September 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] i.MX31L + AC97
by Geoffrey Wossum 10 Sep '09

10 Sep '09

Hi all, I'm currently looking at a board which uses a Freescale i.MX31L connected to a Wolfson WM9715L over the AC97 bus. The WM9715L will be used for both sound and a touchscreen. It doesn't look like there are any AC97 SoC drivers for the i.MX31. Is this correct? If so, should I try to get the board changed to use a codec on the I2S bus (the board is being layed out right now), or would creating the AC97 driver simple enough that we should continue on with this hardware and write the driver when needed? TIA, --- Geoffrey

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[alsa-devel] Alsa plugin unable to be loaded by Pulseaudio
by Ng Oon-Ee 10 Sep '09

10 Sep '09

Hello, please find below an issue I had with loading plugins into pulseaudio. The pulse dev has sent this mail on his list concerning alsa's interaction with pulse, indicating that there is something buggy with how it works. Details below. I'd be willing to post something to the bugtracker, if I only knew what the below means... -------- Forwarded Message -------- From: Lennart Poettering <lennart(a)poettering.net> Reply-to: General PulseAudio Discussion <pulseaudio-discuss(a)mail.0pointer.de> To: pulseaudio-discuss(a)mail.0pointer.de Subject: Re: [pulseaudio-discuss] Alsa output devices, able to use alsa plugin? Date: Fri, 4 Sep 2009 01:20:15 +0200 On Wed, 26.08.09 12:27, Tanu Kaskinen (tanuk(a)iki.fi) wrote: > D: alsa-util.c: Trying test with SND_PCM_NO_AUTO_FORMAT ... > D: alsa-util.c: Managed to open test > D: alsa-util.c: Maximum hw buffer size is 371 ms > I: alsa-util.c: snd_pcm_hw_params_set_periods_near() failed: Tiedostoa tai hakemistoa ei ole > I: alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Tiedostoa tai hakemistoa ei ole > D: alsa-util.c: Trying test without SND_PCM_NO_AUTO_FORMAT ... > D: alsa-util.c: Managed to open test > D: alsa-util.c: Maximum hw buffer size is 371 ms > I: alsa-util.c: snd_pcm_hw_params_set_periods_near() failed: Tiedostoa tai hakemistoa ei ole > I: alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Tiedostoa tai hakemistoa ei ole > D: alsa-util.c: Trying plug:test with SND_PCM_NO_AUTO_FORMAT ... > D: alsa-util.c: Managed to open plug:test > D: alsa-util.c: Maximum hw buffer size is 371 ms > I: alsa-util.c: snd_pcm_hw_params_set_periods_near() failed: Tiedostoa tai hakemistoa ei ole > I: alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Tiedostoa tai hakemistoa ei ole > D: alsa-util.c: Trying plug:test without SND_PCM_NO_AUTO_FORMAT ... > D: alsa-util.c: Managed to open plug:test > D: alsa-util.c: Maximum hw buffer size is 371 ms > I: alsa-util.c: snd_pcm_hw_params_set_periods_near() failed: Tiedostoa tai hakemistoa ei ole > I: alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Tiedostoa tai hakemistoa ei ole > I: alsa-util.c: Failed to set hardware parameters on plug:test: Tiedostoa tai hakemistoa ei ole > E: module.c: Failed to load module "module-alsa-sink" (argument: "device=test"): initialization failed. > > That "Tiedostoa tai hakemistoa ei ole" error means "No such file or > directory". I tried also with device=hw:0,0 and device=plughw:0,0 and > the first worked while the latter gave the same errors. Since the failed > function calls pertain to buffering, I also tried with tsched=0. It > worked. So, maybe this is a bug in the "plug" alsa plugin, ie. it > doesn't implement the *_near() functions properly. > > Lennart, any insights? I am pretty sure that ALSA is broken here. We nowadays first try to set the number of periods, and then set the buffer size in the hwparams struct. If this fails we print a warning message (the ones you see above with all the vowels regarding snd_pcm_hw_params_set_periods_near() resp. snd_pcm_hw_params_set_buffer_size_near()), but we do not consider this fatal. After doing this we call snd_pcm_hw_params() to activate our settings -- and if this fails then we consider this a fatal problem. Now the brokeness in ALSA is that after refusing our parameters it still fails to configure our settings, i.e. calling the two functions that fail breaks the hwparams setup: if we wouldn't call them, no breakage happens at all. The two functions in question should either succeed or fail, but if they fail they should not modify hwparams in a way that it subsequently becomes unusable for snd_pcm_hw_params(). This issue needs to be fixed in ALSA. Lennart

3 9

[alsa-devel] Problems with safe API and snd-cs46xx
by Sophie Hamilton 10 Sep '09

10 Sep '09

Hi there, I was pointed here by Audacious developers, who believe I've found a bug in the snd-cs46xx driver, which I use for my sound card (a Turtle Beach Santa Cruz, whose PCI ID is 1013:6003). Audacious, as of version 2.1, uses the default period time offered by ALSA and limits itself to the "safe API" as per Lennart Poettering. This already uncovered a bug in the Aureal Vortex driver, and I believe, based on the information that the devs gave me, that I have just uncovered another. (I'm also having problems with recent versions of OpenAL. However, I'll explain this further below.) I first started having the problem after having upgraded to Audacious 2.1, from Audacious 1.5.1. (to be more precise, the Gentoo ebuild 1.5.1-r1.) At the time, I was using the Gentoo-patched 2.6.29-series kernel, which I use as my normal kernel. After upgrading, I could no longer play audio directly using ALSA; instead, Audacious would initialise to 00:00, not play anything, and report the following to the owning console every so often: > ERROR: ALSA: alsa-core.c:226 (alsaplug_write_buffer): (write) snd_pcm_recover: Input/output error The thread which was doing this also seemed to be blocking, as when I tried to do anything else in Audacious, the interface would then freeze until the thread was done. OSS output seems to be okay, although as I don't have OSS compiled at all into the kernel (not even as a module), this would be going via ALSA's own OSS emulation. In addition, ALSA output to another audio device from Audacious works fine. I Googled for this and came across another snd-cs46xx user having the exact same problem in bug AUD-53 in the JIRA implementation they were using: http://jira.atheme.org/browse/AUD-53 . Since that user had been asked to upgrade to tip but hadn't, I did so instead (uninstalling the Gentoo ebuilds frst, of course) and reported back the additional debug info: > DEBUG: ALSA: alsa-core.c:226 (alsaplug_write_buffer): snd_pcm_writei error: Input/output error > ERROR: ALSA: alsa-core.c:230 (alsaplug_write_buffer): snd_pcm_recover error: Input/output error I then didn't know how to continue as I wasn't sure whether the bug would be seen as it had previously been closed as unreproducible, so I hopped onto the IRC channel and asked about what should be done. It was there that the probability of it being a driver bug in the snd-cs46xx sound driver was mentioned, and the reasoning - that the use of ALSA's safe API has been exposing driver bugs that were previously hidden. As mentioned previously, I'm also having issues with OpenAL. The version I'm using now is 1.7.411; previously I was using 1.5.304, which I believe worked fine. (I can't easily test this now as the Gentoo ebuild for it has gone, however.) The symptoms I got were similar - I got no sound, and messages were printed to the console every so often (more often than with Audacious, however): > AL lib: alsa.c:194: Wait timeout... buffer size too low? At first I thought this was an issue with Second Life, but on seeing the exact same symptoms using mplayer with the '-ao openal' switch, and also seeing it occur with 'openal-info', it seems to be an OpenAL thing. (Since I don't have many apps that I use with OpenAL, I hadn't known this before.) It seems that this is a symptom of whatever driver bug is causing the Audacious error. For the purposes of making this post, I decided to compile and install a vanilla 2.6.30.5 kernel to boot into, so that I could make sure there were no problems with the latest stable build. (I'd rather not run unstable versions if I can help it, or I would have done so; this is my primary machine. However, I'm willing to apply patches for this specific problem in order to test them.) The results of this were that I no longer get the messages and the threads don't block, but I get entirely new problems now - instead of useable sound, I get what sounds a little like noise, but more crackly, and the seek bar in Audacious seems to be going along as fast as it can - for each wallclock second, the seek bar goes up by about 114 seconds. The same occurs when I use mplayer with the '-ao openal' switch; I get the noise/crackles, and the time position shown on the console zooms by. The description of my sound card given by 'lspci -vv' (under both kernels) is thus: > 05:04.0 Multimedia audio controller: Cirrus Logic CS 4614/22/24/30 [CrystalClear SoundFusion Audio Accelerator] (rev 01) > Subsystem: Voyetra Technologies Santa Cruz > Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- > Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- > Latency: 64 (1000ns min, 6000ns max) > Interrupt: pin A routed to IRQ 22 > Region 0: Memory at fdafe000 (32-bit, non-prefetchable) [size=4K] > Region 1: Memory at fd900000 (32-bit, non-prefetchable) [size=1M] > Capabilities: [40] Power Management version 2 > Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) > Status: D0 PME-Enable- DSel=0 DScale=0 PME- > Kernel driver in use: Sound Fusion CS46xx > Kernel modules: snd-cs46xx Most applications that I have seem to work properly with regards to audio, although I've heard that this is probably because ALSA's safe API isn't very well documented and as such most applications will not be using it, thus not exposing the bug in snd-cs46xx. No information is output to dmesg when this happens, on either the Gentoo-patched 2.6.29 kernel, or the vanilla 2.6.30.5 kernel. If you need any further information, please let me know! As stated above, I'm willing to apply patches to the vanilla sources of the latest stable kernel in order to test them. I don't have experience of downloading the ALSA drivers and compiling them separately but I could also do that if it's needed. Thank you, - Sophie.

