March 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] View alsa-lib plugins setup for any app?
by Eliot Blennerhassett 05 Dec '25

05 Dec '25

Greetings, running aplay -v will show the plugin chain setup. /proc/asound/card/pcm/* shows hw and sw params for the card driver when in use by any app Is there any way to show the alsa-lib plugin settings a-la aplay for an arbitrary application? thanks -- Eliot

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[alsa-devel] Error patching file pcm_native.c
by Delusion's Master 20 Nov '24

20 Nov '24

Hi, first of all I apologize in case you receive multiple copies of this message, but I forgot to specify a subject in the original mail. I have a problem compiling the latest alsa driver snapshots. Specifically, it's a problem when patching pcm_native.c: [..] copying file alsa-kernel/core/info.c patching file info.c copying file alsa-kernel/core/pcm.c patching file pcm.c Hunk #3 succeeded at 739 (offset 2 lines). Hunk #5 succeeded at 954 (offset 2 lines). Hunk #7 succeeded at 1045 (offset 2 lines). copying file alsa-kernel/core/pcm_native.c patching file pcm_native.c Hunk #4 succeeded at 1544 (offset -25 lines). Hunk #5 succeeded at 2827 (offset 12 lines). Hunk #6 succeeded at 2878 (offset 10 lines). Hunk #7 FAILED at 2906. Hunk #8 FAILED at 2922. Hunk #9 succeeded at 2915 (offset -23 lines). Hunk #10 succeeded at 3036 (offset 10 lines). Hunk #11 succeeded at 3027 (offset -23 lines). Hunk #12 succeeded at 3074 (offset 10 lines). Hunk #13 succeeded at 3098 (offset -23 lines). Hunk #14 succeeded at 3143 (offset 10 lines). Hunk #15 succeeded at 3124 (offset -23 lines). Hunk #16 succeeded at 3214 (offset 10 lines). Hunk #17 succeeded at 3208 (offset -23 lines). Hunk #18 succeeded at 3271 (offset 10 lines). Hunk #19 succeeded at 3304 (offset -23 lines). Hunk #20 succeeded at 3386 (offset 10 lines). Hunk #21 succeeded at 3411 (offset -23 lines). Hunk #22 succeeded at 3456 (offset 10 lines). Hunk #23 succeeded at 3450 (offset -23 lines). Hunk #24 succeeded at 3525 (offset 10 lines). Hunk #25 succeeded at 3644 (offset -23 lines). 2 out of 25 hunks FAILED -- saving rejects to file pcm_native.c.rej make[2]: *** [pcm_native.c] Error 1 It's some days I get this, but I waited to report this since I thought it could be a temporary problem. My OS is Fedora 11 x86_64. Previous snapshots didn't give me this problem, but I can't say which one broke the compilation, since I usually try updated snapshots only at kernel updates. I searched for this error and the only relevant report I could find is this one: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016420.html a report from last April. What could the problem be? Thanks _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

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[alsa-devel] hda-intel missing mixer channels
by Orm Finnendahl 26 Aug '13

26 Aug '13

Hi, I sent this to alsa-user a week ago but didn't get any reply. Does anybody know, what is responsible for the missing faders? Did the driver change or is it some module loading mechanism in the distro which prevents the faders from showing up in alsamixer? This is so strange. Please help! -- Orm -------------------------------------------------------------------- Hi everybody, after the last update of Ubuntu (intrepid to jaunty) on my Lenovo Ideapad my microphone input doesn't work anymore (neither internal nor external). I found out that alsamixer lacks quite a lot of faders in the newer kernel/module compared to the older one. Some of the faders now unavailable were crucial for Mic input to work on the older system so I'd very much like to get them back ;-) Some more infos: Soundcard: Intel 82801G (ICH7 Family) Chip: Realtek ALC269 Module: snd_hda_intel (in both kernels) Older kernel: Ubuntu stock 2.6.27-14-generic Newer kernel: Ubuntu stock 2.6.28-13-rt Is there any way to make the missing controls available or did the driver change and I have to compile the older driver for the new kernel? Yours, Orm

2 3

[alsa-devel] [ASoc] TDM over one SSC channel on AT91 possible?
by Friedrich Lobenstock 24 Jun '11

24 Jun '11

Hello! The background: I'm just designing an ATMEL AT91RM9200 based board which got two TLV320AIC33 on board. It is either possible to hook up the two TLV320AIC33 to a separate SSC (synchronous serial control) channel each or use one SSC channel and configure it for TDM (time division multiplex) mode. There is already a driver for TLV320AIC14k/TLV320AIC2k in the development tree, so adding support for the TLV320AIC33 should be feasible. What I'm not sure about is the fact if ASOC is prepared for using channels, especially SSC, in TDM mode. Is the core ASOC structure prepared to handle that? Would we be better off instantiate one drive twice each time with different parameters for the SSC channel? Thanks for any feedback. -- MfG / Regards Friedrich Lobenstock _________________________________________________________ SCOTTY Group Austria GmbH a SCOTTY Group plc company, http://www.scottygroup.com/ _________________________________________________________

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Re: [alsa-devel] [PATCH] ASoC: Add support for Cirrus Logic CS42L52 Lowpower Stereo Codec
by Mark Brown 22 Jun '11

