mux", SND_SOC_NOPM, 0, 0, + &rt286_hpo_mux, + rt286_out_mux_event, + SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_SUPPLY("HP Power", SND_SOC_NOPM, + 0, 0, rt286_hp_pow_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_MUX("SPDIF Mux", SND_SOC_NOPM, 0, 0, + &rt286_spdif_mux), + + /* Output Mixer */ + SND_SOC_DAPM_MIXER("Front", SND_SOC_NOPM, 0, 0, + rt286_front_mix, ARRAY_SIZE(rt286_front_mix)), + SND_SOC_DAPM_PGA("Surround", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_SWITCH_E("SPO", SND_SOC_NOPM, 0, 0, + &spo_enable_control, rt286_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH_E("HPO", SND_SOC_NOPM, 0, 0, + &hpo_enable_control, rt286_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), +}; + +static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + {"DMIC1", NULL, "DMIC Receiver"}, + {"DMIC2", NULL, "DMIC Receiver"}, + + {"RECMIX", "Beep Switch", "Beep"}, + {"RECMIX", "Line1 Switch", "LINE1"}, + {"RECMIX", "Mic1 Switch", "MIC1"}, + + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "Mic", "MIC1"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "Mic", "MIC1"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TX", NULL, "ADC 0"}, + {"AIF2TX", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RX"}, + {"DAC 1", NULL, "AIF2RX"}, + + {"Front", "DAC Switch", "DAC 0"}, + {"Front", "RECMIX Switch", "RECMIX"}, + + {"Surround", NULL, "DAC 1"}, + + {"SPK Mux", "Front", "Front"}, + {"SPK Mux", "Surround", "Surround"}, + + {"HPO Mux", "Front", "Front"}, + {"HPO Mux", "Surround", "Surround"}, + + {"SPO", "Switch", "SPK Mux"}, + {"HPO", "Switch", "HPO Mux"}, + {"HPO", NULL, "HP Power"}, + + {"SPOL", NULL, "SPO"}, + {"SPOR", NULL, "SPO"}, + {"HPO Pin", NULL, "HPO"}, +}; + +static int rt286_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + val |= 0x4000; + break; + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt286->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + d_len_code = 0; + switch (params_format(params)) { + /* bit 6:4 Bits per Sample */ + case SNDRV_PCM_FORMAT_S16_LE: + d_len_code = 0; + val |= (0x1 << 4); + break; + case SNDRV_PCM_FORMAT_S32_LE: + d_len_code = 2; + val |= (0x4 << 4); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + d_len_code = 1; + val |= (0x2 << 4); + break; + case SNDRV_PCM_FORMAT_S24_LE: + d_len_code = 2; + val |= (0x3 << 4); + break; + case SNDRV_PCM_FORMAT_S8: + break; + default: + return -EINVAL; + } + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x0018, d_len_code << 3); + dev_dbg(codec->dev, "format val = 0x%x\n", val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + RT286_DAC_OUT1, 0, 0x407f, val); + else + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + RT286_ADC_IN1, 0, 0x407f, val); + + return 0; +} + +static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x800, 0x800); + break; + case SND_SOC_DAIFMT_CBS_CFS: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x800, 0x0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x300, 0x0); + break; + case SND_SOC_DAIFMT_LEFT_J: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x300, 0x1 << 8); + break; + case SND_SOC_DAIFMT_DSP_A: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x300, 0x2 << 8); + break; + case SND_SOC_DAIFMT_DSP_B: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x300, 0x3 << 8); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + RT286_DAC_OUT1, 0, 0x8000, 0x0); + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + RT286_ADC_IN1, 0, 0x8000, 0x0); + + return 0; +} + +static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq); + + if (RT286_SCLK_S_MCLK == clk_id) { + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL2, 0x0100, 0x0); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_PLL_CTRL1, 0x20, 0x20); + } else { + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL2, 0x0100, 0x0100); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_PLL_CTRL, 0x4, 0x4); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_PLL_CTRL1, 0x20, 0x0); + } + + switch (freq) { + case 19200000: + case 24000000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + break; + case 12288000: + case 11289600: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL2, 0x8, 0x0); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_CLK_DIV, 0xfc1e, 0x0004); + break; + case 24576000: + case 22579200: + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL2, 0x8, 0x8); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_CLK_DIV, 0xfc1e, 0x5406); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt286->sys_clk = freq; + + return 0; +} + +static int rt286_set_dai_dfs(struct snd_soc_dai *dai, unsigned int fs) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s fs=%d\n", __func__, fs); + if (50 == fs) + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x1000, 0x1000); + else + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_I2S_CTRL1, 0x1000, 0x0); + + + return 0; +} + +static int rt286_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) + regmap_write(rt286->regmap, RT286_SET_AUDIO_D0, 0); + break; + + case SND_SOC_BIAS_STANDBY: + regmap_write(rt286->regmap, RT286_SET_AUDIO_D3, 0); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static irqreturn_t rt286_irq(int irq, void *data) +{ + struct rt286_priv *rt286 = data; + bool hp = false; + bool mic = false; + + rt286_jack_detect(rt286->codec, &hp, &mic); + + int status = 0; + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt286->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + pm_wakeup_event(&rt286->i2c->dev, 300); + + return IRQ_HANDLED; +} + +static int rt286_probe(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int i, ret; + + regmap_write(rt286->regmap, RT286_SET_AUDIO_D3, 0); + for (i = 0; i < RT286_POWER_REG_LEN; i++) { + regmap_write(rt286->regmap, + RT286_SET_D3(rt286_support_power_controls[i]), 0); + } + + if (!rt286->pdata.cbj_en) { + rt286_index_write(codec, RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL2, 0x0000); + rt286_index_write(codec, RT286_VENDOR_REGISTERS, + RT286_MIC1_DET_CTRL, 0x0816); + rt286_index_write(codec, RT286_VENDOR_REGISTERS, + RT286_MISC_CTRL1, 0x0000); + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL1, 0xf000, 0xb000); + } else { + rt286_index_update_bits(codec, RT286_VENDOR_REGISTERS, + RT286_CBJ_CTRL1, 0xf000, 0x5000); + } + + mdelay(10); + + if (!rt286->pdata.gpio2_en) + regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT_4000, 0); + else + regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT_0, 0); + + mdelay(10); + + /*Power down LDO2*/ + rt286_index_update_bits(codec, + RT286_VENDOR_REGISTERS, RT286_POWER_CTRL2, 0x8, 0x0); + + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + rt286->codec = codec; + + if (rt286->i2c->irq) { + rt286_index_update_bits(codec, + RT286_VENDOR_REGISTERS, RT286_IRQ_CTRL, 0x2, 0x2); + ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); + if (ret != 0) { + dev_err(codec->dev, + "Failed to reguest IRQ: %d\n", ret); + return ret; + } + } + + return 0; +} + +#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT286_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 rt286_aif_dai_ops = { + .hw_params = rt286_hw_params, + .set_fmt = rt286_set_dai_fmt, + .set_sysclk = rt286_set_dai_sysclk, + .set_dfs = rt286_set_dai_dfs, +}; + +struct snd_soc_dai_driver rt286_dai[] = { + { + .name = "rt286-aif1", + .id = RT286_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + .symmetric_rates = 1, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt286 = { + .probe = rt286_probe, + .set_bias_level = rt286_set_bias_level, + .idle_bias_off = true, + .controls = rt286_snd_controls, + .num_controls = ARRAY_SIZE(rt286_snd_controls), + .dapm_widgets = rt286_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets), + .dapm_routes = rt286_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes), +}; + +static const struct regmap_config rt286_regmap = { + .reg_bits = 32, + .val_bits = 32, + .reg_write = rt286_hw_write, + .reg_read = rt286_hw_read, + .readable_reg = rt286_readable_register, + .cache_type = REGCACHE_NONE, +}; + +static const struct i2c_device_id rt286_i2c_id[] = { + {"rt286", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); + +static struct acpi_device_id rt286_acpi_match[] = { + { "INT33CA", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); + +static int rt286_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt286_priv *rt286; + struct device *dev = &i2c->dev; + int ret; + + rt286 = devm_kzalloc(&i2c->dev, + sizeof(struct rt286_priv), + GFP_KERNEL); + if (NULL == rt286) + return -ENOMEM; + + rt286->regmap = devm_regmap_init(dev, NULL, i2c, &rt286_regmap); + if (IS_ERR(rt286->regmap)) { + ret = PTR_ERR(rt286->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + rt286->i2c = i2c; + i2c_set_clientdata(i2c, rt286); + + if (pdata) + rt286->pdata = *pdata; + + ret = devm_snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286, + rt286_dai, ARRAY_SIZE(rt286_dai)); + + return ret; +} + +struct i2c_driver rt286_i2c_driver = { + .driver = { + .name = "rt286", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt286_acpi_match), + }, + .probe = rt286_i2c_probe, + .id_table = rt286_i2c_id, +}; + +module_i2c_driver(rt286_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT286 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h new file mode 100644 index 0000000..5f9dc99 --- /dev/null +++ b/sound/soc/codecs/rt286.h @@ -0,0 +1,159 @@ +/* + * rt286.h -- RT286 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 __RT286_H__ +#define __RT286_H__ + +#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) + +#define RT286_AUDIO_FUNCTION_GROUP 0x01 +#define RT286_DAC_OUT1 0x02 +#define RT286_DAC_OUT2 0x03 +#define RT286_ADC_IN1 0x09 +#define RT286_ADC_IN2 0x08 +#define RT286_MIXER_IN 0x0b +#define RT286_MIXER_OUT1 0x0c +#define RT286_MIXER_OUT2 0x0d +#define RT286_DMIC1 0x12 +#define RT286_DMIC2 0x13 +#define RT286_SPK_OUT 0x14 +#define RT286_MIC1 0x18 +#define RT286_LINE1 0x1a +#define RT286_BEEP 0x1d +#define RT286_SPDIF 0x1e +#define RT286_VENDOR_REGISTERS 0x20 +#define RT286_HP_OUT 0x21 +#define RT286_MIXER_IN1 0x22 +#define RT286_MIXER_IN2 0x23 + +#define RT286_CONNECTION_INDEX_MIC1 0X0 +#define RT286_CONNECTION_INDEX_DMIC 0X5 + +/* Verb commands */ +#define RT286_SET_D0(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, AC_PWRST_D0) +#define RT286_SET_D3(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, AC_PWRST_D3) + +#define RT286_SET_MIC1_24\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_MIC1, 0x24) +#define RT286_SET_MIC1_20\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_MIC1, 0x20) +#define RT286_SET_AUDIO_D0 RT286_SET_D0(RT286_AUDIO_FUNCTION_GROUP) +#define RT286_SET_AUDIO_D3 RT286_SET_D3(RT286_AUDIO_FUNCTION_GROUP) +#define RT286_SET_HPO_D0 RT286_SET_D0(RT286_HP_OUT) +#define RT286_SET_HPO_D3 RT286_SET_D3(RT286_HP_OUT) +#define RT286_SET_SPK_D0 RT286_SET_D0(RT286_SPK_OUT) +#define RT286_SET_SPK_D3 RT286_SET_D3(RT286_SPK_OUT) +#define RT286_SET_DMIC1_D0 RT286_SET_D0(RT286_DMIC1) +#define RT286_SET_DMIC1_D3 RT286_SET_D3(RT286_DMIC1) +#define RT286_ENABLE_HPO\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_HP_OUT, 0x40) +#define RT286_DISABLE_HPO\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_HP_OUT, 0) +#define RT286_ENABLE_SPK\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_SPK_OUT, 0x40) +#define RT286_DISABLE_SPK\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_SPK_OUT, 0) +#define RT286_ENABLE_DMIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_DMIC1, 0x20) +#define RT286_DISABLE_DMIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_DMIC1, 0) +#define RT286_SPK_EAPD_HIGH\ + VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT286_SPK_OUT, 0x2) +#define RT286_SPK_EAPD_LOW\ + VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT286_SPK_OUT, 0) +#define RT286_UNMUTE_HPO\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xb000) +#define RT286_UNMUTE_HPO_R\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0x9000) +#define RT286_UNMUTE_HPO_L\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xa000) +#define RT286_MUTE_HPO\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xb080) +#define RT286_GET_HP_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_HP_OUT, 0) +#define RT286_GET_MIC1_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_MIC1, 0) +#define RT286_SET_DMIC2_DEFAULT_4000\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT286_DMIC2, 0x4000) +#define RT286_SET_DMIC2_DEFAULT_0\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT286_DMIC2, 0) + +/* Index registers */ +#define RT286_A_BIAS_CTRL1 0x01 +#define RT286_A_BIAS_CTRL2 0x02 +#define RT286_POWER_CTRL1 0x03 +#define RT286_A_BIAS_CTRL3 0x04 +#define RT286_POWER_CTRL2 0x08 +#define RT286_I2S_CTRL1 0x09 +#define RT286_I2S_CTRL2 0x0a +#define RT286_CLK_DIV 0x0b +#define RT286_POWER_CTRL3 0x0f +#define RT286_MIC1_DET_CTRL 0x19 +#define RT286_MISC_CTRL1 0x20 +#define RT286_IRQ_CTRL 0x33 +#define RT286_PLL_CTRL1 0x49 +#define RT286_CBJ_CTRL1 0x4f +#define RT286_CBJ_CTRL2 0x50 +#define RT286_PLL_CTRL 0x63 + +/* SPDIF (0x06) */ +#define RT286_SPDIF_SEL_SFT 0 +#define RT286_SPDIF_SEL_PCM0 0 +#define RT286_SPDIF_SEL_PCM1 1 +#define RT286_SPDIF_SEL_SPOUT 2 +#define RT286_SPDIF_SEL_PP 3 + +/* RECMIX (0x0b) */ +#define RT286_M_REC_BEEP_SFT 0 +#define RT286_M_REC_LINE1_SFT 1 +#define RT286_M_REC_MIC1_SFT 2 +#define RT286_M_REC_I2S_SFT 3 + +/* Front (0x0c) */ +#define RT286_M_FRONT_DAC_SFT 0 +#define RT286_M_FRONT_REC_SFT 1 + +/* SPK-OUT (0x14) */ +#define RT286_M_SPK_MUX_SFT 14 +#define RT286_SPK_SEL_MASK 0x1 +#define RT286_SPK_SEL_SFT 0 +#define RT286_SPK_SEL_F 0 +#define RT286_SPK_SEL_S 1 + +/* HP-OUT (0x21) */ +#define RT286_M_HP_MUX_SFT 14 +#define RT286_HP_SEL_MASK 0x1 +#define RT286_HP_SEL_SFT 0 +#define RT286_HP_SEL_F 0 +#define RT286_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT286_ADC_SEL_MASK 0x7 +#define RT286_ADC_SEL_SFT 0 +#define RT286_ADC_SEL_SURR 0 +#define RT286_ADC_SEL_FRONT 1 +#define RT286_ADC_SEL_DMIC 2 +#define RT286_ADC_SEL_BEEP 4 +#define RT286_ADC_SEL_LINE1 5 +#define RT286_ADC_SEL_I2S 6 +#define RT286_ADC_SEL_MIC1 7 + +#define RT286_SCLK_S_MCLK 0 +#define RT286_SCLK_S_PLL 1 + +enum { + RT286_AIF1, + RT286_AIFS, +}; + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + +#endif /* __RT286_H__ */ -- 1.8.1.1.439.g50a6b54

4 3

[alsa-devel] [PATCH 0/4] ASoC: simple-card: DT fix and multi DAI links extension
by Jean-Francois Moine 25 Feb '14

25 Feb '14

This patch series fixes a small problem about node device locks and also extends the simple card driver to handle many DAI links as this exists in the Cubox audio subsystem. Jean-Francois Moine (4): ASoC: simple-card: Fix device node locks ASoC: simple-card: dynamically allocate the DAI link array ASoC: simple-card: accept many DAI links ASoC: simple-card: add DT documentation for multi-DAI links .../devicetree/bindings/sound/simple-card.txt | 34 ++++- sound/soc/generic/simple-card.c | 148 +++++++++++++++------ 2 files changed, 137 insertions(+), 45 deletions(-) -- 1.9.0

3 9

[alsa-devel] [PATCH 1/2] ASoC: S3C: Remove invalid file reference
by Sachin Kamat 25 Feb '14

25 Feb '14

Remove file references rendered invalid due to relocation. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- include/linux/platform_data/asoc-s3c.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/include/linux/platform_data/asoc-s3c.h b/include/linux/platform_data/asoc-s3c.h index 9efc04dd255a..709c6f7e2f8c 100644 --- a/include/linux/platform_data/asoc-s3c.h +++ b/include/linux/platform_data/asoc-s3c.h @@ -1,5 +1,4 @@ -/* arch/arm/plat-samsung/include/plat/audio.h - * +/* * Copyright (c) 2009 Samsung Electronics Co. Ltd * Author: Jaswinder Singh <jassi.brar(a)samsung.com> * -- 1.7.9.5

2 2

Re: [alsa-devel] ASoC: Intel: Add Haswell/Broadwell IPC
by Dan Carpenter 25 Feb '14

25 Feb '14

Hello Liam Girdwood, The patch 22981243589c: "ASoC: Intel: Add Haswell/Broadwell IPC" from Feb 20, 2014, leads to the following imaginary static checker warning: sound/soc/intel/sst-haswell-ipc.c:898 hsw_irq_thread() warn: this is always true. sound/soc/intel/sst-haswell-ipc.c 895 /* Handle Immediate reply from DSP Core */ 896 handled = hsw_process_reply(hsw, ipcx); ^^^^^^^^^^^^^^^^^ Returns 1 on success/error and -EIO on error. 897 898 if (handled) { 899 /* clear DONE bit - tell DSP we have completed */ 900 sst_dsp_shim_update_bits_unlocked(sst, SST_IPCX, 901 SST_IPCX_DONE, 0); 902 903 /* unmask Done interrupt */ 904 sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, 905 SST_IMRX_DONE, 0); 906 } regards, dan carpenter

1 0

[alsa-devel] [patch 2/2] ASoC: sst-haswell-ipc: incorrect sizeof()
by Dan Carpenter 25 Feb '14

25 Feb '14

This should be sizeof(pos) instead of sizeof(&pos). Most likely they are both 8 bytes though so it doesn't often make a difference in real life. Fixes: 22981243589c ('ASoC: Intel: Add Haswell/Broadwell IPC') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 552aebf2c8f5..1f1576a9586a 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -617,7 +617,7 @@ static void hsw_notification_work(struct work_struct *work) case IPC_POSITION_CHANGED: trace_ipc_notification("DSP stream position changed for", stream->reply.stream_hw_id); - sst_dsp_inbox_read(hsw->dsp, pos, sizeof(&pos)); + sst_dsp_inbox_read(hsw->dsp, pos, sizeof(pos)); if (stream->notify_position) stream->notify_position(stream, stream->pdata);

1 0

[alsa-devel] External sound card issue
by Ionica Bizau 25 Feb '14

25 Feb '14

Hi, I have just bought an external sound card: Asus Xonar U7 I typed in terminal: alsamixer Then I wanted to change the sound card (I have connected an external sound card via USB). I pressed F6, selected the external sound card and pressed enter. alsamixer crashed and outputed: cannot load mixer controls: Broken pipe My ALSA information is located at http://www.alsa-project.org/db/?f=1871c086b58a2965d18cfb38483794788187edfb Please see this bug reported in alsa-driver package: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1267866 Tell me what other information you need. Can you fix this bug in 7 days? If you cannot I will return the sound card to the shop.... :-( Thank you! Good luck! I wait for your answer, Ionica Bizau

6 15

[alsa-devel] [PATCH 0/7] ASoC: Intel: Add support for Baytrail SST DSP
by Jarkko Nikula 25 Feb '14

25 Feb '14

This set adds support for Baytrail SST DSP with RT5640 based machine driver. Design is quite similar and is actually inherited from Intel Haswell SST DSP drivers that were posted recently. Reason for another implementation is that there are more differences than similarities between Haswell and Baytrail. First difference comes from the SHIM registers that are 32-bit on Haswell and 64-bit on Baytrail. Also memory map and message mailbox is different. Second difference is message passing and message headers between host and SST DSP. For instance the DSP notifies the host using different SHIM registers and bits than on Haswell. Biggest difference comes from the DSP firmware. It has different IPC messages, stream setup and how DMA pointer is tracked. Currently firmware has less features than Haswell DSP firmware and it doesn't support compress offload and volume controls support. The RT5640 based machine driver is for our customer reference boards but at quick test it appears to work on Asus T100 too. Note currently the SST is the I2S master on this machine driver and codec is set to use I2C BCLK as it's input clock. Jarkko Nikula (7): ASoC: Intel: Add Baytrail SST ID and Baytrail specific register bits ASoC: Intel: Add Intel Baytrail SST DSP support ASoC: Intel: Add Intel Baytrail SST DSP IPC support ASoC: Intel: Add Intel Baytrail SST PCM platform driver ASoC: Intel: Add machine driver for Baytrail SST with RT5640 codec ASoC: Intel: Add Baytrail SST and byt-rt5640 machine driver probing ASoC: Intel: Add build support for Baytrail SST sound/soc/intel/Kconfig | 14 +- sound/soc/intel/Makefile | 5 + sound/soc/intel/byt-rt5640.c | 194 +++++++++ sound/soc/intel/sst-acpi.c | 17 + sound/soc/intel/sst-baytrail-dsp.c | 372 ++++++++++++++++ sound/soc/intel/sst-baytrail-ipc.c | 866 +++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-baytrail-ipc.h | 69 +++ sound/soc/intel/sst-baytrail-pcm.c | 422 ++++++++++++++++++ sound/soc/intel/sst-dsp-priv.h | 1 + sound/soc/intel/sst-dsp.h | 11 + 10 files changed, 1970 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/byt-rt5640.c create mode 100644 sound/soc/intel/sst-baytrail-dsp.c create mode 100644 sound/soc/intel/sst-baytrail-ipc.c create mode 100644 sound/soc/intel/sst-baytrail-ipc.h create mode 100644 sound/soc/intel/sst-baytrail-pcm.c -- 1.8.5.3

3 10

[alsa-devel] [PATCH] ASoC: wm8958-dsp: Fix firmware block loading
by Lars-Peter Clausen 25 Feb '14

25 Feb '14

The codec->control_data contains a pointer to the device's regmap struct. But wm8994_bulk_write() expects a pointer to the parent wm8998 device. The issue was introduced in commit d9a7666f ("ASoC: Remove ASoC-specific WM8994 I/O code"). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm8958-dsp2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 1974377..7ac2e51 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -153,7 +153,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, data32 &= 0xffffff; - wm8994_bulk_write(codec->control_data, + wm8994_bulk_write(wm8994->wm8994, data32 & 0xffffff, block_len / 2, (void *)(data + 8)); -- 1.8.0

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm8996: Replace codec->control_data with wm8996->regmap
by Lars-Peter Clausen 25 Feb '14

25 Feb '14

With the ongoing component-ization of the ASoC framework and the continuing migration to using regmap for IO the control_data field of the snd_soc_codec struct will eventually be removed. Prepare the wm8996 driver for this by using wm8996->regmap instead of accessing the CODEC's control_data field. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm8996.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 0330165..e2d0dd7 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1608,8 +1608,8 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, msleep(5); } - regcache_cache_only(codec->control_data, false); - regcache_sync(codec->control_data); + regcache_cache_only(wm8996->regmap, false); + regcache_sync(wm8996->regmap); } /* Bypass the MICBIASes for lowest power */ @@ -1620,10 +1620,10 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: - regcache_cache_only(codec->control_data, true); + regcache_cache_only(wm8996->regmap, true); if (wm8996->pdata.ldo_ena >= 0) { gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); - regcache_cache_only(codec->control_data, true); + regcache_cache_only(wm8996->regmap, true); } regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); -- 1.8.0

2 3

[alsa-devel] [PATCH] ASoC: wm8993: Remove unused pointer in wm8993_remove()
by Christian Engelmayer 25 Feb '14

25 Feb '14

Commit 88b5bdfd (ASoC: wm8993: drop regulator_bulk_free of devm_ allocated data) eliminated the last user of driver data pointer 'wm8993' in function wm8993_remove() - Thus remove it. Detected by Coverity: CID 1186208. Signed-off-by: Christian Engelmayer <cengelma(a)gmx.at> --- Applies against branch for-next in tree git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git --- sound/soc/codecs/wm8993.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 2ee23a3..f7978b3 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1559,8 +1559,6 @@ static int wm8993_probe(struct snd_soc_codec *codec) static int wm8993_remove(struct snd_soc_codec *codec) { - struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); - wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: da732x: Use da732x->regmap instead of codec->control_data
by Lars-Peter Clausen 25 Feb '14

25 Feb '14

With the ongoing component-ization of the ASoC framework and the continuing migration to using regmap for IO the control_data field of the snd_soc_codec struct will eventually be removed. Prepare the da732x driver for this by using da732x->regmap instead of accessing the CODEC's control_data field. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/da732x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index f295b65..54db45d 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1487,8 +1487,8 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec, da732x_hp_dc_offset_cancellation(codec); - regcache_cache_only(codec->control_data, false); - regcache_sync(codec->control_data); + regcache_cache_only(da732x->regmap, false); + regcache_sync(da732x->regmap); } else { snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, DA732X_BIAS_BOOST_MASK, @@ -1499,7 +1499,7 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_OFF: - regcache_cache_only(codec->control_data, true); + regcache_cache_only(da732x->regmap, true); da732x_set_charge_pump(codec, DA732X_DISABLE_CP); snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, DA732X_BIAS_EN, DA732X_BIAS_DIS); -- 1.8.0

3 2

[alsa-devel] [PATCH] ASoC: cs42888: Add codec driver support
by Nicolin Chen 25 Feb '14

25 Feb '14

This patch adds support for the Cirrus Logic CS42888 Audio CODEC that has four 24-bit A/D and eight 24-bit D/A converters. [ CS42888 supports both I2C and SPI control ports. As initial patch, this patch only adds the support for I2C. ] Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- .../devicetree/bindings/sound/cs42888.txt | 27 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs42888.c | 552 +++++++++++++++++++++ sound/soc/codecs/cs42888.h | 209 ++++++++ 5 files changed, 795 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs42888.txt create mode 100644 sound/soc/codecs/cs42888.c create mode 100644 sound/soc/codecs/cs42888.h diff --git a/Documentation/devicetree/bindings/sound/cs42888.txt b/Documentation/devicetree/bindings/sound/cs42888.txt new file mode 100644 index 0000000..7736527 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42888.txt @@ -0,0 +1,27 @@ +CS42888 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42888" + + - reg : the I2C address of the device for I2C + + - clocks : phandle to the clock source for MCLK + + - clock-names : must contain "mclk". + + - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt + +Example: + +codec: cs42888@48 { + compatible = "cirrus,cs42888"; + reg = <0x48>; + clocks = <&codec_mclk 0>; + clock-names = "mclk"; + VA-supply = <&reg_audio>; + VD-supply = <&reg_audio>; + VLS-supply = <&reg_audio>; + VLC-supply = <&reg_audio>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f2383eb..3211488 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,6 +44,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI + select SND_SOC_CS42888 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if I2C select SND_SOC_DA7213 if I2C @@ -300,6 +301,10 @@ config SND_SOC_CS4271 tristate "Cirrus Logic CS4271 CODEC" depends on SND_SOC_I2C_AND_SPI +config SND_SOC_CS42888 + tristate "Cirrus Logic CS42888 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6af7a55..75fe757 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -30,6 +30,7 @@ snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o +snd-soc-cs42888-objs := cs42888.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -173,6 +174,7 @@ obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o +obj-$(CONFIG_SND_SOC_CS42888) += snd-soc-cs42888.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs42888.c b/sound/soc/codecs/cs42888.c new file mode 100644 index 0000000..8561a25 --- /dev/null +++ b/sound/soc/codecs/cs42888.c @@ -0,0 +1,552 @@ +/* + * Cirrus Logic CS42888 Audio CODEC Digital Audio Interface (DAI) driver + * + * 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. + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "cs42888.h" + +#define CS42888_NUM_SUPPLIES 4 +static const char *cs42888_supply_names[CS42888_NUM_SUPPLIES] = { + "VA", + "VD", + "VLS", + "VLC", +}; + +#define CS42888_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct cs42888_priv { + struct regulator_bulk_data supplies[CS42888_NUM_SUPPLIES]; + struct regmap *regmap; + struct clk *clk; + + bool slave_mode; + unsigned long sysclk; +}; + +/* -127.5dB to 0dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +/* -64dB to 24dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(adc_tlv, -6400, 50, 0); + +static const char *cs42888_adc_single[] = { "Differential", "Single-Ended" }; +static const char *cs42888_szc[] = { "Immediate Change", "Zero Cross", + "Soft Ramp", "Soft Ramp on Zero Cross" }; + +static const struct soc_enum cs42888_enum[] = { + SOC_ENUM_SINGLE(CS42888_ADCCTL, 4, 2, cs42888_adc_single), + SOC_ENUM_SINGLE(CS42888_ADCCTL, 3, 2, cs42888_adc_single), + SOC_ENUM_SINGLE(CS42888_TXCTL, 5, 4, cs42888_szc), + SOC_ENUM_SINGLE(CS42888_TXCTL, 0, 4, cs42888_szc), +}; + +static const struct snd_kcontrol_new cs42888_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", CS42888_VOLAOUT1, + CS42888_VOLAOUT2, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC2 Playback Volume", CS42888_VOLAOUT3, + CS42888_VOLAOUT4, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC3 Playback Volume", CS42888_VOLAOUT5, + CS42888_VOLAOUT6, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC4 Playback Volume", CS42888_VOLAOUT7, + CS42888_VOLAOUT8, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_S_TLV("ADC1 Capture Volume", CS42888_VOLAIN1, + CS42888_VOLAIN2, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE_R_S_TLV("ADC2 Capture Volume", CS42888_VOLAIN3, + CS42888_VOLAIN4, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE("DAC1 Invert Switch", CS42888_DACINV, 0, 1, 1, 0), + SOC_DOUBLE("DAC2 Invert Switch", CS42888_DACINV, 2, 3, 1, 0), + SOC_DOUBLE("DAC3 Invert Switch", CS42888_DACINV, 4, 5, 1, 0), + SOC_DOUBLE("DAC4 Invert Switch", CS42888_DACINV, 6, 7, 1, 0), + SOC_DOUBLE("ADC1 Invert Switch", CS42888_ADCINV, 0, 1, 1, 0), + SOC_DOUBLE("ADC2 Invert Switch", CS42888_ADCINV, 2, 3, 1, 0), + SOC_SINGLE("ADC High-Pass Filter Switch", CS42888_ADCCTL, 7, 1, 1), + SOC_SINGLE("DAC De-emphasis Switch", CS42888_ADCCTL, 5, 1, 0), + SOC_ENUM("ADC1 Single Ended Mode Switch", cs42888_enum[0]), + SOC_ENUM("ADC2 Single Ended Mode Switch", cs42888_enum[1]), + SOC_SINGLE("DAC Single Volume Control Switch", CS42888_TXCTL, 7, 1, 0), + SOC_ENUM("DAC Soft Ramp & Zero Cross Control Switch", cs42888_enum[2]), + SOC_SINGLE("DAC Auto Mute Switch", CS42888_TXCTL, 4, 1, 0), + SOC_SINGLE("Mute ADC Serial Port Switch", CS42888_TXCTL, 3, 1, 0), + SOC_SINGLE("ADC Single Volume Control Switch", CS42888_TXCTL, 2, 1, 0), + SOC_ENUM("ADC Soft Ramp & Zero Cross Control Switch", cs42888_enum[3]), +}; + +static const struct snd_soc_dapm_widget cs42888_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC1", "Playback", CS42888_PWRCTL, 1, 1), + SND_SOC_DAPM_DAC("DAC2", "Playback", CS42888_PWRCTL, 2, 1), + SND_SOC_DAPM_DAC("DAC3", "Playback", CS42888_PWRCTL, 3, 1), + SND_SOC_DAPM_DAC("DAC4", "Playback", CS42888_PWRCTL, 4, 1), + + SND_SOC_DAPM_OUTPUT("AOUT1L"), + SND_SOC_DAPM_OUTPUT("AOUT1R"), + SND_SOC_DAPM_OUTPUT("AOUT2L"), + SND_SOC_DAPM_OUTPUT("AOUT2R"), + SND_SOC_DAPM_OUTPUT("AOUT3L"), + SND_SOC_DAPM_OUTPUT("AOUT3R"), + SND_SOC_DAPM_OUTPUT("AOUT4L"), + SND_SOC_DAPM_OUTPUT("AOUT4R"), + + SND_SOC_DAPM_ADC("ADC1", "Capture", CS42888_PWRCTL, 5, 1), + SND_SOC_DAPM_ADC("ADC2", "Capture", CS42888_PWRCTL, 6, 1), + + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + + SND_SOC_DAPM_SUPPLY("PWR", CS42888_PWRCTL, 0, 1, NULL, 0), +}; + +static const struct snd_soc_dapm_route cs42888_dapm_routes[] = { + /* Playback */ + { "AOUT1L", NULL, "DAC1" }, + { "AOUT1R", NULL, "DAC1" }, + { "DAC1", NULL, "PWR" }, + + { "AOUT2L", NULL, "DAC2" }, + { "AOUT2R", NULL, "DAC2" }, + { "DAC2", NULL, "PWR" }, + + { "AOUT3L", NULL, "DAC3" }, + { "AOUT3R", NULL, "DAC3" }, + { "DAC3", NULL, "PWR" }, + + { "AOUT4L", NULL, "DAC4" }, + { "AOUT4R", NULL, "DAC4" }, + { "DAC4", NULL, "PWR" }, + + /* Capture */ + { "ADC1", NULL, "AIN1L" }, + { "ADC1", NULL, "AIN1R" }, + { "ADC1", NULL, "PWR" }, + + { "ADC2", NULL, "AIN2L" }, + { "ADC2", NULL, "AIN2R" }, + { "ADC2", NULL, "PWR" }, +}; + +struct cs42888_ratios { + unsigned int ratio; + unsigned char speed; + unsigned char mclk; +}; + +static struct cs42888_ratios cs42888_ratios[] = { + { 64, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_256(4) }, + { 96, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_384(4) }, + { 128, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_512(4) }, + { 192, CS42888_FM_QUAD, CS42888_FUNCMOD_MFREQ_768(4) }, + { 256, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_256(1) }, + { 384, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_384(1) }, + { 512, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_512(1) }, + { 768, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_768(1) }, + { 1024, CS42888_FM_SINGLE, CS42888_FUNCMOD_MFREQ_1024(1) } +}; + +static int cs42888_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + cs42888->sysclk = freq; + + return 0; +} + +static int cs42888_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + u32 val; + + /* Set DAI format */ + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + val = CS42888_INTF_DAC_DIF_LEFTJ | CS42888_INTF_ADC_DIF_LEFTJ; + break; + case SND_SOC_DAIFMT_I2S: + val = CS42888_INTF_DAC_DIF_I2S | CS42888_INTF_ADC_DIF_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val = CS42888_INTF_DAC_DIF_RIGHTJ | CS42888_INTF_ADC_DIF_RIGHTJ; + break; + default: + dev_err(codec->dev, "unsupported dai format\n"); + return -EINVAL; + } + + regmap_update_bits(cs42888->regmap, CS42888_INTF, + CS42888_INTF_DAC_DIF_MASK | + CS42888_INTF_ADC_DIF_MASK, val); + + /* Set master/slave audio interface */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs42888->slave_mode = true; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs42888->slave_mode = false; + break; + default: + dev_err(codec->dev, "unsupported master/slave mode\n"); + return -EINVAL; + } + + return 0; +} + +static int cs42888_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 cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 ratio = cs42888->sysclk / params_rate(params); + u32 i, fm, val, mask; + + for (i = 0; i < ARRAY_SIZE(cs42888_ratios); i++) { + if (cs42888_ratios[i].ratio == ratio) + break; + } + + if (i == ARRAY_SIZE(cs42888_ratios)) { + dev_err(codec->dev, "unsupported sysclk ratio\n"); + return -EINVAL; + } + + mask = CS42888_FUNCMOD_MFREQ_MASK; + val = cs42888_ratios[i].mclk; + + fm = cs42888->slave_mode ? CS42888_FM_AUTO : cs42888_ratios[i].speed; + + regmap_update_bits(cs42888->regmap, CS42888_FUNCMOD, + CS42888_FUNCMOD_xC_FM_MASK(tx) | mask, + CS42888_FUNCMOD_xC_FM(tx, fm) | val); + + return 0; +} + +static int cs42888_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(cs42888->regmap, CS42888_DACMUTE, + CS42888_DACMUTE_ALL, mute ? CS42888_DACMUTE_ALL : 0); + + return 0; +} + +static struct snd_soc_dai_ops cs42888_dai_ops = { + .set_fmt = cs42888_set_dai_fmt, + .set_sysclk = cs42888_set_dai_sysclk, + .hw_params = cs42888_hw_params, + .digital_mute = cs42888_digital_mute, +}; + +static struct snd_soc_dai_driver cs42888_dai = { + .name = "cs42888", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42888_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42888_FORMATS, + }, + .ops = &cs42888_dai_ops, +}; + +static struct reg_default cs42888_reg[] = { + { 0x01, 0x01 }, /* Chip I.D. and Revision Register */ + { 0x02, 0x00 }, /* Power Control */ + { 0x03, 0xF0 }, /* Functional Mode */ + { 0x04, 0x46 }, /* Interface Formats */ + { 0x05, 0x00 }, /* ADC Control & DAC De-Emphasis */ + { 0x06, 0x10 }, /* Transition Control */ + { 0x07, 0x00 }, /* DAC Channel Mute */ + { 0x08, 0x00 }, /* Volume Control AOUT1 */ + { 0x09, 0x00 }, /* Volume Control AOUT2 */ + { 0x0a, 0x00 }, /* Volume Control AOUT3 */ + { 0x0b, 0x00 }, /* Volume Control AOUT4 */ + { 0x0c, 0x00 }, /* Volume Control AOUT5 */ + { 0x0d, 0x00 }, /* Volume Control AOUT6 */ + { 0x0e, 0x00 }, /* Volume Control AOUT7 */ + { 0x0f, 0x00 }, /* Volume Control AOUT8 */ + { 0x10, 0x00 }, /* DAC Channel Invert */ + { 0x11, 0x00 }, /* Volume Control AIN1 */ + { 0x12, 0x00 }, /* Volume Control AIN2 */ + { 0x13, 0x00 }, /* Volume Control AIN3 */ + { 0x14, 0x00 }, /* Volume Control AIN4 */ + { 0x17, 0x00 }, /* ADC Channel Invert */ + { 0x18, 0x00 }, /* Status Control */ + { 0x1a, 0x00 }, /* Status Mask */ + { 0x1b, 0x00 }, /* MUTEC Pin Control */ +}; + +static bool cs42888_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42888_STATUS: + return true; + default: + return false; + } +} + +static bool cs42888_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42888_CHIPID: + case CS42888_STATUS: + return false; + default: + return true; + } +} + +static const struct regmap_config cs42888_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42888_LASTREG, + .reg_defaults = cs42888_reg, + .num_reg_defaults = ARRAY_SIZE(cs42888_reg), + .volatile_reg = cs42888_volatile_register, + .writeable_reg = cs42888_writeable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs42888_probe(struct snd_soc_codec *codec) +{ + struct cs42888_priv *cs42888 = snd_soc_codec_get_drvdata(codec); + + /* Disable auto-mute */ + regmap_update_bits(cs42888->regmap, CS42888_TXCTL, + CS42888_TXCTL_AMUTE | CS42888_TXCTL_DAC_SZC_MASK, + CS42888_TXCTL_DAC_SZC_SR); + + /* Mute all DAC channels */ + regmap_write(cs42888->regmap, CS42888_DACMUTE, CS42888_DACMUTE_ALL); + + return 0; +} + +static struct snd_soc_codec_driver cs42888_driver = { + .probe = cs42888_probe, + .idle_bias_off = true, + + .controls = cs42888_snd_controls, + .num_controls = ARRAY_SIZE(cs42888_snd_controls), + .dapm_widgets = cs42888_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42888_dapm_widgets), + .dapm_routes = cs42888_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs42888_dapm_routes), +}; + +static int cs42888_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct cs42888_priv *cs42888; + int ret, val, i; + + cs42888 = devm_kzalloc(&i2c->dev, sizeof(*cs42888), GFP_KERNEL); + if (cs42888 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, cs42888); + + cs42888->clk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(cs42888->clk)) + dev_warn(&i2c->dev, "failed to get the clock: %ld\n", + PTR_ERR(cs42888->clk)); + else + cs42888->sysclk = clk_get_rate(cs42888->clk); + + for (i = 0; i < ARRAY_SIZE(cs42888->supplies); i++) + cs42888->supplies[i].supply = cs42888_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, + ARRAY_SIZE(cs42888->supplies), cs42888->supplies); + if (ret) { + dev_err(&i2c->dev, "failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + if (ret) { + dev_err(&i2c->dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + /* Make sure hardware reset done */ + msleep(5); + + cs42888->regmap = devm_regmap_init_i2c(i2c, &cs42888_regmap); + if (IS_ERR(cs42888->regmap)) { + ret = PTR_ERR(cs42888->regmap); + dev_err(&i2c->dev, "failed to allocate regmap: %d\n", ret); + goto err_enable; + } + + /* + * We haven't marked the chip revision as volatile due to + * sharing a register with the right input volume; explicitly + * bypass the cache to read it. + */ + regcache_cache_bypass(cs42888->regmap, true); + + /* Validate the chip ID */ + regmap_read(cs42888->regmap, CS42888_CHIPID, &val); + if (val < 0) { + dev_err(&i2c->dev, "failed to get device ID: %x", val); + ret = -EINVAL; + goto err_enable; + } + + /* The top four bits of the chip ID should be 0000 */ + if ((val & CS42888_CHIPID_CHIP_ID_MASK) != 0x00) { + dev_err(&i2c->dev, "unmatched chip ID: %d\n", + val & CS42888_CHIPID_CHIP_ID_MASK); + ret = -EINVAL; + goto err_enable; + } + + dev_info(&i2c->dev, "found device at %X, revision %X\n", + i2c->addr, val & CS42888_CHIPID_REV_ID_MASK); + + regcache_cache_bypass(cs42888->regmap, false); + + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + + ret = snd_soc_register_codec(&i2c->dev, &cs42888_driver, &cs42888_dai, 1); + if (ret) { + dev_err(&i2c->dev, "failed to register codec:%d\n", ret); + goto err_enable; + } + + regcache_cache_only(cs42888->regmap, true); + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), cs42888->supplies); + + return ret; +} + +static int cs42888_i2c_remove(struct i2c_client *i2c_client) +{ + snd_soc_unregister_codec(&i2c_client->dev); + return 0; +} + +#ifdef CONFIG_PM_RUNTIME +static int cs42888_runtime_resume(struct device *dev) +{ + struct cs42888_priv *cs42888 = dev_get_drvdata(dev); + int ret; + + if (!IS_ERR(cs42888->clk)) + clk_prepare_enable(cs42888->clk); + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + /* + * In case the device was put to hard reset during sleep, + * we need to wait 500ns here before any I2C communication + */ + mdelay(5); + + regcache_cache_only(cs42888->regmap, false); + + regcache_sync(cs42888->regmap); + + return 0; +} + +static int cs42888_runtime_suspend(struct device *dev) +{ + struct cs42888_priv *cs42888 = dev_get_drvdata(dev); + + regcache_cache_only(cs42888->regmap, true); + + regulator_bulk_disable(ARRAY_SIZE(cs42888->supplies), + cs42888->supplies); + + if (!IS_ERR(cs42888->clk)) + clk_disable_unprepare(cs42888->clk); + + return 0; +} +#endif + +static const struct dev_pm_ops cs42888_pm = { + SET_RUNTIME_PM_OPS(cs42888_runtime_suspend, cs42888_runtime_resume, NULL) +}; + +static struct i2c_device_id cs42888_i2c_id[] = { + {"cs42888", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs42888_i2c_id); + +static const struct of_device_id cs42888_of_match[] = { + { .compatible = "cirrus,cs42888", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, cs42888_of_match); + +static struct i2c_driver cs42888_i2c_driver = { + .driver = { + .name = "cs42888", + .owner = THIS_MODULE, + .of_match_table = cs42888_of_match, + .pm = &cs42888_pm, + }, + .probe = cs42888_i2c_probe, + .remove = cs42888_i2c_remove, + .id_table = cs42888_i2c_id, +}; + +module_i2c_driver(cs42888_i2c_driver); + +MODULE_DESCRIPTION("Cirrus Logic CS42888 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42888.h b/sound/soc/codecs/cs42888.h new file mode 100644 index 0000000..6d2cdec --- /dev/null +++ b/sound/soc/codecs/cs42888.h @@ -0,0 +1,209 @@ +/* + * cs42888.h - Cirrus Logic CS42888 Audio CODEC driver header file + * + * 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 _CS42888_H +#define _CS42888_H + +/* CS42888 register map */ +#define CS42888_CHIPID 0x01 /* Chip ID */ +#define CS42888_PWRCTL 0x02 /* Power Control */ +#define CS42888_FUNCMOD 0x03 /* Functional Mode */ +#define CS42888_INTF 0x04 /* Interface Formats */ +#define CS42888_ADCCTL 0x05 /* ADC Control */ +#define CS42888_TXCTL 0x06 /* Transition Control */ +#define CS42888_DACMUTE 0x07 /* DAC Mute Control */ +#define CS42888_VOLAOUT1 0x08 /* Volume Control AOUT1 */ +#define CS42888_VOLAOUT2 0x09 /* Volume Control AOUT2 */ +#define CS42888_VOLAOUT3 0x0A /* Volume Control AOUT3 */ +#define CS42888_VOLAOUT4 0x0B /* Volume Control AOUT4 */ +#define CS42888_VOLAOUT5 0x0C /* Volume Control AOUT5 */ +#define CS42888_VOLAOUT6 0x0D /* Volume Control AOUT6 */ +#define CS42888_VOLAOUT7 0x0E /* Volume Control AOUT7 */ +#define CS42888_VOLAOUT8 0x0F /* Volume Control AOUT8 */ +#define CS42888_DACINV 0x10 /* DAC Channel Invert */ +#define CS42888_VOLAIN1 0x11 /* Volume Control AIN1 */ +#define CS42888_VOLAIN2 0x12 /* Volume Control AIN2 */ +#define CS42888_VOLAIN3 0x13 /* Volume Control AIN3 */ +#define CS42888_VOLAIN4 0x14 /* Volume Control AIN4 */ +#define CS42888_ADCINV 0x17 /* ADC Channel Invert */ +#define CS42888_STATUSCTL 0x18 /* Status Control */ +#define CS42888_STATUS 0x19 /* Status */ +#define CS42888_STATUSM 0x1A /* Status Mask */ +#define CS42888_MUTEC 0x1B /* MUTEC Pin Control */ + +#define CS42888_FIRSTREG CS42888_CHIPID +#define CS42888_LASTREG CS42888_MUTEC +#define CS42888_NUMREGS (CS42888_LASTREG - CS42888_FIRSTREG + 1) +#define CS42888_I2C_INCR 0x80 + +/* Chip I.D. and Revision Register (Address 01h) */ +#define CS42888_CHIPID_CHIP_ID_MASK 0xF0 +#define CS42888_CHIPID_REV_ID_MASK 0x0F + +/* Power Control (Address 02h) */ +#define CS42888_PWRCTL_PDN_ADC2_SHIFT 6 +#define CS42888_PWRCTL_PDN_ADC2_MASK (1 << CS42888_PWRCTL_PDN_ADC2_SHIFT) +#define CS42888_PWRCTL_PDN_ADC2 (1 << CS42888_PWRCTL_PDN_ADC2_SHIFT) +#define CS42888_PWRCTL_PDN_ADC1_SHIFT 5 +#define CS42888_PWRCTL_PDN_ADC1_MASK (1 << CS42888_PWRCTL_PDN_ADC1_SHIFT) +#define CS42888_PWRCTL_PDN_ADC1 (1 << CS42888_PWRCTL_PDN_ADC1_SHIFT) +#define CS42888_PWRCTL_PDN_DAC4_SHIFT 4 +#define CS42888_PWRCTL_PDN_DAC4_MASK (1 << CS42888_PWRCTL_PDN_DAC4_SHIFT) +#define CS42888_PWRCTL_PDN_DAC4 (1 << CS42888_PWRCTL_PDN_DAC4_SHIFT) +#define CS42888_PWRCTL_PDN_DAC3_SHIFT 3 +#define CS42888_PWRCTL_PDN_DAC3_MASK (1 << CS42888_PWRCTL_PDN_DAC3_SHIFT) +#define CS42888_PWRCTL_PDN_DAC3 (1 << CS42888_PWRCTL_PDN_DAC3_SHIFT) +#define CS42888_PWRCTL_PDN_DAC2_SHIFT 2 +#define CS42888_PWRCTL_PDN_DAC2_MASK (1 << CS42888_PWRCTL_PDN_DAC2_SHIFT) +#define CS42888_PWRCTL_PDN_DAC2 (1 << CS42888_PWRCTL_PDN_DAC2_SHIFT) +#define CS42888_PWRCTL_PDN_DAC1_SHIFT 1 +#define CS42888_PWRCTL_PDN_DAC1_MASK (1 << CS42888_PWRCTL_PDN_DAC1_SHIFT) +#define CS42888_PWRCTL_PDN_DAC1 (1 << CS42888_PWRCTL_PDN_DAC1_SHIFT) +#define CS42888_PWRCTL_PDN_SHIFT 0 +#define CS42888_PWRCTL_PDN_MASK (1 << CS42888_PWRCTL_PDN_SHIFT) +#define CS42888_PWRCTL_PDN (1 << CS42888_PWRCTL_PDN_SHIFT) + +/* Functional Mode (Address 03h) */ +#define CS42888_FUNCMOD_DAC_FM_SHIFT 6 +#define CS42888_FUNCMOD_DAC_FM_WIDTH 2 +#define CS42888_FUNCMOD_DAC_FM_MASK (((1 << CS42888_FUNCMOD_DAC_FM_WIDTH) - 1) << CS42888_FUNCMOD_DAC_FM_SHIFT) +#define CS42888_FUNCMOD_DAC_FM(v) ((v) << CS42888_FUNCMOD_DAC_FM_SHIFT) +#define CS42888_FUNCMOD_ADC_FM_SHIFT 4 +#define CS42888_FUNCMOD_ADC_FM_WIDTH 2 +#define CS42888_FUNCMOD_ADC_FM_MASK (((1 << CS42888_FUNCMOD_ADC_FM_WIDTH) - 1) << CS42888_FUNCMOD_ADC_FM_SHIFT) +#define CS42888_FUNCMOD_ADC_FM(v) ((v) << CS42888_FUNCMOD_ADC_FM_SHIFT) +#define CS42888_FUNCMOD_xC_FM_MASK(x) ((x) ? CS42888_FUNCMOD_DAC_FM_MASK : CS42888_FUNCMOD_ADC_FM_MASK) +#define CS42888_FUNCMOD_xC_FM(x, v) ((x) ? CS42888_FUNCMOD_DAC_FM(v) : CS42888_FUNCMOD_ADC_FM(v)) +#define CS42888_FUNCMOD_MFREQ_SHIFT 1 +#define CS42888_FUNCMOD_MFREQ_WIDTH 3 +#define CS42888_FUNCMOD_MFREQ_MASK (((1 << CS42888_FUNCMOD_MFREQ_WIDTH) - 1) << CS42888_FUNCMOD_MFREQ_SHIFT) +#define CS42888_FUNCMOD_MFREQ_256(s) ((0 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_384(s) ((1 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_512(s) ((2 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_768(s) ((3 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42888_FUNCMOD_MFREQ_1024(s) ((4 << CS42888_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) + +#define CS42888_FM_SINGLE 0 +#define CS42888_FM_DOUBLE 1 +#define CS42888_FM_QUAD 2 +#define CS42888_FM_AUTO 3 + +/* Interface Formats (Address 04h) */ +#define CS42888_INTF_FREEZE_SHIFT 7 +#define CS42888_INTF_FREEZE_MASK (1 << CS42888_INTF_FREEZE_SHIFT) +#define CS42888_INTF_FREEZE (1 << CS42888_INTF_FREEZE_SHIFT) +#define CS42888_INTF_AUX_DIF_SHIFT 6 +#define CS42888_INTF_AUX_DIF_MASK (1 << CS42888_INTF_AUX_DIF_SHIFT) +#define CS42888_INTF_AUX_DIF (1 << CS42888_INTF_AUX_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_SHIFT 3 +#define CS42888_INTF_DAC_DIF_WIDTH 3 +#define CS42888_INTF_DAC_DIF_MASK (((1 << CS42888_INTF_DAC_DIF_WIDTH) - 1) << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_LEFTJ (0 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_I2S (1 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_RIGHTJ (2 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_RIGHTJ_16 (3 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_ONELINE_20 (4 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_ONELINE_24 (6 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_DAC_DIF_TDM (7 << CS42888_INTF_DAC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_SHIFT 0 +#define CS42888_INTF_ADC_DIF_WIDTH 3 +#define CS42888_INTF_ADC_DIF_MASK (((1 << CS42888_INTF_ADC_DIF_WIDTH) - 1) << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_LEFTJ (0 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_I2S (1 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_RIGHTJ (2 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_RIGHTJ_16 (3 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_ONELINE_20 (4 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_ONELINE_24 (6 << CS42888_INTF_ADC_DIF_SHIFT) +#define CS42888_INTF_ADC_DIF_TDM (7 << CS42888_INTF_ADC_DIF_SHIFT) + +/* ADC Control & DAC De-Emphasis (Address 05h) */ +#define CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT 7 +#define CS42888_ADCCTL_ADC_HPF_FREEZE_MASK (1 << CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42888_ADCCTL_ADC_HPF_FREEZE (1 << CS42888_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42888_ADCCTL_DAC_DEM_SHIFT 5 +#define CS42888_ADCCTL_DAC_DEM_MASK (1 << CS42888_ADCCTL_DAC_DEM_SHIFT) +#define CS42888_ADCCTL_DAC_DEM (1 << CS42888_ADCCTL_DAC_DEM_SHIFT) +#define CS42888_ADCCTL_ADC1_SINGLE_SHIFT 4 +#define CS42888_ADCCTL_ADC1_SINGLE_MASK (1 << CS42888_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC1_SINGLE (1 << CS42888_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC2_SINGLE_SHIFT 3 +#define CS42888_ADCCTL_ADC2_SINGLE_MASK (1 << CS42888_ADCCTL_ADC2_SINGLE_SHIFT) +#define CS42888_ADCCTL_ADC2_SINGLE (1 << CS42888_ADCCTL_ADC2_SINGLE_SHIFT) + +/* Transition Control (Address 06h) */ +#define CS42888_TXCTL_DAC_SNGVOL_SHIFT 7 +#define CS42888_TXCTL_DAC_SNGVOL_MASK (1 << CS42888_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42888_TXCTL_DAC_SNGVOL (1 << CS42888_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SHIFT 5 +#define CS42888_TXCTL_DAC_SZC_WIDTH 2 +#define CS42888_TXCTL_DAC_SZC_MASK (((1 << CS42888_TXCTL_DAC_SZC_WIDTH) - 1) << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_IC (0 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_ZC (1 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SR (2 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_DAC_SZC_SRZC (3 << CS42888_TXCTL_DAC_SZC_SHIFT) +#define CS42888_TXCTL_AMUTE_SHIFT 4 +#define CS42888_TXCTL_AMUTE_MASK (1 << CS42888_TXCTL_AMUTE_SHIFT) +#define CS42888_TXCTL_AMUTE (1 << CS42888_TXCTL_AMUTE_SHIFT) +#define CS42888_TXCTL_MUTE_ADC_SP_SHIFT 3 +#define CS42888_TXCTL_MUTE_ADC_SP_MASK (1 << CS42888_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42888_TXCTL_MUTE_ADC_SP (1 << CS42888_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42888_TXCTL_ADC_SNGVOL_SHIFT 2 +#define CS42888_TXCTL_ADC_SNGVOL_MASK (1 << CS42888_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42888_TXCTL_ADC_SNGVOL (1 << CS42888_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SHIFT 0 +#define CS42888_TXCTL_ADC_SZC_MASK (((1 << CS42888_TXCTL_ADC_SZC_WIDTH) - 1) << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_IC (0 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_ZC (1 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SR (2 << CS42888_TXCTL_ADC_SZC_SHIFT) +#define CS42888_TXCTL_ADC_SZC_SRZC (3 << CS42888_TXCTL_ADC_SZC_SHIFT) + +/* DAC Channel Mute (Address 07h) */ +#define CS42888_DACMUTE_AOUT(n) (0x1 << n) +#define CS42888_DACMUTE_ALL 0xff + +/* Status Control (Address 18h)*/ +#define CS42888_STATUSCTL_INI_SHIFT 2 +#define CS42888_STATUSCTL_INI_WIDTH 2 +#define CS42888_STATUSCTL_INI_MASK (((1 << CS42888_STATUSCTL_INI_WIDTH) - 1) << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_ACTIVE_HIGH (0 << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_ACTIVE_LOW (1 << CS42888_STATUSCTL_INI_SHIFT) +#define CS42888_STATUSCTL_INT_OPEN_DRAIN (2 << CS42888_STATUSCTL_INI_SHIFT) + +/* Status (Address 19h)*/ +#define CS42888_STATUS_DAC_CLK_ERR_SHIFT 4 +#define CS42888_STATUS_DAC_CLK_ERR_MASK (1 << CS42888_STATUS_DAC_CLK_ERR_SHIFT) +#define CS42888_STATUS_ADC_CLK_ERR_SHIFT 3 +#define CS42888_STATUS_ADC_CLK_ERR_MASK (1 << CS42888_STATUS_ADC_CLK_ERR_SHIFT) +#define CS42888_STATUS_ADC2_OVFL_SHIFT 1 +#define CS42888_STATUS_ADC2_OVFL_MASK (1 << CS42888_STATUS_ADC2_OVFL_SHIFT) +#define CS42888_STATUS_ADC1_OVFL_SHIFT 0 +#define CS42888_STATUS_ADC1_OVFL_MASK (1 << CS42888_STATUS_ADC1_OVFL_SHIFT) + +/* Status Mask (Address 1Ah) */ +#define CS42888_STATUS_DAC_CLK_ERR_M_SHIFT 4 +#define CS42888_STATUS_DAC_CLK_ERR_M_MASK (1 << CS42888_STATUS_DAC_CLK_ERR_M_SHIFT) +#define CS42888_STATUS_ADC_CLK_ERR_M_SHIFT 3 +#define CS42888_STATUS_ADC_CLK_ERR_M_MASK (1 << CS42888_STATUS_ADC_CLK_ERR_M_SHIFT) +#define CS42888_STATUS_ADC2_OVFL_M_SHIFT 1 +#define CS42888_STATUS_ADC2_OVFL_M_MASK (1 << CS42888_STATUS_ADC2_OVFL_M_SHIFT) +#define CS42888_STATUS_ADC1_OVFL_M_SHIFT 0 +#define CS42888_STATUS_ADC1_OVFL_M_MASK (1 << CS42888_STATUS_ADC1_OVFL_M_SHIFT) + +/* MUTEC Pin Control (Address 1Bh) */ +#define CS42888_MUTEC_MCPOLARITY_SHIFT 1 +#define CS42888_MUTEC_MCPOLARITY_MASK (1 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MCPOLARITY_ACTIVE_LOW (0 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MCPOLARITY_ACTIVE_HIGH (1 << CS42888_MUTEC_MCPOLARITY_SHIFT) +#define CS42888_MUTEC_MUTEC_ACTIVE_SHIFT 0 +#define CS42888_MUTEC_MUTEC_ACTIVE_MASK (1 << CS42888_MUTEC_MUTEC_ACTIVE_SHIFT) +#define CS42888_MUTEC_MUTEC_ACTIVE (1 << CS42888_MUTEC_MUTEC_ACTIVE_SHIFT) +#endif /* _CS42888_H */ -- 1.8.4

5 19

[alsa-devel] [PATCH] ASOC: dapm: add param_fixup callback
by Nikesh＠alsa-project.org 25 Feb '14

25 Feb '14

From: nikesh <Nikesh.Oswal(a)wolfsonmicro.com> dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional callback to modify these params. Change-Id: Iad6ee3951bc4e8b8bc519c62642a2b4bcd949c18 Signed-off-by: nikesh <Nikesh.Oswal(a)wolfsonmicro.com> --- include/sound/soc-dapm.h | 7 ++++--- include/sound/soc.h | 4 +++- sound/soc/soc-core.c | 4 ++-- sound/soc/soc-dapm.c | 21 +++++++++++++++++---- 4 files changed, 26 insertions(+), 10 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 68d92e3..e3497ed 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -414,9 +414,10 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, - const struct snd_soc_pcm_stream *params, + struct snd_soc_pcm_stream *params, struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink); + struct snd_soc_dapm_widget *sink, + struct snd_soc_dai_link *dai_link); /* dapm path setup */ int snd_soc_dapm_new_widgets(struct snd_soc_card *card); @@ -561,7 +562,7 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ - const struct snd_soc_pcm_stream *params; /* params for dai links */ + struct snd_soc_pcm_stream *params; /* params for dai links */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 9a00147..401c911 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -875,7 +875,9 @@ struct snd_soc_dai_link { const struct device_node *platform_of_node; int be_id; /* optional ID for machine driver BE identification */ - const struct snd_soc_pcm_stream *params; + struct snd_soc_pcm_stream *params; + /* optional params re-writing for dai links */ + int (*params_fixup)(struct snd_soc_dapm_widget *w, int event); unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fe1df50..4f03e88 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1469,7 +1469,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + capture_w, play_w, dai_link); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1481,7 +1481,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + capture_w, play_w, dai_link); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dc8ff13..dc1ef8b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3378,11 +3378,12 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, { struct snd_soc_dapm_path *source_p, *sink_p; struct snd_soc_dai *source, *sink; - const struct snd_soc_pcm_stream *config = w->params; + struct snd_soc_pcm_stream *config = w->params; + struct snd_soc_dai_link *dai_link = w->priv; struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; u64 fmt; - int ret; + int ret = 0; if (WARN_ON(!config) || WARN_ON(list_empty(&w->sources) || list_empty(&w->sinks))) @@ -3402,6 +3403,16 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, source = source_p->source->priv; sink = sink_p->sink->priv; + if (dai_link && dai_link->params_fixup) { + ret = dai_link->params_fixup(w, event); + if (ret < 0) { + dev_err(w->dapm->dev, + "ASoC: params_fixup for dai link widget failed %d\n", + ret); + goto out; + } + } + /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3483,9 +3494,10 @@ out: } int snd_soc_dapm_new_pcm(struct snd_soc_card *card, - const struct snd_soc_pcm_stream *params, + struct snd_soc_pcm_stream *params, struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) + struct snd_soc_dapm_widget *sink, + struct snd_soc_dai_link *dai_link) { struct snd_soc_dapm_route routes[2]; struct snd_soc_dapm_widget template; @@ -3517,6 +3529,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, } w->params = params; + w->priv = (void *)dai_link; memset(&routes, 0, sizeof(routes)); -- 1.7.9.5

2 1

[alsa-devel] [PATCH v7] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 24 Feb '14

24 Feb '14

This patch adds initial support for the Behringer BCD2000 USB DJ controller. At the moment, only the MIDI part of the device is working, i.e. knobs, buttons and LEDs. I also plan to add support for the audio part, but I assume that this will require more effort than the rather simple MIDI interface. Progress can be tracked at https://github.com/anyc/snd-usb-bcd2000. Changes since v6: - applied more style improvements Changes since v5: - use kernel bitmap functions for devices_used Changes since v4: - devices_used as array to support arbitrary number of SNDRV_CARDS - removed unused array "enable" Changes since v3: - applied style and snd_printk changes as suggested by Daniel Mack Changes since v2: - applied more changes from Daniel Mack and Clemens Ladisch Changes since v1: - fixed the various code style issues, thanks to Daniel Mack and checkpatch.pl. Signed-off-by: Mario Kicherer <dev(a)kicherer.org> --- sound/usb/Kconfig | 13 ++ sound/usb/Makefile | 2 +- sound/usb/bcd2000/Makefile | 3 + sound/usb/bcd2000/bcd2000.c | 477 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 494 insertions(+), 1 deletion(-) create mode 100644 sound/usb/bcd2000/Makefile create mode 100644 sound/usb/bcd2000/bcd2000.c diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index de9408b..145653a 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -146,5 +146,18 @@ config SND_USB_HIFACE To compile this driver as a module, choose M here: the module will be called snd-usb-hiface. +config SND_BCD2000 + tristate "Behringer BCD2000 MIDI driver" + select SND_RAWMIDI + help + Say Y here to include MIDI support for the Behringer BCD2000 DJ + controller. + + Audio support is still work-in-progress at + https://github.com/anyc/snd-usb-bcd2000 + + To compile this driver as a module, choose M here: the module + will be called snd-bcd2000. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index abe668f..2b92f0d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ bcd2000/ diff --git a/sound/usb/bcd2000/Makefile b/sound/usb/bcd2000/Makefile new file mode 100644 index 0000000..f09ccc0 --- /dev/null +++ b/sound/usb/bcd2000/Makefile @@ -0,0 +1,3 @@ +snd-bcd2000-y := bcd2000.o + +obj-$(CONFIG_SND_BCD2000) += snd-bcd2000.o \ No newline at end of file diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c new file mode 100644 index 0000000..5085c58 --- /dev/null +++ b/sound/usb/bcd2000/bcd2000.c @@ -0,0 +1,477 @@ +/* + * Behringer BCD2000 driver + * + * Copyright (C) 2014 Mario Kicherer (dev(a)kicherer.org) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/bitmap.h> +#include <linux/usb.h> +#include <linux/usb/audio.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/rawmidi.h> + +#define PREFIX "snd-bcd2000: " +#define BUFSIZE 64 + +static struct usb_device_id id_table[] = { + { USB_DEVICE(0x1397, 0x00bd) }, + { }, +}; + +static unsigned char device_cmd_prefix[] = {0x03, 0x00}; + +static unsigned char bcd2000_init_sequence[] = { + 0x07, 0x00, 0x00, 0x00, 0x78, 0x48, 0x1c, 0x81, + 0xc4, 0x00, 0x00, 0x00, 0x5e, 0x53, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff, 0x6c, 0xf3, 0x90, 0xff, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, + 0x18, 0xfa, 0x11, 0xff, 0x14, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0xf2, 0x34, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff +}; + +struct bcd2000 { + struct usb_device *dev; + struct snd_card *card; + struct usb_interface *intf; + int card_index; + + int midi_out_active; + struct snd_rawmidi *rmidi; + struct snd_rawmidi_substream *midi_receive_substream; + struct snd_rawmidi_substream *midi_out_substream; + + unsigned char midi_in_buf[BUFSIZE]; + unsigned char midi_out_buf[BUFSIZE]; + + struct urb *midi_out_urb; + struct urb *midi_in_urb; + + struct usb_anchor anchor; +}; + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; + +static DEFINE_MUTEX(devices_mutex); +DECLARE_BITMAP(devices_used, SNDRV_CARDS); +static struct usb_driver bcd2000_driver; + +#ifdef CONFIG_SND_DEBUG +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) +{ + print_hex_dump(KERN_DEBUG, prefix, + DUMP_PREFIX_NONE, 16, 1, + buf, len, false); +} +#else +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) {} +#endif + +static int bcd2000_midi_input_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_input_close(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +/* register midi substream */ +static void bcd2000_midi_input_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + bcd2k->midi_receive_substream = up ? substream : NULL; +} + +static void bcd2000_midi_handle_input(struct bcd2000 *bcd2k, + const unsigned char *buf, unsigned int buf_len) +{ + unsigned int payload_length, tocopy; + struct snd_rawmidi_substream *midi_receive_substream; + + midi_receive_substream = ACCESS_ONCE(bcd2k->midi_receive_substream); + if (!midi_receive_substream) + return; + + bcd2000_dump_buffer(PREFIX "received from device: ", buf, buf_len); + + if (buf_len < 2) + return; + + payload_length = buf[0]; + + /* ignore packets without payload */ + if (payload_length == 0) + return; + + tocopy = min(payload_length, buf_len-1); + + bcd2000_dump_buffer(PREFIX "sending to userspace: ", + &buf[1], tocopy); + + snd_rawmidi_receive(midi_receive_substream, + &buf[1], tocopy); +} + +static void bcd2000_midi_send(struct bcd2000 *bcd2k) +{ + int len, ret; + struct snd_rawmidi_substream *midi_out_substream; + + BUILD_BUG_ON(sizeof(device_cmd_prefix) >= BUFSIZE); + + midi_out_substream = ACCESS_ONCE(bcd2k->midi_out_substream); + if (!midi_out_substream) + return; + + /* copy command prefix bytes */ + memcpy(bcd2k->midi_out_buf, device_cmd_prefix, + sizeof(device_cmd_prefix)); + + /* + * get MIDI packet and leave space for command prefix + * and payload length + */ + len = snd_rawmidi_transmit(midi_out_substream, + bcd2k->midi_out_buf + 3, BUFSIZE - 3); + + if (len < 0) + dev_err(&bcd2k->dev->dev, "%s: snd_rawmidi_transmit error %d\n", + __func__, len); + + if (len <= 0) + return; + + /* set payload length */ + bcd2k->midi_out_buf[2] = len; + bcd2k->midi_out_urb->transfer_buffer_length = BUFSIZE; + + bcd2000_dump_buffer(PREFIX "sending to device: ", + bcd2k->midi_out_buf, len+3); + + /* send packet to the BCD2000 */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s (%p): usb_submit_urb() failed, ret=%d, len=%d\n", + __func__, midi_out_substream, ret, len); + else + bcd2k->midi_out_active = 1; +} + +static int bcd2000_midi_output_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_output_close(struct snd_rawmidi_substream *substream) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (bcd2k->midi_out_active) { + usb_kill_urb(bcd2k->midi_out_urb); + bcd2k->midi_out_active = 0; + } + + return 0; +} + +/* register midi substream */ +static void bcd2000_midi_output_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (up) { + bcd2k->midi_out_substream = substream; + /* check if there is data userspace wants to send */ + if (!bcd2k->midi_out_active) + bcd2000_midi_send(bcd2k); + } else { + bcd2k->midi_out_substream = NULL; + } +} + +static void bcd2000_output_complete(struct urb *urb) +{ + struct bcd2000 *bcd2k = urb->context; + + bcd2k->midi_out_active = 0; + + if (urb->status) + dev_warn(&urb->dev->dev, + PREFIX "output urb->status: %d\n", urb->status); + + if (urb->status == -ESHUTDOWN) + return; + + /* check if there is more data userspace wants to send */ + bcd2000_midi_send(bcd2k); +} + +static void bcd2000_input_complete(struct urb *urb) +{ + int ret; + struct bcd2000 *bcd2k = urb->context; + + if (urb->status) + dev_warn(&urb->dev->dev, + PREFIX "input urb->status: %i\n", urb->status); + + if (!bcd2k || urb->status == -ESHUTDOWN) + return; + + if (urb->actual_length > 0) + bcd2000_midi_handle_input(bcd2k, urb->transfer_buffer, + urb->actual_length); + + /* acknowledge received packet */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() failed, ret=%d\n", + __func__, ret); +} + +static struct snd_rawmidi_ops bcd2000_midi_output = { + .open = bcd2000_midi_output_open, + .close = bcd2000_midi_output_close, + .trigger = bcd2000_midi_output_trigger, +}; + +static struct snd_rawmidi_ops bcd2000_midi_input = { + .open = bcd2000_midi_input_open, + .close = bcd2000_midi_input_close, + .trigger = bcd2000_midi_input_trigger, +}; + +static void bcd2000_init_device(struct bcd2000 *bcd2k) +{ + int ret; + + init_usb_anchor(&bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_out_urb, &bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_in_urb, &bcd2k->anchor); + + /* copy init sequence into buffer */ + memcpy(bcd2k->midi_out_buf, bcd2000_init_sequence, 52); + bcd2k->midi_out_urb->transfer_buffer_length = 52; + + /* submit sequence */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() out failed, ret=%d: ", + __func__, ret); + else + bcd2k->midi_out_active = 1; + + /* send empty packet to enable button and controller events */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() in failed, ret=%d: ", + __func__, ret); + + /* ensure initialization is finished */ + usb_wait_anchor_empty_timeout(&bcd2k->anchor, 1000); +} + +static int bcd2000_init_midi(struct bcd2000 *bcd2k) +{ + int ret; + struct snd_rawmidi *rmidi; + + ret = snd_rawmidi_new(bcd2k->card, bcd2k->card->shortname, 0, + 1, /* output */ + 1, /* input */ + &rmidi); + + if (ret < 0) + return ret; + + strlcpy(rmidi->name, bcd2k->card->shortname, sizeof(rmidi->name)); + + rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; + rmidi->private_data = bcd2k; + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &bcd2000_midi_output); + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &bcd2000_midi_input); + + bcd2k->rmidi = rmidi; + + bcd2k->midi_in_urb = usb_alloc_urb(0, GFP_KERNEL); + bcd2k->midi_out_urb = usb_alloc_urb(0, GFP_KERNEL); + + if (!bcd2k->midi_in_urb || !bcd2k->midi_out_urb) { + dev_err(&bcd2k->dev->dev, PREFIX "usb_alloc_urb failed\n"); + return -ENOMEM; + } + + usb_fill_int_urb(bcd2k->midi_in_urb, bcd2k->dev, + usb_rcvintpipe(bcd2k->dev, 0x81), + bcd2k->midi_in_buf, BUFSIZE, + bcd2000_input_complete, bcd2k, 1); + + usb_fill_int_urb(bcd2k->midi_out_urb, bcd2k->dev, + usb_sndintpipe(bcd2k->dev, 0x1), + bcd2k->midi_out_buf, BUFSIZE, + bcd2000_output_complete, bcd2k, 1); + + bcd2000_init_device(bcd2k); + + return 0; +} + +static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k, + struct usb_interface *interface) +{ + /* usb_kill_urb not necessary, urb is aborted automatically */ + + usb_free_urb(bcd2k->midi_out_urb); + usb_free_urb(bcd2k->midi_in_urb); + + if (bcd2k->intf) { + usb_set_intfdata(bcd2k->intf, NULL); + bcd2k->intf = NULL; + } +} + +static int bcd2000_probe(struct usb_interface *interface, + const struct usb_device_id *usb_id) +{ + struct snd_card *card; + struct bcd2000 *bcd2k; + unsigned int card_index; + char usb_path[32]; + int err; + + mutex_lock(&devices_mutex); + + for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) + if (!test_bit(card_index, devices_used)) + break; + + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + + err = snd_card_create(index[card_index], id[card_index], THIS_MODULE, + sizeof(*bcd2k), &card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return err; + } + + bcd2k = card->private_data; + bcd2k->dev = interface_to_usbdev(interface); + bcd2k->card = card; + bcd2k->card_index = card_index; + bcd2k->intf = interface; + + snd_card_set_dev(card, &interface->dev); + + strncpy(card->driver, "snd-bcd2000", sizeof(card->driver)); + strncpy(card->shortname, "BCD2000", sizeof(card->shortname)); + usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); + snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), + "Behringer BCD2000 at %s", + usb_path); + + err = bcd2000_init_midi(bcd2k); + if (err < 0) + goto probe_error; + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + usb_set_intfdata(interface, bcd2k); + set_bit(card_index, devices_used); + + mutex_unlock(&devices_mutex); + return 0; + +probe_error: + dev_info(&bcd2k->dev->dev, PREFIX "error during probing"); + bcd2000_free_usb_related_resources(bcd2k, interface); + snd_card_free(card); + mutex_unlock(&devices_mutex); + return err; +} + +static void bcd2000_disconnect(struct usb_interface *interface) +{ + struct bcd2000 *bcd2k = usb_get_intfdata(interface); + + if (!bcd2k) + return; + + mutex_lock(&devices_mutex); + + /* make sure that userspace cannot create new requests */ + snd_card_disconnect(bcd2k->card); + + bcd2000_free_usb_related_resources(bcd2k, interface); + + clear_bit(bcd2k->card_index, devices_used); + + snd_card_free_when_closed(bcd2k->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_driver bcd2000_driver = { + .name = "snd-bcd2000", + .probe = bcd2000_probe, + .disconnect = bcd2000_disconnect, + .id_table = id_table, +}; + +static int __init bcd2000_init(void) +{ + int retval = 0; + + retval = usb_register(&bcd2000_driver); + if (retval) + pr_info(PREFIX "usb_register failed. Error: %d", retval); + return retval; +} + +static void __exit bcd2000_exit(void) +{ + usb_deregister(&bcd2000_driver); +} + +module_init(bcd2000_init); +module_exit(bcd2000_exit); + +MODULE_DEVICE_TABLE(usb, id_table); +MODULE_AUTHOR("Mario Kicherer, dev(a)kicherer.org"); +MODULE_DESCRIPTION("Behringer BCD2000 driver"); +MODULE_LICENSE("GPL"); -- 1.8.3.2

1 0

[alsa-devel] [PATCH v6] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 24 Feb '14

24 Feb '14

This patch adds initial support for the Behringer BCD2000 USB DJ controller. At the moment, only the MIDI part of the device is working, i.e. knobs, buttons and LEDs. I also plan to add support for the audio part, but I assume that this will require more effort than the rather simple MIDI interface. Progress can be tracked at https://github.com/anyc/snd-usb-bcd2000. Changes since v5: - use kernel bitmap functions for devices_used Changes since v4: - devices_used as array to support arbitrary number of SNDRV_CARDS - removed unused array "enable" Changes since v3: - applied style and snd_printk changes as suggested by Daniel Mack Changes since v2: - applied more changes from Daniel Mack and Clemens Ladisch Changes since v1: - fixed the various code style issues, thanks to Daniel Mack and checkpatch.pl. Signed-off-by: Mario Kicherer <dev(a)kicherer.org> --- sound/usb/Kconfig | 13 ++ sound/usb/Makefile | 2 +- sound/usb/bcd2000/Makefile | 3 + sound/usb/bcd2000/bcd2000.c | 488 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 505 insertions(+), 1 deletion(-) create mode 100644 sound/usb/bcd2000/Makefile create mode 100644 sound/usb/bcd2000/bcd2000.c diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index de9408b..145653a 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -146,5 +146,18 @@ config SND_USB_HIFACE To compile this driver as a module, choose M here: the module will be called snd-usb-hiface. +config SND_BCD2000 + tristate "Behringer BCD2000 MIDI driver" + select SND_RAWMIDI + help + Say Y here to include MIDI support for the Behringer BCD2000 DJ + controller. + + Audio support is still work-in-progress at + https://github.com/anyc/snd-usb-bcd2000 + + To compile this driver as a module, choose M here: the module + will be called snd-bcd2000. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index abe668f..2b92f0d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ bcd2000/ diff --git a/sound/usb/bcd2000/Makefile b/sound/usb/bcd2000/Makefile new file mode 100644 index 0000000..f09ccc0 --- /dev/null +++ b/sound/usb/bcd2000/Makefile @@ -0,0 +1,3 @@ +snd-bcd2000-y := bcd2000.o + +obj-$(CONFIG_SND_BCD2000) += snd-bcd2000.o \ No newline at end of file diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c new file mode 100644 index 0000000..82d5e9d --- /dev/null +++ b/sound/usb/bcd2000/bcd2000.c @@ -0,0 +1,488 @@ +/* + * Behringer BCD2000 driver + * + * Copyright (C) 2014 Mario Kicherer (dev(a)kicherer.org) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/bitmap.h> +#include <linux/usb.h> +#include <linux/usb/audio.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/rawmidi.h> + +#define DEVICE_NAME "Behringer BCD2000" +#define DEVICE_SHORTNAME "bcd2000" + +#define PREFIX DEVICE_SHORTNAME ": " +#define BUFSIZE 64 + +static struct usb_device_id id_table[] = { + { USB_DEVICE(0x1397, 0x00bd) }, + { }, +}; + +static unsigned char device_cmd_prefix[] = {0x03, 0x00}; + +static unsigned char bcd2000_init_sequence[] = { + 0x07, 0x00, 0x00, 0x00, 0x78, 0x48, 0x1c, 0x81, + 0xc4, 0x00, 0x00, 0x00, 0x5e, 0x53, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff, 0x6c, 0xf3, 0x90, 0xff, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, + 0x18, 0xfa, 0x11, 0xff, 0x14, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0xf2, 0x34, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff +}; + +struct bcd2000 { + struct usb_device *dev; + struct snd_card *card; + struct usb_interface *intf; + int card_index; + + int midi_out_active; + struct snd_rawmidi *rmidi; + struct snd_rawmidi_substream *midi_receive_substream; + struct snd_rawmidi_substream *midi_out_substream; + + unsigned char midi_in_buf[BUFSIZE]; + unsigned char midi_out_buf[BUFSIZE]; + + struct urb *midi_out_urb; + struct urb *midi_in_urb; + + struct usb_anchor anchor; +}; + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; + +static DEFINE_MUTEX(devices_mutex); +DECLARE_BITMAP(devices_used, SNDRV_CARDS); +static struct usb_driver bcd2000_driver; + +#ifdef CONFIG_SND_DEBUG +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) +{ + print_hex_dump(KERN_DEBUG, prefix, + DUMP_PREFIX_NONE, 16, 1, + buf, len, false); +} +#else +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) {} +#endif + +static int bcd2000_midi_input_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_input_close(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +/* register midi substream */ +static void bcd2000_midi_input_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + bcd2k->midi_receive_substream = up ? substream : NULL; +} + +static void bcd2000_midi_handle_input(struct bcd2000 *bcd2k, + const unsigned char *buf, unsigned int len) +{ + unsigned int length, tocopy; + struct snd_rawmidi_substream *midi_receive_substream; + + midi_receive_substream = ACCESS_ONCE(bcd2k->midi_receive_substream); + if (!midi_receive_substream) + return; + + bcd2000_dump_buffer(PREFIX "received from device: ", buf, len); + + if (len < 2) + return; + + /* + * Packet structure: mm nn oo (pp) + * mm: payload length + * nn: MIDI command or note + * oo: note or velocity + * pp: velocity + */ + + length = buf[0]; + + /* ignore packets without payload */ + if (length == 0) + return; + + tocopy = min(length, len-1); + + bcd2000_dump_buffer(PREFIX "sending to userspace: ", + &buf[1], tocopy); + + snd_rawmidi_receive(midi_receive_substream, + &buf[1], tocopy); +} + +static void bcd2000_midi_send(struct bcd2000 *bcd2k, + struct snd_rawmidi_substream *substream) +{ + int len, ret; + + BUILD_BUG_ON(sizeof(device_cmd_prefix) >= BUFSIZE); + + /* copy the "set LED" command bytes */ + memcpy(bcd2k->midi_out_buf, device_cmd_prefix, + sizeof(device_cmd_prefix)); + + /* + * get MIDI packet and leave space for command prefix + * and payload length + */ + len = snd_rawmidi_transmit(substream, + bcd2k->midi_out_buf + 3, BUFSIZE - 3); + + if (len < 0) + dev_err(&bcd2k->dev->dev, "%s: snd_rawmidi_transmit error %d\n", + __func__, len); + + if (len <= 0) + return; + + /* set payload length */ + bcd2k->midi_out_buf[2] = len; + bcd2k->midi_out_urb->transfer_buffer_length = BUFSIZE; + + bcd2000_dump_buffer(PREFIX "sending to device: ", + bcd2k->midi_out_buf, len+3); + + /* send packet to the BCD2000 */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s (%p): usb_submit_urb() failed, ret=%d, len=%d\n", + __func__, substream, ret, len); + else + bcd2k->midi_out_active = 1; +} + +static int bcd2000_midi_output_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_output_close(struct snd_rawmidi_substream *substream) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (bcd2k->midi_out_active) { + usb_kill_urb(bcd2k->midi_out_urb); + bcd2k->midi_out_active = 0; + } + + return 0; +} + +/* register midi substream */ +static void bcd2000_midi_output_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (up) { + bcd2k->midi_out_substream = substream; + /* check if there is data userspace wants to send */ + if (!bcd2k->midi_out_active) + bcd2000_midi_send(bcd2k, substream); + } else { + bcd2k->midi_out_substream = NULL; + } +} + +static void bcd2000_output_complete(struct urb *urb) +{ + struct bcd2000 *bcd2k = urb->context; + struct snd_rawmidi_substream *midi_out_substream; + + bcd2k->midi_out_active = 0; + + if (urb->status != 0) + dev_err(&urb->dev->dev, + PREFIX "output urb->status: %d\n", urb->status); + + midi_out_substream = ACCESS_ONCE(bcd2k->midi_out_substream); + if (!midi_out_substream) + return; + + /* check if there is more data userspace wants to send */ + bcd2000_midi_send(bcd2k, midi_out_substream); +} + +static void bcd2000_input_complete(struct urb *urb) +{ + int ret; + struct device *dev = &urb->dev->dev; + struct bcd2000 *bcd2k = urb->context; + + if (urb->status) { + dev_warn(dev, PREFIX "input urb->status: %i\n", urb->status); + return; + } + + if (!bcd2k) + return; + + if (urb->actual_length > 0) + bcd2000_midi_handle_input(bcd2k, urb->transfer_buffer, + urb->actual_length); + + /* acknowledge received packet */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(dev, "unable to submit urb. OOM!?\n"); +} + +static struct snd_rawmidi_ops bcd2000_midi_output = { + .open = bcd2000_midi_output_open, + .close = bcd2000_midi_output_close, + .trigger = bcd2000_midi_output_trigger, +}; + +static struct snd_rawmidi_ops bcd2000_midi_input = { + .open = bcd2000_midi_input_open, + .close = bcd2000_midi_input_close, + .trigger = bcd2000_midi_input_trigger, +}; + +static void bcd2000_init_device(struct bcd2000 *bcd2k) +{ + int ret; + + init_usb_anchor(&bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_out_urb, &bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_in_urb, &bcd2k->anchor); + + /* copy init sequence into buffer */ + memcpy(bcd2k->midi_out_buf, bcd2000_init_sequence, 52); + bcd2k->midi_out_urb->transfer_buffer_length = 52; + + /* submit sequence */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() out failed, ret=%d: ", + __func__, ret); + else + bcd2k->midi_out_active = 1; + + /* send empty packet to enable button and controller events */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() in failed, ret=%d: ", + __func__, ret); + + /* ensure initialization is finished */ + usb_wait_anchor_empty_timeout(&bcd2k->anchor, 1000); +} + +static int bcd2000_init_midi(struct bcd2000 *bcd2k) +{ + int ret; + struct snd_rawmidi *rmidi; + + ret = snd_rawmidi_new(bcd2k->card, bcd2k->card->shortname, 0, + 1, /* output */ + 1, /* input */ + &rmidi); + + if (ret < 0) + return ret; + + strlcpy(rmidi->name, bcd2k->card->shortname, sizeof(rmidi->name)); + + rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; + rmidi->private_data = bcd2k; + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &bcd2000_midi_output); + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &bcd2000_midi_input); + + bcd2k->rmidi = rmidi; + + bcd2k->midi_in_urb = usb_alloc_urb(0, GFP_KERNEL); + bcd2k->midi_out_urb = usb_alloc_urb(0, GFP_KERNEL); + + if (!bcd2k->midi_in_urb || !bcd2k->midi_out_urb) { + dev_err(&bcd2k->dev->dev, PREFIX "usb_alloc_urb failed\n"); + return -ENOMEM; + } + + usb_fill_int_urb(bcd2k->midi_in_urb, bcd2k->dev, + usb_rcvintpipe(bcd2k->dev, 0x81), + bcd2k->midi_in_buf, BUFSIZE, + bcd2000_input_complete, bcd2k, 1); + + usb_fill_int_urb(bcd2k->midi_out_urb, bcd2k->dev, + usb_sndintpipe(bcd2k->dev, 0x1), + bcd2k->midi_out_buf, BUFSIZE, + bcd2000_output_complete, bcd2k, 1); + + bcd2000_init_device(bcd2k); + + return 0; +} + +static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k, + struct usb_interface *interface) +{ + /* usb_kill_urb not necessary, urb is aborted automatically */ + + usb_free_urb(bcd2k->midi_out_urb); + usb_free_urb(bcd2k->midi_in_urb); + + if (bcd2k->intf) { + usb_set_intfdata(bcd2k->intf, NULL); + bcd2k->intf = NULL; + } +} + +static int bcd2000_probe(struct usb_interface *interface, + const struct usb_device_id *usb_id) +{ + struct snd_card *card; + struct bcd2000 *bcd2k; + unsigned int card_index; + char usb_path[32]; + int err; + + mutex_lock(&devices_mutex); + + for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) + if (!test_bit(card_index, devices_used)) + break; + + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + + err = snd_card_create(index[card_index], id[card_index], THIS_MODULE, + sizeof(*bcd2k), &card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return err; + } + + bcd2k = card->private_data; + bcd2k->dev = interface_to_usbdev(interface); + bcd2k->card = card; + bcd2k->card_index = card_index; + bcd2k->intf = interface; + + snd_card_set_dev(card, &interface->dev); + + strcpy(card->driver, DEVICE_NAME); + strcpy(card->shortname, DEVICE_SHORTNAME); + usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); + snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), + DEVICE_NAME ", at %s", + usb_path); + + dev_info(&bcd2k->dev->dev, PREFIX "%s", bcd2k->card->longname); + + err = bcd2000_init_midi(bcd2k); + if (err < 0) + goto probe_error; + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + usb_set_intfdata(interface, bcd2k); + set_bit(card_index, devices_used); + + mutex_unlock(&devices_mutex); + return 0; + +probe_error: + dev_info(&bcd2k->dev->dev, PREFIX "error during probing"); + bcd2000_free_usb_related_resources(bcd2k, interface); + snd_card_free(card); + mutex_unlock(&devices_mutex); + return err; +} + +static void bcd2000_disconnect(struct usb_interface *interface) +{ + struct bcd2000 *bcd2k = usb_get_intfdata(interface); + + if (!bcd2k) + return; + + mutex_lock(&devices_mutex); + + /* make sure that userspace cannot create new requests */ + snd_card_disconnect(bcd2k->card); + + bcd2000_free_usb_related_resources(bcd2k, interface); + + clear_bit(bcd2k->card_index, devices_used); + + snd_card_free_when_closed(bcd2k->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_driver bcd2000_driver = { + .name = DEVICE_SHORTNAME, + .probe = bcd2000_probe, + .disconnect = bcd2000_disconnect, + .id_table = id_table, +}; + +static int __init bcd2000_init(void) +{ + int retval = 0; + + retval = usb_register(&bcd2000_driver); + if (retval) + pr_info(PREFIX "usb_register failed. Error: %d", retval); + return retval; +} + +static void __exit bcd2000_exit(void) +{ + usb_deregister(&bcd2000_driver); +} + +module_init(bcd2000_init); +module_exit(bcd2000_exit); + +MODULE_DEVICE_TABLE(usb, id_table); +MODULE_AUTHOR("Mario Kicherer, dev(a)kicherer.org"); +MODULE_DESCRIPTION("Behringer BCD2000 driver"); +MODULE_LICENSE("GPL"); -- 1.8.3.2

2 3

[alsa-devel] [PATCH 1/6] ASoC: Intel: Add support for Haswell/Broadwell DSP
by Liam Girdwood 24 Feb '14