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Re: [alsa-devel] [PATCH] To avoid the divide by zero error during the first execution, initialize the data type.
by Troy Kisky 10 Sep '09

10 Sep '09

avm(a)ti.com wrote: > From: Arun Mani <a0270733(a)gtcx26221.gt.design.ti.com> > > Signed-off-by: Arun Mani <a0270733(a)gtcx26221.gt.design.ti.com> > --- > sound/soc/davinci/davinci-i2s.c | 2 ++ > 1 files changed, 2 insertions(+), 0 deletions(-) > > diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c > index b1ea52f..61b1763 100644 > --- a/sound/soc/davinci/davinci-i2s.c > +++ b/sound/soc/davinci/davinci-i2s.c > @@ -104,10 +104,12 @@ enum { > > static struct davinci_pcm_dma_params davinci_i2s_pcm_out = { > .name = "I2S PCM Stereo out", > + .data_type = 2, //Initialize the data type for playback to avoid divide by zero You should have ".acnt = 2" also for consistency. > }; > > static struct davinci_pcm_dma_params davinci_i2s_pcm_in = { > .name = "I2S PCM Stereo in", > + .data_type = 2, //Initialize the data type for playback to avoid divide by zero > }; > > struct davinci_mcbsp_dev { And if the 1st stream is an 8-bit audio stream, will it initialize it incorrectly ??? Of course, we have formats = SNDRV_PCM_FMTBIT_S16_LE, currently so it's not an issue yet... I don't see how data_type is not being set in davinci_i2s_hw_params before being used in davinci_pcm_prepare. Can prepare be called before hw_params ? There is a "return -EINVAL" in davinci_i2s_hw_params with a "printk(KERN_WARNING "davinci-i2s: unsupported PCM format\n")" before it. Do you see this message in your log ? Thanks Troy

2 2

[alsa-devel] [PATCH v3] ASoC: AK4671: add ak4671 codec driver
by Joonyoung Shim 10 Sep '09