22 Jun '11

On Thu, Feb 25, 2010 at 06:07:35PM -0600, Brian wrote: This is a quick first pass review only. It'd be best if you could include a Signed-off-by line in submissions so that they can be merged if they're OK, even if you expect problems you may be surprised! Overall this looks fairly good - there's a lot of comments below but the vast majority of them are stylistic issues or needing to update to current APIs rather than major structural problems with the code. The main thing I don't really follow is the register access code, there's quite a bit of stuff going on in there. > + * published by the Free Software Foundation. > + * Revision history > + * Nov 2007 Initial version. > + * Oct 2008 Updated to 2.6.26 > + * Feb 2010 Updated to latest asoc git tree Please remove the changelog, once things are merged git takes care of that. > + */ > +#define DEBUG Not by default in mainline :) > +#ifdef DEBUG > +#define SOCDBG(fmt, arg...) printk(KERN_ERR "%s: %s() " fmt, SOC_CS42L52_NAME, __FUNCTION__, ##arg) > +#else > +#define SOCDBG(fmt, arg...) > +#endif > +#define SOCINF(fmt, args...) printk(KERN_INFO "%s: " fmt, SOC_CS42L52_NAME, ##args) > +#define SOCERR(fmt, args...) printk(KERN_ERR "%s: " fmt, SOC_CS42L52_NAME, ##args) Please convert the driver to use the standard dev_ and pr_ macros for this (where possible the dev_ ones are better). > +static int soc_cs42l52_set_bias_level(struct snd_soc_codec *codec, > + enum snd_soc_bias_level level); > +static int soc_cs42l52_pcm_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params, > + struct snd_soc_dai *dai); > +static int soc_cs42l52_set_sysclk(struct snd_soc_dai *codec_dai, > + int clk_id, u_int freq, int dir); > +static int soc_cs42l52_digital_mute(struct snd_soc_dai *dai, int mute); > +static int soc_cs42l52_set_fmt(struct snd_soc_dai *codec_dai, > + u_int fmt); > + > +static unsigned int soc_cs42l52_read(struct snd_soc_codec *codec, > + u_int reg); Please use 'unsigned int' rather than 'u_int' for consistency with the rest of ASoC. It's a bit surprising that you need the forward declarations at all but they do no harm. > +/** > + * snd_soc_get_volsw - single mixer get callback > + * @kcontrol: mixer control > + * @uinfo: control element information > + * > + * Callback to get the value of a single mixer control. > + * > + * Returns 0 for success. > + */ > +int snd_soc_cs42l5x_get_volsw(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); > + int reg = kcontrol->private_value & 0xff; The indentation looks wrong - should be tabs for indenting the code blocks. scripts/checkpatch.pl is a pretty good check for this sort of stuff, I'll not mention other issues. > + int shift = (kcontrol->private_value >> 8) & 0x0f; > + int rshift = (kcontrol->private_value >> 12) & 0x0f; > + int max = (kcontrol->private_value >> 16) & 0xff; > + int mask = (1 << fls(max)) - 1; > + int min = (kcontrol->private_value >> 24) & 0xff; > + > + ucontrol->value.integer.value[0] = > + ((snd_soc_read(codec, reg) >> shift) - min) & mask; > + if (shift != rshift) > + ucontrol->value.integer.value[1] = > + ((snd_soc_read(codec, reg) >> rshift) - min) & mask; This and most of the other custom control stuff looks an awful lot like it should be done by adding a SND_SOC_DOUBLE_SIGNED() or something along the lines of the existing _S8() but I didn't read entirely closely. I was speaking to someone on IRC only the other day who needed some of those types for a different device. These should also be updated to use struct soc_mixer_control for the private value rather than shifting everything into the private value, it allows wider ranges and is less error prone. > +static const u8 soc_cs42l52_reg_default[] = { > + 0x00, 0xE0, 0x01, 0x07, 0x05, /*4*/ > + 0xa0, 0x00, 0x00, 0x81, /*8*/ Very nitpicky but you're using a mix of upper and lower case for the letters in the hex digits. > +static inline int soc_cs42l52_read_reg_cache(struct snd_soc_codec *codec, > + u_int reg) > +{ > + u8 *cache = codec->reg_cache; > + > + return reg > SOC_CS42L52_REG_NUM ? -EINVAL : cache[reg]; > +} You should be able to replace all your register I/O code with use of soc-cache.c with 8 bit data, 8 bit register. Except... > + if(info->flags & SOC_CS42L52_ALL_IN_ONE) > + { > + for(i = 0; i < codec->num_dai; i++) > + { > + if(info->machine_handler) > + info->machine_handler(i); ...I'm not sure what's going on here but it could really use an explanation? I'm not sure what ALL_IN_ONE is supposed to do, or what the machine_handler() function is for. > + if(info->flags & SOC_CS42L52_CHIP_SWICTH) > + { > + num = info->flags & SOC_CS42L52_CHIP_MASK; > + if(info->machine_handler) > + info->machine_handler(num); > + } This too. There's a lot of unusual code using this machine handler function. I suspect this can all use ASoC standard stuff. > +static inline int soc_cs42l52_get_revison(struct snd_soc_codec *codec) > +{ > + u8 data; > + u8 addr; > + int num, ret = 0; > + struct soc_codec_cs42l52 *info = (struct soc_codec_cs42l52*)codec->private_data; No need to cast away from void (quite a few examples of this in the code). > + return ret < 0 ? ret : data; I'm not a big fan of the ternery operator... > +static unsigned int soc_cs42l52_read(struct snd_soc_codec *codec, > + u_int reg) > +{ > + u8 data; > + u8 addr; > + int i, ret = 0; > + struct soc_codec_cs42l52 *info = (struct soc_codec_cs42l52*)codec->private_data; > +#ifndef CONFIG_CS42L52_DEBUG > + if(reg == CODEC_CS42L52_SPK_STATUS) > + { > +#endif This really shouldn't be ifdefed. If you need to handle registers that report status from the device with the soc-cache stuff use the volatile_register() callback in the CODEC, any registers that need a CODEC read will call through to the hardware rather than using the cache. > +static const char *cs42l52_adc_mux[] = {"AIN1", "AIN2", "AIN3", "AIN4", "PGA"}; > +static const char *cs42l52_mic_mux[] = {"MIC1", "MIC2"}; > +static const char *cs42l52_stereo_mux[] = {"Mono", "Stereo"}; > +static const char *cs42l52_off[] = {"On", "Off"}; > +static const char *cs42l52_hpmux[] = {"Off", "On"}; > + > +static const struct soc_enum soc_cs42l52_enum[] = { > +SOC_ENUM_DOUBLE(CODEC_CS42L52_ANALOG_HPF_CTL, 4, 6, 2, cs42l52_hpf_freeze), /*0*/ > +SOC_ENUM_SINGLE(CODEC_CS42L52_ADC_HPF_FREQ, 0, 4, cs42l52_hpf_corner_freq), Don't use an array of enums, define individual variables for them. Some drivers use the arrays for historical reasons - they're quite painful to work with if you ever want to change anything in the middle and we've found a few off by one errors in the referencing into the arrays before :/. > +static const struct snd_kcontrol_new soc_cs42l52_controls[] = { > + > +SOC_ENUM("Mic VA Capture Switch", soc_cs42l52_enum[18]), /*0*/ enums shouldn't be called Switch - Switch has a specific meaning to ALSA (see Documentation/sound/alsa/ControlNames.txt, there's quite a few other controls could do with having the components of the name reordered and various other things following that). Normally just omit the ' Switch' from the name. > +SOC_DOUBLE("Analog SR Capture Switch", CODEC_CS42L52_ANALOG_HPF_CTL, 1, 3, 1, 1), > +SOC_DOUBLE("Analog ZC Capture Switch", CODEC_CS42L52_ANALOG_HPF_CTL, 0, 2, 1, 1), This'd normally be 'Capture ZC Switch' if it's for zero cross. > +SOC_ENUM("HPF corner freq Capture Switch", soc_cs42l52_enum[1]), /*5*/ > + > +SOC_SINGLE("Ganged Ctl Capture Switch", CODEC_CS42L52_ADC_MISC_CTL, 7, 1, 1), /* should be