24 Feb '14

Add support for low level differentiation functions for Haswell and Broadwell SST DSPs. This includes suppoprt for DSP boot and reset, DSP firmware module parsing and DSP memory block map initialisation. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-dsp.c | 60 +++++ sound/soc/intel/sst-dsp.h | 10 + sound/soc/intel/sst-haswell-dsp.c | 517 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 587 insertions(+) create mode 100644 sound/soc/intel/sst-haswell-dsp.c diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c index 6e22c12..0c129fd 100644 --- a/sound/soc/intel/sst-dsp.c +++ b/sound/soc/intel/sst-dsp.c @@ -27,6 +27,66 @@ #define CREATE_TRACE_POINTS #include <trace/events/intel-sst.h> +/* Internal generic low-level SST IO functions - can be overidden */ +void sst_shim32_write(void __iomem *addr, u32 offset, u32 value) +{ + writel(value, addr + offset); +} +EXPORT_SYMBOL_GPL(sst_shim32_write); + +u32 sst_shim32_read(void __iomem *addr, u32 offset) +{ + return readl(addr + offset); +} +EXPORT_SYMBOL_GPL(sst_shim32_read); + +void sst_shim32_write64(void __iomem *addr, u32 offset, u64 value) +{ + memcpy_toio(addr + offset, &value, sizeof(value)); +} +EXPORT_SYMBOL_GPL(sst_shim32_write64); + +u64 sst_shim32_read64(void __iomem *addr, u32 offset) +{ + u64 val; + + memcpy_fromio(&val, addr + offset, sizeof(val)); + return val; +} +EXPORT_SYMBOL_GPL(sst_shim32_read64); + +static inline void _sst_memcpy_toio_32(volatile u32 __iomem *dest, + u32 *src, size_t bytes) +{ + int i, words = bytes >> 2; + + for (i = 0; i < words; i++) + writel(src[i], dest + i); +} + +static inline void _sst_memcpy_fromio_32(u32 *dest, + const volatile __iomem u32 *src, size_t bytes) +{ + int i, words = bytes >> 2; + + for (i = 0; i < words; i++) + dest[i] = readl(src + i); +} + +void sst_memcpy_toio_32(struct sst_dsp *sst, + void __iomem *dest, void *src, size_t bytes) +{ + _sst_memcpy_toio_32(dest, src, bytes); +} +EXPORT_SYMBOL_GPL(sst_memcpy_toio_32); + +void sst_memcpy_fromio_32(struct sst_dsp *sst, void *dest, + void __iomem *src, size_t bytes) +{ + _sst_memcpy_fromio_32(dest, src, bytes); +} +EXPORT_SYMBOL_GPL(sst_memcpy_fromio_32); + /* Public API */ void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value) { diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index 3730fd3..608418c 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -189,6 +189,16 @@ u64 sst_dsp_shim_read64_unlocked(struct sst_dsp *sst, u32 offset); int sst_dsp_shim_update_bits64_unlocked(struct sst_dsp *sst, u32 offset, u64 mask, u64 value); +/* Internal generic low-level SST IO functions - can be overidden */ +void sst_shim32_write(void __iomem *addr, u32 offset, u32 value); +u32 sst_shim32_read(void __iomem *addr, u32 offset); +void sst_shim32_write64(void __iomem *addr, u32 offset, u64 value); +u64 sst_shim32_read64(void __iomem *addr, u32 offset); +void sst_memcpy_toio_32(struct sst_dsp *sst, + void __iomem *dest, void *src, size_t bytes); +void sst_memcpy_fromio_32(struct sst_dsp *sst, + void *dest, void __iomem *src, size_t bytes); + /* DSP reset & boot */ void sst_dsp_reset(struct sst_dsp *sst); int sst_dsp_boot(struct sst_dsp *sst); diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c new file mode 100644 index 0000000..12f7317 --- /dev/null +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -0,0 +1,517 @@ +/* + * Intel Haswell SST DSP driver + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include <linux/delay.h> +#include <linux/fs.h> +#include <linux/slab.h> +#include <linux/device.h> +#include <linux/sched.h> +#include <linux/export.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/dma-mapping.h> +#include <linux/platform_device.h> +#include <linux/pci.h> +#include <linux/firmware.h> +#include <linux/pm_runtime.h> + +#include <linux/acpi.h> +#include <acpi/acpi_bus.h> + +#include "sst-dsp.h" +#include "sst-dsp-priv.h" +#include "sst-haswell-ipc.h" + +#include <trace/events/hswadsp.h> + +#define SST_HSW_FW_SIGNATURE_SIZE 4 +#define SST_HSW_FW_SIGN "$SST" +#define SST_HSW_FW_LIB_SIGN "$LIB" + +#define SST_WPT_SHIM_OFFSET 0xFB000 +#define SST_LP_SHIM_OFFSET 0xE7000 +#define SST_WPT_IRAM_OFFSET 0xA0000 +#define SST_LP_IRAM_OFFSET 0x80000 + +#define SST_SHIM_PM_REG 0x84 + +#define SST_HSW_IRAM 1 +#define SST_HSW_DRAM 2 +#define SST_HSW_REGS 3 + +struct dma_block_info { + __le32 type; /* IRAM/DRAM */ + __le32 size; /* Bytes */ + __le32 ram_offset; /* Offset in I/DRAM */ + __le32 rsvd; /* Reserved field */ +} __attribute__((packed)); + +struct fw_module_info { + __le32 persistent_size; + __le32 scratch_size; +} __attribute__((packed)); + +struct fw_header { + unsigned char signature[SST_HSW_FW_SIGNATURE_SIZE]; /* FW signature */ + __le32 file_size; /* size of fw minus this header */ + __le32 modules; /* # of modules */ + __le32 file_format; /* version of header format */ + __le32 reserved[4]; +} __attribute__((packed)); + +struct fw_module_header { + unsigned char signature[SST_HSW_FW_SIGNATURE_SIZE]; /* module signature */ + __le32 mod_size; /* size of module */ + __le32 blocks; /* # of blocks */ + __le16 padding; + __le16 type; /* codec type, pp lib */ + __le32 entry_point; + struct fw_module_info info; +} __attribute__((packed)); + +static void hsw_free(struct sst_dsp *sst); + +static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, + struct fw_module_header *module) +{ + struct dma_block_info *block; + struct sst_module *mod; + struct sst_module_data block_data; + struct sst_module_template template; + int count; + void __iomem *ram; + + /* TODO: allowed module types need to be configurable */ + if (module->type != SST_HSW_MODULE_BASE_FW + && module->type != SST_HSW_MODULE_PCM_SYSTEM + && module->type != SST_HSW_MODULE_PCM + && module->type != SST_HSW_MODULE_PCM_REFERENCE + && module->type != SST_HSW_MODULE_PCM_CAPTURE + && module->type != SST_HSW_MODULE_LPAL) + return 0; + + dev_dbg(dsp->dev, "new module sign 0x%s size 0x%x blocks 0x%x type 0x%x\n", + module->signature, module->mod_size, + module->blocks, module->type); + dev_dbg(dsp->dev, " entrypoint 0x%x\n", module->entry_point); + dev_dbg(dsp->dev, " persistent 0x%x scratch 0x%x\n", + module->info.persistent_size, module->info.scratch_size); + + memset(&template, 0, sizeof(template)); + template.id = module->type; + template.entry = module->entry_point; + template.p.size = module->info.persistent_size; + template.p.type = SST_MEM_DRAM; + template.p.data_type = SST_DATA_P; + template.s.size = module->info.scratch_size; + template.s.type = SST_MEM_DRAM; + template.s.data_type = SST_DATA_S; + + mod = sst_module_new(fw, &template, NULL); + if (mod == NULL) + return -ENOMEM; + + block = (void *)module + sizeof(*module); + + for (count = 0; count < module->blocks; count++) { + + if (block->size <= 0) { + dev_err(dsp->dev, + "error: block %d size invalid\n", count); + sst_module_free(mod); + return -EINVAL; + } + + switch (block->type) { + case SST_HSW_IRAM: + ram = dsp->addr.lpe; + block_data.offset = + block->ram_offset + dsp->addr.iram_offset; + block_data.type = SST_MEM_IRAM; + break; + case SST_HSW_DRAM: + ram = dsp->addr.lpe; + block_data.offset = block->ram_offset; + block_data.type = SST_MEM_DRAM; + break; + default: + dev_err(dsp->dev, "error: bad type 0x%x for block 0x%x\n", + block->type, count); + sst_module_free(mod); + return -EINVAL; + } + + block_data.size = block->size; + block_data.data_type = SST_DATA_M; + block_data.data = (void *)block + sizeof(*block); + block_data.data_offset = block_data.data - fw->dma_buf; + + dev_dbg(dsp->dev, "copy firmware block %d type 0x%x " + "size 0x%x ==> ram %p offset 0x%x\n", + count, block->type, block->size, ram, + block->ram_offset); + + sst_module_insert_fixed_block(mod, &block_data); + + block = (void *)block + sizeof(*block) + block->size; + } + return 0; +} + +static int hsw_parse_fw_image(struct sst_fw *sst_fw) +{ + struct fw_header *header; + struct sst_module *scratch; + struct fw_module_header *module; + struct sst_dsp *dsp = sst_fw->dsp; + struct sst_hsw *hsw = sst_fw->private; + int ret, count; + + /* Read the header information from the data pointer */ + header = (struct fw_header *)sst_fw->dma_buf; + + /* verify FW */ + if ((strncmp(header->signature, SST_HSW_FW_SIGN, 4) != 0) || + (sst_fw->size != header->file_size + sizeof(*header))) { + dev_err(dsp->dev, "error: invalid fw sign/filesize mismatch\n"); + return -EINVAL; + } + + dev_dbg(dsp->dev, "header size=0x%x modules=0x%x fmt=0x%x size=%zu\n", + header->file_size, header->modules, + header->file_format, sizeof(*header)); + + /* parse each module */ + module = (void *)sst_fw->dma_buf + sizeof(*header); + for (count = 0; count < header->modules; count++) { + + /* module */ + ret = hsw_parse_module(dsp, sst_fw, module); + if (ret < 0) { + dev_err(dsp->dev, "error: invalid module %d\n", count); + return ret; + } + module = (void *)module + sizeof(*module) + module->mod_size; + } + + /* allocate persistent/scratch mem regions */ + scratch = sst_mem_block_alloc_scratch(dsp); + if (scratch == NULL) + return -ENOMEM; + + sst_hsw_set_scratch_module(hsw, scratch); + + return 0; +} + +static irqreturn_t hsw_irq(int irq, void *context) +{ + struct sst_dsp *sst = (struct sst_dsp *) context; + u32 isr; + int ret = IRQ_NONE; + + spin_lock(&sst->spinlock); + + /* Interrupt arrived, check src */ + isr = sst_dsp_shim_read_unlocked(sst, SST_ISRX); + if (isr & SST_ISRX_DONE) { + trace_sst_irq_done(isr, + sst_dsp_shim_read_unlocked(sst, SST_IMRX)); + + /* Mask Done interrupt before return */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, + SST_IMRX_DONE, SST_IMRX_DONE); + ret = IRQ_WAKE_THREAD; + } + + if (isr & SST_ISRX_BUSY) { + trace_sst_irq_busy(isr, + sst_dsp_shim_read_unlocked(sst, SST_IMRX)); + + /* Mask Busy interrupt before return */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, + SST_IMRX_BUSY, SST_IMRX_BUSY); + ret = IRQ_WAKE_THREAD; + } + + spin_unlock(&sst->spinlock); + return ret; +} + +static void hsw_boot(struct sst_dsp *sst) +{ + /* select SSP1 19.2MHz base clock, SSP clock 0, turn off Low Power Clock */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_S1IOCS | SST_CSR_SBCS1 | SST_CSR_LPCS, 0x0); + + /* stall DSP core, set clk to 192/96Mhz */ + sst_dsp_shim_update_bits_unlocked(sst, + SST_CSR, SST_CSR_STALL | SST_CSR_DCS_MASK, + SST_CSR_STALL | SST_CSR_DCS(4)); + + /* Set 24MHz MCLK, prevent local clock gating, enable SSP0 clock */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL, + SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0, + SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0); + + /* disable DMA finish function for SSP0 & SSP1 */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR2, SST_CSR2_SDFD_SSP1, + SST_CSR2_SDFD_SSP1); + + /* enable DMA engine 0,1 all channels to access host memory */ + sst_dsp_shim_update_bits_unlocked(sst, SST_HDMC, + SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff), + SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff)); + + /* disable all clock gating */ + writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2); + + /* set DSP to RUN */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_STALL, 0x0); +} + +static void hsw_reset(struct sst_dsp *sst) +{ + /* put DSP into reset and stall */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_RST | SST_CSR_STALL, SST_CSR_RST | SST_CSR_STALL); + + /* keep in reset for 10ms */ + mdelay(10); + + /* take DSP out of reset and keep stalled for FW loading */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_RST | SST_CSR_STALL, SST_CSR_STALL); +} + +struct sst_adsp_memregion { + u32 start; + u32 end; + int blocks; + enum sst_mem_type type; +}; + +/* lynx point ADSP mem regions */ +static const struct sst_adsp_memregion lp_region[] = { + {0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */ + {0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */ + {0x80000, 0xE0000, 12, SST_MEM_IRAM}, /* I-SRAM - 12 * 32kB */ +}; + +/* wild cat point ADSP mem regions */ +static const struct sst_adsp_memregion wpt_region[] = { + {0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */ + {0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */ + {0x80000, 0xA0000, 4, SST_MEM_DRAM}, /* D-SRAM2 - 4 * 32kB */ + {0xA0000, 0xF0000, 10, SST_MEM_IRAM}, /* I-SRAM - 10 * 32kB */ +}; + +static int hsw_acpi_resource_map(struct sst_dsp *sst, struct sst_pdata *pdata) +{ + /* ADSP DRAM & IRAM */ + sst->addr.lpe_base = pdata->lpe_base; + sst->addr.lpe = ioremap(pdata->lpe_base, pdata->lpe_size); + if (!sst->addr.lpe) + return -ENODEV; + + /* ADSP PCI MMIO config space */ + sst->addr.pci_cfg = ioremap(pdata->pcicfg_base, pdata->pcicfg_size); + if (!sst->addr.pci_cfg) { + iounmap(sst->addr.lpe); + return -ENODEV; + } + + /* SST Shim */ + sst->addr.shim = sst->addr.lpe + sst->addr.shim_offset; + return 0; +} + +static u32 hsw_block_get_bit(struct sst_mem_block *block) +{ + u32 bit = 0, shift = 0; + + switch (block->type) { + case SST_MEM_DRAM: + shift = 16; + break; + case SST_MEM_IRAM: + shift = 6; + break; + default: + return 0; + } + + bit = 1 << (block->index + shift); + + return bit; +} + +/* enable 32kB memory block - locks held by caller */ +static int hsw_block_enable(struct sst_mem_block *block) +{ + struct sst_dsp *sst = block->dsp; + u32 bit, val; + + if (block->users++ > 0) + return 0; + + dev_dbg(block->dsp->dev, " enabled block %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); + + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + bit = hsw_block_get_bit(block); + writel(val & ~bit, sst->addr.pci_cfg + SST_VDRTCTL0); + + /* wait 18 DSP clock ticks */ + udelay(10); + + return 0; +} + +/* disable 32kB memory block - locks held by caller */ +static int hsw_block_disable(struct sst_mem_block *block) +{ + struct sst_dsp *sst = block->dsp; + u32 bit, val; + + if (--block->users > 0) + return 0; + + dev_dbg(block->dsp->dev, " disabled block %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); + + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + bit = hsw_block_get_bit(block); + writel(val | bit, sst->addr.pci_cfg + SST_VDRTCTL0); + + return 0; +} + +static struct sst_block_ops sst_hsw_ops = { + .enable = hsw_block_enable, + .disable = hsw_block_disable, +}; + +static int hsw_enable_shim(struct sst_dsp *sst) +{ + int tries = 10; + u32 reg; + + /* enable shim */ + reg = readl(sst->addr.pci_cfg + SST_SHIM_PM_REG); + writel(reg & ~0x3, sst->addr.pci_cfg + SST_SHIM_PM_REG); + + /* check that ADSP shim is enabled */ + while (tries--) { + reg = sst_dsp_shim_read_unlocked(sst, SST_CSR); + if (reg != 0xffffffff) + return 0; + + msleep(1); + } + + return -ENODEV; +} + +static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) +{ + const struct sst_adsp_memregion *region; + struct device *dev; + int ret = -ENODEV, i, j, region_count; + u32 offset, size; + + dev = sst->dev; + + switch (sst->id) { + case SST_DEV_ID_LYNX_POINT: + region = lp_region; + region_count = ARRAY_SIZE(lp_region); + sst->addr.iram_offset = SST_LP_IRAM_OFFSET; + sst->addr.shim_offset = SST_LP_SHIM_OFFSET; + break; + case SST_DEV_ID_WILDCAT_POINT: + region = wpt_region; + region_count = ARRAY_SIZE(wpt_region); + sst->addr.iram_offset = SST_WPT_IRAM_OFFSET; + sst->addr.shim_offset = SST_WPT_SHIM_OFFSET; + break; + default: + dev_err(dev, "error: failed to get mem resources\n"); + return ret; + } + + ret = hsw_acpi_resource_map(sst, pdata); + if (ret < 0) { + dev_err(dev, "error: failed to map resources\n"); + return ret; + } + + /* enable the DSP SHIM */ + ret = hsw_enable_shim(sst); + if (ret < 0) { + dev_err(dev, "error: failed to set DSP D0 and reset SHIM\n"); + return ret; + } + + ret = dma_coerce_mask_and_coherent(dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + /* Enable Interrupt from both sides */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, 0x3, 0x0); + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRD, + (0x3 | 0x1 << 16 | 0x3 << 21), 0x0); + + /* register DSP memory blocks - ideally we should get this from ACPI */ + for (i = 0; i < region_count; i++) { + offset = region[i].start; + size = (region[i].end - region[i].start) / region[i].blocks; + + /* register individual memory blocks */ + for (j = 0; j < region[i].blocks; j++) { + sst_mem_block_register(sst, offset, size, + region[i].type, &sst_hsw_ops, j, sst); + offset += size; + } + } + + /* set default power gating mask */ + writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL0); + + return 0; +} + +static void hsw_free(struct sst_dsp *sst) +{ + sst_mem_block_unregister_all(sst); + iounmap(sst->addr.lpe); + iounmap(sst->addr.pci_cfg); +} + +struct sst_ops haswell_ops = { + .reset = hsw_reset, + .boot = hsw_boot, + .write = sst_shim32_write, + .read = sst_shim32_read, + .write64 = sst_shim32_write64, + .read64 = sst_shim32_read64, + .ram_read = sst_memcpy_fromio_32, + .ram_write = sst_memcpy_toio_32, + .irq_handler = hsw_irq, + .init = hsw_init, + .free = hsw_free, + .parse_fw = hsw_parse_fw_image, +}; -- 1.8.3.2

4 21

[alsa-devel] [PATCH] ALSA: hda - Add a fixup for HP Folio 13 mute LED
by Takashi Iwai 24 Feb '14

24 Feb '14

HP Folio 13 may have a broken BIOS that doesn't set up the mute LED GPIO properly, and the driver guesses it wrongly, too. Add a new fixup entry for setting the GPIO pin statically for this laptop. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70991 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a2f11bf8155c..3bc29c9b2529 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -98,6 +98,7 @@ enum { STAC_92HD83XXX_HP_LED, STAC_92HD83XXX_HP_INV_LED, STAC_92HD83XXX_HP_MIC_LED, + STAC_HP_LED_GPIO10, STAC_92HD83XXX_HEADSET_JACK, STAC_92HD83XXX_HP, STAC_HP_ENVY_BASS, @@ -2130,6 +2131,17 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, } } +static void stac92hd83xxx_fixup_hp_led_gpio10(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct sigmatel_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gpio_led = 0x10; /* GPIO4 */ + spec->default_polarity = 0; + } +} + static void stac92hd83xxx_fixup_headset_jack(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -2624,6 +2636,12 @@ static const struct hda_fixup stac92hd83xxx_fixups[] = { .chained = true, .chain_id = STAC_92HD83XXX_HP, }, + [STAC_HP_LED_GPIO10] = { + .type = HDA_FIXUP_FUNC, + .v.func = stac92hd83xxx_fixup_hp_led_gpio10, + .chained = true, + .chain_id = STAC_92HD83XXX_HP, + }, [STAC_92HD83XXX_HEADSET_JACK] = { .type = HDA_FIXUP_FUNC, .v.func = stac92hd83xxx_fixup_headset_jack, @@ -2702,6 +2720,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1888, "HP Envy Spectre", STAC_HP_ENVY_BASS), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1899, + "HP Folio 13", STAC_HP_LED_GPIO10), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x18df, "HP Folio", STAC_HP_BNB13_EQ), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x18F8, -- 1.9.0

1 0

[alsa-devel] [PATCH] ALSA: hda - Avoid codec D3 for keeping mute LED up on Lenovo Yxx0
by Takashi Iwai 24 Feb '14

24 Feb '14

The GPIO line used for the mute LED control on Lenovo Yxx0 laptops is cleared unexpectedly when the codec goes to D3, typically by power-saving. For avoiding it, add a power filter in the fixup. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=16373 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3829cb5ebc2b..817a52efe750 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4832,6 +4832,17 @@ static void alc662_led_gpio1_mute_hook(void *private_data, int enabled) spec->gpio_led); } +/* avoid D3 for keeping GPIO up */ +static unsigned int gpio_led_power_filter(struct hda_codec *codec, + hda_nid_t nid, + unsigned int power_state) +{ + struct alc_spec *spec = codec->spec; + if (nid == codec->afg && power_state == AC_PWRST_D3 && spec->gpio_led) + return AC_PWRST_D0; + return power_state; +} + static void alc662_fixup_led_gpio1(struct hda_codec *codec, const struct hda_fixup *fix, int action) { -- 1.8.5.2

1 0

[alsa-devel] [PATCH v2 1/2] ALSA: Use priority list for managing device list
by Takashi Iwai 24 Feb '14

24 Feb '14

Basically, the device type specifies the priority of the device to be registered / freed, too. However, the priority value isn't well utilized but only it's checked as a group. This results in inconsistent register and free order (where each of them should be in reversed direction). This patch simplifies the device list management code by simply inserting a list entry at creation time in an incremental order for the priority value. Since we can just follow the link for register, disconnect and free calls, we don't have to specify the group; so the whole enum definitions are also simplified as well. The visible change to outside is that the priorities of some object types are revisited. For example, now the SNDRV_DEV_LOWLEVEL object is registered before others (control, PCM, etc) and, in return, released after others. Similarly, SNDRV_DEV_CODEC is in a lower priority than SNDRV_DEV_BUS for ensuring the dependency. Also, the unused SNDRV_DEV_TOPLEVEL, SNDRV_DEV_LOWLEVEL_PRE and SNDRV_DEV_LOWLEVEL_NORMAL are removed as a cleanup. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/core.h | 28 ++++++++-------------- sound/core/device.c | 66 +++++++++++++++++++++++++++++----------------------- sound/core/init.c | 14 +++-------- 3 files changed, 50 insertions(+), 58 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index a3e3e89b63b6..5927378dc692 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -46,24 +46,22 @@ struct completion; /* device allocation stuff */ -#define SNDRV_DEV_TYPE_RANGE_SIZE 0x1000 - +/* type of the object used in snd_device_*() + * this also defines the calling order + */ enum snd_device_type { - SNDRV_DEV_TOPLEVEL = 0, - SNDRV_DEV_CONTROL = 1, - SNDRV_DEV_LOWLEVEL_PRE = 2, - SNDRV_DEV_LOWLEVEL_NORMAL = 0x1000, + SNDRV_DEV_LOWLEVEL, + SNDRV_DEV_CONTROL, + SNDRV_DEV_INFO, + SNDRV_DEV_BUS, + SNDRV_DEV_CODEC, SNDRV_DEV_PCM, + SNDRV_DEV_COMPRESS, SNDRV_DEV_RAWMIDI, SNDRV_DEV_TIMER, SNDRV_DEV_SEQUENCER, SNDRV_DEV_HWDEP, - SNDRV_DEV_INFO, - SNDRV_DEV_BUS, - SNDRV_DEV_CODEC, SNDRV_DEV_JACK, - SNDRV_DEV_COMPRESS, - SNDRV_DEV_LOWLEVEL = 0x2000, }; enum snd_device_state { @@ -72,12 +70,6 @@ enum snd_device_state { SNDRV_DEV_DISCONNECTED, }; -enum snd_device_cmd { - SNDRV_DEV_CMD_PRE, - SNDRV_DEV_CMD_NORMAL, - SNDRV_DEV_CMD_POST, -}; - struct snd_device; struct snd_device_ops { @@ -321,7 +313,7 @@ int snd_device_register_all(struct snd_card *card); int snd_device_disconnect(struct snd_card *card, void *device_data); int snd_device_disconnect_all(struct snd_card *card); int snd_device_free(struct snd_card *card, void *device_data); -int snd_device_free_all(struct snd_card *card, enum snd_device_cmd cmd); +int snd_device_free_all(struct snd_card *card); /* isadma.c */ diff --git a/sound/core/device.c b/sound/core/device.c index 856bfdc79cee..53291ff21a09 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -45,6 +45,7 @@ int snd_device_new(struct snd_card *card, enum snd_device_type type, void *device_data, struct snd_device_ops *ops) { struct snd_device *dev; + struct list_head *p; if (snd_BUG_ON(!card || !device_data || !ops)) return -ENXIO; @@ -53,17 +54,36 @@ int snd_device_new(struct snd_card *card, enum snd_device_type type, dev_err(card->dev, "Cannot allocate device, type=%d\n", type); return -ENOMEM; } + INIT_LIST_HEAD(&dev->list); dev->card = card; dev->type = type; dev->state = SNDRV_DEV_BUILD; dev->device_data = device_data; dev->ops = ops; - list_add(&dev->list, &card->devices); /* add to the head of list */ + + /* insert the entry in an incrementally sorted list */ + list_for_each_prev(p, &card->devices) { + struct snd_device *pdev = list_entry(p, struct snd_device, list); + if ((unsigned int)pdev->type <= (unsigned int)type) + break; + } + + list_add(&dev->list, p); return 0; } - EXPORT_SYMBOL(snd_device_new); +static struct snd_device *look_for_dev(struct snd_card *card, void *device_data) +{ + struct snd_device *dev; + + list_for_each_entry(dev, &card->devices, list) + if (dev->device_data == device_data) + return dev; + + return NULL; +} + /** * snd_device_free - release the device from the card * @card: the card instance @@ -82,9 +102,8 @@ int snd_device_free(struct snd_card *card, void *device_data) if (snd_BUG_ON(!card || !device_data)) return -ENXIO; - list_for_each_entry(dev, &card->devices, list) { - if (dev->device_data != device_data) - continue; + dev = look_for_dev(card, device_data); + if (dev) { /* unlink */ list_del(&dev->list); if (dev->state == SNDRV_DEV_REGISTERED && @@ -103,7 +122,6 @@ int snd_device_free(struct snd_card *card, void *device_data) device_data, __builtin_return_address(0)); return -ENXIO; } - EXPORT_SYMBOL(snd_device_free); /** @@ -125,9 +143,8 @@ int snd_device_disconnect(struct snd_card *card, void *device_data) if (snd_BUG_ON(!card || !device_data)) return -ENXIO; - list_for_each_entry(dev, &card->devices, list) { - if (dev->device_data != device_data) - continue; + dev = look_for_dev(card, device_data); + if (dev) { if (dev->state == SNDRV_DEV_REGISTERED && dev->ops->dev_disconnect) { if (dev->ops->dev_disconnect(dev)) @@ -162,9 +179,8 @@ int snd_device_register(struct snd_card *card, void *device_data) if (snd_BUG_ON(!card || !device_data)) return -ENXIO; - list_for_each_entry(dev, &card->devices, list) { - if (dev->device_data != device_data) - continue; + dev = look_for_dev(card, device_data); + if (dev) { if (dev->state == SNDRV_DEV_BUILD && dev->ops->dev_register) { if ((err = dev->ops->dev_register(dev)) < 0) return err; @@ -177,7 +193,6 @@ int snd_device_register(struct snd_card *card, void *device_data) snd_BUG(); return -ENXIO; } - EXPORT_SYMBOL(snd_device_register); /* @@ -212,7 +227,7 @@ int snd_device_disconnect_all(struct snd_card *card) if (snd_BUG_ON(!card)) return -ENXIO; - list_for_each_entry(dev, &card->devices, list) { + list_for_each_entry_reverse(dev, &card->devices, list) { if (snd_device_disconnect(card, dev->device_data) < 0) err = -ENXIO; } @@ -223,24 +238,17 @@ int snd_device_disconnect_all(struct snd_card *card) * release all the devices on the card. * called from init.c */ -int snd_device_free_all(struct snd_card *card, enum snd_device_cmd cmd) +int snd_device_free_all(struct snd_card *card) { - struct snd_device *dev; - int err; - unsigned int range_low, range_high, type; + struct snd_device *dev, *next; + int ret = 0; if (snd_BUG_ON(!card)) return -ENXIO; - range_low = (unsigned int)cmd * SNDRV_DEV_TYPE_RANGE_SIZE; - range_high = range_low + SNDRV_DEV_TYPE_RANGE_SIZE - 1; - __again: - list_for_each_entry(dev, &card->devices, list) { - type = (unsigned int)dev->type; - if (type >= range_low && type <= range_high) { - if ((err = snd_device_free(card, dev->device_data)) < 0) - return err; - goto __again; - } + list_for_each_entry_safe_reverse(dev, next, &card->devices, list) { + int err = snd_device_free(card, dev->device_data); + if (err < 0) + ret = err; } - return 0; + return ret; } diff --git a/sound/core/init.c b/sound/core/init.c index b7085eb19079..9e7f17b72fb6 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -266,7 +266,7 @@ int snd_card_new(struct device *parent, int idx, const char *xid, return 0; __error_ctl: - snd_device_free_all(card, SNDRV_DEV_CMD_PRE); + snd_device_free_all(card); __error: put_device(&card->card_dev); return err; @@ -454,16 +454,8 @@ static int snd_card_do_free(struct snd_card *card) if (snd_mixer_oss_notify_callback) snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_FREE); #endif - if (snd_device_free_all(card, SNDRV_DEV_CMD_PRE) < 0) { - dev_err(card->dev, "unable to free all devices (pre)\n"); - /* Fatal, but this situation should never occur */ - } - if (snd_device_free_all(card, SNDRV_DEV_CMD_NORMAL) < 0) { - dev_err(card->dev, "unable to free all devices (normal)\n"); - /* Fatal, but this situation should never occur */ - } - if (snd_device_free_all(card, SNDRV_DEV_CMD_POST) < 0) { - dev_err(card->dev, "unable to free all devices (post)\n"); + if (snd_device_free_all(card) < 0) { + dev_err(card->dev, "unable to free all devices\n"); /* Fatal, but this situation should never occur */ } if (card->private_free) -- 1.8.5.2

1 1

[alsa-devel] [PATCH] ASoC: add data field into snd_soc_jack_gpio
by xiangxiao 24 Feb '14

24 Feb '14

so callback could get the context data as needed Change-Id: I5542613ccf9881deb672e71d0f0c4f603ad761fd Signed-off-by: xiangxiao <xiaoxiang(a)xiaomi.com> --- include/sound/soc.h | 3 ++- sound/soc/soc-jack.c | 2 +- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 21038e0..a789a29 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -613,7 +613,8 @@ struct snd_soc_jack_gpio { struct snd_soc_jack *jack; struct delayed_work work; - int (*jack_status_check)(void); + void *data; + int (*jack_status_check)(void *data); }; struct snd_soc_jack { diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index b12fce6..b903f82 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -250,7 +250,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) report = 0; if (gpio->jack_status_check) - report = gpio->jack_status_check(); + report = gpio->jack_status_check(gpio->data); snd_soc_jack_report(jack, report, gpio->report); } -- 1.8.1.2

3 2

[alsa-devel] [PATCH] ASoC: fsl: imx-pcm-fiq: Remove unneeded 'out' label
by Fabio Estevam 24 Feb '14

24 Feb '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Instead of jumping to the 'out' label, just return the error code immediately. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-pcm-fiq.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 6553202..7abf6a0 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -270,18 +270,17 @@ static int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) ret = imx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) - goto out; + return ret; } if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = imx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) - goto out; + return ret; } -out: - return ret; + return 0; } static int ssi_irq = 0; -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: core: add snd_soc_dai_set_tdm_slot_xlate()
by Xiubo Li 24 Feb '14

24 Feb '14

The parameters of rx_mask and tx_mask has no use for snd_soc_dai_set_tdm_slot(), because they could be generated by {XXX_ .}of_xlate_tdm_slot_mask(). This patch add snd_soc_dai_set_tdm_slot_xlate() which will replace the snd_soc_dai_set_tdm_slot() eventually. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- include/sound/soc-dai.h | 3 +++ sound/soc/soc-core.c | 33 ++++++++++++++++++++++++++++----- 2 files changed, 31 insertions(+), 5 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index d86e0fc..68569ee 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -110,6 +110,9 @@ int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); /* Digital Audio interface formatting */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); +int snd_soc_dai_set_tdm_slot_xlate(struct snd_soc_dai *dai, + unsigned int slots, + unsigned int slot_width); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0911856..e5a535b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3687,19 +3687,20 @@ static int snd_soc_of_xlate_tdm_slot_mask(unsigned int slots, } /** - * snd_soc_dai_set_tdm_slot - configure DAI TDM. + * snd_soc_dai_set_tdm_slot_xlate - configure DAI TDM with of xlate. * @dai: DAI - * @tx_mask: bitmask representing active TX slots. - * @rx_mask: bitmask representing active RX slots. * @slots: Number of slots in use. * @slot_width: Width in bits for each slot. * * Configures a DAI for TDM operation. Both mask and slots are codec and DAI * specific. */ -int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +int snd_soc_dai_set_tdm_slot_xlate(struct snd_soc_dai *dai, + unsigned int slots, + unsigned int slot_width) { + unsigned int tx_mask, rx_mask; + if (dai->driver && dai->driver->ops->of_xlate_tdm_slot_mask) dai->driver->ops->of_xlate_tdm_slot_mask(slots, &tx_mask, &rx_mask); @@ -3712,6 +3713,28 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, else return -ENOTSUPP; } +EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot_xlate); + +/** + * snd_soc_dai_set_tdm_slot - configure DAI TDM. + * @dai: DAI + * @tx_mask: bitmask representing active TX slots. + * @rx_mask: bitmask representing active RX slots. + * @slots: Number of slots in use. + * @slot_width: Width in bits for each slot. + * + * Configures a DAI for TDM operation. Both mask and slots are codec and DAI + * specific. + */ +int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + if (dai->driver && dai->driver->ops->set_tdm_slot) + return dai->driver->ops->set_tdm_slot(dai, tx_mask, rx_mask, + slots, slot_width); + else + return -ENOTSUPP; +} EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); /** -- 1.8.4

1 0

[alsa-devel] [PATCH] ASoC: da732x: Remove leftover cache size setting
by Mark Brown 24 Feb '14

24 Feb '14

From: Mark Brown <broonie(a)linaro.org> The da732x driver no longer uses the ASoC level register cache but the cache size setting had been left in the driver by mistake. Remove it. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/da732x.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index f295b65..8e2c74d 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1554,7 +1554,6 @@ static struct snd_soc_codec_driver soc_codec_dev_da732x = { .dapm_routes = da732x_dapm_routes, .num_dapm_routes = ARRAY_SIZE(da732x_dapm_routes), .set_pll = da732x_set_dai_pll, - .reg_cache_size = ARRAY_SIZE(da732x_reg_cache), }; static int da732x_i2c_probe(struct i2c_client *i2c, -- 1.9.0.rc3

1 0

[alsa-devel] [PATCH] ASoC: pcm512x: Add PCM512x driver
by Mark Brown 24 Feb '14

24 Feb '14

From: Mark Brown <broonie(a)linaro.org> The PCM512x devices are a family of monolithic CMOS integrated circuits that include a stereo digital-to-analog converter and additional support circuitry. This is an initial driver which supports some core functionality for the device which covers common use cases but does not cover all features. Currently only slave clocking modes with automatic clock configuration are supported and most of the DSP configuration for the device is not enabled. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- .../devicetree/bindings/sound/pcm512x.txt | 30 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm512x.c | 671 +++++++++++++++++++++ sound/soc/codecs/pcm512x.h | 142 +++++ 5 files changed, 849 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm512x.txt create mode 100644 sound/soc/codecs/pcm512x.c create mode 100644 sound/soc/codecs/pcm512x.h diff --git a/Documentation/devicetree/bindings/sound/pcm512x.txt b/Documentation/devicetree/bindings/sound/pcm512x.txt new file mode 100644 index 000000000000..faff75e64573 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm512x.txt @@ -0,0 +1,30 @@ +PCM512x audio CODECs + +These devices support both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : One of "ti,pcm5121" or "ti,pcm5122" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + + - AVDD-supply, DVDD-supply, and CPVDD-supply : power supplies for the + device, as covered in bindings/regulator/regulator.txt + +Optional properties: + + - clocks : A clock specifier for the clock connected as SCLK. If this + is absent the device will be configured to clock from BCLK. + +Example: + + pcm5122: pcm5122@4c { + compatible = "ti,pcm5122"; + reg = <0x4c>; + + AVDD-supply = <&reg_3v3_analog>; + DVDD-supply = <&reg_1v8>; + CPVDD-supply = <&reg_3v3>; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f0bdcc5abe83..28faa2d0eaef 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -60,6 +60,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 + select SND_SOC_PCM512x if SND_SOC_I2C_AND_SPI select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C @@ -324,6 +325,9 @@ config SND_SOC_PCM1792A config SND_SOC_PCM3008 tristate +config SND_SOC_PCM512x + tristate "Texas Instruments PCM512x CODECs" + config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc126764a44d..d3b536fc075d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -46,6 +46,7 @@ snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-pcm512x-objs := pcm512x.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -179,6 +180,7 @@ obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c new file mode 100644 index 000000000000..886e530ad2d2 --- /dev/null +++ b/sound/soc/codecs/pcm512x.c @@ -0,0 +1,671 @@ +/* + * Driver for the PCM512x CODECs + * + * Author: Mark Brown <broonie(a)linaro.org> + * Copyright 2014 Linaro Ltd + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <linux/spi/spi.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "pcm512x.h" + +#define PCM512x_NUM_SUPPLIES 3 +static const char *pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = { + "AVDD", + "DVDD", + "CPVDD", +}; + +struct pcm512x_priv { + struct regmap *regmap; + struct clk *sclk; + struct regulator_bulk_data supplies[PCM512x_NUM_SUPPLIES]; + struct notifier_block supply_nb[PCM512x_NUM_SUPPLIES]; +}; + +/* + * We can't use the same notifier block for more than one supply and + * there's no way I can see to get from a callback to the caller + * except container_of(). + */ +#define PCM512x_REGULATOR_EVENT(n) \ +static int pcm512x_regulator_event_##n(struct notifier_block *nb, \ + unsigned long event, void *data) \ +{ \ + struct pcm512x_priv *pcm512x = container_of(nb, struct pcm512x_priv, \ + supply_nb[n]); \ + if (event & REGULATOR_EVENT_DISABLE) { \ + regcache_mark_dirty(pcm512x->regmap); \ + regcache_cache_only(pcm512x->regmap, true); \ + } \ + return 0; \ +} + +PCM512x_REGULATOR_EVENT(0) +PCM512x_REGULATOR_EVENT(1) +PCM512x_REGULATOR_EVENT(2) + +static const struct reg_default pcm512x_reg_defaults[] = { + { PCM512x_RESET, 0x00 }, + { PCM512x_POWER, 0x00 }, + { PCM512x_MUTE, 0x00 }, + { PCM512x_DSP, 0x00 }, + { PCM512x_PLL_REF, 0x00 }, + { PCM512x_DAC_ROUTING, 0x11 }, + { PCM512x_DSP_PROGRAM, 0x01 }, + { PCM512x_CLKDET, 0x00 }, + { PCM512x_AUTO_MUTE, 0x00 }, + { PCM512x_ERROR_DETECT, 0x00 }, + { PCM512x_DIGITAL_VOLUME_1, 0x00 }, + { PCM512x_DIGITAL_VOLUME_2, 0x30 }, + { PCM512x_DIGITAL_VOLUME_3, 0x30 }, + { PCM512x_DIGITAL_MUTE_1, 0x22 }, + { PCM512x_DIGITAL_MUTE_2, 0x00 }, + { PCM512x_DIGITAL_MUTE_3, 0x07 }, +}; + +static bool pcm512x_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case PCM512x_RESET: + case PCM512x_POWER: + case PCM512x_MUTE: + case PCM512x_PLL_EN: + case PCM512x_SPI_MISO_FUNCTION: + case PCM512x_DSP: + case PCM512x_GPIO_EN: + case PCM512x_BCLK_LRCLK_CFG: + case PCM512x_DSP_GPIO_INPUT: + case PCM512x_MASTER_MODE: + case PCM512x_PLL_REF: + case PCM512x_PLL_COEFF_0: + case PCM512x_PLL_COEFF_1: + case PCM512x_PLL_COEFF_2: + case PCM512x_PLL_COEFF_3: + case PCM512x_PLL_COEFF_4: + case PCM512x_DSP_CLKDIV: + case PCM512x_DAC_CLKDIV: + case PCM512x_NCP_CLKDIV: + case PCM512x_OSR_CLKDIV: + case PCM512x_MASTER_CLKDIV_1: + case PCM512x_MASTER_CLKDIV_2: + case PCM512x_FS_SPEED_MODE: + case PCM512x_IDAC_1: + case PCM512x_IDAC_2: + case PCM512x_ERROR_DETECT: + case PCM512x_I2S_1: + case PCM512x_I2S_2: + case PCM512x_DAC_ROUTING: + case PCM512x_DSP_PROGRAM: + case PCM512x_CLKDET: + case PCM512x_AUTO_MUTE: + case PCM512x_DIGITAL_VOLUME_1: + case PCM512x_DIGITAL_VOLUME_2: + case PCM512x_DIGITAL_VOLUME_3: + case PCM512x_DIGITAL_MUTE_1: + case PCM512x_DIGITAL_MUTE_2: + case PCM512x_DIGITAL_MUTE_3: + case PCM512x_GPIO_OUTPUT_1: + case PCM512x_GPIO_OUTPUT_2: + case PCM512x_GPIO_OUTPUT_3: + case PCM512x_GPIO_OUTPUT_4: + case PCM512x_GPIO_OUTPUT_5: + case PCM512x_GPIO_OUTPUT_6: + case PCM512x_GPIO_CONTROL_1: + case PCM512x_GPIO_CONTROL_2: + case PCM512x_OVERFLOW: + case PCM512x_RATE_DET_1: + case PCM512x_RATE_DET_2: + case PCM512x_RATE_DET_3: + case PCM512x_RATE_DET_4: + case PCM512x_ANALOG_MUTE_DET: + case PCM512x_GPIN: + case PCM512x_DIGITAL_MUTE_DET: + return true; + default: + return false; + } +} + +static bool pcm512x_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case PCM512x_PLL_EN: + case PCM512x_OVERFLOW: + case PCM512x_RATE_DET_1: + case PCM512x_RATE_DET_2: + case PCM512x_RATE_DET_3: + case PCM512x_RATE_DET_4: + case PCM512x_ANALOG_MUTE_DET: + case PCM512x_GPIN: + case PCM512x_DIGITAL_MUTE_DET: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -10350, 50, 1); + +static const char *pcm512x_dsp_program_texts[] = { + "FIR interpolation with de-emphasis", + "Low latency IIR with de-emphasis", + "High attenuation with de-emphasis", + "Ringing-less low latency FIR", +}; + +static const unsigned int pcm512x_dsp_program_values[] = { + 1, + 2, + 3, + 5, + 7, +}; + +static const SOC_VALUE_ENUM_SINGLE_DECL(pcm512x_dsp_program, + PCM512x_DSP_PROGRAM, 0, 0x1f, + pcm512x_dsp_program_texts, + pcm512x_dsp_program_values); + +static const char *pcm512x_clk_missing_text[] = { + "1s", "2s", "3s", "4s", "5s", "6s", "7s", "8s" +}; + +static const struct soc_enum pcm512x_clk_missing = + SOC_ENUM_SINGLE(PCM512x_CLKDET, 0, 7, pcm512x_clk_missing_text); + +static const char *pcm512x_autom_text[] = { + "21ms", "106ms", "213ms", "533ms", "1.07s", "2.13s", "5.33s", "10.66s" +}; + +static const struct soc_enum pcm512x_autom_l = + SOC_ENUM_SINGLE(PCM512x_AUTO_MUTE, PCM512x_ATML_SHIFT, 7, + pcm512x_autom_text); + +static const struct soc_enum pcm512x_autom_r = + SOC_ENUM_SINGLE(PCM512x_AUTO_MUTE, PCM512x_ATMR_SHIFT, 7, + pcm512x_autom_text); + +static const char *pcm512x_ramp_rate_text[] = { + "1 sample/update", "2 samples/update", "4 samples/update", + "Immediate" +}; + +static const struct soc_enum pcm512x_vndf = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNDF_SHIFT, 4, + pcm512x_ramp_rate_text); + +static const struct soc_enum pcm512x_vnuf = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNUF_SHIFT, 4, + pcm512x_ramp_rate_text); + +static const struct soc_enum pcm512x_vedf = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDF_SHIFT, 4, + pcm512x_ramp_rate_text); + +static const char *pcm512x_ramp_step_text[] = { + "4dB/step", "2dB/step", "1dB/step", "0.5dB/step" +}; + +static const struct soc_enum pcm512x_vnds = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNDS_SHIFT, 4, + pcm512x_ramp_step_text); + +static const struct soc_enum pcm512x_vnus = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNUS_SHIFT, 4, + pcm512x_ramp_step_text); + +static const struct soc_enum pcm512x_veds = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDS_SHIFT, 4, + pcm512x_ramp_step_text); + +static const struct snd_kcontrol_new pcm512x_controls[] = { +SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2, + PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv), +SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, + PCM512x_RQMR_SHIFT, 1, 1), + +SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1), +SOC_VALUE_ENUM("DSP Program", pcm512x_dsp_program), + +SOC_ENUM("Clock Missing Period", pcm512x_clk_missing), +SOC_ENUM("Auto Mute Time Left", pcm512x_autom_l), +SOC_ENUM("Auto Mute Time Right", pcm512x_autom_r), +SOC_SINGLE("Auto Mute Mono Switch", PCM512x_DIGITAL_MUTE_3, + PCM512x_ACTL_SHIFT, 1, 0), +SOC_DOUBLE("Auto Mute Switch", PCM512x_DIGITAL_MUTE_3, PCM512x_AMLE_SHIFT, + PCM512x_AMLR_SHIFT, 1, 0), + +SOC_ENUM("Volume Ramp Down Rate", pcm512x_vndf), +SOC_ENUM("Volume Ramp Down Step", pcm512x_vnds), +SOC_ENUM("Volume Ramp Up Rate", pcm512x_vnuf), +SOC_ENUM("Volume Ramp Up Step", pcm512x_vnus), +SOC_ENUM("Volume Ramp Down Emergency Rate", pcm512x_vedf), +SOC_ENUM("Volume Ramp Down Emergency Step", pcm512x_veds), +}; + +static const struct snd_soc_dapm_widget pcm512x_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DACL", NULL, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_DAC("DACR", NULL, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_OUTPUT("OUTL"), +SND_SOC_DAPM_OUTPUT("OUTR"), +}; + +static const struct snd_soc_dapm_route pcm512x_dapm_routes[] = { + { "DACL", NULL, "Playback" }, + { "DACR", NULL, "Playback" }, + + { "OUTL", NULL, "DACL" }, + { "OUTR", NULL, "DACR" }, +}; + +static int pcm512x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct pcm512x_priv *pcm512x = dev_get_drvdata(codec->dev); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQST, 0); + if (ret != 0) { + dev_err(codec->dev, "Failed to remove standby: %d\n", + ret); + return ret; + } + break; + + case SND_SOC_BIAS_OFF: + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQST, PCM512x_RQST); + if (ret != 0) { + dev_err(codec->dev, "Failed to request standby: %d\n", + ret); + return ret; + } + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +static struct snd_soc_dai_driver pcm512x_dai = { + .name = "pcm512x-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE + }, +}; + +static struct snd_soc_codec_driver pcm512x_codec_driver = { + .set_bias_level = pcm512x_set_bias_level, + .idle_bias_off = true, + + .controls = pcm512x_controls, + .num_controls = ARRAY_SIZE(pcm512x_controls), + .dapm_widgets = pcm512x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm512x_dapm_widgets), + .dapm_routes = pcm512x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm512x_dapm_routes), +}; + +static const struct regmap_config pcm512x_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .readable_reg = pcm512x_readable, + .volatile_reg = pcm512x_volatile, + + .max_register = PCM512x_MAX_REGISTER, + .reg_defaults = pcm512x_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm512x_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + +static const struct of_device_id pcm512x_of_match[] = { + { .compatible = "ti,pcm5121", }, + { .compatible = "ti,pcm5122", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm512x_of_match); + +static int pcm512x_probe(struct device *dev, struct regmap *regmap) +{ + struct pcm512x_priv *pcm512x; + int i, ret; + + pcm512x = devm_kzalloc(dev, sizeof(struct pcm512x_priv), GFP_KERNEL); + if (!pcm512x) + return -ENOMEM; + + dev_set_drvdata(dev, pcm512x); + pcm512x->regmap = regmap; + + for (i = 0; i < ARRAY_SIZE(pcm512x->supplies); i++) + pcm512x->supplies[i].supply = pcm512x_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + if (ret != 0) { + dev_err(dev, "Failed to get supplies: %d\n", ret); + return ret; + } + + pcm512x->supply_nb[0].notifier_call = pcm512x_regulator_event_0; + pcm512x->supply_nb[1].notifier_call = pcm512x_regulator_event_1; + pcm512x->supply_nb[2].notifier_call = pcm512x_regulator_event_2; + + for (i = 0; i < ARRAY_SIZE(pcm512x->supplies); i++) { + ret = regulator_register_notifier(pcm512x->supplies[i].consumer, + &pcm512x->supply_nb[i]); + if (ret != 0) { + dev_err(dev, + "Failed to register regulator notifier: %d\n", + ret); + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + /* Reset the device, verifying I/O in the process for I2C */ + ret = regmap_write(regmap, PCM512x_RESET, + PCM512x_RSTM | PCM512x_RSTR); + if (ret != 0) { + dev_err(dev, "Failed to reset device: %d\n", ret); + goto err; + } + + ret = regmap_write(regmap, PCM512x_RESET, 0); + if (ret != 0) { + dev_err(dev, "Failed to reset device: %d\n", ret); + goto err; + } + + pcm512x->sclk = devm_clk_get(dev, NULL); + if (IS_ERR(pcm512x->sclk)) { + if (PTR_ERR(pcm512x->sclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + + dev_info(dev, "No SCLK, using BCLK: %ld\n", + PTR_ERR(pcm512x->sclk)); + + /* Disable reporting of missing SCLK as an error */ + regmap_update_bits(regmap, PCM512x_ERROR_DETECT, + PCM512x_IDCH, PCM512x_IDCH); + + /* Switch PLL input to BCLK */ + regmap_update_bits(regmap, PCM512x_PLL_REF, + PCM512x_SREF, PCM512x_SREF); + } else { + ret = clk_prepare_enable(pcm512x->sclk); + if (ret != 0) { + dev_err(dev, "Failed to enable SCLK: %d\n", ret); + return ret; + } + } + + /* Default to standby mode */ + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQST, PCM512x_RQST); + if (ret != 0) { + dev_err(dev, "Failed to request standby: %d\n", + ret); + goto err_clk; + } + + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + pm_runtime_idle(dev); + + ret = snd_soc_register_codec(dev, &pcm512x_codec_driver, + &pcm512x_dai, 1); + if (ret != 0) { + dev_err(dev, "Failed to register CODEC: %d\n", ret); + goto err_pm; + } + + return 0; + +err_pm: + pm_runtime_disable(dev); +err_clk: + if (!IS_ERR(pcm512x->sclk)) + clk_disable_unprepare(pcm512x->sclk); +err: + regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + return ret; +} + +static void pcm512x_remove(struct device *dev) +{ + struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); + + snd_soc_unregister_codec(dev); + pm_runtime_disable(dev); + if (!IS_ERR(pcm512x->sclk)) + clk_disable_unprepare(pcm512x->sclk); + regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); +} + +static int pcm512x_suspend(struct device *dev) +{ + struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); + int ret; + + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQPD, PCM512x_RQPD); + if (ret != 0) { + dev_err(dev, "Failed to request power down: %d\n", ret); + return ret; + } + + ret = regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + if (ret != 0) { + dev_err(dev, "Failed to disable supplies: %d\n", ret); + return ret; + } + + if (!IS_ERR(pcm512x->sclk)) + clk_disable_unprepare(pcm512x->sclk); + + return 0; +} + +static int pcm512x_resume(struct device *dev) +{ + struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); + int ret; + + if (!IS_ERR(pcm512x->sclk)) { + ret = clk_prepare_enable(pcm512x->sclk); + if (ret != 0) { + dev_err(dev, "Failed to enable SCLK: %d\n", ret); + return ret; + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + regcache_cache_only(pcm512x->regmap, false); + ret = regcache_sync(pcm512x->regmap); + if (ret != 0) { + dev_err(dev, "Failed to sync cache: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQPD, 0); + if (ret != 0) { + dev_err(dev, "Failed to remove power down: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct dev_pm_ops pcm512x_pm_ops = { + SET_RUNTIME_PM_OPS(pcm512x_suspend, pcm512x_resume, NULL) +}; + +#if IS_ENABLED(CONFIG_I2C) +static int pcm512x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_i2c(i2c, &pcm512x_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return pcm512x_probe(&i2c->dev, regmap); +} + +static int pcm512x_i2c_remove(struct i2c_client *i2c) +{ + pcm512x_remove(&i2c->dev); + return 0; +} + +static const struct i2c_device_id pcm512x_i2c_id[] = { + { "pcm5121", }, + { "pcm5122", }, + { } +}; +MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id); + +static struct i2c_driver pcm512x_i2c_driver = { + .probe = pcm512x_i2c_probe, + .remove = pcm512x_i2c_remove, + .id_table = pcm512x_i2c_id, + .driver = { + .name = "pcm512x", + .owner = THIS_MODULE, + .of_match_table = pcm512x_of_match, + .pm = &pcm512x_pm_ops, + }, +}; +#endif + +#if defined(CONFIG_SPI_MASTER) +static int pcm512x_spi_probe(struct spi_device *spi) +{ + struct regmap *regmap; + int ret; + + regmap = devm_regmap_init_spi(spi, &pcm512x_regmap); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + return ret; + } + + return pcm512x_probe(&spi->dev, regmap); +} + +static int pcm512x_spi_remove(struct spi_device *spi) +{ + pcm512x_remove(&spi->dev); + return 0; +} + +static const struct spi_device_id pcm512x_spi_id[] = { + { "pcm5121", }, + { "pcm5122", }, + { }, +}; +MODULE_DEVICE_TABLE(spi, pcm512x_spi_id); + +static struct spi_driver pcm512x_spi_driver = { + .probe = pcm512x_spi_probe, + .remove = pcm512x_spi_remove, + .id_table = pcm512x_spi_id, + .driver = { + .name = "pcm512x", + .owner = THIS_MODULE, + .of_match_table = pcm512x_of_match, + .pm = &pcm512x_pm_ops, + }, +}; +#endif + +static int __init pcm512x_modinit(void) +{ + int ret = 0; + +#if IS_ENABLED(CONFIG_I2C) + ret = i2c_add_driver(&pcm512x_i2c_driver); + if (ret) { + printk(KERN_ERR "Failed to register pcm512x I2C driver: %d\n", + ret); + } +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&pcm512x_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register pcm512x SPI driver: %d\n", + ret); + } +#endif + return ret; +} +module_init(pcm512x_modinit); + +static void __exit pcm512x_exit(void) +{ +#if IS_ENABLED(CONFIG_I2C) + i2c_del_driver(&pcm512x_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&pcm512x_spi_driver); +#endif +} +module_exit(pcm512x_exit); + +MODULE_DESCRIPTION("ASoC PCM512x codec driver"); +MODULE_AUTHOR("Mark Brown <broonie(a)linaro.org>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h new file mode 100644 index 000000000000..b2f518ecb35c --- /dev/null +++ b/sound/soc/codecs/pcm512x.h @@ -0,0 +1,142 @@ +/* + * Driver for the PCM512x CODECs + * + * Author: Mark Brown <broonie(a)linaro.org> + * Copyright 2014 Linaro Ltd + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef _SND_SOC_PCM512X +#define _SND_SOC_PCM512X + +#define PCM512x_PAGE_0_BASE 0 + +#define PCM512x_PAGE 0 + +#define PCM512x_RESET (PCM512x_PAGE_0_BASE + 1) +#define PCM512x_POWER (PCM512x_PAGE_0_BASE + 2) +#define PCM512x_MUTE (PCM512x_PAGE_0_BASE + 3) +#define PCM512x_PLL_EN (PCM512x_PAGE_0_BASE + 4) +#define PCM512x_SPI_MISO_FUNCTION (PCM512x_PAGE_0_BASE + 6) +#define PCM512x_DSP (PCM512x_PAGE_0_BASE + 7) +#define PCM512x_GPIO_EN (PCM512x_PAGE_0_BASE + 8) +#define PCM512x_BCLK_LRCLK_CFG (PCM512x_PAGE_0_BASE + 9) +#define PCM512x_DSP_GPIO_INPUT (PCM512x_PAGE_0_BASE + 10) +#define PCM512x_MASTER_MODE (PCM512x_PAGE_0_BASE + 12) +#define PCM512x_PLL_REF (PCM512x_PAGE_0_BASE + 13) +#define PCM512x_PLL_COEFF_0 (PCM512x_PAGE_0_BASE + 20) +#define PCM512x_PLL_COEFF_1 (PCM512x_PAGE_0_BASE + 21) +#define PCM512x_PLL_COEFF_2 (PCM512x_PAGE_0_BASE + 22) +#define PCM512x_PLL_COEFF_3 (PCM512x_PAGE_0_BASE + 23) +#define PCM512x_PLL_COEFF_4 (PCM512x_PAGE_0_BASE + 24) +#define PCM512x_DSP_CLKDIV (PCM512x_PAGE_0_BASE + 27) +#define PCM512x_DAC_CLKDIV (PCM512x_PAGE_0_BASE + 28) +#define PCM512x_NCP_CLKDIV (PCM512x_PAGE_0_BASE + 29) +#define PCM512x_OSR_CLKDIV (PCM512x_PAGE_0_BASE + 30) +#define PCM512x_MASTER_CLKDIV_1 (PCM512x_PAGE_0_BASE + 32) +#define PCM512x_MASTER_CLKDIV_2 (PCM512x_PAGE_0_BASE + 33) +#define PCM512x_FS_SPEED_MODE (PCM512x_PAGE_0_BASE + 34) +#define PCM512x_IDAC_1 (PCM512x_PAGE_0_BASE + 35) +#define PCM512x_IDAC_2 (PCM512x_PAGE_0_BASE + 36) +#define PCM512x_ERROR_DETECT (PCM512x_PAGE_0_BASE + 37) +#define PCM512x_I2S_1 (PCM512x_PAGE_0_BASE + 40) +#define PCM512x_I2S_2 (PCM512x_PAGE_0_BASE + 41) +#define PCM512x_DAC_ROUTING (PCM512x_PAGE_0_BASE + 42) +#define PCM512x_DSP_PROGRAM (PCM512x_PAGE_0_BASE + 43) +#define PCM512x_CLKDET (PCM512x_PAGE_0_BASE + 44) +#define PCM512x_AUTO_MUTE (PCM512x_PAGE_0_BASE + 59) +#define PCM512x_DIGITAL_VOLUME_1 (PCM512x_PAGE_0_BASE + 60) +#define PCM512x_DIGITAL_VOLUME_2 (PCM512x_PAGE_0_BASE + 61) +#define PCM512x_DIGITAL_VOLUME_3 (PCM512x_PAGE_0_BASE + 62) +#define PCM512x_DIGITAL_MUTE_1 (PCM512x_PAGE_0_BASE + 63) +#define PCM512x_DIGITAL_MUTE_2 (PCM512x_PAGE_0_BASE + 64) +#define PCM512x_DIGITAL_MUTE_3 (PCM512x_PAGE_0_BASE + 65) +#define PCM512x_GPIO_OUTPUT_1 (PCM512x_PAGE_0_BASE + 80) +#define PCM512x_GPIO_OUTPUT_2 (PCM512x_PAGE_0_BASE + 81) +#define PCM512x_GPIO_OUTPUT_3 (PCM512x_PAGE_0_BASE + 82) +#define PCM512x_GPIO_OUTPUT_4 (PCM512x_PAGE_0_BASE + 83) +#define PCM512x_GPIO_OUTPUT_5 (PCM512x_PAGE_0_BASE + 84) +#define PCM512x_GPIO_OUTPUT_6 (PCM512x_PAGE_0_BASE + 85) +#define PCM512x_GPIO_CONTROL_1 (PCM512x_PAGE_0_BASE + 86) +#define PCM512x_GPIO_CONTROL_2 (PCM512x_PAGE_0_BASE + 87) +#define PCM512x_OVERFLOW (PCM512x_PAGE_0_BASE + 90) +#define PCM512x_RATE_DET_1 (PCM512x_PAGE_0_BASE + 91) +#define PCM512x_RATE_DET_2 (PCM512x_PAGE_0_BASE + 92) +#define PCM512x_RATE_DET_3 (PCM512x_PAGE_0_BASE + 93) +#define PCM512x_RATE_DET_4 (PCM512x_PAGE_0_BASE + 94) +#define PCM512x_ANALOG_MUTE_DET (PCM512x_PAGE_0_BASE + 108) +#define PCM512x_GPIN (PCM512x_PAGE_0_BASE + 119) +#define PCM512x_DIGITAL_MUTE_DET (PCM512x_PAGE_0_BASE + 120) + +#define PCM512x_MAX_REGISTER (PCM512x_PAGE_0_BASE + 120) + +/* Page 0, Register 1 - reset */ +#define PCM512x_RSTR (1 << 0) +#define PCM512x_RSTM (1 << 4) + +/* Page 0, Register 2 - power */ +#define PCM512x_RQPD (1 << 0) +#define PCM512x_RQPD_SHIFT 0 +#define PCM512x_RQST (1 << 4) +#define PCM512x_RQST_SHIFT 4 + +/* Page 0, Register 3 - mute */ +#define PCM512x_RQMR_SHIFT 0 +#define PCM512x_RQML_SHIFT 4 + +/* Page 0, Register 4 - PLL */ +#define PCM512x_PLCE (1 << 0) +#define PCM512x_RLCE_SHIFT 0 +#define PCM512x_PLCK (1 << 4) +#define PCM512x_PLCK_SHIFT 4 + +/* Page 0, Register 7 - DSP */ +#define PCM512x_SDSL (1 << 0) +#define PCM512x_SDSL_SHIFT 0 +#define PCM512x_DEMP (1 << 4) +#define PCM512x_DEMP_SHIFT 4 + +/* Page 0, Register 13 - PLL reference */ +#define PCM512x_SREF (1 << 4) + +/* Page 0, Register 37 - Error detection */ +#define PCM512x_IPLK (1 << 0) +#define PCM512x_DCAS (1 << 1) +#define PCM512x_IDCM (1 << 2) +#define PCM512x_IDCH (1 << 3) +#define PCM512x_IDSK (1 << 4) +#define PCM512x_IDBK (1 << 5) +#define PCM512x_IDFS (1 << 6) + +/* Page 0, Register 42 - DAC routing */ +#define PCM512x_AUPR_SHIFT 0 +#define PCM512x_AUPL_SHIFT 4 + +/* Page 0, Register 59 - auto mute */ +#define PCM512x_ATMR_SHIFT 0 +#define PCM512x_ATML_SHIFT 4 + +/* Page 0, Register 63 - ramp rates */ +#define PCM512x_VNDF_SHIFT 6 +#define PCM512x_VNDS_SHIFT 4 +#define PCM512x_VNUF_SHIFT 2 +#define PCM512x_VNUS_SHIFT 0 + +/* Page 0, Register 64 - emergency ramp rates */ +#define PCM512x_VEDF_SHIFT 6 +#define PCM512x_VEDS_SHIFT 4 + +/* Page 0, Register 65 - Digital mute enables */ +#define PCM512x_ACTL_SHIFT 2 +#define PCM512x_AMLE_SHIFT 1 +#define PCM512x_AMLR_SHIFT 0 + +#endif -- 1.9.rc1

3 12

[alsa-devel] [PATCH] ASoC: delay the initial jack detect by debounce_time
by xiangxiao 24 Feb '14

24 Feb '14

so the hardware could get time to initialize and debounce Change-Id: Iebeffa9bac6c4012ee8f9645771c13b28fe5ae80 Signed-off-by: xiangxiao <xiaoxiang(a)xiaomi.com> --- sound/soc/soc-jack.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 23d43da..b12fce6 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -342,7 +342,8 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, gpio_export(gpios[i].gpio, false); /* Update initial jack status */ - snd_soc_jack_gpio_detect(&gpios[i]); + schedule_delayed_work(&gpios[i].work, + msecs_to_jiffies(gpios[i].debounce_time)); } return 0; -- 1.8.1.2

2 1

Re: [alsa-devel] [FFADO-devel] Call for testing: ALSA driver for Fireworks/BeBoB based devices
by Brand, Mark (Mr) (Summerstrand Campus South) 23 Feb '14

23 Feb '14

Dear Takashi, Firstly: thank you so much for taking the time to do this! I am very, very grateful! You asked for the following: $ cat /proc/asound/card2/#clock Sampling rate: 48000 Clock Source: Internal (MSU-dest: 5) $ cat /proc/asound/card2/#firmware Manufacturer: bridgeCo Protocol Ver: 1 Build Ver: 0 GUID: 0x000D6C0A009D269F Model ID: 0x01 Model Rev: 1 Firmware Date: 20050905 Firmware Time: 091022 Firmware ID: 0xA Firmware Ver: 20 Base Addr: 0x20080000 Max Size: 1572864 Loader Date: 20040915 Loader Time: 175644 $ cat /proc/asound/card2/#formation Output Stream from device: Rate PCM MIDI 22050 0 0 24000 0 0 32000 0 0 44100 4 1 48000 4 1 88200 4 1 96000 4 1 176400 0 0 192000 0 0 Input Stream to device: Rate PCM MIDI 22050 0 0 24000 0 0 32000 0 0 44100 4 1 48000 4 1 88200 4 1 96000 4 1 176400 0 0 192000 0 0 $ cat /proc/asound/card2/#meter Analog In 1: 1328896 Analog In 2: 1216256 Analog In 3: 1326097920 Analog In 4: 8012800 Stream In 1: 0 Stream In 2: 0 Stream In 3: 0 Stream In 4: 0 Analog Out 1: 0 Analog Out 2: 0 Analog Out 3: 0 Analog Out 4: 0 Now, my experience: I am testing on a new, well provisioned PC, freshly loaded with Ubuntu 13.10. My firewire card has a Texas Instruments chipset, and my device is the Ozonic. I install via DKMS, as per your instructions, and then sudo modprobe snd-bebob. The snd_firewire_lib and snd_bebob modules are now listed in the output of lsmod. The device has installed as card2 (hw:2,0). I can now test MIDI input with: $ amidi -p hw:2,0,0 -d ... which satisfies me that all MIDI output from the various controllers on the Ozonic is being received. I have not yet tried to send any MIDI data to the device. I open Audacity, which now lists "Ozonic: Ozonic PCM (hw:2,0)" as ALSA input and output options. However, no audio input is shown when I try to monitor the audio input, and trying to play generated audio (e.g. DTMF tones) results in the message: "Error while opening sound device. Please check the output device settings and the project sample rate." I have repeated the experiment at all supported sampling rates (44100, 48000, 88200 & 96000 Hz), and obtained the same result in each case. I then close Audacity and try: $aplay -D hw:2,0 test.wav (where test.wav has been saved from Audacity) ... and this responds with: aplay: main:722: audio open error: Input/output error Based on my experiences with the Ozonic under FFADO, I might easily have resigned myself to thinking that full functionality will only be apparent once a mixer is implemented. I had hacked together the "unmute-ozonic" utility under FFADO in order to overcome the need the need to dual boot in order to get I/O. However, the fact that an error message is returned upon trying to open the Ozonic's output makes me think that this is another problem entirely. UPDATE: I have just this moment been able to do the following: $ aplay -D hw:2,0 Desktop/test.wav Playing WAVE 'Desktop/test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono aplay: set_params:1239: Channels count not available ... which appears in order to me, except that no audio can be heard. I assume this is due to the lack of a mixer. I will retrace my steps to try and figure out why it is now apparently working. However, I now open Audacity again, and the first thing I notice is that my pass-through audio immediately disappears (I have a sound source playing into the unbalanced inputs throughout). Audacity responds exactly the same as before. I close Audacity and aplay can no longer open hw:2,0 (I/O error as before). Now I modprobe -r the modules (and lsmod to confirm they're gone), cycle the Ozonic's power off and on, and the audio pass-through has reappeared! I also see that lsmod lists both snd_firewire_lib & snd_bebob as having reloaded themselves automatically. My testing is probably not particularly rigorous, and I would sincerely appreciate your advice on what steps to follow in order to verify your driver's functionality. In particular, you mention "(to control mixer channels please use FFADO upstream)". I have not yet set up FFADO on this machine, but I also see under "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." Would you please explain? How should one rely on the "upstream" mixer functionality while keeping jackd out of ALSA's way? Once again, thank you sincerely for doing this. I, for one, feel like a kid on Christmas Eve! Kind regards, Mark Brand. ________________________________________ From: Takashi Sakamoto [o-takashi(a)sakamocchi.jp] Sent: Monday, October 14, 2013 5:17 AM To: alsa-devel(a)alsa-project.org; ffado-devel(a)lists.sf.net; linux1394-devel(a)lists.sourceforge.net; linux-audio-dev(a)lists.linuxaudio.org Subject: [FFADO-devel] Call for testing: ALSA driver for Fireworks/BeBoB based devices 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. [snip] 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 ________________________________ NOTICE: Please note that this eMail, and the contents thereof, is subject to the standard NMMU eMail disclaimer which may be found at: http://www.nmmu.ac.za/disclaimer/email.htm

2 1

Re: [alsa-devel] 答复: [PATCH] ASoC: pass snd_soc_jack_gpio to jack_status_check callback
by Mark Brown 23 Feb '14

23 Feb '14

On Wed, Feb 19, 2014 at 02:20:07PM +0000, 肖翔 wrote: > The new parameter isn't used to get the internal field of snd_soc_jack_gpio. > the usage is very like work_struct, here is the demo code: > struct xxx_priv { > truct snd_soc_jack_gpio gpio; > }; > > static int xxx_ jack_status_check(struct snd_soc_jack_gpio *gpio) > { > struct xxx_priv *xxx = container_of(gpio, struct xxx_priv, gpio); > ...... > } > Without this parameter, codec driver has to save xxx_priv as a global variable. OK, this makes sense as a reason for passing the argument in however how about instead of passing in the structure adding a void * to it which gets passed as the argument instead? The user can then assign it to point to the data they want directly without it looking like they should be using the private data for the core code.

2 1

[alsa-devel] [PATCH v3 0/3] Add TDM slot info OF parsing supports
by Xiubo Li 23 Feb '14

23 Feb '14

Change in v3: - Add .of_xlate_tdm_slot_mask(). - Add snd_soc_of_xlate_tdm_slot_mask(). - Remove TDM slot masks OF parsing... - Split the TDM infomation into two properties. Xiubo Li (3): ASoC: binding: add tdm-slot.txt ASoC: core: add TDM slot parsing from DT supports ASoC: simple-card: add slot information parsing supports .../devicetree/bindings/sound/simple-card.txt | 5 ++ .../devicetree/bindings/sound/tdm-slot.txt | 20 ++++++++ include/sound/simple_card.h | 2 + include/sound/soc-dai.h | 2 + include/sound/soc.h | 3 ++ sound/soc/generic/simple-card.c | 18 +++++++ sound/soc/soc-core.c | 59 ++++++++++++++++++++++ 7 files changed, 109 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tdm-slot.txt -- 1.8.4

2 4

[alsa-devel] [PATCH v6 0/4] ASoC: tlv320aic32x4: DT support
by Markus Pargmann 23 Feb '14

23 Feb '14

Hi, This series adds master clock and regulators to the tlv320aic32x4 driver. In v6 I added error handling for the regulator setup. master clock is required for the driver now but still optional in the DT bindings. Regards, Markus Markus Pargmann (4): ASoC: tlv320aic32x4: Support for master clock ASoC: tlv320aic32x4: Support for regulators ASoC: tlv320aic32x4: Rearrange clock tree shutdown Asoc: codecs: Remove tlv320aic32x4 from compatibles of tlv320aic3x .../devicetree/bindings/sound/tlv320aic32x4.txt | 12 ++ .../devicetree/bindings/sound/tlv320aic3x.txt | 1 - sound/soc/codecs/tlv320aic32x4.c | 177 +++++++++++++++++++-- 3 files changed, 173 insertions(+), 17 deletions(-) -- 1.8.5.3

3 8

[alsa-devel] [PATCH 029/102] ASoC: adau1373: Use SOC_ENUM_SINGLE_DECL()
by Takashi Iwai 23 Feb '14

23 Feb '14

Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/adau1373.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 46b1595d7c50..5765c224cd9a 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -576,8 +576,8 @@ static const char *adau1373_decimator_text[] = { "DMIC1", }; -static const struct soc_enum adau1373_decimator_enum = - SOC_ENUM_SINGLE(0, 0, 2, adau1373_decimator_text); +static SOC_ENUM_SINGLE_DECL(adau1373_decimator_enum, + 0, 0, adau1373_decimator_text); static const struct snd_kcontrol_new adau1373_decimator_mux = SOC_DAPM_ENUM_VIRT("Decimator Mux", adau1373_decimator_enum); -- 1.8.5.2

2 109

[alsa-devel] [PATCH] ASoC: pxa: Remove superfluous locking
by Lars-Peter Clausen 23 Feb '14

23 Feb '14

The locking here was added in commit 71a295602e ("ASoC: Lock the CODEC in PXA external jack controls") to protect the DAPM changes that are made inside of ${board}_ext_control() against concurrent updates. The ASoC core was updated in commit a73fb2df01 ("ASoC: dapm: Use DAPM mutex for DAPM ops instead of codec mutex") to use a card wide lock rather the CODEC mutex to protect DAPM operations. We now have proper locking inside ${board}_ext_control() itself, so taking the CODEC lock can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/pxa/corgi.c | 4 ---- sound/soc/pxa/magician.c | 4 ---- sound/soc/pxa/poodle.c | 4 ---- sound/soc/pxa/spitz.c | 4 ---- sound/soc/pxa/tosa.c | 4 ---- 5 files changed, 20 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 9d9c8ad..b4db3b9 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -103,13 +103,9 @@ static int corgi_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; - mutex_lock(&codec->mutex); - /* check the jack status at stream startup */ corgi_ext_control(&codec->dapm); - mutex_unlock(&codec->mutex); - return 0; } diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 31242be..41ab6678 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -77,13 +77,9 @@ static int magician_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; - mutex_lock(&codec->mutex); - /* check the jack status at stream startup */ magician_ext_control(codec); - mutex_unlock(&codec->mutex); - return 0; } diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index c93e138..27c6c03 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -76,13 +76,9 @@ static int poodle_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; - mutex_lock(&codec->mutex); - /* check the jack status at stream startup */ poodle_ext_control(&codec->dapm); - mutex_unlock(&codec->mutex); - return 0; } diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 04dbb5a..a3a13c9 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -113,13 +113,9 @@ static int spitz_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; - mutex_lock(&codec->mutex); - /* check the jack status at stream startup */ spitz_ext_control(&codec->dapm); - mutex_unlock(&codec->mutex); - return 0; } diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 2a4b438..cead165 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -84,13 +84,9 @@ static int tosa_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; - mutex_lock(&codec->mutex); - /* check the jack status at stream startup */ tosa_ext_control(codec); - mutex_unlock(&codec->mutex); - return 0; } -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: sta32x: Fix cache sync
by Lars-Peter Clausen 23 Feb '14

23 Feb '14

codec->control_data contains a pointer to the regmap struct of the device, not to the device private data. Use snd_soc_codec_get_drvdata() instead. The issue was introduced in commit 29fdf4fbbe ("ASoC: sta32x: Convert to regmap"). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/sta32x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 42c5b45..ea78c17 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -331,7 +331,7 @@ static int sta32x_sync_coef_shadow(struct snd_soc_codec *codec) static int sta32x_cache_sync(struct snd_soc_codec *codec) { - struct sta32x_priv *sta32x = codec->control_data; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); unsigned int mute; int rc; -- 1.8.0

2 1

[alsa-devel] No sound from Headphone Jack over Lenovo Docking Station with T440p Laptop
by Nicolas Kalkhof 22 Feb '14

22 Feb '14

Hello, I have a Lenovo ThinkPad T440p laptop connected to an Ultra Docking Station. Sound works fine over Laptop Speakers and the 3,5mm Headphone Jack. However that is not the case when I connect the T440p to the Docking station Type 40A10065EU and use the Headphone Jack at the Dock. I've tried loading the alsa module with the "model=thinkpad" or "model=lenovo-dock" without success. My Distro is Gentoo and the Kernel is 3.13-rc4. alsa.conf: # ALSA portion alias char-major-116 snd # OSS/Free portion alias char-major-14 soundcore ## ## IMPORTANT: ## You need to customise this section for your specific sound card(s) ## and then run `update-modules' command. ## Read alsa-driver's INSTALL file in /usr/share/doc for more info. ## ## ALSA portion ## alias snd-card-0 snd-interwave ## alias snd-card-1 snd-ens1371 ## OSS/Free portion ## alias sound-slot-0 snd-card-0 ## alias sound-slot-1 snd-card-1 ## # OSS/Free portion - card #1 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss ## OSS/Free portion - card #2 ## alias sound-service-1-0 snd-mixer-oss ## alias sound-service-1-3 snd-pcm-oss ## alias sound-service-1-12 snd-pcm-oss alias /dev/mixer snd-mixer-oss alias /dev/dsp snd-pcm-oss alias /dev/midi snd-seq-oss # Set this to the correct number of cards. options snd cards_limit=1 #options snd-hda-intel model=thinkpad options snd-hda-intel model=thinkpad enable=1 index=0 options snd-hda-intel model=hdmi enable=1 index=-2 lspci -vv and dmesg goes here: [1]http://pastebin.ca/2510939 Any Help? References 1. http://pastebin.ca/2510939

4 9

[alsa-devel] [PATCH 0/3] alsa-lib - improve surround 2.1 support
by David Henningsson 21 Feb '14

21 Feb '14

So, in order to support 2.1 surround on both 4-channels (e g FL FR LFE LFE, on internal subwoofer) and 6-channels (e g FL FR RL RR C LFE, on desktops with 5.1 out), I needed to make the route plugin more chmap aware. This also included allowing a new syntax for the ttable, which I think is quite nice. The new syntax also falls back to ALSAs standard chmap in case the hardware does not support it. David Henningsson (3): route: Allow chmap syntax for slave channels in ttable route: Select slave chmap based on ttable information conf: Allow 2.1 surround to use different number of channels src/conf/pcm/surround21.conf | 7 +- src/pcm/pcm_route.c | 304 ++++++++++++++++++++++++++++++++++++++----- 2 files changed, 271 insertions(+), 40 deletions(-) -- 1.9.0

2 6

[alsa-devel] [PATCH] rawmidi: remove undefined functions.
by Takashi Sakamoto 21 Feb '14

21 Feb '14

'snd_rawmidi_transmit_reset()' and 'snd_rawmidi_receive_reset()' are declared but not defined. This state has been continue over 10 years. So let us remove them. --- include/sound/rawmidi.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index adf0885..311dafe 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -157,10 +157,8 @@ void snd_rawmidi_set_ops(struct snd_rawmidi *rmidi, int stream, /* callbacks */ -void snd_rawmidi_receive_reset(struct snd_rawmidi_substream *substream); int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, const unsigned char *buffer, int count); -void snd_rawmidi_transmit_reset(struct snd_rawmidi_substream *substream); int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream); int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, unsigned char *buffer, int count); -- 1.8.3.2

2 3

[alsa-devel] Enable more HP machine to enable mute led
by Kailang 21 Feb '14

21 Feb '14

Hi Takashi, HP want to enable mute led functions for more machine. I created patch as attach patch. Thanks. BR, Kailang

2 1

[alsa-devel] [PATCH 000/102 - chunk#5] Add SOC_ENUM_SINGLE_CONST() & co
by Takashi Iwai 21 Feb '14

21 Feb '14

This is the last chunk of patch series, which introduces two new macros, SOC_ENUM_SINGLE_CONST() and SOC_ENUM_DOUBLE_CONST() in order to simplify the codes without ARRAY_SIZE() in the caller side. Takashi

2 30

Re: [alsa-devel] Headphone output problems with HP ProDesk 600 G1 (Realtek ALC221)
by Raymond Yau 20 Feb '14

20 Feb '14

>> I have a HP ProDesk 600 G1 machine which seems to have some strange behavior with the front headphone jack(s) running under Fedora 20 (running 3.13.2 kernel): >> This machine has two front jacks: a headphone jack and a microphone/headphone jack. When the headphones are plugged into the headphone jack, there's usually no change in the sound outputs list in the Sound control panel. The audio (mostly) works if the audio output device is set to Analog Output, but sometimes spuriously changes back to Speakers and the headphone output stops working (I suspect maybe from a spurious jack-sensing event, though I'm not certain). More confusingly, a couple of times I have gotten it into the state (through various fiddling with jacks and playing in the Sound control panel) where plugging into this jack is detected and causes the Headphone output to be selected in the control panel. I haven't been able to figure out consistently what causes this to happen. >> If the headphone is plugged into the microphone/headphone jack, then the control panel consistently switches to the Headphone output and consistently switches back the way it was when it's unplugged. However, there's no actual audio output coming out of that jack. (From some of the HP manuals, it sounds like with the Windows driver, when you plug into this jack, it's supposed to pop up a prompt asking you whether you want to use it as a microphone or headphones.) >> The other strange thing is that intermittently, on the headphone jack you'll start getting a bunch of loud static (clicking noises) and any actual audio playback becomes very faint and distorted. >> I'm guessing perhaps something isn't being initialized/setup properly in the codec, causing this kind of erratic behavior. The alsa-info output is uploaded at: http://www.alsa-project.org/db/?f=594e837b7af07ffbe3473cf807465bc81a587880 >> I tried using hda-analyzer but it fails on this machine with: >> ValueError: wrong proc file format (unknown dig1 bit 'KAE') >> Any suggestions for things to try to debug this? Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="PCM Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00010014: OUT EAPD Detect EAPD 0x2: EAPD Pin Default 0x01014020: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 it is a driver bug which assign PCM playback Switch to Line Out Jack try the patch https://bugzilla.kernel.org/attachment.cgi?id=124971 HP ProDesk 600 G1 SFF is an All-In-One pc , should headphone and line out share the volume control if headphone can mute the line out or speaker Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Speaker Only' 'Line Out Speaker' Item0: 'Line Out Speaker' Node 0x02 [Audio Output] wcaps 0x1d: 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=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Control: name="PCM Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC221 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x57 0x57] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM

2 8

Re: [alsa-devel] About 'SNDRV_PCM_CLASS_MULTI' for devices with multi-channels
by David Henningsson 20 Feb '14

20 Feb '14

On 02/19/2014 03:57 PM, Takashi Sakamoto wrote: > Hi all, > > I have two questions about the usage of this macro. > > Should I use this macro for new drivers which I'm working for?' > http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071820.ht… > > > My drivers support some firewire devices, which is desinged for > studio-use and have many channels. Its combination of channels are not > designed for surround sound like 5.1ch. > > Just judging from its name, 'SNDRV_PCM_CLASS_MULTI', I should use this. > But there are no drivers which use this macro in current tree, against > my expectation that some drivers (ICE1712-pro, RME9652, 6fire usb and so > on) should use this macro. > > Are there any reasons? I've never heard of it before. I had a quick look in alsa-lib and couldn't find any difference in handling between CLASS_GENERIC and CLASS_MULTI. It's probably something historic that nobody uses any more. Maybe people who have been along longer knows what it is. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

4 3

[alsa-devel] [PATCH] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 20 Feb '14

20 Feb '14

This patch adds initial support for the Behringer BCD2000 USB DJ controller. At the moment, only the MIDI part of the device is working, i.e. knobs, buttons and LEDs. I also plan to add support for the audio part, but I assume that this will require more effort than the rather simple MIDI interface. Progress can be tracked at https://github.com/anyc/snd-usb-bcd2000. Changes since v4: - devices_used as array to support arbitrary number of SNDRV_CARDS - removed unused array "enable" Changes since v3: - applied style and snd_printk changes as suggested by Daniel Mack Changes since v2: - applied more changes from Daniel Mack and Clemens Ladisch Changes since v1: - fixed the various code style issues, thanks to Daniel Mack and checkpatch.pl. Signed-off-by: Mario Kicherer <dev(a)kicherer.org> --- sound/usb/Kconfig | 13 ++ sound/usb/Makefile | 2 +- sound/usb/bcd2000/Makefile | 3 + sound/usb/bcd2000/bcd2000.c | 490 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 507 insertions(+), 1 deletion(-) create mode 100644 sound/usb/bcd2000/Makefile create mode 100644 sound/usb/bcd2000/bcd2000.c diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index de9408b..145653a 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -146,5 +146,18 @@ config SND_USB_HIFACE To compile this driver as a module, choose M here: the module will be called snd-usb-hiface. +config SND_BCD2000 + tristate "Behringer BCD2000 MIDI driver" + select SND_RAWMIDI + help + Say Y here to include MIDI support for the Behringer BCD2000 DJ + controller. + + Audio support is still work-in-progress at + https://github.com/anyc/snd-usb-bcd2000 + + To compile this driver as a module, choose M here: the module + will be called snd-bcd2000. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index abe668f..2b92f0d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ bcd2000/ diff --git a/sound/usb/bcd2000/Makefile b/sound/usb/bcd2000/Makefile new file mode 100644 index 0000000..f09ccc0 --- /dev/null +++ b/sound/usb/bcd2000/Makefile @@ -0,0 +1,3 @@ +snd-bcd2000-y := bcd2000.o + +obj-$(CONFIG_SND_BCD2000) += snd-bcd2000.o \ No newline at end of file diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c new file mode 100644 index 0000000..847e079 --- /dev/null +++ b/sound/usb/bcd2000/bcd2000.c @@ -0,0 +1,490 @@ +/* + * Behringer BCD2000 driver + * + * Copyright (C) 2014 Mario Kicherer (dev(a)kicherer.org) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/usb.h> +#include <linux/usb/audio.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/rawmidi.h> + +#define DEVICE_NAME "Behringer BCD2000" +#define DEVICE_SHORTNAME "bcd2000" + +#define PREFIX DEVICE_SHORTNAME ": " +#define BUFSIZE 64 + +static struct usb_device_id id_table[] = { + { USB_DEVICE(0x1397, 0x00bd) }, + { }, +}; + +static unsigned char device_cmd_prefix[] = {0x03, 0x00}; + +static unsigned char bcd2000_init_sequence[] = { + 0x07, 0x00, 0x00, 0x00, 0x78, 0x48, 0x1c, 0x81, + 0xc4, 0x00, 0x00, 0x00, 0x5e, 0x53, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff, 0x6c, 0xf3, 0x90, 0xff, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, + 0x18, 0xfa, 0x11, 0xff, 0x14, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0xf2, 0x34, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff +}; + +struct bcd2000 { + struct usb_device *dev; + struct snd_card *card; + struct usb_interface *intf; + int card_index; + + int midi_out_active; + struct snd_rawmidi *rmidi; + struct snd_rawmidi_substream *midi_receive_substream; + struct snd_rawmidi_substream *midi_out_substream; + + unsigned char midi_in_buf[BUFSIZE]; + unsigned char midi_out_buf[BUFSIZE]; + + struct urb *midi_out_urb; + struct urb *midi_in_urb; + + struct usb_anchor anchor; +}; + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; + +static DEFINE_MUTEX(devices_mutex); +static unsigned int devices_used[SNDRV_CARDS/sizeof(int)]; +static struct usb_driver bcd2000_driver; + +#ifdef CONFIG_SND_DEBUG +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) +{ + print_hex_dump(KERN_DEBUG, prefix, + DUMP_PREFIX_NONE, 16, 1, + buf, len, false); +} +#else +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) {} +#endif + +static int bcd2000_midi_input_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_input_close(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +/* register midi substream */ +static void bcd2000_midi_input_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + bcd2k->midi_receive_substream = up ? substream : NULL; +} + +static void bcd2000_midi_handle_input(struct bcd2000 *bcd2k, + const unsigned char *buf, unsigned int len) +{ + unsigned int length, tocopy; + struct snd_rawmidi_substream *midi_receive_substream; + + midi_receive_substream = ACCESS_ONCE(bcd2k->midi_receive_substream); + if (!midi_receive_substream) + return; + + bcd2000_dump_buffer(PREFIX "received from device: ", buf, len); + + if (len < 2) + return; + + /* + * Packet structure: mm nn oo (pp) + * mm: payload length + * nn: MIDI command or note + * oo: note or velocity + * pp: velocity + */ + + length = buf[0]; + + /* ignore packets without payload */ + if (length == 0) + return; + + tocopy = min(length, len-1); + + bcd2000_dump_buffer(PREFIX "sending to userspace: ", + &buf[1], tocopy); + + snd_rawmidi_receive(midi_receive_substream, + &buf[1], tocopy); +} + +static void bcd2000_midi_send(struct bcd2000 *bcd2k, + struct snd_rawmidi_substream *substream) +{ + int len, ret; + + BUILD_BUG_ON(sizeof(device_cmd_prefix) >= BUFSIZE); + + /* copy the "set LED" command bytes */ + memcpy(bcd2k->midi_out_buf, device_cmd_prefix, + sizeof(device_cmd_prefix)); + + /* + * get MIDI packet and leave space for command prefix + * and payload length + */ + len = snd_rawmidi_transmit(substream, + bcd2k->midi_out_buf + 3, BUFSIZE - 3); + + if (len < 0) + dev_err(&bcd2k->dev->dev, "%s: snd_rawmidi_transmit error %d\n", + __func__, len); + + if (len <= 0) + return; + + /* set payload length */ + bcd2k->midi_out_buf[2] = len; + bcd2k->midi_out_urb->transfer_buffer_length = BUFSIZE; + + bcd2000_dump_buffer(PREFIX "sending to device: ", + bcd2k->midi_out_buf, len+3); + + /* send packet to the BCD2000 */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s (%p): usb_submit_urb() failed, ret=%d, len=%d\n", + __func__, substream, ret, len); + else + bcd2k->midi_out_active = 1; +} + +static int bcd2000_midi_output_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_output_close(struct snd_rawmidi_substream *substream) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (bcd2k->midi_out_active) { + usb_kill_urb(bcd2k->midi_out_urb); + bcd2k->midi_out_active = 0; + } + + return 0; +} + +/* register midi substream */ +static void bcd2000_midi_output_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (up) { + bcd2k->midi_out_substream = substream; + /* check if there is data userspace wants to send */ + if (!bcd2k->midi_out_active) + bcd2000_midi_send(bcd2k, substream); + } else { + bcd2k->midi_out_substream = NULL; + } +} + +static void bcd2000_output_complete(struct urb *urb) +{ + struct bcd2000 *bcd2k = urb->context; + struct snd_rawmidi_substream *midi_out_substream; + + bcd2k->midi_out_active = 0; + + if (urb->status != 0) + dev_err(&urb->dev->dev, + PREFIX "output urb->status: %d\n", urb->status); + + midi_out_substream = ACCESS_ONCE(bcd2k->midi_out_substream); + if (!midi_out_substream) + return; + + /* check if there is more data userspace wants to send */ + bcd2000_midi_send(bcd2k, midi_out_substream); +} + +static void bcd2000_input_complete(struct urb *urb) +{ + int ret; + struct device *dev = &urb->dev->dev; + struct bcd2000 *bcd2k = urb->context; + + if (urb->status) { + dev_warn(dev, PREFIX "input urb->status: %i\n", urb->status); + return; + } + + if (!bcd2k) + return; + + if (urb->actual_length > 0) + bcd2000_midi_handle_input(bcd2k, urb->transfer_buffer, + urb->actual_length); + + /* acknowledge received packet */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(dev, "unable to submit urb. OOM!?\n"); +} + +static struct snd_rawmidi_ops bcd2000_midi_output = { + .open = bcd2000_midi_output_open, + .close = bcd2000_midi_output_close, + .trigger = bcd2000_midi_output_trigger, +}; + +static struct snd_rawmidi_ops bcd2000_midi_input = { + .open = bcd2000_midi_input_open, + .close = bcd2000_midi_input_close, + .trigger = bcd2000_midi_input_trigger, +}; + +static void bcd2000_init_device(struct bcd2000 *bcd2k) +{ + int ret; + + init_usb_anchor(&bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_out_urb, &bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_in_urb, &bcd2k->anchor); + + /* copy init sequence into buffer */ + memcpy(bcd2k->midi_out_buf, bcd2000_init_sequence, 52); + bcd2k->midi_out_urb->transfer_buffer_length = 52; + + /* submit sequence */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() out failed, ret=%d: ", + __func__, ret); + else + bcd2k->midi_out_active = 1; + + /* send empty packet to enable button and controller events */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() in failed, ret=%d: ", + __func__, ret); + + /* ensure initialization is finished */ + usb_wait_anchor_empty_timeout(&bcd2k->anchor, 1000); +} + +static int bcd2000_init_midi(struct bcd2000 *bcd2k) +{ + int ret; + struct snd_rawmidi *rmidi; + + ret = snd_rawmidi_new(bcd2k->card, bcd2k->card->shortname, 0, + 1, /* output */ + 1, /* input */ + &rmidi); + + if (ret < 0) + return ret; + + strlcpy(rmidi->name, bcd2k->card->shortname, sizeof(rmidi->name)); + + rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; + rmidi->private_data = bcd2k; + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &bcd2000_midi_output); + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &bcd2000_midi_input); + + bcd2k->rmidi = rmidi; + + bcd2k->midi_in_urb = usb_alloc_urb(0, GFP_KERNEL); + bcd2k->midi_out_urb = usb_alloc_urb(0, GFP_KERNEL); + + if (!bcd2k->midi_in_urb || !bcd2k->midi_out_urb) { + dev_err(&bcd2k->dev->dev, PREFIX "usb_alloc_urb failed\n"); + return -ENOMEM; + } + + usb_fill_int_urb(bcd2k->midi_in_urb, bcd2k->dev, + usb_rcvintpipe(bcd2k->dev, 0x81), + bcd2k->midi_in_buf, BUFSIZE, + bcd2000_input_complete, bcd2k, 1); + + usb_fill_int_urb(bcd2k->midi_out_urb, bcd2k->dev, + usb_sndintpipe(bcd2k->dev, 0x1), + bcd2k->midi_out_buf, BUFSIZE, + bcd2000_output_complete, bcd2k, 1); + + bcd2000_init_device(bcd2k); + + return 0; +} + +static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k, + struct usb_interface *interface) +{ + /* usb_kill_urb not necessary, urb is aborted automatically */ + + usb_free_urb(bcd2k->midi_out_urb); + usb_free_urb(bcd2k->midi_in_urb); + + if (bcd2k->intf) { + usb_set_intfdata(bcd2k->intf, NULL); + bcd2k->intf = NULL; + } +} + +static int bcd2000_probe(struct usb_interface *interface, + const struct usb_device_id *usb_id) +{ + struct snd_card *card; + struct bcd2000 *bcd2k; + unsigned int card_index; + char usb_path[32]; + int err; + + mutex_lock(&devices_mutex); + + for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) + if (!(devices_used[card_index/sizeof(int)*8] & + (1 << card_index % (sizeof(int)*8)))) + break; + + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + + err = snd_card_create(index[card_index], id[card_index], THIS_MODULE, + sizeof(*bcd2k), &card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return err; + } + + bcd2k = card->private_data; + bcd2k->dev = interface_to_usbdev(interface); + bcd2k->card = card; + bcd2k->card_index = card_index; + bcd2k->intf = interface; + + snd_card_set_dev(card, &interface->dev); + + strcpy(card->driver, DEVICE_NAME); + strcpy(card->shortname, DEVICE_SHORTNAME); + usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); + snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), + DEVICE_NAME ", at %s", + usb_path); + + dev_info(&bcd2k->dev->dev, PREFIX "%s", bcd2k->card->longname); + + err = bcd2000_init_midi(bcd2k); + if (err < 0) + goto probe_error; + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + usb_set_intfdata(interface, bcd2k); + devices_used[card_index/sizeof(int)*8] |= + 1 << (card_index%(sizeof(int)*8)); + + mutex_unlock(&devices_mutex); + return 0; + +probe_error: + dev_info(&bcd2k->dev->dev, PREFIX "error during probing"); + bcd2000_free_usb_related_resources(bcd2k, interface); + snd_card_free(card); + mutex_unlock(&devices_mutex); + return err; +} + +static void bcd2000_disconnect(struct usb_interface *interface) +{ + struct bcd2000 *bcd2k = usb_get_intfdata(interface); + + if (!bcd2k) + return; + + mutex_lock(&devices_mutex); + + /* make sure that userspace cannot create new requests */ + snd_card_disconnect(bcd2k->card); + + bcd2000_free_usb_related_resources(bcd2k, interface); + + devices_used[bcd2k->card_index/sizeof(int)*8] &= + ~(1 << (bcd2k->card_index%(sizeof(int)*8))); + + snd_card_free_when_closed(bcd2k->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_driver bcd2000_driver = { + .name = DEVICE_SHORTNAME, + .probe = bcd2000_probe, + .disconnect = bcd2000_disconnect, + .id_table = id_table, +}; + +static int __init bcd2000_init(void) +{ + int retval = 0; + + retval = usb_register(&bcd2000_driver); + if (retval) + pr_info(PREFIX "usb_register failed. Error: %d", retval); + return retval; +} + +static void __exit bcd2000_exit(void) +{ + usb_deregister(&bcd2000_driver); +} + +module_init(bcd2000_init); +module_exit(bcd2000_exit); + +MODULE_DEVICE_TABLE(usb, id_table); +MODULE_AUTHOR("Mario Kicherer, dev(a)kicherer.org"); +MODULE_DESCRIPTION("Behringer BCD2000 driver"); +MODULE_LICENSE("GPL"); -- 1.8.3.2