10 Sep '09

The AK4671 is a stereo CODEC with a built-in Microphone-Amplifier, Receiver-Amplifier and Headphone-Amplifier. The datasheet for the ak4671 can find at the following url: http://www.asahi-kasei.co.jp/akm/en/product/ak4671/ak4671_f01e.pdf Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- Change log: v2: - use soc-cache - remove unused pll_on variable - add event handler for DAC and ADC - remove unnecessary comments and PM functions - modify error handling in ak4671_register function v3: - use SND_SOC_DAPM_SUPPLY instead of SND_SOC_DAPM_DAC_E and SND_SOC_DAPM_ADC_E sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4671.c | 825 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4671.h | 156 +++++++++ 4 files changed, 987 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ak4671.c create mode 100644 sound/soc/codecs/ak4671.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0edca93..a2bb659 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C + select SND_SOC_AK4671 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -96,6 +97,9 @@ config SND_SOC_AK4535 config SND_SOC_AK4642 tristate +config SND_SOC_AK4671 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fb4af28..13f7b4f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -6,6 +6,7 @@ snd-soc-ad73311-objs := ad73311.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o +snd-soc-ak4671-objs := ak4671.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -56,6 +57,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o +obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c new file mode 100644 index 0000000..b61214d --- /dev/null +++ b/sound/soc/codecs/ak4671.c @@ -0,0 +1,825 @@ +/* + * ak4671.c -- audio driver for AK4671 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim(a)samsung.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/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/delay.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "ak4671.h" + +static struct snd_soc_codec *ak4671_codec; + +/* codec private data */ +struct ak4671_priv { + struct snd_soc_codec codec; + u8 reg_cache[AK4671_CACHEREGNUM]; +}; + +/* ak4671 register cache & default register settings */ +static const u8 ak4671_reg[AK4671_CACHEREGNUM] = { + 0x00, /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) */ + 0xf6, /* AK4671_PLL_MODE_SELECT0 (0x01) */ + 0x00, /* AK4671_PLL_MODE_SELECT1 (0x02) */ + 0x02, /* AK4671_FORMAT_SELECT (0x03) */ + 0x00, /* AK4671_MIC_SIGNAL_SELECT (0x04) */ + 0x55, /* AK4671_MIC_AMP_GAIN (0x05) */ + 0x00, /* AK4671_MIXING_POWER_MANAGEMENT0 (0x06) */ + 0x00, /* AK4671_MIXING_POWER_MANAGEMENT1 (0x07) */ + 0xb5, /* AK4671_OUTPUT_VOLUME_CONTROL (0x08) */ + 0x00, /* AK4671_LOUT1_SIGNAL_SELECT (0x09) */ + 0x00, /* AK4671_ROUT1_SIGNAL_SELECT (0x0a) */ + 0x00, /* AK4671_LOUT2_SIGNAL_SELECT (0x0b) */ + 0x00, /* AK4671_ROUT2_SIGNAL_SELECT (0x0c) */ + 0x00, /* AK4671_LOUT3_SIGNAL_SELECT (0x0d) */ + 0x00, /* AK4671_ROUT3_SIGNAL_SELECT (0x0e) */ + 0x00, /* AK4671_LOUT1_POWER_MANAGERMENT (0x0f) */ + 0x00, /* AK4671_LOUT2_POWER_MANAGERMENT (0x10) */ + 0x80, /* AK4671_LOUT3_POWER_MANAGERMENT (0x11) */ + 0x91, /* AK4671_LCH_INPUT_VOLUME_CONTROL (0x12) */ + 0x91, /* AK4671_RCH_INPUT_VOLUME_CONTROL (0x13) */ + 0xe1, /* AK4671_ALC_REFERENCE_SELECT (0x14) */ + 0x00, /* AK4671_DIGITAL_MIXING_CONTROL (0x15) */ + 0x00, /* AK4671_ALC_TIMER_SELECT (0x16) */ + 0x00, /* AK4671_ALC_MODE_CONTROL (0x17) */ + 0x02, /* AK4671_MODE_CONTROL1 (0x18) */ + 0x01, /* AK4671_MODE_CONTROL2 (0x19) */ + 0x18, /* AK4671_LCH_OUTPUT_VOLUME_CONTROL (0x1a) */ + 0x18, /* AK4671_RCH_OUTPUT_VOLUME_CONTROL (0x1b) */ + 0x00, /* AK4671_SIDETONE_A_CONTROL (0x1c) */ + 0x02, /* AK4671_DIGITAL_FILTER_SELECT (0x1d) */ + 0x00, /* AK4671_FIL3_COEFFICIENT0 (0x1e) */ + 0x00, /* AK4671_FIL3_COEFFICIENT1 (0x1f) */ + 0x00, /* AK4671_FIL3_COEFFICIENT2 (0x20) */ + 0x00, /* AK4671_FIL3_COEFFICIENT3 (0x21) */ + 0x00, /* AK4671_EQ_COEFFICIENT0 (0x22) */ + 0x00, /* AK4671_EQ_COEFFICIENT1 (0x23) */ + 0x00, /* AK4671_EQ_COEFFICIENT2 (0x24) */ + 0x00, /* AK4671_EQ_COEFFICIENT3 (0x25) */ + 0x00, /* AK4671_EQ_COEFFICIENT4 (0x26) */ + 0x00, /* AK4671_EQ_COEFFICIENT5 (0x27) */ + 0xa9, /* AK4671_FIL1_COEFFICIENT0 (0x28) */ + 0x1f, /* AK4671_FIL1_COEFFICIENT1 (0x29) */ + 0xad, /* AK4671_FIL1_COEFFICIENT2 (0x2a) */ + 0x20, /* AK4671_FIL1_COEFFICIENT3 (0x2b) */ + 0x00, /* AK4671_FIL2_COEFFICIENT0 (0x2c) */ + 0x00, /* AK4671_FIL2_COEFFICIENT1 (0x2d) */ + 0x00, /* AK4671_FIL2_COEFFICIENT2 (0x2e) */ + 0x00, /* AK4671_FIL2_COEFFICIENT3 (0x2f) */ + 0x00, /* AK4671_DIGITAL_FILTER_SELECT2 (0x30) */ + 0x00, /* this register not used */ + 0x00, /* AK4671_E1_COEFFICIENT0 (0x32) */ + 0x00, /* AK4671_E1_COEFFICIENT1 (0x33) */ + 0x00, /* AK4671_E1_COEFFICIENT2 (0x34) */ + 0x00, /* AK4671_E1_COEFFICIENT3 (0x35) */ + 0x00, /* AK4671_E1_COEFFICIENT4 (0x36) */ + 0x00, /* AK4671_E1_COEFFICIENT5 (0x37) */ + 0x00, /* AK4671_E2_COEFFICIENT0 (0x38) */ + 0x00, /* AK4671_E2_COEFFICIENT1 (0x39) */ + 0x00, /* AK4671_E2_COEFFICIENT2 (0x3a) */ + 0x00, /* AK4671_E2_COEFFICIENT3 (0x3b) */ + 0x00, /* AK4671_E2_COEFFICIENT4 (0x3c) */ + 0x00, /* AK4671_E2_COEFFICIENT5 (0x3d) */ + 0x00, /* AK4671_E3_COEFFICIENT0 (0x3e) */ + 0x00, /* AK4671_E3_COEFFICIENT1 (0x3f) */ + 0x00, /* AK4671_E3_COEFFICIENT2 (0x40) */ + 0x00, /* AK4671_E3_COEFFICIENT3 (0x41) */ + 0x00, /* AK4671_E3_COEFFICIENT4 (0x42) */ + 0x00, /* AK4671_E3_COEFFICIENT5 (0x43) */ + 0x00, /* AK4671_E4_COEFFICIENT0 (0x44) */ + 0x00, /* AK4671_E4_COEFFICIENT1 (0x45) */ + 0x00, /* AK4671_E4_COEFFICIENT2 (0x46) */ + 0x00, /* AK4671_E4_COEFFICIENT3 (0x47) */ + 0x00, /* AK4671_E4_COEFFICIENT4 (0x48) */ + 0x00, /* AK4671_E4_COEFFICIENT5 (0x49) */ + 0x00, /* AK4671_E5_COEFFICIENT0 (0x4a) */ + 0x00, /* AK4671_E5_COEFFICIENT1 (0x4b) */ + 0x00, /* AK4671_E5_COEFFICIENT2 (0x4c) */ + 0x00, /* AK4671_E5_COEFFICIENT3 (0x4d) */ + 0x00, /* AK4671_E5_COEFFICIENT4 (0x4e) */ + 0x00, /* AK4671_E5_COEFFICIENT5 (0x4f) */ + 0x88, /* AK4671_EQ_CONTROL_250HZ_100HZ (0x50) */ + 0x88, /* AK4671_EQ_CONTROL_3500HZ_1KHZ (0x51) */ + 0x08, /* AK4671_EQ_CONTRO_10KHZ (0x52) */ + 0x00, /* AK4671_PCM_IF_CONTROL0 (0x53) */ + 0x00, /* AK4671_PCM_IF_CONTROL1 (0x54) */ + 0x00, /* AK4671_PCM_IF_CONTROL2 (0x55) */ + 0x18, /* AK4671_DIGITAL_VOLUME_B_CONTROL (0x56) */ + 0x18, /* AK4671_DIGITAL_VOLUME_C_CONTROL (0x57) */ + 0x00, /* AK4671_SIDETONE_VOLUME_CONTROL (0x58) */ + 0x00, /* AK4671_DIGITAL_MIXING_CONTROL2 (0x59) */ + 0x00, /* AK4671_SAR_ADC_CONTROL (0x5a) */ +}; + +/* + * LOUT1/ROUT1 output volume control: + * from -24 to 6 dB in 6 dB steps (mute instead of -30 dB) + */ +static DECLARE_TLV_DB_SCALE(out1_tlv, -3000, 600, 1); + +/* + * LOUT2/ROUT2 output volume control: + * from -33 to 6 dB in 3 dB steps (mute instead of -33 dB) + */ +static DECLARE_TLV_DB_SCALE(out2_tlv, -3300, 300, 1); + +/* + * LOUT3/ROUT3 output volume control: + * from -6 to 3 dB in 3 dB steps + */ +static DECLARE_TLV_DB_SCALE(out3_tlv, -600, 300, 0); + +/* + * Mic amp gain control: + * from -15 to 30 dB in 3 dB steps + * REVISIT: The actual min value(0x01) is -12 dB and the reg value 0x00 is not + * available + */ +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -1500, 300, 0); + +static const struct snd_kcontrol_new ak4671_snd_controls[] = { + /* Common playback gain controls */ + SOC_SINGLE_TLV("Line Output1 Playback Volume", + AK4671_OUTPUT_VOLUME_CONTROL, 0, 0x6, 0, out1_tlv), + SOC_SINGLE_TLV("Headphone Output2 Playback Volume", + AK4671_OUTPUT_VOLUME_CONTROL, 4, 0xd, 0, out2_tlv), + SOC_SINGLE_TLV("Line