enabled init */ What does the comment mean here? > +SOC_SINGLE("HP Analog Gain Playback Volume", CODEC_CS42L52_PB_CTL1, 5, 7, 0), Just "Headphone Volume"? > +SOC_SINGLE("Playback B=A Volume Playback Switch", CODEC_CS42L52_PB_CTL1, 4, 1, 0), /*10*/ /*should be enabled init*/ If this is the mute for the headphone volume it should have the same name but with Switch instead of Volume so applications like alsamixer can display the two together. > +static int soc_cs42l52_add_controls(struct snd_soc_codec *codec) > +{ > + int i,ret = 0; > + > + for(i = 0; i < ARRAY_SIZE(soc_cs42l52_controls); i++) > + { > + ret = snd_ctl_add(codec->card, > + snd_soc_cnew(&soc_cs42l52_controls[i], codec, NULL)); > + if(ret < 0) > + { > + SOCDBG("add cs42l52 controls failed\n"); > + break; Replace this with snd_soc_add_controls(). > + /* Sum switches */ > + SND_SOC_DAPM_PGA("AIN1A Switch", CODEC_CS42L52_ADC_PGA_A, 0, 0, NULL, 0), Without having checked the datasheet I *suspect* most of these switches should be modelled as inputs to a mixer (possibly one with no power control of its own)? The "Switch" in the name certainly looks very out of place in the name of a DAPM control. > + /* Output path */ > + SND_SOC_DAPM_MUX("Passthrough Left Playback Switch", SND_SOC_NOPM, 0, 0, &cs42l52_hpa_mux), Normally "Bypass" instead of "Passthrough". > + /* Mic Bias */ > + {"Mic Bias Capture Switch", "On", "PGA MICA"}, > + {"Mic Bias Capture Switch", "On", "PGA MICB"}, > + {"Mic-Bias", NULL, "Mic Bias Capture Switch"}, > + {"Mic-Bias", NULL, "Mic Bias Capture Switch"}, > + {"ADC Mux Left Capture Switch", "PGA", "Mic-Bias"}, > + {"ADC Mux Right Capture Switch", "PGA", "Mic-Bias"}, > + {"Passthrough Left Playback Switch", "On", "Mic-Bias"}, > + {"Passthrough Right Playback Switch", "On", "Mic-Bias"}, This looks odd - normally the microphone bias would be connected to the microphone input externally and would not be hooked up as an input wihtin the CODEC? > + /* terminator */ > + {NULL, NULL, NULL}, Not needed any more (I'm vaugely surprised this doesn't generate an error). > +#define SOC_CS42L52_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 | SNDRV_PCM_RATE_88200 | \ > + SNDRV_PCM_RATE_96000 ) /*refer to cs42l52 datasheet page35*/ This is equivalent to SNDRV_PCM_RATE_8000_96000 > +/* #define CONFIG_MANUAL_CLK */ This might use some comments (or should possibly be deleted)? > + soc_cs42l52_set_bias_level(soc_codec, SND_SOC_BIAS_OFF); > + soc_codec->bias_level = SND_SOC_BIAS_STANDBY; > + schedule_delayed_work(&soc_codec->delayed_work, msecs_to_jiffies(1000)); > + soc_cs42l52_required_setup(soc_codec); A comment about what the delayed work is up to wouldn't hurt? > + /*set hp default volume*/ > + soc_cs42l52_write(soc_codec, CODEC_CS42L52_HPA_VOL, DEFAULT_HP_VOL); > + soc_cs42l52_write(soc_codec, CODEC_CS42L52_HPB_VOL, DEFAULT_HP_VOL); For pretty much all the configuration you're doing here just let userspace set things up, there's no telling what's an appropriate setting on any given system and presumably the chip designers made a reasonable effort to ensure that the default state of the chip is at least not harmful. > +#ifdef CONFIG_MANUAL_CLK > + soc_cs42l52_write(soc_codec, CODEC_CS42L52_CLK_CTL, > + (soc_cs42l52_read(soc_codec, CODEC_CS42L52_CLK_CTL) > + & ~CLK_CTL_AUTODECT_ENABLE)); Pass this configuration in as platform data, this will allow multiple boards to build from the same kernel. > + ret = snd_soc_init_card(soc_dev); > + if(ret) > + { > + SOCERR("add snd card failed\n"); > + goto card_err; > + } Please develop against git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 (or whatever the latest kernel version branch is at any given time.) You should also be calling snd_soc_register_codec() and snd_soc_register_dai() from this function. > +#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) No need for the ifdefs if you only support I2C, the ifdefs are used by drivers that have multiple control interfaces (typically I2C plus SPI). > + } > + > + > + return ret; > +} > + > +static int cs42l52_i2c_remove(struct i2c_client *client) > +{ > + struct snd_soc_codec *soc_codec; > + if(client) > + { > + soc_codec = i2c_get_clientdata(client); > + if(soc_codec) > + kfree(soc_codec->reg_cache); > + } > + return 0; > +} This should be unregistering from ASoC as well, or refusing to remove if not implemented. > + case SND_SOC_DAIFMT_DSP_B: > + SOCINF("unsupported format\n"); > + ret = -EINVAL; > + goto done; > + default: > + SOCINF("invaild format\n"); > + ret = -EINVAL; > + goto done; Could just squash these together. > + case SND_SOC_DAIFMT_IB_IF: > + SOCDBG("codec dai fmt inversed sclk\n"); > + iface |= IFACE_CTL1_INV_SCLK; > + break; > + case SND_SOC_DAIFMT_IB_NF: > + iface |= IFACE_CTL1_INV_SCLK; > + break; It looks like the hardware only supports the inversion of one of the clocks so only two of the clock polarity combinations should be supported? > + info->format = iface; I'd really expect to see interaction with the hardware here? > + /* 96k */ > + {12288000, 96000, CLK_CTL_S_DS_MODE, CLK_CTL_NOT_32K, CLK_CTL_NOT_27M, CLK_CTL_RATIO_128, 0}, > + {18432000, 96000, CLK_CTL_S_DS_MODE, CLK_CTL_NOT_32K, CLK_CTL_NOT_27M, CLK_CTL_RATIO_128, 0},/*29*/ > + {12000000, 96000, CLK_CTL_S_DS_MODE, CLK_CTL_NOT_32K, CLK_CTL_NOT_27M, CLK_CTL_RATIO_125, 0}, > + {24000000, 96000, CLK_CTL_S_DS_MODE, CLK_CTL_NOT_32K, CLK_CTL_NOT_27M, CLK_CTL_RATIO_125, 1}, If you're feeling adventurous you should be able to use this data to set up constraints for the DAI in the startup() callback so only sample rates supported from the system clock can be used by userspace. > +static int soc_cs42l52_pcm_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params, > + struct snd_soc_dai *dai) > +{ > + int ret = 0; > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct snd_soc_device *soc_dev = rtd->socdev; > + struct snd_soc_codec *soc_codec = soc_dev->card->codec; > + struct soc_codec_cs42l52 *info = (struct soc_codec_cs42l52*)soc_codec->private_data; > + > + u32 clk = 0; > + int index = soc_cs42l52_get_clk(info->sysclk, params_rate(params)); > +static int soc_cs42l52_suspend(struct platform_device *pdev, pm_message_t state) > +{ > + struct snd_soc_device *soc_dev = (struct snd_soc_device*)platform_get_drvdata(pdev); > + struct snd_soc_codec *soc_codec = soc_dev->card->codec; > + > + soc_cs42l52_write(soc_codec, CODEC_CS42L52_PWCTL1, PWCTL1_PDN_CODEC); > + soc_cs42l52_set_bias_level(soc_codec, SND_SOC_BIAS_OFF); I'd expect that set_bias_level() would do what the explict write does? > +#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) > + if(soc_codec_data->i2c_addr) > + { > + ret = i2c_add_driver(&cs42l52_i2c_drv); > + > + if(ret) > + { > + SOCERR("register i2c driver failed\n"); > + goto out; > + > + } > + SOCINF("Cirrus Logic ALSA SoC codec cs42l52 driver verison 1.2 2/2010\n"); > + return ret; > + } This should be changed to use standard device model registration - see http://git.kernel.org/?p=linux/kernel/git/broonie/sound-2.6.git;a=commit;h=… for an example of doing the conversion. > +struct soc_codec_cs42l52_data{ > + u_short i2c_addr; > +}; With the device model stuff this won't be needed any more. > +/* > + *CS42L52 internal registers > + */ > +#define CODEC_CS42L52_CHIP 0x01 > +#define CHIP_ID 0xE0 > +#define CHIP_ID_MASK 0xF8 Many of the register names and other definitions for register fields and values need to be namespace to avoid collisions.