2 1

[alsa-devel] [PATCH v4 0/4] ASoC: tlv320aic32x4: DT support
by Markus Pargmann 20 Feb '14

20 Feb '14

Hi, These are the remaining patches from v3 of this version. They add master clock and regulator support. In v4 I added a check for the regulator dependencies and updated the binding documentation for them. I fixed the master clock patch to shutdown the master clock after the PLL and added another patch to shutdown all dividers before the PLL. Regards, Markus Changes in v4: - Fix clock shutdown order - Add regulator dependency checks - Revert the previously added tlv320aic32x4 compatible from the binding documentation Markus Pargmann (4): ASoC: tlv320aic32x4: Support for master clock ASoC: tlv320aic32x4: Support for regulators ASoC: tlv320aic32x4: Rearrange clock tree shutdown Revert "ASoC: codec doc, tlv320aic3x: Add tlv320aic32x4 as compatible" .../devicetree/bindings/sound/tlv320aic32x4.txt | 12 ++ .../devicetree/bindings/sound/tlv320aic3x.txt | 1 - sound/soc/codecs/tlv320aic32x4.c | 126 ++++++++++++++++++--- 3 files changed, 123 insertions(+), 16 deletions(-) -- 1.8.5.3

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[alsa-devel] Need Help: Repeated Jackhammer noise after several hours of play and heavy cpu usage in snd_pcm_dmix_sync_area
by Adarsh 20 Feb '14

20 Feb '14

Hi, We are facing a issue where if gdb is attached to the our long running sound process and continued, jackhammer sound is played over and over again abusing the speakers. Process details: 1. We are using RW_INTERLEAVED access as described in write_loop method in /test/pcm.c and opening 'default' device. 2. However to disable the underruns, the snd_pcm_sw_params_set_stop_threshold is set to 0x7fffffff which is greater than boundary value. 3.Period size is 1024 and buffer size is 2048. We try to write period_size no of frames using snd_pcm_writei. (I am attaching a sample program which shows this behaviour.) Steps to Reproduce: 1. Left process running for over 12 hours. 2. gdb is attached to the process and continued inside gdb. 3. At this point, the thread went into a tight loop taking >100% cpu and repeatedly playing same sound. It did not recover for a long time. Stack trace shows that 'snd_pcm_dmix_sync_area' in pcm_dmix.c is taking a very long time to return because the 'size' is a large unsinged interger and mix_areas is getting called with 'transfer' value of <=2048. If snd_pcm_sw_params_set_stop_threshold is not set or when application is running for shorter duration, attaching gdb does not show this behaviour. Is it that setting of stop threshold to 0x7fffffff is triggering this and attaching gdb is making the buffer pointers misbehave? If so, what can be done if we do not want PCM to stop? I have tried by setting silence threshold and silence size to let alsa play silence by itself but still observed the issue. Below is the stack trace and PCM dump: (gdb) bt #0 0xb7c44463 in mix_areas_32_smp (size=1360, dst=0xb6855000, src=0x84a7510, sum=0xb6845000, dst_step=4, src_step=4, sum_step=4) at pcm_dmix_i386.h:261 #1 0xb7c44d2b in mix_areas (dmix=0x84a7110, src_areas=0x84a51d0, dst_areas=0x84a6f00, src_ofs=0, dst_ofs=7168, size=1024) at pcm_dmix.c:193 #2 0xb7c45422 in snd_pcm_dmix_sync_area (pcm=0x84a71e0) at pcm_dmix.c:371 #3 0xb7c46680 in snd_pcm_dmix_mmap_commit (pcm=0x84a71e0, offset=0, size=1024) at pcm_dmix.c:813 #4 0xb7c07033 in snd_pcm_mmap_commit (pcm=0x84a71e0, offset=0, frames=1024) at pcm.c:6501 #5 0xb7c1804e in snd_pcm_plugin_mmap_commit (pcm=0x84a6d18, offset=0, size=1024) at pcm_plugin.c:434 #6 0xb7c07033 in snd_pcm_mmap_commit (pcm=0x84a6d18, offset=0, frames=1024) at pcm.c:6501 #7 0xb7c17956 in snd_pcm_plugin_write_areas (pcm=0x84a5390, areas=0xb68362b0, offset=0, size=1024) at pcm_plugin.c:290 #8 0xb7c074bb in snd1_pcm_write_areas (pcm=0x84a5390, areas=0xb68362b0, offset=0, size=1024, func=0xb7c177fc <snd_pcm_plugin_write_areas>) at pcm.c:6667 #9 0xb7c17cd5 in snd_pcm_plugin_writei (pcm=0x84a5390, buffer=0x84ab518, size=1024) at pcm_plugin.c:361 #10 0xb7bff1a7 in _snd_pcm_writei (pcm=0x84a6b18, buffer=0x84ab518, size=1024) at pcm_local.h:516 #11 0xb7bff16c in snd_pcm_writei (pcm=0x84a6b18, buffer=0x84ab518, size=1024) at pcm.c:1258 (gdb) f 2 #2 0xb7c45422 in snd_pcm_dmix_sync_area (pcm=0x84a71e0) at pcm_dmix.c:371 371 pcm_dmix.c: No such file or directory. (gdb) info locals dmix = 0x84a7110 slave_hw_ptr = 234156032 slave_appl_ptr = 7168 slave_size = 1057171456 appl_ptr = 0 size = 1018295296 transfer = 1024 src_areas = 0x84a51d0 dst_areas = 0x84a6f00 (gdb) p *dmix $1 = {type = SND_PCM_TYPE_DMIX, ipc_key = 5678293, ipc_perm = 432, ipc_gid = 17, semid = 32768, shmid = 65536, shmptr = 0xb6859000, spcm = 0x84a7340, appl_ptr = 217121792, last_appl_ptr = 233882624, hw_ptr = 233875428, avail_max = 0, slave_appl_ptr = 234156032, slave_hw_ptr = 234147996, slave_period_size = 1024, slave_buffer_size = 8192, slave_boundary = 1073741824, sync_ptr = 0xb7c4557c <snd_pcm_dmix_sync_ptr>, state = SND_PCM_STATE_RUNNING, trigger_tstamp = {tv_sec = 1392384066, tv_nsec = 8308000}, server = 0, client = 0, comm_fd = 0, hw_fd = 4, timer_fd = { fd = 3, events = 41, revents = 0}, poll_fd = 3, tread = 1, timer_need_poll = 0, timer_event_suspend = 131072, timer_event_resume = 262144, server_fd = 0, server_pid = 0, timer = 0x8496b90, interleaved = 1, slowptr = 1, max_periods = 0, channels = 2, bindings = 0x0, u = {dmix = { shmid_sum = 98305, sum_buffer = 0xb6837000, mix_areas_16 = 0xb7c443a8 <mix_areas_16_smp_mmx>, mix_areas_32 = 0xb7c44415 <mix_areas_32_smp>, mix_areas_24 = 0xb7c4451b <mix_areas_24_smp_cmov>, mix_areas_u8 = 0, remix_areas_16 = 0xb7c44638 <remix_areas_16_smp_mmx>, remix_areas_32 = 0xb7c446a5 <remix_areas_32_smp>, remix_areas_24 = 0xb7c447ab <remix_areas_24_smp_cmov>, remix_areas_u8 = 0}, dsnoop = {<No data fields>}, dshare = {chn_mask = 13151478482154389505}}, server_free = 0} (gdb) p *pcm $2 = {dl_handle = 0x0, name = 0x84a6e48 "dmix:0", type = SND_PCM_TYPE_DMIX, stream = SND_PCM_STREAM_PLAYBACK, mode = 0, poll_fd_count = 1, poll_fd = 3, poll_events = 1, setup = -1, monotonic = 0, access = SND_PCM_ACCESS_MMAP_INTERLEAVED, format = SND_PCM_FORMAT_S32_LE, subformat = SND_PCM_SUBFORMAT_STD, channels = 2, rate = 44100, period_size = 1024, period_time = 23219, periods = {min = 0, max = 0, openmin = 0, openmax = 0, integer = 0, empty = 0}, tstamp_mode = SND_PCM_TSTAMP_NONE, period_step = 1, avail_min = 1024, period_event = 0, start_threshold = 1024, stop_threshold = 2147483647, silence_threshold = 0, silence_size = 0, boundary = 1073741824, info = 2151743747, msbits = 32, rate_num = 44100, rate_den = 1, hw_flags = 0, fifo_size = 0, buffer_size = 2048, buffer_time = {min = 0, max = 0, openmin = 0, openmax = 0, integer = 0, empty = 0}, sample_bits = 32, frame_bits = 64, appl = {master = 0x0, ptr = 0x84a7130, fd = -1, offset = 0, link_dst_count = 0, link_dst = 0x0, private_data = 0x0, changed = 0}, hw = { master = 0x0, ptr = 0x84a7138, fd = -1, offset = 0, link_dst_count = 0, link_dst = 0x0, private_data = 0x0, changed = 0}, min_align = 1, mmap_rw = 1, mmap_shadow = 0, donot_close = 0, mmap_channels = 0x84a54a0, running_areas = 0x84a51d0, stopped_areas = 0x0, ops = 0xb7c91860, fast_ops = 0xb7c918a0, op_arg = 0x84a71e0, fast_op_arg = 0x84a71e0, private_data = 0x84a7110, async_handlers = {next = 0x84a72e4, prev = 0x84a72e4}} PCM dump: Plug PCM: Linear conversion PCM (S32_LE) 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 : 2048 period_size : 1024 period_time : 23219 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1024 stop_threshold : 2147483647 silence_threshold: 0 silence_size : 0 boundary : 1073741824 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 : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 2048 period_size : 1024 period_time : 23219 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1024 stop_threshold : 2147483647 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 : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 2048 period_size : 1024 period_time : 23219 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1024 stop_threshold : 2147483647 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 23219 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 Thanks

3 3

[alsa-devel] [PATCH 00/17 v2] Update locking for snd_soc_dapm_xxxx_pin functions
by Charles Keepax 20 Feb '14

20 Feb '14

This patchset updates the locking around the snd_soc_dapm_xxxx_pin functions. First we add an unlocked version of the functions and add locking to the standard call, this also requires us to remove the few places where these functions are externally locked. Then we patch usages of the functions that look like they are expecting to do an atomic update to do so, using external locking. Thanks, Charles Updates since v1: - Added helper functions to lock/unlock the dapm mutex - Added an unlocked version of snd_soc_dapm_sync, so it can be called under the same mutex lock as the pin functions, this is not required just saves a mutex lock Charles Keepax (17): Input - arizona-haptics: Fix double lock of dapm_mutex ASoC: dapm: Add helpers to lock/unlock DAPM mutex ASoC: dapm: Add locking to snd_soc_dapm_xxxx_pin functions ASoC: dapm: Add unlocked version of snd_soc_dapm_sync ASoC: adav80x: Update locking around use of DAPM pin API ASoC: wm5100: Update locking around use of DAPM pin API ASoC: wm8962: Update locking around use of DAPM pin API ASoC: wm8994: Update locking around use of DAPM pin API ASoC: wm8996: Update locking around use of DAPM pin API ASoC: mfld_machine: Update locking around use of DAPM pin API ASoC: ams-delta: Update locking around use of DAPM pin API ASoC: omap: n810: Update locking around use of DAPM pin API ASoC: omap: rx51: Update locking around use of DAPM pin API ASoC: pxa: corgi: Update locking around use of DAPM pin API ASoC: pxa: magician: Update locking around use of DAPM pin API ASoC: pxa: spitz: Update locking around use of DAPM pin API ASoC: pxa: tosa: Update locking around use of DAPM pin API drivers/extcon/extcon-arizona.c | 12 --- drivers/input/misc/arizona-haptics.c | 19 ---- include/sound/soc-dapm.h | 9 ++ include/sound/soc.h | 9 ++ sound/soc/codecs/adav80x.c | 17 +++-- sound/soc/codecs/wm5100.c | 12 ++- sound/soc/codecs/wm8962.c | 13 ++- sound/soc/codecs/wm8994.c | 45 ++++++---- sound/soc/codecs/wm8996.c | 9 ++- sound/soc/intel/mfld_machine.c | 65 +++++++++----- sound/soc/omap/ams-delta.c | 45 ++++++---- sound/soc/omap/n810.c | 22 +++-- sound/soc/omap/rx51.c | 22 +++-- sound/soc/pxa/corgi.c | 42 +++++---- sound/soc/pxa/magician.c | 22 +++-- sound/soc/pxa/spitz.c | 51 ++++++----- sound/soc/pxa/tosa.c | 28 ++++--- sound/soc/soc-dapm.c | 160 ++++++++++++++++++++++++++++++--- 18 files changed, 399 insertions(+), 203 deletions(-) -- 1.7.2.5

4 39

[alsa-devel] [PATCH] ASoC: Intel: sst-acpi: Add support for multiple machine drivers per platform
by Jarkko Nikula 20 Feb '14

20 Feb '14

Initial implementation of this driver focused only matching SST ACPI ID with single machine driver and same firmware file per platform. It was known restriction to be improved incrementally. This patch is now changing this that SST ACPI ID refers purely to platform specific data which refers to machine drivers on this platform, not vice versa. Matching machine driver is found by looking at ACPI ID which would best match with the driver. Typically this would be the ACPI ID of audio codec but is not tied to it. This patch also changes that DSP firmware name is machine not platform specific. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- This goes on top of my "[PATCHv2] ASoC: Intel: sst-acpi: Request firmware before SST platform driver probing". http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/072812.h… --- sound/soc/intel/sst-acpi.c | 84 ++++++++++++++++++++++++++++++++-------------- 1 file changed, 58 insertions(+), 26 deletions(-) diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index 64154a77d904..c7e36c9ee52f 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -26,9 +26,20 @@ #define SST_WPT_DSP_DMA_ADDR_OFFSET 0x0FE000 #define SST_LPT_DSP_DMA_SIZE (1024 - 1) +/* Descriptor for SST ASoC machine driver */ +struct sst_acpi_mach { + /* ACPI ID for the matching machine driver. Audio codec for instance */ + const u8 id[ACPI_ID_LEN]; + /* machine driver name */ + const char *drv_name; + /* firmware file name */ + const char *fw_filename; +}; + /* Descriptor for setting up SST platform data */ struct sst_acpi_desc { const char *drv_name; + struct sst_acpi_mach *machines; /* Platform resource indexes. Must set to -1 if not used */ int resindex_lpe_base; int resindex_pcicfg_base; @@ -37,24 +48,17 @@ struct sst_acpi_desc { int resindex_dma_base; /* Unique number identifying the SST core on platform */ int sst_id; - /* firmware file name */ - const char *fw_filename; /* DMA only valid when resindex_dma_base != -1*/ int dma_engine; int dma_size; }; -/* Descriptor for SST ASoC machine driver */ -struct sst_acpi_mach { - const char *drv_name; - struct sst_acpi_desc *res_desc; -}; - struct sst_acpi_priv { struct platform_device *pdev_mach; struct platform_device *pdev_pcm; struct sst_pdata sst_pdata; struct sst_acpi_desc *desc; + struct sst_acpi_mach *mach; }; static void sst_acpi_fw_cb(const struct firmware *fw, void *context) @@ -64,10 +68,11 @@ static void sst_acpi_fw_cb(const struct firmware *fw, void *context) struct sst_acpi_priv *sst_acpi = platform_get_drvdata(pdev); struct sst_pdata *sst_pdata = &sst_acpi->sst_pdata; struct sst_acpi_desc *desc = sst_acpi->desc; + struct sst_acpi_mach *mach = sst_acpi->mach; sst_pdata->fw = fw; if (!fw) { - dev_err(dev, "Cannot load firmware %s\n", desc->fw_filename); + dev_err(dev, "Cannot load firmware %s\n", mach->fw_filename); return; } @@ -83,6 +88,28 @@ static void sst_acpi_fw_cb(const struct firmware *fw, void *context) return; } +static acpi_status sst_acpi_mach_match(acpi_handle handle, u32 level, + void *context, void **ret) +{ + *(bool *)context = true; + return AE_OK; +} + +static struct sst_acpi_mach *sst_acpi_find_machine( + struct sst_acpi_mach *machines) +{ + struct sst_acpi_mach *mach; + bool found = false; + + for (mach = machines; mach->id[0]; mach++) + if (ACPI_SUCCESS(acpi_get_devices(mach->id, + sst_acpi_mach_match, + &found, NULL)) && found) + return mach; + + return NULL; +} + static int sst_acpi_probe(struct platform_device *pdev) { const struct acpi_device_id *id; @@ -102,8 +129,13 @@ static int sst_acpi_probe(struct platform_device *pdev) if (!id) return -ENODEV; - mach = (struct sst_acpi_mach *)id->driver_data; - desc = mach->res_desc; + desc = (struct sst_acpi_desc *)id->driver_data; + mach = sst_acpi_find_machine(desc->machines); + if (mach == NULL) { + dev_err(dev, "No matching ASoC machine driver found\n"); + return -ENODEV; + } + sst_pdata = &sst_acpi->sst_pdata; sst_pdata->id = desc->sst_id; sst_acpi->desc = desc; @@ -154,7 +186,7 @@ static int sst_acpi_probe(struct platform_device *pdev) return PTR_ERR(sst_acpi->pdev_mach); /* continue SST probing after firmware is loaded */ - ret = request_firmware_nowait(THIS_MODULE, true, desc->fw_filename, + ret = request_firmware_nowait(THIS_MODULE, true, mach->fw_filename, dev, GFP_KERNEL, pdev, sst_acpi_fw_cb); if (ret) platform_device_unregister(sst_acpi->pdev_mach); @@ -175,45 +207,45 @@ static int sst_acpi_remove(struct platform_device *pdev) return 0; } +static struct sst_acpi_mach haswell_machines[] = { + { "INT33CA", "haswell-audio", "intel/IntcSST1.bin" }, + {} +}; + static struct sst_acpi_desc sst_acpi_haswell_desc = { .drv_name = "haswell-pcm-audio", + .machines = haswell_machines, .resindex_lpe_base = 0, .resindex_pcicfg_base = 1, .resindex_fw_base = -1, .irqindex_host_ipc = 0, .sst_id = SST_DEV_ID_LYNX_POINT, - .fw_filename = "intel/IntcSST1.bin", .dma_engine = SST_DMA_TYPE_DW, .resindex_dma_base = SST_LPT_DSP_DMA_ADDR_OFFSET, .dma_size = SST_LPT_DSP_DMA_SIZE, }; +static struct sst_acpi_mach broadwell_machines[] = { + { "INT343A", "broadwell-audio", "intel/IntcSST2.bin" }, + {} +}; + static struct sst_acpi_desc sst_acpi_broadwell_desc = { .drv_name = "haswell-pcm-audio", + .machines = broadwell_machines, .resindex_lpe_base = 0, .resindex_pcicfg_base = 1, .resindex_fw_base = -1, .irqindex_host_ipc = 0, .sst_id = SST_DEV_ID_WILDCAT_POINT, - .fw_filename = "intel/IntcSST2.bin", .dma_engine = SST_DMA_TYPE_DW, .resindex_dma_base = SST_WPT_DSP_DMA_ADDR_OFFSET, .dma_size = SST_LPT_DSP_DMA_SIZE, }; -static struct sst_acpi_mach haswell_mach = { - .drv_name = "haswell-audio", - .res_desc = &sst_acpi_haswell_desc, -}; - -static struct sst_acpi_mach broadwell_mach = { - .drv_name = "broadwell-audio", - .res_desc = &sst_acpi_broadwell_desc, -}; - static struct acpi_device_id sst_acpi_match[] = { - { "INT33C8", (unsigned long)&haswell_mach }, - { "INT3438", (unsigned long)&broadwell_mach }, + { "INT33C8", (unsigned long)&sst_acpi_haswell_desc }, + { "INT3438", (unsigned long)&sst_acpi_broadwell_desc }, { } }; MODULE_DEVICE_TABLE(acpi, sst_acpi_match); -- 1.8.5.3

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[alsa-devel] [PATCH] ALSA: Export snd_pcm_constraint_mask64()
by Mark Brown 20 Feb '14

20 Feb '14

From: Mark Brown <broonie(a)linaro.org> Allow modules to use it, fixing a build failure when the newly added ADAU1977 driver is built as a module. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- OK to apply this to the ASoC tree? sound/core/pcm_lib.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6630a4e..ce83def 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1240,6 +1240,7 @@ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_par return -EINVAL; return 0; } +EXPORT_SYMBOL(snd_pcm_hw_constraint_mask64); /** * snd_pcm_hw_constraint_integer - apply an integer constraint to an interval -- 1.9.0.rc3

2 2

[alsa-devel] [PATCH] ALSA: hda - Enable front audio jacks on one HP desktop model
by Hui Wang 20 Feb '14

20 Feb '14

The front headphone and mic jackes on a HP desktop model (Vendor Id: 0x111d76c7 Subsystem Id: 0x103c2b17) can not work, the codec on this machine has 8 physical ports, 6 of them are routed to rear jackes and all of them work very well, while the remaining 2 ports are routed to front headphone and mic jackes, but the corresponding pin complex node are not defined correctly. After apply this fix, the front audio jackes can work very well. BugLink: https://bugs.launchpad.net/bugs/1282369 Cc: David Henningsson <david.henningsson(a)canonical.com> Tested-by: Gerald Yang <gerald.yang(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7311bad..d75f90a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -83,7 +83,8 @@ enum { STAC_DELL_M6_BOTH, STAC_DELL_EQ, STAC_ALIENWARE_M17X, - STAC_92HD73XX_MODELS + STAC_92HD73XX_MODELS, + STAC_92HD89XX_HP_FRONT_JACK }; enum { @@ -1795,6 +1796,12 @@ static const struct hda_pintbl intel_dg45id_pin_configs[] = { {} }; +static const struct hda_pintbl stac92hd89xx_hp_front_jack_pin_configs[] = { + { 0x0a, 0x02214030 }, + { 0x0b, 0x02A19010 }, + {} +}; + static void stac92hd73xx_fixup_ref(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -1913,6 +1920,10 @@ static const struct hda_fixup stac92hd73xx_fixups[] = { [STAC_92HD73XX_NO_JD] = { .type = HDA_FIXUP_FUNC, .v.func = stac92hd73xx_fixup_no_jd, + }, + [STAC_92HD89XX_HP_FRONT_JACK] = { + .type = HDA_FIXUP_PINS, + .v.pins = stac92hd89xx_hp_front_jack_pin_configs, } }; @@ -1973,6 +1984,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = { "Alienware M17x", STAC_ALIENWARE_M17X), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490, "Alienware M17x R3", STAC_DELL_EQ), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x2b17, + "unknown HP", STAC_92HD89XX_HP_FRONT_JACK), {} /* terminator */ }; -- 1.8.1.2

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[alsa-devel] [PATCH 000/102 - chunk#2] Add const to SOC_ENUM_*_DECL() macros
by Takashi Iwai 20 Feb '14

20 Feb '14

The first patch adds the const modifier to SOC_ENUM_SINGLE_DECL() & co, and the rest are to remove superfluous const modifiers. Takashi

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[alsa-devel] [GIT PULL] ASoC fixes for v3.14
by Mark Brown 20 Feb '14

20 Feb '14

The following changes since commit 6d0abeca3242a88cab8232e4acd7e2bf088f3bc2: Linux 3.14-rc3 (2014-02-16 13:30:25 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.14-rc3 for you to fetch changes up to 895be5b31e5175bef575008aadb4f0a27b850daa: Merge remote-tracking branch 'asoc/fix/wm8993' into asoc-linus (2014-02-19 13:13:58 +0900) ---------------------------------------------------------------- ASoC: Fixes for v3.14 A few fixes, all driver speccific ones. The DaVinci ones aren't as clear as they should be from the subject lines on the commits but they fix issues which will prevent correct operation in some use cases and only affect that particular driver so are reasonably safe. ---------------------------------------------------------------- Adam Thomson (1): ASoC: da9055: Fix device registration of PMIC and CODEC devices Alexander Shiyan (1): ASoC: txx9aclc_ac97: Fix kernel crash on probe Dylan Reid (1): ASoC: max98090: sync regcache on entering STANDBY Jarkko Nikula (1): ASoC: rt5640: Add ACPI ID for Intel Baytrail Lars-Peter Clausen (1): ASoC: blackfin: Fix machine driver Kconfig dependencies Mark Brown (2): Merge remote-tracking branches 'asoc/fix/blackfin', 'asoc/fix/da9055', 'asoc/fix/davinci', 'asoc/fix/fsl', 'asoc/fix/fsl-esai', 'asoc/fix/max98090', 'asoc/fix/rt5640', 'asoc/fix/samsung' and 'asoc/fix/txx9aclc-ac97' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8993' into asoc-linus 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: Consolidate pm_runtime_get/put() use in the driver ASoC: davinci-evm: Add pm callbacks to platform driver Sachin Kamat (2): ASoC: samsung: Remove invalid dependencies ASoC: samsung: Fix trivial typo Shawn Guo (1): ASoC: fsl: fix pm support of machine drivers Stephen Warren (1): ASoC: max98090: make REVISION_ID readable Wei Yongjun (1): ASoC: wm8993: drop regulator_bulk_free of devm_ allocated data Xiubo Li (1): ASoC: fsl-esai: fix ESAI TDM slot setting drivers/mfd/da9055-i2c.c | 12 +++++- sound/soc/blackfin/Kconfig | 11 +++--- sound/soc/codecs/da9055.c | 11 +++++- sound/soc/codecs/max98090.c | 21 +++++----- sound/soc/codecs/rt5640.c | 1 + sound/soc/codecs/wm8993.c | 1 - sound/soc/davinci/davinci-evm.c | 1 + sound/soc/davinci/davinci-mcasp.c | 83 ++++++++++++++++++--------------------- sound/soc/fsl/fsl_esai.c | 4 +- sound/soc/fsl/fsl_esai.h | 2 +- sound/soc/fsl/imx-mc13783.c | 1 - sound/soc/fsl/imx-sgtl5000.c | 10 +++-- sound/soc/fsl/imx-wm8962.c | 11 ++++-- sound/soc/samsung/Kconfig | 6 +-- sound/soc/txx9/txx9aclc-ac97.c | 8 ++-- 15 files changed, 99 insertions(+), 84 deletions(-)

2 1

[alsa-devel] Need your advice: Add a new communication inteface between HD-Audio and Gfx drivers for hotplug notification/ELD update
by Lin, Mengdong 20 Feb '14

20 Feb '14

Dear audio and gfx stakeholders, We hope to add a new interface between audio and gfx driver, for gfx driver to notify audio about HDMI/DP hot-plug and ELD update. Would you please share some comments on the proposal below? Background of this issue: On Intel Haswell/Broadwell platforms, there is a HW restriction that after the display HD-Audio controller is in D3, it cannot be waken up by HDMI/DP hot-plug. Consequently, although the gfx driver can still detect the HDMI/DP hot-plug, audio driver has no idea about this and cannot notify user space whether the external HDMI/DP monitor is available for audio playback, because the audio controller cannot wake up to D0 and receive HW unsolicited event about hot-plug from the audio codec. This limitation will affect user space to decide whether we can output audio over HDMI/DP. To solve the above limitation, Takashi suggested to add a new communication interface between audio and gfx driver: create a common object containing the ops registered by both graphics and audio drivers, then communicate through it, something like vga_switcheroo. Is it okay to create this kernel object in i915 driver? I915 can export an API like "display_register_audio_client" for audio driver to register a client and hot-plug notification ops. I915 can also call some API like "display_register_gfx_client" itself and register ops for audio driver to query monitor presence and ELD info on a specific port. It would be faster for audio driver than quering ELD by command/response over the HD-A bus, thus avoid delay in i915 mode set. This will also avoid waking up the audio devices unnecessarily if the user space does not really want to use HDMI/DP for audio playback. Whenever i195 enables/disables audio on a port in modeset, it can call some API like "display_set_audio_state()" on this kernel object and trigger notifications to the audio driver. When the audio driver is probed (in the delayed probe stage), it can request i915 API symbol to register the audio client for this communication kernel object. Since the 1st i915 mode set may happen before audio driver registers the ops, we'll let audio driver check ELD once after registering the audio client ops. And for the platforms which uses this communication interface, we can disable unsolicited event for HDMI/DP hot-plug in the audio driver. We hope to hear your feedback and start to work out more details. Thanks & Best Regards Mengdong

8 21

[alsa-devel] [PATCH 000/105] Rename soc_enum.max field
by Takashi Iwai 20 Feb '14

20 Feb '14

Hi, this is the first chunk of the whole patch serires, preliminary works for renaming the confusing field name and minor clean ups in the relevant places. After this chunk, SOC_ENUM_*_DECL() macro cleanup and the actual fixes for the wrong number of enum items will follow. Then introduce SOC_ENUM_SINGLE_CONST() & co for omitting the explicit ARRAY_SIZE() call, and apply it in the later patches. Takashi

6 11

[alsa-devel] [PATCH 000/102 - chunk#3] Actually fix wrong enum items
by Takashi Iwai 20 Feb '14

20 Feb '14

Now, this patch chunk fixes the actual bugs in various codec drivers that have either off-by-one number of items or lead to array overflow. sta32x is the latter one, and this can be picked up for 3.14, too. The other off-by-one fixes would change the driver behavior (now user will see more items), thus they are potential regression risks although the "fixes" are correct, per se. Takashi

5 28

[alsa-devel] [PATCH 1/2] ASoC: alc5623: Convert to direct regmap API usage
by Mark Brown 20 Feb '14

20 Feb '14

From: Mark Brown <broonie(a)linaro.org> Convert to directly use the regmap API, allowing us to eliminate the last user of the ASoC level I/O implementations (there are still open coded I/O implementations in drivers), avoiding duplicating code in regmap. We no longer cache the entire CODEC register map on probe since the more advanced cache infrastructure in regmap is able to fill the cache on demand. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/alc5623.c | 87 ++++++++++++++++++++++++++-------------------- 1 file changed, 49 insertions(+), 38 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index d303628..36cf7e8 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -21,6 +21,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -38,26 +39,13 @@ MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)"); /* codec private data */ struct alc5623_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; u8 id; unsigned int sysclk; - u16 reg_cache[ALC5623_VENDOR_ID2+2]; unsigned int add_ctrl; unsigned int jack_det_ctrl; }; -static void alc5623_fill_cache(struct snd_soc_codec *codec) -{ - int i, step = codec->driver->reg_cache_step; - u16 *cache = codec->reg_cache; - - /* not really efficient ... */ - codec->cache_bypass = 1; - for (i = 0 ; i < codec->driver->reg_cache_size ; i += step) - cache[i] = snd_soc_read(codec, i); - codec->cache_bypass = 0; -} - static inline int alc5623_reset(struct snd_soc_codec *codec) { return snd_soc_write(codec, ALC5623_RESET, 0); @@ -869,18 +857,28 @@ static struct snd_soc_dai_driver alc5623_dai = { static int alc5623_suspend(struct snd_soc_codec *codec) { + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(alc5623->regmap, true); + return 0; } static int alc5623_resume(struct snd_soc_codec *codec) { - int i, step = codec->driver->reg_cache_step; - u16 *cache = codec->reg_cache; + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + int ret; /* Sync reg_cache with the hardware */ - for (i = 2 ; i < codec->driver->reg_cache_size ; i += step) - snd_soc_write(codec, i, cache[i]); + regcache_cache_only(alc5623->regmap, false); + ret = regcache_sync(alc5623->regmap); + if (ret != 0) { + dev_err(codec->dev, "Failed to sync register cache: %d\n", + ret); + regcache_cache_only(alc5623->regmap, true); + return ret; + } alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -900,14 +898,14 @@ static int alc5623_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type); + codec->control_data = alc5623->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; } alc5623_reset(codec); - alc5623_fill_cache(codec); /* power on device */ alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -980,9 +978,15 @@ static struct snd_soc_codec_driver soc_codec_device_alc5623 = { .suspend = alc5623_suspend, .resume = alc5623_resume, .set_bias_level = alc5623_set_bias_level, - .reg_cache_size = ALC5623_VENDOR_ID2+2, - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, +}; + +static const struct regmap_config alc5623_regmap = { + .reg_bits = 8, + .val_bits = 16, + .reg_stride = 2, + + .max_register = ALC5623_VENDOR_ID2, + .cache_type = REGCACHE_RBTREE, }; /* @@ -996,19 +1000,32 @@ static int alc5623_i2c_probe(struct i2c_client *client, { struct alc5623_platform_data *pdata; struct alc5623_priv *alc5623; - int ret, vid1, vid2; + unsigned int vid1, vid2; + int ret; - vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1); - if (vid1 < 0) { - dev_err(&client->dev, "failed to read I2C\n"); - return -EIO; + alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv), + GFP_KERNEL); + if (alc5623 == NULL) + return -ENOMEM; + + alc5623->regmap = devm_regmap_init_i2c(client, &alc5623_regmap); + if (IS_ERR(alc5623->regmap)) { + ret = PTR_ERR(alc5623->regmap); + dev_err(&client->dev, "Failed to initialise I/O: %d\n", ret); + return ret; + } + + ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID1, &vid1); + if (ret < 0) { + dev_err(&client->dev, "failed to read vendor ID1: %d\n", ret); + return ret; } vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); - vid2 = i2c_smbus_read_byte_data(client, ALC5623_VENDOR_ID2); - if (vid2 < 0) { - dev_err(&client->dev, "failed to read I2C\n"); - return -EIO; + ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID2, &vid2); + if (ret < 0) { + dev_err(&client->dev, "failed to read vendor ID2: %d\n", ret); + return ret; } if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) { @@ -1021,11 +1038,6 @@ static int alc5623_i2c_probe(struct i2c_client *client, dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2); - alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv), - GFP_KERNEL); - if (alc5623 == NULL) - return -ENOMEM; - pdata = client->dev.platform_data; if (pdata) { alc5623->add_ctrl = pdata->add_ctrl; @@ -1048,7 +1060,6 @@ static int alc5623_i2c_probe(struct i2c_client *client, } i2c_set_clientdata(client, alc5623); - alc5623->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&client->dev, &soc_codec_device_alc5623, &alc5623_dai, 1); -- 1.9.0.rc3

1 1

[alsa-devel] About 'SNDRV_PCM_CLASS_MULTI' for devices with multi-channels
by Takashi Sakamoto 19 Feb '14

19 Feb '14

Hi all, I have two questions about the usage of this macro. Should I use this macro for new drivers which I'm working for?' http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071820.ht… My drivers support some firewire devices, which is desinged for studio-use and have many channels. Its combination of channels are not designed for surround sound like 5.1ch. Just judging from its name, 'SNDRV_PCM_CLASS_MULTI', I should use this. But there are no drivers which use this macro in current tree, against my expectation that some drivers (ICE1712-pro, RME9652, 6fire usb and so on) should use this macro. Are there any reasons? Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

1 0

[alsa-devel] About 'SNDRV_PCM_CLASS_MULTI' for devices with multi-channels
by Takashi Sakamoto 19 Feb '14

19 Feb '14

Hi all, I have two questions about the usage of this macro. Should I use this macro for new drivers which I'm working for?' http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071820.ht… My drivers support some firewire devices, which is desinged for studio-use and have many channels. Its combination of channels are not designed for surround sound like 5.1ch. Just judging from its name, 'SNDRV_PCM_CLASS_MULTI', I should use this. But there are no drivers which use this macro in current tree, against my expectation that some drivers (ICE1712-pro, RME9652, 6fire usb and so on) should use this macro. Are there any reasons? Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

1 0

[alsa-devel] [PATCHv2] ASoC: Intel: Fix build for sst-dsp.c on PPC architecture
by Liam Girdwood 19 Feb '14

19 Feb '14

Disable build on non X86 architectures except for compile testing. This fixes the following build errors on PPC and adds an option for testing the build on other architectures as suggested by Mark Brown :- sound/soc/intel/sst-dsp.c: In function 'sst_dsp_outbox_write': sound/soc/intel/sst-dsp.c:218:2: error: implicit declaration of function 'memcpy_toio' [-Werror=implicit-function-declaration] memcpy_toio(sst->mailbox.out_base, message, bytes); ^ sound/soc/intel/sst-dsp.c: In function 'sst_dsp_outbox_read': sound/soc/intel/sst-dsp.c:231:2: error: implicit declaration of function 'memcpy_fromio' [-Werror=implicit-function-declaration] memcpy_fromio(message, sst->mailbox.out_base, bytes); ^ Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/Kconfig | 1 + sound/soc/intel/sst-dsp.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index c962432..4bb802b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -15,6 +15,7 @@ config SND_SST_MFLD_PLATFORM config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI + depends on (X86 || COMPILE_TEST) help This adds support for Intel(R) Smart Sound Technology (SST). Say Y if you have such a device diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c index e0ad2e5..6e22c12 100644 --- a/sound/soc/intel/sst-dsp.c +++ b/sound/soc/intel/sst-dsp.c @@ -19,6 +19,7 @@ #include <linux/interrupt.h> #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/io.h> #include "sst-dsp.h" #include "sst-dsp-priv.h" -- 1.8.3.2

2 1

[alsa-devel] About 'SNDRV_PCM_CLASS_MULTI' for devices with multi-channels
by Takashi Sakamoto 19 Feb '14

19 Feb '14

Hi all, I have two questions about the usage of this macro. Should I use this macro for new drivers which I'm working for?' http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071820.ht… My drivers support some firewire devices, which is desinged for studio-use and have many channels. Its combination of channels are not designed for surround sound like 5.1ch. Just judging from its name, 'SNDRV_PCM_CLASS_MULTI', I should use this. But there are no drivers which use this macro in current tree, against my expectation that some drivers (ICE1712-pro, RME9652, 6fire usb and so on) should use this macro. Are there any reasons? Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

1 0

[alsa-devel] [PATCH] ASoC: cs4271: claim reset GPIO in bus probe function
by Daniel Mack 19 Feb '14

19 Feb '14

Move the GPIO acquisition from the codec to the bus probe functions. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/cs4271.c | 60 +++++++++++++++++++++++++++++++---------------- 1 file changed, 40 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index ce05fd9..6bfe021 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -540,14 +540,10 @@ static int cs4271_probe(struct snd_soc_codec *codec) struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; int ret; - int gpio_nreset = -EINVAL; bool amutec_eq_bmutec = false; #ifdef CONFIG_OF if (of_match_device(cs4271_dt_ids, codec->dev)) { - gpio_nreset = of_get_named_gpio(codec->dev->of_node, - "reset-gpio", 0); - if (of_get_property(codec->dev->of_node, "cirrus,amutec-eq-bmutec", NULL)) amutec_eq_bmutec = true; @@ -559,27 +555,19 @@ static int cs4271_probe(struct snd_soc_codec *codec) #endif if (cs4271plat) { - if (gpio_is_valid(cs4271plat->gpio_nreset)) - gpio_nreset = cs4271plat->gpio_nreset; - amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec; cs4271->enable_soft_reset = cs4271plat->enable_soft_reset; } - if (gpio_nreset >= 0) - if (devm_gpio_request(codec->dev, gpio_nreset, "CS4271 Reset")) - gpio_nreset = -EINVAL; - if (gpio_nreset >= 0) { + if (gpio_is_valid(cs4271->gpio_nreset)) { /* Reset codec */ - gpio_direction_output(gpio_nreset, 0); + gpio_direction_output(cs4271->gpio_nreset, 0); udelay(1); - gpio_set_value(gpio_nreset, 1); + gpio_set_value(cs4271->gpio_nreset, 1); /* Give the codec time to wake up */ udelay(1); } - cs4271->gpio_nreset = gpio_nreset; - ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, CS4271_MODE2_PDN | CS4271_MODE2_CPEN, CS4271_MODE2_PDN | CS4271_MODE2_CPEN); @@ -641,14 +629,45 @@ static const struct regmap_config cs4271_spi_regmap = { .volatile_reg = cs4271_volatile_reg, }; -static int cs4271_spi_probe(struct spi_device *spi) +static int cs4271_common_probe(struct device *dev, + struct cs4271_private **c) { + struct cs4271_platform_data *cs4271plat = dev->platform_data; struct cs4271_private *cs4271; - cs4271 = devm_kzalloc(&spi->dev, sizeof(*cs4271), GFP_KERNEL); + cs4271 = devm_kzalloc(dev, sizeof(*cs4271), GFP_KERNEL); if (!cs4271) return -ENOMEM; + if (of_match_device(cs4271_dt_ids, dev)) + cs4271->gpio_nreset = + of_get_named_gpio(dev->of_node, "reset-gpio", 0); + + if (cs4271plat) + cs4271->gpio_nreset = cs4271plat->gpio_nreset; + + if (gpio_is_valid(cs4271->gpio_nreset)) { + int ret; + + ret = devm_gpio_request(dev, cs4271->gpio_nreset, + "CS4271 Reset"); + if (ret < 0) + return ret; + } + + *c = cs4271; + return 0; +} + +static int cs4271_spi_probe(struct spi_device *spi) +{ + struct cs4271_private *cs4271; + int ret; + + ret = cs4271_common_probe(&spi->dev, &cs4271); + if (ret < 0) + return ret; + spi_set_drvdata(spi, cs4271); cs4271->regmap = devm_regmap_init_spi(spi, &cs4271_spi_regmap); if (IS_ERR(cs4271->regmap)) @@ -698,10 +717,11 @@ static int cs4271_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct cs4271_private *cs4271; + int ret; - cs4271 = devm_kzalloc(&client->dev, sizeof(*cs4271), GFP_KERNEL); - if (!cs4271) - return -ENOMEM; + ret = cs4271_common_probe(&client->dev, &cs4271); + if (ret < 0) + return ret; i2c_set_clientdata(client, cs4271); cs4271->regmap = devm_regmap_init_i2c(client, &cs4271_i2c_regmap); -- 1.8.5.3

4 5

[alsa-devel] [PATCH v4] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 19 Feb '14

19 Feb '14

This patch adds initial support for the Behringer BCD2000 USB DJ controller. At the moment, only the MIDI part of the device is working, i.e. knobs, buttons and LEDs. I also plan to add support for the audio part, but I assume that this will require more effort than the rather simple MIDI interface. Progress can be tracked at https://github.com/anyc/snd-usb-bcd2000. Changes since v3: - applied style and snd_printk changes as suggested by Daniel Mack Changes since v2: - applied more changes from Daniel Mack and Clemens Ladisch Changes since v1: - fixed the various code style issues, thanks to Daniel Mack and checkpatch.pl. Signed-off-by: Mario Kicherer <dev(a)kicherer.org> --- sound/usb/Kconfig | 13 ++ sound/usb/Makefile | 2 +- sound/usb/bcd2000/Makefile | 3 + sound/usb/bcd2000/bcd2000.c | 488 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 505 insertions(+), 1 deletion(-) create mode 100644 sound/usb/bcd2000/Makefile create mode 100644 sound/usb/bcd2000/bcd2000.c diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index de9408b..145653a 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -146,5 +146,18 @@ config SND_USB_HIFACE To compile this driver as a module, choose M here: the module will be called snd-usb-hiface. +config SND_BCD2000 + tristate "Behringer BCD2000 MIDI driver" + select SND_RAWMIDI + help + Say Y here to include MIDI support for the Behringer BCD2000 DJ + controller. + + Audio support is still work-in-progress at + https://github.com/anyc/snd-usb-bcd2000 + + To compile this driver as a module, choose M here: the module + will be called snd-bcd2000. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index abe668f..2b92f0d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ bcd2000/ diff --git a/sound/usb/bcd2000/Makefile b/sound/usb/bcd2000/Makefile new file mode 100644 index 0000000..f09ccc0 --- /dev/null +++ b/sound/usb/bcd2000/Makefile @@ -0,0 +1,3 @@ +snd-bcd2000-y := bcd2000.o + +obj-$(CONFIG_SND_BCD2000) += snd-bcd2000.o \ No newline at end of file diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c new file mode 100644 index 0000000..6965a89 --- /dev/null +++ b/sound/usb/bcd2000/bcd2000.c @@ -0,0 +1,488 @@ +/* + * Behringer BCD2000 driver + * + * Copyright (C) 2014 Mario Kicherer (dev(a)kicherer.org) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/usb.h> +#include <linux/usb/audio.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/rawmidi.h> + +#define DEVICE_NAME "Behringer BCD2000" +#define DEVICE_SHORTNAME "bcd2000" + +#define PREFIX DEVICE_SHORTNAME ": " +#define BUFSIZE 64 + +static struct usb_device_id id_table[] = { + { USB_DEVICE(0x1397, 0x00bd) }, + { }, +}; + +static unsigned char device_cmd_prefix[] = {0x03, 0x00}; + +static unsigned char bcd2000_init_sequence[] = { + 0x07, 0x00, 0x00, 0x00, 0x78, 0x48, 0x1c, 0x81, + 0xc4, 0x00, 0x00, 0x00, 0x5e, 0x53, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff, 0x6c, 0xf3, 0x90, 0xff, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, + 0x18, 0xfa, 0x11, 0xff, 0x14, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0xf2, 0x34, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff +}; + +struct bcd2000 { + struct usb_device *dev; + struct snd_card *card; + struct usb_interface *intf; + int card_index; + + int midi_out_active; + struct snd_rawmidi *rmidi; + struct snd_rawmidi_substream *midi_receive_substream; + struct snd_rawmidi_substream *midi_out_substream; + + unsigned char midi_in_buf[BUFSIZE]; + unsigned char midi_out_buf[BUFSIZE]; + + struct urb *midi_out_urb; + struct urb *midi_in_urb; + + struct usb_anchor anchor; +}; + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +static DEFINE_MUTEX(devices_mutex); +static unsigned int devices_used; +static struct usb_driver bcd2000_driver; + +#ifdef CONFIG_SND_DEBUG +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) +{ + print_hex_dump(KERN_DEBUG, prefix, + DUMP_PREFIX_NONE, 16, 1, + buf, len, false); +} +#else +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) {} +#endif + +static int bcd2000_midi_input_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_input_close(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +/* register midi substream */ +static void bcd2000_midi_input_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + bcd2k->midi_receive_substream = up ? substream : NULL; +} + +static void bcd2000_midi_handle_input(struct bcd2000 *bcd2k, + const unsigned char *buf, unsigned int len) +{ + unsigned int length, tocopy; + struct snd_rawmidi_substream *midi_receive_substream; + + midi_receive_substream = ACCESS_ONCE(bcd2k->midi_receive_substream); + if (!midi_receive_substream) + return; + + bcd2000_dump_buffer(PREFIX "received from device: ", buf, len); + + if (len < 2) + return; + + /* + * Packet structure: mm nn oo (pp) + * mm: payload length + * nn: MIDI command or note + * oo: note or velocity + * pp: velocity + */ + + length = buf[0]; + + /* ignore packets without payload */ + if (length == 0) + return; + + tocopy = min(length, len-1); + + bcd2000_dump_buffer(PREFIX "sending to userspace: ", + &buf[1], tocopy); + + snd_rawmidi_receive(midi_receive_substream, + &buf[1], tocopy); +} + +static void bcd2000_midi_send(struct bcd2000 *bcd2k, + struct snd_rawmidi_substream *substream) +{ + int len, ret; + + BUILD_BUG_ON(sizeof(device_cmd_prefix) >= BUFSIZE); + + /* copy the "set LED" command bytes */ + memcpy(bcd2k->midi_out_buf, device_cmd_prefix, + sizeof(device_cmd_prefix)); + + /* + * get MIDI packet and leave space for command prefix + * and payload length + */ + len = snd_rawmidi_transmit(substream, + bcd2k->midi_out_buf + 3, BUFSIZE - 3); + + if (len < 0) + dev_err(&bcd2k->dev->dev, "%s: snd_rawmidi_transmit error %d\n", + __func__, len); + + if (len <= 0) + return; + + /* set payload length */ + bcd2k->midi_out_buf[2] = len; + bcd2k->midi_out_urb->transfer_buffer_length = BUFSIZE; + + bcd2000_dump_buffer(PREFIX "sending to device: ", + bcd2k->midi_out_buf, len+3); + + /* send packet to the BCD2000 */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s (%p): usb_submit_urb() failed, ret=%d, len=%d\n", + __func__, substream, ret, len); + else + bcd2k->midi_out_active = 1; +} + +static int bcd2000_midi_output_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_output_close(struct snd_rawmidi_substream *substream) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (bcd2k->midi_out_active) { + usb_kill_urb(bcd2k->midi_out_urb); + bcd2k->midi_out_active = 0; + } + + return 0; +} + +/* register midi substream */ +static void bcd2000_midi_output_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (up) { + bcd2k->midi_out_substream = substream; + /* check if there is data userspace wants to send */ + if (!bcd2k->midi_out_active) + bcd2000_midi_send(bcd2k, substream); + } else { + bcd2k->midi_out_substream = NULL; + } +} + +static void bcd2000_output_complete(struct urb *urb) +{ + struct bcd2000 *bcd2k = urb->context; + struct snd_rawmidi_substream *midi_out_substream; + + bcd2k->midi_out_active = 0; + + if (urb->status != 0) + dev_err(&urb->dev->dev, + PREFIX "output urb->status: %d\n", urb->status); + + midi_out_substream = ACCESS_ONCE(bcd2k->midi_out_substream); + if (!midi_out_substream) + return; + + /* check if there is more data userspace wants to send */ + bcd2000_midi_send(bcd2k, midi_out_substream); +} + +static void bcd2000_input_complete(struct urb *urb) +{ + int ret; + struct device *dev = &urb->dev->dev; + struct bcd2000 *bcd2k = urb->context; + + if (urb->status) { + dev_warn(dev, PREFIX "input urb->status: %i\n", urb->status); + return; + } + + if (!bcd2k) + return; + + if (urb->actual_length > 0) + bcd2000_midi_handle_input(bcd2k, urb->transfer_buffer, + urb->actual_length); + + /* acknowledge received packet */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(dev, "unable to submit urb. OOM!?\n"); +} + +static struct snd_rawmidi_ops bcd2000_midi_output = { + .open = bcd2000_midi_output_open, + .close = bcd2000_midi_output_close, + .trigger = bcd2000_midi_output_trigger, +}; + +static struct snd_rawmidi_ops bcd2000_midi_input = { + .open = bcd2000_midi_input_open, + .close = bcd2000_midi_input_close, + .trigger = bcd2000_midi_input_trigger, +}; + +static void bcd2000_init_device(struct bcd2000 *bcd2k) +{ + int ret; + + init_usb_anchor(&bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_out_urb, &bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_in_urb, &bcd2k->anchor); + + /* copy init sequence into buffer */ + memcpy(bcd2k->midi_out_buf, bcd2000_init_sequence, 52); + bcd2k->midi_out_urb->transfer_buffer_length = 52; + + /* submit sequence */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() out failed, ret=%d: ", + __func__, ret); + else + bcd2k->midi_out_active = 1; + + /* send empty packet to enable button and controller events */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() in failed, ret=%d: ", + __func__, ret); + + /* ensure initialization is finished */ + usb_wait_anchor_empty_timeout(&bcd2k->anchor, 1000); +} + +static int bcd2000_init_midi(struct bcd2000 *bcd2k) +{ + int ret; + struct snd_rawmidi *rmidi; + + ret = snd_rawmidi_new(bcd2k->card, bcd2k->card->shortname, 0, + 1, /* output */ + 1, /* input */ + &rmidi); + + if (ret < 0) + return ret; + + strlcpy(rmidi->name, bcd2k->card->shortname, sizeof(rmidi->name)); + + rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; + rmidi->private_data = bcd2k; + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &bcd2000_midi_output); + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &bcd2000_midi_input); + + bcd2k->rmidi = rmidi; + + bcd2k->midi_in_urb = usb_alloc_urb(0, GFP_KERNEL); + bcd2k->midi_out_urb = usb_alloc_urb(0, GFP_KERNEL); + + if (!bcd2k->midi_in_urb || !bcd2k->midi_out_urb) { + dev_err(&bcd2k->dev->dev, PREFIX "usb_alloc_urb failed\n"); + return -ENOMEM; + } + + usb_fill_int_urb(bcd2k->midi_in_urb, bcd2k->dev, + usb_rcvintpipe(bcd2k->dev, 0x81), + bcd2k->midi_in_buf, BUFSIZE, + bcd2000_input_complete, bcd2k, 1); + + usb_fill_int_urb(bcd2k->midi_out_urb, bcd2k->dev, + usb_sndintpipe(bcd2k->dev, 0x1), + bcd2k->midi_out_buf, BUFSIZE, + bcd2000_output_complete, bcd2k, 1); + + bcd2000_init_device(bcd2k); + + return 0; +} + +static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k, + struct usb_interface *interface) +{ + /* usb_kill_urb not necessary, urb is aborted automatically */ + + usb_free_urb(bcd2k->midi_out_urb); + usb_free_urb(bcd2k->midi_in_urb); + + if (bcd2k->intf) { + usb_set_intfdata(bcd2k->intf, NULL); + bcd2k->intf = NULL; + } +} + +static int bcd2000_probe(struct usb_interface *interface, + const struct usb_device_id *usb_id) +{ + struct snd_card *card; + struct bcd2000 *bcd2k; + unsigned int card_index; + char usb_path[32]; + int err; + + mutex_lock(&devices_mutex); + + for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) + if (enable[card_index] && !(devices_used & (1 << card_index))) + break; + + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + + err = snd_card_create(index[card_index], id[card_index], THIS_MODULE, + sizeof(*bcd2k), &card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return err; + } + + bcd2k = card->private_data; + bcd2k->dev = interface_to_usbdev(interface); + bcd2k->card = card; + bcd2k->card_index = card_index; + bcd2k->intf = interface; + + snd_card_set_dev(card, &interface->dev); + + strcpy(card->driver, DEVICE_NAME); + strcpy(card->shortname, DEVICE_SHORTNAME); + usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); + snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), + DEVICE_NAME ", at %s", + usb_path); + + dev_info(&bcd2k->dev->dev, PREFIX "%s", bcd2k->card->longname); + + err = bcd2000_init_midi(bcd2k); + if (err < 0) + goto probe_error; + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + usb_set_intfdata(interface, bcd2k); + devices_used |= 1 << card_index; + + mutex_unlock(&devices_mutex); + return 0; + +probe_error: + dev_info(&bcd2k->dev->dev, PREFIX "error during probing"); + bcd2000_free_usb_related_resources(bcd2k, interface); + snd_card_free(card); + mutex_unlock(&devices_mutex); + return err; +} + +static void bcd2000_disconnect(struct usb_interface *interface) +{ + struct bcd2000 *bcd2k = usb_get_intfdata(interface); + + if (!bcd2k) + return; + + mutex_lock(&devices_mutex); + + /* make sure that userspace cannot create new requests */ + snd_card_disconnect(bcd2k->card); + + bcd2000_free_usb_related_resources(bcd2k, interface); + + devices_used &= ~(1 << bcd2k->card_index); + + snd_card_free_when_closed(bcd2k->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_driver bcd2000_driver = { + .name = DEVICE_SHORTNAME, + .probe = bcd2000_probe, + .disconnect = bcd2000_disconnect, + .id_table = id_table, +}; + +static int __init bcd2000_init(void) +{ + int retval = 0; + + retval = usb_register(&bcd2000_driver); + if (retval) + pr_info(PREFIX "usb_register failed. Error: %d", retval); + return retval; +} + +static void __exit bcd2000_exit(void) +{ + usb_deregister(&bcd2000_driver); +} + +module_init(bcd2000_init); +module_exit(bcd2000_exit); + +MODULE_DEVICE_TABLE(usb, id_table); +MODULE_AUTHOR("Mario Kicherer, dev(a)kicherer.org"); +MODULE_DESCRIPTION("Behringer BCD2000 driver"); +MODULE_LICENSE("GPL"); -- 1.8.3.2

3 2

[alsa-devel] [PATCHv2] ASoC: Intel: Fix sparse warnings for firmware loader.
by Liam Girdwood 19 Feb '14

19 Feb '14

Changes since V1 :- o Fixed sparse warning for block_module_remove o Fixed additional sparse warnings when building entire driver. Sparse gives us the following warnings on sst-firmware.c CHECK sound/soc/intel/sst-firmware.c sound/soc/intel/sst-firmware.c:39:34: warning: incorrect type in argument 1 (different address spaces) sound/soc/intel/sst-firmware.c:39:34: expected void volatile [noderef] <asn:2>*dst sound/soc/intel/sst-firmware.c:39:34: got void * sound/soc/intel/sst-firmware.c:417:36: warning: incorrect type in argument 1 (different address spaces) sound/soc/intel/sst-firmware.c:417:36: expected void *dest sound/soc/intel/sst-firmware.c:417:36: got void [noderef] <asn:2>* sound/soc/intel/sst-firmware.c:430:5: warning: symbol 'sst_block_module_remove' was not declared. Should it be static? and CC [M] sound/soc/intel/sst-dsp.o sound/soc/intel/sst-dsp-priv.h:271:53: warning: incorrect type in argument 3 (different address spaces) sound/soc/intel/sst-dsp-priv.h:271:53: expected void *src sound/soc/intel/sst-dsp-priv.h:271:53: got void [noderef] <asn:2>* sound/soc/intel/sst-dsp-priv.h:271:53: warning: incorrect type in argument 3 (different address spaces) sound/soc/intel/sst-dsp-priv.h:271:53: expected void *src sound/soc/intel/sst-dsp-priv.h:271:53: got void [noderef] <asn:2>* sound/soc/intel/sst-dsp-priv.h:271:53: warning: incorrect type in argument 3 (different address spaces) sound/soc/intel/sst-dsp-priv.h:271:53: expected void *src sound/soc/intel/sst-dsp-priv.h:271:53: got void [noderef] <asn:2>* This patch removes these warnings Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-dsp-priv.h | 7 +++++-- sound/soc/intel/sst-firmware.c | 2 +- 2 files changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h index 6135564..fe8e81a 100644 --- a/sound/soc/intel/sst-dsp-priv.h +++ b/sound/soc/intel/sst-dsp-priv.h @@ -41,8 +41,10 @@ struct sst_ops { u64 (*read64)(void __iomem *addr, u32 offset); /* DSP I/DRAM IO */ - void (*ram_read)(struct sst_dsp *sst, void *dest, void *src, size_t bytes); - void (*ram_write)(struct sst_dsp *sst, void *dest, void *src, size_t bytes); + void (*ram_read)(struct sst_dsp *sst, void *dest, void __iomem *src, + size_t bytes); + void (*ram_write)(struct sst_dsp *sst, void __iomem *dest, void *src, + size_t bytes); void (*dump)(struct sst_dsp *); @@ -296,6 +298,7 @@ struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id); struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp); void sst_mem_block_free_scratch(struct sst_dsp *dsp, struct sst_module *scratch); +int sst_block_module_remove(struct sst_module *module); /* Register the DSPs memory blocks - would be nice to read from ACPI */ struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset, diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 31cd154..dee7eb5 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -30,7 +30,7 @@ #include "sst-dsp.h" #include "sst-dsp-priv.h" -static void sst_memcpy32(void *dest, void *src, u32 bytes) +static void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes) { u32 i; -- 1.8.3.2

3 9

[alsa-devel] About 'SNDRV_PCM_CLASS_MULTI' for devices with multi-channels
by Takashi Sakamoto 19 Feb '14

19 Feb '14

Hi all, I have two questions about the usage of this macro. Should I use this macro for new drivers which I'm working for?' http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071820.ht… My drivers support some firewire devices, which is desinged for studio-use and have many channels. Its combination of channels are not designed for surround sound like 5.1ch. Just judging from its name, 'SNDRV_PCM_CLASS_MULTI', I should use this. But there are no drivers which use this macro in current tree, against my expectation that some drivers (ICE1712-pro, RME9652, 6fire usb and so on) should use this macro. Are there any reasons? Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

1 0

[alsa-devel] About 'SNDRV_PCM_CLASS_MULTI' for devices with multi-channels
by Takashi Sakamoto 19 Feb '14

19 Feb '14

Hi all, I have two questions about the usage of this macro. Should I use this macro for new drivers which I'm working for?' http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071820.ht… My drivers support some firewire devices, which is desinged for studio-use and have many channels. Its combination of channels are not designed for surround sound like 5.1ch. Just judging from its name, 'SNDRV_PCM_CLASS_MULTI', I should use this. But there are no drivers which use this macro in current tree, against my expectation that some drivers (ICE1712-pro, RME9652, 6fire usb and so on) should use this macro. Are there any reasons? Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

1 0

[alsa-devel] [PATCHv2] ASoC: Intel: sst-acpi: Request firmware before SST platform driver probing
by Jarkko Nikula 19 Feb '14

19 Feb '14

We originally thought to request SST audio DSP firmware during the SST platform driver initialization. However plain request_firmware doesn't work in driver probe paths if userspace is not ready to handle it. For instance when drivers are built-in. Implementing asynchronous firmware request in SST platform driver initialization complicates code needlessly since it anyway will fail if firmware is missing. This is more simple to handle by requesting firmware asynchronously in sst_acpi_probe() and register SST platform only after firmware is loaded. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- V2: - Fix leak in sst_acpi_probe that sst_acpi->pdev_mach wasn't unregistered in case of request_firmware_nowait fails - Test is sst_acpi->pdev_pcm valid before unregistering it in sst_acpi_remove. It can contain not only NULL or valid pointer but an error code too --- sound/soc/intel/sst-acpi.c | 56 ++++++++++++++++++++++++++++++++-------------- sound/soc/intel/sst-dsp.h | 2 +- 2 files changed, 40 insertions(+), 18 deletions(-) diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index aba73ca8a923..64154a77d904 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -16,6 +16,7 @@ #include <linux/acpi.h> #include <linux/device.h> +#include <linux/firmware.h> #include <linux/module.h> #include <linux/platform_device.h> @@ -56,6 +57,32 @@ struct sst_acpi_priv { struct sst_acpi_desc *desc; }; +static void sst_acpi_fw_cb(const struct firmware *fw, void *context) +{ + struct platform_device *pdev = context; + struct device *dev = &pdev->dev; + struct sst_acpi_priv *sst_acpi = platform_get_drvdata(pdev); + struct sst_pdata *sst_pdata = &sst_acpi->sst_pdata; + struct sst_acpi_desc *desc = sst_acpi->desc; + + sst_pdata->fw = fw; + if (!fw) { + dev_err(dev, "Cannot load firmware %s\n", desc->fw_filename); + return; + } + + /* register PCM and DAI driver */ + sst_acpi->pdev_pcm = + platform_device_register_data(dev, desc->drv_name, -1, + sst_pdata, sizeof(*sst_pdata)); + if (IS_ERR(sst_acpi->pdev_pcm)) { + dev_err(dev, "Cannot register device %s. Error %d\n", + desc->drv_name, (int)PTR_ERR(sst_acpi->pdev_pcm)); + } + + return; +} + static int sst_acpi_probe(struct platform_device *pdev) { const struct acpi_device_id *id; @@ -79,7 +106,6 @@ static int sst_acpi_probe(struct platform_device *pdev) desc = mach->res_desc; sst_pdata = &sst_acpi->sst_pdata; sst_pdata->id = desc->sst_id; - sst_pdata->fw_filename = desc->fw_filename; sst_acpi->desc = desc; if (desc->resindex_dma_base >= 0) { @@ -118,37 +144,33 @@ static int sst_acpi_probe(struct platform_device *pdev) } } - /* register PCM and DAI driver */ - sst_acpi->pdev_pcm = - platform_device_register_data(dev, desc->drv_name, -1, - sst_pdata, sizeof(*sst_pdata)); - if (IS_ERR(sst_acpi->pdev_pcm)) - return PTR_ERR(sst_acpi->pdev_pcm); - - /* register machine driver */ platform_set_drvdata(pdev, sst_acpi); + /* register machine driver */ sst_acpi->pdev_mach = platform_device_register_data(dev, mach->drv_name, -1, sst_pdata, sizeof(*sst_pdata)); - if (IS_ERR(sst_acpi->pdev_mach)) { - ret = PTR_ERR(sst_acpi->pdev_mach); - goto sst_err; - } + if (IS_ERR(sst_acpi->pdev_mach)) + return PTR_ERR(sst_acpi->pdev_mach); - return ret; + /* continue SST probing after firmware is loaded */ + ret = request_firmware_nowait(THIS_MODULE, true, desc->fw_filename, + dev, GFP_KERNEL, pdev, sst_acpi_fw_cb); + if (ret) + platform_device_unregister(sst_acpi->pdev_mach); -sst_err: - platform_device_unregister(sst_acpi->pdev_pcm); return ret; } static int sst_acpi_remove(struct platform_device *pdev) { struct sst_acpi_priv *sst_acpi = platform_get_drvdata(pdev); + struct sst_pdata *sst_pdata = &sst_acpi->sst_pdata; platform_device_unregister(sst_acpi->pdev_mach); - platform_device_unregister(sst_acpi->pdev_pcm); + if (!IS_ERR_OR_NULL(sst_acpi->pdev_pcm)) + platform_device_unregister(sst_acpi->pdev_pcm); + release_firmware(sst_pdata->fw); return 0; } diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index d134359fecac..3730fd324455 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -152,7 +152,7 @@ struct sst_pdata { int irq; /* Firmware */ - const char *fw_filename; + const struct firmware *fw; /* DMA */ u32 dma_base; -- 1.8.5.3

3 3

[alsa-devel] [PATCH] ASoC: cs4271: free reset gpio in cs4271_remove()
by Daniel Mack 19 Feb '14

19 Feb '14

The reset GPIO line is acquired in cs4271_probe(), so it should be freed from cs4271_remove() as well. This way, the driver can unloaded and reloaded more than once. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/cs4271.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index ce05fd9..aeaa5c6 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -604,9 +604,11 @@ static int cs4271_remove(struct snd_soc_codec *codec) { struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - if (gpio_is_valid(cs4271->gpio_nreset)) + if (gpio_is_valid(cs4271->gpio_nreset)) { /* Set codec to the reset state */ gpio_set_value(cs4271->gpio_nreset, 0); + devm_gpio_free(codec->dev, cs4271->gpio_nreset); + } return 0; }; -- 1.8.5.3

4 6

[alsa-devel] [PATCH] ALSA: core: Fix missing card sysfs contents
by Takashi Iwai 19 Feb '14

19 Feb '14

While moving the card device into struct snd_card, the reference to the assigned card in sysfs show/store callbacks were forgotten to be refreshed, still accessing to the no longer used drvdata. Fix these places to refer correctly via container_of(). Also, remove the superfluous NULL checks since it's guaranteed to be non-NULL now. Fixes: 8bfb181c17d2 ('ALSA: Embed card device into struct snd_card') Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/init.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 0095a530ff12..b7085eb19079 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -624,15 +624,15 @@ static ssize_t card_id_show_attr(struct device *dev, struct device_attribute *attr, char *buf) { - struct snd_card *card = dev_get_drvdata(dev); - return snprintf(buf, PAGE_SIZE, "%s\n", card ? card->id : "(null)"); + struct snd_card *card = container_of(dev, struct snd_card, card_dev); + return snprintf(buf, PAGE_SIZE, "%s\n", card->id); } static ssize_t card_id_store_attr(struct device *dev, struct device_attribute *attr, const char *buf, size_t count) { - struct snd_card *card = dev_get_drvdata(dev); + struct snd_card *card = container_of(dev, struct snd_card, card_dev); char buf1[sizeof(card->id)]; size_t copy = count > sizeof(card->id) - 1 ? sizeof(card->id) - 1 : count; @@ -664,8 +664,8 @@ static ssize_t card_number_show_attr(struct device *dev, struct device_attribute *attr, char *buf) { - struct snd_card *card = dev_get_drvdata(dev); - return snprintf(buf, PAGE_SIZE, "%i\n", card ? card->number : -1); + struct snd_card *card = container_of(dev, struct snd_card, card_dev); + return snprintf(buf, PAGE_SIZE, "%i\n", card->number); } static DEVICE_ATTR(number, S_IRUGO, card_number_show_attr, NULL); -- 1.8.5.2

1 0

[alsa-devel] [PATCH] ASoC: Intel: Fix sparse warnings for firmware loader.
by Liam Girdwood 19 Feb '14

19 Feb '14

Sparse gives us the following warning on sst-firmware.c CHECK sound/soc/intel/sst-firmware.c sound/soc/intel/sst-firmware.c:39:34: warning: incorrect type in argument 1 (different address spaces) sound/soc/intel/sst-firmware.c:39:34: expected void volatile [noderef] <asn:2>*dst sound/soc/intel/sst-firmware.c:39:34: got void * sound/soc/intel/sst-firmware.c:417:36: warning: incorrect type in argument 1 (different address spaces) sound/soc/intel/sst-firmware.c:417:36: expected void *dest sound/soc/intel/sst-firmware.c:417:36: got void [noderef] <asn:2>* sound/soc/intel/sst-firmware.c:430:5: warning: symbol 'sst_block_module_remove' was not declared. Should it be static? This patch removes this warning. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 31cd154..dee7eb5 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -30,7 +30,7 @@ #include "sst-dsp.h" #include "sst-dsp-priv.h" -static void sst_memcpy32(void *dest, void *src, u32 bytes) +static void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes) { u32 i; -- 1.8.3.2

2 2

[alsa-devel] [PATCH] ALSA: hda - Add QEMU codec vendor ID
by Takashi Iwai 19 Feb '14

19 Feb '14

Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4be322b0d745..115502783b64 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -67,6 +67,7 @@ static struct hda_vendor_id hda_vendor_ids[] = { { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, { 0x1aec, "Wolfson Microelectronics" }, + { 0x1af4, "QEMU" }, { 0x434d, "C-Media" }, { 0x8086, "Intel" }, { 0x8384, "SigmaTel" }, -- 1.8.5.2

1 0

[alsa-devel] snd_intel8x0_inside_vm()
by H. Peter Anvin 19 Feb '14

19 Feb '14

I just stumbled upon the function snd_intel8x0_inside_vm(), and quite frankly the code seems bizarre: /* detect KVM and Parallels virtual environments */ result = kvm_para_available(); #ifdef X86_FEATURE_HYPERVISOR result = result || boot_cpu_has(X86_FEATURE_HYPERVISOR); #endif if (!result) goto fini; Why don't we simply rely on the PCI SSIDs? -hpa

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda/ca0132 - setup/cleanup streams
by Hsin-Yu Chao 19 Feb '14

19 Feb '14

When a HDMI stream is opened with the same stream tag as a following opened stream to ca0132, audio will be heard from two ports simultaneously. Fix this issue by change to use snd_hda_codec_setup_stream and snd_hda_codec_cleanup_stream instead, so that an inactive stream can be marked as 'dirty' when found with a conflict stream tag, and then get purified. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> Reviewed-by: Chih-Chung Chang <chihchung(a)chromium.org> --- sound/pci/hda/patch_ca0132.c | 68 ++++++-------------------------------------- 1 file changed, 9 insertions(+), 59 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 54d1479..59104dc 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2662,60 +2662,6 @@ static bool dspload_wait_loaded(struct hda_codec *codec) } /* - * PCM stuffs - */ -static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid, - u32 stream_tag, - int channel_id, int format) -{ - unsigned int oldval, newval; - - if (!nid) - return; - - snd_printdd( - "ca0132_setup_stream: NID=0x%x, stream=0x%x, " - "channel=%d, format=0x%x\n", - nid, stream_tag, channel_id, format); - - /* update the format-id if changed */ - oldval = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_STREAM_FORMAT, - 0); - if (oldval != format) { - msleep(20); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, - format); - } - - oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - newval = (stream_tag << 4) | channel_id; - if (oldval != newval) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - newval); - } -} - -static void ca0132_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) -{ - unsigned int val; - - if (!nid) - return; - - snd_printdd(KERN_INFO "ca0132_cleanup_stream: NID=0x%x\n", nid); - - val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - if (!val) - return; - - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); -} - -/* * PCM callbacks */ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, @@ -2726,7 +2672,9 @@ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, { struct ca0132_spec *spec = codec->spec; - ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format); + ca0132_toggle_dac_format(codec); + + snd_hda_codec_setup_stream(codec, spec->dacs[0], stream_tag, 0, format); return 0; } @@ -2745,7 +2693,7 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) msleep(50); - ca0132_cleanup_stream(codec, spec->dacs[0]); + snd_hda_codec_cleanup_stream(codec, spec->dacs[0]); return 0; } @@ -2824,8 +2772,8 @@ static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo, { struct ca0132_spec *spec = codec->spec; - ca0132_setup_stream(codec, spec->adcs[substream->number], - stream_tag, 0, format); + snd_hda_codec_setup_stream(codec, spec->adcs[substream->number], + stream_tag, 0, format); return 0; } @@ -2839,7 +2787,7 @@ static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, if (spec->dsp_state == DSP_DOWNLOADING) return 0; - ca0132_cleanup_stream(codec, hinfo->nid); + snd_hda_codec_cleanup_stream(codec, hinfo->nid); return 0; } @@ -4742,6 +4690,8 @@ static int patch_ca0132(struct hda_codec *codec) return err; codec->patch_ops = ca0132_patch_ops; + codec->pcm_format_first = 1; + codec->no_sticky_stream = 1; return 0; } -- 1.9.0.rc1.175.g0b1dcb5

2 7

[alsa-devel] [PATCH 1/2] ASoC: Intel: Move extended fw base and size fields in struct sst_pdata
by Jarkko Nikula 19 Feb '14

19 Feb '14

Move fw_base and fw_size fields in struct sst_pdata under ACPI data for clarifying that these are not related to firmware file but for platform specific extended firmware area reserved by the BIOS. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-dsp.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index 0ce5c8d91794..d134359fecac 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -147,12 +147,12 @@ struct sst_pdata { u32 lpe_size; u32 pcicfg_base; u32 pcicfg_size; + u32 fw_base; + u32 fw_size; int irq; /* Firmware */ const char *fw_filename; - u32 fw_base; - u32 fw_size; /* DMA */ u32 dma_base; -- 1.8.5.3

3 7

[alsa-devel] [PATCH] ASoC: Add ADAU1977 CODEC driver
by Lars-Peter Clausen 19 Feb '14