Output3 Playback Volume", + AK4671_LOUT3_POWER_MANAGERMENT, 6, 0x3, 0, out3_tlv), + + /* Common capture gain controls */ + SOC_DOUBLE_TLV("Mic Amp Capture Volume", + AK4671_MIC_AMP_GAIN, 0, 4, 0xf, 0, mic_amp_tlv), +}; + +/* event handlers */ +static int ak4671_out2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u8 reg; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); + reg |= AK4671_MUTEN; + snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + break; + case SND_SOC_DAPM_PRE_PMD: + reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); + reg &= ~AK4671_MUTEN; + snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + break; + } + + return 0; +} + +/* Output Mixers */ +static const struct snd_kcontrol_new ak4671_lout1_mixer_controls[] = { + SOC_DAPM_SINGLE("DACL", AK4671_LOUT1_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINL1", AK4671_LOUT1_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINL2", AK4671_LOUT1_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINL3", AK4671_LOUT1_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINL4", AK4671_LOUT1_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPL", AK4671_LOUT1_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout1_mixer_controls[] = { + SOC_DAPM_SINGLE("DACR", AK4671_ROUT1_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINR1", AK4671_ROUT1_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINR2", AK4671_ROUT1_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINR3", AK4671_ROUT1_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINR4", AK4671_ROUT1_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPR", AK4671_ROUT1_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_lout2_mixer_controls[] = { + SOC_DAPM_SINGLE("DACHL", AK4671_LOUT2_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINH1", AK4671_LOUT2_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINH2", AK4671_LOUT2_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINH3", AK4671_LOUT2_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINH4", AK4671_LOUT2_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPHL", AK4671_LOUT2_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout2_mixer_controls[] = { + SOC_DAPM_SINGLE("DACHR", AK4671_ROUT2_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINH1", AK4671_ROUT2_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINH2", AK4671_ROUT2_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINH3", AK4671_ROUT2_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINH4", AK4671_ROUT2_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPHR", AK4671_ROUT2_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_lout3_mixer_controls[] = { + SOC_DAPM_SINGLE("DACSL", AK4671_LOUT3_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINS1", AK4671_LOUT3_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINS2", AK4671_LOUT3_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINS3", AK4671_LOUT3_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINS4", AK4671_LOUT3_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPSL", AK4671_LOUT3_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout3_mixer_controls[] = { + SOC_DAPM_SINGLE("DACSR", AK4671_ROUT3_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINS1", AK4671_ROUT3_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINS2", AK4671_ROUT3_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINS3", AK4671_ROUT3_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINS4", AK4671_ROUT3_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPSR", AK4671_ROUT3_SIGNAL_SELECT, 5, 1, 0), +}; + +/* Input MUXs */ +static const char *ak4671_lin_mux_texts[] = + {"LIN1", "LIN2", "LIN3", "LIN4"}; +static const struct soc_enum ak4671_lin_mux_enum = + SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 0, + ARRAY_SIZE(ak4671_lin_mux_texts), + ak4671_lin_mux_texts); +static const struct snd_kcontrol_new ak4671_lin_mux_control = + SOC_DAPM_ENUM("Route", ak4671_lin_mux_enum); + +static const char *ak4671_rin_mux_texts[] = + {"RIN1", "RIN2", "RIN3", "RIN4"}; +static const struct soc_enum ak4671_rin_mux_enum = + SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 2, + ARRAY_SIZE(ak4671_rin_mux_texts), + ak4671_rin_mux_texts); +static const struct snd_kcontrol_new ak4671_rin_mux_control = + SOC_DAPM_ENUM("Route", ak4671_rin_mux_enum); + +static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("LIN1"), + SND_SOC_DAPM_INPUT("RIN1"), + SND_SOC_DAPM_INPUT("LIN2"), + SND_SOC_DAPM_INPUT("RIN2"), + SND_SOC_DAPM_INPUT("LIN3"), + SND_SOC_DAPM_INPUT("RIN3"), + SND_SOC_DAPM_INPUT("LIN4"), + SND_SOC_DAPM_INPUT("RIN4"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + SND_SOC_DAPM_OUTPUT("LOUT3"), + SND_SOC_DAPM_OUTPUT("ROUT3"), + + /* DAC */ + SND_SOC_DAPM_DAC("DAC Left", "Left HiFi Playback", + AK4671_AD_DA_POWER_MANAGEMENT, 6, 0), + SND_SOC_DAPM_DAC("DAC Right", "Right HiFi Playback", + AK4671_AD_DA_POWER_MANAGEMENT, 7, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC Left", "Left HiFi Capture", + AK4671_AD_DA_POWER_MANAGEMENT, 4, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right HiFi Capture", + AK4671_AD_DA_POWER_MANAGEMENT, 5, 0), + + /* PGA */ + SND_SOC_DAPM_PGA("LOUT2 Mix Amp", + AK4671_LOUT2_POWER_MANAGERMENT, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("ROUT2 Mix Amp", + AK4671_LOUT2_POWER_MANAGERMENT, 6, 0, NULL, 0), + + SND_SOC_DAPM_PGA("LIN1 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN1 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN2 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN2 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN3 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN3 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN4 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN4 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 7, 0, NULL, 0), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("LOUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 0, 0, + &ak4671_lout1_mixer_controls[0], + ARRAY_SIZE(ak4671_lout1_mixer_controls)), + SND_SOC_DAPM_MIXER("ROUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 1, 0, + &ak4671_rout1_mixer_controls[0], + ARRAY_SIZE(ak4671_rout1_mixer_controls)), + SND_SOC_DAPM_MIXER_E("LOUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT, + 0, 0, &ak4671_lout2_mixer_controls[0], + ARRAY_SIZE(ak4671_lout2_mixer_controls), + ak4671_out2_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MIXER_E("ROUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT, + 1, 0, &ak4671_rout2_mixer_controls[0], + ARRAY_SIZE(ak4671_rout2_mixer_controls), + ak4671_out2_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MIXER("LOUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 0, 0, + &ak4671_lout3_mixer_controls[0], + ARRAY_SIZE(ak4671_lout3_mixer_controls)), + SND_SOC_DAPM_MIXER("ROUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 1, 0, + &ak4671_rout3_mixer_controls[0], + ARRAY_SIZE(ak4671_rout3_mixer_controls)), + + /* Input MUXs */ + SND_SOC_DAPM_MUX("LIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 2, 0, + &ak4671_lin_mux_control), + SND_SOC_DAPM_MUX("RIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 3, 0, + &ak4671_rin_mux_control), + + /* Mic Power */ + SND_SOC_DAPM_MICBIAS("Mic Bias", AK4671_AD_DA_POWER_MANAGEMENT, 1, 0), + + /* Supply */ + SND_SOC_DAPM_SUPPLY("PMPLL", AK4671_PLL_MODE_SELECT1, 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"DAC Left", "NULL", "PMPLL"}, + {"DAC