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[alsa-devel] Conexant CX20561 - Microphone problems
by Shawn Starr 19 Apr '11

19 Apr '11

Hello ALSA developers, I am having problems with the microphone on the following hardware this is on a Lenovo W500 laptop. Card: HDA Intel Chip: Conexant CX20561 (Hermosa) Here is the problems im noticing: 1) If I use ALSA 'default' with the microphone I get noise interference on output [high/low/high/low]. If I switch this to use the "ALSA Capture on HDA Intel [CONEXANT Analog] I get interference except it is the standard 60Hz electrical noise [steady noise] 2) Using the internal/external microphones gives an inconsistent volume. It also does not seem to work with the microphone cutting out so much. I need to adjust Digital + another microphone but it still drops capturing. Alsamixer shows: Digital/Docking Microphone/External Microphone/Internal Microphone Here is the module options I give to the sound driver: options snd-intel-hda enable_msi=1 power_save=10 power_save_controller=true model=laptop index=0 Assuming power saving/controller causes the microphone to cut out or not. 3) I'm now aware that this card has no pinouts for PCM capture https://bugzilla.redhat.com/show_bug.cgi?id=502293 Thanks for any help, Shawn.

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[alsa-devel] CA0110 taints kernel
by Vedran Miletić 13 Apr '11

13 Apr '11

Hi, I experience frequent kernel crashes with 2.6.31 and CA0110. I know this card doesn't work, and there is no problem with that; however, it crashing non-stop is not very nice. Also, card doesn't get any controls (and I believe it does get some controls when it's the only card using HDA driver, that is when I disable onboard one). dmesg is here: http://pastebin.com/m7c53562f lspci: 02:00.0 PCI bridge: Creative Labs Device 7006 (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=02, secondary=03, subordinate=03, sec-latency=64 Memory behind bridge: fe900000-fe9fffff Capabilities: [50] Power Management version 3 Capabilities: [60] MSI: Enable- Count=1/16 Maskable- 64bit+ Capabilities: [80] Subsystem: Creative Labs Device 0010 Capabilities: [90] Express PCI/PCI-X Bridge, MSI 00 Capabilities: [100] Advanced Error Reporting Kernel modules: shpchp 03:00.0 Audio device: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG Subsystem: Creative Labs Device 0018 Flags: bus master, medium devsel, latency 64, IRQ 16 Memory at fe9fc000 (32-bit, non-prefetchable) [size=16K] Capabilities: [dc] Power Management version 3 Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel Any ideas? -- Vedran Miletić P.S. Please CC me, as I'm not subscribed to mailing list at the moment.

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[alsa-devel] idt STAC927x adat supported (sigmatel HDA codec) ?
by Eric 03 Mar '11

03 Mar '11

Is the ADAT output of the IDT (sigmatel) STAC927x HDA codec supported in alsa ? If not could someone please try to add this incredibly nice feature in the existing alsa driver ? Thank you very much

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[alsa-devel] davinci-i2c,pcm updates
by Troy Kisky 27 Sep '10

27 Sep '10

This patch set is against git://git.alsa-project.org/alsa-kernel.git but the last patch in series won't compile until the sram allocator patch is available.

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[alsa-devel] [PATCH 2.6.32.3] hda_intel: ALSA HD Audio patch for Intel Cougar Point DeviceIDs
by Seth Heasley 13 Sep '10

13 Sep '10

This patch adds the Intel Cougar Point (PCH) HD Audio Controller DeviceIDs. Signed-off-by: Seth Heasley <seth.heasley(a)intel.com> --- linux-2.6.32.3/sound/pci/hda/hda_intel.c.orig 2010-01-06 15:07:45.000000000 -0800 +++ linux-2.6.32.3/sound/pci/hda/hda_intel.c 2010-01-07 13:10:28.000000000 -0800 @@ -116,6 +116,7 @@ "{Intel, ICH9}," "{Intel, ICH10}," "{Intel, PCH}," + "{Intel, CPT}," "{Intel, SCH}," "{ATI, SB450}," "{ATI, SB600}," @@ -2648,6 +2649,8 @@ { PCI_DEVICE(0x8086, 0x3a6e), .driver_data = AZX_DRIVER_ICH }, /* PCH */ { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_ICH }, + /* CPT */ + { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_ICH }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, /* ATI SB 450/600 */

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Re: [alsa-devel] Patchmix-style application for EMU cards
by Camilo Polymeris 30 Aug '10

30 Aug '10

Hello > That is more or less what I had figured out, but it how are playback_1 to > > _16 channels routed to DSP0-15? It seems that it depends on the setting > of > > Front, Surround, LFE, etc. mixer controls. If they are turned up, I > don't > > understand what happens, sound to playback channels appears on various > DSP > > buses, at different levels. > > Really? Well, I didn't much experiment with surround mode. In source code, > I > can see that there are up to playback 8 channels at hw:0,0 with can be > controlled by faders in alsamixer. I tried surround40 and surroun51 on my > emu1616m and it worked as expected. The channles were mapped to first > couple > of DSP ports in a correct order. In fact this should work the same as > audigy > does, except the result is passed to DSP ports instead of physical > outputs. > On my card, playback channels 1 & 2 (hw:0,3) appear on all DSP busses associated with surround. Say you turn up the "Side" fader in alsamixer, channels 1 & 2 will appear at the relevant DSPs (6-7, I think) Whatever, for now I am just keeping all surround controls down (that is, Front, Surround, Center, Side & LFE). > Anyway, I think if there is another layer of routing done in audigy DSP, > it > shouldn't be used in EMU specific application. Or at least not for now. > The > most important thing is to provide all the switches and FPGA routing > scheme > for each EMU model. Or, to be precise, three routing schemes for each EMU > model, because for "double speed" (e.g. 96kHz) and "quadro speed" (e.g. > 192kHz) modes of operation the numebr of both physical and virtual > channels > changes. > I think the switches and enums for the card models are rather trivial, one can handle it with a customized version of qamix (done with a simple XML file), for example. A separate application would not be necessary for only that. > > If I turn the Front/Surround/LFE/etc. faders down, I can mix playback > > channels to DSP using the "Multichannel PCM Send/Routing/Volume" faders, > in > > a more or less intuitive way, besides that routing values do not map > > directly to DSP buses (i.e. 14-15 map to DSP0-1, but 2-3 map to DSP2-3). > Handling of this controls motivated me to write the application in the first place. No sound at all will appear if I don't set the appropiate controls for each playback channel: Multichannel PCM Volume (full), Multichannel PCM Send (an index of 8 controls, the first 4 select the target DSP) & Multichannel PCM Send Volume (again faders 0-3 of 8) By the way, only DSP0-7 work in this way. I have no idea how to make DSP8-31 work. On which card you did tested it? The driver was primary written for the > 1820M/1212M V1, so some things may not work well on other models and need > to > be fixed. I have a 1212m, don't know which version, but I think it's V1-- it does have firewire. I am using driver version 1.0.16. Thanks, Camilo.