19 Feb '14

This patch adds support for the ADAU1977, ADAU1978 and ADAU1979 audio CODEC devices. They are a family of 4-channel differential input audio ADC devices. They can be connected to either a SPI or I2C bus. The driver is implemented in three modules, one main module (adau1977.ko) which implements the device logic and one module each for SPI (adau1977-spi.ko) and I2C (adau1977-i2c.ko) bus access. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Split out devicetree support into a separate patch. Will be submitted after this has been merged. * gpiod_set_value -> gpiod_set_value_cansleep * Split adau1977_set_power() into adau1977_disable_power() and adau1977_enable_power() as most of the code for those two cases was on different paths and was a bit hard to follow due to both being implemented in the same function. * Moved duplicated rate to FS field setting lookup to separate helper function. * Drop 'if (stream == SNDRV_PCM_STREAM_PLAYBACK) ...' in adau1977_mute() since the core should make sure that this never happens --- include/linux/platform_data/adau1977.h | 45 ++ sound/soc/codecs/Kconfig | 15 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/adau1977-i2c.c | 59 ++ sound/soc/codecs/adau1977-spi.c | 76 +++ sound/soc/codecs/adau1977.c | 1018 ++++++++++++++++++++++++++++++++ sound/soc/codecs/adau1977.h | 37 ++ 7 files changed, 1256 insertions(+) create mode 100644 include/linux/platform_data/adau1977.h create mode 100644 sound/soc/codecs/adau1977-i2c.c create mode 100644 sound/soc/codecs/adau1977-spi.c create mode 100644 sound/soc/codecs/adau1977.c create mode 100644 sound/soc/codecs/adau1977.h diff --git a/include/linux/platform_data/adau1977.h b/include/linux/platform_data/adau1977.h new file mode 100644 index 0000000..bed11d9 --- /dev/null +++ b/include/linux/platform_data/adau1977.h @@ -0,0 +1,45 @@ +/* + * ADAU1977/ADAU1978/ADAU1979 driver + * + * Copyright 2014 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#ifndef __LINUX_PLATFORM_DATA_ADAU1977_H__ +#define __LINUX_PLATFORM_DATA_ADAU1977_H__ + +/** + * enum adau1977_micbias - ADAU1977 MICBIAS pin voltage setting + * @ADAU1977_MICBIAS_5V0: MICBIAS is set to 5.0 V + * @ADAU1977_MICBIAS_5V5: MICBIAS is set to 5.5 V + * @ADAU1977_MICBIAS_6V0: MICBIAS is set to 6.0 V + * @ADAU1977_MICBIAS_6V5: MICBIAS is set to 6.5 V + * @ADAU1977_MICBIAS_7V0: MICBIAS is set to 7.0 V + * @ADAU1977_MICBIAS_7V5: MICBIAS is set to 7.5 V + * @ADAU1977_MICBIAS_8V0: MICBIAS is set to 8.0 V + * @ADAU1977_MICBIAS_8V5: MICBIAS is set to 8.5 V + * @ADAU1977_MICBIAS_9V0: MICBIAS is set to 9.0 V + */ +enum adau1977_micbias { + ADAU1977_MICBIAS_5V0 = 0x0, + ADAU1977_MICBIAS_5V5 = 0x1, + ADAU1977_MICBIAS_6V0 = 0x2, + ADAU1977_MICBIAS_6V5 = 0x3, + ADAU1977_MICBIAS_7V0 = 0x4, + ADAU1977_MICBIAS_7V5 = 0x5, + ADAU1977_MICBIAS_8V0 = 0x6, + ADAU1977_MICBIAS_8V5 = 0x7, + ADAU1977_MICBIAS_9V0 = 0x8, +}; + +/** + * struct adau1977_platform_data - Platform configuration data for the ADAU1977 + * @micbias: Specifies the voltage for the MICBIAS pin + */ +struct adau1977_platform_data { + enum adau1977_micbias micbias; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index fee1f64..9a5f85e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -24,6 +24,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADAU1373 if I2C select SND_SOC_ADAV801 if SPI_MASTER select SND_SOC_ADAV803 if I2C + select SND_SOC_ADAU1977_SPI if SPI_MASTER + select SND_SOC_ADAU1977_I2C if I2C select SND_SOC_ADAU1701 if I2C select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER @@ -202,6 +204,19 @@ config SND_SOC_ADAU1701 config SND_SOC_ADAU1373 tristate +config SND_SOC_ADAU1977 + tristate + +config SND_SOC_ADAU1977_SPI + tristate + select SND_SOC_ADAU1977 + select REGMAP_SPI + +config SND_SOC_ADAU1977_I2C + tristate + select SND_SOC_ADAU1977 + select REGMAP_I2C + config SND_SOC_ADAV80X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index eeb2f91..12dc02c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,9 @@ snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-adau1701-objs := adau1701.o snd-soc-adau1373-objs := adau1373.o +snd-soc-adau1977-objs := adau1977.o +snd-soc-adau1977-spi-objs := adau1977-spi.o +snd-soc-adau1977-i2c-objs := adau1977-i2c.o snd-soc-adav80x-objs := adav80x.o snd-soc-adav801-objs := adav801.o snd-soc-adav803-objs := adav803.o @@ -142,6 +145,9 @@ obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o +obj-$(CONFIG_SND_SOC_ADAU1977) += snd-soc-adau1977.o +obj-$(CONFIG_SND_SOC_ADAU1977_SPI) += snd-soc-adau1977-spi.o +obj-$(CONFIG_SND_SOC_ADAU1977_I2C) += snd-soc-adau1977-i2c.o obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o diff --git a/sound/soc/codecs/adau1977-i2c.c b/sound/soc/codecs/adau1977-i2c.c new file mode 100644 index 0000000..9700e8c --- /dev/null +++ b/sound/soc/codecs/adau1977-i2c.c @@ -0,0 +1,59 @@ +/* + * ADAU1977/ADAU1978/ADAU1979 driver + * + * Copyright 2014 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#include <linux/i2c.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/soc.h> + +#include "adau1977.h" + +static int adau1977_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct regmap_config config; + + config = adau1977_regmap_config; + config.val_bits = 8; + config.reg_bits = 8; + + return adau1977_probe(&client->dev, + devm_regmap_init_i2c(client, &config), + id->driver_data, NULL); +} + +static int adau1977_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id adau1977_i2c_ids[] = { + { "adau1977", ADAU1977 }, + { "adau1978", ADAU1978 }, + { "adau1979", ADAU1978 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, adau1977_i2c_ids); + +static struct i2c_driver adau1977_i2c_driver = { + .driver = { + .name = "adau1977", + .owner = THIS_MODULE, + }, + .probe = adau1977_i2c_probe, + .remove = adau1977_i2c_remove, + .id_table = adau1977_i2c_ids, +}; +module_i2c_driver(adau1977_i2c_driver); + +MODULE_DESCRIPTION("ASoC ADAU1977/ADAU1978/ADAU1979 driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1977-spi.c b/sound/soc/codecs/adau1977-spi.c new file mode 100644 index 0000000..b05cf5d --- /dev/null +++ b/sound/soc/codecs/adau1977-spi.c @@ -0,0 +1,76 @@ +/* + * ADAU1977/ADAU1978/ADAU1979 driver + * + * Copyright 2014 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/spi/spi.h> +#include <sound/soc.h> + +#include "adau1977.h" + +static void adau1977_spi_switch_mode(struct device *dev) +{ + struct spi_device *spi = to_spi_device(dev); + + /* + * To get the device into SPI mode CLATCH has to be pulled low three + * times. Do this by issuing three dummy reads. + */ + spi_w8r8(spi, 0x00); + spi_w8r8(spi, 0x00); + spi_w8r8(spi, 0x00); +} + +static int adau1977_spi_probe(struct spi_device *spi) +{ + const struct spi_device_id *id = spi_get_device_id(spi); + struct regmap_config config; + + if (!id) + return -EINVAL; + + config = adau1977_regmap_config; + config.val_bits = 8; + config.reg_bits = 16; + config.read_flag_mask = 0x1; + + return adau1977_probe(&spi->dev, + devm_regmap_init_spi(spi, &config), + id->driver_data, adau1977_spi_switch_mode); +} + +static int adau1977_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static const struct spi_device_id adau1977_spi_ids[] = { + { "adau1977", ADAU1977 }, + { "adau1978", ADAU1978 }, + { "adau1979", ADAU1978 }, + { } +}; +MODULE_DEVICE_TABLE(spi, adau1977_spi_ids); + +static struct spi_driver adau1977_spi_driver = { + .driver = { + .name = "adau1977", + .owner = THIS_MODULE, + }, + .probe = adau1977_spi_probe, + .remove = adau1977_spi_remove, + .id_table = adau1977_spi_ids, +}; +module_spi_driver(adau1977_spi_driver); + +MODULE_DESCRIPTION("ASoC ADAU1977/ADAU1978/ADAU1979 driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c new file mode 100644 index 0000000..fd55da7 --- /dev/null +++ b/sound/soc/codecs/adau1977.c @@ -0,0 +1,1018 @@ +/* + * ADAU1977/ADAU1978/ADAU1979 driver + * + * Copyright 2014 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/gpio/consumer.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_data/adau1977.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <linux/slab.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "adau1977.h" + +#define ADAU1977_REG_POWER 0x00 +#define ADAU1977_REG_PLL 0x01 +#define ADAU1977_REG_BOOST 0x02 +#define ADAU1977_REG_MICBIAS 0x03 +#define ADAU1977_REG_BLOCK_POWER_SAI 0x04 +#define ADAU1977_REG_SAI_CTRL0 0x05 +#define ADAU1977_REG_SAI_CTRL1 0x06 +#define ADAU1977_REG_CMAP12 0x07 +#define ADAU1977_REG_CMAP34 0x08 +#define ADAU1977_REG_SAI_OVERTEMP 0x09 +#define ADAU1977_REG_POST_ADC_GAIN(x) (0x0a + (x)) +#define ADAU1977_REG_MISC_CONTROL 0x0e +#define ADAU1977_REG_DIAG_CONTROL 0x10 +#define ADAU1977_REG_STATUS(x) (0x11 + (x)) +#define ADAU1977_REG_DIAG_IRQ1 0x15 +#define ADAU1977_REG_DIAG_IRQ2 0x16 +#define ADAU1977_REG_ADJUST1 0x17 +#define ADAU1977_REG_ADJUST2 0x18 +#define ADAU1977_REG_ADC_CLIP 0x19 +#define ADAU1977_REG_DC_HPF_CAL 0x1a + +#define ADAU1977_POWER_RESET BIT(7) +#define ADAU1977_POWER_PWUP BIT(0) + +#define ADAU1977_PLL_CLK_S BIT(4) +#define ADAU1977_PLL_MCS_MASK 0x7 + +#define ADAU1977_MICBIAS_MB_VOLTS_MASK 0xf0 +#define ADAU1977_MICBIAS_MB_VOLTS_OFFSET 4 + +#define ADAU1977_BLOCK_POWER_SAI_LR_POL BIT(7) +#define ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE BIT(6) +#define ADAU1977_BLOCK_POWER_SAI_LDO_EN BIT(5) + +#define ADAU1977_SAI_CTRL0_FMT_MASK (0x3 << 6) +#define ADAU1977_SAI_CTRL0_FMT_I2S (0x0 << 6) +#define ADAU1977_SAI_CTRL0_FMT_LJ (0x1 << 6) +#define ADAU1977_SAI_CTRL0_FMT_RJ_24BIT (0x2 << 6) +#define ADAU1977_SAI_CTRL0_FMT_RJ_16BIT (0x3 << 6) + +#define ADAU1977_SAI_CTRL0_SAI_MASK (0x7 << 3) +#define ADAU1977_SAI_CTRL0_SAI_I2S (0x0 << 3) +#define ADAU1977_SAI_CTRL0_SAI_TDM_2 (0x1 << 3) +#define ADAU1977_SAI_CTRL0_SAI_TDM_4 (0x2 << 3) +#define ADAU1977_SAI_CTRL0_SAI_TDM_8 (0x3 << 3) +#define ADAU1977_SAI_CTRL0_SAI_TDM_16 (0x4 << 3) + +#define ADAU1977_SAI_CTRL0_FS_MASK (0x7) +#define ADAU1977_SAI_CTRL0_FS_8000_12000 (0x0) +#define ADAU1977_SAI_CTRL0_FS_16000_24000 (0x1) +#define ADAU1977_SAI_CTRL0_FS_32000_48000 (0x2) +#define ADAU1977_SAI_CTRL0_FS_64000_96000 (0x3) +#define ADAU1977_SAI_CTRL0_FS_128000_192000 (0x4) + +#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_MASK (0x3 << 5) +#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_32 (0x0 << 5) +#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_24 (0x1 << 5) +#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_16 (0x2 << 5) +#define ADAU1977_SAI_CTRL1_DATA_WIDTH_MASK (0x1 << 4) +#define ADAU1977_SAI_CTRL1_DATA_WIDTH_16BIT (0x1 << 4) +#define ADAU1977_SAI_CTRL1_DATA_WIDTH_24BIT (0x0 << 4) +#define ADAU1977_SAI_CTRL1_LRCLK_PULSE BIT(3) +#define ADAU1977_SAI_CTRL1_MSB BIT(2) +#define ADAU1977_SAI_CTRL1_BCLKRATE_16 (0x1 << 1) +#define ADAU1977_SAI_CTRL1_BCLKRATE_32 (0x0 << 1) +#define ADAU1977_SAI_CTRL1_BCLKRATE_MASK (0x1 << 1) +#define ADAU1977_SAI_CTRL1_MASTER BIT(0) + +#define ADAU1977_SAI_OVERTEMP_DRV_C(x) BIT(4 + (x)) +#define ADAU1977_SAI_OVERTEMP_DRV_HIZ BIT(3) + +#define ADAU1977_MISC_CONTROL_SUM_MODE_MASK (0x3 << 6) +#define ADAU1977_MISC_CONTROL_SUM_MODE_1CH (0x2 << 6) +#define ADAU1977_MISC_CONTROL_SUM_MODE_2CH (0x1 << 6) +#define ADAU1977_MISC_CONTROL_SUM_MODE_4CH (0x0 << 6) +#define ADAU1977_MISC_CONTROL_MMUTE BIT(4) +#define ADAU1977_MISC_CONTROL_DC_CAL BIT(0) + +#define ADAU1977_CHAN_MAP_SECOND_SLOT_OFFSET 4 +#define ADAU1977_CHAN_MAP_FIRST_SLOT_OFFSET 0 + +struct adau1977 { + struct regmap *regmap; + bool right_j; + unsigned int sysclk; + enum adau1977_sysclk_src sysclk_src; + struct gpio_desc *reset_gpio; + enum adau1977_type type; + + struct regulator *avdd_reg; + struct regulator *dvdd_reg; + + struct snd_pcm_hw_constraint_list constraints; + + struct device *dev; + void (*switch_mode)(struct device *dev); + + unsigned int max_master_fs; + unsigned int slot_width; + bool enabled; + bool master; +}; + +static const struct reg_default adau1977_reg_defaults[] = { + { 0x00, 0x00 }, + { 0x01, 0x41 }, + { 0x02, 0x4a }, + { 0x03, 0x7d }, + { 0x04, 0x3d }, + { 0x05, 0x02 }, + { 0x06, 0x00 }, + { 0x07, 0x10 }, + { 0x08, 0x32 }, + { 0x09, 0xf0 }, + { 0x0a, 0xa0 }, + { 0x0b, 0xa0 }, + { 0x0c, 0xa0 }, + { 0x0d, 0xa0 }, + { 0x0e, 0x02 }, + { 0x10, 0x0f }, + { 0x15, 0x20 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, + { 0x18, 0x00 }, + { 0x1a, 0x00 }, +}; + +static const DECLARE_TLV_DB_MINMAX_MUTE(adau1977_adc_gain, -3562, 6000); + +static const struct snd_soc_dapm_widget adau1977_micbias_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("MICBIAS", ADAU1977_REG_MICBIAS, + 3, 0, NULL, 0) +}; + +static const struct snd_soc_dapm_widget adau1977_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("Vref", ADAU1977_REG_BLOCK_POWER_SAI, + 4, 0, NULL, 0), + + SND_SOC_DAPM_ADC("ADC1", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 0, 0), + SND_SOC_DAPM_ADC("ADC2", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 1, 0), + SND_SOC_DAPM_ADC("ADC3", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 2, 0), + SND_SOC_DAPM_ADC("ADC4", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 3, 0), + + SND_SOC_DAPM_INPUT("AIN1"), + SND_SOC_DAPM_INPUT("AIN2"), + SND_SOC_DAPM_INPUT("AIN3"), + SND_SOC_DAPM_INPUT("AIN4"), + + SND_SOC_DAPM_OUTPUT("VREF"), +}; + +static const struct snd_soc_dapm_route adau1977_dapm_routes[] = { + { "ADC1", NULL, "AIN1" }, + { "ADC2", NULL, "AIN2" }, + { "ADC3", NULL, "AIN3" }, + { "ADC4", NULL, "AIN4" }, + + { "ADC1", NULL, "Vref" }, + { "ADC2", NULL, "Vref" }, + { "ADC3", NULL, "Vref" }, + { "ADC4", NULL, "Vref" }, + + { "VREF", NULL, "Vref" }, +}; + +#define ADAU1977_VOLUME(x) \ + SOC_SINGLE_TLV("ADC" #x " Capture Volume", \ + ADAU1977_REG_POST_ADC_GAIN((x) - 1), \ + 0, 255, 1, adau1977_adc_gain) + +#define ADAU1977_HPF_SWITCH(x) \ + SOC_SINGLE("ADC" #x " Highpass-Filter Capture Switch", \ + ADAU1977_REG_DC_HPF_CAL, (x) - 1, 1, 0) + +#define ADAU1977_DC_SUB_SWITCH(x) \ + SOC_SINGLE("ADC" #x " DC Substraction Capture Switch", \ + ADAU1977_REG_DC_HPF_CAL, (x) + 3, 1, 0) + +static const struct snd_kcontrol_new adau1977_snd_controls[] = { + ADAU1977_VOLUME(1), + ADAU1977_VOLUME(2), + ADAU1977_VOLUME(3), + ADAU1977_VOLUME(4), + + ADAU1977_HPF_SWITCH(1), + ADAU1977_HPF_SWITCH(2), + ADAU1977_HPF_SWITCH(3), + ADAU1977_HPF_SWITCH(4), + + ADAU1977_DC_SUB_SWITCH(1), + ADAU1977_DC_SUB_SWITCH(2), + ADAU1977_DC_SUB_SWITCH(3), + ADAU1977_DC_SUB_SWITCH(4), +}; + +static int adau1977_reset(struct adau1977 *adau1977) +{ + int ret; + + /* + * The reset bit is obviously volatile, but we need to be able to cache + * the other bits in the register, so we can't just mark the whole + * register as volatile. Since this is the only place where we'll ever + * touch the reset bit just bypass the cache for this operation. + */ + regcache_cache_bypass(adau1977->regmap, true); + ret = regmap_write(adau1977->regmap, ADAU1977_REG_POWER, + ADAU1977_POWER_RESET); + regcache_cache_bypass(adau1977->regmap, false); + if (ret) + return ret; + + return ret; +} + +/* + * Returns the appropriate setting for ths FS field in the CTRL0 register + * depending on the rate. + */ +static int adau1977_lookup_fs(unsigned int rate) +{ + if (rate >= 8000 && rate <= 12000) + return ADAU1977_SAI_CTRL0_FS_8000_12000; + else if (rate >= 16000 && rate <= 24000) + return ADAU1977_SAI_CTRL0_FS_16000_24000; + else if (rate >= 32000 && rate <= 48000) + return ADAU1977_SAI_CTRL0_FS_32000_48000; + else if (rate >= 64000 && rate <= 96000) + return ADAU1977_SAI_CTRL0_FS_64000_96000; + else if (rate >= 128000 && rate <= 192000) + return ADAU1977_SAI_CTRL0_FS_128000_192000; + else + return -EINVAL; +} + +static int adau1977_lookup_mcs(struct adau1977 *adau1977, unsigned int rate, + unsigned int fs) +{ + unsigned int mcs; + + /* + * rate = sysclk / (512 * mcs_lut[mcs]) * 2**fs + * => mcs_lut[mcs] = sysclk / (512 * rate) * 2**fs + * => mcs_lut[mcs] = sysclk / ((512 / 2**fs) * rate) + */ + + rate *= 512 >> fs; + + if (adau1977->sysclk % rate != 0) + return -EINVAL; + + mcs = adau1977->sysclk / rate; + + /* The factors configured by MCS are 1, 2, 3, 4, 6 */ + if (mcs < 1 || mcs > 6 || mcs == 5) + return -EINVAL; + + mcs = mcs - 1; + if (mcs == 5) + mcs = 4; + + return mcs; +} + +static int adau1977_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + unsigned int slot_width; + unsigned int ctrl0, ctrl0_mask; + unsigned int ctrl1; + int mcs, fs; + int ret; + + fs = adau1977_lookup_fs(rate); + if (fs < 0) + return fs; + + if (adau1977->sysclk_src == ADAU1977_SYSCLK_SRC_MCLK) { + mcs = adau1977_lookup_mcs(adau1977, rate, fs); + if (mcs < 0) + return mcs; + } else { + mcs = 0; + } + + ctrl0_mask = ADAU1977_SAI_CTRL0_FS_MASK; + ctrl0 = fs; + + if (adau1977->right_j) { + switch (params_width(params)) { + case 16: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_RJ_16BIT; + break; + case 24: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_RJ_24BIT; + break; + default: + return -EINVAL; + } + ctrl0_mask |= ADAU1977_SAI_CTRL0_FMT_MASK; + } + + if (adau1977->master) { + switch (params_width(params)) { + case 16: + ctrl1 = ADAU1977_SAI_CTRL1_DATA_WIDTH_16BIT; + slot_width = 16; + break; + case 24: + case 32: + ctrl1 = ADAU1977_SAI_CTRL1_DATA_WIDTH_24BIT; + slot_width = 32; + break; + default: + return -EINVAL; + } + + /* In TDM mode there is a fixed slot width */ + if (adau1977->slot_width) + slot_width = adau1977->slot_width; + + if (slot_width == 16) + ctrl1 |= ADAU1977_SAI_CTRL1_BCLKRATE_16; + else + ctrl1 |= ADAU1977_SAI_CTRL1_BCLKRATE_32; + + ret = regmap_update_bits(adau1977->regmap, + ADAU1977_REG_SAI_CTRL1, + ADAU1977_SAI_CTRL1_DATA_WIDTH_MASK | + ADAU1977_SAI_CTRL1_BCLKRATE_MASK, + ctrl1); + if (ret < 0) + return ret; + } + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL0, + ctrl0_mask, ctrl0); + if (ret < 0) + return ret; + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_PLL, + ADAU1977_PLL_MCS_MASK, mcs); +} + +static int adau1977_power_disable(struct adau1977 *adau1977) +{ + int ret = 0; + + if (!adau1977->enabled) + return 0; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_POWER, + ADAU1977_POWER_PWUP, 0); + if (ret) + return ret; + + regcache_mark_dirty(adau1977->regmap); + + if (adau1977->reset_gpio) + gpiod_set_value_cansleep(adau1977->reset_gpio, 0); + + regcache_cache_only(adau1977->regmap, true); + + regulator_disable(adau1977->avdd_reg); + if (adau1977->dvdd_reg) + regulator_disable(adau1977->dvdd_reg); + + adau1977->enabled = false; + + return 0; +} + +static int adau1977_power_enable(struct adau1977 *adau1977) +{ + unsigned int val; + int ret = 0; + + if (adau1977->enabled) + return 0; + + ret = regulator_enable(adau1977->avdd_reg); + if (ret) + return ret; + + if (adau1977->dvdd_reg) { + ret = regulator_enable(adau1977->dvdd_reg); + if (ret) + goto err_disable_avdd; + } + + if (adau1977->reset_gpio) + gpiod_set_value_cansleep(adau1977->reset_gpio, 1); + + regcache_cache_only(adau1977->regmap, false); + + if (adau1977->switch_mode) + adau1977->switch_mode(adau1977->dev); + + ret = adau1977_reset(adau1977); + if (ret) + goto err_disable_dvdd; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_POWER, + ADAU1977_POWER_PWUP, ADAU1977_POWER_PWUP); + if (ret) + goto err_disable_dvdd; + + ret = regcache_sync(adau1977->regmap); + if (ret) + goto err_disable_dvdd; + + /* + * The PLL register is not affected by the software reset. It is + * possible that the value of the register was changed to the + * default value while we were in cache only mode. In this case + * regcache_sync will skip over it and we have to manually sync + * it. + */ + ret = regmap_read(adau1977->regmap, ADAU1977_REG_PLL, &val); + if (ret) + goto err_disable_dvdd; + + if (val == 0x41) { + regcache_cache_bypass(adau1977->regmap, true); + ret = regmap_write(adau1977->regmap, ADAU1977_REG_PLL, + 0x41); + if (ret) + goto err_disable_dvdd; + regcache_cache_bypass(adau1977->regmap, false); + } + + adau1977->enabled = true; + + return ret; + +err_disable_dvdd: + if (adau1977->dvdd_reg) + regulator_disable(adau1977->dvdd_reg); +err_disable_avdd: + regulator_disable(adau1977->avdd_reg); + return ret; +} + +static int adau1977_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + ret = adau1977_power_enable(adau1977); + break; + case SND_SOC_BIAS_OFF: + ret = adau1977_power_disable(adau1977); + break; + } + + if (ret) + return ret; + + codec->dapm.bias_level = level; + + return 0; +} + +static int adau1977_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int width) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl0, ctrl1, drv; + unsigned int slot[4]; + unsigned int i; + int ret; + + if (slots == 0) { + /* 0 = No fixed slot width */ + adau1977->slot_width = 0; + adau1977->max_master_fs = 192000; + return regmap_update_bits(adau1977->regmap, + ADAU1977_REG_SAI_CTRL0, ADAU1977_SAI_CTRL0_SAI_MASK, + ADAU1977_SAI_CTRL0_SAI_I2S); + } + + if (rx_mask == 0 || tx_mask != 0) + return -EINVAL; + + drv = 0; + for (i = 0; i < 4; i++) { + slot[i] = __ffs(rx_mask); + drv |= ADAU1977_SAI_OVERTEMP_DRV_C(i); + rx_mask &= ~(1 << slot[i]); + if (slot[i] >= slots) + return -EINVAL; + if (rx_mask == 0) + break; + } + + if (rx_mask != 0) + return -EINVAL; + + switch (width) { + case 16: + ctrl1 = ADAU1977_SAI_CTRL1_SLOT_WIDTH_16; + break; + case 24: + /* We can only generate 16 bit or 32 bit wide slots */ + if (adau1977->master) + return -EINVAL; + ctrl1 = ADAU1977_SAI_CTRL1_SLOT_WIDTH_24; + break; + case 32: + ctrl1 = ADAU1977_SAI_CTRL1_SLOT_WIDTH_32; + break; + default: + return -EINVAL; + } + + switch (slots) { + case 2: + ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_2; + break; + case 4: + ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_4; + break; + case 8: + ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_8; + break; + case 16: + ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_16; + break; + default: + return -EINVAL; + } + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_OVERTEMP, + ADAU1977_SAI_OVERTEMP_DRV_C(0) | + ADAU1977_SAI_OVERTEMP_DRV_C(1) | + ADAU1977_SAI_OVERTEMP_DRV_C(2) | + ADAU1977_SAI_OVERTEMP_DRV_C(3), drv); + if (ret) + return ret; + + ret = regmap_write(adau1977->regmap, ADAU1977_REG_CMAP12, + (slot[1] << ADAU1977_CHAN_MAP_SECOND_SLOT_OFFSET) | + (slot[0] << ADAU1977_CHAN_MAP_FIRST_SLOT_OFFSET)); + if (ret) + return ret; + + ret = regmap_write(adau1977->regmap, ADAU1977_REG_CMAP34, + (slot[3] << ADAU1977_CHAN_MAP_SECOND_SLOT_OFFSET) | + (slot[2] << ADAU1977_CHAN_MAP_FIRST_SLOT_OFFSET)); + if (ret) + return ret; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL0, + ADAU1977_SAI_CTRL0_SAI_MASK, ctrl0); + if (ret) + return ret; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL1, + ADAU1977_SAI_CTRL1_SLOT_WIDTH_MASK, ctrl1); + if (ret) + return ret; + + adau1977->slot_width = width; + + /* In master mode the maximum bitclock is 24.576 MHz */ + adau1977->max_master_fs = min(192000, 24576000 / width / slots); + + return 0; +} + +static int adau1977_mute(struct snd_soc_dai *dai, int mute, int stream) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int val; + + if (mute) + val = ADAU1977_MISC_CONTROL_MMUTE; + else + val = 0; + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_MISC_CONTROL, + ADAU1977_MISC_CONTROL_MMUTE, val); +} + +static int adau1977_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl0 = 0, ctrl1 = 0, block_power = 0; + bool invert_lrclk; + int ret; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + adau1977->master = false; + break; + case SND_SOC_DAIFMT_CBM_CFM: + ctrl1 |= ADAU1977_SAI_CTRL1_MASTER; + adau1977->master = true; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + invert_lrclk = false; + break; + case SND_SOC_DAIFMT_IB_NF: + block_power |= ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE; + invert_lrclk = false; + break; + case SND_SOC_DAIFMT_NB_IF: + invert_lrclk = true; + break; + case SND_SOC_DAIFMT_IB_IF: + block_power |= ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE; + invert_lrclk = true; + break; + default: + return -EINVAL; + } + + adau1977->right_j = false; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_LJ; + invert_lrclk = !invert_lrclk; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_RJ_24BIT; + adau1977->right_j = true; + invert_lrclk = !invert_lrclk; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1 |= ADAU1977_SAI_CTRL1_LRCLK_PULSE; + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_I2S; + invert_lrclk = false; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1 |= ADAU1977_SAI_CTRL1_LRCLK_PULSE; + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_LJ; + invert_lrclk = false; + break; + default: + return -EINVAL; + } + + if (invert_lrclk) + block_power |= ADAU1977_BLOCK_POWER_SAI_LR_POL; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_BLOCK_POWER_SAI, + ADAU1977_BLOCK_POWER_SAI_LR_POL | + ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE, block_power); + if (ret) + return ret; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL0, + ADAU1977_SAI_CTRL0_FMT_MASK, + ctrl0); + if (ret) + return ret; + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL1, + ADAU1977_SAI_CTRL1_MASTER | ADAU1977_SAI_CTRL1_LRCLK_PULSE, + ctrl1); +} + +static int adau1977_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + u64 formats = 0; + + if (adau1977->slot_width == 16) + formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE; + else if (adau1977->right_j || adau1977->slot_width == 24) + formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE; + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &adau1977->constraints); + + if (adau1977->master) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, 8000, adau1977->max_master_fs); + + if (formats != 0) + snd_pcm_hw_constraint_mask64(substream->runtime, + SNDRV_PCM_HW_PARAM_FORMAT, formats); + + return 0; +} + +static int adau1977_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int val; + + if (tristate) + val = ADAU1977_SAI_OVERTEMP_DRV_HIZ; + else + val = 0; + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_OVERTEMP, + ADAU1977_SAI_OVERTEMP_DRV_HIZ, val); +} + +static const struct snd_soc_dai_ops adau1977_dai_ops = { + .startup = adau1977_startup, + .hw_params = adau1977_hw_params, + .mute_stream = adau1977_mute, + .set_fmt = adau1977_set_dai_fmt, + .set_tdm_slot = adau1977_set_tdm_slot, + .set_tristate = adau1977_set_tristate, +}; + +static struct snd_soc_dai_driver adau1977_dai = { + .name = "adau1977-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .sig_bits = 24, + }, + .ops = &adau1977_dai_ops, +}; + +static const unsigned int adau1977_rates[] = { + 8000, 16000, 32000, 64000, 128000, + 11025, 22050, 44100, 88200, 172400, + 12000, 24000, 48000, 96000, 192000, +}; + +#define ADAU1977_RATE_CONSTRAINT_MASK_32000 0x001f +#define ADAU1977_RATE_CONSTRAINT_MASK_44100 0x03e0 +#define ADAU1977_RATE_CONSTRAINT_MASK_48000 0x7c00 +/* All rates >= 32000 */ +#define ADAU1977_RATE_CONSTRAINT_MASK_LRCLK 0x739c + +static bool adau1977_check_sysclk(unsigned int mclk, unsigned int base_freq) +{ + unsigned int mcs; + + if (mclk % (base_freq * 128) != 0) + return false; + + mcs = mclk / (128 * base_freq); + if (mcs < 1 || mcs > 6 || mcs == 5) + return false; + + return true; +} + +static int adau1977_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec); + unsigned int mask = 0; + unsigned int clk_src; + unsigned int ret; + + if (dir != SND_SOC_CLOCK_IN) + return -EINVAL; + + if (clk_id != ADAU1977_SYSCLK) + return -EINVAL; + + switch (source) { + case ADAU1977_SYSCLK_SRC_MCLK: + clk_src = 0; + break; + case ADAU1977_SYSCLK_SRC_LRCLK: + clk_src = ADAU1977_PLL_CLK_S; + break; + default: + return -EINVAL; + } + + if (freq != 0 && source == ADAU1977_SYSCLK_SRC_MCLK) { + if (freq < 4000000 || freq > 36864000) + return -EINVAL; + + if (adau1977_check_sysclk(freq, 32000)) + mask |= ADAU1977_RATE_CONSTRAINT_MASK_32000; + if (adau1977_check_sysclk(freq, 44100)) + mask |= ADAU1977_RATE_CONSTRAINT_MASK_44100; + if (adau1977_check_sysclk(freq, 48000)) + mask |= ADAU1977_RATE_CONSTRAINT_MASK_48000; + + if (mask == 0) + return -EINVAL; + } else if (source == ADAU1977_SYSCLK_SRC_LRCLK) { + mask = ADAU1977_RATE_CONSTRAINT_MASK_LRCLK; + } + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_PLL, + ADAU1977_PLL_CLK_S, clk_src); + if (ret) + return ret; + + adau1977->constraints.mask = mask; + adau1977->sysclk_src = source; + adau1977->sysclk = freq; + + return 0; +} + +static int adau1977_codec_probe(struct snd_soc_codec *codec) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (adau1977->type) { + case ADAU1977: + ret = snd_soc_dapm_new_controls(&codec->dapm, + adau1977_micbias_dapm_widgets, + ARRAY_SIZE(adau1977_micbias_dapm_widgets)); + if (ret < 0) + return ret; + break; + default: + break; + } + + return 0; +} + +static struct snd_soc_codec_driver adau1977_codec_driver = { + .probe = adau1977_codec_probe, + .set_bias_level = adau1977_set_bias_level, + .set_sysclk = adau1977_set_sysclk, + .idle_bias_off = true, + + .controls = adau1977_snd_controls, + .num_controls = ARRAY_SIZE(adau1977_snd_controls), + .dapm_widgets = adau1977_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau1977_dapm_widgets), + .dapm_routes = adau1977_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(adau1977_dapm_routes), +}; + +static int adau1977_setup_micbias(struct adau1977 *adau1977) +{ + struct adau1977_platform_data *pdata = adau1977->dev->platform_data; + unsigned int micbias; + + if (pdata) { + micbias = pdata->micbias; + if (micbias > ADAU1977_MICBIAS_9V0) + return -EINVAL; + + } else { + micbias = ADAU1977_MICBIAS_8V5; + } + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_MICBIAS, + ADAU1977_MICBIAS_MB_VOLTS_MASK, + micbias << ADAU1977_MICBIAS_MB_VOLTS_OFFSET); +} + +int adau1977_probe(struct device *dev, struct regmap *regmap, + enum adau1977_type type, void (*switch_mode)(struct device *dev)) +{ + unsigned int power_off_mask; + struct adau1977 *adau1977; + int ret; + + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + adau1977 = devm_kzalloc(dev, sizeof(*adau1977), GFP_KERNEL); + if (adau1977 == NULL) + return -ENOMEM; + + adau1977->dev = dev; + adau1977->type = type; + adau1977->regmap = regmap; + adau1977->switch_mode = switch_mode; + adau1977->max_master_fs = 192000; + + adau1977->constraints.list = adau1977_rates; + adau1977->constraints.count = ARRAY_SIZE(adau1977_rates); + + adau1977->avdd_reg = devm_regulator_get(dev, "AVDD"); + if (IS_ERR(adau1977->avdd_reg)) + return PTR_ERR(adau1977->avdd_reg); + + adau1977->dvdd_reg = devm_regulator_get_optional(dev, "DVDD"); + if (IS_ERR(adau1977->dvdd_reg)) { + if (PTR_ERR(adau1977->dvdd_reg) != -ENODEV) + return PTR_ERR(adau1977->dvdd_reg); + adau1977->dvdd_reg = NULL; + } + + adau1977->reset_gpio = devm_gpiod_get(dev, "reset"); + if (IS_ERR(adau1977->reset_gpio)) { + ret = PTR_ERR(adau1977->reset_gpio); + if (ret != -ENOENT && ret != -ENOSYS) + return PTR_ERR(adau1977->reset_gpio); + adau1977->reset_gpio = NULL; + } + + dev_set_drvdata(dev, adau1977); + + if (adau1977->reset_gpio) { + ret = gpiod_direction_output(adau1977->reset_gpio, 0); + if (ret) + return ret; + ndelay(100); + } + + ret = adau1977_power_enable(adau1977); + if (ret) + return ret; + + if (type == ADAU1977) { + ret = adau1977_setup_micbias(adau1977); + if (ret) + goto err_poweroff; + } + + if (adau1977->dvdd_reg) + power_off_mask = ~0; + else + power_off_mask = ~ADAU1977_BLOCK_POWER_SAI_LDO_EN; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_BLOCK_POWER_SAI, + power_off_mask, 0x00); + if (ret) + goto err_poweroff; + + ret = adau1977_power_disable(adau1977); + if (ret) + return ret; + + return snd_soc_register_codec(dev, &adau1977_codec_driver, + &adau1977_dai, 1); + +err_poweroff: + adau1977_power_disable(adau1977); + return ret; + +} +EXPORT_SYMBOL_GPL(adau1977_probe); + +static bool adau1977_register_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADAU1977_REG_STATUS(0): + case ADAU1977_REG_STATUS(1): + case ADAU1977_REG_STATUS(2): + case ADAU1977_REG_STATUS(3): + case ADAU1977_REG_ADC_CLIP: + return true; + } + + return false; +} + +const struct regmap_config adau1977_regmap_config = { + .max_register = ADAU1977_REG_DC_HPF_CAL, + .volatile_reg = adau1977_register_volatile, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = adau1977_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adau1977_reg_defaults), +}; +EXPORT_SYMBOL_GPL(adau1977_regmap_config); + +MODULE_DESCRIPTION("ASoC ADAU1977/ADAU1978/ADAU1979 driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1977.h b/sound/soc/codecs/adau1977.h new file mode 100644 index 0000000..95e7143 --- /dev/null +++ b/sound/soc/codecs/adau1977.h @@ -0,0 +1,37 @@ +/* + * ADAU1977/ADAU1978/ADAU1979 driver + * + * Copyright 2014 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#ifndef __SOUND_SOC_CODECS_ADAU1977_H__ +#define __SOUND_SOC_CODECS_ADAU1977_H__ + +#include <linux/regmap.h> + +struct device; + +enum adau1977_type { + ADAU1977, + ADAU1978, + ADAU1979, +}; + +int adau1977_probe(struct device *dev, struct regmap *regmap, + enum adau1977_type type, void (*switch_mode)(struct device *dev)); + +extern const struct regmap_config adau1977_regmap_config; + +enum adau1977_clk_id { + ADAU1977_SYSCLK, +}; + +enum adau1977_sysclk_src { + ADAU1977_SYSCLK_SRC_MCLK, + ADAU1977_SYSCLK_SRC_LRCLK, +}; + +#endif -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: dapm: Correct regulator bypass error messages
by Charles Keepax 19 Feb '14

19 Feb '14

The error messages for bypassing/unbypassing a regulator appear to be swapped round, this patch corrects these. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2c2d751..7e10be8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1219,7 +1219,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, ret = regulator_allow_bypass(w->regulator, false); if (ret != 0) dev_warn(w->dapm->dev, - "ASoC: Failed to bypass %s: %d\n", + "ASoC: Failed to unbypass %s: %d\n", w->name, ret); } @@ -1229,7 +1229,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) dev_warn(w->dapm->dev, - "ASoC: Failed to unbypass %s: %d\n", + "ASoC: Failed to bypass %s: %d\n", w->name, ret); } @@ -3249,7 +3249,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) dev_warn(w->dapm->dev, - "ASoC: Failed to unbypass %s: %d\n", + "ASoC: Failed to bypass %s: %d\n", w->name, ret); } break; -- 1.7.2.5

2 1

[alsa-devel] [PATCHv2] ASoC: Intel: Rename SST trace event header to be less generic.
by Liam Girdwood 19 Feb '14

19 Feb '14

The Intel audio DSP SST trace event header has been renamed from sst.h to intel-sst.h in order to avoid any confusion with any future Samoa Standard Time drivers ;) Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/trace/events/{sst.h => intel-sst.h} | 6 +++--- sound/soc/intel/sst-dsp.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) rename include/trace/events/{sst.h => intel-sst.h} (94%) diff --git a/include/trace/events/sst.h b/include/trace/events/intel-sst.h similarity index 94% rename from include/trace/events/sst.h rename to include/trace/events/intel-sst.h index 1a0d6b3..76c72d3 100644 --- a/include/trace/events/sst.h +++ b/include/trace/events/intel-sst.h @@ -1,8 +1,8 @@ #undef TRACE_SYSTEM -#define TRACE_SYSTEM sst +#define TRACE_SYSTEM intel-sst -#if !defined(_TRACE_SST_H) || defined(TRACE_HEADER_MULTI_READ) -#define _TRACE_SST_H +#if !defined(_TRACE_INTEL_SST_H) || defined(TRACE_HEADER_MULTI_READ) +#define _TRACE_INTEL_SST_H #include <linux/types.h> #include <linux/ktime.h> diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c index 1888de5..e0ad2e5 100644 --- a/sound/soc/intel/sst-dsp.c +++ b/sound/soc/intel/sst-dsp.c @@ -24,7 +24,7 @@ #include "sst-dsp-priv.h" #define CREATE_TRACE_POINTS -#include <trace/events/sst.h> +#include <trace/events/intel-sst.h> /* Public API */ void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value) -- 1.8.3.2

2 1

[alsa-devel] [PATCH 1/2] ASoC: Intel: Rename SST trace event header to be less generic.
by Liam Girdwood 19 Feb '14

19 Feb '14

The Intel audio DSP SST trace event header has been renamed from sst.h to intel-sst.h in order to avoid any confusion with any future Samoa Standard Time drivers ;) Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/trace/events/{sst.h => intel-sst.h} | 0 sound/soc/intel/sst-dsp.c | 2 +- 2 files changed, 1 insertion(+), 1 deletion(-) rename include/trace/events/{sst.h => intel-sst.h} (100%) diff --git a/include/trace/events/sst.h b/include/trace/events/intel-sst.h similarity index 100% rename from include/trace/events/sst.h rename to include/trace/events/intel-sst.h diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c index 1888de5..e0ad2e5 100644 --- a/sound/soc/intel/sst-dsp.c +++ b/sound/soc/intel/sst-dsp.c @@ -24,7 +24,7 @@ #include "sst-dsp-priv.h" #define CREATE_TRACE_POINTS -#include <trace/events/sst.h> +#include <trace/events/intel-sst.h> /* Public API */ void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value) -- 1.8.3.2

2 3

[alsa-devel] [PATCH 1/1] ASoC: cs42l51: Remove unused variable
by Sachin Kamat 19 Feb '14

19 Feb '14

‘cs42l51’ is not used. Remove it. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/cs42l51.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index e53c8714591f..3eab46020a30 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -487,7 +487,6 @@ static struct snd_soc_dai_driver cs42l51_dai = { static int cs42l51_probe(struct snd_soc_codec *codec) { - struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); int ret, reg; ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); -- 1.7.9.5

3 2

[alsa-devel] Failed RPM build RHEL 6.4 /boot/ vmlinuz-2.6.32-279.22.1.el6.x86_64
by Ionut V. 18 Feb '14

18 Feb '14

Not too much of a Linux user but i got this when trying to install alsa on RHEL 6.4 with kernel - 2.6.32-279.22.1.el6.x86_64 Hope attachments work, if not i will paste the whole output in the replay. -- Ionut V.

1 0

[alsa-devel] SOC_VALUE_ENUM usages in cs42l52 & cs42l73 drivers
by Takashi Iwai 18 Feb '14

18 Feb '14

Hi, as I audited through ASoC codec drivers recently, I noticed that cs42l52 and cs42l73 drivers have some strange SOC_VALUE_ENUM usages. For example, cs42l52.c has the following code snippet: static const char * const left_swap_text[] = { "Left", "LR 2", "Right"}; .... static const unsigned int swap_values[] = { 0, 1, 3 }; static const struct soc_enum adca_swap_enum = SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 1, ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); Note that the third argument of SOC_VALUE_ENUM_SINGLE() is the mask bits. Passing 1 means only the least bit is evaluated no matter what value is assigned while swap_values can be 0, 1 or 3. I suppose here meant 3, but would like to know how the code is supposed to work. There are three value-enum definitions in cs42l52.c, and all pass 1 there. Similarly, cs42l73.c has four definitions, but with different masks: SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 1, ... SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 4, 3, ... SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 5, ... SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 7, ... while values can be either 0, 1 or 2. Something really fishy here. thanks, Takashi

2 2

[alsa-devel] [PATCH 0/2] alsa-lib surround 2.1 support
by David Henningsson 18 Feb '14

18 Feb '14

I'm currently experimenting with different ways to make the system autodetect if surround2.1 is available through a four channel setup (e g laptop surround speaker) and fallback to 5.1 if it isn't. It's a bit tricky, especially given that the LFE speaker could be on left channel, right channel, and both, so meanwhile, this fallback should be better than nothing and help establish a standard for upper parts of the stack (i e PulseAudio). David Henningsson (2): conf: Add basic infrastructure for 2.1 surround sound conf: Add surround 2.1 support to all cards src/conf/alsa.conf | 3 ++ src/conf/cards/ATIIXP-SPDMA.conf | 1 + src/conf/cards/ATIIXP.conf | 1 + src/conf/cards/Audigy.conf | 1 + src/conf/cards/Audigy2.conf | 1 + src/conf/cards/Aureon51.conf | 1 + src/conf/cards/Aureon71.conf | 1 + src/conf/cards/CA0106.conf | 1 + src/conf/cards/CMI8738-MC6.conf | 1 + src/conf/cards/CMI8738-MC8.conf | 1 + src/conf/cards/CMI8788.conf | 2 +- src/conf/cards/CS46xx.conf | 1 + src/conf/cards/EMU10K1.conf | 1 + src/conf/cards/EMU10K1X.conf | 1 + src/conf/cards/FM801.conf | 1 + src/conf/cards/FireWave.conf | 1 + src/conf/cards/HDA-Intel.conf | 1 + src/conf/cards/ICE1712.conf | 1 + src/conf/cards/ICE1724.conf | 1 + src/conf/cards/ICH.conf | 1 + src/conf/cards/ICH4.conf | 1 + src/conf/cards/Loopback.conf | 1 + src/conf/cards/NFORCE.conf | 1 + src/conf/cards/SB-XFi.conf | 1 + src/conf/cards/SI7018.conf | 1 + src/conf/cards/USB-Audio.conf | 1 + src/conf/cards/VIA8233.conf | 1 + src/conf/cards/VIA8233A.conf | 1 + src/conf/cards/VIA8237.conf | 1 + src/conf/pcm/Makefile.am | 2 +- src/conf/pcm/surround21.conf | 62 ++++++++++++++++++++++++++++++++++++++++ 31 files changed, 94 insertions(+), 2 deletions(-) create mode 100644 src/conf/pcm/surround21.conf -- 1.9.rc1

2 3

[alsa-devel] [PATCH 00/15] Update locking for snd_soc_dapm_xxxx_pin functions
by Charles Keepax 18 Feb '14

18 Feb '14

This patchset updates the locking around the snd_soc_dapm_xxxx_pin functions. First we add a locked version of the functions and add locking to the standard call, this also requires us to remove the few places where these functions are externally locked. Then we patch usages of the functions that look like they are expecting to do an atomic update to do so. Thanks, Charles Charles Keepax (15): Input - arizona-haptics: Fix double lock of dapm_mutex ASoC: dapm: Add locked version of snd_soc_dapm_xxxx_pin functions ASoC: adav80x: Update locking around use of DAPM pin API ASoC: wm5100: Update locking around use of DAPM pin API ASoC: wm8962: Update locking around use of DAPM pin API ASoC: wm8994: Update locking around use of DAPM pin API ASoC: wm8996: Update locking around use of DAPM pin API ASoC: mfld_machine: Update locking around use of DAPM pin API ASoC: ams-delta: Update locking around use of DAPM pin API ASoC: omap: n810: Update locking around use of DAPM pin API ASoC: omap: rx51: Update locking around use of DAPM pin API ASoC: pxa: corgi: Update locking around use of DAPM pin API ASoC: pxa: magician: Update locking around use of DAPM pin API ASoC: pxa: spitz: Update locking around use of DAPM pin API ASoC: pxa: tosa: Update locking around use of DAPM pin API drivers/extcon/extcon-arizona.c | 12 --- drivers/input/misc/arizona-haptics.c | 19 ----- include/sound/soc-dapm.h | 8 ++ sound/soc/codecs/adav80x.c | 14 +++- sound/soc/codecs/wm5100.c | 9 ++- sound/soc/codecs/wm8962.c | 12 ++- sound/soc/codecs/wm8994.c | 40 ++++++---- sound/soc/codecs/wm8996.c | 8 ++- sound/soc/intel/mfld_machine.c | 43 ++++++++---- sound/soc/omap/ams-delta.c | 38 ++++++---- sound/soc/omap/n810.c | 20 +++-- sound/soc/omap/rx51.c | 20 +++-- sound/soc/pxa/corgi.c | 40 ++++++----- sound/soc/pxa/magician.c | 20 +++-- sound/soc/pxa/spitz.c | 49 +++++++------ sound/soc/pxa/tosa.c | 26 ++++--- sound/soc/soc-dapm.c | 133 +++++++++++++++++++++++++++++++--- 17 files changed, 338 insertions(+), 173 deletions(-) -- 1.7.2.5

4 24

[alsa-devel] Heavy looping inside snd_pcm_dmix_sync_area with snd_pcm_sw_params_set_stop_threshold > boundary and several hours of play
by Adarsh 18 Feb '14

18 Feb '14

Hi, We are using RW_INTERLEAVED access as described in write_loop method in /test/pcm.c and opening 'default' device. However to disable the underruns, the snd_pcm_sw_params_set_stop_threshold is set to 0x7fffffff which is greater than boundary value. Period size is 1024 and buffer size is 2048. We try to write period_size no of frames using snd_pcm_writei. This setup runs fine for several hours. It was left runing overnight. After some 12 hours of play, gdb is attached to the process and continued inside gdb. At this point, the thread went into a tight loop taking >100% cpu and playing same sound over and over again abusing speakers. It did not recover for a long time. Stack trace shows that 'snd_pcm_dmix_sync_area' in pcm_dmix.c is taking a very long time to return because the 'size' is a large unsinged interger and mix_areas is getting called with 'transfer' value of <=2048. Below is the stack trace: (gdb) bt #0 0xb7c44463 in mix_areas_32_smp (size=1360, dst=0xb6855000, src=0x84a7510, sum=0xb6845000, dst_step=4, src_step=4, sum_step=4) at pcm_dmix_i386.h:261 #1 0xb7c44d2b in mix_areas (dmix=0x84a7110, src_areas=0x84a51d0, dst_areas=0x84a6f00, src_ofs=0, dst_ofs=7168, size=1024) at pcm_dmix.c:193 #2 0xb7c45422 in snd_pcm_dmix_sync_area (pcm=0x84a71e0) at pcm_dmix.c:371 #3 0xb7c46680 in snd_pcm_dmix_mmap_commit (pcm=0x84a71e0, offset=0, size=1024) at pcm_dmix.c:813 #4 0xb7c07033 in snd_pcm_mmap_commit (pcm=0x84a71e0, offset=0, frames=1024) at pcm.c:6501 #5 0xb7c1804e in snd_pcm_plugin_mmap_commit (pcm=0x84a6d18, offset=0, size=1024) at pcm_plugin.c:434 #6 0xb7c07033 in snd_pcm_mmap_commit (pcm=0x84a6d18, offset=0, frames=1024) at pcm.c:6501 #7 0xb7c17956 in snd_pcm_plugin_write_areas (pcm=0x84a5390, areas=0xb68362b0, offset=0, size=1024) at pcm_plugin.c:290 #8 0xb7c074bb in snd1_pcm_write_areas (pcm=0x84a5390, areas=0xb68362b0, offset=0, size=1024, func=0xb7c177fc <snd_pcm_plugin_write_areas>) at pcm.c:6667 #9 0xb7c17cd5 in snd_pcm_plugin_writei (pcm=0x84a5390, buffer=0x84ab518, size=1024) at pcm_plugin.c:361 #10 0xb7bff1a7 in _snd_pcm_writei (pcm=0x84a6b18, buffer=0x84ab518, size=1024) at pcm_local.h:516 #11 0xb7bff16c in snd_pcm_writei (pcm=0x84a6b18, buffer=0x84ab518, size=1024) at pcm.c:1258 (gdb) f 2 #2 0xb7c45422 in snd_pcm_dmix_sync_area (pcm=0x84a71e0) at pcm_dmix.c:371 371 pcm_dmix.c: No such file or directory. (gdb) info locals dmix = 0x84a7110 slave_hw_ptr = 234156032 slave_appl_ptr = 7168 slave_size = 1057171456 appl_ptr = 0 size = 1018295296 transfer = 1024 src_areas = 0x84a51d0 dst_areas = 0x84a6f00 (gdb) p *dmix $1 = {type = SND_PCM_TYPE_DMIX, ipc_key = 5678293, ipc_perm = 432, ipc_gid = 17, semid = 32768, shmid = 65536, shmptr = 0xb6859000, spcm = 0x84a7340, appl_ptr = 217121792, last_appl_ptr = 233882624, hw_ptr = 233875428, avail_max = 0, slave_appl_ptr = 234156032, slave_hw_ptr = 234147996, slave_period_size = 1024, slave_buffer_size = 8192, slave_boundary = 1073741824, sync_ptr = 0xb7c4557c <snd_pcm_dmix_sync_ptr>, state = SND_PCM_STATE_RUNNING, trigger_tstamp = {tv_sec = 1392384066, tv_nsec = 8308000}, server = 0, client = 0, comm_fd = 0, hw_fd = 4, timer_fd = { fd = 3, events = 41, revents = 0}, poll_fd = 3, tread = 1, timer_need_poll = 0, timer_event_suspend = 131072, timer_event_resume = 262144, server_fd = 0, server_pid = 0, timer = 0x8496b90, interleaved = 1, slowptr = 1, max_periods = 0, channels = 2, bindings = 0x0, u = {dmix = { shmid_sum = 98305, sum_buffer = 0xb6837000, mix_areas_16 = 0xb7c443a8 <mix_areas_16_smp_mmx>, mix_areas_32 = 0xb7c44415 <mix_areas_32_smp>, mix_areas_24 = 0xb7c4451b <mix_areas_24_smp_cmov>, mix_areas_u8 = 0, remix_areas_16 = 0xb7c44638 <remix_areas_16_smp_mmx>, remix_areas_32 = 0xb7c446a5 <remix_areas_32_smp>, remix_areas_24 = 0xb7c447ab <remix_areas_24_smp_cmov>, remix_areas_u8 = 0}, dsnoop = {<No data fields>}, dshare = {chn_mask = 13151478482154389505}}, server_free = 0} (gdb) p *pcm $2 = {dl_handle = 0x0, name = 0x84a6e48 "dmix:0", type = SND_PCM_TYPE_DMIX, stream = SND_PCM_STREAM_PLAYBACK, mode = 0, poll_fd_count = 1, poll_fd = 3, poll_events = 1, setup = -1, monotonic = 0, access = SND_PCM_ACCESS_MMAP_INTERLEAVED, format = SND_PCM_FORMAT_S32_LE, subformat = SND_PCM_SUBFORMAT_STD, channels = 2, rate = 44100, period_size = 1024, period_time = 23219, periods = {min = 0, max = 0, openmin = 0, openmax = 0, integer = 0, empty = 0}, tstamp_mode = SND_PCM_TSTAMP_NONE, period_step = 1, avail_min = 1024, period_event = 0, start_threshold = 1024, stop_threshold = 2147483647, silence_threshold = 0, silence_size = 0, boundary = 1073741824, info = 2151743747, msbits = 32, rate_num = 44100, rate_den = 1, hw_flags = 0, fifo_size = 0, buffer_size = 2048, buffer_time = {min = 0, max = 0, openmin = 0, openmax = 0, integer = 0, empty = 0}, sample_bits = 32, frame_bits = 64, appl = {master = 0x0, ptr = 0x84a7130, fd = -1, offset = 0, link_dst_count = 0, link_dst = 0x0, private_data = 0x0, changed = 0}, hw = { master = 0x0, ptr = 0x84a7138, fd = -1, offset = 0, link_dst_count = 0, link_dst = 0x0, private_data = 0x0, changed = 0}, min_align = 1, mmap_rw = 1, mmap_shadow = 0, donot_close = 0, mmap_channels = 0x84a54a0, running_areas = 0x84a51d0, stopped_areas = 0x0, ops = 0xb7c91860, fast_ops = 0xb7c918a0, op_arg = 0x84a71e0, fast_op_arg = 0x84a71e0, private_data = 0x84a7110, async_handlers = {next = 0x84a72e4, prev = 0x84a72e4}} Attaching gdb does not show this behaviour if snd_pcm_sw_params_set_stop_threshold is not set, which then defaults to buffer size or when the application is running for shorter duration. Is it that setting of stop threshold to 0x7fffffff is triggering this and attaching gdb is making the buffer pointers misbehave? If so, what can be done if we do not want PCM to stop. Below is the pcm dump: Plug PCM: Linear conversion PCM (S32_LE) 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 : 2048 period_size : 1024 period_time : 23219 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1024 stop_threshold : 2147483647 silence_threshold: 0 silence_size : 0 boundary : 1073741824 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 : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 2048 period_size : 1024 period_time : 23219 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1024 stop_threshold : 2147483647 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 : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 2048 period_size : 1024 period_time : 23219 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1024 stop_threshold : 2147483647 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 23219 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 Thanks

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[alsa-devel] Heavy looping inside snd_pcm_dmix_sync_area with snd_pcm_sw_params_set_stop_threshold > boundary and several hours of play
by Adarsh 18 Feb '14

18 Feb '14

Hi, We are using RW_INTERLEAVED access as described in write_loop method in /test/pcm.c and opening 'default' device. However to disable the underruns, the snd_pcm_sw_params_set_stop_threshold is set to 0x7fffffff which is greater than boundary value. Period size is 1024 and buffer size is 2048. We try to write period_size no of frames using snd_pcm_writei. This setup runs fine for several hours. It was left runing overnight. After some 12 hours of play, gdb is attached to the process and continued inside gdb. At this point, the thread went into a tight loop taking >100% cpu and playing same sound over and over again abusing speakers. It did not recover for a long time. Stack trace shows that 'snd_pcm_dmix_sync_area' in pcm_dmix.c is taking a very long time to return because the 'size' is a large unsinged interger and mix_areas is getting called with 'transfer' value of <=2048. Below is the stack trace: (gdb) bt #0 0xb7c44463 in mix_areas_32_smp (size=1360, dst=0xb6855000, src=0x84a7510, sum=0xb6845000, dst_step=4, src_step=4, sum_step=4) at pcm_dmix_i386.h:261 #1 0xb7c44d2b in mix_areas (dmix=0x84a7110, src_areas=0x84a51d0, dst_areas=0x84a6f00, src_ofs=0, dst_ofs=7168, size=1024) at pcm_dmix.c:193 #2 0xb7c45422 in snd_pcm_dmix_sync_area (pcm=0x84a71e0) at pcm_dmix.c:371 #3 0xb7c46680 in snd_pcm_dmix_mmap_commit (pcm=0x84a71e0, offset=0, size=1024) at pcm_dmix.c:813 #4 0xb7c07033 in snd_pcm_mmap_commit (pcm=0x84a71e0, offset=0, frames=1024) at pcm.c:6501 #5 0xb7c1804e in snd_pcm_plugin_mmap_commit (pcm=0x84a6d18, offset=0, size=1024) at pcm_plugin.c:434 #6 0xb7c07033 in snd_pcm_mmap_commit (pcm=0x84a6d18, offset=0, frames=1024) at pcm.c:6501 #7 0xb7c17956 in snd_pcm_plugin_write_areas (pcm=0x84a5390, areas=0xb68362b0, offset=0, size=1024) at pcm_plugin.c:290 #8 0xb7c074bb in snd1_pcm_write_areas (pcm=0x84a5390, areas=0xb68362b0, offset=0, size=1024, func=0xb7c177fc <snd_pcm_plugin_write_areas>) at pcm.c:6667 #9 0xb7c17cd5 in snd_pcm_plugin_writei (pcm=0x84a5390, buffer=0x84ab518, size=1024) at pcm_plugin.c:361 #10 0xb7bff1a7 in _snd_pcm_writei (pcm=0x84a6b18, buffer=0x84ab518, size=1024) at pcm_local.h:516 #11 0xb7bff16c in snd_pcm_writei (pcm=0x84a6b18, buffer=0x84ab518, size=1024) at pcm.c:1258 (gdb) f 2 #2 0xb7c45422 in snd_pcm_dmix_sync_area (pcm=0x84a71e0) at pcm_dmix.c:371 371 pcm_dmix.c: No such file or directory. (gdb) info locals dmix = 0x84a7110 slave_hw_ptr = 234156032 slave_appl_ptr = 7168 slave_size = 1057171456 appl_ptr = 0 size = 1018295296 transfer = 1024 src_areas = 0x84a51d0 dst_areas = 0x84a6f00 (gdb) p *dmix $1 = {type = SND_PCM_TYPE_DMIX, ipc_key = 5678293, ipc_perm = 432, ipc_gid = 17, semid = 32768, shmid = 65536, shmptr = 0xb6859000, spcm = 0x84a7340, appl_ptr = 217121792, last_appl_ptr = 233882624, hw_ptr = 233875428, avail_max = 0, slave_appl_ptr = 234156032, slave_hw_ptr = 234147996, slave_period_size = 1024, slave_buffer_size = 8192, slave_boundary = 1073741824, sync_ptr = 0xb7c4557c <snd_pcm_dmix_sync_ptr>, state = SND_PCM_STATE_RUNNING, trigger_tstamp = {tv_sec = 1392384066, tv_nsec = 8308000}, server = 0, client = 0, comm_fd = 0, hw_fd = 4, timer_fd = { fd = 3, events = 41, revents = 0}, poll_fd = 3, tread = 1, timer_need_poll = 0, timer_event_suspend = 131072, timer_event_resume = 262144, server_fd = 0, server_pid = 0, timer = 0x8496b90, interleaved = 1, slowptr = 1, max_periods = 0, channels = 2, bindings = 0x0, u = {dmix = { shmid_sum = 98305, sum_buffer = 0xb6837000, mix_areas_16 = 0xb7c443a8 <mix_areas_16_smp_mmx>, mix_areas_32 = 0xb7c44415 <mix_areas_32_smp>, mix_areas_24 = 0xb7c4451b <mix_areas_24_smp_cmov>, mix_areas_u8 = 0, remix_areas_16 = 0xb7c44638 <remix_areas_16_smp_mmx>, remix_areas_32 = 0xb7c446a5 <remix_areas_32_smp>, remix_areas_24 = 0xb7c447ab <remix_areas_24_smp_cmov>, remix_areas_u8 = 0}, dsnoop = {<No data fields>}, dshare = {chn_mask = 13151478482154389505}}, server_free = 0} (gdb) p *pcm $2 = {dl_handle = 0x0, name = 0x84a6e48 "dmix:0", type = SND_PCM_TYPE_DMIX, stream = SND_PCM_STREAM_PLAYBACK, mode = 0, poll_fd_count = 1, poll_fd = 3, poll_events = 1, setup = -1, monotonic = 0, access = SND_PCM_ACCESS_MMAP_INTERLEAVED, format = SND_PCM_FORMAT_S32_LE, subformat = SND_PCM_SUBFORMAT_STD, channels = 2, rate = 44100, period_size = 1024, period_time = 23219, periods = {min = 0, max = 0, openmin = 0, openmax = 0, integer = 0, empty = 0}, tstamp_mode = SND_PCM_TSTAMP_NONE, period_step = 1, avail_min = 1024, period_event = 0, start_threshold = 1024, stop_threshold = 2147483647, silence_threshold = 0, silence_size = 0, boundary = 1073741824, info = 2151743747, msbits = 32, rate_num = 44100, rate_den = 1, hw_flags = 0, fifo_size = 0, buffer_size = 2048, buffer_time = {min = 0, max = 0, openmin = 0, openmax = 0, integer = 0, empty = 0}, sample_bits = 32, frame_bits = 64, appl = {master = 0x0, ptr = 0x84a7130, fd = -1, offset = 0, link_dst_count = 0, link_dst = 0x0, private_data = 0x0, changed = 0}, hw = { master = 0x0, ptr = 0x84a7138, fd = -1, offset = 0, link_dst_count = 0, link_dst = 0x0, private_data = 0x0, changed = 0}, min_align = 1, mmap_rw = 1, mmap_shadow = 0, donot_close = 0, mmap_channels = 0x84a54a0, running_areas = 0x84a51d0, stopped_areas = 0x0, ops = 0xb7c91860, fast_ops = 0xb7c918a0, op_arg = 0x84a71e0, fast_op_arg = 0x84a71e0, private_data = 0x84a7110, async_handlers = {next = 0x84a72e4, prev = 0x84a72e4}} Attaching gdb does not show this behaviour if snd_pcm_sw_params_set_stop_threshold is not set, which then defaults to buffer size or when the application is running for shorter duration. Is it that setting of stop threshold to 0x7fffffff is triggering this and attaching gdb is making the buffer pointers misbehave? If so, what can be done if we do not want PCM to stop. Below is the pcm dump: Plug PCM: Linear conversion PCM (S32_LE) 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 : 2048 period_size : 1024 period_time : 23219 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1024 stop_threshold : 2147483647 silence_threshold: 0 silence_size : 0 boundary : 1073741824 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 : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 2048 period_size : 1024 period_time : 23219 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1024 stop_threshold : 2147483647 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 : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 2048 period_size : 1024 period_time : 23219 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1024 stop_threshold : 2147483647 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 23219 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 Thanks

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[alsa-devel] [PATCHv2 1/5] ASoC: Intel: Add Intel SST audio DSP low level shim driver.
by Liam Girdwood 18 Feb '14

18 Feb '14

Add support for Intel Smart Sound Technology (SST) audio DSPs. This driver provides the low level IO, reset, boot and IRQ management for Intel audio DSPs. These files make up the low level part of the SST audio driver stack and will be used by many Intel SST cores like Haswell, Broadwell and Baytrail. SST DSPs expose a memory mapped region (shim) for config and control. The shim layout is mostly shared without much modification across cores and this driver provides a uniform API to access the shim and to enable basic shim functions. It also provides functionality to abstract some shim functions for cores with different shim features. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-dsp-priv.h | 305 ++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-dsp.c | 324 +++++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-dsp.h | 212 +++++++++++++++++++++++++++ 3 files changed, 841 insertions(+) create mode 100644 sound/soc/intel/sst-dsp-priv.h create mode 100644 sound/soc/intel/sst-dsp.c create mode 100644 sound/soc/intel/sst-dsp.h diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h new file mode 100644 index 0000000..35de547 --- /dev/null +++ b/sound/soc/intel/sst-dsp-priv.h @@ -0,0 +1,305 @@ +/* + * Intel Smart Sound Technology + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#ifndef __SOUND_SOC_SST_DSP_PRIV_H +#define __SOUND_SOC_SST_DSP_PRIV_H + +#include <linux/kernel.h> +#include <linux/types.h> +#include <linux/interrupt.h> +#include <linux/firmware.h> + +struct sst_mem_block; +struct sst_module; +struct sst_fw; + +/* + * DSP Operations exported by platform Audio DSP driver. + */ +struct sst_ops { + /* DSP core boot / reset */ + void (*boot)(struct sst_dsp *); + void (*reset)(struct sst_dsp *); + + /* Shim IO */ + void (*write)(void __iomem *addr, u32 offset, u32 value); + u32 (*read)(void __iomem *addr, u32 offset); + void (*write64)(void __iomem *addr, u32 offset, u64 value); + u64 (*read64)(void __iomem *addr, u32 offset); + + /* DSP I/DRAM IO */ + void (*ram_read)(struct sst_dsp *sst, void *dest, void *src, size_t bytes); + void (*ram_write)(struct sst_dsp *sst, void *dest, void *src, size_t bytes); + + void (*dump)(struct sst_dsp *); + + /* IRQ handlers */ + irqreturn_t (*irq_handler)(int irq, void *context); + + /* SST init and free */ + int (*init)(struct sst_dsp *sst, struct sst_pdata *pdata); + void (*free)(struct sst_dsp *sst); + + /* FW module parser/loader */ + int (*parse_fw)(struct sst_fw *sst_fw); +}; + +/* + * Audio DSP memory offsets and addresses. + */ +struct sst_addr { + u32 lpe_base; + u32 shim_offset; + u32 iram_offset; + void __iomem *lpe; + void __iomem *shim; + void __iomem *pci_cfg; + void __iomem *fw_ext; +}; + +/* + * Audio DSP Mailbox configuration. + */ +struct sst_mailbox { + void __iomem *in_base; + void __iomem *out_base; + size_t in_size; + size_t out_size; +}; + +/* + * Audio DSP Firmware data types. + */ +enum sst_data_type { + SST_DATA_M = 0, /* module block data */ + SST_DATA_P = 1, /* peristant data (text, data) */ + SST_DATA_S = 2, /* scratch data (usually buffers) */ +}; + +/* + * Audio DSP memory block types. + */ +enum sst_mem_type { + SST_MEM_IRAM = 0, + SST_MEM_DRAM = 1, + SST_MEM_ANY = 2, + SST_MEM_CACHE= 3, +}; + +/* + * Audio DSP Generic Firmware File. + * + * SST Firmware files can consist of 1..N modules. This generic structure is + * used to manage each firmware file and it's modules regardless of SST firmware + * type. A SST driver may load multiple FW files. + */ +struct sst_fw { + struct sst_dsp *dsp; + + /* base addresses of FW file data */ + dma_addr_t dmable_fw_paddr; /* physical address of fw data */ + void *dma_buf; /* virtual address of fw data */ + u32 size; /* size of fw data */ + + /* lists */ + struct list_head list; /* DSP list of FW */ + struct list_head module_list; /* FW list of modules */ + + void *private; /* core doesn't touch this */ +}; + +/* + * Audio DSP Generic Module data. + * + * This is used to dsecribe any sections of persistent (text and data) and + * scratch (buffers) of module data in ADSP memory space. + */ +struct sst_module_data { + + enum sst_mem_type type; /* destination memory type */ + enum sst_data_type data_type; /* type of module data */ + + u32 size; /* size in bytes */ + u32 offset; /* offset in FW file */ + u32 data_offset; /* offset in ADSP memory space */ + void *data; /* module data */ +}; + +/* + * Audio DSP Generic Module Template. + * + * Used to define and register a new FW module. This data is extracted from + * FW module header information. + */ +struct sst_module_template { + u32 id; + u32 entry; /* entry point */ + struct sst_module_data s; /* scratch data */ + struct sst_module_data p; /* peristant data */ +}; + +/* + * Audio DSP Generic Module. + * + * Each Firmware file can consist of 1..N modules. A module can span multiple + * ADSP memory blocks. The simplest FW will be a file with 1 module. + */ +struct sst_module { + struct sst_dsp *dsp; + struct sst_fw *sst_fw; /* parent FW we belong too */ + + /* module configuration */ + u32 id; + u32 entry; /* module entry point */ + u32 offset; /* module offset in firmware file */ + u32 size; /* module size */ + struct sst_module_data s; /* scratch data */ + struct sst_module_data p; /* peristant data */ + + /* runtime */ + u32 usage_count; /* can be unloaded if count == 0 */ + void *private; /* core doesn't touch this */ + + /* lists */ + struct list_head block_list; /* Module list of blocks in use */ + struct list_head list; /* DSP list of modules */ + struct list_head list_fw; /* FW list of modules */ +}; + +/* + * SST Memory Block operations. + */ +struct sst_block_ops { + int (*enable)(struct sst_mem_block *block); + int (*disable)(struct sst_mem_block *block); +}; + +/* + * SST Generic Memory Block. + * + * SST ADP memory has multiple IRAM and DRAM blocks. Some ADSP blocks can be + * power gated. + */ +struct sst_mem_block { + struct sst_dsp *dsp; + struct sst_module *module; /* module that uses this block */ + + /* block config */ + u32 offset; /* offset from base */ + u32 size; /* block size */ + u32 index; /* block index 0..N */ + enum sst_mem_type type; /* block memory type IRAM/DRAM */ + struct sst_block_ops *ops; /* block operations, if any */ + + /* block status */ + enum sst_data_type data_type; /* data type held in this block */ + u32 bytes_used; /* bytes in use by modules */ + void *private; /* generic core does not touch this */ + int users; /* number of modules using this block */ + + /* block lists */ + struct list_head module_list; /* Module list of blocks */ + struct list_head list; /* Map list of free/used blocks */ +}; + +/* + * Generic SST Shim Interface. + */ +struct sst_dsp { + + /* runtime */ + struct sst_dsp_device *sst_dev; + spinlock_t spinlock; /* IPC locking */ + struct mutex mutex; /* DSP FW lock */ + struct device *dev; + void *thread_context; + int irq; + u32 id; + + /* list of free and used ADSP memory blocks */ + struct list_head used_block_list; + struct list_head free_block_list; + + /* operations */ + struct sst_ops *ops; + + /* debug FS */ + struct dentry *debugfs_root; + + /* base addresses */ + struct sst_addr addr; + + /* mailbox */ + struct sst_mailbox mailbox; + + /* SST FW files loaded and their modules */ + struct list_head module_list; + struct list_head fw_list; + + /* platform data */ + struct sst_pdata *pdata; + + /* DMA FW loading */ + struct sst_dma *dma; + bool fw_use_dma; +}; + +/* Size optimised DRAM/IRAM memcpy */ +static inline void sst_dsp_write(struct sst_dsp *sst, void *src, + u32 dest_offset, size_t bytes) +{ + sst->ops->ram_write(sst, sst->addr.lpe + dest_offset, src, bytes); +} + +static inline void sst_dsp_read(struct sst_dsp *sst, void *dest, + u32 src_offset, size_t bytes) +{ + sst->ops->ram_read(sst, dest, sst->addr.lpe + src_offset, bytes); +} + +static inline void *sst_dsp_get_thread_context(struct sst_dsp *sst) +{ + return sst->thread_context; +} + +/* Create/Free FW files - can contain multiple modules */ +struct sst_fw *sst_fw_new(struct sst_dsp *dsp, + const struct firmware *fw, void *private); +void sst_fw_free(struct sst_fw *sst_fw); +void sst_fw_free_all(struct sst_dsp *dsp); + +/* Create/Free firmware modules */ +struct sst_module *sst_module_new(struct sst_fw *sst_fw, + struct sst_module_template *template, void *private); +void sst_module_free(struct sst_module *sst_module); +int sst_module_insert(struct sst_module *sst_module); +int sst_module_remove(struct sst_module *sst_module); +int sst_module_insert_fixed_block(struct sst_module *module, + struct sst_module_data *data); +struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id); + +/* allocate/free pesistent/scratch memory regions managed by drv */ +struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp); +void sst_mem_block_free_scratch(struct sst_dsp *dsp, + struct sst_module *scratch); + +/* Register the DSPs memory blocks - would be nice to read from ACPI */ +struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset, + u32 size, enum sst_mem_type type, struct sst_block_ops *ops, u32 index, + void *private); +void sst_mem_block_unregister_all(struct sst_dsp *dsp); + +#endif diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c new file mode 100644 index 0000000..1888de5 --- /dev/null +++ b/sound/soc/intel/sst-dsp.c @@ -0,0 +1,324 @@ +/* + * Intel Smart Sound Technology (SST) DSP Core Driver + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include <linux/slab.h> +#include <linux/export.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> + +#include "sst-dsp.h" +#include "sst-dsp-priv.h" + +#define CREATE_TRACE_POINTS +#include <trace/events/sst.h> + +/* Public API */ +void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value) +{ + unsigned long flags; + + spin_lock_irqsave(&sst->spinlock, flags); + sst->ops->write(sst->addr.shim, offset, value); + spin_unlock_irqrestore(&sst->spinlock, flags); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_write); + +u32 sst_dsp_shim_read(struct sst_dsp *sst, u32 offset) +{ + unsigned long flags; + u32 val; + + spin_lock_irqsave(&sst->spinlock, flags); + val = sst->ops->read(sst->addr.shim, offset); + spin_unlock_irqrestore(&sst->spinlock, flags); + + return val; +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_read); + +void sst_dsp_shim_write64(struct sst_dsp *sst, u32 offset, u64 value) +{ + unsigned long flags; + + spin_lock_irqsave(&sst->spinlock, flags); + sst->ops->write64(sst->addr.shim, offset, value); + spin_unlock_irqrestore(&sst->spinlock, flags); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_write64); + +u64 sst_dsp_shim_read64(struct sst_dsp *sst, u32 offset) +{ + unsigned long flags; + u64 val; + + spin_lock_irqsave(&sst->spinlock, flags); + val = sst->ops->read64(sst->addr.shim, offset); + spin_unlock_irqrestore(&sst->spinlock, flags); + + return val; +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_read64); + +void sst_dsp_shim_write_unlocked(struct sst_dsp *sst, u32 offset, u32 value) +{ + sst->ops->write(sst->addr.shim, offset, value); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_write_unlocked); + +u32 sst_dsp_shim_read_unlocked(struct sst_dsp *sst, u32 offset) +{ + return sst->ops->read(sst->addr.shim, offset); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_read_unlocked); + +void sst_dsp_shim_write64_unlocked(struct sst_dsp *sst, u32 offset, u64 value) +{ + sst->ops->write64(sst->addr.shim, offset, value); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_write64_unlocked); + +u64 sst_dsp_shim_read64_unlocked(struct sst_dsp *sst, u32 offset) +{ + return sst->ops->read64(sst->addr.shim, offset); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_read64_unlocked); + +int sst_dsp_shim_update_bits_unlocked(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value) +{ + bool change; + unsigned int old, new; + u32 ret; + + ret = sst_dsp_shim_read_unlocked(sst, offset); + + old = ret; + new = (old & (~mask)) | (value & mask); + + change = (old != new); + if (change) + sst_dsp_shim_write_unlocked(sst, offset, new); + + return change; +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits_unlocked); + +int sst_dsp_shim_update_bits64_unlocked(struct sst_dsp *sst, u32 offset, + u64 mask, u64 value) +{ + bool change; + u64 old, new; + + old = sst_dsp_shim_read64_unlocked(sst, offset); + + new = (old & (~mask)) | (value & mask); + + change = (old != new); + if (change) + sst_dsp_shim_write64_unlocked(sst, offset, new); + + return change; +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64_unlocked); + +int sst_dsp_shim_update_bits(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value) +{ + unsigned long flags; + bool change; + + spin_lock_irqsave(&sst->spinlock, flags); + change = sst_dsp_shim_update_bits_unlocked(sst, offset, mask, value); + spin_unlock_irqrestore(&sst->spinlock, flags); + return change; +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits); + +int sst_dsp_shim_update_bits64(struct sst_dsp *sst, u32 offset, + u64 mask, u64 value) +{ + unsigned long flags; + bool change; + + spin_lock_irqsave(&sst->spinlock, flags); + change = sst_dsp_shim_update_bits64_unlocked(sst, offset, mask, value); + spin_unlock_irqrestore(&sst->spinlock, flags); + return change; +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64); + +void sst_dsp_dump(struct sst_dsp *sst) +{ + sst->ops->dump(sst); +} +EXPORT_SYMBOL_GPL(sst_dsp_dump); + +void sst_dsp_reset(struct sst_dsp *sst) +{ + sst->ops->reset(sst); +} +EXPORT_SYMBOL_GPL(sst_dsp_reset); + +int sst_dsp_boot(struct sst_dsp *sst) +{ + sst->ops->boot(sst); + return 0; +} +EXPORT_SYMBOL_GPL(sst_dsp_boot); + +void sst_dsp_ipc_msg_tx(struct sst_dsp *dsp, u32 msg) +{ + sst_dsp_shim_write_unlocked(dsp, SST_IPCX, msg | SST_IPCX_BUSY); + trace_sst_ipc_msg_tx(msg); +} +EXPORT_SYMBOL_GPL(sst_dsp_ipc_msg_tx); + +u32 sst_dsp_ipc_msg_rx(struct sst_dsp *dsp) +{ + u32 msg; + + msg = sst_dsp_shim_read_unlocked(dsp, SST_IPCX); + trace_sst_ipc_msg_rx(msg); + + return msg; +} +EXPORT_SYMBOL_GPL(sst_dsp_ipc_msg_rx); + +int sst_dsp_mailbox_init(struct sst_dsp *sst, u32 inbox_offset, size_t inbox_size, + u32 outbox_offset, size_t outbox_size) +{ + sst->mailbox.in_base = sst->addr.lpe + inbox_offset; + sst->mailbox.out_base = sst->addr.lpe + outbox_offset; + sst->mailbox.in_size = inbox_size; + sst->mailbox.out_size = outbox_size; + return 0; +} +EXPORT_SYMBOL_GPL(sst_dsp_mailbox_init); + +void sst_dsp_outbox_write(struct sst_dsp *sst, void *message, size_t bytes) +{ + u32 i; + + trace_sst_ipc_outbox_write(bytes); + + memcpy_toio(sst->mailbox.out_base, message, bytes); + + for (i = 0; i < bytes; i += 4) + trace_sst_ipc_outbox_wdata(i, *(u32 *)(message + i)); +} +EXPORT_SYMBOL_GPL(sst_dsp_outbox_write); + +void sst_dsp_outbox_read(struct sst_dsp *sst, void *message, size_t bytes) +{ + u32 i; + + trace_sst_ipc_outbox_read(bytes); + + memcpy_fromio(message, sst->mailbox.out_base, bytes); + + for (i = 0; i < bytes; i += 4) + trace_sst_ipc_outbox_rdata(i, *(u32 *)(message + i)); +} +EXPORT_SYMBOL_GPL(sst_dsp_outbox_read); + +void sst_dsp_inbox_write(struct sst_dsp *sst, void *message, size_t bytes) +{ + u32 i; + + trace_sst_ipc_inbox_write(bytes); + + memcpy_toio(sst->mailbox.in_base, message, bytes); + + for (i = 0; i < bytes; i += 4) + trace_sst_ipc_inbox_wdata(i, *(u32 *)(message + i)); +} +EXPORT_SYMBOL_GPL(sst_dsp_inbox_write); + +void sst_dsp_inbox_read(struct sst_dsp *sst, void *message, size_t bytes) +{ + u32 i; + + trace_sst_ipc_inbox_read(bytes); + + memcpy_fromio(message, sst->mailbox.in_base, bytes); + + for (i = 0; i < bytes; i += 4) + trace_sst_ipc_inbox_rdata(i, *(u32 *)(message + i)); +} +EXPORT_SYMBOL_GPL(sst_dsp_inbox_read); + +struct sst_dsp *sst_dsp_new(struct device *dev, + struct sst_dsp_device *sst_dev, struct sst_pdata *pdata) +{ + struct sst_dsp *sst; + int err; + + dev_dbg(dev, "initialising audio DSP id 0x%x\n", pdata->id); + + sst = devm_kzalloc(dev, sizeof(*sst), GFP_KERNEL); + if (sst == NULL) + return NULL; + + spin_lock_init(&sst->spinlock); + mutex_init(&sst->mutex); + sst->dev = dev; + sst->thread_context = sst_dev->thread_context; + sst->sst_dev = sst_dev; + sst->id = pdata->id; + sst->irq = pdata->irq; + sst->ops = sst_dev->ops; + sst->pdata = pdata; + INIT_LIST_HEAD(&sst->used_block_list); + INIT_LIST_HEAD(&sst->free_block_list); + INIT_LIST_HEAD(&sst->module_list); + INIT_LIST_HEAD(&sst->fw_list); + + /* Initialise SST Audio DSP */ + if (sst->ops->init) { + err = sst->ops->init(sst, pdata); + if (err < 0) + return NULL; + } + + /* Register the ISR */ + err = request_threaded_irq(sst->irq, sst->ops->irq_handler, + sst_dev->thread, IRQF_SHARED, "AudioDSP", sst); + if (err) + goto irq_err; + + return sst; + +irq_err: + if (sst->ops->free) + sst->ops->free(sst); + + return NULL; +} +EXPORT_SYMBOL_GPL(sst_dsp_new); + +void sst_dsp_free(struct sst_dsp *sst) +{ + free_irq(sst->irq, sst); + if (sst->ops->free) + sst->ops->free(sst); +} +EXPORT_SYMBOL_GPL(sst_dsp_free); + +/* Module information */ +MODULE_AUTHOR("Liam Girdwood"); +MODULE_DESCRIPTION("Intel SST Core"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h new file mode 100644 index 0000000..0ce5c8d --- /dev/null +++ b/sound/soc/intel/sst-dsp.h @@ -0,0 +1,212 @@ +/* + * Intel Smart Sound Technology (SST) Core + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#ifndef __SOUND_SOC_SST_DSP_H +#define __SOUND_SOC_SST_DSP_H + +#include <linux/kernel.h> +#include <linux/types.h> +#include <linux/interrupt.h> + +/* SST Device IDs */ +#define SST_DEV_ID_LYNX_POINT 0x33C8 +#define SST_DEV_ID_WILDCAT_POINT 0x3438 + +/* Supported SST DMA Devices */ +#define SST_DMA_TYPE_DW 1 +#define SST_DMA_TYPE_MID 2 + +/* SST Shim register map + * The register naming can differ between products. Some products also + * contain extra functionality. + */ +#define SST_CSR 0x00 +#define SST_PISR 0x08 +#define SST_PIMR 0x10 +#define SST_ISRX 0x18 +#define SST_ISRD 0x20 +#define SST_IMRX 0x28 +#define SST_IMRD 0x30 +#define SST_IPCX 0x38 /* IPC IA -> SST */ +#define SST_IPCD 0x40 /* IPC SST -> IA */ +#define SST_ISRSC 0x48 +#define SST_ISRLPESC 0x50 +#define SST_IMRSC 0x58 +#define SST_IMRLPESC 0x60 +#define SST_IPCSC 0x68 +#define SST_IPCLPESC 0x70 +#define SST_CLKCTL 0x78 +#define SST_CSR2 0x80 +#define SST_LTRC 0xE0 +#define SST_HDMC 0xE8 +#define SST_DBGO 0xF0 + +#define SST_SHIM_SIZE 0x100 +#define SST_PWMCTRL 0x1000 + +/* SST Shim Register bits + * The register bit naming can differ between products. Some products also + * contain extra functionality. + */ + +/* CSR / CS */ +#define SST_CSR_RST (0x1 << 1) +#define SST_CSR_SBCS0 (0x1 << 2) +#define SST_CSR_SBCS1 (0x1 << 3) +#define SST_CSR_DCS(x) (x << 4) +#define SST_CSR_DCS_MASK (0x7 << 4) +#define SST_CSR_STALL (0x1 << 10) +#define SST_CSR_S0IOCS (0x1 << 21) +#define SST_CSR_S1IOCS (0x1 << 23) +#define SST_CSR_LPCS (0x1 << 31) + +/* ISRX / ISC */ +#define SST_ISRX_BUSY (0x1 << 1) +#define SST_ISRX_DONE (0x1 << 0) + +/* ISRD / ISD */ +#define SST_ISRD_BUSY (0x1 << 1) +#define SST_ISRD_DONE (0x1 << 0) + +/* IMRX / IMC */ +#define SST_IMRX_BUSY (0x1 << 1) +#define SST_IMRX_DONE (0x1 << 0) + +/* IPCX / IPCC */ +#define SST_IPCX_DONE (0x1 << 30) +#define SST_IPCX_BUSY (0x1 << 31) + +/* IPCD */ +#define SST_IPCD_DONE (0x1 << 30) +#define SST_IPCD_BUSY (0x1 << 31) + +/* CLKCTL */ +#define SST_CLKCTL_SMOS(x) (x << 24) +#define SST_CLKCTL_MASK (3 << 24) +#define SST_CLKCTL_DCPLCG (1 << 18) +#define SST_CLKCTL_SCOE1 (1 << 17) +#define SST_CLKCTL_SCOE0 (1 << 16) + +/* CSR2 / CS2 */ +#define SST_CSR2_SDFD_SSP0 (1 << 1) +#define SST_CSR2_SDFD_SSP1 (1 << 2) + +/* LTRC */ +#define SST_LTRC_VAL(x) (x << 0) + +/* HDMC */ +#define SST_HDMC_HDDA0(x) (x << 0) +#define SST_HDMC_HDDA1(x) (x << 7) + + +/* SST Vendor Defined Registers and bits */ +#define SST_VDRTCTL0 0xa0 +#define SST_VDRTCTL1 0xa4 +#define SST_VDRTCTL2 0xa8 +#define SST_VDRTCTL3 0xaC + +/* VDRTCTL0 */ +#define SST_VDRTCL0_DSRAMPGE_SHIFT 16 +#define SST_VDRTCL0_DSRAMPGE_MASK (0xffff << SST_VDRTCL0_DSRAMPGE_SHIFT) +#define SST_VDRTCL0_ISRAMPGE_SHIFT 6 +#define SST_VDRTCL0_ISRAMPGE_MASK (0x3ff << SST_VDRTCL0_ISRAMPGE_SHIFT) + +struct sst_dsp; + +/* + * SST Device. + * + * This structure is populated by the SST core driver. + */ +struct sst_dsp_device { + /* Mandatory fields */ + struct sst_ops *ops; + irqreturn_t (*thread)(int irq, void *context); + void *thread_context; +}; + +/* + * SST Platform Data. + */ +struct sst_pdata { + /* ACPI data */ + u32 lpe_base; + u32 lpe_size; + u32 pcicfg_base; + u32 pcicfg_size; + int irq; + + /* Firmware */ + const char *fw_filename; + u32 fw_base; + u32 fw_size; + + /* DMA */ + u32 dma_base; + u32 dma_size; + int dma_engine; + + /* DSP */ + u32 id; + void *dsp; +}; + +/* Initialization */ +struct sst_dsp *sst_dsp_new(struct device *dev, + struct sst_dsp_device *sst_dev, struct sst_pdata *pdata); +void sst_dsp_free(struct sst_dsp *sst); + +/* SHIM Read / Write */ +void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value); +u32 sst_dsp_shim_read(struct sst_dsp *sst, u32 offset); +int sst_dsp_shim_update_bits(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value); +void sst_dsp_shim_write64(struct sst_dsp *sst, u32 offset, u64 value); +u64 sst_dsp_shim_read64(struct sst_dsp *sst, u32 offset); +int sst_dsp_shim_update_bits64(struct sst_dsp *sst, u32 offset, + u64 mask, u64 value); + +/* SHIM Read / Write Unlocked for callers already holding sst lock */ +void sst_dsp_shim_write_unlocked(struct sst_dsp *sst, u32 offset, u32 value); +u32 sst_dsp_shim_read_unlocked(struct sst_dsp *sst, u32 offset); +int sst_dsp_shim_update_bits_unlocked(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value); +void sst_dsp_shim_write64_unlocked(struct sst_dsp *sst, u32 offset, u64 value); +u64 sst_dsp_shim_read64_unlocked(struct sst_dsp *sst, u32 offset); +int sst_dsp_shim_update_bits64_unlocked(struct sst_dsp *sst, u32 offset, + u64 mask, u64 value); + +/* DSP reset & boot */ +void sst_dsp_reset(struct sst_dsp *sst); +int sst_dsp_boot(struct sst_dsp *sst); + +/* Msg IO */ +void sst_dsp_ipc_msg_tx(struct sst_dsp *dsp, u32 msg); +u32 sst_dsp_ipc_msg_rx(struct sst_dsp *dsp); + +/* Mailbox management */ +int sst_dsp_mailbox_init(struct sst_dsp *dsp, u32 inbox_offset, + size_t inbox_size, u32 outbox_offset, size_t outbox_size); +void sst_dsp_inbox_write(struct sst_dsp *dsp, void *message, size_t bytes); +void sst_dsp_inbox_read(struct sst_dsp *dsp, void *message, size_t bytes); +void sst_dsp_outbox_write(struct sst_dsp *dsp, void *message, size_t bytes); +void sst_dsp_outbox_read(struct sst_dsp *dsp, void *message, size_t bytes); +void sst_dsp_mailbox_dump(struct sst_dsp *dsp, size_t bytes); + +/* Debug */ +void sst_dsp_dump(struct sst_dsp *sst); + +#endif -- 1.8.3.2

4 13

[alsa-devel] Assert in pcm_params.c
by Alan Horstmann 18 Feb '14

18 Feb '14

Hi, I am trying to assist in a case that is getting this assert:- pcm_params.c:2249: snd1_pcm_hw_params_slave: Assertion `err >= 0' failed. and wondered if anyone could shed some light on what is likely to cause that, and whether it indicates an underlying fault? The system (not mine) runs Linux Mint 13 (kernel 3.2.0) with PulseAudio, and we are debugging weird behaviour during Portaudio initialisation. It is struggling to get consistent access to the Alsa devices, often retrying many times. Any pointers would be appreciated. Thanks Alan

2 1

[alsa-devel] [PATCH] ALSA: hda - add headset mic detect quirks for two Dell laptops
by Hui Wang 18 Feb '14

18 Feb '14

When we plug a 3-ring headset on the Dell machines (Vendor ID: 0x10ec0255, Subsystem ID: 0x10280657; Vendor ID: 0x10ec0255, Subsystem ID: 0x1028065f), 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: Cyrus Lien <cyrus.lien(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 | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2145148..0d80c7f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4309,7 +4309,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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, 0x0657, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0658, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x065f, "Dell", ALC255_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), -- 1.8.1.2

2 1

[alsa-devel] [PATCH 1/4] ASoC: ssm2602: Split SPI and I2C code into different modules
by Lars-Peter Clausen 18 Feb '14

18 Feb '14

There are a few known (minor) problems with having the support code for both I2C and SPI in the same module: * We need to be extra careful to make sure to not build the driver into the kernel if one of the subsystems is build as a module (Currently only SPI can be build as a module). * The module init path error handling is rather ugly. E.g. what should be done if either the SPI or the I2C driver fails to register? Most drivers that implement SPI and I2C in the same module currently fallback to undefined behavior in that case. Splitting the the driver into two modules, one for each bus allows the registration of the other bus driver to continue without problems if one of them fails. This patch splits the AD193X driver into 3 modules. One core module that implements the device logic, but is independent of the bus method used. And one module for SPI and I2C each that registers the drivers and sets up the regmap struct for the bus. While we are at it also cleanup the include section of the ssm2602 driver and remove unneeded includes. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/blackfin/Kconfig | 3 +- sound/soc/codecs/Kconfig | 11 ++- sound/soc/codecs/Makefile | 4 ++ sound/soc/codecs/ssm2602-i2c.c | 57 +++++++++++++++ sound/soc/codecs/ssm2602-spi.c | 41 +++++++++++ sound/soc/codecs/ssm2602.c | 158 ++++++----------------------------------- sound/soc/codecs/ssm2602.h | 14 ++++ 7 files changed, 148 insertions(+), 140 deletions(-) create mode 100644 sound/soc/codecs/ssm2602-i2c.c create mode 100644 sound/soc/codecs/ssm2602-spi.c diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 4544d8e..90262b9 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -14,7 +14,8 @@ config SND_BF5XX_SOC_SSM2602 depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S if !BF60x select SND_BF6XX_SOC_I2S if BF60x - select SND_SOC_SSM2602 + select SND_SOC_SSM2602_SPI if SPI_MASTER + select SND_SOC_SSM2602_I2C if I2C help Say Y if you want to add support for the Analog Devices SSM2602 Audio Codec Add-On Card. diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cc56201..0de36ae 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -70,7 +70,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF select SND_SOC_SSM2518 if I2C - select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI + select SND_SOC_SSM2602_SPI if SPI_MASTER + select SND_SOC_SSM2602_I2C if I2C select SND_SOC_STA32X if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS @@ -361,6 +362,14 @@ config SND_SOC_SSM2518 config SND_SOC_SSM2602 tristate +config SND_SOC_SSM2602_SPI + select SND_SOC_SSM2602 + tristate + +config SND_SOC_SSM2602_I2C + select SND_SOC_SSM2602 + tristate + config SND_SOC_STA32X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d3b536f..55d4973 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -59,6 +59,8 @@ snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-ssm2602-spi-objs := ssm2602-spi.o +snd-soc-ssm2602-i2c-objs := ssm2602-i2c.o snd-soc-sta32x-objs := sta32x.o snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o @@ -190,6 +192,8 @@ obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o +obj-$(CONFIG_SND_SOC_SSM2602_I2C) += snd-soc-ssm2602-i2c.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o diff --git a/sound/soc/codecs/ssm2602-i2c.c b/sound/soc/codecs/ssm2602-i2c.c new file mode 100644 index 0000000..abd63d5 --- /dev/null +++ b/sound/soc/codecs/ssm2602-i2c.c @@ -0,0 +1,57 @@ +/* + * SSM2602/SSM2603/SSM2604 I2C audio driver + * + * Copyright 2014 Analog Devices Inc. + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/regmap.h> + +#include <sound/soc.h> + +#include "ssm2602.h" + +/* + * ssm2602 2 wire address is determined by GPIO5 + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +static int ssm2602_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + return ssm2602_probe(&client->dev, id->driver_data, + devm_regmap_init_i2c(client, &ssm2602_regmap_config)); +} + +static int ssm2602_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ssm2602_i2c_id[] = { + { "ssm2602", SSM2602 }, + { "ssm2603", SSM2602 }, + { "ssm2604", SSM2604 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id); + +static struct i2c_driver ssm2602_i2c_driver = { + .driver = { + .name = "ssm2602", + .owner = THIS_MODULE, + }, + .probe = ssm2602_i2c_probe, + .remove = ssm2602_i2c_remove, + .id_table = ssm2602_i2c_id, +}; +module_i2c_driver(ssm2602_i2c_driver); + +MODULE_DESCRIPTION("ASoC SSM2602/SSM2603/SSM2604 I2C driver"); +MODULE_AUTHOR("Cliff Cai"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2602-spi.c b/sound/soc/codecs/ssm2602-spi.c new file mode 100644 index 0000000..2bf55e2 --- /dev/null +++ b/sound/soc/codecs/ssm2602-spi.c @@ -0,0 +1,41 @@ +/* + * SSM2602 SPI audio driver + * + * Copyright 2014 Analog Devices Inc. + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/spi/spi.h> +#include <linux/regmap.h> + +#include <sound/soc.h> + +#include "ssm2602.h" + +static int ssm2602_spi_probe(struct spi_device *spi) +{ + return ssm2602_probe(&spi->dev, SSM2602, + devm_regmap_init_spi(spi, &ssm2602_regmap_config)); +} + +static int ssm2602_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver ssm2602_spi_driver = { + .driver = { + .name = "ssm2602", + .owner = THIS_MODULE, + }, + .probe = ssm2602_spi_probe, + .remove = ssm2602_spi_remove, +}; +module_spi_driver(ssm2602_spi_driver); + +MODULE_DESCRIPTION("ASoC SSM2602 SPI driver"); +MODULE_AUTHOR("Cliff Cai"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index f444d58..49d28ea 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -27,28 +27,16 @@ */ #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/spi/spi.h> #include <linux/regmap.h> #include <linux/slab.h> -#include <sound/core.h> + #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <sound/initval.h> #include <sound/tlv.h> #include "ssm2602.h" -enum ssm2602_type { - SSM2602, - SSM2604, -}; - /* codec private data */ struct ssm2602_priv { unsigned int sysclk; @@ -529,7 +517,7 @@ static int ssm2602_resume(struct snd_soc_codec *codec) return 0; } -static int ssm2602_probe(struct snd_soc_codec *codec) +static int ssm2602_codec_probe(struct snd_soc_codec *codec) { struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; @@ -554,7 +542,7 @@ static int ssm2602_probe(struct snd_soc_codec *codec) ARRAY_SIZE(ssm2602_routes)); } -static int ssm2604_probe(struct snd_soc_codec *codec) +static int ssm2604_codec_probe(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; @@ -568,7 +556,7 @@ static int ssm2604_probe(struct snd_soc_codec *codec) ARRAY_SIZE(ssm2604_routes)); } -static int ssm260x_probe(struct snd_soc_codec *codec) +static int ssm260x_codec_probe(struct snd_soc_codec *codec) { struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); int ret; @@ -597,10 +585,10 @@ static int ssm260x_probe(struct snd_soc_codec *codec) switch (ssm2602->type) { case SSM2602: - ret = ssm2602_probe(codec); + ret = ssm2602_codec_probe(codec); break; case SSM2604: - ret = ssm2604_probe(codec); + ret = ssm2604_codec_probe(codec); break; } @@ -620,7 +608,7 @@ static int ssm2602_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { - .probe = ssm260x_probe, + .probe = ssm260x_codec_probe, .remove = ssm2602_remove, .suspend = ssm2602_suspend, .resume = ssm2602_resume, @@ -639,7 +627,7 @@ static bool ssm2602_register_volatile(struct device *dev, unsigned int reg) return reg == SSM2602_RESET; } -static const struct regmap_config ssm2602_regmap_config = { +const struct regmap_config ssm2602_regmap_config = { .val_bits = 9, .reg_bits = 7, @@ -650,134 +638,28 @@ static const struct regmap_config ssm2602_regmap_config = { .reg_defaults_raw = ssm2602_reg, .num_reg_defaults_raw = ARRAY_SIZE(ssm2602_reg), }; +EXPORT_SYMBOL_GPL(ssm2602_regmap_config); -#if defined(CONFIG_SPI_MASTER) -static int ssm2602_spi_probe(struct spi_device *spi) +int ssm2602_probe(struct device *dev, enum ssm2602_type type, + struct regmap *regmap) { struct ssm2602_priv *ssm2602; - int ret; - - ssm2602 = devm_kzalloc(&spi->dev, sizeof(struct ssm2602_priv), - GFP_KERNEL); - if (ssm2602 == NULL) - return -ENOMEM; - - spi_set_drvdata(spi, ssm2602); - ssm2602->type = SSM2602; - - ssm2602->regmap = devm_regmap_init_spi(spi, &ssm2602_regmap_config); - if (IS_ERR(ssm2602->regmap)) - return PTR_ERR(ssm2602->regmap); - - ret = snd_soc_register_codec(&spi->dev, - &soc_codec_dev_ssm2602, &ssm2602_dai, 1); - return ret; -} -static int ssm2602_spi_remove(struct spi_device *spi) -{ - snd_soc_unregister_codec(&spi->dev); - return 0; -} - -static struct spi_driver ssm2602_spi_driver = { - .driver = { - .name = "ssm2602", - .owner = THIS_MODULE, - }, - .probe = ssm2602_spi_probe, - .remove = ssm2602_spi_remove, -}; -#endif - -#if IS_ENABLED(CONFIG_I2C) -/* - * ssm2602 2 wire address is determined by GPIO5 - * state during powerup. - * low = 0x1a - * high = 0x1b - */ -static int ssm2602_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) -{ - struct ssm2602_priv *ssm2602; - int ret; + if (IS_ERR(regmap)) + return PTR_ERR(regmap); - ssm2602 = devm_kzalloc(&i2c->dev, sizeof(struct ssm2602_priv), - GFP_KERNEL); + ssm2602 = devm_kzalloc(dev, sizeof(*ssm2602), GFP_KERNEL); if (ssm2602 == NULL) return -ENOMEM; - i2c_set_clientdata(i2c, ssm2602); - ssm2602->type = id->driver_data; - - ssm2602->regmap = devm_regmap_init_i2c(i2c, &ssm2602_regmap_config); - if (IS_ERR(ssm2602->regmap)) - return PTR_ERR(ssm2602->regmap); - - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_ssm2602, &ssm2602_dai, 1); - return ret; -} - -static int ssm2602_i2c_remove(struct i2c_client *client) -{ - snd_soc_unregister_codec(&client->dev); - return 0; -} - -static const struct i2c_device_id ssm2602_i2c_id[] = { - { "ssm2602", SSM2602 }, - { "ssm2603", SSM2602 }, - { "ssm2604", SSM2604 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id); - -/* corgi i2c codec control layer */ -static struct i2c_driver ssm2602_i2c_driver = { - .driver = { - .name = "ssm2602", - .owner = THIS_MODULE, - }, - .probe = ssm2602_i2c_probe, - .remove = ssm2602_i2c_remove, - .id_table = ssm2602_i2c_id, -}; -#endif - - -static int __init ssm2602_modinit(void) -{ - int ret = 0; - -#if defined(CONFIG_SPI_MASTER) - ret = spi_register_driver(&ssm2602_spi_driver); - if (ret) - return ret; -#endif - -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&ssm2602_i2c_driver); - if (ret) - return ret; -#endif - - return ret; -} -module_init(ssm2602_modinit); - -static void __exit ssm2602_exit(void) -{ -#if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&ssm2602_spi_driver); -#endif + dev_set_drvdata(dev, ssm2602); + ssm2602->type = SSM2602; + ssm2602->regmap = regmap; -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&ssm2602_i2c_driver); -#endif + return snd_soc_register_codec(dev, &soc_codec_dev_ssm2602, + &ssm2602_dai, 1); } -module_exit(ssm2602_exit); +EXPORT_SYMBOL_GPL(ssm2602_probe); MODULE_DESCRIPTION("ASoC SSM2602/SSM2603/SSM2604 driver"); MODULE_AUTHOR("Cliff Cai"); diff --git a/sound/soc/codecs/ssm2602.h b/sound/soc/codecs/ssm2602.h index fbd07d7..7475388 100644 --- a/sound/soc/codecs/ssm2602.h +++ b/sound/soc/codecs/ssm2602.h @@ -28,6 +28,20 @@ #ifndef _SSM2602_H #define _SSM2602_H +#include <linux/regmap.h> + +struct device; + +enum ssm2602_type { + SSM2602, + SSM2604, +}; + +extern const struct regmap_config ssm2602_regmap_config; + +int ssm2602_probe(struct device *dev, enum ssm2602_type type, + struct regmap *regmap); + /* SSM2602 Codec Register definitions */ #define SSM2602_LINVOL 0x00 -- 1.8.0