Right", "NULL", "PMPLL"}, + {"ADC Left", "NULL", "PMPLL"}, + {"ADC Right", "NULL", "PMPLL"}, + + /* Outputs */ + {"LOUT1", "NULL", "LOUT1 Mixer"}, + {"ROUT1", "NULL", "ROUT1 Mixer"}, + {"LOUT2", "NULL", "LOUT2 Mix Amp"}, + {"ROUT2", "NULL", "ROUT2 Mix Amp"}, + {"LOUT3", "NULL", "LOUT3 Mixer"}, + {"ROUT3", "NULL", "ROUT3 Mixer"}, + + {"LOUT1 Mixer", "DACL", "DAC Left"}, + {"ROUT1 Mixer", "DACR", "DAC Right"}, + {"LOUT2 Mixer", "DACHL", "DAC Left"}, + {"ROUT2 Mixer", "DACHR", "DAC Right"}, + {"LOUT2 Mix Amp", "NULL", "LOUT2 Mixer"}, + {"ROUT2 Mix Amp", "NULL", "ROUT2 Mixer"}, + {"LOUT3 Mixer", "DACSL", "DAC Left"}, + {"ROUT3 Mixer", "DACSR", "DAC Right"}, + + /* Inputs */ + {"LIN MUX", "LIN1", "LIN1"}, + {"LIN MUX", "LIN2", "LIN2"}, + {"LIN MUX", "LIN3", "LIN3"}, + {"LIN MUX", "LIN4", "LIN4"}, + + {"RIN MUX", "RIN1", "RIN1"}, + {"RIN MUX", "RIN2", "RIN2"}, + {"RIN MUX", "RIN3", "RIN3"}, + {"RIN MUX", "RIN4", "RIN4"}, + + {"LIN1", NULL, "Mic Bias"}, + {"RIN1", NULL, "Mic Bias"}, + {"LIN2", NULL, "Mic Bias"}, + {"RIN2", NULL, "Mic Bias"}, + + {"ADC Left", "NULL", "LIN MUX"}, + {"ADC Right", "NULL", "RIN MUX"}, + + /* Analog Loops */ + {"LIN1 Mixing Circuit", "NULL", "LIN1"}, + {"RIN1 Mixing Circuit", "NULL", "RIN1"}, + {"LIN2 Mixing Circuit", "NULL", "LIN2"}, + {"RIN2 Mixing Circuit", "NULL", "RIN2"}, + {"LIN3 Mixing Circuit", "NULL", "LIN3"}, + {"RIN3 Mixing Circuit", "NULL", "RIN3"}, + {"LIN4 Mixing Circuit", "NULL", "LIN4"}, + {"RIN4 Mixing Circuit", "NULL", "RIN4"}, + + {"LOUT1 Mixer", "LINL1", "LIN1 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR1", "RIN1 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH1", "LIN1 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH1", "RIN1 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS1", "LIN1 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS1", "RIN1 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL2", "LIN2 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR2", "RIN2 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH2", "LIN2 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH2", "RIN2 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS2", "LIN2 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS2", "RIN2 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL3", "LIN3 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR3", "RIN3 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH3", "LIN3 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH3", "RIN3 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS3", "LIN3 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS3", "RIN3 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL4", "LIN4 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR4", "RIN4 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH4", "LIN4 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH4", "RIN4 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS4", "LIN4 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS4", "RIN4 Mixing Circuit"}, +}; + +static int ak4671_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, ak4671_dapm_widgets, + ARRAY_SIZE(ak4671_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int ak4671_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; + u8 fs; + + fs = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0); + fs &= ~AK4671_FS; + + switch (params_rate(params)) { + case 8000: + fs |= AK4671_FS_8KHZ; + break; + case 12000: + fs |= AK4671_FS_12KHZ; + break; + case 16000: + fs |= AK4671_FS_16KHZ; + break; + case 24000: + fs |= AK4671_FS_24KHZ; + break; + case 11025: + fs |= AK4671_FS_11_025KHZ; + break; + case 22050: + fs |= AK4671_FS_22_05KHZ; + break; + case 32000: + fs |= AK4671_FS_32KHZ; + break; + case 44100: + fs |= AK4671_FS_44_1KHZ; + break; + case 48000: + fs |= AK4671_FS_48KHZ; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, fs); + + return 0; +} + +static int ak4671_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + u8 pll; + + pll = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0); + pll &= ~AK4671_PLL; + + switch (freq) { + case 11289600: + pll |= AK4671_PLL_11_2896MHZ; + break; + case 12000000: + pll |= AK4671_PLL_12MHZ; + break; + case 12288000: + pll |= AK4671_PLL_12_288MHZ; + break; + case 13000000: + pll |= AK4671_PLL_13MHZ; + break; + case 13500000: + pll |= AK4671_PLL_13_5MHZ; + break; + case 19200000: + pll |= AK4671_PLL_19_2MHZ; + break; + case 24000000: + pll |= AK4671_PLL_24MHZ; + break; + case 26000000: + pll |= AK4671_PLL_26MHZ; + break; + case 27000000: + pll |= AK4671_PLL_27MHZ; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, pll); + + return 0; +} + +static int ak4671_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mode; + u8 format; + + /* set master/slave audio interface */ + mode = snd_soc_read(codec, AK4671_PLL_MODE_SELECT1); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mode |= AK4671_M_S; + break; + case SND_SOC_DAIFMT_CBM_CFS: + mode &= ~(AK4671_M_S); + break; + default: + return -EINVAL; + } + + /* interface format */ + format = snd_soc_read(codec, AK4671_FORMAT_SELECT); + format &= ~AK4671_DIF; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + format |= AK4671_DIF_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + format |= AK4671_DIF_MSB_MODE; + break; + case SND_SOC_DAIFMT_DSP_A: + format |= AK4671_DIF_DSP_MODE; + format |= AK4671_BCKP; + format |= AK4671_MSBS; + break; + default: + return -EINVAL; + } + + /* set mode and format */ + snd_soc_write(codec, AK4671_PLL_MODE_SELECT1, mode); + snd_soc_write(codec, AK4671_FORMAT_SELECT, format); + + return 0; +} + +static int ak4671_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u8 reg; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + reg = snd_soc_read(codec, AK4671_AD_DA_POWER_MANAGEMENT); + snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, + reg | AK4671_PMVCM); + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00); + break; + } + codec->bias_level = level; + return 0; +} + +#define AK4671_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) + +#define AK4671_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +static struct snd_soc_dai_ops ak4671_dai_ops = { + .hw_params = ak4671_hw_params, + .set_sysclk = ak4671_set_dai_sysclk, + .set_fmt = ak4671_set_dai_fmt, +}; + +struct snd_soc_dai ak4671_dai = { + .name = "AK4671", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AK4671_RATES, + .formats = AK4671_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AK4671_RATES, + .formats = AK4671_FORMATS,}, + .ops = &ak4671_dai_ops, +}; +EXPORT_SYMBOL_GPL(ak4671_dai); + +static int ak4671_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (ak4671_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ak4671_codec; + codec = ak4671_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, ak4671_snd_controls, + ARRAY_SIZE(ak4671_snd_controls)); + ak4671_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +static int ak4671_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak4671 = { + .probe = ak4671_probe, + .remove = ak4671_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4671); + +static int ak4671_register(struct ak4671_priv *ak4671, + enum snd_soc_control_type control) +{ + int ret; + struct snd_soc_codec *codec = &ak4671->codec; + + if (ak4671_codec) { + dev_err(codec->dev, "Another AK4671 is registered\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = ak4671; + codec->name = "AK4671"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = ak4671_set_bias_level; + codec->dai = &ak4671_dai; + codec->num_dai = 1; + codec->reg_cache_size = AK4671_CACHEREGNUM; + codec->reg_cache = &ak4671->reg_cache; + + memcpy(codec->reg_cache, ak4671_reg, sizeof(ak4671_reg)); + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, control); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + ak4671_dai.dev = codec->dev; + ak4671_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&ak4671_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(ak4671); + return ret; +} + +static void ak4671_unregister(struct ak4671_priv *ak4671) +{ + ak4671_set_bias_level(&ak4671->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&ak4671_dai); + snd_soc_unregister_codec(&ak4671->codec); + kfree(ak4671); + ak4671_codec = NULL; +} + +static int __devinit ak4671_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct ak4671_priv *ak4671; + struct snd_soc_codec *codec; + + ak4671 = kzalloc(sizeof(struct ak4671_priv), GFP_KERNEL); + if (ak4671 == NULL) + return -ENOMEM; + + codec = &ak4671->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(client, ak4671); + codec->control_data = client; + + codec->dev = &client->dev; + + return ak4671_register(ak4671, SND_SOC_I2C); +} + +static __devexit int ak4671_i2c_remove(struct i2c_client *client) +{ + struct ak4671_priv *ak4671 = i2c_get_clientdata(client); + + ak4671_unregister(ak4671); + + return 0; +} + +static const struct i2c_device_id ak4671_i2c_id[] = { + { "ak4671", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4671_i2c_id); + +static struct i2c_driver ak4671_i2c_driver = { + .driver = { + .name = "ak4671", + .owner = THIS_MODULE, + }, + .probe = ak4671_i2c_probe, + .remove = __devexit_p(ak4671_i2c_remove), + .id_table = ak4671_i2c_id, +}; + +static int __init ak4671_modinit(void) +{ + return i2c_add_driver(&ak4671_i2c_driver); +} +module_init(ak4671_modinit); + +static void __exit ak4671_exit(void) +{ + i2c_del_driver(&ak4671_i2c_driver); +} +module_exit(ak4671_exit); + +MODULE_DESCRIPTION("ASoC AK4671 codec driver"); +MODULE_AUTHOR("Joonyoung Shim <jy0922.shim(a)samsung.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4671.h b/sound/soc/codecs/ak4671.h new file mode 100644 index 0000000..e2fad96 --- /dev/null +++ b/sound/soc/codecs/ak4671.h @@ -0,0 +1,156 @@ +/* + * ak4671.h -- audio driver for AK4671 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim(a)samsung.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. + * + */ + +#ifndef _AK4671_H +#define _AK4671_H + +#define AK4671_AD_DA_POWER_MANAGEMENT 0x00 +#define AK4671_PLL_MODE_SELECT0 0x01 +#define AK4671_PLL_MODE_SELECT1 0x02 +#define AK4671_FORMAT_SELECT 0x03 +#define AK4671_MIC_SIGNAL_SELECT 0x04 +#define AK4671_MIC_AMP_GAIN 0x05 +#define AK4671_MIXING_POWER_MANAGEMENT0 0x06 +#define AK4671_MIXING_POWER_MANAGEMENT1 0x07 +#define AK4671_OUTPUT_VOLUME_CONTROL 0x08 +#define AK4671_LOUT1_SIGNAL_SELECT 0x09 +#define AK4671_ROUT1_SIGNAL_SELECT 0x0a +#define AK4671_LOUT2_SIGNAL_SELECT 0x0b +#define AK4671_ROUT2_SIGNAL_SELECT 0x0c +#define AK4671_LOUT3_SIGNAL_SELECT 0x0d +#define AK4671_ROUT3_SIGNAL_SELECT 0x0e +#define AK4671_LOUT1_POWER_MANAGERMENT 0x0f +#define AK4671_LOUT2_POWER_MANAGERMENT 0x10 +#define AK4671_LOUT3_POWER_MANAGERMENT 0x11 +#define AK4671_LCH_INPUT_VOLUME_CONTROL 0x12 +#define AK4671_RCH_INPUT_VOLUME_CONTROL 0x13 +#define AK4671_ALC_REFERENCE_SELECT 0x14 +#define AK4671_DIGITAL_MIXING_CONTROL 0x15 +#define AK4671_ALC_TIMER_SELECT 0x16 +#define AK4671_ALC_MODE_CONTROL 0x17 +#define AK4671_MODE_CONTROL1 0x18 +#define AK4671_MODE_CONTROL2 0x19 +#define AK4671_LCH_OUTPUT_VOLUME_CONTROL 0x1a +#define AK4671_RCH_OUTPUT_VOLUME_CONTROL 0x1b +#define AK4671_SIDETONE_A_CONTROL 0x1c +#define AK4671_DIGITAL_FILTER_SELECT 0x1d +#define AK4671_FIL3_COEFFICIENT0 0x1e +#define AK4671_FIL3_COEFFICIENT1 0x1f +#define AK4671_FIL3_COEFFICIENT2 0x20 +#define AK4671_FIL3_COEFFICIENT3 0x21 +#define AK4671_EQ_COEFFICIENT0 0x22 +#define AK4671_EQ_COEFFICIENT1 0x23 +#define AK4671_EQ_COEFFICIENT2 0x24 +#define AK4671_EQ_COEFFICIENT3 0x25 +#define AK4671_EQ_COEFFICIENT4 0x26 +#define AK4671_EQ_COEFFICIENT5 0x27 +#define AK4671_FIL1_COEFFICIENT0 0x28 +#define AK4671_FIL1_COEFFICIENT1 0x29 +#define AK4671_FIL1_COEFFICIENT2 0x2a +#define AK4671_FIL1_COEFFICIENT3 0x2b +#define AK4671_FIL2_COEFFICIENT0 0x2c +#define AK4671_FIL2_COEFFICIENT1 0x2d +#define AK4671_FIL2_COEFFICIENT2 0x2e +#define AK4671_FIL2_COEFFICIENT3 0x2f +#define AK4671_DIGITAL_FILTER_SELECT2 0x30 +#define AK4671_E1_COEFFICIENT0 0x32 +#define AK4671_E1_COEFFICIENT1 0x33 +#define AK4671_E1_COEFFICIENT2 0x34 +#define AK4671_E1_COEFFICIENT3 0x35 +#define AK4671_E1_COEFFICIENT4 0x36 +#define AK4671_E1_COEFFICIENT5 0x37 +#define AK4671_E2_COEFFICIENT0 0x38 +#define AK4671_E2_COEFFICIENT1 0x39 +#define AK4671_E2_COEFFICIENT2 0x3a +#define AK4671_E2_COEFFICIENT3 0x3b +#define AK4671_E2_COEFFICIENT4 0x3c +#define AK4671_E2_COEFFICIENT5 0x3d +#define AK4671_E3_COEFFICIENT0 0x3e +#define AK4671_E3_COEFFICIENT1 0x3f +#define AK4671_E3_COEFFICIENT2 0x40 +#define AK4671_E3_COEFFICIENT3 0x41 +#define AK4671_E3_COEFFICIENT4 0x42 +#define AK4671_E3_COEFFICIENT5 0x43 +#define AK4671_E4_COEFFICIENT0 0x44 +#define AK4671_E4_COEFFICIENT1 0x45 +#define AK4671_E4_COEFFICIENT2 0x46 +#define AK4671_E4_COEFFICIENT3 0x47 +#define AK4671_E4_COEFFICIENT4 0x48 +#define AK4671_E4_COEFFICIENT5 0x49 +#define AK4671_E5_COEFFICIENT0 0x4a +#define AK4671_E5_COEFFICIENT1 0x4b +#define AK4671_E5_COEFFICIENT2 0x4c +#define AK4671_E5_COEFFICIENT3 0x4d +#define AK4671_E5_COEFFICIENT4 0x4e +#define AK4671_E5_COEFFICIENT5 0x4f +#define AK4671_EQ_CONTROL_250HZ_100HZ 0x50 +#define AK4671_EQ_CONTROL_3500HZ_1KHZ 0x51 +#define AK4671_EQ_CONTRO_10KHZ 0x52 +#define AK4671_PCM_IF_CONTROL0 0x53 +#define AK4671_PCM_IF_CONTROL1 0x54 +#define AK4671_PCM_IF_CONTROL2 0x55 +#define AK4671_DIGITAL_VOLUME_B_CONTROL 0x56 +#define AK4671_DIGITAL_VOLUME_C_CONTROL 0x57 +#define AK4671_SIDETONE_VOLUME_CONTROL 0x58 +#define AK4671_DIGITAL_MIXING_CONTROL2 0x59 +#define AK4671_SAR_ADC_CONTROL 0x5a + +#define AK4671_CACHEREGNUM (AK4671_SAR_ADC_CONTROL + 1) + +/* Bitfield Definitions */ + +/* AK4671_AD_DA_POWER_MANAGEMENT (0x00) Fields */ +#define AK4671_PMVCM 0x01 + +/* AK4671_PLL_MODE_SELECT0 (0x01) Fields */ +#define AK4671_PLL 0x0f +#define AK4671_PLL_11_2896MHZ (4 << 0) +#define AK4671_PLL_12_288MHZ (5 << 0) +#define AK4671_PLL_12MHZ (6 << 0) +#define AK4671_PLL_24MHZ (7 << 0) +#define AK4671_PLL_19_2MHZ (8 << 0) +#define AK4671_PLL_13_5MHZ (12 << 0) +#define AK4671_PLL_27MHZ (13 << 0) +#define AK4671_PLL_13MHZ (14 << 0) +#define AK4671_PLL_26MHZ (15 << 0) +#define AK4671_FS 0xf0 +#define AK4671_FS_8KHZ (0 << 4) +#define AK4671_FS_12KHZ (1 << 4) +#define AK4671_FS_16KHZ (2 << 4) +#define AK4671_FS_24KHZ (3 << 4) +#define AK4671_FS_11_025KHZ (5 << 4) +#define AK4671_FS_22_05KHZ (7 << 4) +#define AK4671_FS_32KHZ (10 << 4) +#define AK4671_FS_48KHZ (11 << 4) +#define AK4671_FS_44_1KHZ (15 << 4) + +/* AK4671_PLL_MODE_SELECT1 (0x02) Fields */ +#define AK4671_PMPLL 0x01 +#define AK4671_M_S 0x02 + +/* AK4671_FORMAT_SELECT (0x03) Fields */ +#define AK4671_DIF 0x03 +#define AK4671_DIF_DSP_MODE (0 << 0) +#define AK4671_DIF_MSB_MODE (2 << 0) +#define AK4671_DIF_I2S_MODE (3 << 0) +#define AK4671_BCKP 0x04 +#define AK4671_MSBS 0x08 +#define AK4671_SDOD 0x10 + +/* AK4671_LOUT2_POWER_MANAGEMENT (0x10) Fields */ +#define AK4671_MUTEN 0x04 + +extern struct snd_soc_dai ak4671_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4671; + +#endif -- 1.6.0.4