3 3

[alsa-devel] [PATCH 1/4] ASoC: DaVinci: remove requirement that dma_params is 1st in structure
by Troy Kisky 13 Jul '10

13 Jul '10

Remove requirement that dma_params is 1st in the structures davinci_audio_dev and davinci_mcbsp_dev. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- sound/soc/davinci/davinci-i2s.c | 6 +----- sound/soc/davinci/davinci-mcasp.c | 1 + sound/soc/davinci/davinci-mcasp.h | 5 ----- sound/soc/davinci/davinci-pcm.c | 7 ++++--- 4 files changed, 6 insertions(+), 13 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 4ae7070..9e69a4e 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -98,11 +98,6 @@ enum { }; struct davinci_mcbsp_dev { - /* - * dma_params must be first because rtd->dai->cpu_dai->private_data - * is cast to a pointer of an array of struct davinci_pcm_dma_params in - * davinci_pcm_open. - */ struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; #define MOD_DSP_A 0 @@ -547,6 +542,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; davinci_i2s_dai.private_data = dev; + davinci_i2s_dai.dma_data = dev->dma_params; ret = snd_soc_register_dai(&davinci_i2s_dai); if (ret != 0) goto err_free_mem; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 5d1f98a..5023915 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -907,6 +907,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->channel = res->start; davinci_mcasp_dai[pdata->op_mode].private_data = dev; + davinci_mcasp_dai[pdata->op_mode].dma_data = dev->dma_params; davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev; ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]); diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 9d179cc..582c924 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -39,11 +39,6 @@ enum { }; struct davinci_audio_dev { - /* - * dma_params must be first because rtd->dai->cpu_dai->private_data - * is cast to a pointer of an array of struct davinci_pcm_dma_params in - * davinci_pcm_open. - */ struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; int sample_rate; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index c73a915..c2c2e85 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -238,10 +238,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) struct davinci_runtime_data *prtd; int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->private_data; - struct davinci_pcm_dma_params *params = &pa[substream->stream]; - if (!params) + struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params *params; + if (!pa) return -ENODEV; + params = &pa[substream->stream]; snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware); /* ensure that buffer size is a multiple of period size */ -- 1.5.6.3

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[alsa-devel] Verifying mixer dB data/Invalid dB data from USB cards, especially Aureon 5.1 MkII
by Lennart Poettering 25 Jun '10

25 Jun '10

Heya, a while back I wrote a little tool "dbmeasure" which can be used to determine dB attenuation data for alsa mixer controls. I have now extended this little dB tools suite to include a tiny tool "dbverify" that can be used to verify the correctness of existing dB data exposed by the drivers. It should be relatively easy to use that tool, suitable even for non-guru folks who want to debug the dB information from alsa. http://git.0pointer.de/?p=dbmeasure.git git://git.0pointer.de/dbmeasure.git Just run "make" in a git checkout and then try: $ ./dbverify Aureon51MkII Master 30 200 That will verify the dB data of the "Master" control of the sound card that goes by the id "Aureon51MkII". It will compare the discrete volume steps 30 and 200: first it will play a full amplitude sine wave at mixer volume step 30, and then will change to mixer volume step 200, but attenuate the sine wave in software compensating for the volume change. So once you hear the sine wave attenuated by the hw, and once by the software. If the card's dB data is correct both sines should have the same volume. Now, playing around with this, I could verify what I already mentioned to Jarsolav elsewhere earlier: it seems that particularly USB cards seem to expose invalid dB data (at least all mine do). For example, for the above mentioned Aureon 5.1 it's really *way* off -- if you posess that card just run the line above, and listen to the difference. It's really bad. OTOH the dB data of the HDA cards I own is pretty accurate as it seems, at least I was unable to decide just by listening which sine wave was attenuated by sw, and which by hw. Getting back to the invalid dB data from the USB cards: the question is whether the USB descriptor data is badly parsed, and the dB mismatch hence systematic for USB cards, or if these cards are just crappy and include invalid data? In which case I wonder what we could do about that? The Aureon's dB range already looks suspicous, since both the max and the min dB value are < 0 (-47.69 to -1.97), so maybe we could add some heuristics to filter out data that already looks suspicous? Invalid dB data from the driver is a real problem for the "flat volume" logic in PA. We basically allow each app to control the full hw volume range individually, and then set the hw volume to the max of what all apps wanted and attenuate the other streams accordingly. On the Aureon this doesn't work at all, since the attenuation of the streams is miscalculated due to the invalid dB data. Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] no audio output with TYAN S7025 motherboard and Realtek ALC262
by Bryan Green 20 Jun '10

20 Jun '10

Hi, I have several systems with a TYAN S7025 motherboard and Realtek ALC262 chipset. I have not been able to produce any audio output. Well, that is except for one of the systems: after a variety of tweeks and tests, I got sound working on one system, but I do not know what change enabled sound to start working, and I have not been able to reproduce the effect on the other systems. I have tested with kernels up to 2.6.33, and with ALSA version 1.0.22.1. For a non-working system, 'alsaconf' gives the following output: Unknown hardware: "HDA-Intel" "Realtek ALC262" "HDA:10ec0262,10ec0262,00100202" "0x8086" "0x3a3e" And dmesg gives the following: HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 HDA Intel 0000:00:1b.0: irq 89 for MSI/MSI-X HDA Intel 0000:00:1b.0: setting latency timer to 64 hda_codec: ALC262: BIOS auto-probing. autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) hp_outs=0 (0x0/0x0/0x0/0x0/0x0) mono: mono_out=0x0 dig-out=0x1e/0x0 inputs: mic=0x18, fmic=0x19, line=0x1a, fline=0x0, cd=0x1c, aux=0x0 dig-in=0x1f realtek: No valid SSID, checking pincfg 0x411111f0 for NID 0x1d realtek: Enable default setup for auto mode as fallback input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input8 For the working system, the dmesg output is the same except for two lines: autoconfig: line_outs=1 (0x1b/0x0/0x0/0x0/0x0) inputs: mic=0x18, fmic=0x19, line=0x1a, fline=0x0, cd=0x1c, aux=0x14 Is there a method for tweaking these values on the non-working system to match the working system, to see if that works? I've uploaded the alsa-info output from a non-working system here: http://www.alsa-project.org/db/?f=91057bfc12790688f1d257743e53d5d0ec2ce1e2 I've uploaded the alsa-info output from the working system here: http://www.alsa-project.org/db/?f=198420f14cd0062713636408640636e6d474cf94 Thanks, -bryan

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Re: [alsa-devel] Crash 2.6.32.x, Echo 3G, Alsa, SCSI
by Clemens Ladisch 03 Jun '10

03 Jun '10

Frederick V Heitkamp wrote: > Repeatable Hard Crash. > What else do I need to provide? > > [ 5484.995249] WriteControlReg: not written, no change > [ 5485.069621] divide error: 0000 [#1] pcm_hw_params ok > [ 5485.070159] Prepare rate=44100 format=2 channels=2 > [ 5485.070161] set_audio_format[14] = 5 > [ 5485.070166] Prepare rate=44100 format=2 channels=2 > [ 5485.070167] set_audio_format[14] = 5 > [ 5485.070003] PREEMPT SMP > ... > [ 5485.070003] EIP is at pcm_pointer+0x37/0x70 [snd_echo3g] > ... > [ 5485.965788] [<c10041c0>] ? do_divide_error+0x0/0x90 > [ 5485.980619] [<f87aa037>] ? pcm_pointer+0x37/0x70 [snd_echo3g] > [ 5485.998047] [<c104764e>] ? run_timer_softirq+0x17e/0x2e0 > [ 5486.014175] [<f87ac9bf>] ? snd_echo_interrupt+0x11f/0x240 [snd_echo3g] > [ 5486.033940] [<c107a5d5>] ? handle_IRQ_event+0x45/0x190 bytes_to_frames() divides by runtime->frame_bits which is not set until after the hw_params callback has succeeded, but the corresponding chip->substream[] entry is set in that callback, by init_engine(). It should probably have been set in the prepare callback. Frederick, please try the patch below. Takashi, is there any reason why the runtime-> fields are initialized after the hw_params callback? Regards, Clemens --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -403,12 +403,6 @@ if (err < 0) goto _error; - if (substream->ops->hw_params != NULL) { - err = substream->ops->hw_params(substream, params); - if (err < 0) - goto _error; - } - runtime->access = params_access(params); runtime->format = params_format(params); runtime->subformat = params_subformat(params); @@ -446,6 +440,13 @@ runtime->boundary *= 2; snd_pcm_timer_resolution_change(substream); + + if (substream->ops->hw_params != NULL) { + err = substream->ops->hw_params(substream, params); + if (err < 0) + goto _error; + } + runtime->status->state = SNDRV_PCM_STATE_SETUP; pm_qos_remove_requirement(PM_QOS_CPU_DMA_LATENCY,