2 7

[alsa-devel] [PATCH] ASoC: txx9aclc_ac97: Fix kernel crash on probe
by Alexander Shiyan 18 Feb '14

18 Feb '14

This patch fixes a crash caused by commit af5b2d7519 (ASoC: txx9aclc_ac97: Convert to devm_ioremap_resource()). This is an attempt to assign "drvdata->base" while memory for "drvdata" is not already allocated. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/txx9/txx9aclc-ac97.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index e0305a1..9edd68d 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -183,14 +183,16 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) irq = platform_get_irq(pdev, 0); if (irq < 0) return irq; + + drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); + if (!drvdata) + return -ENOMEM; + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); drvdata->base = devm_ioremap_resource(&pdev->dev, r); if (IS_ERR(drvdata->base)) return PTR_ERR(drvdata->base); - drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); - if (!drvdata) - return -ENOMEM; platform_set_drvdata(pdev, drvdata); drvdata->physbase = r->start; if (sizeof(drvdata->physbase) > sizeof(r->start) && -- 1.8.3.2

2 4

[alsa-devel] [PATCH] ASoC: pass snd_soc_jack_gpio to jack_status_check callback
by xiangxiao 18 Feb '14

18 Feb '14

so the implementation could get the context data Change-Id: I5542613ccf9881deb672e71d0f0c4f603ad761fd Signed-off-by: xiangxiao <xiaoxiang(a)xiaomi.com> --- include/sound/soc.h | 2 +- sound/soc/soc-jack.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 21038e0..e287462 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -613,7 +613,7 @@ struct snd_soc_jack_gpio { struct snd_soc_jack *jack; struct delayed_work work; - int (*jack_status_check)(void); + int (*jack_status_check)(struct snd_soc_jack_gpio *gpio); }; struct snd_soc_jack { diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index b12fce6..6028af5 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -250,7 +250,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) report = 0; if (gpio->jack_status_check) - report = gpio->jack_status_check(); + report = gpio->jack_status_check(gpio); snd_soc_jack_report(jack, report, gpio->report); } -- 1.8.1.2

2 1

[alsa-devel] [PATCH] tinycompress: compress: use int for samples in compress_get_tstamp
by Vinod Koul 17 Feb '14

17 Feb '14

in 64bit updates we made the sample rate as 32bit explicitly, so update the library API as well Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- compress.c | 2 +- include/tinycompress/tinycompress.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/compress.c b/compress.c index d1a2283..15dfdb7 100644 --- a/compress.c +++ b/compress.c @@ -333,7 +333,7 @@ int compress_get_hpointer(struct compress *compress, } int compress_get_tstamp(struct compress *compress, - unsigned long *samples, unsigned int *sampling_rate) + unsigned int *samples, unsigned int *sampling_rate) { struct snd_compr_tstamp ktstamp; diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index e340cca..03c396f 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -125,7 +125,7 @@ int compress_get_hpointer(struct compress *compress, * @sampling_rate: sampling rate of decoded samples */ int compress_get_tstamp(struct compress *compress, - unsigned long *samples, unsigned int *sampling_rate); + unsigned int *samples, unsigned int *sampling_rate); /* * compress_write: write data to the compress stream -- 1.7.0.4

1 0

[alsa-devel] [TINYCOMPRESS PATCH 1/2 v4] crec: Initial version of a compressed capture utility
by Charles Keepax 17 Feb '14

17 Feb '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 Changes since v3: - O_CREAT flag was missing from open, add this 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..568e4b7 --- /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 | O_CREAT); + 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

2 2

[alsa-devel] [PATCHv2] 0/5 Intel SST Low level drivers.
by Liam Girdwood 17 Feb '14

17 Feb '14

Changes from V1 :- o Fixed firmware_release() usage. o Removed HSW tristate in Kconfig (it's part of the next series) o Block alloc now iterative instead of recursive. o Fixed symbols to be GPL o Locked update_bits() now simplified. o unified type's used in mailbox IO. o Made sst_get_thread_context() and a few other small functions inline. o Moved FW files to be read from intel/ sub directory. One other change I've made that was not part of review comments is that I've removed the DMA FW loader for the time being. I've been made aware a new effort is almost ready that allows registration of ACPI devices that have multiple sub devices in the ACPI space (e.g. ADSP has DMA and SSP devices alongside the DSP). I'll then re-submit the DMA FW loader as a separate patch using the new ACPI infrastructure. In the mean time the driver can safely use memcopy for FW loading. Liam

1 0

[alsa-devel] [PATCH v5] ARM: DTS: imx5* imx6*, use imx51-ssi
by Markus Pargmann 17 Feb '14

17 Feb '14

imx51-ssi and imx21-ssi are different IPs. imx51-ssi supports online reconfiguration and needs this for correct interaction with SDMA. This patch adds imx51-ssi before each imx21-ssi for all imx5/imx6 SoCs. Signed-off-by: Markus Pargmann <mpa(a)pengutronix.de> --- Hi Shawn, This is the patch including the two added compatibles in imx50.dtsi for fsl,imx51-ssi. That is the only change in v5. Regards, Markus arch/arm/boot/dts/imx50.dtsi | 6 +++++- arch/arm/boot/dts/imx53.dtsi | 10 +++++++--- arch/arm/boot/dts/imx6qdl.dtsi | 12 +++++++++--- arch/arm/boot/dts/imx6sl.dtsi | 12 +++++++++--- 4 files changed, 30 insertions(+), 10 deletions(-) diff --git a/arch/arm/boot/dts/imx50.dtsi b/arch/arm/boot/dts/imx50.dtsi index 01c0499..47df15f 100644 --- a/arch/arm/boot/dts/imx50.dtsi +++ b/arch/arm/boot/dts/imx50.dtsi @@ -140,7 +140,9 @@ }; ssi2: ssi@50014000 { - compatible = "fsl,imx50-ssi", "fsl,imx21-ssi"; + compatible = "fsl,imx50-ssi", + "fsl,imx51-ssi", + "fsl,imx21-ssi"; reg = <0x50014000 0x4000>; interrupts = <30>; clocks = <&clks IMX5_CLK_SSI2_IPG_GATE>; @@ -446,6 +448,8 @@ ssi1: ssi@63fcc000 { compatible = "fsl,imx50-ssi", "fsl,imx21-ssi"; + compatible = "fsl,imx50-ssi", "fsl,imx51-ssi", + "fsl,imx21-ssi"; reg = <0x63fcc000 0x4000>; interrupts = <29>; clocks = <&clks IMX5_CLK_SSI1_IPG_GATE>; diff --git a/arch/arm/boot/dts/imx53.dtsi b/arch/arm/boot/dts/imx53.dtsi index 4dbde25..73e103b 100644 --- a/arch/arm/boot/dts/imx53.dtsi +++ b/arch/arm/boot/dts/imx53.dtsi @@ -171,7 +171,9 @@ }; ssi2: ssi@50014000 { - compatible = "fsl,imx53-ssi", "fsl,imx21-ssi"; + compatible = "fsl,imx53-ssi", + "fsl,imx51-ssi", + "fsl,imx21-ssi"; reg = <0x50014000 0x4000>; interrupts = <30>; clocks = <&clks IMX5_CLK_SSI2_IPG_GATE>; @@ -590,7 +592,8 @@ }; ssi1: ssi@63fcc000 { - compatible = "fsl,imx53-ssi", "fsl,imx21-ssi"; + compatible = "fsl,imx53-ssi", "fsl,imx51-ssi", + "fsl,imx21-ssi"; reg = <0x63fcc000 0x4000>; interrupts = <29>; clocks = <&clks IMX5_CLK_SSI1_IPG_GATE>; @@ -617,7 +620,8 @@ }; ssi3: ssi@63fe8000 { - compatible = "fsl,imx53-ssi", "fsl,imx21-ssi"; + compatible = "fsl,imx53-ssi", "fsl,imx51-ssi", + "fsl,imx21-ssi"; reg = <0x63fe8000 0x4000>; interrupts = <96>; clocks = <&clks IMX5_CLK_SSI3_IPG_GATE>; diff --git a/arch/arm/boot/dts/imx6qdl.dtsi b/arch/arm/boot/dts/imx6qdl.dtsi index 8a86502..cf7956e 100644 --- a/arch/arm/boot/dts/imx6qdl.dtsi +++ b/arch/arm/boot/dts/imx6qdl.dtsi @@ -247,7 +247,9 @@ }; ssi1: ssi@02028000 { - compatible = "fsl,imx6q-ssi","fsl,imx21-ssi"; + compatible = "fsl,imx6q-ssi", + "fsl,imx51-ssi", + "fsl,imx21-ssi"; reg = <0x02028000 0x4000>; interrupts = <0 46 IRQ_TYPE_LEVEL_HIGH>; clocks = <&clks 178>; @@ -260,7 +262,9 @@ }; ssi2: ssi@0202c000 { - compatible = "fsl,imx6q-ssi","fsl,imx21-ssi"; + compatible = "fsl,imx6q-ssi", + "fsl,imx51-ssi", + "fsl,imx21-ssi"; reg = <0x0202c000 0x4000>; interrupts = <0 47 IRQ_TYPE_LEVEL_HIGH>; clocks = <&clks 179>; @@ -273,7 +277,9 @@ }; ssi3: ssi@02030000 { - compatible = "fsl,imx6q-ssi","fsl,imx21-ssi"; + compatible = "fsl,imx6q-ssi", + "fsl,imx51-ssi", + "fsl,imx21-ssi"; reg = <0x02030000 0x4000>; interrupts = <0 48 IRQ_TYPE_LEVEL_HIGH>; clocks = <&clks 180>; diff --git a/arch/arm/boot/dts/imx6sl.dtsi b/arch/arm/boot/dts/imx6sl.dtsi index a449c4f..95bb37b 100644 --- a/arch/arm/boot/dts/imx6sl.dtsi +++ b/arch/arm/boot/dts/imx6sl.dtsi @@ -226,7 +226,9 @@ }; ssi1: ssi@02028000 { - compatible = "fsl,imx6sl-ssi","fsl,imx21-ssi"; + compatible = "fsl,imx6sl-ssi", + "fsl,imx51-ssi", + "fsl,imx21-ssi"; reg = <0x02028000 0x4000>; interrupts = <0 46 IRQ_TYPE_LEVEL_HIGH>; clocks = <&clks IMX6SL_CLK_SSI1>; @@ -238,7 +240,9 @@ }; ssi2: ssi@0202c000 { - compatible = "fsl,imx6sl-ssi","fsl,imx21-ssi"; + compatible = "fsl,imx6sl-ssi", + "fsl,imx51-ssi", + "fsl,imx21-ssi"; reg = <0x0202c000 0x4000>; interrupts = <0 47 IRQ_TYPE_LEVEL_HIGH>; clocks = <&clks IMX6SL_CLK_SSI2>; @@ -250,7 +254,9 @@ }; ssi3: ssi@02030000 { - compatible = "fsl,imx6sl-ssi","fsl,imx21-ssi"; + compatible = "fsl,imx6sl-ssi", + "fsl,imx51-ssi", + "fsl,imx21-ssi"; reg = <0x02030000 0x4000>; interrupts = <0 48 IRQ_TYPE_LEVEL_HIGH>; clocks = <&clks IMX6SL_CLK_SSI3>; -- 1.8.5.2

2 5

[alsa-devel] [PATCH v3] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 17 Feb '14

17 Feb '14

This patch adds initial support for the Behringer BCD2000 USB DJ controller. At the moment, only the MIDI part of the device is working, i.e. knobs, buttons and LEDs. I also plan to add support for the audio part, but I assume that this will require more effort than the rather simple MIDI interface. Progress can be tracked at https://github.com/anyc/snd-usb-bcd2000. Changes since v2: - applied more changes from Daniel Mack and Clemens Ladisch Changes since v1: - fixed the various code style issues, thanks to Daniel Mack and checkpatch.pl. Signed-off-by: Mario Kicherer <dev(a)kicherer.org> --- sound/usb/Kconfig | 13 ++ sound/usb/Makefile | 2 +- sound/usb/bcd2000/Makefile | 3 + sound/usb/bcd2000/bcd2000.c | 484 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 501 insertions(+), 1 deletion(-) create mode 100644 sound/usb/bcd2000/Makefile create mode 100644 sound/usb/bcd2000/bcd2000.c diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index de9408b..145653a 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -146,5 +146,18 @@ config SND_USB_HIFACE To compile this driver as a module, choose M here: the module will be called snd-usb-hiface. +config SND_BCD2000 + tristate "Behringer BCD2000 MIDI driver" + select SND_RAWMIDI + help + Say Y here to include MIDI support for the Behringer BCD2000 DJ + controller. + + Audio support is still work-in-progress at + https://github.com/anyc/snd-usb-bcd2000 + + To compile this driver as a module, choose M here: the module + will be called snd-bcd2000. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index abe668f..2b92f0d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ bcd2000/ diff --git a/sound/usb/bcd2000/Makefile b/sound/usb/bcd2000/Makefile new file mode 100644 index 0000000..f09ccc0 --- /dev/null +++ b/sound/usb/bcd2000/Makefile @@ -0,0 +1,3 @@ +snd-bcd2000-y := bcd2000.o + +obj-$(CONFIG_SND_BCD2000) += snd-bcd2000.o \ No newline at end of file diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c new file mode 100644 index 0000000..7a24611 --- /dev/null +++ b/sound/usb/bcd2000/bcd2000.c @@ -0,0 +1,484 @@ +/* + * Behringer BCD2000 driver + * + * Copyright (C) 2014 Mario Kicherer (dev(a)kicherer.org) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/usb.h> +#include <linux/usb/audio.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/rawmidi.h> + +#define DEVICE_NAME "Behringer BCD2000" +#define DEVICE_SHORTNAME "bcd2000" + +#define PREFIX DEVICE_SHORTNAME ": " +#define BUFSIZE 64 + +static struct usb_device_id id_table[] = { + { USB_DEVICE(0x1397, 0x00bd) }, + { }, +}; + +static unsigned char device_cmd_prefix[] = {0x03, 0x00}; + +static unsigned char bcd2000_init_sequence[] = { + 0x07, 0x00, 0x00, 0x00, 0x78, 0x48, 0x1c, 0x81, + 0xc4, 0x00, 0x00, 0x00, 0x5e, 0x53, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff, 0x6c, 0xf3, 0x90, 0xff, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, + 0x18, 0xfa, 0x11, 0xff, 0x14, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0xf2, 0x34, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff}; + +struct bcd2000 { + struct usb_device *dev; + struct snd_card *card; + struct usb_interface *intf; + int card_index; + + int midi_out_active; + struct snd_rawmidi *rmidi; + struct snd_rawmidi_substream *midi_receive_substream; + struct snd_rawmidi_substream *midi_out_substream; + + unsigned char midi_in_buf[BUFSIZE]; + unsigned char midi_out_buf[BUFSIZE]; + + struct urb *midi_out_urb; + struct urb *midi_in_urb; + + struct usb_anchor anchor; +}; + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +static DEFINE_MUTEX(devices_mutex); +static unsigned int devices_used; +static struct usb_driver bcd2000_driver; + +#ifdef CONFIG_SND_DEBUG +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) +{ + print_hex_dump(KERN_DEBUG, prefix, + DUMP_PREFIX_NONE, 16, 1, + buf, len, false); +} +#else +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) {} +#endif + +static int bcd2000_midi_input_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_input_close(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +/* register midi substream */ +static void bcd2000_midi_input_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + bcd2k->midi_receive_substream = up ? substream : NULL; +} + +static void bcd2000_midi_handle_input(struct bcd2000 *bcd2k, + const unsigned char *buf, int len) +{ + unsigned char length, tocopy; + + if (!bcd2k->midi_receive_substream) + return; + + bcd2000_dump_buffer(PREFIX "received from device: ", buf, len); + + if (len < 2) + return; + + /* + * Packet structure: mm nn oo (pp) + * mm: payload length + * nn: MIDI command or note + * oo: note or velocity + * pp: velocity + */ + + length = buf[0]; + + /* ignore packets without payload */ + if (length == 0) + return; + + tocopy = min(length, (unsigned char) (len-1)); + + bcd2000_dump_buffer(PREFIX "sending to userspace: ", + &buf[1], tocopy); + + snd_rawmidi_receive(bcd2k->midi_receive_substream, + &buf[1], tocopy); +} + +static void bcd2000_midi_send(struct bcd2000 *bcd2k, + struct snd_rawmidi_substream *substream) +{ + int len, ret; + + BUILD_BUG_ON(sizeof(device_cmd_prefix) >= BUFSIZE); + /* copy the "set LED" command bytes */ + memcpy(bcd2k->midi_out_buf, device_cmd_prefix, + sizeof(device_cmd_prefix)); + + /* + * get MIDI packet and leave space for command prefix + * and payload length + */ + len = snd_rawmidi_transmit(substream, + bcd2k->midi_out_buf + 3, BUFSIZE - 3); + + if (len < 0) + snd_printk("%s: snd_rawmidi_transmit error %d\n", + __func__, len); + + if (len <= 0) + return; + + /* set payload length */ + bcd2k->midi_out_buf[2] = len; + bcd2k->midi_out_urb->transfer_buffer_length = BUFSIZE; + + bcd2000_dump_buffer(PREFIX "sending to device: ", + bcd2k->midi_out_buf, len+3); + + /* send packet to the BCD2000 */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s (%p): usb_submit_urb() failed, ret=%d, len=%d\n", + __func__, substream, ret, len); + else + bcd2k->midi_out_active = 1; +} + +static int bcd2000_midi_output_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_output_close(struct snd_rawmidi_substream *substream) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (bcd2k->midi_out_active) { + usb_kill_urb(bcd2k->midi_out_urb); + bcd2k->midi_out_active = 0; + } + + return 0; +} + +/* register midi substream */ +static void bcd2000_midi_output_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (up) { + bcd2k->midi_out_substream = substream; + /* check if there is data userspace wants to send */ + if (!bcd2k->midi_out_active) + bcd2000_midi_send(bcd2k, substream); + } else { + bcd2k->midi_out_substream = NULL; + } +} + +static void bcd2000_output_complete(struct urb *urb) +{ + struct bcd2000 *bcd2k = urb->context; + struct snd_rawmidi_substream *midi_out_substream; + + bcd2k->midi_out_active = 0; + + if (urb->status != 0) + dev_err(&urb->dev->dev, + PREFIX "output urb->status: %d\n", urb->status); + + midi_out_substream = ACCESS_ONCE(bcd2k->midi_out_substream); + if (!midi_out_substream) + return; + + /* check if there is more data userspace wants to send */ + bcd2000_midi_send(bcd2k, midi_out_substream); +} + +static void bcd2000_input_complete(struct urb *urb) +{ + int ret; + struct device *dev = &urb->dev->dev; + struct bcd2000 *bcd2k = urb->context; + + if (urb->status) { + dev_warn(dev, PREFIX "input urb->status: %i\n", urb->status); + return; + } + + if (!bcd2k) + return; + + if (urb->actual_length > 0) + bcd2000_midi_handle_input(bcd2k, urb->transfer_buffer, + urb->actual_length); + + /* acknowledge received packet */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(dev, "unable to submit urb. OOM!?\n"); +} + +static struct snd_rawmidi_ops bcd2000_midi_output = { + .open = bcd2000_midi_output_open, + .close = bcd2000_midi_output_close, + .trigger = bcd2000_midi_output_trigger, +}; + +static struct snd_rawmidi_ops bcd2000_midi_input = { + .open = bcd2000_midi_input_open, + .close = bcd2000_midi_input_close, + .trigger = bcd2000_midi_input_trigger, +}; + +static void bcd2000_init_device(struct bcd2000 *bcd2k) +{ + int ret; + + init_usb_anchor(&bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_out_urb, &bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_in_urb, &bcd2k->anchor); + + /* copy init sequence into buffer */ + memcpy(bcd2k->midi_out_buf, bcd2000_init_sequence, 52); + bcd2k->midi_out_urb->transfer_buffer_length = 52; + + /* submit sequence */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() out failed, ret=%d: ", + __func__, ret); + else + bcd2k->midi_out_active = 1; + + /* send empty packet to enable button and controller events */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() in failed, ret=%d: ", + __func__, ret); + + /* ensure initialization is finished */ + usb_wait_anchor_empty_timeout(&bcd2k->anchor, 1000); +} + +static int bcd2000_init_midi(struct bcd2000 *bcd2k) +{ + int ret; + struct snd_rawmidi *rmidi; + + ret = snd_rawmidi_new(bcd2k->card, bcd2k->card->shortname, 0, + 1, /* output */ + 1, /* input */ + &rmidi); + + if (ret < 0) + return; + + strlcpy(rmidi->name, bcd2k->card->shortname, sizeof(rmidi->name)); + + rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; + rmidi->private_data = bcd2k; + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &bcd2000_midi_output); + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &bcd2000_midi_input); + + bcd2k->rmidi = rmidi; + + bcd2k->midi_in_urb = usb_alloc_urb(0, GFP_KERNEL); + bcd2k->midi_out_urb = usb_alloc_urb(0, GFP_KERNEL); + + if (!bcd2k->midi_in_urb || !bcd2k->midi_out_urb) { + snd_printk(KERN_ERR PREFIX "usb_alloc_urb failed\n"); + return -ENOMEM; + } + + usb_fill_int_urb(bcd2k->midi_in_urb, bcd2k->dev, + usb_rcvintpipe(bcd2k->dev, 0x81), + bcd2k->midi_in_buf, BUFSIZE, + bcd2000_input_complete, bcd2k, 1); + + usb_fill_int_urb(bcd2k->midi_out_urb, bcd2k->dev, + usb_sndintpipe(bcd2k->dev, 0x1), + bcd2k->midi_out_buf, BUFSIZE, + bcd2000_output_complete, bcd2k, 1); + + bcd2000_init_device(bcd2k); + + return 0; +} + +static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k, + struct usb_interface *interface) +{ + /* usb_kill_urb not necessary, urb is aborted automatically */ + + usb_free_urb(bcd2k->midi_out_urb); + usb_free_urb(bcd2k->midi_in_urb); + + if (bcd2k->intf) { + usb_set_intfdata(bcd2k->intf, NULL); + bcd2k->intf = NULL; + } +} + +static int bcd2000_probe(struct usb_interface *interface, + const struct usb_device_id *usb_id) +{ + struct snd_card *card; + struct bcd2000 *bcd2k; + unsigned int card_index; + char usb_path[32]; + int err; + + mutex_lock(&devices_mutex); + + for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) + if (enable[card_index] && !(devices_used & (1 << card_index))) + break; + + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + + err = snd_card_create(index[card_index], id[card_index], THIS_MODULE, + sizeof(*bcd2k), &card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return err; + } + + bcd2k = card->private_data; + bcd2k->dev = interface_to_usbdev(interface); + bcd2k->card = card; + bcd2k->card_index = card_index; + bcd2k->intf = interface; + + snd_card_set_dev(card, &interface->dev); + + strcpy(card->driver, DEVICE_NAME); + strcpy(card->shortname, DEVICE_SHORTNAME); + usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); + snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), + DEVICE_NAME ", at %s", + usb_path); + + dev_info(&bcd2k->dev->dev, PREFIX "%s", bcd2k->card->longname); + + err = bcd2000_init_midi(bcd2k); + if (err < 0) + goto probe_error; + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + usb_set_intfdata(interface, bcd2k); + devices_used |= 1 << card_index; + + mutex_unlock(&devices_mutex); + return 0; + +probe_error: + dev_info(&bcd2k->dev->dev, PREFIX "error during probing"); + bcd2000_free_usb_related_resources(bcd2k, interface); + snd_card_free(card); + mutex_unlock(&devices_mutex); + return err; +} + +static void bcd2000_disconnect(struct usb_interface *interface) +{ + struct bcd2000 *bcd2k = usb_get_intfdata(interface); + + if (!bcd2k) + return; + + mutex_lock(&devices_mutex); + + /* make sure that userspace cannot create new requests */ + snd_card_disconnect(bcd2k->card); + + bcd2000_free_usb_related_resources(bcd2k, interface); + + devices_used &= ~(1 << bcd2k->card_index); + + snd_card_free_when_closed(bcd2k->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_driver bcd2000_driver = { + .name = DEVICE_SHORTNAME, + .probe = bcd2000_probe, + .disconnect = bcd2000_disconnect, + .id_table = id_table, +}; + +static int __init bcd2000_init(void) +{ + int retval = 0; + + retval = usb_register(&bcd2000_driver); + if (retval) + snd_printk(KERN_INFO "usb_register failed. Error: %d", retval); + return retval; +} + +static void __exit bcd2000_exit(void) +{ + usb_deregister(&bcd2000_driver); +} + +module_init(bcd2000_init); +module_exit(bcd2000_exit); + +MODULE_DEVICE_TABLE(usb, id_table); +MODULE_AUTHOR("Mario Kicherer, dev(a)kicherer.org"); +MODULE_DESCRIPTION("Behringer BCD2000 driver"); +MODULE_LICENSE("GPL"); -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ALSA: usx2y: Don't peep the card internal object
by Takashi Iwai 17 Feb '14

17 Feb '14

Avoid traversing the device object list of the card instance just for checking the PCM streams. The driver's private object already contains the array of substream pointers, so it can be simply looked through. The card internal may be restructured in future, thus better not to rely on it. Also, this fixes the possible deadlocks in PCM mutex. Instead of taking multiple PCM mutexes, just take the common mutex in all places. Along with it, rename prepare_mutex as pcm_mutex. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/usx2y/usbusx2y.c | 2 +- sound/usb/usx2y/usbusx2y.h | 2 +- sound/usb/usx2y/usbusx2yaudio.c | 60 ++++++++++++++++++-------------- sound/usb/usx2y/usx2yhwdeppcm.c | 76 ++++++++++++++--------------------------- 4 files changed, 61 insertions(+), 79 deletions(-) diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index e38c87302b15..91e0e2a4808c 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -353,7 +353,7 @@ static int usX2Y_create_card(struct usb_device *device, card->private_free = snd_usX2Y_card_private_free; usX2Y(card)->dev = device; init_waitqueue_head(&usX2Y(card)->prepare_wait_queue); - mutex_init(&usX2Y(card)->prepare_mutex); + mutex_init(&usX2Y(card)->pcm_mutex); INIT_LIST_HEAD(&usX2Y(card)->midi_list); strcpy(card->driver, "USB "NAME_ALLCAPS""); sprintf(card->shortname, "TASCAM "NAME_ALLCAPS""); diff --git a/sound/usb/usx2y/usbusx2y.h b/sound/usb/usx2y/usbusx2y.h index e43c0a86441a..6ae6b0806938 100644 --- a/sound/usb/usx2y/usbusx2y.h +++ b/sound/usb/usx2y/usbusx2y.h @@ -36,7 +36,7 @@ struct usX2Ydev { unsigned int rate, format; int chip_status; - struct mutex prepare_mutex; + struct mutex pcm_mutex; struct us428ctls_sharedmem *us428ctls_sharedmem; int wait_iso_frame; wait_queue_head_t us428ctls_wait_queue_head; diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 6234a51625b1..a63330dd1407 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -752,36 +752,44 @@ static int snd_usX2Y_pcm_hw_params(struct snd_pcm_substream *substream, unsigned int rate = params_rate(hw_params); snd_pcm_format_t format = params_format(hw_params); struct snd_card *card = substream->pstr->pcm->card; - struct list_head *list; + struct usX2Ydev *dev = usX2Y(card); + int i; + mutex_lock(&usX2Y(card)->pcm_mutex); snd_printdd("snd_usX2Y_hw_params(%p, %p)\n", substream, hw_params); - // all pcm substreams off one usX2Y have to operate at the same rate & format - list_for_each(list, &card->devices) { - struct snd_device *dev; - struct snd_pcm *pcm; - int s; - dev = snd_device(list); - if (dev->type != SNDRV_DEV_PCM) + /* all pcm substreams off one usX2Y have to operate at the same + * rate & format + */ + for (i = 0; i < dev->pcm_devs * 2; i++) { + struct snd_usX2Y_substream *subs = dev->subs[i]; + struct snd_pcm_substream *test_substream; + + if (!subs) + continue; + test_substream = subs->pcm_substream; + if (!test_substream || test_substream == substream || + !test_substream->runtime) continue; - pcm = dev->device_data; - for (s = 0; s < 2; ++s) { - struct snd_pcm_substream *test_substream; - test_substream = pcm->streams[s].substream; - if (test_substream && test_substream != substream && - test_substream->runtime && - ((test_substream->runtime->format && - test_substream->runtime->format != format) || - (test_substream->runtime->rate && - test_substream->runtime->rate != rate))) - return -EINVAL; + if ((test_substream->runtime->format && + test_substream->runtime->format != format) || + (test_substream->runtime->rate && + test_substream->runtime->rate != rate)) { + err = -EINVAL; + goto error; } } - if (0 > (err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)))) { + + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (err < 0) { snd_printk(KERN_ERR "snd_pcm_lib_malloc_pages(%p, %i) returned %i\n", substream, params_buffer_bytes(hw_params), err); - return err; + goto error; } - return 0; + + error: + mutex_unlock(&usX2Y(card)->pcm_mutex); + return err; } /* @@ -791,7 +799,7 @@ static int snd_usX2Y_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_usX2Y_substream *subs = runtime->private_data; - mutex_lock(&subs->usX2Y->prepare_mutex); + mutex_lock(&subs->usX2Y->pcm_mutex); snd_printdd("snd_usX2Y_hw_free(%p)\n", substream); if (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) { @@ -812,7 +820,7 @@ static int snd_usX2Y_pcm_hw_free(struct snd_pcm_substream *substream) usX2Y_urbs_release(subs); } } - mutex_unlock(&subs->usX2Y->prepare_mutex); + mutex_unlock(&subs->usX2Y->pcm_mutex); return snd_pcm_lib_free_pages(substream); } /* @@ -829,7 +837,7 @@ static int snd_usX2Y_pcm_prepare(struct snd_pcm_substream *substream) int err = 0; snd_printdd("snd_usX2Y_pcm_prepare(%p)\n", substream); - mutex_lock(&usX2Y->prepare_mutex); + mutex_lock(&usX2Y->pcm_mutex); usX2Y_subs_prepare(subs); // Start hardware streams // SyncStream first.... @@ -849,7 +857,7 @@ static int snd_usX2Y_pcm_prepare(struct snd_pcm_substream *substream) err = usX2Y_urbs_start(subs); up_prepare_mutex: - mutex_unlock(&usX2Y->prepare_mutex); + mutex_unlock(&usX2Y->pcm_mutex); return err; } diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 814d0e887c62..90766a92e7fd 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -358,7 +358,7 @@ static int snd_usX2Y_usbpcm_hw_free(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_usX2Y_substream *subs = runtime->private_data, *cap_subs2 = subs->usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE + 2]; - mutex_lock(&subs->usX2Y->prepare_mutex); + mutex_lock(&subs->usX2Y->pcm_mutex); snd_printdd("snd_usX2Y_usbpcm_hw_free(%p)\n", substream); if (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) { @@ -387,7 +387,7 @@ static int snd_usX2Y_usbpcm_hw_free(struct snd_pcm_substream *substream) usX2Y_usbpcm_urbs_release(cap_subs2); } } - mutex_unlock(&subs->usX2Y->prepare_mutex); + mutex_unlock(&subs->usX2Y->pcm_mutex); return snd_pcm_lib_free_pages(substream); } @@ -493,7 +493,7 @@ static int snd_usX2Y_usbpcm_prepare(struct snd_pcm_substream *substream) memset(usX2Y->hwdep_pcm_shm, 0, sizeof(struct snd_usX2Y_hwdep_pcm_shm)); } - mutex_lock(&usX2Y->prepare_mutex); + mutex_lock(&usX2Y->pcm_mutex); usX2Y_subs_prepare(subs); // Start hardware streams // SyncStream first.... @@ -534,7 +534,7 @@ static int snd_usX2Y_usbpcm_prepare(struct snd_pcm_substream *substream) usX2Y->hwdep_pcm_shm->capture_iso_start = -1; up_prepare_mutex: - mutex_unlock(&usX2Y->prepare_mutex); + mutex_unlock(&usX2Y->pcm_mutex); return err; } @@ -600,59 +600,30 @@ static struct snd_pcm_ops snd_usX2Y_usbpcm_ops = }; -static int usX2Y_pcms_lock_check(struct snd_card *card) +static int usX2Y_pcms_busy_check(struct snd_card *card) { - struct list_head *list; - struct snd_device *dev; - struct snd_pcm *pcm; - int err = 0; - list_for_each(list, &card->devices) { - dev = snd_device(list); - if (dev->type != SNDRV_DEV_PCM) - continue; - pcm = dev->device_data; - mutex_lock(&pcm->open_mutex); - } - list_for_each(list, &card->devices) { - int s; - dev = snd_device(list); - if (dev->type != SNDRV_DEV_PCM) - continue; - pcm = dev->device_data; - for (s = 0; s < 2; ++s) { - struct snd_pcm_substream *substream; - substream = pcm->streams[s].substream; - if (substream && SUBSTREAM_BUSY(substream)) - err = -EBUSY; - } - } - return err; -} - + struct usX2Ydev *dev = usX2Y(card); + int i; -static void usX2Y_pcms_unlock(struct snd_card *card) -{ - struct list_head *list; - struct snd_device *dev; - struct snd_pcm *pcm; - list_for_each(list, &card->devices) { - dev = snd_device(list); - if (dev->type != SNDRV_DEV_PCM) - continue; - pcm = dev->device_data; - mutex_unlock(&pcm->open_mutex); + for (i = 0; i < dev->pcm_devs * 2; i++) { + struct snd_usX2Y_substream *subs = dev->subs[i]; + if (subs && subs->pcm_substream && + SUBSTREAM_BUSY(subs->pcm_substream)) + return -EBUSY; } + return 0; } - static int snd_usX2Y_hwdep_pcm_open(struct snd_hwdep *hw, struct file *file) { - // we need to be the first struct snd_card *card = hw->card; - int err = usX2Y_pcms_lock_check(card); - if (0 == err) + int err; + + mutex_lock(&usX2Y(card)->pcm_mutex); + err = usX2Y_pcms_busy_check(card); + if (!err) usX2Y(card)->chip_status |= USX2Y_STAT_CHIP_MMAP_PCM_URBS; - usX2Y_pcms_unlock(card); + mutex_unlock(&usX2Y(card)->pcm_mutex); return err; } @@ -660,10 +631,13 @@ static int snd_usX2Y_hwdep_pcm_open(struct snd_hwdep *hw, struct file *file) static int snd_usX2Y_hwdep_pcm_release(struct snd_hwdep *hw, struct file *file) { struct snd_card *card = hw->card; - int err = usX2Y_pcms_lock_check(card); - if (0 == err) + int err; + + mutex_lock(&usX2Y(card)->pcm_mutex); + err = usX2Y_pcms_busy_check(card); + if (!err) usX2Y(hw->card)->chip_status &= ~USX2Y_STAT_CHIP_MMAP_PCM_URBS; - usX2Y_pcms_unlock(card); + mutex_unlock(&usX2Y(card)->pcm_mutex); return err; } -- 1.8.5.2

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: work around KEF X300A firmware bug
by Clemens Ladisch 17 Feb '14

17 Feb '14

When the driver tries to access Function Unit 10, the KEF X300A speakers' firmware apparently locks up, making even PCM streaming impossible. Work around this by ignoring this FU. Cc: <stable(a)vger.kernel.org> Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/mixer_maps.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 32af6b7..d1d72ff 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -328,6 +328,11 @@ static struct usbmix_name_map gamecom780_map[] = { {} }; +static const struct usbmix_name_map kef_x300a_map[] = { + { 10, NULL }, /* firmware locks up (?) when we try to access this FU */ + { 0 } +}; + /* * Control map entries */ @@ -419,6 +424,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x200c, 0x1018), .map = ebox44_map, }, + { + .id = USB_ID(0x27ac, 0x1000), + .map = kef_x300a_map, + }, { 0 } /* terminator */ };

2 1

[alsa-devel] [PATCH] ASoC: uda134x: Use core for applying symmetry constraitns
by Lars-Peter Clausen 17 Feb '14

17 Feb '14

Let the core take care of applying sample rate and sample bits constraints instead of open-coding this in the driver. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Only compile tested --- sound/soc/codecs/uda134x.c | 52 +++------------------------------------------- 1 file changed, 3 insertions(+), 49 deletions(-) diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index c94d4c1..43b88ee 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -34,9 +34,6 @@ struct uda134x_priv { int sysclk; int dai_fmt; - - struct snd_pcm_substream *master_substream; - struct snd_pcm_substream *slave_substream; }; /* In-data addresses are hard-coded into the reg-cache values */ @@ -156,49 +153,6 @@ static int uda134x_mute(struct snd_soc_dai *dai, int mute) return 0; } -static int uda134x_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); - struct snd_pcm_runtime *master_runtime; - - if (uda134x->master_substream) { - master_runtime = uda134x->master_substream->runtime; - - pr_debug("%s constraining to %d bits at %d\n", __func__, - master_runtime->sample_bits, - master_runtime->rate); - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - master_runtime->rate, - master_runtime->rate); - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - master_runtime->sample_bits, - master_runtime->sample_bits); - - uda134x->slave_substream = substream; - } else - uda134x->master_substream = substream; - - return 0; -} - -static void uda134x_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); - - if (uda134x->master_substream == substream) - uda134x->master_substream = uda134x->slave_substream; - - uda134x->slave_substream = NULL; -} - static int uda134x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -208,7 +162,7 @@ static int uda134x_hw_params(struct snd_pcm_substream *substream, struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); u8 hw_params; - if (substream == uda134x->slave_substream) { + if (dai->active) { pr_debug("%s ignoring hw_params for slave substream\n", __func__); return 0; @@ -447,8 +401,6 @@ static const struct snd_soc_dapm_route uda134x_dapm_routes[] = { }; static const struct snd_soc_dai_ops uda134x_dai_ops = { - .startup = uda134x_startup, - .shutdown = uda134x_shutdown, .hw_params = uda134x_hw_params, .digital_mute = uda134x_mute, .set_sysclk = uda134x_set_dai_sysclk, @@ -475,6 +427,8 @@ static struct snd_soc_dai_driver uda134x_dai = { }, /* pcm operations */ .ops = &uda134x_dai_ops, + .symmetric_rates = 1, + .symmetric_samplebits = 1, }; static int uda134x_soc_probe(struct snd_soc_codec *codec) -- 1.8.0

1 1

[alsa-devel] [PATCH 0/1] dmix: Add support for S24_3BE format.
by Vincent Vinel 17 Feb '14

17 Feb '14

Hello, This patch adds swapped versions for generic_(re)mix_areas_24() functions. I am unsure if the same code could handle S24_BE properly, so this format is left out and only S24_3BE is supported. Thanks ! Vincent Vinel (1): dmix: Add support for S24_3BE format. include/pcm.h | 4 +++ src/pcm/pcm_direct.c | 3 +- src/pcm/pcm_dmix.c | 2 ++ src/pcm/pcm_dmix_generic.c | 77 +++++++++++++++++++++++++++++++++++++++++++--- 4 files changed, 80 insertions(+), 6 deletions(-) -- 1.8.5.5

3 4

[alsa-devel] Intel HDA + HDMI - snd_pcm_drain buggy behaviour.
by Sander Jansen 17 Feb '14

17 Feb '14

All, When I play regular PCM audio through the HDMI that's connected to my amplifier I noticed buggy behaviour when using the snd_pcm_drain() call. >From my understanding of ALSA, during playback snd_pcm_drain() normally blocks until the playback buffer is empty. Here's what I'm experiencing: a. If the PCM handle is in BLOCKING mode, the snd_pcm_drain() function will block indefinitely (I've waited 30 seconds, before I gave up). b. In NON-BLOCKING mode, the first call to snd_pcm_drain results in a " Resource temporarily unavailable", subsequent call works but the pcm state has changed to DRAINING and takes about 1 sec to change to SETUP. Even though the audio has less than 0.5 seconds of actual playback left. So to make it work, I changed my single snd_pcm_drain() call into the following pseudo code: while(state==RUNNING && snd_pcm_drain()<0) { state=snd_pcm_state() } while(state==DRAINING && sleep(100)){ state=snd_pcm_state() } Even though this seem to "sort of work", another problem with this approach I've noticed, is that part of the beginning of the playback buffer may be replayed at the end of the drain: 1. start playback at beginning of file 2. seek to close to the end of the file (snd_pcm_drop() gets called) 3. end of file reached, drain gets called, hear the end of the song + part of the beginning again? So I'm wondering whether this may be related to the drain issue I've noticed. I've attached the alsa-info. Thanks, Sander

2 3

[alsa-devel] Stereo audio not coming through with a TI codec
by Barnaby Jones 17 Feb '14

17 Feb '14

Hey all, I'm working on a project that has a TI TLV320AIC3256 codec hooked up to an OMAP 4 processor. My biggest issue right now is that I'm only getting one of the channels. E.g. if I run: % tinycap test.wav -d 0 -r 48000 -b 16 Only one of the channels (left, in this case) has any data. The right channel is all 0s. I'm using the tlv320aic325x.c driver that's publicly available from TI[1]. I've verified that the I2S data sent from the codec to the processor has data on both channels (i.e. I see I2S data on both the high and low phases of the frame sync). I'm looking for any help figuring out where to start looking in the sound system to debug this. I'm not really familiar with how the I2S data makes its way from the hardware buffers to the PCM driver; any help there would be appreciated too! Thanks - B [1]: https://gitorious.org/ti-codecs/ti-codecs/source/721c6e957db39a9d2174cecd4c…

1 1

[alsa-devel] [PATCH] ASoC: io: Remove SPI support
by Mark Brown 16 Feb '14

16 Feb '14

From: Mark Brown <broonie(a)linaro.org> All ASoC CODEC drivers that use SPI have now been converted to use regmap so we can delete SND_SOC_SPI, preventing any new users being added. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/soc.h | 1 - sound/soc/soc-io.c | 7 ------- 2 files changed, 8 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 21d025e..21038e0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -356,7 +356,6 @@ extern struct snd_ac97_bus_ops *soc_ac97_ops; enum snd_soc_control_type { SND_SOC_I2C = 1, - SND_SOC_SPI, SND_SOC_REGMAP, }; diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index aa886cc..3a0d99e 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -106,13 +106,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, break; #endif -#if IS_ENABLED(CONFIG_REGMAP_SPI) - case SND_SOC_SPI: - codec->control_data = regmap_init_spi(to_spi_device(codec->dev), - &config); - break; -#endif - case SND_SOC_REGMAP: /* Device has made its own regmap arrangements */ codec->using_regmap = true; -- 1.9.0.rc3

1 0

[alsa-devel] [PATCH] ASoC: max98090: make REVISION_ID readable
by Stephen Warren 16 Feb '14

16 Feb '14

From: Stephen Warren <swarren(a)nvidia.com> The REVISION_ID register is not currently marked readable. snd_soc_read() refuses to read the register, and hence probe() fails. Fixes: d4807ad2c4c0 ("regmap: Check readable regs in _regmap_read") [exposed the bug, by checking for readability] Fixes: 685e42154dcf ("ASoC: Replace max98090 Device Driver") [left out this register from the readable list] Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/max98090.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 51f9b3d16b41..141091c62db5 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -336,6 +336,7 @@ static bool max98090_readable_register(struct device *dev, unsigned int reg) case M98090_REG_RECORD_TDM_SLOT: case M98090_REG_SAMPLE_RATE: case M98090_REG_DMIC34_BIQUAD_BASE ... M98090_REG_DMIC34_BIQUAD_BASE + 0x0E: + case M98090_REG_REVISION_ID: return true; default: return false; -- 1.8.1.5

2 1

[alsa-devel] [PATCH] ASoC: pcm512x: Add regmap select
by Mark Brown 16 Feb '14

16 Feb '14

From: Mark Brown <broonie(a)linaro.org> We need at least the core regmap code to build. 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 56d0c28..fa47a83 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -316,6 +316,7 @@ config SND_SOC_PCM3008 config SND_SOC_PCM512x tristate "Texas Instruments PCM512x CODECs" + select REGMAP config SND_SOC_RT5631 tristate -- 1.9.0.rc3

1 0

[alsa-devel] user defined plugin and namehint
by Olivier Langlois 15 Feb '14

15 Feb '14

Hi, I created a new plugin in asound.conf and I would like it to appear in the list presented with aplay -L as this is the list used by my media player (vlc) to offer the option to choose my playback device. I have tried this: pcm.rate96000Hz { type hw card 0 device 0 rate 96000 hint { show on description "96000Hz rate output" } } It seems that it is missing the small magic used by standard specs. ie: from front.conf: hint { show { @func refer name defaults.namehint.basic } description "Front speakers" device $DEV } Background problem: I have a flac audio file with 88.2 kHz sampling rate 24 bits and the player is opening the playback device at 48kHz and downsample my file. >From the ui, there doesn't seem to be a way to fine tune these params. I will bring the discussion to vlc dev as well. Thank you,

1 1

[alsa-devel] Roland V-Studio 20
by Keith A. Milner 15 Feb '14

15 Feb '14

I've recently picked up a cheap Roland V-Studio 20 device. This is described in the product spec as supporting 24-bit, but it is appearing as 16-bit only. Unfortunately, Roland seem to insist on their own proprietary descriptors, which I don't know how to decode, so I'm not sure whether this is genuinely a 24-bit interface that isn't being interpreted properly by snd-usb-audio, or whether it's marketing spin, and it's actually 16-bit. I know that some patches were put into snd-usb-audio to "unify" some of the individual quirks that were used for Roland gear, but I don;t know if these support 24-bit interfaces. Any ideas/suggestions? The lsusb -v is as follows: Bus 007 Device 002: ID 0582:0117 Roland Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 8 idVendor 0x0582 Roland Corp. idProduct 0x0117 bcdDevice 1.00 iManufacturer 1 Cakewalk iProduct 2 VS-20 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 167 bNumInterfaces 3 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 360mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 06 24 f1 01 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 03 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0120 1x 288 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 03 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0120 1x 288 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 02 02 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1 Device Status: 0x0000 (Bus Powered)

2 2

[alsa-devel] Using HDA on an SoC without PCI
by Dylan Reid 14 Feb '14

14 Feb '14

Hi, I'm working on an system with an HDMI codec on an HDA interface, but that interface doesn't use PCI. I want to re-use as much of the code as possible, but I want to avoid scattering ifdefs all over hda_intel.c. Would gathering the pci specific functions from hda_intel.c into an interface struct of some kind make sense? If azx was probed from pci, then a pci interface would be used, if it was a platform driver, then the correct interface for the platform would be used. It would add some overhead to operations such as azx_writel, but that would be measured to make sure it isn't detrimental to performance. Any other ideas on how to approach this? Thanks, Dylan

3 3

[alsa-devel] [PATCH 1/5] ASoC: Intel: Add Intel SST audio DSP low level shim driver.
by Liam Girdwood 14 Feb '14

14 Feb '14

Add support for Intel Smart Sound Technology (SST) audio DSPs. This driver provides the low level IO, reset, boot and IRQ management for Intel audio DSPs. These files make up the low level part of the SST audio driver stack and will be used by many Intel SST cores like Haswell, Broadwell and Baytrail. SST DSPs expose a memory mapped region (shim) for config and control. The shim layout is mostly shared without much modification across cores and this driver provides a uniform API to access the shim and to enable basic shim functions. It also provides functionality to abstract some shim functions for cores with different shim features. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-dsp-priv.h | 291 ++++++++++++++++++++++++++++++++ sound/soc/intel/sst-dsp.c | 370 +++++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-dsp.h | 226 +++++++++++++++++++++++++ 3 files changed, 887 insertions(+) create mode 100644 sound/soc/intel/sst-dsp-priv.h create mode 100644 sound/soc/intel/sst-dsp.c create mode 100644 sound/soc/intel/sst-dsp.h diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h new file mode 100644 index 0000000..c4db4c7 --- /dev/null +++ b/sound/soc/intel/sst-dsp-priv.h @@ -0,0 +1,291 @@ +/* + * Intel Smart Sound Technology + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#ifndef __SOUND_SOC_SST_DSP_PRIV_H +#define __SOUND_SOC_SST_DSP_PRIV_H + +#include <linux/kernel.h> +#include <linux/types.h> +#include <linux/interrupt.h> +#include <linux/firmware.h> + +struct sst_mem_block; +struct sst_module; +struct sst_fw; + +/* + * DSP Operations exported by platform Audio DSP driver. + */ +struct sst_ops { + /* DSP core boot / reset */ + void (*boot)(struct sst_dsp *); + void (*reset)(struct sst_dsp *); + + /* Shim IO */ + void (*write)(void __iomem *addr, u32 offset, u32 value); + u32 (*read)(void __iomem *addr, u32 offset); + void (*write64)(void __iomem *addr, u32 offset, u64 value); + u64 (*read64)(void __iomem *addr, u32 offset); + + /* DSP I/DRAM IO */ + void (*ram_read)(struct sst_dsp *sst, void *dest, void *src, size_t bytes); + void (*ram_write)(struct sst_dsp *sst, void *dest, void *src, size_t bytes); + + void (*dump)(struct sst_dsp *); + + /* IRQ handlers */ + irqreturn_t (*irq_handler)(int irq, void *context); + + /* SST init and free */ + int (*init)(struct sst_dsp *sst, struct sst_pdata *pdata); + void (*free)(struct sst_dsp *sst); + + /* FW module parser/loader */ + int (*parse_fw)(struct sst_fw *sst_fw); +}; + +/* + * Audio DSP memory offsets and addresses. + */ +struct sst_addr { + u32 lpe_base; + u32 shim_offset; + u32 iram_offset; + void __iomem *lpe; + void __iomem *shim; + void __iomem *pci_cfg; + void __iomem *fw_ext; +}; + +/* + * Audio DSP Mailbox configuration. + */ +struct sst_mailbox { + void __iomem *in_base; + void __iomem *out_base; + size_t in_size; + size_t out_size; +}; + +/* + * Audio DSP Firmware data types. + */ +enum sst_data_type { + SST_DATA_M = 0, /* module block data */ + SST_DATA_P = 1, /* peristant data (text, data) */ + SST_DATA_S = 2, /* scratch data (usually buffers) */ +}; + +/* + * Audio DSP memory block types. + */ +enum sst_mem_type { + SST_MEM_IRAM = 0, + SST_MEM_DRAM = 1, + SST_MEM_ANY = 2, + SST_MEM_CACHE= 3, +}; + +/* + * Audio DSP Generic Firmware File. + * + * SST Firmware files can consist of 1..N modules. This generic structure is + * used to manage each firmware file and it's modules regardless of SST firmware + * type. A SST driver may load multiple FW files. + */ +struct sst_fw { + struct sst_dsp *dsp; + + /* base addresses of FW file data */ + dma_addr_t dmable_fw_paddr; /* physical address of fw data */ + void *dma_buf; /* virtual address of fw data */ + u32 size; /* size of fw data */ + + /* lists */ + struct list_head list; /* DSP list of FW */ + struct list_head module_list; /* FW list of modules */ + + void *private; /* core doesn't touch this */ +}; + +/* + * Audio DSP Generic Module data. + * + * This is used to dsecribe any sections of persistent (text and data) and + * scratch (buffers) of module data in ADSP memory space. + */ +struct sst_module_data { + + enum sst_mem_type type; /* destination memory type */ + enum sst_data_type data_type; /* type of module data */ + + u32 size; /* size in bytes */ + u32 offset; /* offset in FW file */ + u32 data_offset; /* offset in ADSP memory space */ + void *data; /* module data */ +}; + +/* + * Audio DSP Generic Module Template. + * + * Used to define and register a new FW module. This data is extracted from + * FW module header information. + */ +struct sst_module_template { + u32 id; + u32 entry; /* entry point */ + struct sst_module_data s; /* scratch data */ + struct sst_module_data p; /* peristant data */ +}; + +/* + * Audio DSP Generic Module. + * + * Each Firmware file can consist of 1..N modules. A module can span multiple + * ADSP memory blocks. The simplest FW will be a file with 1 module. + */ +struct sst_module { + struct sst_dsp *dsp; + struct sst_fw *sst_fw; /* parent FW we belong too */ + + /* module configuration */ + u32 id; + u32 entry; /* module entry point */ + u32 offset; /* module offset in firmware file */ + u32 size; /* module size */ + struct sst_module_data s; /* scratch data */ + struct sst_module_data p; /* peristant data */ + + /* runtime */ + u32 usage_count; /* can be unloaded if count == 0 */ + void *private; /* core doesn't touch this */ + + /* lists */ + struct list_head block_list; /* Module list of blocks in use */ + struct list_head list; /* DSP list of modules */ + struct list_head list_fw; /* FW list of modules */ +}; + +/* + * SST Memory Block operations. + */ +struct sst_block_ops { + int (*enable)(struct sst_mem_block *block); + int (*disable)(struct sst_mem_block *block); +}; + +/* + * SST Generic Memory Block. + * + * SST ADP memory has multiple IRAM and DRAM blocks. Some ADSP blocks can be + * power gated. + */ +struct sst_mem_block { + struct sst_dsp *dsp; + struct sst_module *module; /* module that uses this block */ + + /* block config */ + u32 offset; /* offset from base */ + u32 size; /* block size */ + u32 index; /* block index 0..N */ + enum sst_mem_type type; /* block memory type IRAM/DRAM */ + struct sst_block_ops *ops; /* block operations, if any */ + + /* block status */ + enum sst_data_type data_type; /* data type held in this block */ + u32 bytes_used; /* bytes in use by modules */ + void *private; /* generic core does not touch this */ + int users; /* number of modules using this block */ + + /* block lists */ + struct list_head module_list; /* Module list of blocks */ + struct list_head list; /* Map list of free/used blocks */ +}; + +/* + * Generic SST Shim Interface. + */ +struct sst_dsp { + + /* runtime */ + struct sst_dsp_device *sst_dev; + spinlock_t spinlock; /* IPC locking */ + struct mutex mutex; /* DSP FW lock */ + struct device *dev; + void *thread_context; + int irq; + u32 id; + + /* list of free and used ADSP memory blocks */ + struct list_head used_block_list; + struct list_head free_block_list; + + /* operations */ + struct sst_ops *ops; + + /* debug FS */ + struct dentry *debugfs_root; + + /* base addresses */ + struct sst_addr addr; + + /* mailbox */ + struct sst_mailbox mailbox; + + /* SST FW files loaded and their modules */ + struct list_head module_list; + struct list_head fw_list; + + /* platform data */ + struct sst_pdata *pdata; + + /* DMA FW loading */ + struct sst_dma *dma; + bool fw_use_dma; +}; + +/* Create/Free FW files - can contain multiple modules */ +struct sst_fw *sst_fw_new(struct sst_dsp *dsp, + const struct firmware *fw, void *private); +void sst_fw_free(struct sst_fw *sst_fw); +void sst_fw_free_all(struct sst_dsp *dsp); + +/* Create/Free firmware modules */ +struct sst_module *sst_module_new(struct sst_fw *sst_fw, + struct sst_module_template *template, void *private); +void sst_module_free(struct sst_module *sst_module); +int sst_module_insert(struct sst_module *sst_module); +int sst_module_remove(struct sst_module *sst_module); +int sst_module_insert_fixed_block(struct sst_module *module, + struct sst_module_data *data); +struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id); + +/* allocate/free pesistent/scratch memory regions managed by drv */ +struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp); +void sst_mem_block_free_scratch(struct sst_dsp *dsp, + struct sst_module *scratch); + +/* Register the DSPs memory blocks - would be nice to read from ACPI */ +struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset, + u32 size, enum sst_mem_type type, struct sst_block_ops *ops, u32 index, + void *private); +void sst_mem_block_unregister_all(struct sst_dsp *dsp); + +/* DMA */ +int sst_dma_new(struct sst_dsp *sst); +void sst_dma_free(struct sst_dma *dma); + +#endif diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c new file mode 100644 index 0000000..beef2ef --- /dev/null +++ b/sound/soc/intel/sst-dsp.c @@ -0,0 +1,370 @@ +/* + * Intel Smart Sound Technology (SST) DSP Core Driver + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include <linux/slab.h> +#include <linux/export.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> + +#include "sst-dsp.h" +#include "sst-dsp-priv.h" + +#define CREATE_TRACE_POINTS +#include <trace/events/sst.h> + +/* Public API */ +void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value) +{ + unsigned long flags; + + spin_lock_irqsave(&sst->spinlock, flags); + sst->ops->write(sst->addr.shim, offset, value); + spin_unlock_irqrestore(&sst->spinlock, flags); +} +EXPORT_SYMBOL(sst_dsp_shim_write); + +u32 sst_dsp_shim_read(struct sst_dsp *sst, u32 offset) +{ + unsigned long flags; + u32 val; + + spin_lock_irqsave(&sst->spinlock, flags); + val = sst->ops->read(sst->addr.shim, offset); + spin_unlock_irqrestore(&sst->spinlock, flags); + + return val; +} +EXPORT_SYMBOL(sst_dsp_shim_read); + +void sst_dsp_shim_write64(struct sst_dsp *sst, u32 offset, u64 value) +{ + unsigned long flags; + + spin_lock_irqsave(&sst->spinlock, flags); + sst->ops->write64(sst->addr.shim, offset, value); + spin_unlock_irqrestore(&sst->spinlock, flags); +} +EXPORT_SYMBOL(sst_dsp_shim_write64); + +u64 sst_dsp_shim_read64(struct sst_dsp *sst, u32 offset) +{ + unsigned long flags; + u64 val; + + spin_lock_irqsave(&sst->spinlock, flags); + val = sst->ops->read64(sst->addr.shim, offset); + spin_unlock_irqrestore(&sst->spinlock, flags); + + return val; +} +EXPORT_SYMBOL(sst_dsp_shim_read64); + +void sst_dsp_shim_write_unlocked(struct sst_dsp *sst, u32 offset, u32 value) +{ + sst->ops->write(sst->addr.shim, offset, value); +} +EXPORT_SYMBOL(sst_dsp_shim_write_unlocked); + +u32 sst_dsp_shim_read_unlocked(struct sst_dsp *sst, u32 offset) +{ + return sst->ops->read(sst->addr.shim, offset); +} +EXPORT_SYMBOL(sst_dsp_shim_read_unlocked); + +void sst_dsp_shim_write64_unlocked(struct sst_dsp *sst, u32 offset, u64 value) +{ + sst->ops->write64(sst->addr.shim, offset, value); +} +EXPORT_SYMBOL(sst_dsp_shim_write64_unlocked); + +u64 sst_dsp_shim_read64_unlocked(struct sst_dsp *sst, u32 offset) +{ + return sst->ops->read64(sst->addr.shim, offset); +} +EXPORT_SYMBOL(sst_dsp_shim_read64_unlocked); + +int sst_dsp_shim_update_bits(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value) +{ + unsigned long flags; + bool change; + u32 old, new; + + spin_lock_irqsave(&sst->spinlock, flags); + old = sst_dsp_shim_read_unlocked(sst, offset); + + new = (old & (~mask)) | (value & mask); + + change = (old != new); + if (change) + sst_dsp_shim_write_unlocked(sst, offset, new); + + spin_unlock_irqrestore(&sst->spinlock, flags); + return change; +} +EXPORT_SYMBOL(sst_dsp_shim_update_bits); + +int sst_dsp_shim_update_bits64(struct sst_dsp *sst, u32 offset, + u64 mask, u64 value) +{ + unsigned long flags; + bool change; + u64 old, new; + + spin_lock_irqsave(&sst->spinlock, flags); + old = sst_dsp_shim_read64_unlocked(sst, offset); + + new = (old & (~mask)) | (value & mask); + + change = (old != new); + if (change) + sst_dsp_shim_write64_unlocked(sst, offset, new); + + spin_unlock_irqrestore(&sst->spinlock, flags); + return change; +} +EXPORT_SYMBOL(sst_dsp_shim_update_bits64); + +int sst_dsp_shim_update_bits_unlocked(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value) +{ + bool change; + unsigned int old, new; + u32 ret; + + ret = sst_dsp_shim_read_unlocked(sst, offset); + + old = ret; + new = (old & (~mask)) | (value & mask); + + change = (old != new); + if (change) + sst_dsp_shim_write_unlocked(sst, offset, new); + + return change; +} +EXPORT_SYMBOL(sst_dsp_shim_update_bits_unlocked); + +int sst_dsp_shim_update_bits64_unlocked(struct sst_dsp *sst, u32 offset, + u64 mask, u64 value) +{ + bool change; + u64 old, new; + + old = sst_dsp_shim_read64_unlocked(sst, offset); + + new = (old & (~mask)) | (value & mask); + + change = (old != new); + if (change) + sst_dsp_shim_write64_unlocked(sst, offset, new); + + return change; +} +EXPORT_SYMBOL(sst_dsp_shim_update_bits64_unlocked); + +void sst_dsp_dump(struct sst_dsp *sst) +{ + sst->ops->dump(sst); +} +EXPORT_SYMBOL(sst_dsp_dump); + +void sst_dsp_reset(struct sst_dsp *sst) +{ + sst->ops->reset(sst); +} +EXPORT_SYMBOL(sst_dsp_reset); + +int sst_dsp_boot(struct sst_dsp *sst) +{ + sst->ops->boot(sst); + return 0; +} +EXPORT_SYMBOL(sst_dsp_boot); + +void sst_dsp_ipc_msg_tx(struct sst_dsp *dsp, u32 msg) +{ + sst_dsp_shim_write_unlocked(dsp, SST_IPCX, msg | SST_IPCX_BUSY); + trace_sst_ipc_msg_tx(msg); +} +EXPORT_SYMBOL_GPL(sst_dsp_ipc_msg_tx); + +u32 sst_dsp_ipc_msg_rx(struct sst_dsp *dsp) +{ + u32 msg; + + msg = sst_dsp_shim_read_unlocked(dsp, SST_IPCX); + trace_sst_ipc_msg_rx(msg); + + return msg; +} +EXPORT_SYMBOL_GPL(sst_dsp_ipc_msg_rx); + +void sst_dsp_write(struct sst_dsp *sst, void *src, u32 dest_offset, + size_t bytes) +{ + sst->ops->ram_write(sst, sst->addr.lpe + dest_offset, src, bytes); +} +EXPORT_SYMBOL(sst_dsp_write); + +void sst_dsp_read(struct sst_dsp *sst, void *dest, u32 src_offset, + size_t bytes) +{ + sst->ops->ram_read(sst, dest, sst->addr.lpe + src_offset, bytes); +} +EXPORT_SYMBOL(sst_dsp_read); + +int sst_dsp_mailbox_init(struct sst_dsp *sst, u32 inbox_offset, size_t inbox_size, + u32 outbox_offset, size_t outbox_size) +{ + sst->mailbox.in_base = sst->addr.lpe + inbox_offset; + sst->mailbox.out_base = sst->addr.lpe + outbox_offset; + sst->mailbox.in_size = inbox_size; + sst->mailbox.out_size = outbox_size; + return 0; +} +EXPORT_SYMBOL(sst_dsp_mailbox_init); + +void sst_dsp_outbox_write(struct sst_dsp *sst, void *message, size_t bytes) +{ + int i; + + trace_sst_ipc_outbox_write(bytes); + + memcpy_toio(sst->mailbox.out_base, message, bytes); + + for (i = 0; i < bytes; i += 4) + trace_sst_ipc_outbox_wdata(i, *(uint32_t *)(message + i)); +} +EXPORT_SYMBOL(sst_dsp_outbox_write); + +void sst_dsp_outbox_read(struct sst_dsp *sst, void *message, size_t bytes) +{ + int i; + + trace_sst_ipc_outbox_read(bytes); + + memcpy_fromio(message, sst->mailbox.out_base, bytes); + + for (i = 0; i < bytes; i += 4) + trace_sst_ipc_outbox_rdata(i, *(uint32_t *)(message + i)); +} +EXPORT_SYMBOL(sst_dsp_outbox_read); + +void sst_dsp_inbox_write(struct sst_dsp *sst, void *message, size_t bytes) +{ + int i; + + trace_sst_ipc_inbox_write(bytes); + + memcpy_toio(sst->mailbox.in_base, message, bytes); + + for (i = 0; i < bytes; i += 4) + trace_sst_ipc_inbox_wdata(i, *(uint32_t *)(message + i)); +} +EXPORT_SYMBOL(sst_dsp_inbox_write); + +void sst_dsp_inbox_read(struct sst_dsp *sst, void *message, size_t bytes) +{ + int i; + + trace_sst_ipc_inbox_read(bytes); + + memcpy_fromio(message, sst->mailbox.in_base, bytes); + + for (i = 0; i < bytes; i += 4) + trace_sst_ipc_inbox_rdata(i, *(uint32_t *)(message + i)); +} +EXPORT_SYMBOL(sst_dsp_inbox_read); + +void *sst_dsp_get_thread_context(struct sst_dsp *sst) +{ + return sst->thread_context; +} +EXPORT_SYMBOL(sst_dsp_get_thread_context); + +struct sst_dsp *sst_dsp_new(struct device *dev, + struct sst_dsp_device *sst_dev, struct sst_pdata *pdata) +{ + struct sst_dsp *sst; + int err; + + dev_dbg(dev, "initialising audio DSP id 0x%x\n", pdata->id); + + sst = devm_kzalloc(dev, sizeof(*sst), GFP_KERNEL); + if (sst == NULL) + return NULL; + + spin_lock_init(&sst->spinlock); + mutex_init(&sst->mutex); + sst->dev = dev; + sst->thread_context = sst_dev->thread_context; + sst->sst_dev = sst_dev; + sst->id = pdata->id; + sst->irq = pdata->irq; + sst->ops = sst_dev->ops; + sst->pdata = pdata; + INIT_LIST_HEAD(&sst->used_block_list); + INIT_LIST_HEAD(&sst->free_block_list); + INIT_LIST_HEAD(&sst->module_list); + INIT_LIST_HEAD(&sst->fw_list); + + /* Initialise SST Audio DSP */ + if (sst->ops->init) { + err = sst->ops->init(sst, pdata); + if (err < 0) + return NULL; + } + + /* Register the ISR */ + err = request_threaded_irq(sst->irq, sst->ops->irq_handler, + sst_dev->thread, IRQF_SHARED, "AudioDSP", sst); + if (err) + goto irq_err; + + /* Register the FW loader DMA controller if we have one */ + if (pdata->dma_engine) + err = sst_dma_new(sst); + if (err) + goto dma_err; + + return sst; + +dma_err: + free_irq(sst->irq, sst); +irq_err: + if (sst->ops->free) + sst->ops->free(sst); + + return NULL; +} +EXPORT_SYMBOL(sst_dsp_new); + +void sst_dsp_free(struct sst_dsp *sst) +{ + if (sst->pdata->dma_engine) + sst_dma_free(sst->dma); + free_irq(sst->irq, sst); + if (sst->ops->free) + sst->ops->free(sst); +} +EXPORT_SYMBOL(sst_dsp_free); + +/* Module information */ +MODULE_AUTHOR("Liam Girdwood"); +MODULE_DESCRIPTION("Intel SST Core"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h new file mode 100644 index 0000000..8ea3cbe --- /dev/null +++ b/sound/soc/intel/sst-dsp.h @@ -0,0 +1,226 @@ +/* + * Intel Smart Sound Technology (SST) Core + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#ifndef __SOUND_SOC_SST_DSP_H +#define __SOUND_SOC_SST_DSP_H + +#include <linux/kernel.h> +#include <linux/types.h> +#include <linux/interrupt.h> + +/* SST Device IDs */ +#define SST_DEV_ID_LYNX_POINT 0x33C8 +#define SST_DEV_ID_WILDCAT_POINT 0x3438 + +/* Supported SST DMA Devices */ +#define SST_DMA_TYPE_DW 1 +#define SST_DMA_TYPE_MID 2 + +/* SST Shim register map + * The register naming can differ between products. Some products also + * contain extra functionality. + */ +#define SST_CSR 0x00 +#define SST_PISR 0x08 +#define SST_PIMR 0x10 +#define SST_ISRX 0x18 +#define SST_ISRD 0x20 +#define SST_IMRX 0x28 +#define SST_IMRD 0x30 +#define SST_IPCX 0x38 /* IPC IA -> SST */ +#define SST_IPCD 0x40 /* IPC SST -> IA */ +#define SST_ISRSC 0x48 +#define SST_ISRLPESC 0x50 +#define SST_IMRSC 0x58 +#define SST_IMRLPESC 0x60 +#define SST_IPCSC 0x68 +#define SST_IPCLPESC 0x70 +#define SST_CLKCTL 0x78 +#define SST_CSR2 0x80 +#define SST_LTRC 0xE0 +#define SST_HDMC 0xE8 +#define SST_DBGO 0xF0 + +#define SST_SHIM_SIZE 0x100 +#define SST_PWMCTRL 0x1000 + +/* SST Shim Register bits + * The register bit naming can differ between products. Some products also + * contain extra functionality. + */ + +/* CSR / CS */ +#define SST_CSR_RST (0x1 << 1) +#define SST_CSR_SBCS0 (0x1 << 2) +#define SST_CSR_SBCS1 (0x1 << 3) +#define SST_CSR_DCS(x) (x << 4) +#define SST_CSR_DCS_MASK (0x7 << 4) +#define SST_CSR_STALL (0x1 << 10) +#define SST_CSR_S0IOCS (0x1 << 21) +#define SST_CSR_S1IOCS (0x1 << 23) +#define SST_CSR_LPCS (0x1 << 31) + +/* ISRX / ISC */ +#define SST_ISRX_BUSY (0x1 << 1) +#define SST_ISRX_DONE (0x1 << 0) + +/* ISRD / ISD */ +#define SST_ISRD_BUSY (0x1 << 1) +#define SST_ISRD_DONE (0x1 << 0) + +/* IMRX / IMC */ +#define SST_IMRX_BUSY (0x1 << 1) +#define SST_IMRX_DONE (0x1 << 0) + +/* IPCX / IPCC */ +#define SST_IPCX_DONE (0x1 << 30) +#define SST_IPCX_BUSY (0x1 << 31) + +/* IPCD */ +#define SST_IPCD_DONE (0x1 << 30) +#define SST_IPCD_BUSY (0x1 << 31) + +/* CLKCTL */ +#define SST_CLKCTL_SMOS(x) (x << 24) +#define SST_CLKCTL_MASK (3 << 24) +#define SST_CLKCTL_DCPLCG (1 << 18) +#define SST_CLKCTL_SCOE1 (1 << 17) +#define SST_CLKCTL_SCOE0 (1 << 16) + +/* CSR2 / CS2 */ +#define SST_CSR2_SDFD_SSP0 (1 << 1) +#define SST_CSR2_SDFD_SSP1 (1 << 2) + +/* LTRC */ +#define SST_LTRC_VAL(x) (x << 0) + +/* HDMC */ +#define SST_HDMC_HDDA0(x) (x << 0) +#define SST_HDMC_HDDA1(x) (x << 7) + + +/* SST Vendor Defined Registers and bits */ +#define SST_VDRTCTL0 0xa0 +#define SST_VDRTCTL1 0xa4 +#define SST_VDRTCTL2 0xa8 +#define SST_VDRTCTL3 0xaC + +/* VDRTCTL0 */ +#define SST_VDRTCL0_DSRAMPGE_SHIFT 16 +#define SST_VDRTCL0_DSRAMPGE_MASK (0xffff << SST_VDRTCL0_DSRAMPGE_SHIFT) +#define SST_VDRTCL0_ISRAMPGE_SHIFT 6 +#define SST_VDRTCL0_ISRAMPGE_MASK (0x3ff << SST_VDRTCL0_ISRAMPGE_SHIFT) + +struct sst_dsp; + +/* + * SST Device. + * + * This structure is populated by the SST core driver. + */ +struct sst_dsp_device { + /* Mandatory fields */ + struct sst_ops *ops; + irqreturn_t (*thread)(int irq, void *context); + void *thread_context; +}; + +/* + * SST Platform Data. + */ +struct sst_pdata { + /* ACPI data */ + u32 lpe_base; + u32 lpe_size; + u32 pcicfg_base; + u32 pcicfg_size; + int irq; + + /* Firmware */ + const char *fw_filename; + u32 fw_base; + u32 fw_size; + + /* DMA */ + u32 dma_base; + u32 dma_size; + int dma_engine; + + /* DSP */ + u32 id; + void *dsp; +}; + +/* Initialization */ +struct sst_dsp *sst_dsp_new(struct device *dev, + struct sst_dsp_device *sst_dev, struct sst_pdata *pdata); +void sst_dsp_free(struct sst_dsp *sst); + +/* SHIM Read / Write */ +void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value); +u32 sst_dsp_shim_read(struct sst_dsp *sst, u32 offset); +int sst_dsp_shim_update_bits(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value); +void sst_dsp_shim_write64(struct sst_dsp *sst, u32 offset, u64 value); +u64 sst_dsp_shim_read64(struct sst_dsp *sst, u32 offset); +int sst_dsp_shim_update_bits64(struct sst_dsp *sst, u32 offset, + u64 mask, u64 value); + +/* SHIM Read / Write Unlocked for callers already holding sst lock */ +void sst_dsp_shim_write_unlocked(struct sst_dsp *sst, u32 offset, u32 value); +u32 sst_dsp_shim_read_unlocked(struct sst_dsp *sst, u32 offset); +int sst_dsp_shim_update_bits_unlocked(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value); +void sst_dsp_shim_write64_unlocked(struct sst_dsp *sst, u32 offset, u64 value); +u64 sst_dsp_shim_read64_unlocked(struct sst_dsp *sst, u32 offset); +int sst_dsp_shim_update_bits64_unlocked(struct sst_dsp *sst, u32 offset, + u64 mask, u64 value); + +/* Size optimised DRAM/IRAM memcpy */ +void sst_dsp_write(struct sst_dsp *sst, void *src, u32 dest_offset, + size_t bytes); +void sst_dsp_bzero(struct sst_dsp *sst, u32 src_offset, size_t bytes); +void sst_dsp_read(struct sst_dsp *sst, void *dest, u32 src_offset, + size_t bytes); + +/* DSP reset & boot */ +void sst_dsp_reset(struct sst_dsp *sst); +int sst_dsp_boot(struct sst_dsp *sst); + +/* Msg IO */ +void sst_dsp_ipc_msg_tx(struct sst_dsp *dsp, u32 msg); +u32 sst_dsp_ipc_msg_rx(struct sst_dsp *dsp); +void *sst_dsp_get_thread_context(struct sst_dsp *sst); + +/* Mailbox management */ +int sst_dsp_mailbox_init(struct sst_dsp *dsp, u32 inbox_offset, + size_t inbox_size, u32 outbox_offset, size_t outbox_size); +void sst_dsp_inbox_write(struct sst_dsp *dsp, void *message, size_t bytes); +void sst_dsp_inbox_read(struct sst_dsp *dsp, void *message, size_t bytes); +void sst_dsp_outbox_write(struct sst_dsp *dsp, void *message, size_t bytes); +void sst_dsp_outbox_read(struct sst_dsp *dsp, void *message, size_t bytes); +void sst_dsp_mailbox_dump(struct sst_dsp *dsp, size_t bytes); + +/* DMA FW maangement */ +int sst_dsp_dma_copy(struct sst_dsp *sst, dma_addr_t src_addr, + dma_addr_t dstn_addr, size_t size); +int sst_dsp_dma_get_channel(struct sst_dsp *dsp, int chan_id); +void sst_dsp_dma_put_channel(struct sst_dsp *dsp); + +/* Debug */ +void sst_dsp_dump(struct sst_dsp *sst); + +#endif -- 1.8.3.2

4 14

[alsa-devel] [PATCH] ak4117: Do not free priv until timer handler hasn't actually stopped using it
by Kirill Tkhai 14 Feb '14

14 Feb '14

Function del_timer() does not guarantee that timer was really deleted. If the timer handler is beeing executed at the moment, the function does nothing. So, it's possible to use already freed memory in the handler: [ref: Documentation/DocBook/kernel-locking.tmpl] This was found using grep and compile-tested only. Signed-off-by: Kirill Tkhai <ktkhai(a)parallels.com> CC: Jaroslav Kysela <perex(a)perex.cz> CC: Takashi Iwai <tiwai(a)suse.de> --- sound/i2c/other/ak4117.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/i2c/other/ak4117.c b/sound/i2c/other/ak4117.c index 40e33c9..88452e8 100644 --- a/sound/i2c/other/ak4117.c +++ b/sound/i2c/other/ak4117.c @@ -62,7 +62,7 @@ static void reg_dump(struct ak4117 *ak4117) static void snd_ak4117_free(struct ak4117 *chip) { - del_timer(&chip->timer); + del_timer_sync(&chip->timer); kfree(chip); }

2 1

[alsa-devel] [PATCH RFC 00/10] Use standard printk helpers for ALSA core codes
by Takashi Iwai 14 Feb '14

14 Feb '14

OK, this is the last batch. This series converts the use of snd_printk() and co with the standard ones like dev_err() in ALSA core codes. This became possible now by the new card device management. Takashi

1 13

[alsa-devel] [PATCH RFC 0/3] Embed card device into struct snd_card
by Takashi Iwai 14 Feb '14

14 Feb '14

The third batch is a more intensive change; now struct device for card-object is embedded in snd_card struct. This makes the presence of the device object presistent, which can be later referred by other devices at the object creation time. Also, along with this change, the refcounting of the card object was changed just to use the standard device refcount. This needs to move the destructor into the device release callback. Onre more batch will follow. Takashi

1 4

[alsa-devel] [PATCH RFC 00/20] snd_device_* cleanups
by Takashi Iwai 14 Feb '14

14 Feb '14

This batch is aimed for cleanups of snd_device_*() usages. The major change is to introduce SNDRV_DEV_CARD type and use it for the card-level object instead of SNDRV_DEV_LOWLEVEL. The rest are simplification of snd_device_*() calls and management. Takashi

1 21

[alsa-devel] [RFC PATCH 00/24] Mandate to pass a device pointer at card creation
by Takashi Iwai 14 Feb '14

14 Feb '14

Hi, this is the first series of patches for a slight restructuring of device management in ALSA core codes. This batch makes the association of struct device mandatory to snd_card at the object creation time, so that the driver can refer to the device at any time even before registring the device. Basically, it rewrites snd_card_create() with the (re-)new snd_card_new() that requires the struct device pointer. The only upstream driver that had no device assignment was atmel/au1x00, and I revived the old patch to convert to platform_device for that. Takashi

4 30

[alsa-devel] Sound distorted on Intel HDA (Realtek ALC889) (Kernel 3.x)
by Vincent Thiele 14 Feb '14

14 Feb '14

Audio device: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller (rev 05) Mainboard is: http://www.asus.com/Motherboards/MAXIMUS_IV_GENEZ/ Only fix at the moment is changing "load-module module-udev-detect use_ucm=0" in "/etc/pulse/default.pa " to "load-module module-udev-detect tsched=0". With Kernel 3.13.2 and Ubuntu 13.10 sound without "load-module module-udev-detect tsched=0" is better but sometimes there is still some crackling noise. Hope you can fix this. Best regards

1 0

[alsa-devel] Maya44
by marcin＠saepia.net 14 Feb '14

14 Feb '14

Hi, what is the exact status for maya44 driver? I've found some files in alsa source named maya44.* but card is not recognized. There are also some patches to 1.0.17 floating in the net. I've also found some legacy information about the driver [1] and statement dated 2008 from this mailing list that the driver "cannot be merged without a proper sign-off" [2]. Is there any chance that it will be merged upstream? Is it technical or legal problem that prevents this from being done? How can I help? Thanks, Marcin [1] https://www.kernel.org/doc/readme/Documeantation-sound-alsa-README.maya44 [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2008-May/007719.html

1 0

[alsa-devel] [PATCH 1/2] ASoC: S3C: Remove invalid file reference
by Sachin Kamat 14 Feb '14

14 Feb '14

Remove file references rendered invalid due to relocation. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- include/linux/platform_data/asoc-s3c.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/include/linux/platform_data/asoc-s3c.h b/include/linux/platform_data/asoc-s3c.h index 9efc04dd255a..709c6f7e2f8c 100644 --- a/include/linux/platform_data/asoc-s3c.h +++ b/include/linux/platform_data/asoc-s3c.h @@ -1,5 +1,4 @@ -/* arch/arm/plat-samsung/include/plat/audio.h - * +/* * Copyright (c) 2009 Samsung Electronics Co. Ltd * Author: Jaswinder Singh <jassi.brar(a)samsung.com> * -- 1.7.9.5

1 1

[alsa-devel] [PATCH] MAINTAINERS: Add entry for Cirrus CODEC drivers
by Mark Brown 14 Feb '14

14 Feb '14

From: Mark Brown <broonie(a)linaro.org> I noticed that some recent patches hadn't been CCed to the Cirrus people - add an entry in MAINTAINERS to help make that happen. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- Brian, Paul - are you OK with this? MAINTAINERS | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index b97ce2594f16..5ef6c0277075 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -2200,6 +2200,13 @@ L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Odd Fixes F: sound/soc/codecs/cs4270* +CIRRUS LOGIC AUDIO CODEC DRIVERS +M: Brian Austin <brian.austin(a)cirrus.com> +M: Paul Handrigan <Paul.Handrigan(a)cirrus.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: sound/soc/codecs/cs* + CLEANCACHE API M: Konrad Rzeszutek Wilk <konrad.wilk(a)oracle.com> L: linux-kernel(a)vger.kernel.org -- 1.9.rc1

5 9

[alsa-devel] [PATCH] ASoC: dapm: Consistently use unsigned int for register values
by Lars-Peter Clausen 13 Feb '14

13 Feb '14

Commit f7d3c1709 ("ASoC: dapm: Change prototype of soc_widget_read") changed the signature of soc_widget_read() so that it, instead of return the register value as a int, takes a pointer to a unsigned int and stores the register value in that pointer. There are still a few places that pass a int type value to that function though. Change these to unsigned int. For more consistency also change the signature of soc_widget_write() to take a unsigned int instead of an int for the register value. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dc8ff13..2c2d751 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -386,7 +386,8 @@ static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg, return -1; } -static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) +static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, + unsigned int val) { if (w->codec) return snd_soc_write(w->codec, reg, val); @@ -506,7 +507,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_switch: case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: { - int val; + unsigned int val; struct soc_mixer_control *mc = (struct soc_mixer_control *) w->kcontrol_news[i].private_value; int reg = mc->reg; @@ -530,7 +531,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_mux: { struct soc_enum *e = (struct soc_enum *) w->kcontrol_news[i].private_value; - int val, item; + unsigned int val, item; soc_widget_read(w, e->reg, &val); item = (val >> e->shift_l) & e->mask; @@ -559,7 +560,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_value_mux: { struct soc_enum *e = (struct soc_enum *) w->kcontrol_news[i].private_value; - int val, item; + unsigned int val, item; soc_widget_read(w, e->reg, &val); val = (val >> e->shift_l) & e->mask; -- 1.8.0

2 1

[alsa-devel] [PATCH v2 0/3] Add tdm slot support
by Xiubo Li 13 Feb '14

13 Feb '14

Xiubo Li (3): ASoC: binding: add tdm-slot.txt ASoC: core: add slot information parsing supports ASoC: simple-card: add slot information parsing supports .../devicetree/bindings/sound/simple-card.txt | 1 + .../devicetree/bindings/sound/tdm-slot.txt | 17 +++++++++++ include/sound/simple_card.h | 1 + include/sound/soc.h | 10 +++++++ sound/soc/generic/simple-card.c | 35 ++++++++++++++++++++-- sound/soc/soc-core.c | 23 ++++++++++++++ 6 files changed, 84 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tdm-slot.txt -- 1.8.4

4 6

[alsa-devel] [PATCH] ASoC: max98090: sync regcache on entering STANDBY
by Dylan Reid 12 Feb '14

12 Feb '14

Sync regcache when entering STANDBY from OFF. ON isn't entered with OFF as the current state, so the registers were not being re-synced after suspend/resume. The 98088 and 98095 already call regcache_sync from STANDBY. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/max98090.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 51f9b3d..149b57f 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1769,16 +1769,6 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = regcache_sync(max98090->regmap); - - if (ret != 0) { - dev_err(codec->dev, - "Failed to sync cache: %d\n", ret); - return ret; - } - } - if (max98090->jack_state == M98090_JACK_STATE_HEADSET) { /* * Set to normal bias level. @@ -1792,6 +1782,16 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regcache_sync(max98090->regmap); + if (ret != 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + break; + case SND_SOC_BIAS_OFF: /* Set internal pull-up to lowest power mode */ snd_soc_update_bits(codec, M98090_REG_JACK_DETECT, -- 1.8.1.3.605.g02339dd

2 1

[alsa-devel] [BUG]3.14-rc2 [PATCH] Revert "ALSA: hda/realtek - Avoid invalid COEFs for ALC271X"
by Martin Kepplinger 12 Feb '14

12 Feb '14

This reverts commit d3c56568f43807135f2c2a09582a69f809f0d8b7. The reverted commit breaks audio through headphone line out on the Acer TravelMate B113 (Type1Sku0) Notebook, my main work machine. I don't know much about it but this fixes my problem. Bisected and tested. Tested-by: Martin Kepplinger <martink(a)posteo.de> Signed-off-by: Martin Kepplinger <martink(a)posteo.de> --- sound/pci/hda/patch_realtek.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d9693ca..0f5af34 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4434,9 +4434,6 @@ static void alc269_fill_coef(struct hda_codec *codec) if (spec->codec_variant != ALC269_TYPE_ALC269VB) return; - /* ALC271X doesn't seem to support these COEFs (bko#52181) */ - if (!strcmp(codec->chip_name, "ALC271X")) - return; if ((alc_get_coef0(codec) & 0x00ff) < 0x015) { alc_write_coef_idx(codec, 0xf, 0x960b); -- 1.7.10.4

2 4

[alsa-devel] [PATCH v3 0/2] ASoC: atmel_ssc_dai: add option to choose clock
by Bo Shen 12 Feb '14

12 Feb '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 v3: - Move the property from card to ssc device Series-changes: 2 - using "-" replace "_" in binding document Bo Shen (2): ASoC: atmel_ssc_dai: make option to choose clock Binding: atmel-ssc: add option to choose clock Documentation/devicetree/bindings/misc/atmel-ssc.txt | 8 ++++++++ drivers/misc/atmel-ssc.c | 6 ++++++ include/linux/atmel-ssc.h | 1 + sound/soc/atmel/atmel_ssc_dai.c | 13 +++++++++---- 4 files changed, 24 insertions(+), 4 deletions(-) -- 1.8.5.2

3 5

Re: [alsa-devel] linux-next: Tree for Feb 10 (snd_hda)
by Randy Dunlap 12 Feb '14

12 Feb '14

On 02/09/2014 09:17 PM, Stephen Rothwell wrote: > Hi all, > > This tree fails (more than usual) the powerpc allyesconfig build. > > Changes since 20140207: > on x86_64: sound/built-in.o: In function `snd_hda_codec_configure': (.text+0x520a2): undefined reference to `snd_hda_parse_hdmi_codec' sound/built-in.o: In function `snd_hda_codec_configure': (.text+0x520a9): undefined reference to `snd_hda_parse_hdmi_codec' Lots of builtin drivers, but CONFIG_SND_HDA_CODEC_HDMI=m Full randconfig file is attached. -- ~Randy

2 1

[alsa-devel] [PATCH] ASoC: blackfin: Fix machine driver Kconfig dependencies
by Lars-Peter Clausen 12 Feb '14

12 Feb '14

Since the machine driver selects the CODEC driver we need to make sure that the machine driver is only selectable if the CODEC driver can be build. This avoids build errors under some configurations (which typically only result from randconfig). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/blackfin/Kconfig | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 54f74f8..4544d8e 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -11,7 +11,7 @@ config SND_BF5XX_I2S config SND_BF5XX_SOC_SSM2602 tristate "SoC SSM2602 Audio Codec Add-On Card support" - depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) + depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S if !BF60x select SND_BF6XX_SOC_I2S if BF60x select SND_SOC_SSM2602 @@ -21,10 +21,9 @@ config SND_BF5XX_SOC_SSM2602 config SND_SOC_BFIN_EVAL_ADAU1701 tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && I2C select SND_BF5XX_SOC_I2S select SND_SOC_ADAU1701 - select I2C help Say Y if you want to add support for the Analog Devices EVAL-ADAU1701MINIZ board connected to one of the Blackfin evaluation boards like the @@ -45,7 +44,7 @@ config SND_SOC_BFIN_EVAL_ADAU1373 config SND_SOC_BFIN_EVAL_ADAV80X tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards" - depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) + depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S select SND_SOC_ADAV80X help @@ -58,7 +57,7 @@ config SND_SOC_BFIN_EVAL_ADAV80X config SND_BF5XX_SOC_AD1836 tristate "SoC AD1836 Audio support for BF5xx" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && SPI_MASTER select SND_BF5XX_SOC_I2S select SND_SOC_AD1836 help @@ -66,7 +65,7 @@ config SND_BF5XX_SOC_AD1836 config SND_BF5XX_SOC_AD193X tristate "SoC AD193X Audio support for Blackfin" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S select SND_SOC_AD193X help -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: dapm: Add locking to snd_soc_dapm_xxxx_pin functions
by Charles Keepax 12 Feb '14