2 1

[alsa-devel] Is there a way to figure out if the clocks of two snd_pcm_t's deviate? (snd_pcm_info_get_sync?)
by Lennart Poettering 09 Sep '09

09 Sep '09

Heya! I am looking for a way to find out whether the clocks of two snd_pcm_t's are perfectly in sync and don't deviate (or only deviate by a constant phase). i.e. for real duplex devices that use the same crystal for input and output this should tell me "Yes" for the playback snd_pcm_t tested against the recording snd_pcm_t; and for all other, unrelated audio devices it should return "No". I found snd_pcm_info_get_sync() in the docs, but its documentation is a bit terse, so I am wondering if this might be what I want. Is it? Does this (untested) code snippet do what I hope it does? <snip> snd_pcm_info(a, info_a); snd_pcm_info(b, info_b); in_sync = memcmp(&snd_pcm_info_get_sync(info_a), &snd_pcm_info_get_sync(info_b)) == 0; </snip> Thanks, Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4

4 6

[alsa-devel] pts value using alsa for audio capture
by Nicolas Krieger 09 Sep '09

09 Sep '09

Hi, I have a question about how pts are computed using alsa while grabbing audio. I take an example to explain : I do grab audio during 10 seconds, the first 2 seconds, there is no problem, so the pts are like that (I take examples values) : 1 2 3 4 5 6 .... x <- x is the value of the last pts (2nd second). Then, for any reason, no capture during 3 seconds. And then no problem until the end. Nicolas Krieger

1 1

Re: [alsa-devel] [PATCH 2/3] ASoC: DaVinci: pcm, rename variables in prep for ping/pong
by Nori, Sekhar 09 Sep '09

09 Sep '09

On Wed, Sep 09, 2009 at 18:03:58, Caglar Akyuz wrote: > On Wednesday 09 September 2009 15:08:42 Nori, Sekhar wrote: [...] > > > > However, with or without your patch, I noticed a segmentation fault for the > > first time the 'arecord | aplay' loop is created. There is no fault the > > second time around. I haven't spent time debugging this yet. > > > > This is due to these lines in davinci_pcm_enqueue_dma: > > data_type = prtd->params->data_type; > count = period_size / data_type; > > First time running data_type is set to 0. This is a long time issue, I > remember modifying my kernels to check for zero before this division.. Ah, thanks! Planning to submit a fix for this? We can work on the patch if you are busy otherwise. Regards, Sekhar

3 2

[alsa-devel] Layla 3G does not recover from ACPI Suspend
by Mark Hills 09 Sep '09