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[alsa-devel] usbaudio: Support for USB audio v2.0 devices
by Daniel Mack 27 May '10

27 May '10

Hi, I've been working on support for USB devices compliant to audio class v2.0 and would like to share what I got. First problem is that they literally changed every single descriptor type by adding or removing fields or changing their width, respectively. Even worse, some enum values were dropped and reassigned so their numerical value has a different meaning now. My first step was to clean up most of the existing descriptor parsers by moving away from anonymous byte array access to real structs so we can see which fields we're actually dereferencing. In a second step, I added definitions for the replacement descriptors and use them at appropriate places. The bInterfaceProtocol field must always be set correctly of course, otherwise we'll parse garbage. Other things that have changed from v1.0 to v2.0 are: * The number of streaming interfaces is now reported by a standard interface association descriptor. The old approach using a proprietary descriptor extension is deprecated. * The number of channels per interface is now stored in the AS_GENERAL descriptor (used to be part of the FORMAT_TYPE descriptor). * The list of supported sample rates is no longer stored in a variable length appendix of the FORMAT_TYPE descriptor but is retrieved from the device using a class specific GET_RANGE command. * Supported sample formats are now reported as 32bit bitmap rather than as a fixed value. For now, this is worked around by choosing just one of them. Eventually, the code should propagate the full ability of the device. * A devices needs to have at least one CLOCK_SOURCE descriptor which denotes a clockID that is needed as argument to the class request command. If can have more than one though and the host driver is in charge to choose the right one. There are also new types for clock selectors and sample rate converters that a device might report. This is all unsupported at the moment. We only parse the first CLOCK_SOURCE and take this one. * Many descriptors (format_type, ...) have changed their layout. Handle this by casting the descriptors to the appropriate structs. With the following patches applied, a v2.0 device is recognized and reported as ALSA device. I can even hear at least some kind of music, but the sample rate setting does not yet seem to be correct. Also, the existing mixer code does not work at all for v2.0 devices. This will be quite some work to support that and I fear the whole thing must be reimplemented for v2.0 specifically. For now, I worked around that by not parsing them at all but bailing out very early. However, I would like to make sure the changes I made so far won't cause any regression for v1.0 devices, and so it would be good to see them applied as a preliminary support. I'll continue working on this and send more updates along. Let me know what you think. Thanks, Daniel

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[alsa-devel] safe support for rewind in ALSA
by pl bossart 04 May '10

04 May '10

All, I'd like to reopen a thread on concerns with the current implementation of rewind in ALSA. This was already discussed on the mailing list last year without any progress. (see http://thread.gmane.org/gmane.linux.alsa.devel/65759/focus=65773 and http://article.gmane.org/gmane.linux.alsa.devel/66902) The problem is that snd_pcm_rewind() allows in theory to go back to the next sample to be rendered. This however isn't possible on any existing hardware. DMAs do prefetch samples ahead of time, and output interfaces have FIFOs on top of this. This isn't too bad in the PC world with HDAudio, since the prefetching is limited. However in the embedded space it has become a headache. To reduce power consumption, most of the recent embedded interfaces do embed some local internal memory (5kB for TI's McBSP3, a lot more for others), and we should provide a means for driver developers to set a bound on how much can be rewound safely when power-saving features are enabled. Note that there already are a set of definitions for FIFOs, but they are either obsolete or no one uses them (actually HDAudio makes use of this value internally in azx_dev but doesn't export it). My suggestion would be to add a 'fifo' parameter to the runtime structure (so that it's visible to the core), let the driver developer fill the value in the .open routine, and use this parameter in rewind to prevent rewinding too much. This parameter would include both hardware FIFOs and DMA buffers. Comments welcome. - Pierre

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[alsa-devel] Reforming S3C I2S towards supporting I2Sv4
by Jassi Brar 02 May '10

02 May '10

Hello, I have prepared some patches that debug and modify the behaviour of extant S3C I2S support towards managing I2Sv4 controllers as well. Overall pic... For now the 'new' I2Sv4 CPU driver(s3c64xx-i2s-v4.c) is almost a copy of s3c64xx-i2s.c The driver will evolve as support for more features is added. Minor quirks in otherwise similar controllers are handled by defining a new field(feature) in 'struct s3c_i2sv2_info' that is meant to contain 1-bit flags for absence/presence of such quirks. The header with I2S register map and bit definitions has been copied to where the drivers are(sound/soc/s3c24xx/) since the header has nothing usable for platform code. Also, it will help avoid need for co-ordination between ASoC and S3C ARCH trees. For now, the header regs-s3c2412-iis.h is left intact but rendered useless by making ASoC drivers include the newly copied version of it (sound/soc/s3c24xx/regs-i2s-v2.h) Later the header could be dropped by patches to S3C PLAT tree. The patches apply against 'for-2.6.35' of http://www.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git Regards. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com>

6 16

[alsa-devel] [PATCH 01/20] ASoC: S3C: I2Sv2: Add missing semicolon
by Jassi Brar 27 Apr '10

27 Apr '10

Add missing semicolon after s3c2412_i2s_delay Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/s3c-i2s-v2.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index aa84f4c..8f08508 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -753,7 +753,7 @@ int s3c_i2sv2_register_dai(struct snd_soc_dai *dai) /* Allow overriding by (for example) IISv4 */ if (!ops->delay) - ops->delay = s3c2412_i2s_delay, + ops->delay = s3c2412_i2s_delay; dai->suspend = s3c2412_i2s_suspend; dai->resume = s3c2412_i2s_resume; -- 1.6.2.5

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[alsa-devel] [PATCH 1/4] pxa: add namespace on ssp
by Haojian Zhuang 17 Apr '10

17 Apr '10

3 6

[alsa-devel] [PATCH 3/4] ASoC: support pxa168 ssp in ASoC
by Haojian Zhuang 12 Apr '10

12 Apr '10

2 1

[alsa-devel] [PATCH]Support MCP89 and GT21x hdmi audio
by Wei Ni 07 Apr '10

07 Apr '10

Hi, Takashi I developed the hdmi audio driver for new chipset MCP89 and GT21x. The new HAD controller and codec support standard HDMI operation. I attached the patch file, please check it. Thanks Wei. ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

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[alsa-devel] [PATCH 00/37] OF: move device node pointer into struct device.
by Grant Likely 06 Apr '10