12 Feb '14

snd_soc_dapm_disable_pin, snd_soc_dapm_enable_pin and snd_soc_dapm_force_enable_pin all require the dapm_mutex to be held when they are called as they edit the dirty list. There are 385 usages of these functions and only 7 hold the lock whilst calling. This patch moves the locking into snd_soc_dapm_set_pin and fixes up the places where the lock was held on the caller side. This saves on fixing up all the current users and also is much more consistant with the rest of the DAPM API which all handles the locking internally. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Hi, I have put the changes in a single patch to avoid bisect problems, but let me know if it would be better split into seperate patches. Thanks, Charles drivers/extcon/extcon-arizona.c | 11 ----------- drivers/input/misc/arizona-haptics.c | 19 ------------------- sound/soc/soc-dapm.c | 8 ++++---- 3 files changed, 4 insertions(+), 34 deletions(-) diff --git a/drivers/extcon/extcon-arizona.c b/drivers/extcon/extcon-arizona.c index c20602f..84167f4 100644 --- a/drivers/extcon/extcon-arizona.c +++ b/drivers/extcon/extcon-arizona.c @@ -222,26 +222,19 @@ static void arizona_extcon_pulse_micbias(struct arizona_extcon_info *info) struct snd_soc_dapm_context *dapm = arizona->dapm; int ret; - mutex_lock(&dapm->card->dapm_mutex); - ret = snd_soc_dapm_force_enable_pin(dapm, widget); if (ret != 0) dev_warn(arizona->dev, "Failed to enable %s: %d\n", widget, ret); - mutex_unlock(&dapm->card->dapm_mutex); - snd_soc_dapm_sync(dapm); if (!arizona->pdata.micd_force_micbias) { - mutex_lock(&dapm->card->dapm_mutex); - ret = snd_soc_dapm_disable_pin(arizona->dapm, widget); if (ret != 0) dev_warn(arizona->dev, "Failed to disable %s: %d\n", widget, ret); - mutex_unlock(&dapm->card->dapm_mutex); snd_soc_dapm_sync(dapm); } @@ -304,16 +297,12 @@ static void arizona_stop_mic(struct arizona_extcon_info *info) ARIZONA_MICD_ENA, 0, &change); - mutex_lock(&dapm->card->dapm_mutex); - ret = snd_soc_dapm_disable_pin(dapm, widget); if (ret != 0) dev_warn(arizona->dev, "Failed to disable %s: %d\n", widget, ret); - mutex_unlock(&dapm->card->dapm_mutex); - snd_soc_dapm_sync(dapm); if (info->micd_reva) { diff --git a/drivers/input/misc/arizona-haptics.c b/drivers/input/misc/arizona-haptics.c index 7a04f54..ef2e281 100644 --- a/drivers/input/misc/arizona-haptics.c +++ b/drivers/input/misc/arizona-haptics.c @@ -37,7 +37,6 @@ static void arizona_haptics_work(struct work_struct *work) struct arizona_haptics, work); struct arizona *arizona = haptics->arizona; - struct mutex *dapm_mutex = &arizona->dapm->card->dapm_mutex; int ret; if (!haptics->arizona->dapm) { @@ -67,13 +66,10 @@ static void arizona_haptics_work(struct work_struct *work) return; } - mutex_lock_nested(dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - ret = snd_soc_dapm_enable_pin(arizona->dapm, "HAPTICS"); if (ret != 0) { dev_err(arizona->dev, "Failed to start HAPTICS: %d\n", ret); - mutex_unlock(dapm_mutex); return; } @@ -81,21 +77,14 @@ static void arizona_haptics_work(struct work_struct *work) if (ret != 0) { dev_err(arizona->dev, "Failed to sync DAPM: %d\n", ret); - mutex_unlock(dapm_mutex); return; } - - mutex_unlock(dapm_mutex); - } else { /* This disable sequence will be a noop if already enabled */ - mutex_lock_nested(dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - ret = snd_soc_dapm_disable_pin(arizona->dapm, "HAPTICS"); if (ret != 0) { dev_err(arizona->dev, "Failed to disable HAPTICS: %d\n", ret); - mutex_unlock(dapm_mutex); return; } @@ -103,12 +92,9 @@ static void arizona_haptics_work(struct work_struct *work) if (ret != 0) { dev_err(arizona->dev, "Failed to sync DAPM: %d\n", ret); - mutex_unlock(dapm_mutex); return; } - mutex_unlock(dapm_mutex); - ret = regmap_update_bits(arizona->regmap, ARIZONA_HAPTICS_CONTROL_1, ARIZONA_HAP_CTRL_MASK, @@ -155,16 +141,11 @@ static int arizona_haptics_play(struct input_dev *input, void *data, static void arizona_haptics_close(struct input_dev *input) { struct arizona_haptics *haptics = input_get_drvdata(input); - struct mutex *dapm_mutex = &haptics->arizona->dapm->card->dapm_mutex; cancel_work_sync(&haptics->work); - mutex_lock_nested(dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (haptics->arizona->dapm) snd_soc_dapm_disable_pin(haptics->arizona->dapm, "HAPTICS"); - - mutex_unlock(dapm_mutex); } static int arizona_haptics_probe(struct platform_device *pdev) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dc8ff13..a44b560 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2325,6 +2325,8 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, { struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + if (!w) { dev_err(dapm->dev, "ASoC: DAPM unknown pin %s\n", pin); return -EINVAL; @@ -2337,6 +2339,8 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, if (status == 0) w->force = 0; + mutex_unlock(&dapm->card->dapm_mutex); + return 0; } @@ -3210,15 +3214,11 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (ucontrol->value.integer.value[0]) snd_soc_dapm_enable_pin(&card->dapm, pin); else snd_soc_dapm_disable_pin(&card->dapm, pin); - mutex_unlock(&card->dapm_mutex); - snd_soc_dapm_sync(&card->dapm); return 0; } -- 1.7.2.5

3 7

[alsa-devel] [PATCH v2 1/2] ASoC: fsl-spdif: big-endian support
by Xiubo Li 12 Feb '14

12 Feb '14

For most platforms, the CPU and SPDIF device is in the same endianess mode. While for the LS1 platform, the CPU is in LE mode and the SPDIF is in BE mode. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> Cc: Nicolin Chen <Guangyu.Chen(a)freescale.com> Acked-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Fix some issues from Nicolin Chen's comments. Documentation/devicetree/bindings/sound/fsl,spdif.txt | 5 +++++ sound/soc/fsl/fsl_spdif.c | 5 ++++- 2 files changed, 9 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt index f2ae335..3e9e82c8 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.txt @@ -29,6 +29,10 @@ Required properties: can also be referred to TxClk_Source bit of register SPDIF_STC. + - big-endian : If this property is absent, the native endian mode will + be in use as default, or the big endian mode will be in use for all the + device registers. + Example: spdif: spdif@02004000 { @@ -50,5 +54,6 @@ spdif: spdif@02004000 { "rxtx5", "rxtx6", "rxtx7"; + big-endian; status = "okay"; }; diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 8f36f49..6452ca8 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -985,7 +985,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) } } -static const struct regmap_config fsl_spdif_regmap_config = { +static struct regmap_config fsl_spdif_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, @@ -1105,6 +1105,9 @@ static int fsl_spdif_probe(struct platform_device *pdev) memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai)); spdif_priv->cpu_dai_drv.name = spdif_priv->name; + if (of_property_read_bool(np, "big-endian")) + fsl_spdif_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; + /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); -- 1.8.4

2 3

[alsa-devel] [PATCH] ASoC: Intel: Add a mfld prefix to Intel SST drivers.
by Liam Girdwood 12 Feb '14

12 Feb '14

Resent with correct email for Mark. In order to differentiate the different Intel SST audio core drivers we need to rename the current drivers with a mfld prefix. This also includes renaming in the Makefile and Kconfig Acked-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/Kconfig | 4 ++-- sound/soc/intel/Makefile | 4 ++-- sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} | 8 ++++---- sound/soc/intel/{sst_platform.c => sst-mfld-platform.c} | 8 ++++---- sound/soc/intel/{sst_platform.h => sst-mfld-platform.h} | 4 ++-- 5 files changed, 14 insertions(+), 14 deletions(-) rename sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} (96%) rename sound/soc/intel/{sst_platform.c => sst-mfld-platform.c} (99%) rename sound/soc/intel/{sst_platform.h => sst-mfld-platform.h} (97%) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 61c10bf..4d9d0a5 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -2,12 +2,12 @@ config SND_MFLD_MACHINE tristate "SOC Machine Audio driver for Intel Medfield MID platform" depends on INTEL_SCU_IPC select SND_SOC_SN95031 - select SND_SST_PLATFORM + select SND_SST_MFLD_PLATFORM help This adds support for ASoC machine driver for Intel(R) MID Medfield platform used as alsa device in audio substem in Intel(R) MID devices Say Y if you have such a device If unsure select "N". -config SND_SST_PLATFORM +config SND_SST_MFLD_PLATFORM tristate diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 6398833..eb899fc 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -1,5 +1,5 @@ -snd-soc-sst-platform-objs := sst_platform.o +snd-soc-sst-mfld-platform-objs := sst-mfld-platform.o snd-soc-mfld-machine-objs := mfld_machine.o -obj-$(CONFIG_SND_SST_PLATFORM) += snd-soc-sst-platform.o +obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o diff --git a/sound/soc/intel/sst_dsp.h b/sound/soc/intel/sst-mfld-dsp.h similarity index 96% rename from sound/soc/intel/sst_dsp.h rename to sound/soc/intel/sst-mfld-dsp.h index 0fce1de..3b63edc 100644 --- a/sound/soc/intel/sst_dsp.h +++ b/sound/soc/intel/sst-mfld-dsp.h @@ -1,7 +1,7 @@ -#ifndef __SST_DSP_H__ -#define __SST_DSP_H__ +#ifndef __SST_MFLD_DSP_H__ +#define __SST_MFLD_DSP_H__ /* - * sst_dsp.h - Intel SST Driver for audio engine + * sst_mfld_dsp.h - Intel SST Driver for audio engine * * Copyright (C) 2008-12 Intel Corporation * Authors: Vinod Koul <vinod.koul(a)linux.intel.com> @@ -131,4 +131,4 @@ struct snd_sst_params { struct snd_sst_alloc_params_ext aparams; }; -#endif /* __SST_DSP_H__ */ +#endif /* __SST_MFLD_DSP_H__ */ diff --git a/sound/soc/intel/sst_platform.c b/sound/soc/intel/sst-mfld-platform.c similarity index 99% rename from sound/soc/intel/sst_platform.c rename to sound/soc/intel/sst-mfld-platform.c index f465a81..840306c 100644 --- a/sound/soc/intel/sst_platform.c +++ b/sound/soc/intel/sst-mfld-platform.c @@ -1,5 +1,5 @@ /* - * sst_platform.c - Intel MID Platform driver + * sst_mfld_platform.c - Intel MID Platform driver * * Copyright (C) 2010-2013 Intel Corp * Author: Vinod Koul <vinod.koul(a)intel.com> @@ -33,7 +33,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/compress_driver.h> -#include "sst_platform.h" +#include "sst-mfld-platform.h" static struct sst_device *sst; static DEFINE_MUTEX(sst_lock); @@ -709,7 +709,7 @@ static int sst_platform_remove(struct platform_device *pdev) static struct platform_driver sst_platform_driver = { .driver = { - .name = "sst-platform", + .name = "sst-mfld-platform", .owner = THIS_MODULE, }, .probe = sst_platform_probe, @@ -722,4 +722,4 @@ MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver"); MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); MODULE_AUTHOR("Harsha Priya <priya.harsha(a)intel.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:sst-platform"); +MODULE_ALIAS("platform:sst-mfld-platform"); diff --git a/sound/soc/intel/sst_platform.h b/sound/soc/intel/sst-mfld-platform.h similarity index 97% rename from sound/soc/intel/sst_platform.h rename to sound/soc/intel/sst-mfld-platform.h index bee64fb..0c4e2dd 100644 --- a/sound/soc/intel/sst_platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -1,5 +1,5 @@ /* - * sst_platform.h - Intel MID Platform driver header file + * sst_mfld_platform.h - Intel MID Platform driver header file * * Copyright (C) 2010 Intel Corp * Author: Vinod Koul <vinod.koul(a)intel.com> @@ -27,7 +27,7 @@ #ifndef __SST_PLATFORMDRV_H__ #define __SST_PLATFORMDRV_H__ -#include "sst_dsp.h" +#include "sst-mfld-dsp.h" #define SST_MONO 1 #define SST_STEREO 2 -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: Intel: Add a mfld prefix to Intel SST drivers.
by Liam Girdwood 12 Feb '14

12 Feb '14

In order to differentiate the different Intel SST audio core drivers we need to rename the current drivers with a mfld prefix. This also includes renaming in the Makefile and Kconfig Acked-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/Kconfig | 4 ++-- sound/soc/intel/Makefile | 4 ++-- sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} | 8 ++++---- sound/soc/intel/{sst_platform.c => sst-mfld-platform.c} | 8 ++++---- sound/soc/intel/{sst_platform.h => sst-mfld-platform.h} | 4 ++-- 5 files changed, 14 insertions(+), 14 deletions(-) rename sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} (96%) rename sound/soc/intel/{sst_platform.c => sst-mfld-platform.c} (99%) rename sound/soc/intel/{sst_platform.h => sst-mfld-platform.h} (97%) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 61c10bf..4d9d0a5 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -2,12 +2,12 @@ config SND_MFLD_MACHINE tristate "SOC Machine Audio driver for Intel Medfield MID platform" depends on INTEL_SCU_IPC select SND_SOC_SN95031 - select SND_SST_PLATFORM + select SND_SST_MFLD_PLATFORM help This adds support for ASoC machine driver for Intel(R) MID Medfield platform used as alsa device in audio substem in Intel(R) MID devices Say Y if you have such a device If unsure select "N". -config SND_SST_PLATFORM +config SND_SST_MFLD_PLATFORM tristate diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 6398833..eb899fc 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -1,5 +1,5 @@ -snd-soc-sst-platform-objs := sst_platform.o +snd-soc-sst-mfld-platform-objs := sst-mfld-platform.o snd-soc-mfld-machine-objs := mfld_machine.o -obj-$(CONFIG_SND_SST_PLATFORM) += snd-soc-sst-platform.o +obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o diff --git a/sound/soc/intel/sst_dsp.h b/sound/soc/intel/sst-mfld-dsp.h similarity index 96% rename from sound/soc/intel/sst_dsp.h rename to sound/soc/intel/sst-mfld-dsp.h index 0fce1de..3b63edc 100644 --- a/sound/soc/intel/sst_dsp.h +++ b/sound/soc/intel/sst-mfld-dsp.h @@ -1,7 +1,7 @@ -#ifndef __SST_DSP_H__ -#define __SST_DSP_H__ +#ifndef __SST_MFLD_DSP_H__ +#define __SST_MFLD_DSP_H__ /* - * sst_dsp.h - Intel SST Driver for audio engine + * sst_mfld_dsp.h - Intel SST Driver for audio engine * * Copyright (C) 2008-12 Intel Corporation * Authors: Vinod Koul <vinod.koul(a)linux.intel.com> @@ -131,4 +131,4 @@ struct snd_sst_params { struct snd_sst_alloc_params_ext aparams; }; -#endif /* __SST_DSP_H__ */ +#endif /* __SST_MFLD_DSP_H__ */ diff --git a/sound/soc/intel/sst_platform.c b/sound/soc/intel/sst-mfld-platform.c similarity index 99% rename from sound/soc/intel/sst_platform.c rename to sound/soc/intel/sst-mfld-platform.c index f465a81..840306c 100644 --- a/sound/soc/intel/sst_platform.c +++ b/sound/soc/intel/sst-mfld-platform.c @@ -1,5 +1,5 @@ /* - * sst_platform.c - Intel MID Platform driver + * sst_mfld_platform.c - Intel MID Platform driver * * Copyright (C) 2010-2013 Intel Corp * Author: Vinod Koul <vinod.koul(a)intel.com> @@ -33,7 +33,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/compress_driver.h> -#include "sst_platform.h" +#include "sst-mfld-platform.h" static struct sst_device *sst; static DEFINE_MUTEX(sst_lock); @@ -709,7 +709,7 @@ static int sst_platform_remove(struct platform_device *pdev) static struct platform_driver sst_platform_driver = { .driver = { - .name = "sst-platform", + .name = "sst-mfld-platform", .owner = THIS_MODULE, }, .probe = sst_platform_probe, @@ -722,4 +722,4 @@ MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver"); MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); MODULE_AUTHOR("Harsha Priya <priya.harsha(a)intel.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:sst-platform"); +MODULE_ALIAS("platform:sst-mfld-platform"); diff --git a/sound/soc/intel/sst_platform.h b/sound/soc/intel/sst-mfld-platform.h similarity index 97% rename from sound/soc/intel/sst_platform.h rename to sound/soc/intel/sst-mfld-platform.h index bee64fb..0c4e2dd 100644 --- a/sound/soc/intel/sst_platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -1,5 +1,5 @@ /* - * sst_platform.h - Intel MID Platform driver header file + * sst_mfld_platform.h - Intel MID Platform driver header file * * Copyright (C) 2010 Intel Corp * Author: Vinod Koul <vinod.koul(a)intel.com> @@ -27,7 +27,7 @@ #ifndef __SST_PLATFORMDRV_H__ #define __SST_PLATFORMDRV_H__ -#include "sst_dsp.h" +#include "sst-mfld-dsp.h" #define SST_MONO 1 #define SST_STEREO 2 -- 1.8.3.2

1 0

[alsa-devel] [PATCH] ASoC: rsnd: print error if there is SRC settings mismatch
by Kuninori Morimoto 12 Feb '14

12 Feb '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd request clock master if SRC is used Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/scu.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 6e5c763..3984d4b 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -122,6 +122,14 @@ static int rsnd_scu_ssi_mode_init(struct rsnd_mod *mod, struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct rsnd_scu *scu = rsnd_mod_to_scu(mod); int id = rsnd_mod_id(mod); + u32 convert_rate = rsnd_scu_convert_rate(scu); + + if (convert_rate && !rsnd_dai_is_clk_master(rdai)) { + struct device *dev = rsnd_priv_to_dev(priv); + + dev_err(dev, "rsnd should be clk master when you rate convert\n"); + return -EINVAL; + } /* * SSI_MODE0 -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: rsnd: set DIV_EN register on rsnd_adg_set_convert_clk_gen2()
by Kuninori Morimoto 12 Feb '14

12 Feb '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> DIV_EN register enable bit is required when you use Gen2 SRC Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 16 +++++++++++++--- sound/soc/sh/rcar/gen.c | 1 + sound/soc/sh/rcar/rsnd.h | 1 + 3 files changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 821791e..fc1dce6 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -110,8 +110,8 @@ int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod, struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct rsnd_adg *adg = rsnd_priv_to_adg(priv); struct device *dev = rsnd_priv_to_dev(priv); - int idx, sel, div, step; - u32 val; + int idx, sel, div, step, ret; + u32 val, en; unsigned int min, diff; unsigned int sel_rate [] = { clk_get_rate(adg->clk[CLKA]), /* 0000: CLKA */ @@ -123,6 +123,7 @@ int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod, min = ~0; val = 0; + en = 0; for (sel = 0; sel < ARRAY_SIZE(sel_rate); sel++) { idx = 0; step = 2; @@ -135,6 +136,7 @@ int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod, if (min > diff) { val = (sel << 8) | idx; min = diff; + en = 1 << (sel + 1); /* fixme */ } /* @@ -156,7 +158,15 @@ int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod, return -EIO; } - return rsnd_adg_set_src_timsel_gen2(rdai, mod, io, val); + ret = rsnd_adg_set_src_timsel_gen2(rdai, mod, io, val); + if (ret < 0) { + dev_err(dev, "timsel error\n"); + return ret; + } + + rsnd_mod_bset(mod, DIV_EN, en, en); + + return 0; } int rsnd_adg_set_convert_timing_gen2(struct rsnd_mod *mod, diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 3e03a8b..0a43b90 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -253,6 +253,7 @@ static int rsnd_gen2_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL0, 0x0c), RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL1, 0x10), RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL2, 0x14), + RSND_GEN2_S_REG(gen, ADG, DIV_EN, 0x30), RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL0, 0x34), RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL1, 0x38), RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL2, 0x3c), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 8b66dc1..5691bfc 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -67,6 +67,7 @@ enum rsnd_reg { RSND_REG_AUDIO_CLK_SEL3, /* for Gen1 */ RSND_REG_AUDIO_CLK_SEL4, /* for Gen1 */ RSND_REG_AUDIO_CLK_SEL5, /* for Gen1 */ + RSND_REG_DIV_EN, /* for Gen2 */ RSND_REG_SRCIN_TIMSEL0, /* for Gen2 */ RSND_REG_SRCIN_TIMSEL1, /* for Gen2 */ RSND_REG_SRCIN_TIMSEL2, /* for Gen2 */ -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: rsnd: tidyup original for_each_rsnd_xxx macro
by Kuninori Morimoto 12 Feb '14

12 Feb '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current for_each_rsnd_xxx macro will read out-of-array's memory after last loop operation. It was not good C language operation, and the binary which was compiled by (at least) gcc 4.8.1 is broken This patch tidyup these issues Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 7 ++++--- sound/soc/sh/rcar/rsnd.h | 7 ++++--- 2 files changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 8d3a82e..bc8961c 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -25,9 +25,10 @@ struct rsnd_adg { }; #define for_each_rsnd_clk(pos, adg, i) \ - for (i = 0, (pos) = adg->clk[i]; \ - i < CLKMAX; \ - i++, (pos) = adg->clk[i]) + for (i = 0; \ + (i < CLKMAX) && \ + ((pos) = adg->clk[i]); \ + i++) #define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 8b66dc1..9e4efb4 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -216,9 +216,10 @@ struct rsnd_dai { #define rsnd_dai_nr(priv) ((priv)->dai_nr) #define for_each_rsnd_dai(rdai, priv, i) \ - for (i = 0, (rdai) = rsnd_dai_get(priv, i); \ - i < rsnd_dai_nr(priv); \ - i++, (rdai) = rsnd_dai_get(priv, i)) + for (i = 0; \ + (i < rsnd_dai_nr(priv)) && \ + ((rdai) = rsnd_dai_get(priv, i)); \ + i++) struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id); int rsnd_dai_disconnect(struct rsnd_mod *mod); -- 1.7.9.5

2 1

[alsa-devel] [PATCH][RFC] alsa-lib: heavy pcm atomics cleanup
by riku.voipio＠linaro.org 12 Feb '14

12 Feb '14

From: Riku Voipio <riku.voipio(a)linaro.org> The following patch comes from the realization that at least ARM code for atomics is quite broken and nobody has cared for a decade. A quick dive shows that only snd_atomic_{read,write}_{begin,end} appear to be used widely. These are implemented using wmb/rmb. Only other use of atomic functions is in pcm_meter.c. The #SND_PCM_TYPE_METER plugin type appears rarely, if ever, used. I presume these days anyone who wants a meter/scope will do in pulseaudio layer instead of alsa. It would seem better fit to have pcm_meter in alsa-plugins instead of alsa-lib, but I guess that would be an ABI break... So instead, I'm proposing here 1. Removal of all hand-crafted atomics from iatomic.h apart from barriers, which are used in snd_atomic_{read,write}_{begin,end}. 2. Using __sync_synchronize as the default fallback for barriers. This has been available since gcc 4.1, so it shouldn't be a problem. 3. Defining the few atomics used by pcm_meter.c withing pcm_meter.c itself, using gcc atomic builtins[1]. 4. Since gcc atomic builtins are available only since gcc 4.7, add a check for that in gcc configure.in, and don't build pcm meter plugin if using older gcc. The last point has the impact, that if there actually is someone who 1) uses the meter plugin 2) wants to upgrade to 2014 alsa-lib 3) but does not want to use a 2012+ gcc - that someone will be inconvenienced. Finally remove the unneeded configure check for cpu type. We can trust the gcc to set right flags for us. [1] http://gcc.gnu.org/onlinedocs/gcc/_005f_005fatomic-Builtins.html Signed-off-by: Riku Voipio <riku.voipio(a)linaro.org> --- configure.in | 53 +-- include/iatomic.h | 1044 +-------------------------------------------------- src/pcm/pcm_meter.c | 9 +- 3 files changed, 33 insertions(+), 1073 deletions(-) diff --git a/configure.in b/configure.in index 35fd89f..bb56eb6 100644 --- a/configure.in +++ b/configure.in @@ -309,40 +309,6 @@ fi AC_SUBST(ALSA_DEPLIBS) -dnl Check for architecture -AC_MSG_CHECKING(for architecture) -case "$host" in -i?86*) - AC_MSG_RESULT(x86) - ;; -x86_64*) - AC_MSG_RESULT(x86) - ;; -alpha*) - AC_MSG_RESULT(alpha) - ;; -powerpc*|ppc*) - AC_MSG_RESULT(ppc) - CPPFLAGS="$CPPFLAGS -D__ppc__" - ;; -ia64*) - AC_MSG_RESULT(ia64) - CPPFLAGS="$CPPFLAGS -D__ia64__" - ;; -mips*) - AC_MSG_RESULT(mips) - CPPFLAGS="$CPPFLAGS -D__mips__" - ;; -arm*) - AC_MSG_RESULT(arm) - CPPFLAGS="$CPPFLAGS -D__arm__" - ;; -*) - AC_MSG_RESULT($host_cpu) - echo "No atomic operations supported.." - ;; -esac - dnl Check for wordexp.h AC_CHECK_HEADERS([wordexp.h]) @@ -488,6 +454,21 @@ else pcm_plugins="" fi +dnl check atomics for pcm_meter + +AC_MSG_CHECKING([whether GCC supports builtin atomic intrinsics]) +if test -z "$gcc_have_atomics"; then + gcc_have_atomics=no + AC_TRY_LINK([], + [int i; + __atomic_load_n(&i, __ATOMIC_SEQ_CST); + __atomic_add_fetch(&i, 0, __ATOMIC_SEQ_CST); + ], + [gcc_have_atomics=yes], + [gcc_have_atomics=no]) +fi +AC_MSG_RESULT($gcc_have_atomics) + PCM_PLUGIN_LIST="copy linear route mulaw alaw adpcm rate plug multi shm file null empty share meter hooks lfloat ladspa dmix dshare dsnoop asym iec958 softvol extplug ioplug mmap_emul" build_pcm_plugin="no" @@ -531,6 +512,10 @@ if test "$softfloat" = "yes"; then build_pcm_ladspa="no" fi +if test "$gcc_have_atomics" != "yes"; then + build_pcm_meter="no" +fi + AM_CONDITIONAL([BUILD_PCM_PLUGIN], [test x$build_pcm_plugin = xyes]) AM_CONDITIONAL([BUILD_PCM_PLUGIN_COPY], [test x$build_pcm_copy = xyes]) AM_CONDITIONAL([BUILD_PCM_PLUGIN_LINEAR], [test x$build_pcm_linear = xyes]) diff --git a/include/iatomic.h b/include/iatomic.h index 2393297..acdd3e2 100644 --- a/include/iatomic.h +++ b/include/iatomic.h @@ -1,349 +1,21 @@ #ifndef __ALSA_IATOMIC_H #define __ALSA_IATOMIC_H -#if defined(__i386__) || defined(__x86_64__) - -/* - * Atomic operations that C can't guarantee us. Useful for - * resource counting etc.. - */ - -#define ATOMIC_SMP_LOCK "lock ; " - -/* - * Make sure gcc doesn't try to be clever and move things around - * on us. We need to use _exactly_ the address the user gave us, - * not some alias that contains the same information. - */ -typedef struct { volatile int counter; } atomic_t; - -#define ATOMIC_INIT(i) { (i) } - -/** - * atomic_read - read atomic variable - * @v: pointer of type atomic_t - * - * Atomically reads the value of @v. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -#define atomic_read(v) ((v)->counter) - -/** - * atomic_set - set atomic variable - * @v: pointer of type atomic_t - * @i: required value - * - * Atomically sets the value of @v to @i. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -#define atomic_set(v,i) (((v)->counter) = (i)) - -/** - * atomic_add - add integer to atomic variable - * @i: integer value to add - * @v: pointer of type atomic_t - * - * Atomically adds @i to @v. Note that the guaranteed useful range - * of an atomic_t is only 24 bits. - */ -static __inline__ void atomic_add(int i, atomic_t *v) -{ - __asm__ __volatile__( - ATOMIC_SMP_LOCK "addl %1,%0" - :"=m" (v->counter) - :"ir" (i), "m" (v->counter)); -} - -/** - * atomic_sub - subtract the atomic variable - * @i: integer value to subtract - * @v: pointer of type atomic_t - * - * Atomically subtracts @i from @v. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -static __inline__ void atomic_sub(int i, atomic_t *v) -{ - __asm__ __volatile__( - ATOMIC_SMP_LOCK "subl %1,%0" - :"=m" (v->counter) - :"ir" (i), "m" (v->counter)); -} - -/** - * atomic_sub_and_test - subtract value from variable and test result - * @i: integer value to subtract - * @v: pointer of type atomic_t - * - * Atomically subtracts @i from @v and returns - * true if the result is zero, or false for all - * other cases. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -static __inline__ int atomic_sub_and_test(int i, atomic_t *v) -{ - unsigned char c; - - __asm__ __volatile__( - ATOMIC_SMP_LOCK "subl %2,%0; sete %1" - :"=m" (v->counter), "=qm" (c) - :"ir" (i), "m" (v->counter) : "memory"); - return c; -} - -/** - * atomic_inc - increment atomic variable - * @v: pointer of type atomic_t - * - * Atomically increments @v by 1. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -static __inline__ void atomic_inc(atomic_t *v) -{ - __asm__ __volatile__( - ATOMIC_SMP_LOCK "incl %0" - :"=m" (v->counter) - :"m" (v->counter)); -} - -/** - * atomic_dec - decrement atomic variable - * @v: pointer of type atomic_t - * - * Atomically decrements @v by 1. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -static __inline__ void atomic_dec(atomic_t *v) -{ - __asm__ __volatile__( - ATOMIC_SMP_LOCK "decl %0" - :"=m" (v->counter) - :"m" (v->counter)); -} - -/** - * atomic_dec_and_test - decrement and test - * @v: pointer of type atomic_t - * - * Atomically decrements @v by 1 and - * returns true if the result is 0, or false for all other - * cases. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -static __inline__ int atomic_dec_and_test(atomic_t *v) -{ - unsigned char c; - - __asm__ __volatile__( - ATOMIC_SMP_LOCK "decl %0; sete %1" - :"=m" (v->counter), "=qm" (c) - :"m" (v->counter) : "memory"); - return c != 0; -} - -/** - * atomic_inc_and_test - increment and test - * @v: pointer of type atomic_t - * - * Atomically increments @v by 1 - * and returns true if the result is zero, or false for all - * other cases. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -static __inline__ int atomic_inc_and_test(atomic_t *v) -{ - unsigned char c; - - __asm__ __volatile__( - ATOMIC_SMP_LOCK "incl %0; sete %1" - :"=m" (v->counter), "=qm" (c) - :"m" (v->counter) : "memory"); - return c != 0; -} - -/** - * atomic_add_negative - add and test if negative - * @v: pointer of type atomic_t - * @i: integer value to add - * - * Atomically adds @i to @v and returns true - * if the result is negative, or false when - * result is greater than or equal to zero. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -static __inline__ int atomic_add_negative(int i, atomic_t *v) -{ - unsigned char c; - - __asm__ __volatile__( - ATOMIC_SMP_LOCK "addl %2,%0; sets %1" - :"=m" (v->counter), "=qm" (c) - :"ir" (i), "m" (v->counter) : "memory"); - return c; -} - -/* These are x86-specific, used by some header files */ -#define atomic_clear_mask(mask, addr) \ -__asm__ __volatile__(ATOMIC_SMP_LOCK "andl %0,%1" \ -: : "r" (~(mask)),"m" (*addr) : "memory") - -#define atomic_set_mask(mask, addr) \ -__asm__ __volatile__(ATOMIC_SMP_LOCK "orl %0,%1" \ -: : "r" (mask),"m" (*addr) : "memory") - -/* - * Force strict CPU ordering. - * And yes, this is required on UP too when we're talking - * to devices. - * - * For now, "wmb()" doesn't actually do anything, as all - * Intel CPU's follow what Intel calls a *Processor Order*, - * in which all writes are seen in the program order even - * outside the CPU. - * - * I expect future Intel CPU's to have a weaker ordering, - * but I'd also expect them to finally get their act together - * and add some real memory barriers if so. - */ - #ifdef __i386__ #define mb() __asm__ __volatile__ ("lock; addl $0,0(%%esp)": : :"memory") #define rmb() mb() #define wmb() __asm__ __volatile__ ("": : :"memory") -#else +#define IATOMIC_DEFINED 1 +#endif + +#ifdef __x86_64__ #define mb() asm volatile("mfence":::"memory") #define rmb() asm volatile("lfence":::"memory") #define wmb() asm volatile("sfence":::"memory") -#endif - -#undef ATOMIC_SMP_LOCK - #define IATOMIC_DEFINED 1 - -#endif /* __i386__ */ +#endif #ifdef __ia64__ - -/* - * On IA-64, counter must always be volatile to ensure that that the - * memory accesses are ordered. - */ -typedef struct { volatile int counter; } atomic_t; - -#define ATOMIC_INIT(i) ((atomic_t) { (i) }) - -#define atomic_read(v) ((v)->counter) -#define atomic_set(v,i) (((v)->counter) = (i)) - -/* stripped version - we need only 4byte version */ -#define ia64_cmpxchg(sem,ptr,old,new,size) \ -({ \ - __typeof__(ptr) _p_ = (ptr); \ - __typeof__(new) _n_ = (new); \ - unsigned long _o_, _r_; \ - _o_ = (unsigned int) (long) (old); \ - __asm__ __volatile__ ("mov ar.ccv=%0;;" :: "rO"(_o_)); \ - __asm__ __volatile__ ("cmpxchg4."sem" %0=[%1],%2,ar.ccv" \ - : "=r"(_r_) : "r"(_p_), "r"(_n_) : "memory"); \ - (__typeof__(old)) _r_; \ -}) - -static __inline__ int -ia64_atomic_add (int i, atomic_t *v) -{ - int old, new; - // CMPXCHG_BUGCHECK_DECL - - do { - // CMPXCHG_BUGCHECK(v); - old = atomic_read(v); - new = old + i; - } while (ia64_cmpxchg("acq", v, old, old + i, sizeof(atomic_t)) != old); - return new; -} - -static __inline__ int -ia64_atomic_sub (int i, atomic_t *v) -{ - int old, new; - // CMPXCHG_BUGCHECK_DECL - - do { - // CMPXCHG_BUGCHECK(v); - old = atomic_read(v); - new = old - i; - } while (ia64_cmpxchg("acq", v, old, new, sizeof(atomic_t)) != old); - return new; -} - -#define IA64_FETCHADD(tmp,v,n,sz) \ -({ \ - switch (sz) { \ - case 4: \ - __asm__ __volatile__ ("fetchadd4.rel %0=[%1],%2" \ - : "=r"(tmp) : "r"(v), "i"(n) : "memory"); \ - break; \ - \ - case 8: \ - __asm__ __volatile__ ("fetchadd8.rel %0=[%1],%2" \ - : "=r"(tmp) : "r"(v), "i"(n) : "memory"); \ - break; \ - } \ -}) - -#define ia64_fetch_and_add(i,v) \ -({ \ - unsigned long _tmp; \ - volatile __typeof__(*(v)) *_v = (v); \ - switch (i) { \ - case -16: IA64_FETCHADD(_tmp, _v, -16, sizeof(*(v))); break; \ - case -8: IA64_FETCHADD(_tmp, _v, -8, sizeof(*(v))); break; \ - case -4: IA64_FETCHADD(_tmp, _v, -4, sizeof(*(v))); break; \ - case -1: IA64_FETCHADD(_tmp, _v, -1, sizeof(*(v))); break; \ - case 1: IA64_FETCHADD(_tmp, _v, 1, sizeof(*(v))); break; \ - case 4: IA64_FETCHADD(_tmp, _v, 4, sizeof(*(v))); break; \ - case 8: IA64_FETCHADD(_tmp, _v, 8, sizeof(*(v))); break; \ - case 16: IA64_FETCHADD(_tmp, _v, 16, sizeof(*(v))); break; \ - } \ - (__typeof__(*v)) (_tmp + (i)); /* return new value */ \ -}) - -/* - * Atomically add I to V and return TRUE if the resulting value is - * negative. - */ -static __inline__ int -atomic_add_negative (int i, atomic_t *v) -{ - return ia64_atomic_add(i, v) < 0; -} - -#define atomic_add_return(i,v) \ - ((__builtin_constant_p(i) && \ - ( (i == 1) || (i == 4) || (i == 8) || (i == 16) \ - || (i == -1) || (i == -4) || (i == -8) || (i == -16))) \ - ? ia64_fetch_and_add(i, &(v)->counter) \ - : ia64_atomic_add(i, v)) - -#define atomic_sub_return(i,v) \ - ((__builtin_constant_p(i) && \ - ( (i == 1) || (i == 4) || (i == 8) || (i == 16) \ - || (i == -1) || (i == -4) || (i == -8) || (i == -16))) \ - ? ia64_fetch_and_add(-(i), &(v)->counter) \ - : ia64_atomic_sub(i, v)) - -#define atomic_dec_return(v) atomic_sub_return(1, (v)) -#define atomic_inc_return(v) atomic_add_return(1, (v)) - -#define atomic_sub_and_test(i,v) (atomic_sub_return((i), (v)) == 0) -#define atomic_dec_and_test(v) (atomic_sub_return(1, (v)) == 0) -#define atomic_inc_and_test(v) (atomic_add_return(1, (v)) != 0) - -#define atomic_add(i,v) atomic_add_return((i), (v)) -#define atomic_sub(i,v) atomic_sub_return((i), (v)) -#define atomic_inc(v) atomic_add(1, (v)) -#define atomic_dec(v) atomic_sub(1, (v)) - /* * Macros to force memory ordering. In these descriptions, "previous" * and "subsequent" refer to program order; "visible" means that all @@ -376,111 +48,6 @@ atomic_add_negative (int i, atomic_t *v) #ifdef __alpha__ -/* - * Atomic operations that C can't guarantee us. Useful for - * resource counting etc... - * - * But use these as seldom as possible since they are much slower - * than regular operations. - */ - - -/* - * Counter is volatile to make sure gcc doesn't try to be clever - * and move things around on us. We need to use _exactly_ the address - * the user gave us, not some alias that contains the same information. - */ -typedef struct { volatile int counter; } atomic_t; - -#define ATOMIC_INIT(i) ( (atomic_t) { (i) } ) - -#define atomic_read(v) ((v)->counter) -#define atomic_set(v,i) ((v)->counter = (i)) - -/* - * To get proper branch prediction for the main line, we must branch - * forward to code at the end of this object's .text section, then - * branch back to restart the operation. - */ - -static __inline__ void atomic_add(int i, atomic_t * v) -{ - unsigned long temp; - __asm__ __volatile__( - "1: ldl_l %0,%1\n" - " addl %0,%2,%0\n" - " stl_c %0,%1\n" - " beq %0,2f\n" - ".subsection 2\n" - "2: br 1b\n" - ".previous" - :"=&r" (temp), "=m" (v->counter) - :"Ir" (i), "m" (v->counter)); -} - -static __inline__ void atomic_sub(int i, atomic_t * v) -{ - unsigned long temp; - __asm__ __volatile__( - "1: ldl_l %0,%1\n" - " subl %0,%2,%0\n" - " stl_c %0,%1\n" - " beq %0,2f\n" - ".subsection 2\n" - "2: br 1b\n" - ".previous" - :"=&r" (temp), "=m" (v->counter) - :"Ir" (i), "m" (v->counter)); -} - -/* - * Same as above, but return the result value - */ -static __inline__ long atomic_add_return(int i, atomic_t * v) -{ - long temp, result; - __asm__ __volatile__( - "1: ldl_l %0,%1\n" - " addl %0,%3,%2\n" - " addl %0,%3,%0\n" - " stl_c %0,%1\n" - " beq %0,2f\n" - " mb\n" - ".subsection 2\n" - "2: br 1b\n" - ".previous" - :"=&r" (temp), "=m" (v->counter), "=&r" (result) - :"Ir" (i), "m" (v->counter) : "memory"); - return result; -} - -static __inline__ long atomic_sub_return(int i, atomic_t * v) -{ - long temp, result; - __asm__ __volatile__( - "1: ldl_l %0,%1\n" - " subl %0,%3,%2\n" - " subl %0,%3,%0\n" - " stl_c %0,%1\n" - " beq %0,2f\n" - " mb\n" - ".subsection 2\n" - "2: br 1b\n" - ".previous" - :"=&r" (temp), "=m" (v->counter), "=&r" (result) - :"Ir" (i), "m" (v->counter) : "memory"); - return result; -} - -#define atomic_dec_return(v) atomic_sub_return(1,(v)) -#define atomic_inc_return(v) atomic_add_return(1,(v)) - -#define atomic_sub_and_test(i,v) (atomic_sub_return((i), (v)) == 0) -#define atomic_dec_and_test(v) (atomic_sub_return(1, (v)) == 0) - -#define atomic_inc(v) atomic_add(1,(v)) -#define atomic_dec(v) atomic_sub(1,(v)) - #define mb() \ __asm__ __volatile__("mb": : :"memory") @@ -496,168 +63,6 @@ __asm__ __volatile__("wmb": : :"memory") #ifdef __powerpc__ -typedef struct { volatile int counter; } atomic_t; - -#define ATOMIC_INIT(i) { (i) } - -#define atomic_read(v) ((v)->counter) -#define atomic_set(v,i) (((v)->counter) = (i)) - -extern void atomic_clear_mask(unsigned long mask, unsigned long *addr); -extern void atomic_set_mask(unsigned long mask, unsigned long *addr); - -#define SMP_ISYNC "\n\tisync" - -static __inline__ void atomic_add(int a, atomic_t *v) -{ - int t; - - __asm__ __volatile__( -"1: lwarx %0,0,%3 # atomic_add\n\ - add %0,%2,%0\n\ - stwcx. %0,0,%3\n\ - bne- 1b" - : "=&r" (t), "=m" (v->counter) - : "r" (a), "r" (&v->counter), "m" (v->counter) - : "cc"); -} - -static __inline__ int atomic_add_return(int a, atomic_t *v) -{ - int t; - - __asm__ __volatile__( -"1: lwarx %0,0,%2 # atomic_add_return\n\ - add %0,%1,%0\n\ - stwcx. %0,0,%2\n\ - bne- 1b" - SMP_ISYNC - : "=&r" (t) - : "r" (a), "r" (&v->counter) - : "cc", "memory"); - - return t; -} - -static __inline__ void atomic_sub(int a, atomic_t *v) -{ - int t; - - __asm__ __volatile__( -"1: lwarx %0,0,%3 # atomic_sub\n\ - subf %0,%2,%0\n\ - stwcx. %0,0,%3\n\ - bne- 1b" - : "=&r" (t), "=m" (v->counter) - : "r" (a), "r" (&v->counter), "m" (v->counter) - : "cc"); -} - -static __inline__ int atomic_sub_return(int a, atomic_t *v) -{ - int t; - - __asm__ __volatile__( -"1: lwarx %0,0,%2 # atomic_sub_return\n\ - subf %0,%1,%0\n\ - stwcx. %0,0,%2\n\ - bne- 1b" - SMP_ISYNC - : "=&r" (t) - : "r" (a), "r" (&v->counter) - : "cc", "memory"); - - return t; -} - -static __inline__ void atomic_inc(atomic_t *v) -{ - int t; - - __asm__ __volatile__( -"1: lwarx %0,0,%2 # atomic_inc\n\ - addic %0,%0,1\n\ - stwcx. %0,0,%2\n\ - bne- 1b" - : "=&r" (t), "=m" (v->counter) - : "r" (&v->counter), "m" (v->counter) - : "cc"); -} - -static __inline__ int atomic_inc_return(atomic_t *v) -{ - int t; - - __asm__ __volatile__( -"1: lwarx %0,0,%1 # atomic_inc_return\n\ - addic %0,%0,1\n\ - stwcx. %0,0,%1\n\ - bne- 1b" - SMP_ISYNC - : "=&r" (t) - : "r" (&v->counter) - : "cc", "memory"); - - return t; -} - -static __inline__ void atomic_dec(atomic_t *v) -{ - int t; - - __asm__ __volatile__( -"1: lwarx %0,0,%2 # atomic_dec\n\ - addic %0,%0,-1\n\ - stwcx. %0,0,%2\n\ - bne- 1b" - : "=&r" (t), "=m" (v->counter) - : "r" (&v->counter), "m" (v->counter) - : "cc"); -} - -static __inline__ int atomic_dec_return(atomic_t *v) -{ - int t; - - __asm__ __volatile__( -"1: lwarx %0,0,%1 # atomic_dec_return\n\ - addic %0,%0,-1\n\ - stwcx. %0,0,%1\n\ - bne- 1b" - SMP_ISYNC - : "=&r" (t) - : "r" (&v->counter) - : "cc", "memory"); - - return t; -} - -#define atomic_sub_and_test(a, v) (atomic_sub_return((a), (v)) == 0) -#define atomic_dec_and_test(v) (atomic_dec_return((v)) == 0) - -/* - * Atomically test *v and decrement if it is greater than 0. - * The function returns the old value of *v minus 1. - */ -static __inline__ int atomic_dec_if_positive(atomic_t *v) -{ - int t; - - __asm__ __volatile__( -"1: lwarx %0,0,%1 # atomic_dec_if_positive\n\ - addic. %0,%0,-1\n\ - blt- 2f\n\ - stwcx. %0,0,%1\n\ - bne- 1b" - SMP_ISYNC - "\n\ -2:" : "=&r" (t) - : "r" (&v->counter) - : "cc", "memory"); - - return t; -} - /* * Memory barrier. * The sync instruction guarantees that all memory accesses initiated @@ -682,447 +87,14 @@ static __inline__ int atomic_dec_if_positive(atomic_t *v) #endif /* __powerpc__ */ -#ifdef __mips__ - -typedef struct { volatile int counter; } atomic_t; - -#define ATOMIC_INIT(i) { (i) } - -/* - * atomic_read - read atomic variable - * @v: pointer of type atomic_t - * - * Atomically reads the value of @v. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -#define atomic_read(v) ((v)->counter) - -/* - * atomic_set - set atomic variable - * @v: pointer of type atomic_t - * @i: required value - * - * Atomically sets the value of @v to @i. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -#define atomic_set(v,i) ((v)->counter = (i)) - -/* - * for MIPS II and better we can use ll/sc instruction, and kernel 2.4.3+ - * will emulate it on MIPS I. - */ - -/* - * atomic_add - add integer to atomic variable - * @i: integer value to add - * @v: pointer of type atomic_t - * - * Atomically adds @i to @v. Note that the guaranteed useful range - * of an atomic_t is only 24 bits. - */ -static __inline__ void atomic_add(int i, atomic_t * v) -{ - unsigned long temp; - - __asm__ __volatile__( - ".set push \n" - ".set mips2 \n" - "1: ll %0, %1 # atomic_add\n" - " addu %0, %2 \n" - " sc %0, %1 \n" - " beqz %0, 1b \n" - ".set pop \n" - : "=&r" (temp), "=m" (v->counter) - : "Ir" (i), "m" (v->counter)); -} - -/* - * atomic_sub - subtract the atomic variable - * @i: integer value to subtract - * @v: pointer of type atomic_t - * - * Atomically subtracts @i from @v. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -static __inline__ void atomic_sub(int i, atomic_t * v) -{ - unsigned long temp; - - __asm__ __volatile__( - ".set push \n" - ".set mips2 \n" - "1: ll %0, %1 # atomic_sub\n" - " subu %0, %2 \n" - " sc %0, %1 \n" - " beqz %0, 1b \n" - ".set pop \n" - : "=&r" (temp), "=m" (v->counter) - : "Ir" (i), "m" (v->counter)); -} - -/* - * Same as above, but return the result value - */ -static __inline__ int atomic_add_return(int i, atomic_t * v) -{ - unsigned long temp, result; - - __asm__ __volatile__( - ".set push # atomic_add_return\n" - ".set noreorder \n" - ".set mips2 \n" - "1: ll %1, %2 \n" - " addu %0, %1, %3 \n" - " sc %0, %2 \n" - " beqz %0, 1b \n" - " addu %0, %1, %3 \n" - ".set pop \n" - : "=&r" (result), "=&r" (temp), "=m" (v->counter) - : "Ir" (i), "m" (v->counter) - : "memory"); - - return result; -} - -static __inline__ int atomic_sub_return(int i, atomic_t * v) -{ - unsigned long temp, result; - - __asm__ __volatile__( - ".set push \n" - ".set mips2 \n" - ".set noreorder # atomic_sub_return\n" - "1: ll %1, %2 \n" - " subu %0, %1, %3 \n" - " sc %0, %2 \n" - " beqz %0, 1b \n" - " subu %0, %1, %3 \n" - ".set pop \n" - : "=&r" (result), "=&r" (temp), "=m" (v->counter) - : "Ir" (i), "m" (v->counter) - : "memory"); - - return result; -} - -#define atomic_dec_return(v) atomic_sub_return(1,(v)) -#define atomic_inc_return(v) atomic_add_return(1,(v)) - -/* - * atomic_sub_and_test - subtract value from variable and test result - * @i: integer value to subtract - * @v: pointer of type atomic_t - * - * Atomically subtracts @i from @v and returns - * true if the result is zero, or false for all - * other cases. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -#define atomic_sub_and_test(i,v) (atomic_sub_return((i), (v)) == 0) - -/* - * atomic_inc_and_test - increment and test - * @v: pointer of type atomic_t - * - * Atomically increments @v by 1 - * and returns true if the result is zero, or false for all - * other cases. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -#define atomic_inc_and_test(v) (atomic_inc_return(1, (v)) == 0) - -/* - * atomic_dec_and_test - decrement by 1 and test - * @v: pointer of type atomic_t - * - * Atomically decrements @v by 1 and - * returns true if the result is 0, or false for all other - * cases. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -#define atomic_dec_and_test(v) (atomic_sub_return(1, (v)) == 0) - -/* - * atomic_inc - increment atomic variable - * @v: pointer of type atomic_t - * - * Atomically increments @v by 1. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -#define atomic_inc(v) atomic_add(1,(v)) - -/* - * atomic_dec - decrement and test - * @v: pointer of type atomic_t - * - * Atomically decrements @v by 1. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - */ -#define atomic_dec(v) atomic_sub(1,(v)) - -/* - * atomic_add_negative - add and test if negative - * @v: pointer of type atomic_t - * @i: integer value to add - * - * Atomically adds @i to @v and returns true - * if the result is negative, or false when - * result is greater than or equal to zero. Note that the guaranteed - * useful range of an atomic_t is only 24 bits. - * - * Currently not implemented for MIPS. - */ - -#define mb() \ -__asm__ __volatile__( \ - "# prevent instructions being moved around\n\t" \ - ".set\tnoreorder\n\t" \ - "# 8 nops to fool the R4400 pipeline\n\t" \ - "nop;nop;nop;nop;nop;nop;nop;nop\n\t" \ - ".set\treorder" \ - : /* no output */ \ - : /* no input */ \ - : "memory") -#define rmb() mb() -#define wmb() mb() - -#define IATOMIC_DEFINED 1 - -#endif /* __mips__ */ - -#ifdef __arm__ - -/* - * FIXME: bellow code is valid only for SA11xx - */ - -/* - * Save the current interrupt enable state & disable IRQs - */ -#define local_irq_save(x) \ - ({ \ - unsigned long temp; \ - __asm__ __volatile__( \ - "mrs %0, cpsr @ local_irq_save\n" \ -" orr %1, %0, #128\n" \ -" msr cpsr_c, %1" \ - : "=r" (x), "=r" (temp) \ - : \ - : "memory"); \ - }) - -/* - * restore saved IRQ & FIQ state - */ -#define local_irq_restore(x) \ - __asm__ __volatile__( \ - "msr cpsr_c, %0 @ local_irq_restore\n" \ - : \ - : "r" (x) \ - : "memory") - -#define __save_flags_cli(x) local_irq_save(x) -#define __restore_flags(x) local_irq_restore(x) - -typedef struct { volatile int counter; } atomic_t; - -#define ATOMIC_INIT(i) { (i) } - -#define atomic_read(v) ((v)->counter) -#define atomic_set(v,i) (((v)->counter) = (i)) - -static __inline__ void atomic_add(int i, volatile atomic_t *v) -{ - unsigned long flags; - - __save_flags_cli(flags); - v->counter += i; - __restore_flags(flags); -} - -static __inline__ void atomic_sub(int i, volatile atomic_t *v) -{ - unsigned long flags; - - __save_flags_cli(flags); - v->counter -= i; - __restore_flags(flags); -} - -static __inline__ void atomic_inc(volatile atomic_t *v) -{ - unsigned long flags; - - __save_flags_cli(flags); - v->counter += 1; - __restore_flags(flags); -} - -static __inline__ void atomic_dec(volatile atomic_t *v) -{ - unsigned long flags; - - __save_flags_cli(flags); - v->counter -= 1; - __restore_flags(flags); -} - -static __inline__ int atomic_dec_and_test(volatile atomic_t *v) -{ - unsigned long flags; - int result; - - __save_flags_cli(flags); - v->counter -= 1; - result = (v->counter == 0); - __restore_flags(flags); - - return result; -} - -static __inline__ int atomic_add_negative(int i, volatile atomic_t *v) -{ - unsigned long flags; - int result; - - __save_flags_cli(flags); - v->counter += i; - result = (v->counter < 0); - __restore_flags(flags); - - return result; -} - -static __inline__ void atomic_clear_mask(unsigned long mask, unsigned long *addr) -{ - unsigned long flags; - - __save_flags_cli(flags); - *addr &= ~mask; - __restore_flags(flags); -} - -#define mb() __asm__ __volatile__ ("" : : : "memory") -#define rmb() mb() -#define wmb() mb() - -#define IATOMIC_DEFINED 1 - -#endif /* __arm__ */ - -#ifdef __sh__ - -typedef struct { volatile int counter; } atomic_t; - -#define ATOMIC_INIT(i) { (i) } - -#define atomic_read(v) ((v)->counter) -#define atomic_set(v,i) (((v)->counter) = (i)) - -#define atomic_dec_return(v) atomic_sub_return(1,(v)) -#define atomic_inc_return(v) atomic_add_return(1,(v)) - -#define atomic_sub_and_test(i,v) (atomic_sub_return((i), (v)) == 0) -#define atomic_dec_and_test(v) (atomic_sub_return(1, (v)) == 0) -#define atomic_inc_and_test(v) (atomic_add_return(1, (v)) != 0) - -#define atomic_add(i,v) atomic_add_return((i),(v)) -#define atomic_sub(i,v) atomic_sub_return((i),(v)) -#define atomic_inc(v) atomic_add(1,(v)) -#define atomic_dec(v) atomic_sub(1,(v)) - -static __inline__ int atomic_add_return(int i, volatile atomic_t *v) -{ - int result; - - asm volatile ( - " .align 2\n" - " mova 99f, r0\n" - " mov r15, r1\n" - " mov #-6, r15\n" - " mov.l @%2, %0\n" - " add %1, %0\n" - " mov.l %0, @%2\n" - "99: mov r1, r15" - : "=&r"(result) - : "r"(i), "r"(v) - : "r0", "r1"); - - return result; -} - -static __inline__ int atomic_sub_return(int i, volatile atomic_t *v) -{ - int result; - - asm volatile ( - " .align 2\n" - " mova 99f, r0\n" - " mov r15, r1\n" - " mov #-6, r15\n" - " mov.l @%2, %0\n" - " sub %1, %0\n" - " mov.l %0, @%2\n" - "99: mov r1, r15" - : "=&r"(result) - : "r"(i), "r"(v) - : "r0", "r1"); - - return result; -} - -#define mb() __asm__ __volatile__ ("" : : : "memory") -#define rmb() mb() -#define wmb() mb() - -#define IATOMIC_DEFINED 1 - -#endif /* __sh__ */ - -#ifdef __bfin__ - -#include <bfin_fixed_code.h> - -typedef struct { volatile int counter; } atomic_t; - -#define ATOMIC_INIT(i) { (i) } +#ifndef IATOMIC_DEFINED -#define atomic_read(v) ((v)->counter) -#define atomic_set(v,i) (((v)->counter) = (i)) -#define atomic_add(i,v) bfin_atomic_add32(&(v)->counter, i) -#define atomic_sub(i,v) bfin_atomic_sub32(&(v)->counter, i) -#define atomic_inc(v) bfin_atomic_inc32(&(v)->counter); -#define atomic_dec(v) bfin_atomic_dec32(&(v)->counter); +/* Generic __sync_synchronize is available from gcc 4.1 */ -#define mb() __asm__ __volatile__ ("" : : : "memory") +#define mb() __sync_synchronize() #define rmb() mb() #define wmb() mb() -#define IATOMIC_DEFINED 1 - -#endif /* __bfin__ */ - -#ifndef IATOMIC_DEFINED -/* - * non supported architecture. - */ -#warning "Atomic operations are not supported on this architecture." - -typedef struct { volatile int counter; } atomic_t; - -#define ATOMIC_INIT(i) { (i) } - -#define atomic_read(v) ((v)->counter) -#define atomic_set(v,i) (((v)->counter) = (i)) -#define atomic_add(i,v) (((v)->counter) += (i)) -#define atomic_sub(i,v) (((v)->counter) -= (i)) -#define atomic_inc(v) (((v)->counter)++) -#define atomic_dec(v) (((v)->counter)--) - -#define mb() -#define rmb() -#define wmb() - #define IATOMIC_DEFINED 1 #endif /* IATOMIC_DEFINED */ diff --git a/src/pcm/pcm_meter.c b/src/pcm/pcm_meter.c index 573b6fe..676fbef 100644 --- a/src/pcm/pcm_meter.c +++ b/src/pcm/pcm_meter.c @@ -33,7 +33,10 @@ #include <dlfcn.h> #include "pcm_local.h" #include "pcm_plugin.h" -#include "iatomic.h" + +#define atomic_read(ptr) __atomic_load_n(ptr, __ATOMIC_SEQ_CST ) +#define atomic_add(ptr, n) __atomic_add_fetch(ptr, n, __ATOMIC_SEQ_CST) +#define atomic_dec(ptr) __atomic_sub_fetch(ptr, 1, __ATOMIC_SEQ_CST) #ifndef PIC /* entry for static linking */ @@ -61,7 +64,7 @@ typedef struct _snd_pcm_meter { struct list_head scopes; int closed; int running; - atomic_t reset; + int reset; pthread_t thread; pthread_mutex_t update_mutex; pthread_mutex_t running_mutex; @@ -288,7 +291,7 @@ static int snd_pcm_meter_prepare(snd_pcm_t *pcm) { snd_pcm_meter_t *meter = pcm->private_data; int err; - atomic_inc(&meter->reset); + atomic_add(&meter->reset, 1); err = snd_pcm_prepare(meter->gen.slave); if (err >= 0) { if (pcm->stream == SND_PCM_STREAM_PLAYBACK) -- 1.8.1.2

3 2

[alsa-devel] [PATCH 1/2] ASoC: ak4671: Convert to table based control init
by Mark Brown 11 Feb '14

11 Feb '14

From: Mark Brown <broonie(a)linaro.org> Saves code and adds error handling. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ak4671.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 25bdf6ad4a54..456bd0a065b1 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -628,9 +628,6 @@ static int ak4671_probe(struct snd_soc_codec *codec) return ret; } - snd_soc_add_codec_controls(codec, ak4671_snd_controls, - ARRAY_SIZE(ak4671_snd_controls)); - ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return ret; @@ -646,6 +643,8 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4671 = { .probe = ak4671_probe, .remove = ak4671_remove, .set_bias_level = ak4671_set_bias_level, + .controls = ak4671_snd_controls, + .num_controls = ARRAY_SIZE(ak4671_snd_controls), .reg_cache_size = AK4671_CACHEREGNUM, .reg_word_size = sizeof(u8), .reg_cache_default = ak4671_reg, -- 1.9.0.rc3

1 1

[alsa-devel] [PATCH 1/2] ASoC: cs42l51: Don't log if we fail to allocate memory
by Mark Brown 11 Feb '14

11 Feb '14

From: Mark Brown <broonie(a)linaro.org> The VM subsystem already logs quite loudly if we run out of memory so don't bother here. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/cs42l51.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 6e9ea8379a91..50c08c8fc9bc 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -574,10 +574,8 @@ static int cs42l51_i2c_probe(struct i2c_client *i2c_client, cs42l51 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l51_private), GFP_KERNEL); - if (!cs42l51) { - dev_err(&i2c_client->dev, "could not allocate codec\n"); + if (!cs42l51) return -ENOMEM; - } i2c_set_clientdata(i2c_client, cs42l51); cs42l51->control_type = SND_SOC_I2C; -- 1.9.0.rc3

2 3

[alsa-devel] [PATCH] ASoC: smdk_wm8994: Fix typo "wm8894"
by Paul Bolle 11 Feb '14

11 Feb '14

Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- 0) This typo was introduced in commit 0d47acc4ffaa ("ASoC: smdk_wm8994: Make driver name more unique"). 1) So currently there's a difference between platform_drive.driver.name and the (sub)string used in MODULE_ALIAS(). If I'd understand the platform_driver interface I might have been able to say whether that difference is only cosmetic or might actually break stuff. But I don't understand it so I can't say. I can't even say whether this typo shows up in some interface to user space. 2) Entirely untested. sound/soc/samsung/smdk_wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index d38ae98..682eb4f 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -202,7 +202,7 @@ static int smdk_audio_probe(struct platform_device *pdev) static struct platform_driver smdk_audio_driver = { .driver = { - .name = "smdk-audio-wm8894", + .name = "smdk-audio-wm8994", .owner = THIS_MODULE, .of_match_table = of_match_ptr(samsung_wm8994_of_match), .pm = &snd_soc_pm_ops, -- 1.8.5.3

2 1

[alsa-devel] [PATCH] ASoC: fsl_esai: Check the return value from clk_prepare_enable()
by Fabio Estevam 11 Feb '14

11 Feb '14

clk_prepare_enable() may fail, so let's check its return value and propagate it in the case of error. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_esai.c | 27 ++++++++++++++++++++++----- 1 file changed, 22 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index d0c72ed..e64115c 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -431,17 +431,26 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) static int fsl_esai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { + int ret; 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); + ret = clk_prepare_enable(esai_priv->coreclk); + if (ret) + return ret; + if (!IS_ERR(esai_priv->extalclk)) { + ret = clk_prepare_enable(esai_priv->extalclk); + if (ret) + goto err_extalck; + } + if (!IS_ERR(esai_priv->fsysclk)) { + ret = clk_prepare_enable(esai_priv->fsysclk); + if (ret) + goto err_fsysclk; + } if (!dai->active) { /* Reset Port C */ @@ -463,6 +472,14 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, } return 0; + +err_fsysclk: + if (!IS_ERR(esai_priv->extalclk)) + clk_disable_unprepare(esai_priv->extalclk); +err_extalck: + clk_disable_unprepare(esai_priv->coreclk); + + return ret; } static int fsl_esai_hw_params(struct snd_pcm_substream *substream, -- 1.8.1.2

2 1

[alsa-devel] [PATCH 0/7] Audio support for Armada 370 DB
by Thomas Petazzoni 11 Feb '14

11 Feb '14

Hello, This series of patches enable audio support on the Marvell Armada 370 Development Board. Since both the I2S controller on the SoC side and the I2C audio codec are already supported by the kernel, the amount of work is fairly limited. Also, since the DT bindings that allows to replace the ASoC board driver are still being ironed out, I'm proposing to add an old-style ASoC board driver for now. This set of patches only provide support for the analog output and input, I am waiting for the optical cable to arrive to get the digital input and output to work. Patches 1 to 3 are to be taken by the ASoC maintainer, while patches 4 to 7 are to be taken by the mvebu maintainers. Note that the audio support for Armada 370 also needs a fix to the CS42L51, which is being discussed with the author of the change that apparently introduced the problem (see discussion at http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071916.ht…). Thanks, Thomas Thomas Petazzoni (7): sound: codec: add Device Tree binding to cs42l51 sound: soc: enable Kirkwood driver for mvebu platforms sound: soc: add ASoC board driver for Armada 370 DB ARM: mvebu: add audio I2S controller to Armada 370 Device Tree ARM: mvebu: add I2C0 muxing option for Armada 370 SoC ARM: mvebu: add audio support to Armada 370 DB ARM: mvebu: enable audio options in mvebu_defconfig .../devicetree/bindings/i2c/trivial-devices.txt | 1 + .../devicetree/bindings/sound/mvebu-audio.txt | 1 + arch/arm/boot/dts/armada-370-db.dts | 46 +++++++ arch/arm/boot/dts/armada-370.dtsi | 29 +++++ arch/arm/configs/mvebu_defconfig | 5 + sound/soc/codecs/cs42l51.c | 7 + sound/soc/kirkwood/Kconfig | 10 +- sound/soc/kirkwood/Makefile | 2 + sound/soc/kirkwood/armada-370-db.c | 144 +++++++++++++++++++++ sound/soc/kirkwood/kirkwood-i2s.c | 1 + 10 files changed, 245 insertions(+), 1 deletion(-) create mode 100644 sound/soc/kirkwood/armada-370-db.c -- 1.8.3.2

3 15

[alsa-devel] [PATCH] ASoC: wm8995: Remove unused pointer in hp_supply_event()
by Christian Engelmayer 11 Feb '14

11 Feb '14

Remove unused driver data pointer 'wm8995' in function hp_supply_event(). Detected by Coverity: CID 141181. Signed-off-by: Christian Engelmayer <cengelma(a)gmx.at> --- Applies against v3.14-rc1 as well as branch for-next in tree git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git. --- sound/soc/codecs/wm8995.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 4300caf..403e1db 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -561,10 +561,8 @@ static int hp_supply_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec; - struct wm8995_priv *wm8995; codec = w->codec; - wm8995 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_PRE_PMU: -- 1.8.3.2

2 1

[alsa-devel] [PATCH 1/2] ASoC: fsl-spdif: big-endian support
by Xiubo Li 11 Feb '14

11 Feb '14

For most platforms, the CPU and SPDIF device is in the same endianess mode. While for the LS1 platform, the CPU is in LE mode and the SPDIF is in BE mode. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> Cc: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Documentation/devicetree/bindings/sound/fsl,spdif.txt | 4 ++++ sound/soc/fsl/fsl_spdif.c | 8 +++++++- 2 files changed, 11 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt index f2ae335..433ad00 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.txt @@ -29,6 +29,10 @@ Required properties: can also be referred to TxClk_Source bit of register SPDIF_STC. + - big-endian : If this property is absent, the native endian mode will + be in use as default, or the big endian mode will be in use for all the + device registers. + Example: spdif: spdif@02004000 { diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 8f36f49..2f8ad75 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -83,6 +83,8 @@ struct fsl_spdif_priv { struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; + bool big_endian; + /* The name space will be allocated dynamically */ char name[0]; }; @@ -985,7 +987,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) } } -static const struct regmap_config fsl_spdif_regmap_config = { +static struct regmap_config fsl_spdif_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, @@ -1105,6 +1107,10 @@ static int fsl_spdif_probe(struct platform_device *pdev) memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai)); spdif_priv->cpu_dai_drv.name = spdif_priv->name; + spdif_priv->big_endian = of_property_read_bool(np, "big-endian"); + if (spdif_priv->big_endian) + fsl_spdif_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; + /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); -- 1.8.4

3 7

[alsa-devel] Noise on right channel of WM9715
by Andreas Zeiler 11 Feb '14

11 Feb '14

Hi, we have a board with an iMX6 processor from freescale and a Wolfson WM9715. I can play music/sound files with it. We're on Linux 3.0.35. The problem is that there is a noise on the right channel and this noise changes with the frequency. I don't mean the sample rate with that. I'm playing a wav-file with a Sinus from 440Hz to 880Hz or higher. I can hear how the noise is changing. Then I played a silence file with nothing in it. When I see this file it has some small artefacts (in audacity). On the left channel I hear nothing as expected. On the right channel I hear some clicking. It's not periodic but I hear it very often. So I turned off everything I don't need in the WM9715. So just the DAC (PCM (Register 0x18)) and the Headphone Mixer (register 0x04) are turned on. Everything else is in default state or muted. AC97 communication is working. I have a good clock etc. I think that there is something wrong when my audio data gets into the ac97 slot. Maybe it gets shifted wrong or something else. Can someone help me with this? Does someone know what can be wrong here?? Mit freundlichen Grüßen / With kind regards Andreas Zeiler (zeiler(a)fs-net.de<mailto:zeiler@fs-net.de>) ________________________________ armStone: Pico-ITX ARM based Single Board Computer ________________________________ F&S Elektronik Systeme GmbH, Stuttgart, Germany Geschaeftsfuehrer: Holger Froelich, Guenther Scholz Handelsregister Stuttgart HRB 18493, VAT DE178964418

1 0

[alsa-devel] [PATCH v2] ASoC: fsl-sai: convert to use regmap API for Freeacale SAI
by Xiubo Li 11 Feb '14

11 Feb '14

Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> Cc: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- This version applies against sound-3.14-rc1. sound/soc/fsl/fsl_sai.c | 255 +++++++++++++++++++++++++++--------------------- sound/soc/fsl/fsl_sai.h | 47 +++++---- 2 files changed, 166 insertions(+), 136 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index cdd3fa8..faa65af 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) { @@ -81,7 +55,7 @@ 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); + regmap_write(sai->regmap, reg_cr2, val_cr2); return 0; } @@ -89,32 +63,22 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, 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 +97,8 @@ 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 +147,25 @@ 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 +189,12 @@ 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 +212,9 @@ 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; } @@ -268,44 +223,34 @@ 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_cr3, reg_cr3; - - 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; rcsr &= ~FSL_SAI_CSR_FRDE; - reg_cr3 = FSL_SAI_TCR3; } else { rcsr |= FSL_SAI_CSR_FRDE; tcsr &= ~FSL_SAI_CSR_FRDE; - reg_cr3 = FSL_SAI_RCR3; } - val_cr3 = sai_readl(sai, sai->base + reg_cr3); - switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: tcsr |= FSL_SAI_CSR_TERE; rcsr |= FSL_SAI_CSR_TERE; - val_cr3 |= FSL_SAI_CR3_TRCE; - sai_writel(sai, val_cr3, sai->base + reg_cr3); - 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: @@ -314,11 +259,8 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, rcsr &= ~FSL_SAI_CSR_TERE; } - val_cr3 &= ~FSL_SAI_CR3_TRCE; - - sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); - sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); - sai_writel(sai, val_cr3, sai->base + reg_cr3); + regmap_write(sai->regmap, FSL_SAI_TCSR, tcsr); + regmap_write(sai->regmap, FSL_SAI_RCSR, rcsr); break; default: return -EINVAL; @@ -331,16 +273,32 @@ 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 reg; - return clk_prepare_enable(sai->clk); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = FSL_SAI_TCR3; + else + reg = FSL_SAI_RCR3; + + regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, + FSL_SAI_CR3_TRCE); + + 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 reg; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = FSL_SAI_TCR3; + else + reg = FSL_SAI_RCR3; - 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 = { @@ -355,18 +313,13 @@ 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); @@ -397,26 +350,109 @@ static const struct snd_soc_component_driver fsl_component = { .name = "fsl-sai", }; +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 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); if (!sai) return -ENOMEM; + sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs"); + if (sai->big_endian_regs) + fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; + + sai->big_endian_data = of_property_read_bool(np, "big-endian-data"); + 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; @@ -424,9 +460,6 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; sai->dma_params_tx.maxburst = FSL_SAI_MAXBURST_TX; - sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs"); - sai->big_endian_data = of_property_read_bool(np, "big-endian-data"); - platform_set_drvdata(pdev, sai); ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, 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

3 2

[alsa-devel] Register wm8782 codec driver
by Ezequiel RG 10 Feb '14

10 Feb '14

Hi everyone, i'm trying to interface the wm8782 codec with my Beaglebone Black running 3.8.13 kernel. I'm getting this error when i load the device tree file: davinci_evm sound.11: ASoC: CODEC (null) not registered What i'm doing wrong?? The correponding source file are attached. Thanks for your help! wm8782.c /* * sound/soc/codecs/wm8782.c * simple, strap-pin configured 24bit 2ch ADC * * Copyright: 2011 Raumfeld GmbH * Author: Johannes Stezenbach <js(a)sig21.net> * * based on ad73311.c * Copyright:Analog Device Inc. * Author:Cliff Cai <cliff.cai(a)analog.com> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. */ #include <linux/init.h> #include <linux/slab.h> #include <linux/module.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> #include <sound/initval.h> #include <sound/soc.h> #include <linux/of_device.h> #include <linux/of_device.h> static const struct of_device_id wm8782_of_match[] = { { .compatible = "wlf,wm8782", }, { } }; MODULE_DEVICE_TABLE(of, wm8782_of_match); static const struct snd_soc_dapm_widget wm8782_dapm_widgets[] = { SND_SOC_DAPM_INPUT("AINL"), SND_SOC_DAPM_INPUT("AINR"), }; static const struct snd_soc_dapm_route wm8782_dapm_routes[] = { { "Capture", NULL, "AINL" }, { "Capture", NULL, "AINR" }, }; static struct snd_soc_dai_driver wm8782_dai = { .name = "wm8782-hifi", .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, /* For configurations with FSAMPEN=0 */ .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, }, }; static struct snd_soc_codec_driver soc_codec_dev_wm8782 = { .dapm_widgets = wm8782_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8782_dapm_widgets), .dapm_routes = wm8782_dapm_routes, .num_dapm_routes = ARRAY_SIZE(wm8782_dapm_routes), }; static int wm8782_probe(struct platform_device *pdev) { return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8782, &wm8782_dai, 1); } static int wm8782_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; } static struct platform_driver wm8782_codec_driver = { .driver = { .name = "wm8782", .owner = THIS_MODULE, }, .probe = wm8782_probe, .remove = wm8782_remove, }; module_platform_driver(wm8782_codec_driver); MODULE_DESCRIPTION("ASoC WM8782 driver"); MODULE_AUTHOR("Johannes Stezenbach <js(a)sig21.net>"); MODULE_LICENSE("GPL"); davinci-evm.c /* * ASoC driver for TI DAVINCI EVM platform * * Author: Vladimir Barinov, <vbarinov(a)embeddedalley.com> * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.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/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> #include <linux/platform_data/edma.h> #include <linux/i2c.h> #include <linux/of_platform.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <asm/dma.h> #include <asm/mach-types.h> #include <linux/edma.h> #include "davinci-pcm.h" #include "davinci-i2s.h" #include "davinci-mcasp.h" #include <linux/of_gpio.h> #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF) static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *soc_card = codec->card; struct device_node *np = soc_card->dev->of_node; int ret = 0; unsigned sysclk; if (np) { ret = of_property_read_u32(np, "ti,codec-clock-rate", &sysclk); if (ret < 0) return ret; } else { /* ASP1 on DM355 EVM is clocked by an external oscillator */ if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() || machine_is_davinci_dm365_evm()) sysclk = 27000000; /* * ASP0 in DM6446 EVM is clocked by U55, as configured by * board-dm644x-evm.c using GPIOs from U18. There are six * options; here we "know" we use a 48 KHz sample rate. */ else if (machine_is_davinci_evm()) sysclk = 12288000; else if (machine_is_davinci_da830_evm() || machine_is_davinci_da850_evm()) sysclk = 24576000; else return -EINVAL; } /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT); if (ret < 0) return ret; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); if (ret < 0) return ret; /* set the codec system clock */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, SND_SOC_CLOCK_OUT); if (ret < 0) return ret; /* set the CPU system clock */ ret = snd_soc_dai_set_sysclk(cpu_dai, 0, sysclk, SND_SOC_CLOCK_OUT); if (ret < 0) return ret; return 0; } static int evm_spdif_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; /* set cpu DAI configuration */ return snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); } static struct snd_soc_ops evm_ops = { .hw_params = evm_hw_params, }; static struct snd_soc_ops evm_spdif_ops = { .hw_params = evm_spdif_hw_params, }; /* davinci-evm machine dapm widgets */ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_LINE("Line Out", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), SND_SOC_DAPM_LINE("Line In", NULL), }; /* davinci-evm machine audio_mapnections to the codec pins */ static const struct snd_soc_dapm_route audio_map[] = { /* Headphone connected to HPLOUT, HPROUT */ {"Headphone Jack", NULL, "HPLOUT"}, {"Headphone Jack", NULL, "HPROUT"}, /* Line Out connected to LLOUT, RLOUT */ {"Line Out", NULL, "LLOUT"}, {"Line Out", NULL, "RLOUT"}, /* Mic connected to (MIC3L | MIC3R) */ {"MIC3L", NULL, "Mic Bias 2V"}, {"MIC3R", NULL, "Mic Bias 2V"}, {"Mic Bias 2V", NULL, "Mic Jack"}, /* Line In connected to (LINE1L | LINE2L), (LINE1R | LINE2R) */ {"LINE1L", NULL, "Line In"}, {"LINE2L", NULL, "Line In"}, {"LINE1R", NULL, "Line In"}, {"LINE2R", NULL, "Line In"}, }; /* Logic for a tda998x as connected on a davinci-evm */ static int evm_tda998x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; ret = snd_soc_dai_set_clkdiv(cpu_dai, 0, 1); if (ret < 0) return ret; ret = snd_soc_dai_set_clkdiv(cpu_dai, 1, 8); if (ret < 0) return ret; ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, SND_SOC_CLOCK_IN); if (ret < 0) return ret; ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_IB_NF); if (ret < 0) return ret; return 0; } /* Logic for a aic3x as connected on a davinci-evm */ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dapm_context *dapm = &codec->dapm; struct device_node *np = codec->card->dev->of_node; int ret; /* Add davinci-evm specific widgets */ snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, ARRAY_SIZE(aic3x_dapm_widgets)); if (np) { ret = snd_soc_of_parse_audio_routing(codec->card, "ti,audio-routing"); if (ret) return ret; } else { /* Set up davinci-evm specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); } /* Divide McASP MCLK by 2 to provide 12MHz to codec */ ret = snd_soc_dai_set_clkdiv(cpu_dai, 0, 2); if (ret < 0) return ret; /* not connected */ snd_soc_dapm_disable_pin(dapm, "MONO_LOUT"); snd_soc_dapm_disable_pin(dapm, "HPLCOM"); snd_soc_dapm_disable_pin(dapm, "HPRCOM"); /* always connected */ snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Line Out"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); snd_soc_dapm_enable_pin(dapm, "Line In"); return 0; } /* davinci-evm digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link dm6446_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name = "davinci-mcbsp", .codec_dai_name = "tlv320aic3x-hifi", .codec_name = "tlv320aic3x-codec.1-001b", .platform_name = "davinci-mcbsp", .init = evm_aic3x_init, .ops = &evm_ops, }; static struct snd_soc_dai_link dm355_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name = "davinci-mcbsp.1", .codec_dai_name = "tlv320aic3x-hifi", .codec_name = "tlv320aic3x-codec.1-001b", .platform_name = "davinci-mcbsp.1", .init = evm_aic3x_init, .ops = &evm_ops, }; static struct snd_soc_dai_link dm365_evm_dai = { #ifdef CONFIG_SND_DM365_AIC3X_CODEC .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name = "davinci-mcbsp", .codec_dai_name = "tlv320aic3x-hifi", .init = evm_aic3x_init, .codec_name = "tlv320aic3x-codec.1-0018", .ops = &evm_ops, .platform_name = "davinci-mcbsp", #elif defined(CONFIG_SND_DM365_VOICE_CODEC) .name = "Voice Codec - CQ93VC", .stream_name = "CQ93", .cpu_dai_name = "davinci-vcif", .codec_dai_name = "cq93vc-hifi", .codec_name = "cq93vc-codec", .platform_name = "davinci-vcif", #endif }; static struct snd_soc_dai_link dm6467_evm_dai[] = { { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "davinci-mcasp.0", .codec_name = "tlv320aic3x-codec.0-001a", .init = evm_aic3x_init, .ops = &evm_ops, }, { .name = "McASP", .stream_name = "spdif", .cpu_dai_name= "davinci-mcasp.1", .codec_dai_name = "dit-hifi", .codec_name = "spdif_dit", .platform_name = "davinci-mcasp.1", .ops = &evm_spdif_ops, }, }; static struct snd_soc_dai_link da830_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name = "davinci-mcasp.1", .codec_dai_name = "tlv320aic3x-hifi", .codec_name = "tlv320aic3x-codec.1-0018", .platform_name = "davinci-mcasp.1", .init = evm_aic3x_init, .ops = &evm_ops, }; static struct snd_soc_dai_link da850_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", .codec_dai_name = "tlv320aic3x-hifi", .codec_name = "tlv320aic3x-codec.1-0018", .platform_name = "davinci-mcasp.0", .init = evm_aic3x_init, .ops = &evm_ops, }; /* davinci dm6446 evm audio machine driver */ static struct snd_soc_card dm6446_snd_soc_card_evm = { .name = "DaVinci DM6446 EVM", .owner = THIS_MODULE, .dai_link = &dm6446_evm_dai, .num_links = 1, }; /* davinci dm355 evm audio machine driver */ static struct snd_soc_card dm355_snd_soc_card_evm = { .name = "DaVinci DM355 EVM", .owner = THIS_MODULE, .dai_link = &dm355_evm_dai, .num_links = 1, }; /* davinci dm365 evm audio machine driver */ static struct snd_soc_card dm365_snd_soc_card_evm = { .name = "DaVinci DM365 EVM", .owner = THIS_MODULE, .dai_link = &dm365_evm_dai, .num_links = 1, }; /* davinci dm6467 evm audio machine driver */ static struct snd_soc_card dm6467_snd_soc_card_evm = { .name = "DaVinci DM6467 EVM", .owner = THIS_MODULE, .dai_link = dm6467_evm_dai, .num_links = ARRAY_SIZE(dm6467_evm_dai), }; static struct snd_soc_card da830_snd_soc_card = { .name = "DA830/OMAP-L137 EVM", .owner = THIS_MODULE, .dai_link = &da830_evm_dai, .num_links = 1, }; static struct snd_soc_card da850_snd_soc_card = { .name = "DA850/OMAP-L138 EVM", .owner = THIS_MODULE, .dai_link = &da850_evm_dai, .num_links = 1, }; #if defined(CONFIG_OF) enum { MACHINE_VERSION_1 = 0,/* DM365 with Voice Codec */ MACHINE_VERSION_2,/* DM365/DA8xx/OMAPL1x/AM33xx */ MACHINE_VERSION_3,/* AM33xx BeagleBone Black */ }; static const struct of_device_id davinci_evm_dt_ids[] = { { .compatible = "ti,dm365-voice-codec-audio", .data = (void *)MACHINE_VERSION_1, }, { .compatible = "ti,da830-evm-audio", .data = (void *)MACHINE_VERSION_2, }, { .compatible = "ti,am33xx-beaglebone-black", .data = (void *)MACHINE_VERSION_3, }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, davinci_mcasp_dt_ids); /* * This struct is just used as place holder. It will be filled with * data from dt node */ static struct snd_soc_dai_link evm_dai = { .name= "WM8782", .stream_name= "AIC3X", .codec_dai_name= "wm8782-hifi", }; /* davinci evm audio machine driver */ static struct snd_soc_card evm_soc_card = { .owner = THIS_MODULE, .dai_link = &evm_dai, .num_links = 1, }; static int davinci_evm_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; const struct of_device_id *match = of_match_device(of_match_ptr(davinci_evm_dt_ids), &pdev->dev); u32 machine_ver, clk_gpio; int ret = 0; machine_ver = (u32)match->data; switch (machine_ver) { case MACHINE_VERSION_1: evm_dai.name= "Voice Codec - CQ93VC"; evm_dai.stream_name= "CQ93"; evm_dai.codec_dai_name= "cq93vc-hifi"; break; case MACHINE_VERSION_2: evm_dai.ops = &evm_ops; evm_dai.init = evm_aic3x_init; break; case MACHINE_VERSION_3: evm_dai.name= "NXP TDA HDMI Chip"; evm_dai.stream_name= "HDMI"; evm_dai.codec_dai_name= "nxp-hdmi-hifi"; evm_dai.init = evm_tda998x_init; /* * Move GPIO handling out of the probe, if probe gets * deferred, the gpio will have been claimed on previous * probe and will fail on the second and susequent probes */ clk_gpio = of_get_named_gpio(np, "mcasp_clock_enable", 0); if (clk_gpio < 0) { dev_err(&pdev->dev, "failed to find mcasp_clock enable GPIO!\n"); return -EINVAL; } ret = gpio_request_one(clk_gpio, GPIOF_OUT_INIT_HIGH, "McASP Clock Enable Pin"); if (ret < 0) { dev_err(&pdev->dev, "Failed to claim McASP Clock Enable pin\n"); return -EINVAL; } gpio_set_value(clk_gpio, 1); break; } evm_dai.codec_of_node = of_parse_phandle(np, "ti,audio-codec", 0); if (!evm_dai.codec_of_node) return -EINVAL; evm_dai.cpu_of_node = of_parse_phandle(np, "ti,mcasp-controller", 0); if (!evm_dai.cpu_of_node) return -EINVAL; evm_dai.platform_of_node = evm_dai.cpu_of_node; evm_soc_card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&evm_soc_card, "ti,model"); if (ret) return ret; ret = snd_soc_register_card(&evm_soc_card); if (ret) dev_err(&pdev->dev, "snd_soc_register_card fuck failed (%d)\n", ret); return ret; } static int davinci_evm_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); snd_soc_unregister_card(card); return 0; } static struct platform_driver davinci_evm_driver = { .probe= davinci_evm_probe, .remove= davinci_evm_remove, .driver= { .name= "davinci_evm", .owner= THIS_MODULE, .of_match_table = of_match_ptr(davinci_evm_dt_ids), }, }; #endif static struct platform_device *evm_snd_device; static int __init evm_init(void) { struct snd_soc_card *evm_snd_dev_data; int index; int ret; #if defined(CONFIG_OF) /* * If dtb is there, the devices will be created dynamically. * Only register platfrom driver structure. */ if (of_have_populated_dt()) return platform_driver_register(&davinci_evm_driver); #endif if (machine_is_davinci_evm()) { evm_snd_dev_data = &dm6446_snd_soc_card_evm; index = 0; } else if (machine_is_davinci_dm355_evm()) { evm_snd_dev_data = &dm355_snd_soc_card_evm; index = 1; } else if (machine_is_davinci_dm365_evm()) { evm_snd_dev_data = &dm365_snd_soc_card_evm; index = 0; } else if (machine_is_davinci_dm6467_evm()) { evm_snd_dev_data = &dm6467_snd_soc_card_evm; index = 0; } else if (machine_is_davinci_da830_evm()) { evm_snd_dev_data = &da830_snd_soc_card; index = 1; } else if (machine_is_davinci_da850_evm()) { evm_snd_dev_data = &da850_snd_soc_card; index = 0; } else return -EINVAL; evm_snd_device = platform_device_alloc("soc-audio", index); if (!evm_snd_device) return -ENOMEM; platform_set_drvdata(evm_snd_device, evm_snd_dev_data); ret = platform_device_add(evm_snd_device); if (ret) platform_device_put(evm_snd_device); return ret; } static void __exit evm_exit(void) { #if defined(CONFIG_OF) if (of_have_populated_dt()) { platform_driver_unregister(&davinci_evm_driver); return; } #endif platform_device_unregister(evm_snd_device); } module_init(evm_init); module_exit(evm_exit); MODULE_AUTHOR("Vladimir Barinov"); MODULE_DESCRIPTION("TI DAVINCI EVM ASoC driver"); MODULE_LICENSE("GPL"); device tree file /* * Copyright (C) 2014 John Rhoades * * 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. */ /dts-v1/; /plugin/; / { compatible = "ti,beaglebone-black"; part-number = "jsr-cs4271"; version = "00A0"; /* state the resources this cape uses */ exclusive-use = /* the pin header uses */ "P9.17", /* cs4271 reset */ /* i2c2 */ "P9.19", /* i2c2: scl */ "P9.20", /* i2c2: sca */ /* mcasp0 */ "P9.31", /* mcasp0: aclkx 0x190 */ "P9.29", /* mcasp0: fsx 0x194 */ /* not used "P9.30", /* mcasp0: axr0 0x198 */ "P9.28", /* mcasp0: axr2 0x19c */ "P9.42", /* mcasp0: aclkr 0x1a0 */ "P9.27", /* mcasp0: fsr 0x1a4 */ /* not used "P9.41", /* mcasp0: axr1 0x1a8 */ "P9.25", /* mcasp0: axr3 0x1ac */ /* the hardware IP uses */ "gpio1_27", "snd-soc-wm8782", // needed to get module loaded "mcasp0"; fragment@0 { target = <&am33xx_pinmux>; __overlay__ { wm8782_i2c2_reset: wm8782_i2c2_reset { pinctrl-single,pins = < 0x078 0x0f /* gpio0_5, mode7 */ >; }; wm8782_i2c2_pins: wm8782_i2c2_pins { pinctrl-single,pins = < 0x178 0x73 /* I2C2_sda */ 0x17c 0x73 /* I2C2_scl */ >; }; mcasp0_pins: mcasp0_pins { pinctrl-single,pins = < 0x190 0x20 /* P9.31 mcasp0 0->aclkx */ 0x194 0x20 /* P9.29 mcasp0 0->fsx */ 0x198 0x20 /* P9.30 mcasp0 0->axr0 */ 0x19c 0x22 /* P9.28 mcasp0 0->ahclkr, 2->axr2 */ 0x1a0 0x20 /* P9.42 mcasp0 0->aclkr, 2->axr2 */ 0x1a4 0x20 /* P9.27 mcasp0 0->fsr, 2->axr3 */ 0x1a8 0x20 /* P9.41 mcasp0 0->axr1 */ 0x1ac 0x22 /* P9.25 mcasp0 0->ahclkx, 2->axr3 */ >; }; }; }; fragment@2 { target = <&i2c2>; __overlay__ { #address-cells = <1>; #size-cells = <0>; wm8782: wm8782@10 { compatible = ",wm8782"; reg = <0x10>; status = "okay"; }; }; }; fragment@3 { target = <&mcasp0>; __overlay__ { pinctrl-names = "default"; pinctrl-0 = <&mcasp0_pins>; status = "okay"; op-mode = <0>; /* MCASP_I2S_MODE */ tdm-slots = <2>; num-serializer = <16>; serial-dir = < /* 0: INACTIVE, 1: TX, 2: RX */ 0 0 2 1 /* axr2: capture data, axr3: play data */ 0 0 0 0 0 0 0 0 0 0 0 0 >; tx-num-evt = <1>; rx-num-evt = <1>; }; }; fragment@4 { target = <&ocp>; __overlay__ { sound { compatible = "ti,da830-evm-audio"; //"ti,am33xx-beaglebone-black"; ti,model = "DA830 EVM"; // "wm8782"; ti,audio-codec = <&wm8782>; ti,mcasp-controller = <&mcasp0>; ti,codec-clock-rate = <24576000>; #mcasp_clock_enable = <&gpio2 27 0>; /* BeagleBone Black Clk enable on GPIO1_27 */ ti,audio-routing = "Headphone Jack", "HPLOUT", "Headphone Jack", "HPROUT", "LINE1L", "Line In", "LINE1R", "Line In"; }; }; }; };

1 0

Re: [alsa-devel] Fw: [Bug 1155202] [Intel DZ77SL-50K, Intel PantherPoint HDMI, Digital Out, HDMI] No sound at all
by niraj kulkarni 10 Feb '14

10 Feb '14

Removing these 3 usleep ranges is working now. Although I have a doubt for applying this patch on all chips. There have been a number of boards released after this particular board, but nobody else is facing this problem. Since spec does not specify a max on deasserting RST, some boards might actually take more than 100uS to reset. Will this patch work for those boards? @@ -1149,7 +1149,7 @@ timeout = jiffies + msecs_to_jiffies(100); while ((azx_readb(chip, GCTL) & ICH6_GCTL_RESET) && time_before(jiffies, timeout)) - usleep_range(500, 1000); + cpu_relax(); } /* exit link reset */ @@ -1162,7 +1162,7 @@ timeout = jiffies + msecs_to_jiffies(100); while (!azx_readb(chip, GCTL) && time_before(jiffies, timeout)) - usleep_range(500, 1000); + cpu_relax(); } /* reset codec link */ @@ -1180,7 +1180,7 @@ /* delay for >= 100us for codec PLL to settle per spec * Rev 0.9 section 5.5.1 */ - usleep_range(500, 1000); + //usleep_range(500, 1000); /* Bring controller out of reset */ azx_exit_link_reset(chip);

2 3

[alsa-devel] [PATCH] ALSA: hda - Disable static quirks for C-Media codecs
by Takashi Iwai 10 Feb '14

10 Feb '14

According to alsa-info.sh outputs, all three entries with static quirks have the correct pin configs, so it's safe to remove static quirks. For now, turn the static quirks off via ifdef. The dead codes will be removed in later release. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_cmedia.c | 31 +++++++++++++++++++++---------- 1 file changed, 21 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 9c6ce73b03c5..139ef301c914 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -32,6 +32,9 @@ #include "hda_jack.h" #include "hda_generic.h" +#undef ENABLE_CMI_STATIC_QUIRKS + +#ifdef ENABLE_CMI_STATIC_QUIRKS #define NUM_PINS 11 @@ -45,10 +48,12 @@ enum { CMI_AUTO, /* let driver guess it */ CMI_MODELS }; +#endif /* ENABLE_CMI_STATIC_QUIRKS */ struct cmi_spec { struct hda_gen_spec gen; +#ifdef ENABLE_CMI_STATIC_QUIRKS /* below are only for static models */ int board_config; @@ -81,8 +86,10 @@ struct cmi_spec { /* multichannel pins */ struct hda_verb multi_init[9]; /* 2 verbs for each pin + terminator */ +#endif /* ENABLE_CMI_STATIC_QUIRKS */ }; +#ifdef ENABLE_CMI_STATIC_QUIRKS /* * input MUX */ @@ -566,6 +573,7 @@ static const struct hda_codec_ops cmi9880_patch_ops = { .init = cmi9880_init, .free = cmi9880_free, }; +#endif /* ENABLE_CMI_STATIC_QUIRKS */ /* * stuff for auto-parser @@ -588,15 +596,20 @@ static int cmi_parse_auto_config(struct hda_codec *codec) err = snd_hda_parse_pin_defcfg(codec, cfg, NULL, 0); if (err < 0) - return err; + goto error; err = snd_hda_gen_parse_auto_config(codec, cfg); if (err < 0) - return err; + goto error; codec->patch_ops = cmi_auto_patch_ops; return 0; + + error: + snd_hda_gen_free(codec); + return err; } + static int patch_cmi9880(struct hda_codec *codec) { struct cmi_spec *spec; @@ -606,6 +619,7 @@ static int patch_cmi9880(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; +#ifdef ENABLE_CMI_STATIC_QUIRKS spec->board_config = snd_hda_check_board_config(codec, CMI_MODELS, cmi9880_models, cmi9880_cfg_tbl); @@ -615,14 +629,8 @@ static int patch_cmi9880(struct hda_codec *codec) spec->board_config = CMI_AUTO; /* try everything */ } - if (spec->board_config == CMI_AUTO) { - int err = cmi_parse_auto_config(codec); - if (err < 0) { - snd_hda_gen_free(codec); - return err; - } - return 0; - } + if (spec->board_config == CMI_AUTO) + return cmi_parse_auto_config(codec); /* copy default DAC NIDs */ memcpy(spec->dac_nids, cmi9880_dac_nids, sizeof(spec->dac_nids)); @@ -669,6 +677,9 @@ static int patch_cmi9880(struct hda_codec *codec) codec->patch_ops = cmi9880_patch_ops; return 0; +#else + return cmi_parse_auto_config(codec); +#endif } /* -- 1.8.5.2