09 Sep '09

I have an Echo Layla 3G in my workstation. It works for audio, but does not recover from ACPI suspend to RAM. On recovery the system is fine, and the Layla exists in /proc/asound/cards. But when the Layla is used it prints this message to dmesg, multiple times: wait_handshake(): Timeout waiting for DSP Here are the relevant dmesg lines after awakening: Echoaudio Echo3G 0000:03:03.0: restoring config space at offset 0xf (was 0x100, writing 0x104) Echoaudio Echo3G 0000:03:03.0: restoring config space at offset 0x4 (was 0x0, writing 0xe7d00000) Echoaudio Echo3G 0000:03:03.0: restoring config space at offset 0x3 (was 0x0, writing 0xc010) Echoaudio Echo3G 0000:03:03.0: restoring config space at offset 0x1 (was 0x2800000, writing 0x2800112) Echoaudio Echo3G 0000:03:03.0: PCI INT A -> GSI 19 (level, low) -> IRQ 19 I couldn't see anything loading the firmware. I thought unloading and loading snd-echo3g after recovery would help. This shows the firmware being loaded, but then no ALSA device is shown in /proc/asound/cards. On loading snd-echo3g: Echoaudio Echo3G 0000:03:03.0: PCI INT A -> GSI 19 (level, low) -> IRQ 19 Echoaudio Echo3G 0000:03:03.0: firmware: requesting ea/echo3g_dsp.fw Echoaudio Echo3G 0000:03:03.0: firmware: requesting ea/loader_dsp.fw Echoaudio Echo3G 0000:03:03.0: PCI INT A disabled Echoaudio Echo3G: probe of 0000:03:03.0 failed with error -5 Is the firmware not being loaded when it should be? Or is there some extra initialisation not being done (eg. init_hw() in echo3g_dsp.c)? All other devices in the system recover fine, but as far as I'm aware no other PCI devices require firmware. It's a Dell x86 system, kernel 2.6.31-rc7. Thanks for any help, -- Mark

3 3

[alsa-devel] [PATCH v2] ASoC: au1x: PSC-AC97 bugfixes
by Manuel Lauss 08 Sep '09

08 Sep '09

This patch fixes the following bugs: - only reprogram bitdepth if it has changed since last call to hw_params. - add locking inside ac97_read/write functions: When reprogramming sample depth, the ac97 unit has to be disabled, which should not be done in the middle of codec register accesses. - retry timed-out codec register accesses. - wait for status bits to set/clear when starting/stopping various functional blocks; very important after reenabling AC97 unit else sound may be distorted (e.g. high-pitch noise in 1kHz sine wave). - clear fifos before/after starting/stopping RX/TX. - longer timeouts waiting for PSC/AC97 ready after cold reset with certain codecs this can take ridiculous amounts of time. Run-tested on various Au1200 platforms with various codecs. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- The previous patch missed a git add for psc.h, sorry! sound/soc/au1x/psc-ac97.c | 129 +++++++++++++++++++++++++++++++++------------ sound/soc/au1x/psc.h | 1 + 2 files changed, 97 insertions(+), 33 deletions(-) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 479d7bd..a521aa9 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.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 @@ -19,6 +19,7 @@ #include <linux/module.h> #include <linux/device.h> #include <linux/delay.h> +#include <linux/mutex.h> #include <linux/suspend.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,6 +30,9 @@ #include "psc.h" +/* how often to retry failed codec register reads/writes */ +#define AC97_RW_RETRIES 5 + #define AC97_DIR \ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) @@ -45,6 +49,9 @@ #define AC97PCR_CLRFIFO(stype) \ ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC) +#define AC97STAT_BUSY(stype) \ + ((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB) + /* instance data. There can be only one, MacLeod!!!! */ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; @@ -54,24 +61,33 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned short data, tmo; + unsigned short data, retry, tmo; - au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) - udelay(2); + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), + AC97_CDC(pscdata)); + au_sync(); + + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); - if (!tmo) - data = 0xffff; - else data = au_readl(AC97_CDC(pscdata)) & 0xffff; - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); - return data; + return retry ? data : 0xffff; } /* AC97 controller writes to codec register */ @@ -80,16 +96,29 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned int tmo; + unsigned int tmo, retry; - au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) + + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), + AC97_CDC(pscdata)); au_sync(); - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); + + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); } /* AC97 controller asserts a warm reset */ @@ -129,9 +158,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for PSC to indicate it's ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) { printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n"); @@ -143,9 +172,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for AC97 core to become ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n"); } @@ -165,12 +194,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned long r, stat; + unsigned long r, ro, stat; int chans, stype = SUBSTREAM_TYPE(substream); chans = params_channels(params); - r = au_readl(AC97_CFG(pscdata)); + r = ro = au_readl(AC97_CFG(pscdata)); stat = au_readl(AC97_STAT(pscdata)); /* already active? */ @@ -180,9 +209,6 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, (pscdata->rate != params_rate(params))) return -EINVAL; } else { - /* disable AC97 device controller first */ - au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); - au_sync(); /* set sample bitdepth: REG[24:21]=(BITS-2)/2 */ r &= ~PSC_AC97CFG_LEN_MASK; @@ -199,14 +225,40 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, r |= PSC_AC97CFG_RXSLOT_ENA(4); } - /* finally enable the AC97 controller again */ + /* do we need to poke the hardware? */ + if (!(r ^ ro)) + goto out; + + /* ac97 engine is about to be disabled */ + mutex_lock(&pscdata->lock); + + /* disable AC97 device controller first... */ + au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + /* ...wait for it... */ + while (au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) + asm volatile ("nop"); + + /* ...write config... */ + au_writel(r, AC97_CFG(pscdata)); + au_sync(); + + /* ...enable the AC97 controller again... */ au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); au_sync(); + /* ...and wait for ready bit */ + while (!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) + asm volatile ("nop"); + + mutex_unlock(&pscdata->lock); + pscdata->cfg = r; pscdata->rate = params_rate(params); } +out: return 0; } @@ -222,6 +274,8 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); au_writel(AC97PCR_START(stype), AC97_PCR(pscdata)); au_sync(); break; @@ -229,6 +283,13 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_SUSPEND: au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata)); au_sync(); + + while (au_readl(AC97_STAT(pscdata)) & AC97STAT_BUSY(stype)) + asm volatile ("nop"); + + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); + break; default: ret = -EINVAL; @@ -251,6 +312,8 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, if (!au1xpsc_ac97_workdata) return -ENOMEM; + mutex_init(&au1xpsc_ac97_workdata->lock); + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!r) { ret = -ENODEV; @@ -269,9 +332,9 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, goto out1; /* configuration: max dma trigger threshold, enable ac97 */ - au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | - PSC_AC97CFG_TT_FIFO8 | - PSC_AC97CFG_DE_ENABLE; + au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | + PSC_AC97CFG_TT_FIFO8 | + PSC_AC97CFG_DE_ENABLE; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) @@ -386,4 +449,4 @@ module_exit(au1xpsc_ac97_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss <manuel.lauss(a)gmail.com>"); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index 8fdb1a0..3f474e8 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -29,6 +29,7 @@ struct au1xpsc_audio_data { unsigned long pm[2]; struct resource *ioarea; + struct mutex lock; }; #define PCM_TX 0 -- 1.6.4.2

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