06 Apr '10

Currently on OF enabled platforms, the device node pointer is stored in both device.archdata.of_node (.prom_node on sparc), and in of_device.node OF device tree support is being generalized to work on any platform, so instead of adding the of_node pointer to each platform's archdata individually, this patch series moves the of_node pointer into struct device proper and fixes up all users to reference the new location. The last 3 patches in this series remove the old .archdata.of_node, .archdata.prom_node and of_device.node instances. This series has been compile tested on powerpc, sparc and microblaze, (allmodconfig used on sparc and powerpc) and boot tested on an mpc5200 powerpc. All known (or at least all I could find) in-tree users of the old names have been removed. This series should also be fully bisectable. After I collect acks, I would like to get this whole series into linux-next to see if I've missed any references, but I can probably hold off merging the last three patches when the 2.6.35 merge window opens to give out of tree users a bit more time to adapt. In fact, I'd probably wait a few extra days after pushing out the bulk of this series before I push the last 3 patches into my linux-next branch to isolate the drivers that I missed while fixing. Cheers, g. --- Grant Likely (37): arch/sparc: Remove .archdata.of_node and use .of_node in struct device arch/microblaze: Remove .archdata.of_node and use .of_node in struct device arch/powerpc: Remove .archdata.of_node and use .of_node in struct device sound/sparc: use .dev.of_node instead of .node in struct of_device sound/soc: use .dev.of_node instead of .node in struct of_device sound/aoa: use .dev.of_node instead of .node in struct of_device drivers/of: use .of_node instead of .archdata.of_node in struct device drivers/cdrom: use .of_node instead of .archdata.of_node in struct device drivers/char: use .of_node instead of .archdata.of_node in struct device drivers/scsi: use .of_node instead of .archdata.of_node in struct device drivers/of: use .dev.of_node instead of .node in struct of_device drivers/mtd: use .dev.of_node instead of .node in struct of_device drivers/infiniband: use .dev.of_node instead of .node in struct of_device drivers/char: use .dev.of_node instead of .node in struct of_device drivers/i2c: use .dev.of_node instead of .node in struct of_device drivers/leds: use .dev.of_node instead of .node in struct of_device drivers/ata: use .dev.of_node instead of .node in struct of_device drivers/input: use .dev.of_node instead of .node in struct of_device drivers/serial: use .dev.of_node instead of .node in struct of_device drivers/pcmcia: use .dev.of_node instead of .node in struct of_device drivers/net: use .dev.of_node instead of .node in struct of_device drivers/sbus: use .dev.of_node instead of .node in struct of_device drivers/mmc: use .dev.of_node instead of .node in struct of_device drivers/spi: use .dev.of_node instead of .node in struct of_device drivers/macintosh: use .dev.of_node instead of .node in struct of_device drivers/usb: use .dev.of_node instead of .node in struct of_device drivers/scsi: use .dev.of_node instead of .node in struct of_device drivers/video: use .dev.of_node instead of .node in struct of_device drivers/atm: use .dev.of_node instead of .node in struct of_device drivers/watchdog: use .dev.of_node instead of .node in struct of_device drivers/serial: use .dev.of_node instead of .node in struct of_device arch/microblaze: use .dev.of_node instead of .node in struct of_device arch/powerpc: use .dev.of_node instead of .node in struct of_device arch/sparc: use .dev.of_node instead of .node in struct of_device of/sparc: use dev->of_node instead of dev->archdata.prom_node of: Eliminate dev_archdata_get_node() i2c/of: Allow device node to be passed via i2c_board_info arch/microblaze/include/asm/device.h | 14 ------------ arch/microblaze/include/asm/of_device.h | 1 - arch/microblaze/kernel/of_device.c | 12 ++++------ arch/microblaze/kernel/of_platform.c | 4 ++- arch/powerpc/include/asm/device.h | 22 +++++++------------ arch/powerpc/include/asm/macio.h | 2 +- arch/powerpc/include/asm/of_device.h | 1 - arch/powerpc/kernel/ibmebus.c | 4 ++- arch/powerpc/kernel/of_device.c | 12 ++++------ arch/powerpc/kernel/of_platform.c | 16 +++++++------- arch/powerpc/kernel/pci-common.c | 1 - arch/powerpc/kernel/vio.c | 17 +++++++-------- arch/powerpc/platforms/52xx/mpc52xx_gpio.c | 4 ++- arch/powerpc/platforms/52xx/mpc52xx_gpt.c | 16 +++++++------- arch/powerpc/platforms/52xx/mpc52xx_lpbfifo.c | 6 +++-- arch/powerpc/platforms/82xx/ep8248e.c | 6 +++-- arch/powerpc/platforms/83xx/suspend.c | 2 +- arch/powerpc/platforms/cell/axon_msi.c | 2 +- arch/powerpc/platforms/cell/iommu.c | 5 ++-- arch/powerpc/platforms/pasemi/gpio_mdio.c | 2 +- arch/powerpc/platforms/pasemi/setup.c | 4 ++- arch/powerpc/platforms/ps3/system-bus.c | 1 - arch/powerpc/platforms/pseries/iommu.c | 2 +- arch/powerpc/sysdev/axonram.c | 6 +++-- arch/powerpc/sysdev/bestcomm/bestcomm.c | 8 +++---- arch/powerpc/sysdev/fsl_msi.c | 14 ++++++------ arch/powerpc/sysdev/fsl_pmc.c | 2 +- arch/powerpc/sysdev/pmi.c | 2 +- arch/sparc/include/asm/device.h | 15 ------------- arch/sparc/include/asm/fb.h | 2 +- arch/sparc/include/asm/floppy_64.h | 4 ++- arch/sparc/include/asm/of_device.h | 1 - arch/sparc/include/asm/parport.h | 2 +- arch/sparc/kernel/auxio_64.c | 2 +- arch/sparc/kernel/central.c | 4 ++- arch/sparc/kernel/chmc.c | 10 ++++----- arch/sparc/kernel/ioport.c | 2 +- arch/sparc/kernel/of_device_32.c | 18 +++++++--------- arch/sparc/kernel/of_device_64.c | 29 ++++++++++++------------- arch/sparc/kernel/of_device_common.c | 4 ++- arch/sparc/kernel/pci.c | 13 ++++------- arch/sparc/kernel/pci_common.c | 9 ++++---- arch/sparc/kernel/pci_fire.c | 4 ++- arch/sparc/kernel/pci_msi.c | 18 ++++++++-------- arch/sparc/kernel/pci_psycho.c | 4 ++- arch/sparc/kernel/pci_sabre.c | 4 ++- arch/sparc/kernel/pci_schizo.c | 14 ++++++------ arch/sparc/kernel/pci_sun4v.c | 8 +++---- arch/sparc/kernel/power.c | 4 ++- arch/sparc/kernel/psycho_common.c | 2 +- arch/sparc/kernel/sbus.c | 16 +++++++------- arch/sparc/kernel/time_32.c | 2 +- arch/sparc/kernel/time_64.c | 6 +++-- drivers/ata/pata_macio.c | 2 +- drivers/ata/pata_of_platform.c | 2 +- drivers/atm/fore200e.c | 16 ++++++++------ drivers/cdrom/viocd.c | 2 +- drivers/char/hw_random/n2-drv.c | 2 +- drivers/char/hw_random/pasemi-rng.c | 2 +- drivers/char/ipmi/ipmi_si_intf.c | 4 ++- drivers/char/viotape.c | 2 +- drivers/gpio/pca953x.c | 2 +- drivers/i2c/busses/i2c-mpc.c | 17 ++++++++------- drivers/i2c/i2c-core.c | 3 +++ drivers/ide/pmac.c | 10 ++++----- drivers/infiniband/hw/ehca/ehca_main.c | 11 +++++---- drivers/input/serio/i8042-sparcio.h | 2 +- drivers/input/serio/xilinx_ps2.c | 8 +++---- drivers/leds/leds-gpio.c | 2 +- drivers/macintosh/macio_asic.c | 16 +++++++------- drivers/macintosh/macio_sysfs.c | 6 +++-- drivers/macintosh/rack-meter.c | 4 ++- drivers/macintosh/therm_pm72.c | 2 +- drivers/mmc/host/of_mmc_spi.c | 4 ++- drivers/mmc/host/sdhci-of-core.c | 2 +- drivers/mtd/maps/physmap_of.c | 6 +++-- drivers/mtd/maps/sun_uflash.c | 2 +- drivers/mtd/nand/fsl_elbc_nand.c | 6 +++-- drivers/mtd/nand/pasemi_nand.c | 2 +- drivers/net/can/sja1000/sja1000_of_platform.c | 4 ++- drivers/net/ehea/ehea_main.c | 14 ++++++------ drivers/net/greth.c | 3 ++- drivers/net/ibm_newemac/core.c | 13 ++++++----- drivers/net/ibm_newemac/debug.c | 9 ++++---- drivers/net/ibm_newemac/debug.h | 4 ++- drivers/net/ibm_newemac/mal.c | 28 ++++++++++++------------ drivers/net/ibm_newemac/rgmii.c | 12 +++++----- drivers/net/ibm_newemac/tah.c | 7 +++--- drivers/net/ibm_newemac/zmii.c | 9 ++++---- drivers/net/myri_sbus.c | 2 +- drivers/net/niu.c | 10 ++++----- drivers/net/phy/mdio-gpio.c | 6 +++-- drivers/net/sunbmac.c | 6 +++-- drivers/net/sunhme.c | 8 +++---- drivers/net/sunlance.c | 6 +++-- drivers/net/sunqe.c | 6 +++-- drivers/net/xilinx_emaclite.c | 8 +++---- drivers/of/device.c | 20 +++++++++-------- drivers/of/of_i2c.c | 4 ++- drivers/of/of_mdio.c | 5 ++-- drivers/of/of_spi.c | 1 - drivers/pcmcia/electra_cf.c | 2 +- drivers/sbus/char/bbc_envctrl.c | 4 ++- drivers/sbus/char/bbc_i2c.c | 4 ++- drivers/sbus/char/display7seg.c | 2 +- drivers/sbus/char/envctrl.c | 2 +- drivers/sbus/char/flash.c | 4 ++- drivers/sbus/char/uctrl.c | 2 +- drivers/scsi/ibmvscsi/ibmvfc.c | 2 +- drivers/scsi/ibmvscsi/ibmvscsi.c | 2 +- drivers/scsi/qlogicpti.c | 10 ++++----- drivers/scsi/sun_esp.c | 16 +++++++------- drivers/serial/apbuart.c | 2 +- drivers/serial/mpc52xx_uart.c | 8 +++---- drivers/serial/nwpserial.c | 2 +- drivers/serial/of_serial.c | 4 ++- drivers/serial/pmac_zilog.c | 2 +- drivers/serial/sunhv.c | 2 +- drivers/serial/sunsab.c | 6 +++-- drivers/serial/sunsu.c | 6 +++-- drivers/serial/sunzilog.c | 8 +++---- drivers/serial/uartlite.c | 6 +++-- drivers/spi/mpc52xx_psc_spi.c | 11 +++++---- drivers/spi/mpc52xx_spi.c | 14 ++++++------ drivers/spi/spi_mpc8xxx.c | 6 +++-- drivers/usb/host/ehci-ppc-of.c | 2 +- drivers/usb/host/isp1760-if.c | 2 +- drivers/usb/host/ohci-ppc-of.c | 2 +- drivers/video/cg6.c | 2 +- drivers/video/ffb.c | 2 +- drivers/watchdog/cpwd.c | 2 +- include/linux/i2c.h | 4 +++ sound/aoa/fabrics/layout.c | 2 +- sound/aoa/soundbus/core.c | 8 +++---- sound/aoa/soundbus/i2sbus/control.c | 2 +- sound/aoa/soundbus/i2sbus/core.c | 4 ++- sound/aoa/soundbus/sysfs.c | 4 ++- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/sparc/cs4231.c | 6 +++-- sound/sparc/dbri.c | 2 +- 140 files changed, 422 insertions(+), 458 deletions(-) -- Signature