1 0

[alsa-devel] [PATCH v3 0/2] *** SUBJECT HERE ***
by Jean-Francois Moine 10 Feb '14

10 Feb '14

*** BLURB HERE *** Jean-Francois Moine (2): drivers/base: permit base components to omit the bind/unbind ops drivers/base: declare phandle DT nodes as components drivers/base/component.c | 21 +++++++++++++++++++-- drivers/base/core.c | 18 ++++++++++++++++++ include/linux/of.h | 2 ++ 3 files changed, 39 insertions(+), 2 deletions(-) -- 1.9.rc1

4 7

[alsa-devel] [PATCH v2 0/8] da9055: Driver initialisation fixes, add DT support
by Adam Thomson 10 Feb '14

10 Feb '14

This patch series provides the following updates for DA9055 drivers: - Fixes an issue with da9055 driver initialisation (conflicting device ids) of PMIC (MFD) and CODEC drivers which means they cannot be used together on the same I2C bus. - Add initial DT support for DA9055 related drivers, including binding documentation. - Remove use of regmap_irq_get_virq() in driver probes which was conflicting with use of platform_get_irq_byname(). platform_get_irq_byname() already returns the VIRQ number due to MFD core translation so using regmap_irq_get_virq() on that returned value results in an incorrect IRQ being requested. The driver probes then fail because of this. [Changes since v1]: - Regulator: * fixed coding style issue (lack of braces for if,else statement) * removal of 'ifdef' around DT related headers * Use of 'of_regulator_match()' for retrieving regulator related info from DT nodes. - General: * Instead of removing platform_get_irq_byname() usage and device resources to fix IRQ conflict in driver probes, now removed 'regmap_irq_get_virq()' as advised. This has been tested on target for both DT and non-DT setups and works as expected. Adam Thomson (8): ASoC: da9055: Fix device registration of PMIC and CODEC devices ASoC: da9055: Add DT support for CODEC mfd: da9055: Add DT support for PMIC regulator: da9055: Add DT support onkey: da9055: Remove use of regmap_irq_get_virq() hwmon: da9055: Remove use of regmap_irq_get_virq() rtc: da9055: Remove use of regmap_irq_get_virq() regulator: da9055: Remove use of regmap_irq_get_virq() Documentation/devicetree/bindings/mfd/da9055.txt | 73 ++++++++++++++++++++ Documentation/devicetree/bindings/sound/da9055.txt | 22 ++++++ drivers/hwmon/da9055-hwmon.c | 4 - drivers/input/misc/da9055_onkey.c | 1 - drivers/mfd/da9055-i2c.c | 20 +++++- drivers/regulator/da9055-regulator.c | 73 ++++++++++++++++++-- drivers/rtc/rtc-da9055.c | 4 +- sound/soc/codecs/da9055.c | 19 +++++- 8 files changed, 201 insertions(+), 15 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/da9055.txt create mode 100644 Documentation/devicetree/bindings/sound/da9055.txt

5 24

[alsa-devel] [PATCH 0/3] Add i.MX6q Kosagi Novena support
by Sean Cross 10 Feb '14

10 Feb '14

The Kosagi Novena board contains an i.MX6 Quad processor. This patchset adds support for this platform, along with support for its own audio codec. The platform uses an ES8328-based codec, which is currently unsupported in the Linux kernel. The audio codec supports playback and recording, as well as jack detection support for the headset. Audio codec support is required as an uninitialized audio codec pulls I2C2 lines down, preventing the bus from working. There are two known issues with this patch, and I would like some feedback on them: * I'm unclear on how audio routing works. The ES8328 has five outputs: LOUT1, ROUT1, LOUT2, ROUT2, and OUT3. It also has four inputs. The board maps Headphone L/R to L/ROUT2, and Speaker to L/ROUT1. What is the proper way to describe this audio routing? * The snd-soc-imx-novena.ko module oopses when removing it. The backtrace shows that it's caused by "Unable to handle kernel paging request at virtual address", due to snd_jack_dev_free possibly getting called twice. What could be causing this? I never explicitly free the jack device: (snd_ctl_remove) from [<804c73d0>] (snd_jack_dev_free+0x48/0x68) (snd_jack_dev_free) from [<804c6ef8>] (snd_device_free+0xd0/0x148) (snd_device_free) from [<804c72b0>] (snd_device_free_all+0x90/0xb0) (snd_device_free_all) from [<804c20ac>] (snd_card_do_free+0x40/0xec) (snd_card_do_free) from [<804c22b4>] (snd_card_free+0x74/0x80) (snd_card_free) from [<804de444>] (soc_cleanup_card_resources+0xb4/0xbc) (soc_cleanup_card_resources) from [<804de460>] (snd_soc_unregister_card+0x14/0x1c) (snd_soc_unregister_card) from [<7f032014>] (imx_novena_remove+0x14/0x1c [snd_soc_imx_novena]) (imx_novena_remove [snd_soc_imx_novena]) from [<8035c258>] (__device_release_driver+0x64/0xb0) (__device_release_driver) from [<8035c81c>] (driver_detach+0xa4/0xcc) (driver_detach) from [<8035be88>] (bus_remove_driver+0x64/0x9c) (bus_remove_driver) from [<8007ea44>] (SyS_delete_module+0x110/0x188) (SyS_delete_module) from [<8000e080>] (ret_fast_syscall+0x0/0x30) Sean Cross (3): sound: soc: codecs: Add es8328 codec sound: soc: fsl: Add support for Novena onboard audio dts: imx: add kosagi novena imx6q dts file arch/arm/boot/dts/imx6q-novena.dts | 517 +++++++++++++++++++++++++++++++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8328.c | 611 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/es8328.h | 240 +++++++++++++++ sound/soc/fsl/Kconfig | 13 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-novena.c | 344 +++++++++++++++++++++ 8 files changed, 1733 insertions(+) create mode 100644 arch/arm/boot/dts/imx6q-novena.dts create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-novena.c -- 1.8.3.2

3 13

[alsa-devel] [PATCH] ASoC: fsl-esai: fix ESAI TDM slot setting
by Xiubo Li 10 Feb '14

10 Feb '14

Cc: Nicolin Chen <Guangyu.Chen(a)freescale.com> Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_esai.c | 4 ++-- sound/soc/fsl/fsl_esai.h | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index d0c72ed..c84026c 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -326,7 +326,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, 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)); + ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(tx_mask)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); @@ -334,7 +334,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, 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_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask)); esai_priv->slot_width = slot_width; diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 9c9f957..75e1403 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -322,7 +322,7 @@ #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) +#define ESAI_xSMB_xS(v) (((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMB_xS_MASK) /* Port C Direction Register -- REG_ESAI_PRRC 0xF8 */ #define ESAI_PRRC_PDC_SHIFT 0 -- 1.8.4

4 5

[alsa-devel] [PATCH v3 0/4] off-codec widgets
by Xiubo Li 10 Feb '14

10 Feb '14

This patch series has removed the code about tdm slot, and need to do more study. This adds "ASoC: binding: add widgets.txt" new patch. This also revises some others from Mark's comment. Xiubo Li (4): ASoC: add snd_soc_of_parse_audio_simple_widgets for DT ASoC: simple-card: add off-codec widgets supports. ASoC: binding: add widgets.txt ASoC: binding: for new properties documenting and usage .../devicetree/bindings/sound/simple-card.txt | 12 ++- .../devicetree/bindings/sound/widgets.txt | 20 +++++ include/sound/soc.h | 2 + sound/soc/generic/simple-card.c | 8 ++ sound/soc/soc-core.c | 87 ++++++++++++++++++++++ 5 files changed, 127 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/widgets.txt -- 1.8.4

2 5

[alsa-devel] [PATCH] ASoC: fsl: fix pm support of machine drivers
by Shawn Guo 10 Feb '14

10 Feb '14

The commit 1abe729 (ASoC: fsl: Add missing pm to current machine drivers) enables pm support for a few IMX machine drivers. But it does not update dev drvdata to be the pointer to 'card'. This causes the kernel dump below in system suspend, because snd_soc_suspend() expects that the dev drvdata points to 'card', while it still points to the private data of machine driver. This patch fixes imx-sgtl5000 and imx-wm8962 by attaching 'card' to dev drvdata and private data to card drvdata. For imx-mc13783, I simply revert the pm change because it must be broken for the same reason and I don't have hardware to test pm enabling code. $ echo mem > /sys/power/state PM: Syncing filesystems ... done. PM: Preparing system for mem sleep mmc1: card e624 removed Freezing user space processes ... (elapsed 0.002 seconds) done. Freezing remaining freezable tasks ... (elapsed 0.002 seconds) done. PM: Entering mem sleep INFO: trying to register non-static key. the code is fine but needs lockdep annotation. turning off the locking correctness validator. CPU: 0 PID: 1861 Comm: bash Not tainted 3.14.0-rc1+ #1648 Backtrace: [<80012144>] (dump_backtrace) from [<800122e4>] (show_stack+0x18/0x1c) r6:8079c77c r5:00000c5a r4:00000000 r3:00000000 [<800122cc>] (show_stack) from [<80637ac0>] (dump_stack+0x78/0x94) [<80637a48>] (dump_stack) from [<80028918>] (warn_slowpath_common+0x6c/0x8c) r4:bdb21c38 r3:be62df00 [<800288ac>] (warn_slowpath_common) from [<800289dc>] (warn_slowpath_fmt+0x38/0x40) r8:be62e3a8 r7:bf122960 r6:00000005 r5:00000000 r4:00000000 [<800289a8>] (warn_slowpath_fmt) from [<8006518c>] (__lock_acquire+0x1ae0/0x1ce0) r3:8079d598 r2:80799e70 [<800636ac>] (__lock_acquire) from [<80065894>] (lock_acquire+0x68/0x7c) r10:bdb20000 r9:be62df00 r8:00000000 r7:00000000 r6:60000013 r5:bdb20000 r4:00000000 [<8006582c>] (lock_acquire) from [<8063c938>] (mutex_lock_nested+0x5c/0x3b8) r7:00000000 r6:80dfc78c r5:804be444 r4:bf122928 [<8063c8dc>] (mutex_lock_nested) from [<804be444>] (snd_soc_suspend+0x34/0x42c) r10:00000000 r9:00000000 r8:00000000 r7:bf1c4444 r6:bf1c4410 r5:be978150 r4:be978010 [<804be410>] (snd_soc_suspend) from [<8034392c>] (platform_pm_suspend+0x34/0x64) r10:00000000 r8:00000000 r7:bf1c4444 r6:bf1c4410 r5:803438f8 r4:bf1c4410 [<803438f8>] (platform_pm_suspend) from [<80348e18>] (dpm_run_callback.isra.7+0x34/0x6c) [<80348de4>] (dpm_run_callback.isra.7) from [<80349354>] (__device_suspend+0x10c/0x220) r9:808dd974 r8:808c4a5c r6:00000002 r5:80e5001c r4:bf1c4410 [<80349248>] (__device_suspend) from [<8034a338>] (dpm_suspend+0x60/0x220) r7:bf1c4410 r6:808dd90c r5:80e5001c r4:bf1c44c0 [<8034a2d8>] (dpm_suspend) from [<8034a790>] (dpm_suspend_start+0x60/0x68) r10:8079a818 r9:00000000 r8:00000004 r7:80dfbe90 r6:80641eec r5:00000000 r4:00000002 [<8034a730>] (dpm_suspend_start) from [<8006a788>] (suspend_devices_and_enter+0x74/0x318) r4:00000003 r3:80dfbe98 [<8006a714>] (suspend_devices_and_enter) from [<8006abd8>] (pm_suspend+0x1ac/0x244) r10:8079a818 r8:00000004 r7:00000003 r6:80641eec r5:00000000 r4:00000003 [<8006aa2c>] (pm_suspend) from [<80069a4c>] (state_store+0x70/0xc0) r5:00000003 r4:bd85ea40 [<800699dc>] (state_store) from [<80294034>] (kobj_attr_store+0x1c/0x28) r10:beb9fe08 r8:00000000 r7:bdb21f78 r6:bd85ea40 r5:00000004 r4:beb9fe00 [<80294018>] (kobj_attr_store) from [<80140f90>] (sysfs_kf_write+0x54/0x58) [<80140f3c>] (sysfs_kf_write) from [<8014474c>] (kernfs_fop_write+0xc4/0x160) r6:bd85ea40 r5:beb9fe00 r4:00000004 r3:80140f3c [<80144688>] (kernfs_fop_write) from [<800dfa14>] (vfs_write+0xbc/0x184) r10:00000000 r9:00000000 r8:00000000 r7:bdb21f78 r6:00500c08 r5:00000004 r4:be782600 [<800df958>] (vfs_write) from [<800dfe00>] (SyS_write+0x48/0x70) r10:00000000 r8:00000000 r7:00000004 r6:00500c08 r5:00000000 r4:be782600 [<800dfdb8>] (SyS_write) from [<8000e800>] (ret_fast_syscall+0x0/0x48) r9:bdb20000 r8:8000e9c4 r7:00000004 r6:00500c08 r5:00000004 r4:76eb65e0 Fixes: 1abe729 (ASoC: fsl: Add missing pm to current machine drivers) Cc: stable(a)vger.kernel.org Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/fsl/imx-mc13783.c | 1 - sound/soc/fsl/imx-sgtl5000.c | 10 ++++++---- sound/soc/fsl/imx-wm8962.c | 11 +++++++---- 3 files changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index 79cee78..a2fd732 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -160,7 +160,6 @@ static struct platform_driver imx_mc13783_audio_driver = { .driver = { .name = "imx_mc13783", .owner = THIS_MODULE, - .pm = &snd_soc_pm_ops, }, .probe = imx_mc13783_probe, .remove = imx_mc13783_remove diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index f2beae7..1cb22dd 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -33,8 +33,7 @@ struct imx_sgtl5000_data { static int imx_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct imx_sgtl5000_data *data = container_of(rtd->card, - struct imx_sgtl5000_data, card); + struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(rtd->card); struct device *dev = rtd->card->dev; int ret; @@ -159,13 +158,15 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dapm_widgets = imx_sgtl5000_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets); + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto fail; } - platform_set_drvdata(pdev, data); of_node_put(ssi_np); of_node_put(codec_np); @@ -184,7 +185,8 @@ fail: static int imx_sgtl5000_remove(struct platform_device *pdev) { - struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(card); clk_put(data->codec_clk); diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3fd76bc..3a3d17c 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -71,7 +71,7 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, { 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 imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; unsigned int pll_out; int ret; @@ -137,7 +137,7 @@ static int imx_wm8962_late_probe(struct snd_soc_card *card) { 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 imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; int ret; @@ -264,13 +264,15 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->card.late_probe = imx_wm8962_late_probe; data->card.set_bias_level = imx_wm8962_set_bias_level; + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto clk_fail; } - platform_set_drvdata(pdev, data); of_node_put(ssi_np); of_node_put(codec_np); @@ -289,7 +291,8 @@ fail: static int imx_wm8962_remove(struct platform_device *pdev) { - struct imx_wm8962_data *data = platform_get_drvdata(pdev); + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); if (!IS_ERR(data->codec_clk)) clk_disable_unprepare(data->codec_clk); -- 1.7.9.5

3 2

[alsa-devel] [PATCH 0/8] New and Resend the old patches
by Xiubo Li 10 Feb '14

10 Feb '14

New and Resend the old patches basing the newest code version. And this patch series has been tested based the VF610 Tower board.

5 24

[alsa-devel] [PATCH] ALSA: hda - Fix undefined symbol due to builtin/module mixup
by Takashi Iwai 10 Feb '14

10 Feb '14

Even after the fix for leftover kconfig handling (commit f8f1becf), the current code still doesn't handle properly the builtin/module mixup case between the core snd-hda-codec and other codec drivers. For example, when CONFIG_SND_HDA_INTEL=y and CONFIG_SND_HDA_CODEC_HDMI=m, it'll end up with an unresolved symbol snd_hda_parse_hdmi_codec. This patch fixes the issue. Now codec->parser points to the parser object *only* when a module (either generic or HDMI parser) is loaded and bound. When a builtin symbol is used, codec->parser still points to NULL. This is the difference from the previous versions. Fixes: f8f1becfa4ac ('ALSA: hda - Fix leftover ifdef checks after modularization') Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 69acbf758bbb..dafcf82139e2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1339,23 +1339,15 @@ get_hda_cvt_setup(struct hda_codec *codec, hda_nid_t nid) /* * Dynamic symbol binding for the codec parsers */ -#ifdef MODULE -#define load_parser_sym(sym) ((int (*)(struct hda_codec *))symbol_request(sym)) -#define unload_parser_addr(addr) symbol_put_addr(addr) -#else -#define load_parser_sym(sym) (sym) -#define unload_parser_addr(addr) do {} while (0) -#endif #define load_parser(codec, sym) \ - ((codec)->parser = load_parser_sym(sym)) + ((codec)->parser = (int (*)(struct hda_codec *))symbol_request(sym)) static void unload_parser(struct hda_codec *codec) { - if (codec->parser) { - unload_parser_addr(codec->parser); - codec->parser = NULL; - } + if (codec->parser) + symbol_put_addr(codec->parser); + codec->parser = NULL; } /* @@ -1620,12 +1612,20 @@ int snd_hda_codec_configure(struct hda_codec *codec) patch = codec->preset->patch; if (!patch) { unload_parser(codec); /* to be sure */ - if (is_likely_hdmi_codec(codec)) + if (is_likely_hdmi_codec(codec)) { +#if IS_MODULE(CONFIG_SND_HDA_CODEC_HDMI) patch = load_parser(codec, snd_hda_parse_hdmi_codec); -#if IS_ENABLED(CONFIG_SND_HDA_GENERIC) - if (!patch) +#elif IS_BUILTIN(CONFIG_SND_HDA_CODEC_HDMI) + patch = snd_hda_parse_hdmi_codec; +#endif + } + if (!patch) { +#if IS_MODULE(CONFIG_SND_HDA_GENERIC) patch = load_parser(codec, snd_hda_parse_generic_codec); +#elif IS_BUILTIN(CONFIG_SND_HDA_GENERIC) + patch = snd_hda_parse_generic_codec; #endif + } if (!patch) { printk(KERN_ERR "hda-codec: No codec parser is available\n"); return -ENODEV; -- 1.8.5.2

1 0

[alsa-devel] [PATCH v2] ALSA: Replace with IS_ENABLED()
by Takashi Iwai 10 Feb '14

10 Feb '14

Replace the lengthy #if defined(XXX) || defined(XXX_MODULE) with the new IS_ENABLED() macro. The patch still doesn't cover all ifdefs. For example, the dependency on CONFIG_GAMEPORT is still open-coded because this also has an extra dependency on MODULE. Similarly, an open-coded ifdef in pcm_oss.c and some sequencer-related stuff are left untouched. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Remove conversions wrt sequencer that unsatisfy config dependency properly sound/core/control.c | 2 +- sound/core/info.c | 4 ++-- sound/core/init.c | 8 ++++---- sound/core/pcm.c | 10 +++++----- sound/core/pcm_native.c | 2 +- sound/core/rtctimer.c | 4 ++-- sound/core/sound_oss.c | 2 +- sound/core/timer.c | 4 ++-- sound/drivers/opl3/opl3_synth.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- 10 files changed, 20 insertions(+), 20 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index d8aa206e8bde..c228f00e5701 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -151,7 +151,7 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, if (snd_BUG_ON(!card || !id)) return; read_lock(&card->ctl_files_rwlock); -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) card->mixer_oss_change_count++; #endif list_for_each_entry(ctl, &card->ctl_files, list) { diff --git a/sound/core/info.c b/sound/core/info.c index e79baa11b60e..7916c07b9324 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -540,7 +540,7 @@ int __init snd_info_init(void) snd_oss_root = entry; } #endif -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) { struct snd_info_entry *entry; if ((entry = snd_info_create_module_entry(THIS_MODULE, "seq", NULL)) == NULL) @@ -567,7 +567,7 @@ int __exit snd_info_done(void) snd_minor_info_done(); snd_info_version_done(); if (snd_proc_root) { -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) snd_info_free_entry(snd_seq_root); #endif #ifdef CONFIG_SND_OSSEMUL diff --git a/sound/core/init.c b/sound/core/init.c index 0d42fcda0de2..a16d765cdf47 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -94,7 +94,7 @@ static int module_slot_match(struct module *module, int idx) return match; } -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) int (*snd_mixer_oss_notify_callback)(struct snd_card *card, int free_flag); EXPORT_SYMBOL(snd_mixer_oss_notify_callback); #endif @@ -394,7 +394,7 @@ int snd_card_disconnect(struct snd_card *card) /* phase 3: notify all connected devices about disconnection */ /* at this point, they cannot respond to any calls except release() */ -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) if (snd_mixer_oss_notify_callback) snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_DISCONNECT); #endif @@ -430,7 +430,7 @@ EXPORT_SYMBOL(snd_card_disconnect); */ static int snd_card_do_free(struct snd_card *card) { -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) if (snd_mixer_oss_notify_callback) snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_FREE); #endif @@ -723,7 +723,7 @@ int snd_card_register(struct snd_card *card) snd_cards[card->number] = card; mutex_unlock(&snd_card_mutex); init_info_for_card(card); -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) if (snd_mixer_oss_notify_callback) snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_REGISTER); #endif diff --git a/sound/core/pcm.c b/sound/core/pcm.c index e1e9e0c999fe..091a05c1d5b3 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -295,7 +295,7 @@ static const char *snd_pcm_state_name(snd_pcm_state_t state) return snd_pcm_state_names[(__force int)state]; } -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) #include <linux/soundcard.h> static const char *snd_pcm_oss_format_name(int format) @@ -398,7 +398,7 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "rate: %u (%u/%u)\n", runtime->rate, runtime->rate_num, runtime->rate_den); snd_iprintf(buffer, "period_size: %lu\n", runtime->period_size); snd_iprintf(buffer, "buffer_size: %lu\n", runtime->buffer_size); -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) if (substream->oss.oss) { snd_iprintf(buffer, "OSS format: %s\n", snd_pcm_oss_format_name(runtime->oss.format)); snd_iprintf(buffer, "OSS channels: %u\n", runtime->oss.channels); @@ -651,7 +651,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) struct snd_pcm_str *pstr = &pcm->streams[stream]; struct snd_pcm_substream *substream, *prev; -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) mutex_init(&pstr->oss.setup_mutex); #endif pstr->stream = stream; @@ -807,7 +807,7 @@ EXPORT_SYMBOL(snd_pcm_new_internal); static void snd_pcm_free_stream(struct snd_pcm_str * pstr) { struct snd_pcm_substream *substream, *substream_next; -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) struct snd_pcm_oss_setup *setup, *setupn; #endif substream = pstr->substream; @@ -819,7 +819,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) substream = substream_next; } snd_pcm_stream_proc_done(pstr); -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) for (setup = pstr->oss.setup_list; setup; setup = setupn) { setupn = setup->next; kfree(setup->task_name); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 10b5d1a9dec0..e3664116736b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -399,7 +399,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, return -EBADFD; } snd_pcm_stream_unlock_irq(substream); -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) if (!substream->oss.oss) #endif if (atomic_read(&substream->mmap_count)) diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c index e85e72baff9e..b27223746b0e 100644 --- a/sound/core/rtctimer.c +++ b/sound/core/rtctimer.c @@ -27,7 +27,7 @@ #include <sound/core.h> #include <sound/timer.h> -#if defined(CONFIG_RTC) || defined(CONFIG_RTC_MODULE) +#if IS_ENABLED(CONFIG_RTC) #include <linux/mc146818rtc.h> @@ -185,4 +185,4 @@ MODULE_LICENSE("GPL"); MODULE_ALIAS("snd-timer-" __stringify(SNDRV_TIMER_GLOBAL_RTC)); -#endif /* CONFIG_RTC || CONFIG_RTC_MODULE */ +#endif /* IS_ENABLED(CONFIG_RTC) */ diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index 726a49ac9725..5391c5e139ef 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -21,7 +21,7 @@ #ifdef CONFIG_SND_OSSEMUL -#if !defined(CONFIG_SOUND) && !(defined(MODULE) && defined(CONFIG_SOUND_MODULE)) +#if !IS_ENABLED(CONFIG_SOUND) #error "Enable the OSS soundcore multiplexer (CONFIG_SOUND) in the kernel." #endif diff --git a/sound/core/timer.c b/sound/core/timer.c index 6ddcf06f52f9..cbec51464c5b 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -35,9 +35,9 @@ #include <sound/initval.h> #include <linux/kmod.h> -#if defined(CONFIG_SND_HRTIMER) || defined(CONFIG_SND_HRTIMER_MODULE) +#if IS_ENABLED(CONFIG_SND_HRTIMER) #define DEFAULT_TIMER_LIMIT 4 -#elif defined(CONFIG_SND_RTCTIMER) || defined(CONFIG_SND_RTCTIMER_MODULE) +#elif IS_ENABLED(CONFIG_SND_RTCTIMER) #define DEFAULT_TIMER_LIMIT 2 #else #define DEFAULT_TIMER_LIMIT 1 diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index 742a4b642fd9..ddcc1a325a61 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -24,7 +24,7 @@ #include <sound/opl3.h> #include <sound/asound_fm.h> -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) #define OPL3_SUPPORT_SYNTH #endif diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e87eac552ec5..17c83fbc9f84 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -4137,7 +4137,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { .driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA }, { PCI_DEVICE(0x1102, 0x0012), .driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA }, -#if !defined(CONFIG_SND_CTXFI) && !defined(CONFIG_SND_CTXFI_MODULE) +#if !IS_ENABLED(CONFIG_SND_CTXFI) /* the following entry conflicts with snd-ctxfi driver, * as ctxfi driver mutates from HD-audio to native mode with * a special command sequence. -- 1.8.5.2

1 0

[alsa-devel] [PATCH] Refresh patch, .reset_resume added to struct usb_driver
by eliot＠blennerhassett.gen.nz 10 Feb '14

10 Feb '14

From: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> Signed-off-by: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> --- alsa/usb/card.patch | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/alsa/usb/card.patch b/alsa/usb/card.patch index 679bf15..a37c93d 100644 --- a/alsa/usb/card.patch +++ b/alsa/usb/card.patch @@ -1,5 +1,5 @@ ---- ../alsa-kernel/usb/card.c 2013-10-09 14:00:50.128683771 +0200 -+++ card.c 2013-10-15 15:45:31.779430219 +0200 +--- ../alsa-kernel/usb/card.c 2014-02-09 14:35:45.872517538 +1300 ++++ card.c 2014-02-09 16:45:45.116673978 +1300 @@ -1,3 +1,4 @@ +#include "card.inc" /* @@ -67,15 +67,15 @@ /* * new 2.5 USB kernel API */ -@@ -727,6 +742,7 @@ - #define usb_audio_suspend NULL +@@ -738,6 +753,7 @@ #define usb_audio_resume NULL + #define usb_audio_reset_resume NULL #endif /* CONFIG_PM */ +#endif /* OLD_USB */ static struct usb_device_id usb_audio_ids [] = { #include "quirks-table.h" -@@ -742,13 +758,25 @@ +@@ -753,14 +769,25 @@ */ static struct usb_driver usb_audio_driver = { @@ -88,6 +88,7 @@ +#ifndef OLD_USB .suspend = usb_audio_suspend, .resume = usb_audio_resume, +- .reset_resume = usb_audio_reset_resume, +#endif +#ifdef OLD_USB + .driver_list = LIST_HEAD_INIT(usb_audio_driver.driver_list), -- 1.7.10.4

3 2

[alsa-devel] [PATCH] [trivial] alsa: Fix typos in alsa-driver-api.xml
by Masanari Iida 10 Feb '14

10 Feb '14

This patch fixed 2 typos in DocBook/alsa-driver-api.xml. It is because this file is generated by make xmldocs, I have to fix typos within source files. Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> --- sound/core/pcm_native.c | 2 +- sound/core/rawmidi.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 01a5e05..10b5d1a 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -954,7 +954,7 @@ static struct action_ops snd_pcm_action_stop = { * * The state of each stream is then changed to the given state unconditionally. * - * Return: Zero if succesful, or a negative error code. + * Return: Zero if successful, or a negative error code. */ int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t state) { diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 7b596b5..f016be7 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1101,7 +1101,7 @@ int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, /** * snd_rawmidi_transmit_ack - acknowledge the transmission * @substream: the rawmidi substream - * @count: the tranferred count + * @count: the transferred count * * Advances the hardware pointer for the internal output buffer with * the given size and updates the condition. -- 1.9.rc1

2 1

[alsa-devel] [PATCH] ALSA: Replace with IS_ENABLED()
by Takashi Iwai 10 Feb '14

10 Feb '14

Replace the lengthy #if defined(XXX) || defined(XXX_MODULE) with the new IS_ENABLED() macro. The sequencer-related ifdefs in some places (e.g. rawmidi.c) are a bit tricky. Instead of adding the extra dependency on CONFIG_MODULE, use the another Kconfig such as CONFIG_SND_RAQMIDI_SEQ that already takes such a dependency into account. The patch still doesn't cover all ifdefs. For example, the dependency on CONFIG_GAMEPORT is still open-coded because this also has an extra dependency on CONFIG_MODULE. Similarly, an open-coded ifdef is still left in pcm_oss.c for OSS mixer ioctl. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/control.c | 2 +- sound/core/info.c | 4 ++-- sound/core/init.c | 8 ++++---- sound/core/pcm.c | 10 +++++----- sound/core/pcm_native.c | 2 +- sound/core/rawmidi.c | 4 ++-- sound/core/rtctimer.c | 4 ++-- sound/core/sound_oss.c | 2 +- sound/core/timer.c | 4 ++-- sound/drivers/opl3/opl3_lib.c | 2 +- sound/drivers/opl3/opl3_synth.c | 2 +- sound/drivers/opl4/opl4_lib.c | 4 ++-- sound/drivers/opl4/opl4_local.h | 2 +- sound/isa/sb/emu8000.c | 2 +- sound/isa/sb/sb16.c | 2 +- sound/pci/emu10k1/emu10k1.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- 17 files changed, 29 insertions(+), 29 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index d8aa206e8bde..c228f00e5701 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -151,7 +151,7 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, if (snd_BUG_ON(!card || !id)) return; read_lock(&card->ctl_files_rwlock); -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) card->mixer_oss_change_count++; #endif list_for_each_entry(ctl, &card->ctl_files, list) { diff --git a/sound/core/info.c b/sound/core/info.c index e79baa11b60e..7916c07b9324 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -540,7 +540,7 @@ int __init snd_info_init(void) snd_oss_root = entry; } #endif -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) { struct snd_info_entry *entry; if ((entry = snd_info_create_module_entry(THIS_MODULE, "seq", NULL)) == NULL) @@ -567,7 +567,7 @@ int __exit snd_info_done(void) snd_minor_info_done(); snd_info_version_done(); if (snd_proc_root) { -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) snd_info_free_entry(snd_seq_root); #endif #ifdef CONFIG_SND_OSSEMUL diff --git a/sound/core/init.c b/sound/core/init.c index 0d42fcda0de2..a16d765cdf47 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -94,7 +94,7 @@ static int module_slot_match(struct module *module, int idx) return match; } -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) int (*snd_mixer_oss_notify_callback)(struct snd_card *card, int free_flag); EXPORT_SYMBOL(snd_mixer_oss_notify_callback); #endif @@ -394,7 +394,7 @@ int snd_card_disconnect(struct snd_card *card) /* phase 3: notify all connected devices about disconnection */ /* at this point, they cannot respond to any calls except release() */ -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) if (snd_mixer_oss_notify_callback) snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_DISCONNECT); #endif @@ -430,7 +430,7 @@ EXPORT_SYMBOL(snd_card_disconnect); */ static int snd_card_do_free(struct snd_card *card) { -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) if (snd_mixer_oss_notify_callback) snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_FREE); #endif @@ -723,7 +723,7 @@ int snd_card_register(struct snd_card *card) snd_cards[card->number] = card; mutex_unlock(&snd_card_mutex); init_info_for_card(card); -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) if (snd_mixer_oss_notify_callback) snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_REGISTER); #endif diff --git a/sound/core/pcm.c b/sound/core/pcm.c index e1e9e0c999fe..091a05c1d5b3 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -295,7 +295,7 @@ static const char *snd_pcm_state_name(snd_pcm_state_t state) return snd_pcm_state_names[(__force int)state]; } -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) #include <linux/soundcard.h> static const char *snd_pcm_oss_format_name(int format) @@ -398,7 +398,7 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "rate: %u (%u/%u)\n", runtime->rate, runtime->rate_num, runtime->rate_den); snd_iprintf(buffer, "period_size: %lu\n", runtime->period_size); snd_iprintf(buffer, "buffer_size: %lu\n", runtime->buffer_size); -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) if (substream->oss.oss) { snd_iprintf(buffer, "OSS format: %s\n", snd_pcm_oss_format_name(runtime->oss.format)); snd_iprintf(buffer, "OSS channels: %u\n", runtime->oss.channels); @@ -651,7 +651,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) struct snd_pcm_str *pstr = &pcm->streams[stream]; struct snd_pcm_substream *substream, *prev; -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) mutex_init(&pstr->oss.setup_mutex); #endif pstr->stream = stream; @@ -807,7 +807,7 @@ EXPORT_SYMBOL(snd_pcm_new_internal); static void snd_pcm_free_stream(struct snd_pcm_str * pstr) { struct snd_pcm_substream *substream, *substream_next; -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) struct snd_pcm_oss_setup *setup, *setupn; #endif substream = pstr->substream; @@ -819,7 +819,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) substream = substream_next; } snd_pcm_stream_proc_done(pstr); -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) for (setup = pstr->oss.setup_list; setup; setup = setupn) { setupn = setup->next; kfree(setup->task_name); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 10b5d1a9dec0..e3664116736b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -399,7 +399,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, return -EBADFD; } snd_pcm_stream_unlock_irq(substream); -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) if (!substream->oss.oss) #endif if (atomic_read(&substream->mmap_count)) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index f016be72b100..f15dbb8001f1 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1530,7 +1530,7 @@ static int snd_rawmidi_dev_free(struct snd_device *device) return snd_rawmidi_free(rmidi); } -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_ENABLED(CONFIG_SND_RAWMIDI_SEQ) static void snd_rawmidi_dev_seq_free(struct snd_seq_device *device) { struct snd_rawmidi *rmidi = device->private_data; @@ -1605,7 +1605,7 @@ static int snd_rawmidi_dev_register(struct snd_device *device) } } rmidi->proc_entry = entry; -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_ENABLED(CONFIG_SND_RAWMIDI_SEQ) if (!rmidi->ops || !rmidi->ops->dev_register) { /* own registration mechanism */ if (snd_seq_device_new(rmidi->card, rmidi->device, SNDRV_SEQ_DEV_ID_MIDISYNTH, 0, &rmidi->seq_dev) >= 0) { rmidi->seq_dev->private_data = rmidi; diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c index e85e72baff9e..b27223746b0e 100644 --- a/sound/core/rtctimer.c +++ b/sound/core/rtctimer.c @@ -27,7 +27,7 @@ #include <sound/core.h> #include <sound/timer.h> -#if defined(CONFIG_RTC) || defined(CONFIG_RTC_MODULE) +#if IS_ENABLED(CONFIG_RTC) #include <linux/mc146818rtc.h> @@ -185,4 +185,4 @@ MODULE_LICENSE("GPL"); MODULE_ALIAS("snd-timer-" __stringify(SNDRV_TIMER_GLOBAL_RTC)); -#endif /* CONFIG_RTC || CONFIG_RTC_MODULE */ +#endif /* IS_ENABLED(CONFIG_RTC) */ diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index 726a49ac9725..5391c5e139ef 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -21,7 +21,7 @@ #ifdef CONFIG_SND_OSSEMUL -#if !defined(CONFIG_SOUND) && !(defined(MODULE) && defined(CONFIG_SOUND_MODULE)) +#if !IS_ENABLED(CONFIG_SOUND) #error "Enable the OSS soundcore multiplexer (CONFIG_SOUND) in the kernel." #endif diff --git a/sound/core/timer.c b/sound/core/timer.c index 6ddcf06f52f9..cbec51464c5b 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -35,9 +35,9 @@ #include <sound/initval.h> #include <linux/kmod.h> -#if defined(CONFIG_SND_HRTIMER) || defined(CONFIG_SND_HRTIMER_MODULE) +#if IS_ENABLED(CONFIG_SND_HRTIMER) #define DEFAULT_TIMER_LIMIT 4 -#elif defined(CONFIG_SND_RTCTIMER) || defined(CONFIG_SND_RTCTIMER_MODULE) +#elif IS_ENABLED(CONFIG_SND_RTCTIMER) #define DEFAULT_TIMER_LIMIT 2 #else #define DEFAULT_TIMER_LIMIT 1 diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c index 33d9a857a262..d7091e56d0d0 100644 --- a/sound/drivers/opl3/opl3_lib.c +++ b/sound/drivers/opl3/opl3_lib.c @@ -530,7 +530,7 @@ int snd_opl3_hwdep_new(struct snd_opl3 * opl3, opl3->hwdep = hw; opl3->seq_dev_num = seq_device; -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_ENABLED(CONFIG_SND_OPL3_LIB_SEQ) if (snd_seq_device_new(card, seq_device, SNDRV_SEQ_DEV_ID_OPL3, sizeof(struct snd_opl3 *), &opl3->seq_dev) >= 0) { strcpy(opl3->seq_dev->name, hw->name); diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index 742a4b642fd9..ddcc1a325a61 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -24,7 +24,7 @@ #include <sound/opl3.h> #include <sound/asound_fm.h> -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) #define OPL3_SUPPORT_SYNTH #endif diff --git a/sound/drivers/opl4/opl4_lib.c b/sound/drivers/opl4/opl4_lib.c index b953fb4aa298..c88679291091 100644 --- a/sound/drivers/opl4/opl4_lib.c +++ b/sound/drivers/opl4/opl4_lib.c @@ -153,7 +153,7 @@ static int snd_opl4_detect(struct snd_opl4 *opl4) return 0; } -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_ENABLED(CONFIG_SND_OPL4_LIB_SEQ) static void snd_opl4_seq_dev_free(struct snd_seq_device *seq_dev) { struct snd_opl4 *opl4 = seq_dev->private_data; @@ -253,7 +253,7 @@ int snd_opl4_create(struct snd_card *card, snd_opl4_create_proc(opl4); #endif -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_ENABLED(SND_OPL4_LIB_SEQ) opl4->seq_client = -1; if (opl4->hardware < OPL3_HW_OPL4_ML) snd_opl4_create_seq_dev(opl4, seq_device); diff --git a/sound/drivers/opl4/opl4_local.h b/sound/drivers/opl4/opl4_local.h index 470e5a758a02..be10b3981365 100644 --- a/sound/drivers/opl4/opl4_local.h +++ b/sound/drivers/opl4/opl4_local.h @@ -184,7 +184,7 @@ struct snd_opl4 { #endif struct mutex access_mutex; -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(SND_OPL4_LIB_SEQ) int used; int seq_dev_num; diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 45fcdff611f9..83de44cd74a3 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -1139,7 +1139,7 @@ snd_emu8000_new(struct snd_card *card, int index, long port, int seq_ports, snd_emu8000_free(hw); return err; } -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_ENABLED(CONFIG_SND_SBAWE_SEQ) if (snd_seq_device_new(card, index, SNDRV_SEQ_DEV_ID_EMU8000, sizeof(struct snd_emu8000*), &awe) >= 0) { strcpy(awe->name, "EMU-8000"); diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index a4130993955f..2bfbb3aaf51b 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -62,7 +62,7 @@ MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB AWE 32}," #define SNDRV_DEBUG_IRQ #endif -#if defined(SNDRV_SBAWE) && (defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE))) +#if defined(SNDRV_SBAWE) && IS_ENABLED(CONFIG_SND_SBAWE_SEQ) #define SNDRV_SBAWE_EMU8000 #endif diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 9e1bd0c39a8c..887540ff5c62 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -37,7 +37,7 @@ MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB Live!/PCI512/E-mu APS}," "{Creative Labs,SB Audigy}}"); -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_ENABLED(CONFIG_SND_EMU10K1_SEQ) #define ENABLE_SYNTH #include <sound/emu10k1_synth.h> #endif diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e87eac552ec5..17c83fbc9f84 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -4137,7 +4137,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { .driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA }, { PCI_DEVICE(0x1102, 0x0012), .driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA }, -#if !defined(CONFIG_SND_CTXFI) && !defined(CONFIG_SND_CTXFI_MODULE) +#if !IS_ENABLED(CONFIG_SND_CTXFI) /* the following entry conflicts with snd-ctxfi driver, * as ctxfi driver mutates from HD-audio to native mode with * a special command sequence. -- 1.8.5.2

1 0

[alsa-devel] [RFC v2][PATCH 00/38] Enhancement for support of some firewire devices
by Takashi Sakamoto 09 Feb '14

09 Feb '14

This series of patch is to update patches which I've posted below: [RFC][PATCH 00/17] Enhancement for firewire-lib http://mailman.alsa-project.org/pipermail/alsa-devel/2013-November/069163.h… [RFC][PATCH 00/13] a new driver for BridgeCo BeBoB based device http://mailman.alsa-project.org/pipermail/alsa-devel/2013-November/069183.h… [PATCH v2 0/8][RFC] a driver for Fireworks based devices http://mailman.alsa-project.org/pipermail/alsa-devel/2013-December/070043.h… In this time, I'll post all patches because now they have dependency each other. Each driver has lines in asound.h/firewire.h. Changes since previous RFC version of my patches: [firewire-lib] - support 32bit sample for capturing (remove 16bit sample support) - limit the number of MIDI conformant data channel by 1 - change the way to trigger MIDI messages - fix some bugs to playback/capture samples in 'dual wire' mode - fix wrong mappling of channels in 'dual wire' mode - add unit/subunit support for PLUG INFO command - add limit for dev_err() [fireworks] - improve return value for hwdep interface [bebob] - reconfirm supported devices - read root directory in config rom to get vendor name charactors - add a new way to detect clock source - remove model specific operations for PreSonus - split codes for M-Audio to three patches - remove control interface except for M-Audio special devices Best regards, Takashi Sakamoto Takashi Sakamoto (38): firewire-lib: Rename functions, structure, member for AMDTP firewire-lib: Add macros instead of fixed value for AMDTP firewire-lib: Add 'direction' member to 'amdtp_stream' structure firewire-lib: Split some codes into functions to reuse for both streams firewire-lib: Add support for AMDTP in-stream and PCM capture firewire-lib: Add support for MIDI capture/playback firewire-lib: Give syt value as parameter to handle_out_packet() firewire-lib: Add support for duplex streams synchronization in blocking mode firewire-lib: Add sort function for transmitted packet firewire-lib: Add transfer delay to synchronized duplex streams firewire-lib: Add support for channel mapping firewire-lib: Rename macros, variables and functions for CMP firewire-lib: Add 'direction' member to 'cmp_connection' structure firewire-lib: Add handling output connection by CMP firewire-lib: Add a new function to check others' connection firewire-lib: Add some AV/C general commands firewire-lib: Add quirks for Fireworks fireworks: Add skelton for Fireworks based devices fireworks: Add transaction and some commands fireworks: Add connection and stream management fireworks: Add proc interface for debugging purpose fireworks: Add MIDI interface fireworks: Add PCM interface fireworks: Add hwdep interface fireworks: Add command/response functionality into hwdep interface bebob: Add skelton for BeBoB based devices bebob: Add commands and connections/streams management bebob: Add proc interface for debugging purpose bebob: Add MIDI interface bebob: Add PCM interface bebob: Add hwdep interface bebob: Prepare for device specific operations bebob: Add support for Terratec PHASE, EWS series and Aureon bebob: Add support for Yamaha GO series bebob: Add support for Focusrite Saffire/SaffirePro series bebob: Add support for M-Audio usual Firewire series bebob: Add support for M-Audio special Firewire series bebob: Send a cue to load firmware for M-Audio Firewire series include/uapi/sound/asound.h | 4 +- include/uapi/sound/firewire.h | 20 + sound/firewire/Kconfig | 55 ++ sound/firewire/Makefile | 2 + sound/firewire/amdtp.c | 917 +++++++++++++++++------ sound/firewire/amdtp.h | 163 +++- sound/firewire/bebob/Makefile | 4 + sound/firewire/bebob/bebob.c | 430 +++++++++++ sound/firewire/bebob/bebob.h | 268 +++++++ sound/firewire/bebob/bebob_command.c | 352 +++++++++ sound/firewire/bebob/bebob_focusrite.c | 289 +++++++ sound/firewire/bebob/bebob_hwdep.c | 197 +++++ sound/firewire/bebob/bebob_maudio.c | 852 +++++++++++++++++++++ sound/firewire/bebob/bebob_midi.c | 156 ++++ sound/firewire/bebob/bebob_pcm.c | 430 +++++++++++ sound/firewire/bebob/bebob_proc.c | 185 +++++ sound/firewire/bebob/bebob_stream.c | 853 +++++++++++++++++++++ sound/firewire/bebob/bebob_terratec.c | 68 ++ sound/firewire/bebob/bebob_yamaha.c | 50 ++ sound/firewire/cmp.c | 217 ++++-- sound/firewire/cmp.h | 14 +- sound/firewire/dice.c | 46 +- sound/firewire/fcp.c | 151 ++++ sound/firewire/fcp.h | 21 + sound/firewire/fireworks/Makefile | 4 + sound/firewire/fireworks/fireworks.c | 330 ++++++++ sound/firewire/fireworks/fireworks.h | 222 ++++++ sound/firewire/fireworks/fireworks_command.c | 397 ++++++++++ sound/firewire/fireworks/fireworks_hwdep.c | 299 ++++++++ sound/firewire/fireworks/fireworks_midi.c | 166 ++++ sound/firewire/fireworks/fireworks_pcm.c | 449 +++++++++++ sound/firewire/fireworks/fireworks_proc.c | 206 +++++ sound/firewire/fireworks/fireworks_stream.c | 354 +++++++++ sound/firewire/fireworks/fireworks_transaction.c | 327 ++++++++ sound/firewire/speakers.c | 94 +-- 35 files changed, 8196 insertions(+), 396 deletions(-) create mode 100644 sound/firewire/bebob/Makefile create mode 100644 sound/firewire/bebob/bebob.c create mode 100644 sound/firewire/bebob/bebob.h create mode 100644 sound/firewire/bebob/bebob_command.c create mode 100644 sound/firewire/bebob/bebob_focusrite.c create mode 100644 sound/firewire/bebob/bebob_hwdep.c create mode 100644 sound/firewire/bebob/bebob_maudio.c create mode 100644 sound/firewire/bebob/bebob_midi.c create mode 100644 sound/firewire/bebob/bebob_pcm.c create mode 100644 sound/firewire/bebob/bebob_proc.c create mode 100644 sound/firewire/bebob/bebob_stream.c create mode 100644 sound/firewire/bebob/bebob_terratec.c create mode 100644 sound/firewire/bebob/bebob_yamaha.c create mode 100644 sound/firewire/fireworks/Makefile create mode 100644 sound/firewire/fireworks/fireworks.c create mode 100644 sound/firewire/fireworks/fireworks.h create mode 100644 sound/firewire/fireworks/fireworks_command.c create mode 100644 sound/firewire/fireworks/fireworks_hwdep.c create mode 100644 sound/firewire/fireworks/fireworks_midi.c create mode 100644 sound/firewire/fireworks/fireworks_pcm.c create mode 100644 sound/firewire/fireworks/fireworks_proc.c create mode 100644 sound/firewire/fireworks/fireworks_stream.c create mode 100644 sound/firewire/fireworks/fireworks_transaction.c -- 1.8.3.2

4 50

[alsa-devel] Workaround for "[AO_ALSA] Unable to set buffer time near: Invalid argument" error on HDMI 5.1 output
by Thilo Schulz 09 Feb '14

09 Feb '14

Hi, the purpose of this message is just to document my workaround solution I found to a bug (probably in ALSA) on a stable Debian system. This took me the better of a fine sunday and I'm posting this to spare anyone who might run into a similar problem the same trouble. And I'm posting this so the alsa devs may look into this (if it's not already fixed in recent alsa versions). Using mplayer, I recently tried outputting a 6 channel test wave file to the HDMI output on my Acer Aspire with a HDA Intel PCH card. 2 channels work fine, but trying to output 6 channels results in: $ mplayer test.wav -ao alsa:device=hdmi -channels 6 [...] [AO_ALSA] Unable to set buffer time near: Invalid argument Failed to initialize audio driver 'alsa:device=hdmi' Obviously, this code from mplayer's ao_alsa.c fails: unsigned int alsa_buffer_time = 500000; /* 0.5 s */ [...] if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, \ alsa_hwparams, &alsa_buffer_time, NULL)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetBufferTimeNear, snd_strerror(err)); return 0; } A similar issue seems to have been discussed in this thread on alsa-devel: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-March/037450.html And as a workaround it is suggested there to raise some memory preallocation parameter: $ cat /proc/asound/card0/pcm3p/sub0/prealloc 64 like so: $ echo 128 > /proc/asound/card0/pcm3p/sub0/prealloc this indeed fixes the problem and mplayer plays fine. Now here comes the curious part. _Lowering_ the parameter like this: $ echo 63 > /proc/asound/card0/pcm3p/sub0/prealloc makes the problem go away as well. Strange. For followup replies, please CC me, as I'm subscribed to neither mailing lists. To ALSA devs: I would be willing to help debug this issue if you'd give a few instructions (where to put a few printk messages in the module, etc) -- Beste regards, Thilo Schulz

1 0

[alsa-devel] [PATCH 25/28] Remove MACH_SMDKV310
by Richard Weinberger 09 Feb '14

09 Feb '14

The symbol is an orphan, get rid of it. Signed-off-by: Richard Weinberger <richard(a)nod.at> --- 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 7ea576c..33fd372 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) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help -- 1.8.4.2

1 0

[alsa-devel] [PATCH v2 0/7] simple card
by Xiubo Li 08 Feb '14

08 Feb '14

This patch series has been test on VF610 Tower board. Changed in V2: - Revise the problem from Jean-Francois Moine. - Add binding documentation patch. Xiubo Li (7): ASoC: simple-card: fix __asoc_simple_card_dai_init ASoC: simple-card: simplify the daifmt code ASoC: simple-card: Add snd_card's name parsing from DT node support ASoC: simple-card: add tdm slot supports ASoC: add snd_soc_of_parse_audio_simple_widgets for DT ASoC: simple-card: add off-codec widgets supports. ASoC: binding: for new properties documenting and usage .../devicetree/bindings/sound/simple-card.txt | 17 +++- include/sound/simple_card.h | 8 ++ include/sound/soc.h | 2 + sound/soc/generic/simple-card.c | 112 +++++++++++++++++---- sound/soc/soc-core.c | 87 ++++++++++++++++ 5 files changed, 204 insertions(+), 22 deletions(-) -- 1.8.4

3 16

[alsa-devel] v3.14-rc1 locking bug in system suspend
by Shawn Guo 08 Feb '14

08 Feb '14

I start seeing the following locking bug on IMX6 with suspend operation after moving to v3.14-rc1. Before I start looking into the issue, I would like to confirm if it's an IMX specific problem or one seen on other platforms. Please ask if you need more info. Shawn root@arm:~# echo mem > /sys/power/state PM: Syncing filesystems ... done. PM: Preparing system for mem sleep mmc1: card e624 removed Freezing user space processes ... (elapsed 0.002 seconds) done. Freezing remaining freezable tasks ... (elapsed 0.002 seconds) done. PM: Entering mem sleep INFO: trying to register non-static key. the code is fine but needs lockdep annotation. turning off the locking correctness validator. CPU: 0 PID: 1849 Comm: bash Not tainted 3.14.0-rc1+ #1648 Backtrace: [<80012144>] (dump_backtrace) from [<800122e4>] (show_stack+0x18/0x1c) r6:808c48a0 r5:00000000 r4:00000000 r3:00000000 [<800122cc>] (show_stack) from [<80637ac0>] (dump_stack+0x78/0x94) [<80637a48>] (dump_stack) from [<80064fa0>] (__lock_acquire+0x18f4/0x1ce0) r4:00000002 r3:bda3f200 [<800636ac>] (__lock_acquire) from [<80065894>] (lock_acquire+0x68/0x7c) r10:bd906000 r9:bda3f200 r8:00000000 r7:00000000 r6:60000013 r5:bd906000 r4:00000000 [<8006582c>] (lock_acquire) from [<8063c938>] (mutex_lock_nested+0x5c/0x3b8) r7:00000000 r6:80dfc78c r5:804be444 r4:bf122928 [<8063c8dc>] (mutex_lock_nested) from [<804be444>] (snd_soc_suspend+0x34/0x42c) r10:00000000 r9:00000000 r8:00000000 r7:bf1c4444 r6:bf1c4410 r5:be8fd150 r4:be8fd010 [<804be410>] (snd_soc_suspend) from [<8034392c>] (platform_pm_suspend+0x34/0x64) r10:00000000 r8:00000000 r7:bf1c4444 r6:bf1c4410 r5:803438f8 r4:bf1c4410 [<803438f8>] (platform_pm_suspend) from [<80348e18>] (dpm_run_callback.isra.7+0x34/0x6c) [<80348de4>] (dpm_run_callback.isra.7) from [<80349354>] (__device_suspend+0x10c/0x220) r9:808dd974 r8:808c4a5c r6:00000002 r5:80e5001c r4:bf1c4410 [<80349248>] (__device_suspend) from [<8034a338>] (dpm_suspend+0x60/0x220) r7:bf1c4410 r6:808dd90c r5:80e5001c r4:bf1c44c0 [<8034a2d8>] (dpm_suspend) from [<8034a790>] (dpm_suspend_start+0x60/0x68) r10:8079a818 r9:00000000 r8:00000004 r7:80dfbe90 r6:80641eec r5:00000000 r4:00000002 [<8034a730>] (dpm_suspend_start) from [<8006a788>] (suspend_devices_and_enter+0x74/0x318) r4:00000003 r3:80dfbe98 [<8006a714>] (suspend_devices_and_enter) from [<8006abd8>] (pm_suspend+0x1ac/0x244) r10:8079a818 r8:00000004 r7:00000003 r6:80641eec r5:00000000 r4:00000003 [<8006aa2c>] (pm_suspend) from [<80069a4c>] (state_store+0x70/0xc0) r5:00000003 r4:be9c7680 [<800699dc>] (state_store) from [<80294034>] (kobj_attr_store+0x1c/0x28) r10:bd99cb88 r8:00000000 r7:bd907f78 r6:be9c7680 r5:00000004 r4:bd99cb80 [<80294018>] (kobj_attr_store) from [<80140f90>] (sysfs_kf_write+0x54/0x58) [<80140f3c>] (sysfs_kf_write) from [<8014474c>] (kernfs_fop_write+0xc4/0x160) r6:be9c7680 r5:bd99cb80 r4:00000004 r3:80140f3c [<80144688>] (kernfs_fop_write) from [<800dfa14>] (vfs_write+0xbc/0x184) r10:00000000 r9:00000000 r8:00000000 r7:bd907f78 r6:00281c08 r5:00000004 r4:bda1bd00 [<800df958>] (vfs_write) from [<800dfe00>] (SyS_write+0x48/0x70) r10:00000000 r8:00000000 r7:00000004 r6:00281c08 r5:00000000 r4:bda1bd00 [<800dfdb8>] (SyS_write) from [<8000e800>] (ret_fast_syscall+0x0/0x48) r9:bd906000 r8:8000e9c4 r7:00000004 r6:00281c08 r5:00000004 r4:76f575e0 INFO: rcu_sched detected stalls on CPUs/tasks: { 0} (detected by 1, t=2104 jiffies, g=576, c=575, q=2) Task dump for CPU 0: bash R running 0 1849 1848 0x00000002 Backtrace: [<80065894>] (lock_acquire) from [<80061410>] (trace_hardirqs_off+0x14/0x18) Backtrace aborted due to bad frame pointer <bd907b3c> INFO: rcu_sched detected stalls on CPUs/tasks: { 0} (detected by 1, t=8424 jiffies, g=576, c=575, q=3) Task dump for CPU 0: bash R running 0 1849 1848 0x00000002 Backtrace: [<80065894>] (lock_acquire) from [<80061410>] (trace_hardirqs_off+0x14/0x18) Backtrace aborted due to bad frame pointer <bd907b3c>

1 2

[alsa-devel] [PATCH RFC 0/2] drivers/base: simplify simple DT-based components
by Jean-Francois Moine 08 Feb '14

08 Feb '14

This patch series tries to simplify the code of simple devices in case they are part of componentised subsystems, are declared in a DT, and are not using the component bin/unbind functions. Jean-Francois Moine (2): drivers/base: permit base components to omit the bind/unbind ops drivers/base: declare phandle DT nodes as components drivers/base/component.c | 21 +++++++++++++++++++-- drivers/base/core.c | 18 ++++++++++++++++++ include/linux/of.h | 2 ++ 3 files changed, 39 insertions(+), 2 deletions(-) -- 1.9.rc1

2 9

[alsa-devel] Error trying to compile driver
by Michael Weiner 08 Feb '14

08 Feb '14

I am trying to fix the sound on a laptop i am working on. Has CentOS 5.10 64bit installed and i am trying to use the alsa-compile.sh script but it fails with the following error: make -C /lib/modules/2.6.18-371.4.1.el5/build SUBDIRS=/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d CPP="gcc -E" CC="gcc" modules make[1]: Entering directory `/usr/src/kernels/2.6.18-371.4.1.el5-x86_64' CC [M] /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/hwdep.o In file included from /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/include/adriver.h:1695, from /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/hwdep.c:1: include/pcmcia/cistpl.h:557: error: expected specifier-qualifier-list before 'cisdata_t' include/pcmcia/cistpl.h:586: error: expected specifier-qualifier-list before 'cisdata_t' include/pcmcia/cistpl.h:590: warning: 'struct pcmcia_socket' declared inside parameter list include/pcmcia/cistpl.h:590: warning: its scope is only this definition or declaration, which is probably not what you want include/pcmcia/cistpl.h:593: warning: 'struct pcmcia_socket' declared inside parameter list include/pcmcia/cistpl.h:594: warning: 'struct pcmcia_socket' declared inside parameter list include/pcmcia/cistpl.h:595: warning: 'struct pcmcia_socket' declared inside parameter list include/pcmcia/cistpl.h:598: warning: 'struct pcmcia_socket' declared inside parameter list In file included from include/pcmcia/ds.h:23, from /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/include/adriver.h:1696, from /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/hwdep.c:1: include/pcmcia/bulkmem.h:27: error: expected specifier-qualifier-list before 'memory_handle_t' include/pcmcia/bulkmem.h:38: warning: 'struct pcmcia_device' declared inside parameter list include/pcmcia/bulkmem.h:39: warning: 'struct pcmcia_device' declared inside parameter list In file included from /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/include/adriver.h:1696, from /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/hwdep.c:1: include/pcmcia/ds.h:35: error: expected specifier-qualifier-list before 'win_req_t' include/pcmcia/ds.h:55: error: expected specifier-qualifier-list before 'adjust_t' make[3]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/hwdep.o] Error 1 make[2]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore] Error 2 make[1]: *** [_module_/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.18-371.4.1.el5-x86_64' make: *** [compile] Error 2 Working directory /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d. Compilation of alsa-driver failed. Report this problem to the <alsa-devel(a)alsa-project.org> mailing list. Can anyone help me figure out why this is failing? Kernel is 2.6.18-371.4.1 Thank you in advance Michael Weiner --

1 0

[alsa-devel] [PATCH v3] ASoC: add RT286 CODEC driver
by bardliao＠realtek.com 07 Feb '14

07 Feb '14

From: Bard Liao <bardliao(a)realtek.com> This patch adds the ALC286 codec driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * Initialize all cache registers in rt286_reg_cache, and remove rt286_reg_cache size define. --- include/sound/rt286.h | 19 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt286.c | 1373 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 89 +++ 5 files changed, 1487 insertions(+) create mode 100644 include/sound/rt286.h create mode 100644 sound/soc/codecs/rt286.c create mode 100644 sound/soc/codecs/rt286.h diff --git a/include/sound/rt286.h b/include/sound/rt286.h new file mode 100644 index 0000000..6c0c1c9 --- /dev/null +++ b/include/sound/rt286.h @@ -0,0 +1,19 @@ +/* + * linux/sound/rt286.h -- Platform data for RT286 + * + * Copyright 2013 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT286_H +#define __LINUX_SND_RT286_H + +struct rt286_platform_data { + bool cbj_en; /*combo jack enable*/ + bool irq_en; /*IRQ enable*/ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087a..755a39d 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_RT286 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_RT286 + tristate + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc12676..b31615c 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-rt286-objs := rt286.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-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(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/rt286.c b/sound/soc/codecs/rt286.c new file mode 100644 index 0000000..214e5fc --- /dev/null +++ b/sound/soc/codecs/rt286.c @@ -0,0 +1,1373 @@ +/* + * rt286.c -- RT286 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/rt286.h> + +#include "../../pci/hda/hda_codec.h" +#include "rt286.h" + +struct rt286_priv { + struct regmap *regmap; + struct rt286_platform_data pdata; + int sys_clk; +}; + +static unsigned int rt286_reg_cache[] = { + [NODE_ID_AUDIO_FUNCTION_GROUP] = 0x0000, + [NODE_ID_DAC_OUT1] = 0x7f7f, + [NODE_ID_DAC_OUT2] = 0x7f7f, + [NODE_ID_SPDIF] = 0x0000, + [NODE_ID_ADC_IN1] = 0x4343, + [NODE_ID_ADC_IN2] = 0x4343, + [NODE_ID_MIC1] = 0x0000, + [NODE_ID_MIXER_IN] = 0x000b, + [NODE_ID_MIXER_OUT1] = 0x0002, + [NODE_ID_MIXER_OUT2] = 0x0000, + [NODE_ID_DMIC1] = 0x0000, + [NODE_ID_DMIC2] = 0x0000, + [NODE_ID_LINE1] = 0x0000, + [NODE_ID_BEEP] = 0x0000, + [NODE_ID_VENDOR_REGISTERS] = 0x0000, + [NODE_ID_SPK_OUT] = 0x0000, + [NODE_ID_HP_OUT] = 0x0000, + [NODE_ID_MIXER_IN1] = 0x0000, + [NODE_ID_MIXER_IN2] = 0x0000, +}; + +static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + int ret; + unsigned int buf = 0x0; + + if (reg <= 0xff) { /*read cache*/ + *value = rt286_reg_cache[reg]; + return 0; + } + reg = cpu_to_be32(reg); + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 4; + xfer[0].buf = (u8 *)® + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 4; + xfer[1].buf = (u8 *)&buf; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret < 0) + return ret; + else if (ret != 2) + return -EIO; + + *value = be32_to_cpu(buf); + + return 0; +} + +static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) +{ + struct i2c_client *client = context; + u8 data[4]; + int ret; + + if (reg <= 0xff) { /*write cache*/ + rt286_reg_cache[reg] = value; + return 0; + } + data[0] = (reg >> 24) & 0xff; + data[1] = (reg >> 16) & 0xff; + data[2] = (reg >> 8) & 0xff; + data[3] = reg & 0xff; + + ret = i2c_master_send(client, data, 4); + if (ret == 4) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +static unsigned int rt286_read(struct snd_soc_codec *codec, + unsigned int vid, + unsigned int nid, + unsigned int data) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb; + unsigned int buf = 0x0; + + verb = (nid << 20) | (vid << 8) | data; + + regmap_read(rt286->regmap, verb, &buf); + return buf; +} + +static int rt286_write(struct snd_soc_codec *codec, + unsigned int vid, + unsigned int nid, + unsigned int value) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + + reg = (nid << 20) | (vid << 8) | value; + + return regmap_write(rt286->regmap, reg, 0); +} + +static int rt286_update_bits(struct snd_soc_codec *codec, unsigned int vid, + unsigned int nid, unsigned int data, + unsigned int mask, unsigned int value) +{ + unsigned int old, new; + int change, ret; + + old = rt286_read(codec, vid | 0x800, nid, data); + new = (old & ~mask) | (value & mask); + change = old != new; + + if (change) { + ret = rt286_write(codec, vid, nid, new); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write private reg: %d\n", ret); + goto err; + } + } + return change; + +err: + return ret; +} + +static int rt286_index_write(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index, unsigned int data) +{ + int ret; + + ret = rt286_write(codec, AC_VERB_SET_COEF_INDEX, + wid, index); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + goto err; + } + ret = rt286_write(codec, AC_VERB_SET_PROC_COEF, + wid, data); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + + return 0; +err: + return ret; +} + +static unsigned int rt286_index_read(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index) +{ + int ret; + + ret = rt286_write(codec, AC_VERB_SET_COEF_INDEX, + wid, index); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + return ret; + } + return rt286_read(codec, AC_VERB_GET_PROC_COEF, + wid, index); +} + +static int rt286_index_update_bits(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index, + unsigned int mask, unsigned int data) +{ + unsigned int old, new; + int change, ret; + + old = rt286_index_read(codec, wid, index); + new = (old & ~mask) | (data & mask); + change = old != new; + + if (change) { + ret = rt286_index_write(codec, wid, index, new); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write private reg: %d\n", ret); + goto err; + } + } + return change; + +err: + return ret; +} + +static int rt286_support_power_controls[] = { + NODE_ID_DAC_OUT1, + NODE_ID_DAC_OUT2, + NODE_ID_ADC_IN1, + NODE_ID_ADC_IN2, + NODE_ID_MIC1, + NODE_ID_DMIC1, + NODE_ID_DMIC2, + NODE_ID_SPK_OUT, + NODE_ID_HP_OUT, +}; +#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls) + +static const struct reg_default rt286_reg[] = { + { NODE_ID_DAC_OUT1, 0x7f7f }, + { NODE_ID_DAC_OUT2, 0x7f7f }, + { NODE_ID_SPDIF, 0x0000 }, + { NODE_ID_ADC_IN1, 0x4343 }, + { NODE_ID_ADC_IN2, 0x4343 }, + { NODE_ID_MIC1, 0x0000 }, + { NODE_ID_MIXER_IN, 0x000b }, + { NODE_ID_MIXER_OUT1, 0x0002 }, + { NODE_ID_MIXER_OUT2, 0x0000 }, + { NODE_ID_SPK_OUT, 0x0000 }, + { NODE_ID_HP_OUT, 0x0000 }, + { NODE_ID_MIXER_IN1, 0x0005 }, + { NODE_ID_MIXER_IN2, 0x0005 }, +}; + +static bool rt286_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NODE_ID_DAC_OUT1: + case NODE_ID_DAC_OUT2: + case NODE_ID_ADC_IN1: + case NODE_ID_ADC_IN2: + case NODE_ID_MIC1: + case NODE_ID_SPDIF: + case NODE_ID_MIXER_IN: + case NODE_ID_MIXER_OUT1: + case NODE_ID_MIXER_OUT2: + case NODE_ID_DMIC1: + case NODE_ID_DMIC2: + case NODE_ID_SPK_OUT: + case NODE_ID_HP_OUT: + case NODE_ID_MIXER_IN1: + case NODE_ID_MIXER_IN2: + return true; + default: + return false; + } +} + +int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int i; + + *hp = false; + *mic = false; + + if (rt286->pdata.cbj_en) { + *hp = rt286_read(codec, AC_VERB_GET_PIN_SENSE, + NODE_ID_HP_OUT, 0); + if (*hp) { + /* power on HV,VERF */ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x03, 0x1001, 0x0); + /* power LDO1 */ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x08, 0x4, 0x4); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_MIC1, 0x24); + rt286_write(codec, AC_VERB_SET_COEF_INDEX, + 0x20, 0x50); + val = rt286_read(codec, AC_VERB_GET_PROC_COEF, + 0x20, 0x50); + + msleep(200); + i = 40; + while (((val & 0x0800) == 0) && (i > 0)) { + rt286_write(codec, AC_VERB_SET_COEF_INDEX, + 0x20, 0x50); + val = rt286_read(codec, + AC_VERB_GET_PROC_COEF, 0x20, 0x50); + i--; + msleep(20); + } + + if (0x0400 == (val & 0x0700)) { + *mic = false; + + rt286_write(codec, + AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_MIC1, 0x20); + /* power off HV,VERF */ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x03, 0x1001, 0x1001); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x04, 0xC000, 0x0000); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x4f, 0x0030, 0x0000); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x02, 0xc000, 0x0000); + } else if ((0x0200 == (val & 0x0700)) || + (0x0100 == (val & 0x0700))) { + *mic = true; + } else { + *mic = false; + } + + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x20, 0x0060, 0x0000); + } else { + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x20, 0x0060, 0x0020); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x04, 0xC000, 0x8000); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x4f, 0x0030, 0x0020); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x02, 0xc000, 0x8000); + + *mic = false; + } + } else { + *hp = rt286_read(codec, AC_VERB_GET_PIN_SENSE, + NODE_ID_HP_OUT, 0); + *mic = rt286_read(codec, AC_VERB_GET_PIN_SENSE, + NODE_ID_MIC1, 0); + } + + + /* Clear IRQ */ + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x33, 0x1, 0x1); + return 0; +} +EXPORT_SYMBOL_GPL(rt286_jack_detect); + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static int rt286_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int io = 0x8000; + + if (NODE_ID_ADC_IN1 == mc->reg) + io = 0; + ucontrol->value.integer.value[0] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x2000 | io) & 0x7f; + ucontrol->value.integer.value[1] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, io) & 0x7f; + + return 0; +} + +static int rt286_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int vall, valr, io; + + if (NODE_ID_ADC_IN1 == mc->reg) + io = 0x4000; + else + io = 0x8000; + vall = ucontrol->value.integer.value[0]; + valr = ucontrol->value.integer.value[1]; + if (vall == valr) { + vall = vall | 0x3000 | io; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + } else { + vall = vall | 0x2000 | io; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + valr = valr | 0x1000 | io; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, valr); + } + + return 0; +} + +static int rt286_mic_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x0) & 0x3; + + return 0; +} + +static int rt286_mic_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int val; + + val = ucontrol->value.integer.value[0]; + val = val | 0x7000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, val); + + return 0; +} + +static int rt286_playback_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + + ucontrol->value.integer.value[0] = + !(rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0xa000) & 0x80); + ucontrol->value.integer.value[1] = + !(rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x8000) & 0x80); + + return 0; +} + +static int rt286_playback_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int vall, valr, val = 0; + + vall = (!ucontrol->value.integer.value[0] << 7); + valr = (!ucontrol->value.integer.value[1] << 7); + if (vall) + val |= 0x8000; + if (valr) + val |= 0x0080; + + if (vall == valr) { + vall = vall | 0xb000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + } else { + vall = vall | 0xa000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + valr = valr | 0x9000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, valr); + } + snd_soc_update_bits(codec, mc->reg, 0x8080, val); + + return 0; +} + +static const struct snd_kcontrol_new rt286_snd_controls[] = { + SOC_DOUBLE_EXT_TLV("DAC0 Playback Volume", NODE_ID_DAC_OUT1, + 8, 0, 0x7f, 0, + rt286_vol_get, rt286_vol_put, + out_vol_tlv), + SOC_DOUBLE_EXT_TLV("ADC0 Capture Volume", NODE_ID_ADC_IN1, + 8, 0, 0x7f, 0, + rt286_vol_get, rt286_vol_put, + out_vol_tlv), + SOC_SINGLE_EXT_TLV("AMIC Volume", NODE_ID_MIC1, 0, 0x3, 0, + rt286_mic_gain_get, rt286_mic_gain_put, + mic_vol_tlv), + SOC_DOUBLE_EXT("Headphone Playback Switch", NODE_ID_HP_OUT, + 15, 8, 1, 1, rt286_playback_switch_get, + rt286_playback_switch_put), + SOC_DOUBLE_EXT("Speaker Playback Switch", NODE_ID_SPK_OUT, + 15, 8, 1, 1, rt286_playback_switch_get, + rt286_playback_switch_put), +}; + +static void hw_configure(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int i; + + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_AUDIO_FUNCTION_GROUP, AC_PWRST_D3); + for (i = 0; i < RT286_POWER_REG_LEN; i++) { + rt286_write(codec, AC_VERB_SET_POWER_STATE, + rt286_support_power_controls[i], AC_PWRST_D3); + } + + if (rt286->pdata.irq_en) + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x33, 0x0002, 0x0002); + + if (!rt286->pdata.cbj_en) { + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x50, 0x0000); + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x19, 0x0816); + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x20, 0x0000); + } + + mdelay(10); + + /* Set configuration default */ + rt286_write(codec, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + NODE_ID_MIC1, 0x00); + rt286_write(codec, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + NODE_ID_DMIC1, 0x00); + rt286_write(codec, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + NODE_ID_DMIC2, 0x00); + + mdelay(10); + + if (rt286->pdata.cbj_en) + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x4f, 0x5029); + else + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x4f, 0xb029); + + /*Power down LDO2*/ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x08, 0x0008, 0x0000); +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt286_front_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", NODE_ID_MIXER_OUT1, + RT286_M_FRONT_DAC_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIX Switch", NODE_ID_MIXER_OUT1, + RT286_M_FRONT_REC_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt286_rec_mix[] = { + SOC_DAPM_SINGLE("Beep Switch", NODE_ID_MIXER_IN, + RT286_M_REC_BEEP_SFT, 1, 1), + SOC_DAPM_SINGLE("Line1 Switch", NODE_ID_MIXER_IN, + RT286_M_REC_LINE1_SFT, 1, 1), + SOC_DAPM_SINGLE("Mic1 Switch", NODE_ID_MIXER_IN, + RT286_M_REC_MIC1_SFT, 1, 1), + SOC_DAPM_SINGLE("I2S Switch", NODE_ID_MIXER_IN, + RT286_M_REC_I2S_SFT, 1, 1), +}; + +/* ADC0 source */ +static const char * const rt286_adc_src[] = { + "Mic", "Dmic" +}; + +static int rt286_adc_values[] = { + 0, 5, +}; + +static const SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc0_enum, NODE_ID_MIXER_IN1, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc0_mux = + SOC_DAPM_VALUE_ENUM("ADC 0 source", rt286_adc0_enum); + +static const SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc1_enum, NODE_ID_MIXER_IN2, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc1_mux = + SOC_DAPM_VALUE_ENUM("ADC 1 source", rt286_adc1_enum); + +/* HP-OUT source */ +static const char * const rt286_hpo_src[] = { + "Front", "Surround" +}; + +static const SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, NODE_ID_HP_OUT, + RT286_HP_SEL_SFT, rt286_hpo_src); + +static const struct snd_kcontrol_new rt286_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt286_hpo_enum); + +/* SPK-OUT source */ +static const char * const rt286_spo_src[] = { + "Front", "Surround" +}; + +static const SOC_ENUM_SINGLE_DECL(rt286_spo_enum, NODE_ID_SPK_OUT, + RT286_SPK_SEL_SFT, rt286_spo_src); + +static const struct snd_kcontrol_new rt286_spo_mux = +SOC_DAPM_ENUM("SPO source", rt286_spo_enum); + +/* SPDIF source */ +static const char * const rt286_spdif_src[] = { + "PCM-IN 0", "PCM-IN 1", "SP-OUT", "PP" +}; + +static const SOC_ENUM_SINGLE_DECL(rt286_spdif_enum, NODE_ID_SPDIF, + RT286_SPDIF_SEL_SFT, rt286_spdif_src); + +static const struct snd_kcontrol_new rt286_spdif_mux = +SOC_DAPM_ENUM("SPDIF source", rt286_spdif_enum); + + +static int rt286_spk_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: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_SPK_OUT, AC_PWRST_D0); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_SPK_OUT, 0x40); + rt286_write(codec, AC_VERB_SET_EAPD_BTLENABLE, + NODE_ID_SPK_OUT, 0x0002); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_EAPD_BTLENABLE, + NODE_ID_SPK_OUT, 0x0000); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_SPK_OUT, 0x00); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_SPK_OUT, AC_PWRST_D3); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int val; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + val = snd_soc_read(codec, NODE_ID_HP_OUT) & 0x8080; + switch (val) { + case 0x0: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0xb000); + break; + case 0x8000: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0x9000); + break; + case 0x0080: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0xa000); + break; + } + if (val != 0x8080) + mdelay(100); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0xb080); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_set_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: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D0); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D3); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_set_dmic1_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: + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_DMIC1, 0x20); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_DMIC1, AC_PWRST_D0); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_DMIC1, AC_PWRST_D3); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_DMIC1, 0x0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_hp_pow_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: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_HP_OUT, AC_PWRST_D0); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_HP_OUT, 0x40); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_HP_OUT, 0x00); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_HP_OUT, AC_PWRST_D3); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (rt286->pdata.cbj_en) { + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x04, 0xC000, 0x8000); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x4f, 0x0030, 0x0020); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x02, 0xc000, 0x8000); + } + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0f, 0x0020, 0x0000); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D0); + rt286_update_bits(codec, AC_VERB_SET_AMP_GAIN_MUTE, + w->reg, 0, 0x7080, 0x7000); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_update_bits(codec, AC_VERB_SET_AMP_GAIN_MUTE, + w->reg, 0, 0x7080, 0x7080); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D3); + break; + case SND_SOC_DAPM_POST_REG: + e = (struct soc_enum *)kcontrol->private_value; + rt286_write(codec, AC_VERB_SET_CONNECT_SEL, e->reg, + snd_soc_read(codec, e->reg) & RT286_ADC_SEL_MASK); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_out_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + + switch (event) { + case SND_SOC_DAPM_POST_REG: + rt286_write(codec, AC_VERB_SET_CONNECT_SEL, + e->reg, snd_soc_read(codec, e->reg) & 0x1); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("Beep"), + + /* DMIC */ + SND_SOC_DAPM_PGA_E("DMIC1", NODE_ID_DMIC1, 15, 0, + NULL, 0, rt286_set_dmic1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("DMIC2", NODE_ID_DMIC2, 15, 0, + NULL, 0, rt286_set_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0, + rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX_E("ADC 0 Mux", NODE_ID_ADC_IN1, 15, 0, + &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_MUX_E("ADC 1 Mux", NODE_ID_ADC_IN2, 15, 0, + &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_REG), + + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + + /* Output Side */ + + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC 0", NULL, NODE_ID_DAC_OUT1, + 15, 0, rt286_set_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("DAC 1", NULL, NODE_ID_DAC_OUT2, + 15, 0, rt286_set_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Output Mux */ + SND_SOC_DAPM_MUX_E("SPK Mux", SND_SOC_NOPM, 0, 0, + &rt286_spo_mux, + rt286_out_mux_event, + SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_MUX_E("HPO Mux", SND_SOC_NOPM, 0, 0, + &rt286_hpo_mux, + rt286_out_mux_event, + SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_SUPPLY("HP Power", SND_SOC_NOPM, + 0, 0, rt286_hp_pow_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_MUX("SPDIF Mux", SND_SOC_NOPM, 0, 0, + &rt286_spdif_mux), + + /* Output Mixer */ + SND_SOC_DAPM_MIXER("Front", SND_SOC_NOPM, 0, 0, + rt286_front_mix, ARRAY_SIZE(rt286_front_mix)), + SND_SOC_DAPM_PGA("Surround", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_PGA_E("SPO", SND_SOC_NOPM, 0, 0, + NULL, 0, rt286_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("HPO", SND_SOC_NOPM, 0, 0, + NULL, 0, rt286_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), +}; + +static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + {"DMIC1", NULL, "DMIC Receiver"}, + {"DMIC2", NULL, "DMIC Receiver"}, + + {"RECMIX", "Beep Switch", "Beep"}, + {"RECMIX", "Line1 Switch", "LINE1"}, + {"RECMIX", "Mic1 Switch", "MIC1"}, + + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "Mic", "MIC1"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "Mic", "MIC1"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TX", NULL, "ADC 0"}, + {"AIF2TX", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RX"}, + {"DAC 1", NULL, "AIF2RX"}, + + {"Front", "DAC Switch", "DAC 0"}, + {"Front", "RECMIX Switch", "RECMIX"}, + + {"Surround", NULL, "DAC 1"}, + + {"SPK Mux", "Front", "Front"}, + {"SPK Mux", "Surround", "Surround"}, + + {"HPO Mux", "Front", "Front"}, + {"HPO Mux", "Surround", "Surround"}, + + {"SPO", NULL, "SPK Mux"}, + {"HPO", NULL, "HPO Mux"}, + {"HPO", NULL, "HP Power"}, + + {"SPOL", NULL, "SPO"}, + {"SPOR", NULL, "SPO"}, + {"HPO Pin", NULL, "HPO"}, +}; + +static int rt286_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 rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + val |= 0x4000; + break; + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt286->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + d_len_code = 0; + switch (params_format(params)) { + /* bit 6:4 Bits per Sample */ + case SNDRV_PCM_FORMAT_S16_LE: + d_len_code = 0; + val |= (0x1 << 4); + break; + case SNDRV_PCM_FORMAT_S32_LE: + d_len_code = 2; + val |= (0x4 << 4); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + d_len_code = 1; + val |= (0x2 << 4); + break; + case SNDRV_PCM_FORMAT_S24_LE: + d_len_code = 2; + val |= (0x3 << 4); + break; + case SNDRV_PCM_FORMAT_S8: + break; + default: + return -EINVAL; + } + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0018, d_len_code << 3); + pr_debug("format val = 0x%lx\n", val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_DAC_OUT1, 0, 0x407f, val); + else + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_ADC_IN1, 0, 0x407f, val); + + return 0; +} + +static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0800, 0x800); + break; + case SND_SOC_DAIFMT_CBS_CFS: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0800, 0x0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x0); + break; + case SND_SOC_DAIFMT_LEFT_J: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x1 << 8); + break; + case SND_SOC_DAIFMT_DSP_A: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x2 << 8); + break; + case SND_SOC_DAIFMT_DSP_B: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x3 << 8); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_DAC_OUT1, 0, 0x8000, 0x0); + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_ADC_IN1, 0, 0x8000, 0x0); + + return 0; +} + +static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + pr_debug("%s freq=%d\n", __func__, freq); + + if (RT286_SCLK_S_MCLK == clk_id) { + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x0100, 0x0); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x49, 0x0020, 0x0020); + } else { + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x0100, 0x0100); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x63, 0x4, 0x4); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x49, 0x20, 0x0); + } + + switch (freq) { + case 19200000: + case 24000000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + break; + case 12288000: + case 11289600: + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x8, 0x0); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0b, 0xfc1e, 0x0004); + break; + case 24576000: + case 22579200: + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x8, 0x8); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0b, 0xfc1e, 0x5406); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt286->sys_clk = freq; + + return 0; +} + +static int rt286_set_dai_dfs(struct snd_soc_dai *dai, unsigned int fs) +{ + struct snd_soc_codec *codec = dai->codec; + + pr_debug("%s fs=%d\n", __func__, fs); + if (50 == fs) + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x1000, 0x1000); + else + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x1000, 0x0); + + + return 0; +} + +static int rt286_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) + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_AUDIO_FUNCTION_GROUP, AC_PWRST_D0); + break; + + case SND_SOC_BIAS_OFF: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_AUDIO_FUNCTION_GROUP, AC_PWRST_D3); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt286_probe(struct snd_soc_codec *codec) +{ + hw_configure(codec); + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + + return 0; +} + +static int rt286_remove(struct snd_soc_codec *codec) +{ + rt286_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT286_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 rt286_aif_dai_ops = { + .hw_params = rt286_hw_params, + .set_fmt = rt286_set_dai_fmt, + .set_sysclk = rt286_set_dai_sysclk, + .set_dfs = rt286_set_dai_dfs, +}; + +struct snd_soc_dai_driver rt286_dai[] = { + { + .name = "rt286-aif1", + .id = RT286_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt286 = { + .probe = rt286_probe, + .remove = rt286_remove, + .set_bias_level = rt286_set_bias_level, + .idle_bias_off = true, + .controls = rt286_snd_controls, + .num_controls = ARRAY_SIZE(rt286_snd_controls), + .dapm_widgets = rt286_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets), + .dapm_routes = rt286_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes), +}; + +static const struct regmap_config rt286_regmap = { + .reg_bits = 32, + .val_bits = 32, + .reg_write = rt286_hw_write, + .reg_read = rt286_hw_read, + .readable_reg = rt286_readable_register, + .cache_type = REGCACHE_NONE, +}; + +static const struct i2c_device_id rt286_i2c_id[] = { + {"rt5640", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); + +static struct acpi_device_id rt286_acpi_match[] = { + { "INT33CA", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); + +static int rt286_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt286_priv *rt286; + struct device *dev = &i2c->dev; + int ret; + + rt286 = devm_kzalloc(&i2c->dev, + sizeof(struct rt286_priv), + GFP_KERNEL); + if (NULL == rt286) + return -ENOMEM; + + rt286->regmap = devm_regmap_init(dev, NULL, i2c, &rt286_regmap); + if (IS_ERR(rt286->regmap)) { + ret = PTR_ERR(rt286->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + i2c_set_clientdata(i2c, rt286); + + if (pdata) + rt286->pdata = *pdata; + + ret = devm_snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286, + rt286_dai, ARRAY_SIZE(rt286_dai)); + if (ret < 0) + goto err; + + return 0; +err: + return ret; +} + +static int rt286_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +struct i2c_driver rt286_i2c_driver = { + .driver = { + .name = "rt5640", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt286_acpi_match), + }, + .probe = rt286_i2c_probe, + .remove = rt286_i2c_remove, + .id_table = rt286_i2c_id, +}; + +static int __init rt286_modinit(void) +{ + return i2c_add_driver(&rt286_i2c_driver); +} + +module_init(rt286_modinit); + +static void __exit rt286_modexit(void) +{ + i2c_del_driver(&rt286_i2c_driver); +} + +module_exit(rt286_modexit); + +MODULE_DESCRIPTION("ASoC RT286 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h new file mode 100644 index 0000000..c579c53 --- /dev/null +++ b/sound/soc/codecs/rt286.h @@ -0,0 +1,89 @@ +/* + * rt286.h -- RT286 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 __RT286_H__ +#define __RT286_H__ + +#define NODE_ID_AUDIO_FUNCTION_GROUP 0x01 +#define NODE_ID_DAC_OUT1 0x02 +#define NODE_ID_DAC_OUT2 0x03 +#define NODE_ID_ADC_IN1 0x09 +#define NODE_ID_ADC_IN2 0x08 +#define NODE_ID_MIXER_IN 0x0b +#define NODE_ID_MIXER_OUT1 0x0c +#define NODE_ID_MIXER_OUT2 0x0d +#define NODE_ID_DMIC1 0x12 +#define NODE_ID_DMIC2 0x13 +#define NODE_ID_SPK_OUT 0x14 +#define NODE_ID_MIC1 0x18 +#define NODE_ID_LINE1 0x1a +#define NODE_ID_BEEP 0x1d +#define NODE_ID_SPDIF 0x1e +#define NODE_ID_VENDOR_REGISTERS 0x20 +#define NODE_ID_HP_OUT 0x21 +#define NODE_ID_MIXER_IN1 0x22 +#define NODE_ID_MIXER_IN2 0x23 +#define TOTAL_NODE_ID 0x23 + +#define CONNECTION_INDEX_MIC1 0X0 +#define CONNECTION_INDEX_DMIC 0X5 + +/* SPDIF (0x06) */ +#define RT286_SPDIF_SEL_SFT 0 +#define RT286_SPDIF_SEL_PCM0 0 +#define RT286_SPDIF_SEL_PCM1 1 +#define RT286_SPDIF_SEL_SPOUT 2 +#define RT286_SPDIF_SEL_PP 3 + +/* RECMIX (0x0b) */ +#define RT286_M_REC_BEEP_SFT 0 +#define RT286_M_REC_LINE1_SFT 1 +#define RT286_M_REC_MIC1_SFT 2 +#define RT286_M_REC_I2S_SFT 3 + +/* Front (0x0c) */ +#define RT286_M_FRONT_DAC_SFT 0 +#define RT286_M_FRONT_REC_SFT 1 + +/* SPK-OUT (0x14) */ +#define RT286_SPK_SEL_MASK 0x1 +#define RT286_SPK_SEL_SFT 0 +#define RT286_SPK_SEL_F 0 +#define RT286_SPK_SEL_S 1 + +/* HP-OUT (0x21) */ +#define RT286_HP_SEL_MASK 0x1 +#define RT286_HP_SEL_SFT 0 +#define RT286_HP_SEL_F 0 +#define RT286_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT286_ADC_SEL_MASK 0x7 +#define RT286_ADC_SEL_SFT 0 +#define RT286_ADC_SEL_SURR 0 +#define RT286_ADC_SEL_FRONT 1 +#define RT286_ADC_SEL_DMIC 2 +#define RT286_ADC_SEL_BEEP 4 +#define RT286_ADC_SEL_LINE1 5 +#define RT286_ADC_SEL_I2S 6 +#define RT286_ADC_SEL_MIC1 7 + +#define RT286_SCLK_S_MCLK 0 +#define RT286_SCLK_S_PLL 1 + +enum { + RT286_AIF1, + RT286_AIFS, +}; + +int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic); + +#endif /* __RT286_H__ */ -- 1.8.1.1.439.g50a6b54

3 5

[alsa-devel] [PATCH] SoC: wm8991: Remove unused pointer in wm8991_probe()
by Christian Engelmayer 07 Feb '14

07 Feb '14

Remove unused pointer 'wm8991' in function wm8991_probe(). The last user vanished with a86652e5 (ASoC: wm8991: Convert to direct regmap API usage) Detected by Coverity: CID 1162831 Signed-off-by: Christian Engelmayer <cengelma(a)gmx.at> --- sound/soc/codecs/wm8991.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index dba0306..244eb09 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1251,11 +1251,8 @@ static int wm8991_remove(struct snd_soc_codec *codec) static int wm8991_probe(struct snd_soc_codec *codec) { - struct wm8991_priv *wm8991; int ret; - wm8991 = snd_soc_codec_get_drvdata(codec); - 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); -- 1.8.3.2

2 3

[alsa-devel] [PATCH] ASoC: tobermory: Stop the FLL if we fail to switch SYSCLK to it
by Mark Brown 07 Feb '14

07 Feb '14

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/tobermory.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index f21ff608a819..1807b75ccc12 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -44,6 +44,8 @@ static int tobermory_set_bias_level(struct snd_soc_card *card, SND_SOC_CLOCK_IN); if (ret < 0) { pr_err("Failed to set SYSCLK: %d\n", ret); + snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + 0, 0, 0); return ret; } } -- 1.9.rc1

1 0

[alsa-devel] [PATCH 1/2] ASoC: pcm512x: Implement paging support
by Mark Brown 07 Feb '14

07 Feb '14

From: Mark Brown <broonie(a)linaro.org> The PCM512x devices use a paged register map covering the entire register range. Implement support for this, mapping pages in at addresses starting at 0x100 for ease of use (though since the pages are numbered from 0 there is going to be an off by one when looking at the first byte as a page number). Also mark the new registers as accessible with the exception of the coefficient RAM which is a bit fiddly and may benefit from some extra handling to linearise the blocks. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm512x.c | 65 ++++++++++++++++------- sound/soc/codecs/pcm512x.h | 126 +++++++++++++++++++++++++-------------------- 2 files changed, 116 insertions(+), 75 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 1150381fc373..cdcb51e4c86f 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -66,22 +66,29 @@ PCM512x_REGULATOR_EVENT(1) PCM512x_REGULATOR_EVENT(2) static const struct reg_default pcm512x_reg_defaults[] = { - { PCM512x_RESET, 0x00 }, - { PCM512x_POWER, 0x00 }, - { PCM512x_MUTE, 0x00 }, - { PCM512x_DSP, 0x00 }, - { PCM512x_PLL_REF, 0x00 }, - { PCM512x_DAC_ROUTING, 0x11 }, - { PCM512x_DSP_PROGRAM, 0x01 }, - { PCM512x_CLKDET, 0x00 }, - { PCM512x_AUTO_MUTE, 0x00 }, - { PCM512x_ERROR_DETECT, 0x00 }, - { PCM512x_DIGITAL_VOLUME_1, 0x00 }, - { PCM512x_DIGITAL_VOLUME_2, 0x30 }, - { PCM512x_DIGITAL_VOLUME_3, 0x30 }, - { PCM512x_DIGITAL_MUTE_1, 0x22 }, - { PCM512x_DIGITAL_MUTE_2, 0x00 }, - { PCM512x_DIGITAL_MUTE_3, 0x07 }, + { PCM512x_RESET, 0x00 }, + { PCM512x_POWER, 0x00 }, + { PCM512x_MUTE, 0x00 }, + { PCM512x_DSP, 0x00 }, + { PCM512x_PLL_REF, 0x00 }, + { PCM512x_DAC_ROUTING, 0x11 }, + { PCM512x_DSP_PROGRAM, 0x01 }, + { PCM512x_CLKDET, 0x00 }, + { PCM512x_AUTO_MUTE, 0x00 }, + { PCM512x_ERROR_DETECT, 0x00 }, + { PCM512x_DIGITAL_VOLUME_1, 0x00 }, + { PCM512x_DIGITAL_VOLUME_2, 0x30 }, + { PCM512x_DIGITAL_VOLUME_3, 0x30 }, + { PCM512x_DIGITAL_MUTE_1, 0x22 }, + { PCM512x_DIGITAL_MUTE_2, 0x00 }, + { PCM512x_DIGITAL_MUTE_3, 0x07 }, + { PCM512x_OUTPUT_AMPLITUDE, 0x00 }, + { PCM512x_ANALOG_GAIN_CTRL, 0x00 }, + { PCM512x_UNDERVOLTAGE_PROT, 0x00 }, + { PCM512x_ANALOG_MUTE_CTRL, 0x00 }, + { PCM512x_ANALOG_GAIN_BOOST, 0x00 }, + { PCM512x_VCOM_CTRL_1, 0x00 }, + { PCM512x_VCOM_CTRL_2, 0x01 }, }; static bool pcm512x_readable(struct device *dev, unsigned int reg) @@ -141,9 +148,18 @@ static bool pcm512x_readable(struct device *dev, unsigned int reg) case PCM512x_ANALOG_MUTE_DET: case PCM512x_GPIN: case PCM512x_DIGITAL_MUTE_DET: + case PCM512x_OUTPUT_AMPLITUDE: + case PCM512x_ANALOG_GAIN_CTRL: + case PCM512x_UNDERVOLTAGE_PROT: + case PCM512x_ANALOG_MUTE_CTRL: + case PCM512x_ANALOG_GAIN_BOOST: + case PCM512x_VCOM_CTRL_1: + case PCM512x_VCOM_CTRL_2: + case PCM512x_CRAM_CTRL: return true; default: - return false; + /* There are 256 raw register addresses */ + return reg < 0xff; } } @@ -159,9 +175,11 @@ static bool pcm512x_volatile(struct device *dev, unsigned int reg) case PCM512x_ANALOG_MUTE_DET: case PCM512x_GPIN: case PCM512x_DIGITAL_MUTE_DET: + case PCM512x_CRAM_CTRL: return true; default: - return false; + /* There are 256 raw register addresses */ + return reg < 0xff; } } @@ -343,6 +361,14 @@ static struct snd_soc_codec_driver pcm512x_codec_driver = { .num_dapm_routes = ARRAY_SIZE(pcm512x_dapm_routes), }; +static const struct regmap_range_cfg pcm512x_range = { + .name = "Pages", .range_min = PCM512x_VIRT_BASE, + .range_max = PCM512x_MAX_REGISTER, + .selector_reg = PCM512x_PAGE, + .selector_mask = 0xff, + .window_start = 0, .window_len = 0x100, +}; + static const struct regmap_config pcm512x_regmap = { .reg_bits = 8, .val_bits = 8, @@ -350,6 +376,9 @@ static const struct regmap_config pcm512x_regmap = { .readable_reg = pcm512x_readable, .volatile_reg = pcm512x_volatile, + .ranges = &pcm512x_range, + .num_ranges = 1, + .max_register = PCM512x_MAX_REGISTER, .reg_defaults = pcm512x_reg_defaults, .num_reg_defaults = ARRAY_SIZE(pcm512x_reg_defaults), diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h index b2f518ecb35c..e58743c965d6 100644 --- a/sound/soc/codecs/pcm512x.h +++ b/sound/soc/codecs/pcm512x.h @@ -17,66 +17,78 @@ #ifndef _SND_SOC_PCM512X #define _SND_SOC_PCM512X -#define PCM512x_PAGE_0_BASE 0 +#define PCM512x_VIRT_BASE 0x100 +#define PCM512x_PAGE_LEN 0x100 +#define PCM512x_PAGE_BASE(n) (PCM512x_VIRT_BASE + (PCM512x_PAGE_LEN * n)) #define PCM512x_PAGE 0 -#define PCM512x_RESET (PCM512x_PAGE_0_BASE + 1) -#define PCM512x_POWER (PCM512x_PAGE_0_BASE + 2) -#define PCM512x_MUTE (PCM512x_PAGE_0_BASE + 3) -#define PCM512x_PLL_EN (PCM512x_PAGE_0_BASE + 4) -#define PCM512x_SPI_MISO_FUNCTION (PCM512x_PAGE_0_BASE + 6) -#define PCM512x_DSP (PCM512x_PAGE_0_BASE + 7) -#define PCM512x_GPIO_EN (PCM512x_PAGE_0_BASE + 8) -#define PCM512x_BCLK_LRCLK_CFG (PCM512x_PAGE_0_BASE + 9) -#define PCM512x_DSP_GPIO_INPUT (PCM512x_PAGE_0_BASE + 10) -#define PCM512x_MASTER_MODE (PCM512x_PAGE_0_BASE + 12) -#define PCM512x_PLL_REF (PCM512x_PAGE_0_BASE + 13) -#define PCM512x_PLL_COEFF_0 (PCM512x_PAGE_0_BASE + 20) -#define PCM512x_PLL_COEFF_1 (PCM512x_PAGE_0_BASE + 21) -#define PCM512x_PLL_COEFF_2 (PCM512x_PAGE_0_BASE + 22) -#define PCM512x_PLL_COEFF_3 (PCM512x_PAGE_0_BASE + 23) -#define PCM512x_PLL_COEFF_4 (PCM512x_PAGE_0_BASE + 24) -#define PCM512x_DSP_CLKDIV (PCM512x_PAGE_0_BASE + 27) -#define PCM512x_DAC_CLKDIV (PCM512x_PAGE_0_BASE + 28) -#define PCM512x_NCP_CLKDIV (PCM512x_PAGE_0_BASE + 29) -#define PCM512x_OSR_CLKDIV (PCM512x_PAGE_0_BASE + 30) -#define PCM512x_MASTER_CLKDIV_1 (PCM512x_PAGE_0_BASE + 32) -#define PCM512x_MASTER_CLKDIV_2 (PCM512x_PAGE_0_BASE + 33) -#define PCM512x_FS_SPEED_MODE (PCM512x_PAGE_0_BASE + 34) -#define PCM512x_IDAC_1 (PCM512x_PAGE_0_BASE + 35) -#define PCM512x_IDAC_2 (PCM512x_PAGE_0_BASE + 36) -#define PCM512x_ERROR_DETECT (PCM512x_PAGE_0_BASE + 37) -#define PCM512x_I2S_1 (PCM512x_PAGE_0_BASE + 40) -#define PCM512x_I2S_2 (PCM512x_PAGE_0_BASE + 41) -#define PCM512x_DAC_ROUTING (PCM512x_PAGE_0_BASE + 42) -#define PCM512x_DSP_PROGRAM (PCM512x_PAGE_0_BASE + 43) -#define PCM512x_CLKDET (PCM512x_PAGE_0_BASE + 44) -#define PCM512x_AUTO_MUTE (PCM512x_PAGE_0_BASE + 59) -#define PCM512x_DIGITAL_VOLUME_1 (PCM512x_PAGE_0_BASE + 60) -#define PCM512x_DIGITAL_VOLUME_2 (PCM512x_PAGE_0_BASE + 61) -#define PCM512x_DIGITAL_VOLUME_3 (PCM512x_PAGE_0_BASE + 62) -#define PCM512x_DIGITAL_MUTE_1 (PCM512x_PAGE_0_BASE + 63) -#define PCM512x_DIGITAL_MUTE_2 (PCM512x_PAGE_0_BASE + 64) -#define PCM512x_DIGITAL_MUTE_3 (PCM512x_PAGE_0_BASE + 65) -#define PCM512x_GPIO_OUTPUT_1 (PCM512x_PAGE_0_BASE + 80) -#define PCM512x_GPIO_OUTPUT_2 (PCM512x_PAGE_0_BASE + 81) -#define PCM512x_GPIO_OUTPUT_3 (PCM512x_PAGE_0_BASE + 82) -#define PCM512x_GPIO_OUTPUT_4 (PCM512x_PAGE_0_BASE + 83) -#define PCM512x_GPIO_OUTPUT_5 (PCM512x_PAGE_0_BASE + 84) -#define PCM512x_GPIO_OUTPUT_6 (PCM512x_PAGE_0_BASE + 85) -#define PCM512x_GPIO_CONTROL_1 (PCM512x_PAGE_0_BASE + 86) -#define PCM512x_GPIO_CONTROL_2 (PCM512x_PAGE_0_BASE + 87) -#define PCM512x_OVERFLOW (PCM512x_PAGE_0_BASE + 90) -#define PCM512x_RATE_DET_1 (PCM512x_PAGE_0_BASE + 91) -#define PCM512x_RATE_DET_2 (PCM512x_PAGE_0_BASE + 92) -#define PCM512x_RATE_DET_3 (PCM512x_PAGE_0_BASE + 93) -#define PCM512x_RATE_DET_4 (PCM512x_PAGE_0_BASE + 94) -#define PCM512x_ANALOG_MUTE_DET (PCM512x_PAGE_0_BASE + 108) -#define PCM512x_GPIN (PCM512x_PAGE_0_BASE + 119) -#define PCM512x_DIGITAL_MUTE_DET (PCM512x_PAGE_0_BASE + 120) - -#define PCM512x_MAX_REGISTER (PCM512x_PAGE_0_BASE + 120) +#define PCM512x_RESET (PCM512x_PAGE_BASE(0) + 1) +#define PCM512x_POWER (PCM512x_PAGE_BASE(0) + 2) +#define PCM512x_MUTE (PCM512x_PAGE_BASE(0) + 3) +#define PCM512x_PLL_EN (PCM512x_PAGE_BASE(0) + 4) +#define PCM512x_SPI_MISO_FUNCTION (PCM512x_PAGE_BASE(0) + 6) +#define PCM512x_DSP (PCM512x_PAGE_BASE(0) + 7) +#define PCM512x_GPIO_EN (PCM512x_PAGE_BASE(0) + 8) +#define PCM512x_BCLK_LRCLK_CFG (PCM512x_PAGE_BASE(0) + 9) +#define PCM512x_DSP_GPIO_INPUT (PCM512x_PAGE_BASE(0) + 10) +#define PCM512x_MASTER_MODE (PCM512x_PAGE_BASE(0) + 12) +#define PCM512x_PLL_REF (PCM512x_PAGE_BASE(0) + 13) +#define PCM512x_PLL_COEFF_0 (PCM512x_PAGE_BASE(0) + 20) +#define PCM512x_PLL_COEFF_1 (PCM512x_PAGE_BASE(0) + 21) +#define PCM512x_PLL_COEFF_2 (PCM512x_PAGE_BASE(0) + 22) +#define PCM512x_PLL_COEFF_3 (PCM512x_PAGE_BASE(0) + 23) +#define PCM512x_PLL_COEFF_4 (PCM512x_PAGE_BASE(0) + 24) +#define PCM512x_DSP_CLKDIV (PCM512x_PAGE_BASE(0) + 27) +#define PCM512x_DAC_CLKDIV (PCM512x_PAGE_BASE(0) + 28) +#define PCM512x_NCP_CLKDIV (PCM512x_PAGE_BASE(0) + 29) +#define PCM512x_OSR_CLKDIV (PCM512x_PAGE_BASE(0) + 30) +#define PCM512x_MASTER_CLKDIV_1 (PCM512x_PAGE_BASE(0) + 32) +#define PCM512x_MASTER_CLKDIV_2 (PCM512x_PAGE_BASE(0) + 33) +#define PCM512x_FS_SPEED_MODE (PCM512x_PAGE_BASE(0) + 34) +#define PCM512x_IDAC_1 (PCM512x_PAGE_BASE(0) + 35) +#define PCM512x_IDAC_2 (PCM512x_PAGE_BASE(0) + 36) +#define PCM512x_ERROR_DETECT (PCM512x_PAGE_BASE(0) + 37) +#define PCM512x_I2S_1 (PCM512x_PAGE_BASE(0) + 40) +#define PCM512x_I2S_2 (PCM512x_PAGE_BASE(0) + 41) +#define PCM512x_DAC_ROUTING (PCM512x_PAGE_BASE(0) + 42) +#define PCM512x_DSP_PROGRAM (PCM512x_PAGE_BASE(0) + 43) +#define PCM512x_CLKDET (PCM512x_PAGE_BASE(0) + 44) +#define PCM512x_AUTO_MUTE (PCM512x_PAGE_BASE(0) + 59) +#define PCM512x_DIGITAL_VOLUME_1 (PCM512x_PAGE_BASE(0) + 60) +#define PCM512x_DIGITAL_VOLUME_2 (PCM512x_PAGE_BASE(0) + 61) +#define PCM512x_DIGITAL_VOLUME_3 (PCM512x_PAGE_BASE(0) + 62) +#define PCM512x_DIGITAL_MUTE_1 (PCM512x_PAGE_BASE(0) + 63) +#define PCM512x_DIGITAL_MUTE_2 (PCM512x_PAGE_BASE(0) + 64) +#define PCM512x_DIGITAL_MUTE_3 (PCM512x_PAGE_BASE(0) + 65) +#define PCM512x_GPIO_OUTPUT_1 (PCM512x_PAGE_BASE(0) + 80) +#define PCM512x_GPIO_OUTPUT_2 (PCM512x_PAGE_BASE(0) + 81) +#define PCM512x_GPIO_OUTPUT_3 (PCM512x_PAGE_BASE(0) + 82) +#define PCM512x_GPIO_OUTPUT_4 (PCM512x_PAGE_BASE(0) + 83) +#define PCM512x_GPIO_OUTPUT_5 (PCM512x_PAGE_BASE(0) + 84) +#define PCM512x_GPIO_OUTPUT_6 (PCM512x_PAGE_BASE(0) + 85) +#define PCM512x_GPIO_CONTROL_1 (PCM512x_PAGE_BASE(0) + 86) +#define PCM512x_GPIO_CONTROL_2 (PCM512x_PAGE_BASE(0) + 87) +#define PCM512x_OVERFLOW (PCM512x_PAGE_BASE(0) + 90) +#define PCM512x_RATE_DET_1 (PCM512x_PAGE_BASE(0) + 91) +#define PCM512x_RATE_DET_2 (PCM512x_PAGE_BASE(0) + 92) +#define PCM512x_RATE_DET_3 (PCM512x_PAGE_BASE(0) + 93) +#define PCM512x_RATE_DET_4 (PCM512x_PAGE_BASE(0) + 94) +#define PCM512x_ANALOG_MUTE_DET (PCM512x_PAGE_BASE(0) + 108) +#define PCM512x_GPIN (PCM512x_PAGE_BASE(0) + 119) +#define PCM512x_DIGITAL_MUTE_DET (PCM512x_PAGE_BASE(0) + 120) + +#define PCM512x_OUTPUT_AMPLITUDE (PCM512x_PAGE_BASE(1) + 1) +#define PCM512x_ANALOG_GAIN_CTRL (PCM512x_PAGE_BASE(1) + 2) +#define PCM512x_UNDERVOLTAGE_PROT (PCM512x_PAGE_BASE(1) + 5) +#define PCM512x_ANALOG_MUTE_CTRL (PCM512x_PAGE_BASE(1) + 6) +#define PCM512x_ANALOG_GAIN_BOOST (PCM512x_PAGE_BASE(1) + 7) +#define PCM512x_VCOM_CTRL_1 (PCM512x_PAGE_BASE(1) + 8) +#define PCM512x_VCOM_CTRL_2 (PCM512x_PAGE_BASE(1) + 9) + +#define PCM512x_CRAM_CTRL (PCM512x_PAGE_BASE(44) + 1) + +#define PCM512x_MAX_REGISTER (PCM512x_PAGE_BASE(44) + 1) /* Page 0, Register 1 - reset */ #define PCM512x_RSTR (1 << 0) -- 1.9.rc1

1 1

[alsa-devel] Sony Vaio Pro svp112a1cm
by Hendrik-Jan Heins 07 Feb '14

07 Feb '14

Dear list, to report the Spny Vaio Pro svp112a1cm THe sound device in this notebook works, but the internal microphone not. This seeems to be the same device as in teh 13 inch version of this laptop, but they seem to have changed the ID of the device. Maybe the same fix can be applied to fix? All details of the machine are available here: http://www.alsa-project.org/db/?f=970d897015298925b86d4d2c68a106fba22eb184 thank you. best regards, Hendrik-Jan

2 4

[alsa-devel] ASoC driver for multi-codec sound device
by Christopher Seifert 07 Feb '14

07 Feb '14

Hi all, we are currently working on an ASoC driver for a multi-codec sound device. The kernel we use is version 3.10. Each codec has it's own I2C address and are connected via I2S to a single CPU DAI. We'd like to use the codecs in TDM mode later on, so we can use them as a single multi channel ALSA device. Currently we can use one codec in stereo mode with playback and capturing working using a device tree. We were able to get a second codec working (using the snd_soc_aux_dev structure in our driver) in so far we can see and use the codec's controls via alsamixer, however the codec seems to ignore the commands or does not receive any at all. What we'd like to know: 1) How to we achieve proper initialisation of all codecs? More specifically, which function (ALSA side) or callback (codec side) is used to set up the codecs clocks (via I2C)? 2) As far as we know the driver we are building upon does not implement own PCM or DMA drivers. Is it necessary to do this on our own to get the codecs working in TDM mode? Also when working in I2S (stereo) mode on ALSA side, is it possible to tell ALSA to use more than 2 channels of the sound buffer? We currently have interleaved audio samples from more than one codec in our sound buffer (transferred by DMA) and we'd like to select a subset of these depending on how the sound device was opened (for example "arecord -c 2" vs. "arecord -c 4" and so on). Which are the audio sample order ALSA supports or expects, specifically in the case of more than 2 audio channels? (Interleaved "L0, L1, L2, ..., Ln, R0, R1, ..., Rn" vs. "L0, R0, L1, R1, ..., Ln, Rn") 3) If there is no custom DMA driver implemented how does the ALSA framework use the DMA controller? In which case do we need to write our own DMA driver? Are there possibilities to use the ALSA framework's internal DMA handling for our purposes? 4) How are PCM and DMA correlated in ALSA framework? Specifically, do the implementations depend on each other or are they independent, so we can for example implement a DMA driver and use ALSA's default PCM? We're stuck with this, so we appreciate any kind of help pointing us in the right direction. If there are any required informations missing please don't hesitate to ask. Thanks, Achim and Christopher

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[alsa-devel] [PATCH] ASoC: rsnd: use device dependency clock
by Kuninori Morimoto 07 Feb '14

07 Feb '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current R-Car sound driver is using device independent audio clock, but it is not good design for DT (= common clock) support. This patch adds device dependent clock support. But, there are some platform which is using old style clock. Old style clock is still supported at this point, but it will be removed soon. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 35 ++++++++++++++++++++++++++++++----- 1 file changed, 30 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index a53235c..e82ab75 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -259,8 +259,9 @@ int rsnd_adg_probe(struct platform_device *pdev, { struct rsnd_adg *adg; struct device *dev = rsnd_priv_to_dev(priv); - struct clk *clk; + struct clk *clk, *clk_orig; int i; + bool use_old_style = false; adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL); if (!adg) { @@ -268,10 +269,34 @@ int rsnd_adg_probe(struct platform_device *pdev, return -ENOMEM; } - adg->clk[CLKA] = clk_get(NULL, "audio_clk_a"); - adg->clk[CLKB] = clk_get(NULL, "audio_clk_b"); - adg->clk[CLKC] = clk_get(NULL, "audio_clk_c"); - adg->clk[CLKI] = clk_get(NULL, "audio_clk_internal"); + clk_orig = clk_get(dev, NULL); + adg->clk[CLKA] = clk_get(dev, "clk_a"); + adg->clk[CLKB] = clk_get(dev, "clk_b"); + adg->clk[CLKC] = clk_get(dev, "clk_c"); + adg->clk[CLKI] = clk_get(dev, "clk_i"); + + for_each_rsnd_clk(clk, adg, i) { + if (clk_orig == clk) { + dev_warn(dev, + "Audio clock doesn't support new style," + "use old style\n"); + use_old_style = true; + break; + } + } + + /* + * note: + * these exist in order to keep compatible with old style, + * but will be removed soon + */ + if (use_old_style) { + adg->clk[CLKA] = clk_get(NULL, "audio_clk_a"); + adg->clk[CLKB] = clk_get(NULL, "audio_clk_b"); + adg->clk[CLKC] = clk_get(NULL, "audio_clk_c"); + adg->clk[CLKI] = clk_get(NULL, "audio_clk_internal"); + } + for_each_rsnd_clk(clk, adg, i) { if (IS_ERR(clk)) { dev_err(dev, "Audio clock failed\n"); -- 1.7.9.5

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[alsa-devel] [PATCH 1/3] sound: Include appropriate header file in pcsp/pcsp_input.c
by Rashika Kheria 07 Feb '14

07 Feb '14

Include appropriate header file in drivers/pcsp/pcsp_input.c because it defines function whose prototype definitions are present in drivers/pcsp/pcsp_input.h. This eliminates the following warning in drivers/pcsp/pcsp_input.c: sound/drivers/pcsp/pcsp_input.c:42:6: warning: no previous prototype for ‘pcspkr_stop_sound’ [-Wmissing-prototypes] sound/drivers/pcsp/pcsp_input.c:80:5: warning: no previous prototype for ‘pcspkr_input_init’ [-Wmissing-prototypes] sound/drivers/pcsp/pcsp_input.c:110:5: warning: no previous prototype for ‘pcspkr_input_remove’ [-Wmissing-prototypes] Signed-off-by: Rashika Kheria <rashika.kheria(a)gmail.com> --- sound/drivers/pcsp/pcsp_input.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/drivers/pcsp/pcsp_input.c b/sound/drivers/pcsp/pcsp_input.c index b874b0a..0ecf8a4 100644 --- a/sound/drivers/pcsp/pcsp_input.c +++ b/sound/drivers/pcsp/pcsp_input.c @@ -16,6 +16,7 @@ #include <linux/input.h> #include <asm/io.h> #include "pcsp.h" +#include "pcsp_input.h" static void pcspkr_do_sound(unsigned int count) { -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ASoC: codecs: Put the CODEC drivers in a menu
by Mark Brown 07 Feb '14

07 Feb '14

From: Mark Brown <broonie(a)linaro.org> Now they're visible they get a bit noisy. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/Kconfig | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 28faa2d0eaef..6b67e86f3f2c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -8,6 +8,8 @@ config SND_SOC_I2C_AND_SPI default y if I2C=y default y if SPI_MASTER=y +menu "CODEC drivers" + config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" depends on COMPILE_TEST @@ -588,3 +590,5 @@ config SND_SOC_ML26124 config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C + +endmenu -- 1.9.rc1

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[alsa-devel] [PATCH] ASoC: rt5640: Add ACPI ID for Intel Baytrail
by Jarkko Nikula 07 Feb '14

07 Feb '14

Realtek RT5640 uses ACPI ID "10EC5640" for Intel Baytrail platforms. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Cc: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index a3fb41179636..886924934aa5 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2093,6 +2093,7 @@ MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); #ifdef CONFIG_ACPI static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, + { "10EC5640", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); -- 1.8.5.3

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Re: [alsa-devel] [PATCH RFC 26/46] drivers/base: provide an infrastructure for componentised subsystems
by Daniel Vetter 07 Feb '14

07 Feb '14

I've chatted a bit with Hans Verkuil about this topic at fosdem and apparently both v4l and alsa have something like this already in their helper libraries. Adding more people as fyi in case they want to switch to the new driver core stuff from Russell. -Daniel On Thu, Jan 2, 2014 at 10:27 PM, Russell King <rmk+kernel(a)arm.linux.org.uk> wrote: > Subsystems such as ALSA, DRM and others require a single card-level > device structure to represent a subsystem. However, firmware tends to > describe the individual devices and the connections between them. > > Therefore, we need a way to gather up the individual component devices > together, and indicate when we have all the component devices. > > We do this in DT by providing a "superdevice" node which specifies > the components, eg: > > imx-drm { > compatible = "fsl,drm"; > crtcs = <&ipu1>; > connectors = <&hdmi>; > }; > > The superdevice is declared into the component support, along with the > subcomponents. The superdevice receives callbacks to locate the > subcomponents, and identify when all components are present. At this > point, we bind the superdevice, which causes the appropriate subsystem > to be initialised in the conventional way. > > When any of the components or superdevice are removed from the system, > we unbind the superdevice, thereby taking the subsystem down. > > Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> > --- > drivers/base/Makefile | 2 +- > drivers/base/component.c | 379 +++++++++++++++++++++++++++++++++++++++++++++ > include/linux/component.h | 31 ++++ > 3 files changed, 411 insertions(+), 1 deletions(-) > create mode 100644 drivers/base/component.c > create mode 100644 include/linux/component.h > > diff --git a/drivers/base/Makefile b/drivers/base/Makefile > index 94e8a80e87f8..870ecfd503af 100644 > --- a/drivers/base/Makefile > +++ b/drivers/base/Makefile > @@ -1,6 +1,6 @@ > # Makefile for the Linux device tree > > -obj-y := core.o bus.o dd.o syscore.o \ > +obj-y := component.o core.o bus.o dd.o syscore.o \ > driver.o class.o platform.o \ > cpu.o firmware.o init.o map.o devres.o \ > attribute_container.o transport_class.o \ > diff --git a/drivers/base/component.c b/drivers/base/component.c > new file mode 100644 > index 000000000000..5492cd8d2247 > --- /dev/null > +++ b/drivers/base/component.c > @@ -0,0 +1,379 @@ > +/* > + * Componentized device handling. > + * > + * This program is free software; you can redistribute it and/or modify > + * it under the terms of the GNU General Public License version 2 as > + * published by the Free Software Foundation. > + * > + * This is work in progress. We gather up the component devices into a list, > + * and bind them when instructed. At the moment, we're specific to the DRM > + * subsystem, and only handles one master device, but this doesn't have to be > + * the case. > + */ > +#include <linux/component.h> > +#include <linux/device.h> > +#include <linux/kref.h> > +#include <linux/list.h> > +#include <linux/module.h> > +#include <linux/mutex.h> > +#include <linux/slab.h> > + > +struct master { > + struct list_head node; > + struct list_head components; > + bool bound; > + > + const struct component_master_ops *ops; > + struct device *dev; > +}; > + > +struct component { > + struct list_head node; > + struct list_head master_node; > + struct master *master; > + bool bound; > + > + const struct component_ops *ops; > + struct device *dev; > +}; > + > +static DEFINE_MUTEX(component_mutex); > +static LIST_HEAD(component_list); > +static LIST_HEAD(masters); > + > +static struct master *__master_find(struct device *dev, const struct component_master_ops *ops) > +{ > + struct master *m; > + > + list_for_each_entry(m, &masters, node) > + if (m->dev == dev && (!ops || m->ops == ops)) > + return m; > + > + return NULL; > +} > + > +/* Attach an unattached component to a master. */ > +static void component_attach_master(struct master *master, struct component *c) > +{ > + c->master = master; > + > + list_add_tail(&c->master_node, &master->components); > +} > + > +/* Detach a component from a master. */ > +static void component_detach_master(struct master *master, struct component *c) > +{ > + list_del(&c->master_node); > + > + c->master = NULL; > +} > + > +int component_master_add_child(struct master *master, > + int (*compare)(struct device *, void *), void *compare_data) > +{ > + struct component *c; > + int ret = -ENXIO; > + > + list_for_each_entry(c, &component_list, node) { > + if (c->master) > + continue; > + > + if (compare(c->dev, compare_data)) { > + component_attach_master(master, c); > + ret = 0; > + break; > + } > + } > + > + return ret; > +} > +EXPORT_SYMBOL_GPL(component_master_add_child); > + > +/* Detach all attached components from this master */ > +static void master_remove_components(struct master *master) > +{ > + while (!list_empty(&master->components)) { > + struct component *c = list_first_entry(&master->components, > + struct component, master_node); > + > + WARN_ON(c->master != master); > + > + component_detach_master(master, c); > + } > +} > + > +/* > + * Try to bring up a master. If component is NULL, we're interested in > + * this master, otherwise it's a component which must be present to try > + * and bring up the master. > + * > + * Returns 1 for successful bringup, 0 if not ready, or -ve errno. > + */ > +static int try_to_bring_up_master(struct master *master, > + struct component *component) > +{ > + int ret = 0; > + > + if (!master->bound) { > + /* > + * Search the list of components, looking for components that > + * belong to this master, and attach them to the master. > + */ > + if (master->ops->add_components(master->dev, master)) { > + /* Failed to find all components */ > + master_remove_components(master); > + ret = 0; > + goto out; > + } > + > + if (component && component->master != master) { > + master_remove_components(master); > + ret = 0; > + goto out; > + } > + > + /* Found all components */ > + ret = master->ops->bind(master->dev); > + if (ret < 0) { > + master_remove_components(master); > + goto out; > + } > + > + master->bound = true; > + ret = 1; > + } > +out: > + > + return ret; > +} > + > +static int try_to_bring_up_masters(struct component *component) > +{ > + struct master *m; > + int ret = 0; > + > + list_for_each_entry(m, &masters, node) { > + ret = try_to_bring_up_master(m, component); > + if (ret != 0) > + break; > + } > + > + return ret; > +} > + > +static void take_down_master(struct master *master) > +{ > + if (master->bound) { > + master->ops->unbind(master->dev); > + master->bound = false; > + } > + > + master_remove_components(master); > +} > + > +int component_master_add(struct device *dev, const struct component_master_ops *ops) > +{ > + struct master *master; > + int ret; > + > + master = kzalloc(sizeof(*master), GFP_KERNEL); > + if (!master) > + return -ENOMEM; > + > + master->dev = dev; > + master->ops = ops; > + INIT_LIST_HEAD(&master->components); > + > + /* Add to the list of available masters. */ > + mutex_lock(&component_mutex); > + list_add(&master->node, &masters); > + > + ret = try_to_bring_up_master(master, NULL); > + > + if (ret < 0) { > + /* Delete off the list if we weren't successful */ > + list_del(&master->node); > + kfree(master); > + } > + mutex_unlock(&component_mutex); > + > + return ret < 0 ? ret : 0; > +} > +EXPORT_SYMBOL_GPL(component_master_add); > + > +void component_master_del(struct device *dev, const struct component_master_ops *ops) > +{ > + struct master *master; > + > + mutex_lock(&component_mutex); > + master = __master_find(dev, ops); > + if (master) { > + take_down_master(master); > + > + list_del(&master->node); > + kfree(master); > + } > + mutex_unlock(&component_mutex); > +} > +EXPORT_SYMBOL_GPL(component_master_del); > + > +static void component_unbind(struct component *component, > + struct master *master, void *data) > +{ > + WARN_ON(!component->bound); > + > + component->ops->unbind(component->dev, master->dev, data); > + component->bound = false; > + > + /* Release all resources claimed in the binding of this component */ > + devres_release_group(component->dev, component); > +} > + > +void component_unbind_all(struct device *master_dev, void *data) > +{ > + struct master *master; > + struct component *c; > + > + WARN_ON(!mutex_is_locked(&component_mutex)); > + > + master = __master_find(master_dev, NULL); > + if (!master) > + return; > + > + list_for_each_entry_reverse(c, &master->components, master_node) > + component_unbind(c, master, data); > +} > +EXPORT_SYMBOL_GPL(component_unbind_all); > + > +static int component_bind(struct component *component, struct master *master, > + void *data) > +{ > + int ret; > + > + /* > + * Each component initialises inside its own devres group. > + * This allows us to roll-back a failed component without > + * affecting anything else. > + */ > + if (!devres_open_group(master->dev, NULL, GFP_KERNEL)) > + return -ENOMEM; > + > + /* > + * Also open a group for the device itself: this allows us > + * to release the resources claimed against the sub-device > + * at the appropriate moment. > + */ > + if (!devres_open_group(component->dev, component, GFP_KERNEL)) { > + devres_release_group(master->dev, NULL); > + return -ENOMEM; > + } > + > + dev_dbg(master->dev, "binding %s (ops %ps)\n", > + dev_name(component->dev), component->ops); > + > + ret = component->ops->bind(component->dev, master->dev, data); > + if (!ret) { > + component->bound = true; > + > + /* > + * Close the component device's group so that resources > + * allocated in the binding are encapsulated for removal > + * at unbind. Remove the group on the DRM device as we > + * can clean those resources up independently. > + */ > + devres_close_group(component->dev, NULL); > + devres_remove_group(master->dev, NULL); > + > + dev_info(master->dev, "bound %s (ops %ps)\n", > + dev_name(component->dev), component->ops); > + } else { > + devres_release_group(component->dev, NULL); > + devres_release_group(master->dev, NULL); > + > + dev_err(master->dev, "failed to bind %s (ops %ps): %d\n", > + dev_name(component->dev), component->ops, ret); > + } > + > + return ret; > +} > + > +int component_bind_all(struct device *master_dev, void *data) > +{ > + struct master *master; > + struct component *c; > + int ret = 0; > + > + WARN_ON(!mutex_is_locked(&component_mutex)); > + > + master = __master_find(master_dev, NULL); > + if (!master) > + return -EINVAL; > + > + list_for_each_entry(c, &master->components, master_node) { > + ret = component_bind(c, master, data); > + if (ret) > + break; > + } > + > + if (ret != 0) { > + list_for_each_entry_continue_reverse(c, &master->components, > + master_node) > + component_unbind(c, master, data); > + } > + > + return ret; > +} > +EXPORT_SYMBOL_GPL(component_bind_all); > + > +int component_add(struct device *dev, const struct component_ops *ops) > +{ > + struct component *component; > + int ret; > + > + component = kzalloc(sizeof(*component), GFP_KERNEL); > + if (!component) > + return -ENOMEM; > + > + component->ops = ops; > + component->dev = dev; > + > + dev_dbg(dev, "adding component (ops %ps)\n", ops); > + > + mutex_lock(&component_mutex); > + list_add_tail(&component->node, &component_list); > + > + ret = try_to_bring_up_masters(component); > + if (ret < 0) { > + list_del(&component->node); > + > + kfree(component); > + } > + mutex_unlock(&component_mutex); > + > + return ret < 0 ? ret : 0; > +} > +EXPORT_SYMBOL_GPL(component_add); > + > +void component_del(struct device *dev, const struct component_ops *ops) > +{ > + struct component *c, *component = NULL; > + > + mutex_lock(&component_mutex); > + list_for_each_entry(c, &component_list, node) > + if (c->dev == dev && c->ops == ops) { > + list_del(&c->node); > + component = c; > + break; > + } > + > + if (component && component->master) > + take_down_master(component->master); > + > + mutex_unlock(&component_mutex); > + > + WARN_ON(!component); > + kfree(component); > +} > +EXPORT_SYMBOL_GPL(component_del); > + > +MODULE_LICENSE("GPL v2"); > diff --git a/include/linux/component.h b/include/linux/component.h > new file mode 100644 > index 000000000000..73657636db0b > --- /dev/null > +++ b/include/linux/component.h > @@ -0,0 +1,31 @@ > +#ifndef COMPONENT_H > +#define COMPONENT_H > + > +struct device; > + > +struct component_ops { > + int (*bind)(struct device *, struct device *, void *); > + void (*unbind)(struct device *, struct device *, void *); > +}; > + > +int component_add(struct device *, const struct component_ops *); > +void component_del(struct device *, const struct component_ops *); > + > +int component_bind_all(struct device *, void *); > +void component_unbind_all(struct device *, void *); > + > +struct master; > + > +struct component_master_ops { > + int (*add_components)(struct device *, struct master *); > + int (*bind)(struct device *); > + void (*unbind)(struct device *); > +}; > + > +int component_master_add(struct device *, const struct component_master_ops *); > +void component_master_del(struct device *, const struct component_master_ops *); > + > +int component_master_add_child(struct master *master, > + int (*compare)(struct device *, void *), void *compare_data); > + > +#endif > -- > 1.7.4.4 > > _______________________________________________ > dri-devel mailing list > dri-devel(a)lists.freedesktop.org > http://lists.freedesktop.org/mailman/listinfo/dri-devel -- Daniel Vetter Software Engineer, Intel Corporation +41 (0) 79 365 57 48 - http://blog.ffwll.ch

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[alsa-devel] [PATCH] ALSA: hda - Fix leftover ifdef checks after modularization
by Takashi Iwai 07 Feb '14

07 Feb '14

Since the commit [595fe1b702c3: ALSA: hda - Make CONFIG_SND_HDA_CODEC_* tristate], the kconfig variables for the generic parser and codec drivers can be "m" instead of boolean, but some codes are left unchanged to check only #ifdef CONFIG_SND_HDA_CODEC_XXX, which is no longer true for modules. This patch fixes them by replacing with IS_ENABLED() macros. Fixes: 595fe1b702c3 ('ALSA: hda - Make CONFIG_SND_HDA_CODEC_* tristate') Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70161 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 6 +++--- sound/pci/hda/hda_intel.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ec4536c8d8d4..69acbf758bbb 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -932,7 +932,7 @@ int snd_hda_bus_new(struct snd_card *card, } EXPORT_SYMBOL_GPL(snd_hda_bus_new); -#ifdef CONFIG_SND_HDA_GENERIC +#if IS_ENABLED(CONFIG_SND_HDA_GENERIC) #define is_generic_config(codec) \ (codec->modelname && !strcmp(codec->modelname, "generic")) #else @@ -1570,7 +1570,7 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec) EXPORT_SYMBOL_GPL(snd_hda_codec_update_widgets); -#ifdef CONFIG_SND_HDA_CODEC_HDMI +#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) /* if all audio out widgets are digital, let's assume the codec as a HDMI/DP */ static bool is_likely_hdmi_codec(struct hda_codec *codec) { @@ -1622,7 +1622,7 @@ int snd_hda_codec_configure(struct hda_codec *codec) unload_parser(codec); /* to be sure */ if (is_likely_hdmi_codec(codec)) patch = load_parser(codec, snd_hda_parse_hdmi_codec); -#ifdef CONFIG_SND_HDA_GENERIC +#if IS_ENABLED(CONFIG_SND_HDA_GENERIC) if (!patch) patch = load_parser(codec, snd_hda_parse_generic_codec); #endif diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fa2879a21a50..e354ab1ec20f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -198,7 +198,7 @@ MODULE_DESCRIPTION("Intel HDA driver"); #endif #if defined(CONFIG_PM) && defined(CONFIG_VGA_SWITCHEROO) -#ifdef CONFIG_SND_HDA_CODEC_HDMI +#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) #define SUPPORT_VGA_SWITCHEROO #endif #endif -- 1.8.5.2

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[alsa-devel] [PATCH] ALSA: hda - Add a headset quirk for Dell XPS 13
by David Henningsson 07 Feb '14

07 Feb '14

This quirk is needed for the headset microphone to work. Alsa-info at http://www.alsa-project.org/db/?f=8c7dfe857ceff462ca2de133e67023c0f68de9cb Cc: stable(a)vger.kernel.org (3.10+) Reported-by: Po-Hsu Lin <po-hsu.lin(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(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 bee4b8c..2cb6a39 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5146,6 +5146,7 @@ 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, 0x060a, "Dell XPS 13", 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), -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Rename ASUS subwoofer quirks
by David Henningsson 07 Feb '14

07 Feb '14

Just a small refactoring to make the next patch slightly simpler. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bee4b8c..0295efe 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4925,9 +4925,9 @@ enum { ALC662_FIXUP_INV_DMIC, ALC668_FIXUP_DELL_MIC_NO_PRESENCE, ALC668_FIXUP_HEADSET_MODE, - ALC662_FIXUP_BASS_CHMAP, + ALC662_FIXUP_BASS_MODE4_CHMAP, ALC662_FIXUP_BASS_1A, - ALC662_FIXUP_BASS_1A_CHMAP, + ALC662_FIXUP_BASS_CHMAP, ALC668_FIXUP_AUTO_MUTE, }; @@ -5115,7 +5115,7 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_alc668, }, - [ALC662_FIXUP_BASS_CHMAP] = { + [ALC662_FIXUP_BASS_MODE4_CHMAP] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_bass_chmap, .chained = true, @@ -5127,12 +5127,12 @@ static const struct hda_fixup alc662_fixups[] = { {0x1a, 0x80106111}, /* bass speaker */ {} }, + .chained = true, + .chain_id = ALC662_FIXUP_BASS_CHMAP, }, - [ALC662_FIXUP_BASS_1A_CHMAP] = { + [ALC662_FIXUP_BASS_CHMAP] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_bass_chmap, - .chained = true, - .chain_id = ALC662_FIXUP_BASS_1A, }, }; @@ -5153,9 +5153,9 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { 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), - SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_CHMAP), + SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A), + SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), + SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), -- 1.7.9.5

2 2

[alsa-devel] [PATCH v2] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 06 Feb '14

06 Feb '14

This patch adds initial support for the Behringer BCD2000 USB DJ controller. At the moment, only the MIDI part of the device is working, i.e. knobs, buttons and LEDs. I also plan to add support for the audio part, but I assume that this will require more effort than the rather simple MIDI interface. Progress can be tracked at https://github.com/anyc/snd-usb-bcd2000. Changes since v1: - fixed the various code style issues, thanks to Daniel Mack and checkpatch.pl. Signed-off-by: Mario Kicherer <dev(a)kicherer.org> --- sound/usb/Kconfig | 14 ++ sound/usb/Makefile | 2 +- sound/usb/bcd2000/Makefile | 3 + sound/usb/bcd2000/bcd2000.c | 553 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 571 insertions(+), 1 deletion(-) create mode 100644 sound/usb/bcd2000/Makefile create mode 100644 sound/usb/bcd2000/bcd2000.c diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index de9408b..2abafea 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -146,5 +146,19 @@ config SND_USB_HIFACE To compile this driver as a module, choose M here: the module will be called snd-usb-hiface. +config SND_USB_BCD2000 + tristate "Behringer BCD2000 MIDI driver" + select SND_HWDEP + select SND_RAWMIDI + help + Say Y here to include MIDI support for the Behringer BCD2000 DJ + controller. + + Audio support is still work-in-progress at + https://github.com/anyc/snd-usb-bcd2000 + + To compile this driver as a module, choose M here: the module + will be called snd-usb-bcd2000. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index abe668f..2b92f0d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ bcd2000/ diff --git a/sound/usb/bcd2000/Makefile b/sound/usb/bcd2000/Makefile new file mode 100644 index 0000000..6540136 --- /dev/null +++ b/sound/usb/bcd2000/Makefile @@ -0,0 +1,3 @@ +snd-usb-bcd2000-y := bcd2000.o + +obj-$(CONFIG_SND_USB_BCD2000) += snd-usb-bcd2000.o \ No newline at end of file diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c new file mode 100644 index 0000000..d8c5525 --- /dev/null +++ b/sound/usb/bcd2000/bcd2000.c @@ -0,0 +1,553 @@ +/* + * Behringer BCD2000 driver + * + * Copyright (C) 2014 Mario Kicherer (dev(a)kicherer.org) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/usb.h> +#include <linux/usb/audio.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/rawmidi.h> +#include "../usbaudio.h" +#include "../midi.h" + +#define DEVICE_NAME "Behringer BCD2000" +#define DEVICE_SHORTNAME "bcd2000" + +#define PREFIX DEVICE_SHORTNAME ": " +#define BUFSIZE 64 + +static struct usb_device_id id_table[] = { + { USB_DEVICE(0x1397, 0x00bd) }, + { }, +}; + +static int debug; +module_param(debug, int, S_IRUGO | S_IWUSR); +MODULE_PARM_DESC(debug, "enable debug output (default: 0)"); + +static unsigned char device_cmd_prefix[] = {0x03, 0x00}; + +static unsigned char bcd2000_init_sequence[] = { + 0x07, 0x00, 0x00, 0x00, 0x78, 0x48, 0x1c, 0x81, + 0xc4, 0x00, 0x00, 0x00, 0x5e, 0x53, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff, 0x6c, 0xf3, 0x90, 0xff, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, + 0x18, 0xfa, 0x11, 0xff, 0x14, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0xf2, 0x34, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff}; + +struct bcd2000 { + struct usb_device *dev; + struct snd_card *card; + struct usb_interface *intf; + int card_index; + struct snd_pcm *pcm; + struct list_head midi_list; + spinlock_t lock; + struct mutex mutex; + + int midi_out_active; + struct snd_rawmidi *rmidi; + struct snd_rawmidi_substream *midi_receive_substream; + struct snd_rawmidi_substream *midi_out_substream; + + unsigned char midi_in_buf[BUFSIZE]; + unsigned char midi_out_buf[BUFSIZE]; + + unsigned char midi_cmd_buf[3]; + unsigned char midi_cmd_offset; + + struct urb *midi_out_urb; + struct urb *midi_in_urb; +}; + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +static DEFINE_MUTEX(devices_mutex); +static unsigned int devices_used; +static struct usb_driver bcd2000_driver; + + + + + + +static int bcd2000_midi_input_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_input_close(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static void bcd2000_midi_input_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (!bcd2k) + return; + + bcd2k->midi_receive_substream = up ? substream : NULL; +} + +static void bcd2000_midi_handle_input(struct bcd2000 *bcd2k, + const unsigned char *buf, int len) +{ + unsigned char length, cmd, buf_offset, tocopy; + + if (!bcd2k->midi_receive_substream) + return; + + if (debug) { + print_hex_dump(KERN_DEBUG, "received from device: ", + DUMP_PREFIX_NONE, 16, 1, buf, len, false); + } + + if (len < 2) + return; + + /* + * The BCD2000 sends MIDI commands with 2 or 3 bytes length. In case of + * 2 bytes, the command is the same as before. + * + * Packet structure: mm nn oo (pp) + * mm: payload length + * nn: MIDI command or note + * oo: note or velocity + * pp: velocity + */ + + length = buf[0]; + + /* ignore packets without payload */ + if (length == 0) + return; + + /* + * The MIDI packets the BCD2000 sends can be arbitrarily truncated or + * concatenated. Therefore, this loop accumulates the bytes from the + * input buffer until a full valid MIDI packet is in the MIDI command + * buffer "midi_cmd_buf". + */ + + buf_offset = 1; + while (buf_offset <= length) { + cmd = buf[buf_offset]; + + if (bcd2k->midi_cmd_offset == 0 && cmd != 0x90 && cmd != 0xb0) { + /* + * this is a 2-byte midi packet -> reuse command byte + * from last midi packet + */ + bcd2k->midi_cmd_offset = 1; + } + + /* determine the number of bytes we want to copy this time */ + tocopy = min(3 - bcd2k->midi_cmd_offset, + length - (buf_offset - 1)); + + /* safety check */ + if (bcd2k->midi_cmd_offset + tocopy < BUFSIZE && + buf_offset + tocopy < len) { + memcpy(&bcd2k->midi_cmd_buf[bcd2k->midi_cmd_offset], + &buf[buf_offset], tocopy); + } else { + snd_printk(KERN_ERR PREFIX "access violation in %s\n", + __func__); + } + + bcd2k->midi_cmd_offset += tocopy; + buf_offset += tocopy; + + /* is our MIDI packet complete? */ + if (bcd2k->midi_cmd_offset == 3) { + if (debug) { + print_hex_dump(KERN_DEBUG, + "sending to userspace: ", + DUMP_PREFIX_NONE, 16, 1, + bcd2k->midi_cmd_buf, + bcd2k->midi_cmd_offset, false); + } + + /* send MIDI packet */ + snd_rawmidi_receive(bcd2k->midi_receive_substream, + bcd2k->midi_cmd_buf, 3); + + bcd2k->midi_cmd_offset = 0; + } + } +} + +static int bcd2000_midi_output_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_output_close(struct snd_rawmidi_substream *substream) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + if (bcd2k->midi_out_active) { + usb_free_urb(bcd2k->midi_out_urb); + bcd2k->midi_out_active = 0; + } + return 0; +} + +static void bcd2000_midi_send(struct bcd2000 *bcd2k, + struct snd_rawmidi_substream *substream) +{ + int len, ret; + + BUILD_BUG_ON(sizeof(device_cmd_prefix) >= BUFSIZE); + /* copy the "set LED" command bytes */ + memcpy(bcd2k->midi_out_buf, device_cmd_prefix, + sizeof(device_cmd_prefix)); + + /* + * get MIDI packet and leave space for command prefix + * and payload length + */ + len = snd_rawmidi_transmit(substream, + bcd2k->midi_out_buf + 3, BUFSIZE - 3); + + if (len <= 0) + return; + + /* set payload length */ + bcd2k->midi_out_buf[2] = len; + bcd2k->midi_out_urb->transfer_buffer_length = BUFSIZE; + + if (debug) { + print_hex_dump(KERN_DEBUG, "sending to device: ", + DUMP_PREFIX_NONE, 16, 1, + bcd2k->midi_out_buf, len+3, false); + } + + /* send packet to the BCD2000 */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) { + dev_err(&bcd2k->dev->dev, PREFIX + "%s (%p): usb_submit_urb() failed" + "ret=%d, len=%d\n", __func__, substream, ret, len); + } else { + bcd2k->midi_out_active = 1; + } +} + +static void bcd2000_midi_output_done(struct urb *urb) +{ + struct bcd2000 *bcd2k = urb->context; + + bcd2k->midi_out_active = 0; + if (urb->status != 0) { + dev_err(&urb->dev->dev, PREFIX "urb status: %d\n", urb->status); + return; + } + + if (!bcd2k->midi_out_substream) + return; + + bcd2000_midi_send(bcd2k, bcd2k->midi_out_substream); +} + +static void bcd2000_command_reply_dispatch(struct urb *urb) +{ + int ret; + struct device *dev = &urb->dev->dev; + struct bcd2000 *bcd2k = urb->context; + + if (urb->status || !bcd2k) { + dev_warn(dev, "received urb->status = %i\n", urb->status); + return; + } + + if (urb->actual_length > 0) + bcd2000_midi_handle_input(bcd2k, urb->transfer_buffer, + urb->actual_length); + + bcd2k->midi_in_urb->actual_length = 0; + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(dev, "unable to submit urb. OOM!?\n"); +} + +static void bcd2000_midi_output_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (up) { + bcd2k->midi_out_substream = substream; + if (!bcd2k->midi_out_active) + bcd2000_midi_send(bcd2k, substream); + } else { + bcd2k->midi_out_substream = NULL; + } +} + +static struct snd_rawmidi_ops bcd2000_midi_output = { + .open = bcd2000_midi_output_open, + .close = bcd2000_midi_output_close, + .trigger = bcd2000_midi_output_trigger, +}; + +static struct snd_rawmidi_ops bcd2000_midi_input = { + .open = bcd2000_midi_input_open, + .close = bcd2000_midi_input_close, + .trigger = bcd2000_midi_input_trigger, +}; + +static void bcd2000_init_device(struct bcd2000 *bcd2k) +{ + int ret; + + /* copy init sequence into buffer */ + memcpy(bcd2k->midi_out_buf, bcd2000_init_sequence, 52); + bcd2k->midi_out_urb->transfer_buffer_length = 52; + + /* submit sequence */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) { + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() failed," + "ret=%d: ", __func__, ret); + } else { + bcd2k->midi_out_active = 1; + } + + /* send empty packet to enable button and controller events */ + bcd2k->midi_in_urb->actual_length = 0; + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); +} + +static void bcd2000_init_midi(struct bcd2000 *bcd2k) +{ + int ret; + struct snd_rawmidi *rmidi; + + ret = snd_rawmidi_new(bcd2k->card, bcd2k->card->shortname, 0, + 1, /* output */ + 1, /* input */ + &rmidi); + + if (ret < 0) + return; + + strlcpy(rmidi->name, bcd2k->card->shortname, sizeof(rmidi->name)); + + rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; + rmidi->private_data = bcd2k; + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &bcd2000_midi_output); + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &bcd2000_midi_input); + + bcd2k->rmidi = rmidi; + + bcd2k->midi_in_urb = usb_alloc_urb(0, GFP_KERNEL); + bcd2k->midi_out_urb = usb_alloc_urb(0, GFP_KERNEL); + + usb_fill_int_urb(bcd2k->midi_in_urb, bcd2k->dev, + usb_rcvbulkpipe(bcd2k->dev, 0x81), + bcd2k->midi_in_buf, BUFSIZE, + bcd2000_command_reply_dispatch, bcd2k, 1); + + usb_fill_int_urb(bcd2k->midi_out_urb, bcd2k->dev, + usb_sndbulkpipe(bcd2k->dev, 0x1), + bcd2k->midi_out_buf, BUFSIZE, + bcd2000_midi_output_done, bcd2k, 1); + + bcd2000_init_device(bcd2k); +} + +static void bcd2000_close_midi(struct bcd2000 *bcd2k) +{ + usb_free_urb(bcd2k->midi_out_urb); + usb_free_urb(bcd2k->midi_in_urb); +} + + + + + +/* + * general part + */ + +static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k, + struct usb_interface *interface) +{ + struct usb_interface *intf; + + mutex_lock(&bcd2k->mutex); + intf = bcd2k->intf; + bcd2k->intf = NULL; + mutex_unlock(&bcd2k->mutex); + if (intf) { + usb_set_intfdata(intf, NULL); + if (intf != interface) + usb_driver_release_interface(&bcd2000_driver, + intf); + } +} + +static void bcd2000_card_free(struct snd_card *card) +{ + /* empty for now */ +} + +static int bcd2000_probe(struct usb_interface *interface, + const struct usb_device_id *usb_id) +{ + struct snd_card *card; + struct bcd2000 *bcd2k; + unsigned int card_index; + char usb_path[32]; + int err; + + + mutex_lock(&devices_mutex); + + for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) + if (enable[card_index] && !(devices_used & (1 << card_index))) + break; + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + err = snd_card_create(index[card_index], id[card_index], THIS_MODULE, + sizeof(*bcd2k), &card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return err; + } + card->private_free = bcd2000_card_free; + bcd2k = card->private_data; + bcd2k->dev = interface_to_usbdev(interface); + bcd2k->card = card; + bcd2k->card_index = card_index; + INIT_LIST_HEAD(&bcd2k->midi_list); + spin_lock_init(&bcd2k->lock); + mutex_init(&bcd2k->mutex); + + bcd2k->intf = interface; + + snd_card_set_dev(card, &interface->dev); + + strcpy(card->driver, DEVICE_NAME); + strcpy(card->shortname, DEVICE_SHORTNAME); + usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); + snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), + DEVICE_NAME ", at %s, %s speed", + usb_path, + bcd2k->dev->speed == USB_SPEED_HIGH ? "high" : "full"); + + dev_info(&bcd2k->dev->dev, PREFIX "%s", bcd2k->card->longname); + + + + bcd2000_init_midi(bcd2k); + + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + usb_set_intfdata(interface, bcd2k); + devices_used |= 1 << card_index; + + mutex_unlock(&devices_mutex); + return 0; + +probe_error: + bcd2000_free_usb_related_resources(bcd2k, interface); + snd_card_free(card); + mutex_unlock(&devices_mutex); + return err; +} + +static void bcd2000_disconnect(struct usb_interface *interface) +{ + struct bcd2000 *bcd2k = usb_get_intfdata(interface); + struct list_head *midi; + + if (!bcd2k) + return; + + bcd2000_close_midi(bcd2k); + + mutex_lock(&devices_mutex); + + /* make sure that userspace cannot create new requests */ + snd_card_disconnect(bcd2k->card); + + /* make sure that there are no pending USB requests */ + list_for_each(midi, &bcd2k->midi_list) + snd_usbmidi_disconnect(midi); + + bcd2000_free_usb_related_resources(bcd2k, interface); + + devices_used &= ~(1 << bcd2k->card_index); + + snd_card_free_when_closed(bcd2k->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_driver bcd2000_driver = { + .name = DEVICE_SHORTNAME, + .probe = bcd2000_probe, + .disconnect = bcd2000_disconnect, + .id_table = id_table, +}; + +static int __init bcd2000_init(void) +{ + int retval = 0; + + retval = usb_register(&bcd2000_driver); + if (retval) + snd_printk(KERN_INFO "usb_register failed. Error: %d", retval); + return retval; +} + +static void __exit bcd2000_exit(void) +{ + usb_deregister(&bcd2000_driver); +} + +module_init(bcd2000_init); +module_exit(bcd2000_exit); + +MODULE_DEVICE_TABLE(usb, id_table); +MODULE_AUTHOR("Mario Kicherer, dev(a)kicherer.org"); +MODULE_DESCRIPTION("Behringer BCD2000 driver"); +MODULE_LICENSE("GPL"); -- 1.8.3.2

4 7

[alsa-devel] [PATCH] ASoC: pcm512x: More constification
by Mark Brown 06 Feb '14

06 Feb '14

From: Mark Brown <broonie(a)linaro.org> Since the core now takes const strings for enums we should be constifying them (and the regulator supplies while we're at it). Reported-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm512x.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 5ad3e9aa3cb4..1150381fc373 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -30,7 +30,7 @@ #include "pcm512x.h" #define PCM512x_NUM_SUPPLIES 3 -static const char *pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = { +static const char * const pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = { "AVDD", "DVDD", "CPVDD", @@ -167,7 +167,7 @@ static bool pcm512x_volatile(struct device *dev, unsigned int reg) static const DECLARE_TLV_DB_SCALE(digital_tlv, -10350, 50, 1); -static const char *pcm512x_dsp_program_texts[] = { +static const char * const pcm512x_dsp_program_texts[] = { "FIR interpolation with de-emphasis", "Low latency IIR with de-emphasis", "Fixed process flow", @@ -188,14 +188,14 @@ static const SOC_VALUE_ENUM_SINGLE_DECL(pcm512x_dsp_program, pcm512x_dsp_program_texts, pcm512x_dsp_program_values); -static const char *pcm512x_clk_missing_text[] = { +static const char * const pcm512x_clk_missing_text[] = { "1s", "2s", "3s", "4s", "5s", "6s", "7s", "8s" }; static const struct soc_enum pcm512x_clk_missing = SOC_ENUM_SINGLE(PCM512x_CLKDET, 0, 8, pcm512x_clk_missing_text); -static const char *pcm512x_autom_text[] = { +static const char * const pcm512x_autom_text[] = { "21ms", "106ms", "213ms", "533ms", "1.07s", "2.13s", "5.33s", "10.66s" }; @@ -207,7 +207,7 @@ static const struct soc_enum pcm512x_autom_r = SOC_ENUM_SINGLE(PCM512x_AUTO_MUTE, PCM512x_ATMR_SHIFT, 8, pcm512x_autom_text); -static const char *pcm512x_ramp_rate_text[] = { +static const char * const pcm512x_ramp_rate_text[] = { "1 sample/update", "2 samples/update", "4 samples/update", "Immediate" }; @@ -224,7 +224,7 @@ static const struct soc_enum pcm512x_vedf = SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDF_SHIFT, 4, pcm512x_ramp_rate_text); -static const char *pcm512x_ramp_step_text[] = { +static const char * const pcm512x_ramp_step_text[] = { "4dB/step", "2dB/step", "1dB/step", "0.5dB/step" }; -- 1.9.rc1

1 0

[alsa-devel] [PATCH 0/8] da9055: Driver initialisation fixes, add DT support
by Adam Thomson 06 Feb '14

06 Feb '14

This patch series provides the following updates for DA9055 drivers: - Fixes an issue with da9055 driver initialisation (conflicting device ids) of PMIC (MFD) and CODEC drivers. - Add initial DT support for DA9055 related drivers, including binding documentation. - Remove conflicting use of platform_get_irq_byname() in driver probes. - Remove unnecessary resource structures for MFD cells as a result of removing use of platform_get_irq_byname(). Adam Thomson (8): ASoC: da9055: Fix device registration of PMIC and CODEC devices ASoC: da9055: Add DT support for CODEC mfd: da9055: Add DT support for PMIC regulator: da9055: Add DT support onkey: da9055: Remove use of platform_get_irq_byname() hwmon: da9055: Remove use of platform_get_irq_byname() rtc: da9055: Remove use of platform_get_irq_byname() mfd: da9055: Remove unused resource structures for mfd cells. Documentation/devicetree/bindings/mfd/da9055.txt | 73 ++++++++++++++++++++ Documentation/devicetree/bindings/sound/da9055.txt | 22 ++++++ drivers/hwmon/da9055-hwmon.c | 7 +-- drivers/input/misc/da9055_onkey.c | 9 +-- drivers/mfd/da9055-core.c | 46 ------------ drivers/mfd/da9055-i2c.c | 20 +++++- drivers/regulator/da9055-regulator.c | 59 ++++++++++++++-- drivers/rtc/rtc-da9055.c | 3 +- sound/soc/codecs/da9055.c | 19 +++++- 9 files changed, 188 insertions(+), 70 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/da9055.txt create mode 100644 Documentation/devicetree/bindings/sound/da9055.txt

4 18

[alsa-devel] [PATCH] ALSA: hda - Fix inconsistent Mic mute LED
by Takashi Iwai 06 Feb '14

06 Feb '14

The current code for controlling mic mute LED in patch_sigmatel.c blindly assumes that there is a single capture switch. But, there can be multiple multiple ones, and each of them flips the state, ended up in an inconsistent state. For fixing this problem, this patch adds kcontrol to be passed to the hook function so that the callee can check which switch is being accessed. In stac_capture_led_hook(), the state is checked as a bitmask, and turns on the LED when all capture switches are off. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 8 ++++---- sound/pci/hda/hda_generic.h | 1 + sound/pci/hda/patch_conexant.c | 3 ++- sound/pci/hda/patch_realtek.c | 9 ++++++--- sound/pci/hda/patch_sigmatel.c | 27 +++++++++++++++++---------- sound/pci/hda/thinkpad_helper.c | 1 + 6 files changed, 31 insertions(+), 18 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8321a97d5c05..d9a09bdd09db 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3269,7 +3269,7 @@ static int cap_put_caller(struct snd_kcontrol *kcontrol, mutex_unlock(&codec->control_mutex); snd_hda_codec_flush_cache(codec); /* flush the updates */ if (err >= 0 && spec->cap_sync_hook) - spec->cap_sync_hook(codec, ucontrol); + spec->cap_sync_hook(codec, kcontrol, ucontrol); return err; } @@ -3390,7 +3390,7 @@ static int cap_single_sw_put(struct snd_kcontrol *kcontrol, return ret; if (spec->cap_sync_hook) - spec->cap_sync_hook(codec, ucontrol); + spec->cap_sync_hook(codec, kcontrol, ucontrol); return ret; } @@ -3795,7 +3795,7 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx, return 0; snd_hda_activate_path(codec, path, true, false); if (spec->cap_sync_hook) - spec->cap_sync_hook(codec, NULL); + spec->cap_sync_hook(codec, NULL, NULL); path_power_down_sync(codec, old_path); return 1; } @@ -5270,7 +5270,7 @@ static void init_input_src(struct hda_codec *codec) } if (spec->cap_sync_hook) - spec->cap_sync_hook(codec, NULL); + spec->cap_sync_hook(codec, NULL, NULL); } /* set right pin controls for digital I/O */ diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 07f767231c9f..c908afbe4d94 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -274,6 +274,7 @@ struct hda_gen_spec { void (*init_hook)(struct hda_codec *codec); void (*automute_hook)(struct hda_codec *codec); void (*cap_sync_hook)(struct hda_codec *codec, + struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); /* PCM hooks */ diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4e0ec146553d..bcf91bea3317 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3291,7 +3291,8 @@ static void cxt_update_headset_mode(struct hda_codec *codec) } static void cxt_update_headset_mode_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { cxt_update_headset_mode(codec); } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 56a8f1876603..6aabaa1c19f5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -708,7 +708,8 @@ static void alc_inv_dmic_sync(struct hda_codec *codec, bool force) } static void alc_inv_dmic_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { alc_inv_dmic_sync(codec, false); } @@ -3211,7 +3212,8 @@ static void alc269_fixup_hp_gpio_mute_hook(void *private_data, int enabled) /* turn on/off mic-mute LED per capture hook */ static void alc269_fixup_hp_gpio_mic_mute_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct alc_spec *spec = codec->spec; unsigned int oldval = spec->gpio_led; @@ -3521,7 +3523,8 @@ static void alc_update_headset_mode(struct hda_codec *codec) } static void alc_update_headset_mode_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { alc_update_headset_mode(codec); } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6998cf29b9bc..7311badf6a94 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -194,7 +194,7 @@ struct sigmatel_spec { int default_polarity; unsigned int mic_mute_led_gpio; /* capture mute LED GPIO */ - bool mic_mute_led_on; /* current mic mute state */ + unsigned int mic_enabled; /* current mic mute state (bitmask) */ /* stream */ unsigned int stream_delay; @@ -324,19 +324,26 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, /* hook for controlling mic-mute LED GPIO */ static void stac_capture_led_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct sigmatel_spec *spec = codec->spec; - bool mute; + unsigned int mask; + bool cur_mute, prev_mute; - if (!ucontrol) + if (!kcontrol || !ucontrol) return; - mute = !(ucontrol->value.integer.value[0] || - ucontrol->value.integer.value[1]); - if (spec->mic_mute_led_on != mute) { - spec->mic_mute_led_on = mute; - if (mute) + mask = 1U << snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + prev_mute = !spec->mic_enabled; + if (ucontrol->value.integer.value[0] || + ucontrol->value.integer.value[1]) + spec->mic_enabled |= mask; + else + spec->mic_enabled &= ~mask; + cur_mute = !spec->mic_enabled; + if (cur_mute != prev_mute) { + if (cur_mute) spec->gpio_data |= spec->mic_mute_led_gpio; else spec->gpio_data &= ~spec->mic_mute_led_gpio; @@ -4462,7 +4469,7 @@ static void stac_setup_gpio(struct hda_codec *codec) if (spec->mic_mute_led_gpio) { spec->gpio_mask |= spec->mic_mute_led_gpio; spec->gpio_dir |= spec->mic_mute_led_gpio; - spec->mic_mute_led_on = true; + spec->mic_enabled = 0; spec->gpio_data |= spec->mic_mute_led_gpio; spec->gen.cap_sync_hook = stac_capture_led_hook; diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 5799fbc24c28..8fe3b8c18ed4 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -39,6 +39,7 @@ static void update_tpacpi_mute_led(void *private_data, int enabled) } static void update_tpacpi_micmute_led(struct hda_codec *codec, + struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { if (!ucontrol || !led_set_func) -- 1.8.5.2

2 6

[alsa-devel] [alsa-plugins] [PATCH 2/2] Allow number of channels
by Corfu, ValentinX 06 Feb '14

06 Feb '14

>From 92cf8488d3d2e271fdb4b1fb41fbc006cc59677b Mon Sep 17 00:00:00 2001 From: Valentin Corfu <valentinx.corfu(a)intel.com> Date: Fri, 31 Jan 2014 08:08:51 -0500 Subject: [PATCH 2/2] 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 475c619..feef4a0 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -325,7 +325,7 @@ 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, long num_channels) { snd_pcm_jack_t *jack; int err; @@ -341,11 +341,19 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, jack->fd = -1; jack->io.poll_fd = -1; - err = parse_ports(jack, stream == SND_PCM_STREAM_PLAYBACK ? - playback_conf : capture_conf); - if (err) { - snd_pcm_jack_free(jack); - return err; + if (num_channels > 0) { + jack->num_ports = num_channels; + jack->port_names = calloc(num_channels, sizeof(char*)); + if (jack->port_names == NULL) + return -ENOMEM; + } + else { + err = parse_ports(jack, stream == SND_PCM_STREAM_PLAYBACK ? + playback_conf : capture_conf); + if (err) { + snd_pcm_jack_free(jack); + return err; + } } jack->channels = jack->num_ports; @@ -422,6 +430,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) snd_config_t *playback_conf = NULL; snd_config_t *capture_conf = NULL; int err; + long num_channels = 0; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); @@ -434,6 +443,10 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) snd_config_get_string(n, &name); 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); @@ -454,7 +467,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) return -EINVAL; } - err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode, num_channels); 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 to specify a client name as parameters
by Corfu, ValentinX 06 Feb '14

06 Feb '14

>From 1ac8268330df7564edf252a542a27f3cf4f4472f Mon Sep 17 00:00:00 2001 From: Valentin Corfu <valentinx.corfu(a)intel.com> Date: Fri, 31 Jan 2014 08:06:23 -0500 Subject: [PATCH 1/2] 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 c51fe84..475c619 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -355,12 +355,19 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, snd_pcm_jack_free(jack); 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 ( name == NULL ) { + 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); @@ -423,6 +430,10 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0 || strcmp(id, "hint") == 0) continue; + if (strcmp(id, "name") == 0) { + snd_config_get_string(n, &name); + continue; + } if (strcmp(id, "playback_ports") == 0) { if (snd_config_get_type(n) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for %s", id); -- 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: pcm: export some constraint functions
by Jean-Francois Moine 06 Feb '14

06 Feb '14

The functions snd_pcm_hw_constraint_mask() and snd_pcm_hw_constraint_mask64() are needed to add audio constraints to substreams at open/startup time. This patch makes them available to ALSA modules. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/core/pcm_lib.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a210467..aea4972 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1219,6 +1219,7 @@ int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param return -EINVAL; return 0; } +EXPORT_SYMBOL(snd_pcm_hw_constraint_mask); /** * snd_pcm_hw_constraint_mask64 - apply the given bitmap mask constraint @@ -1242,6 +1243,7 @@ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_par return -EINVAL; return 0; } +EXPORT_SYMBOL(snd_pcm_hw_constraint_mask64); /** * snd_pcm_hw_constraint_integer - apply an integer constraint to an interval -- 1.9.rc1

2 1

[alsa-devel] [PATCH v3 0/5] add a TDA998x CODEC
by Jean-Francois Moine 05 Feb '14

05 Feb '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 - v3 - move the DT description of the CODEC as a subnode of the TDA998x I2C (Mark Brown remark - is this the right way?) - change CODEC loading mechanism - simplify tda998x i2c reference search - CODEC DT documentation in drm/i2c/tda998x.txt - 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 the TDA998x ASoC: tda998x: add DT documentation of the tda998x CODEC ASoC: tda998x: adjust the audio hw parameters from EDID ASoC: tda998x: adjust the audio CTS_N pre-divider from audio format .../devicetree/bindings/drm/i2c/tda998x.txt | 17 ++ drivers/gpu/drm/i2c/tda998x_drv.c | 86 +++++- include/drm/i2c/tda998x.h | 11 +- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 303 +++++++++++++++++++++ 6 files changed, 419 insertions(+), 6 deletions(-) create mode 100644 sound/soc/codecs/tda998x.c -- 1.9.rc1

4 25

[alsa-devel] [PATCH] ASoC: cs42l73: Don't mix SNDRV_PCM_RATE_KNOT with specific rates
by Lars-Peter Clausen 05 Feb '14

05 Feb '14

SNDRV_PCM_RATE_KNOT means that the device can support rates that can not be expressed using the rate bits. The driver will provide a list of those rates specified through constraints. Any rate bits that are set in the rates mask will be ignored. So setting other rate bits besides SNDRV_PCM_RATE_KNOT wont have any effect, but might be confusing to the casual reader, so remove them. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/cs42l73.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 549d5d6..7cae046 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1255,9 +1255,6 @@ static int cs42l73_pcm_startup(struct snd_pcm_substream *substream, return 0; } -/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ -#define CS42L73_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) - #define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) @@ -1278,14 +1275,14 @@ static struct snd_soc_dai_driver cs42l73_dai[] = { .stream_name = "XSP Playback", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .capture = { .stream_name = "XSP Capture", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .ops = &cs42l73_ops, @@ -1298,14 +1295,14 @@ static struct snd_soc_dai_driver cs42l73_dai[] = { .stream_name = "ASP Playback", .channels_min = 2, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .capture = { .stream_name = "ASP Capture", .channels_min = 2, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .ops = &cs42l73_ops, @@ -1318,14 +1315,14 @@ static struct snd_soc_dai_driver cs42l73_dai[] = { .stream_name = "VSP Playback", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .capture = { .stream_name = "VSP Capture", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .ops = &cs42l73_ops, -- 1.8.0

5 10

[alsa-devel] [PATCH] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 05 Feb '14

05 Feb '14

This patch adds initial support for the Behringer BCD2000 USB DJ controller. At the moment, only the MIDI part of the device is working, i.e. knobs, buttons and LEDs. I also plan to add support for the audio part, but I assume that this will require more effort than the rather simple MIDI interface. Progress can be tracked at https://github.com/anyc/snd-usb-bcd2000. Signed-off-by: Mario Kicherer <dev(a)kicherer.org> --- sound/usb/Kconfig | 14 + sound/usb/Makefile | 2 +- sound/usb/bcd2000/Makefile | 3 + sound/usb/bcd2000/bcd2000.c | 629 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 647 insertions(+), 1 deletion(-) create mode 100644 sound/usb/bcd2000/Makefile create mode 100644 sound/usb/bcd2000/bcd2000.c diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index de9408b..50af405 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -146,5 +146,19 @@ config SND_USB_HIFACE To compile this driver as a module, choose M here: the module will be called snd-usb-hiface. +config SND_USB_BCD2000 + tristate "Behringer BCD2000 MIDI driver" + select SND_HWDEP + select SND_RAWMIDI + help + Say Y here to include MIDI support for the Behringer BCD2000 DJ + controller. + + Audio support is still work-in-progress at + https://github.com/anyc/snd-usb-bcd2000 + + To compile this driver as a module, choose M here: the module + will be called snd-usb-bcd2000. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index abe668f..2b92f0d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ bcd2000/ diff --git a/sound/usb/bcd2000/Makefile b/sound/usb/bcd2000/Makefile new file mode 100644 index 0000000..6540136 --- /dev/null +++ b/sound/usb/bcd2000/Makefile @@ -0,0 +1,3 @@ +snd-usb-bcd2000-y := bcd2000.o + +obj-$(CONFIG_SND_USB_BCD2000) += snd-usb-bcd2000.o \ No newline at end of file diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c new file mode 100644 index 0000000..100c39f --- /dev/null +++ b/sound/usb/bcd2000/bcd2000.c @@ -0,0 +1,629 @@ +/* + * Behringer BCD2000 driver + * + * Copyright (C) 2014 Mario Kicherer (dev(a)kicherer.org) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifdef CONFIG_USB_DEBUG +#define DEBUG 1 +#else +#define DEBUG 0 +#endif +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/usb.h> +#include <linux/usb/audio.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/rawmidi.h> +#include "../usbaudio.h" +#include "../midi.h" + +#define DEVICE_NAME "Behringer BCD2000" +#define DEVICE_SHORTNAME "bcd2000" + +#define PREFIX DEVICE_SHORTNAME ": " +#define BUFSIZE 64 + +static struct usb_device_id id_table [] = { + { USB_DEVICE(0x1397, 0x00bd) }, + { }, +}; +MODULE_DEVICE_TABLE (usb, id_table); +MODULE_AUTHOR("Mario Kicherer, dev(a)kicherer.org"); +MODULE_DESCRIPTION("Behringer BCD2000 driver"); +MODULE_LICENSE("GPL"); + +static int debug = 0; +module_param(debug, int, S_IRUGO | S_IWUSR); +MODULE_PARM_DESC(debug, "enable debug output (default: 0)"); + +static unsigned char set_led_prefix[] = {0x03, 0x00}; + +static unsigned char bcd2000_init_sequence[] = { + 0x07, 0x00, 0x00, 0x00, 0x78, 0x48, 0x1c, 0x81, + 0xc4, 0x00, 0x00, 0x00, 0x5e, 0x53, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff, 0x6c, 0xf3, 0x90, 0xff, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, + 0x18, 0xfa, 0x11, 0xff, 0x14, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0xf2, 0x34, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff}; + +struct bcd2000 { + struct usb_device *dev; + struct snd_card *card; + struct usb_interface *intf; + int card_index; + struct snd_pcm *pcm; + struct list_head midi_list; + spinlock_t lock; + struct mutex mutex; + + int midi_out_active; + struct snd_rawmidi *rmidi; + struct snd_rawmidi_substream *midi_receive_substream; + struct snd_rawmidi_substream *midi_out_substream; + + unsigned char midi_in_buf[BUFSIZE]; + unsigned char midi_out_buf[BUFSIZE]; + + unsigned char midi_cmd_buf[3]; + unsigned char midi_cmd_offset; + + struct urb * midi_out_urb; + struct urb * midi_in_urb; +}; + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +static DEFINE_MUTEX(devices_mutex); +static unsigned int devices_used; +static struct usb_driver bcd2000_driver; + + + + + + +static int bcd2000_midi_input_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_input_close(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static void bcd2000_midi_input_trigger(struct snd_rawmidi_substream *substream, int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (!bcd2k) + return; + + bcd2k->midi_receive_substream = up ? substream : NULL; +} + +static void bcd2000_midi_handle_input(struct bcd2000 *bcd2k, + const unsigned char *buf, int len) +{ + unsigned char length, cmd, buf_offset, tocopy; + int ret; + + if (!bcd2k->midi_receive_substream) + return; + + if (DEBUG || debug) { + printk(KERN_DEBUG PREFIX "received (%d bytes): ", len); + for (ret=0; ret < len; ret++) { + printk("%x ", buf[ret]); + } + printk("\n"); + } + + if (len < 2) + return; + + /* + * The BCD2000 sends MIDI commands with 2 or 3 bytes length. In case of + * 2 bytes, the command is the same as before. + * + * Packet structure: mm nn oo (pp) + * mm: payload length + * nn: MIDI command or note + * oo: note or velocity + * pp: velocity + */ + + length = buf[0]; + + /* ignore packets without payload, should not happen */ + if (length == 0) + return; + + /* + * The MIDI packets the BCD2000 sends can be arbitrarily truncated or + * concatenated. Therefore, this loop accumulates the bytes from the + * input buffer until a full valid MIDI packet is in the MIDI command + * buffer "midi_cmd_buf". + */ + + buf_offset = 1; + while (buf_offset <= length) { + cmd = buf[buf_offset]; + + if (bcd2k->midi_cmd_offset == 0 && cmd != 0x90 && cmd != 0xb0) { + /* + * this is a 2-byte midi packet -> reuse command byte from + * last midi packet + */ + bcd2k->midi_cmd_offset = 1; + } + + /* determine the number of bytes we want to copy this time */ + tocopy = min( 3 - bcd2k->midi_cmd_offset, length - (buf_offset - 1)); + + if (tocopy > 0) { + memcpy(&bcd2k->midi_cmd_buf[bcd2k->midi_cmd_offset], + &buf[buf_offset], tocopy); + } else + printk(KERN_ERR PREFIX "error parsing\n"); + + bcd2k->midi_cmd_offset += tocopy; + buf_offset += tocopy; + + /* is our MIDI packet complete? */ + if (bcd2k->midi_cmd_offset == 3) { + if (DEBUG || debug) { + printk(KERN_DEBUG PREFIX "send midi packet (%d bytes): ", + bcd2k->midi_cmd_offset); + for (ret=0;ret< bcd2k->midi_cmd_offset;ret++) { + printk("%x ", bcd2k->midi_cmd_buf[ret]); + } + printk("\n"); + } + + /* send MIDI packet */ + snd_rawmidi_receive(bcd2k->midi_receive_substream, + bcd2k->midi_cmd_buf, 3); + + bcd2k->midi_cmd_offset = 0; + } + } +} + +static int bcd2000_midi_output_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_output_close(struct snd_rawmidi_substream *substream) +{ + struct bcd2000 * bcd2k = substream->rmidi->private_data; + if (bcd2k->midi_out_active) { + usb_free_urb(bcd2k->midi_out_urb); + bcd2k->midi_out_active = 0; + } + return 0; +} + +static void bcd2000_midi_send(struct bcd2000 *bcd2k, + struct snd_rawmidi_substream *substream) +{ + int len, ret; + + /* copy the "set LED" command bytes */ + memcpy(bcd2k->midi_out_buf, set_led_prefix, sizeof(set_led_prefix)); + + /* get MIDI packet */ + len = snd_rawmidi_transmit(substream, bcd2k->midi_out_buf + 3, BUFSIZE - 3); + + if (len <= 0) + return; + + /* set payload length */ + bcd2k->midi_out_buf[2] = len; + bcd2k->midi_out_urb->transfer_buffer_length = BUFSIZE; + + if (DEBUG || debug) { + printk(KERN_DEBUG PREFIX "sending: "); + for (ret=0;ret< 3 + len;ret++) { + printk("%x ", bcd2k->midi_out_buf[ret]); + } + printk("\n"); + } + + /* send packet to the BCD2000 */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) { + printk(KERN_ERR PREFIX + "bcd2000_midi_send(%p): usb_submit_urb() failed," + "ret=%d, len=%d: ", substream, ret, len); + switch (ret) { + case -ENOMEM: printk("ENOMEM\n"); break; + case -ENODEV: printk("ENODEV\n"); break; + case -ENXIO: printk("ENXIO\n"); break; + case -EINVAL: printk("EINVAL\n"); break; + case -EAGAIN: printk("EAGAIN\n"); break; + case -EFBIG: printk("EFBIG\n"); break; + case -EPIPE: printk("EPIPE\n"); break; + case -EMSGSIZE: printk("EMSGSIZE\n"); break; + default: printk("\n"); + } + } else { + bcd2k->midi_out_active = 1; + } +} + +static void bcd2000_midi_output_done(struct urb* urb) +{ + struct bcd2000 * bcd2k = urb->context; + + bcd2k->midi_out_active = 0; + if (urb->status != 0) { + printk(KERN_ERR PREFIX "urb status: %d\n", urb->status); + return; + } + + if (!bcd2k->midi_out_substream) + return; + + bcd2000_midi_send(bcd2k, bcd2k->midi_out_substream); +} + +static void bcd2000_command_reply_dispatch(struct urb* urb) +{ + int ret; + struct device *dev = &urb->dev->dev; + struct bcd2000 *bcd2k = urb->context; + + if (urb->status || !bcd2k) { + dev_warn(dev, "received urb->status = %i\n", urb->status); + return; + } + + if (urb->actual_length > 0) + bcd2000_midi_handle_input(bcd2k, urb->transfer_buffer, urb->actual_length); + + bcd2k->midi_in_urb->actual_length = 0; + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(dev, "unable to submit urb. OOM!?\n"); +} + +static void bcd2000_midi_output_trigger(struct snd_rawmidi_substream *substream, int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (up) { + bcd2k->midi_out_substream = substream; + if (!bcd2k->midi_out_active) + bcd2000_midi_send(bcd2k, substream); + } else { + bcd2k->midi_out_substream = NULL; + } +} + +static struct snd_rawmidi_ops bcd2000_midi_output = +{ + .open = bcd2000_midi_output_open, + .close = bcd2000_midi_output_close, + .trigger = bcd2000_midi_output_trigger, +}; + +static struct snd_rawmidi_ops bcd2000_midi_input = +{ + .open = bcd2000_midi_input_open, + .close = bcd2000_midi_input_close, + .trigger = bcd2000_midi_input_trigger, +}; + +static void bcd2000_init_device(struct bcd2000 * bcd2k) { + int ret; + + /* copy init sequence into buffer */ + memcpy(bcd2k->midi_out_buf, bcd2000_init_sequence, 52); + bcd2k->midi_out_urb->transfer_buffer_length = 52; + + /* submit sequence */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) { + printk(KERN_ERR PREFIX + "bcd2000_init_device: usb_submit_urb() failed," + "ret=%d: ", ret); + switch (ret) { + case -ENOMEM: printk("ENOMEM\n"); break; + case -ENODEV: printk("ENODEV\n"); break; + case -ENXIO: printk("ENXIO\n"); break; + case -EINVAL: printk("EINVAL\n"); break; + case -EAGAIN: printk("EAGAIN\n"); break; + case -EFBIG: printk("EFBIG\n"); break; + case -EPIPE: printk("EPIPE\n"); break; + case -EMSGSIZE: printk("EMSGSIZE\n"); break; + default: printk("\n"); + } + } else { + bcd2k->midi_out_active = 1; + } + + /* send empty packet to enable button and controller events */ + bcd2k->midi_in_urb->actual_length = 0; + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); +} + +static void bcd2000_init_midi(struct bcd2000 * bcd2k) { + int ret; + struct snd_rawmidi *rmidi; + + ret = snd_rawmidi_new(bcd2k->card, bcd2k->card->shortname, 0, + 1, /* output */ + 1, /* input */ + &rmidi); + + if (ret < 0) + return; + + strlcpy(rmidi->name, bcd2k->card->shortname, sizeof(rmidi->name)); + + rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; + rmidi->private_data = bcd2k; + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &bcd2000_midi_output); + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &bcd2000_midi_input); + + bcd2k->rmidi = rmidi; + + bcd2k->midi_in_urb = usb_alloc_urb(0, GFP_KERNEL); + bcd2k->midi_out_urb = usb_alloc_urb(0, GFP_KERNEL); + + usb_fill_int_urb(bcd2k->midi_in_urb, bcd2k->dev, + usb_rcvbulkpipe(bcd2k->dev, 0x81), + bcd2k->midi_in_buf, BUFSIZE, + bcd2000_command_reply_dispatch, bcd2k, 1); + + usb_fill_int_urb(bcd2k->midi_out_urb, bcd2k->dev, + usb_sndbulkpipe(bcd2k->dev, 0x1), + bcd2k->midi_out_buf, BUFSIZE, + bcd2000_midi_output_done, bcd2k, 1); + + bcd2000_init_device(bcd2k); +} + +static void bcd2000_close_midi(struct bcd2000 * bcd2k) { + usb_free_urb(bcd2k->midi_out_urb); + usb_free_urb(bcd2k->midi_in_urb); +} + + + + + +/* + * general part + */ + +static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k, + struct usb_interface *interface) +{ + unsigned int i; + struct usb_interface *intf; + +// mutex_lock(&ua->mutex); +// free_stream_urbs(&ua->capture); +// free_stream_urbs(&ua->playback); +// mutex_unlock(&ua->mutex); +// free_stream_buffers(ua, &ua->capture); +// free_stream_buffers(ua, &ua->playback); + + for (i = 0; i < 1; ++i) { + mutex_lock(&bcd2k->mutex); + intf = bcd2k->intf; + bcd2k->intf = NULL; + mutex_unlock(&bcd2k->mutex); + if (intf) { + usb_set_intfdata(intf, NULL); + if (intf != interface) + usb_driver_release_interface(&bcd2000_driver, + intf); + } + } +} + +static void bcd2000_card_free(struct snd_card *card) +{ +// struct bcd2000 *bcd2k = card->private_data; +// mutex_destroy(&ua->mutex); +} + +static int bcd2000_probe(struct usb_interface *interface, const struct usb_device_id *usb_id) +{ + struct snd_card *card; + struct bcd2000 * bcd2k; + unsigned int card_index; + char usb_path[32]; + int err; + + + mutex_lock(&devices_mutex); + + for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) + if (enable[card_index] && !(devices_used & (1 << card_index))) + break; + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + err = snd_card_create(index[card_index], id[card_index], THIS_MODULE, + sizeof(*bcd2k), &card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return err; + } + card->private_free = bcd2000_card_free; + bcd2k = card->private_data; + bcd2k->dev = interface_to_usbdev(interface); + bcd2k->card = card; + bcd2k->card_index = card_index; + INIT_LIST_HEAD(&bcd2k->midi_list); + spin_lock_init(&bcd2k->lock); + mutex_init(&bcd2k->mutex); + +// INIT_LIST_HEAD(&bcd2k->ready_playback_urbs); +// tasklet_init(&bcd2k->playback_tasklet, +// playback_tasklet, (unsigned long)bcd2k); +// init_waitqueue_head(&bcd2k->alsa_capture_wait); +// init_waitqueue_head(&bcd2k->rate_feedback_wait); +// init_waitqueue_head(&bcd2k->alsa_playback_wait); + + bcd2k->intf = interface; + + snd_card_set_dev(card, &interface->dev); + + strcpy(card->driver, DEVICE_NAME); + strcpy(card->shortname, DEVICE_SHORTNAME); + usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); + snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), + DEVICE_NAME ", at %s, %s speed", + usb_path, + bcd2k->dev->speed == USB_SPEED_HIGH ? "high" : "full"); + + printk(KERN_INFO PREFIX "%s", bcd2k->card->longname); + +// err = alloc_stream_buffers(bcd2k, &bcd2k->capture); +// if (err < 0) +// goto probe_error; +// err = alloc_stream_buffers(bcd2k, &bcd2k->playback); +// if (err < 0) +// goto probe_error; +// +// err = alloc_stream_urbs(bcd2k, &bcd2k->capture, capture_urb_complete); +// if (err < 0) +// goto probe_error; +// err = alloc_stream_urbs(bcd2k, &bcd2k->playback, playback_urb_complete); +// if (err < 0) +// goto probe_error; +// +// err = snd_pcm_new(card, "bcd2000", 0, 1, 1, &bcd2k->pcm); +// if (err < 0) +// goto probe_error; +// bcd2k->pcm->private_data = bcd2k; +// strcpy(bcd2k->pcm->name, "bcd2000"); +// snd_pcm_set_ops(bcd2k->pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_pcm_ops); +// snd_pcm_set_ops(bcd2k->pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_pcm_ops); + +// err = snd_usbmidi_create(card, bcd2k->intf, +// &bcd2k->midi_list, &midi_quirk); +// if (err < 0) +// goto probe_error; + + + + bcd2000_init_midi(bcd2k); + + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + usb_set_intfdata(interface, bcd2k); + devices_used |= 1 << card_index; + + mutex_unlock(&devices_mutex); + return 0; + +probe_error: + bcd2000_free_usb_related_resources(bcd2k, interface); + snd_card_free(card); + mutex_unlock(&devices_mutex); + return err; +} + +static void bcd2000_disconnect(struct usb_interface *interface) +{ + struct bcd2000 *bcd2k = usb_get_intfdata(interface); + struct list_head *midi; + + if (!bcd2k) + return; + + bcd2000_close_midi(bcd2k); + + mutex_lock(&devices_mutex); + +// set_bit(DISCONNECTED, &bcd2k->states); +// wake_up(&bcd2k->rate_feedback_wait); + + /* make sure that userspace cannot create new requests */ + snd_card_disconnect(bcd2k->card); + + /* make sure that there are no pending USB requests */ + list_for_each(midi, &bcd2k->midi_list) + snd_usbmidi_disconnect(midi); + +// abort_alsa_playback(bcd2k); +// abort_alsa_capture(bcd2k); +// mutex_lock(&bcd2k->mutex); +// stop_usb_playback(bcd2k); +// stop_usb_capture(bcd2k); +// mutex_unlock(&bcd2k->mutex); + + bcd2000_free_usb_related_resources(bcd2k, interface); + + devices_used &= ~(1 << bcd2k->card_index); + + snd_card_free_when_closed(bcd2k->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_driver bcd2000_driver = { + .name = DEVICE_SHORTNAME, + .probe = bcd2000_probe, + .disconnect = bcd2000_disconnect, + .id_table = id_table, +}; + +static int __init bcd2000_init(void) +{ + int retval = 0; + + retval = usb_register(&bcd2000_driver); + if (retval) + printk(KERN_INFO "usb_register failed. Error number %d", retval); + return retval; +} + +static void __exit bcd2000_exit(void) +{ + usb_deregister(&bcd2000_driver); +} + +module_init (bcd2000_init); +module_exit (bcd2000_exit); -- 1.8.3.2

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