10 52

[alsa-devel] [PATCH] WM8750: Convert to new API
by Marek Vasut 05 Apr '10

05 Apr '10

Register the WM8750 as a SPI or I2C device. This patch mostly shuffles code around. Hugely inspired by WM8753 which was already converted. Signed-off-by: Marek Vasut <marek.vasut(a)gmail.com> --- sound/soc/codecs/wm8750.c | 297 +++++++++++++++++++++------------------------ 1 files changed, 140 insertions(+), 157 deletions(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index ee08408..f01513e 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -29,11 +29,6 @@ #include "wm8750.h" -/* codec private data */ -struct wm8750_priv { - unsigned int sysclk; -}; - /* * wm8750 register cache * We can't read the WM8750 register space when we @@ -53,6 +48,13 @@ static const u16 wm8750_reg[] = { 0x0079, 0x0079, 0x0079, /* 40 */ }; +/* codec private data */ +struct wm8750_priv { + unsigned int sysclk; + struct snd_soc_codec codec; + u16 reg_cache[ARRAY_SIZE(wm8750_reg)]; +}; + #define wm8750_reset(c) snd_soc_write(c, WM8750_RESET, 0) /* @@ -695,25 +697,90 @@ static int wm8750_resume(struct platform_device *pdev) return 0; } +static struct snd_soc_codec *wm8750_codec; + +static int wm8750_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (!wm8750_codec) { + dev_err(&pdev->dev, "WM8750 codec not yet registered\n"); + return -EINVAL; + } + + socdev->card->codec = wm8750_codec; + codec = wm8750_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8750: failed to create pcms\n"); + goto err; + } + + snd_soc_add_controls(codec, wm8750_snd_controls, + ARRAY_SIZE(wm8750_snd_controls)); + wm8750_add_widgets(codec); + + return 0; + +err: + return ret; +} + +/* power down chip */ +static int wm8750_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_wm8750 = { + .probe = wm8750_probe, + .remove = wm8750_remove, + .suspend = wm8750_suspend, + .resume = wm8750_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); + /* * initialise the WM8750 driver * register the mixer and dsp interfaces with the kernel */ -static int wm8750_init(struct snd_soc_device *socdev, - enum snd_soc_control_type control) +static int wm8750_register(struct wm8750_priv *wm8750, + enum snd_soc_control_type control) { - struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_codec *codec = &wm8750->codec; int reg, ret = 0; + if (wm8750_codec) { + dev_err(codec->dev, "Multiple WM8750 devices not supported\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->name = "WM8750"; codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_STANDBY; codec->set_bias_level = wm8750_set_bias_level; codec->dai = &wm8750_dai; codec->num_dai = 1; - codec->reg_cache_size = ARRAY_SIZE(wm8750_reg); - codec->reg_cache = kmemdup(wm8750_reg, sizeof(wm8750_reg), GFP_KERNEL); - if (codec->reg_cache == NULL) - return -ENOMEM; + codec->private_data = wm8750; + codec->reg_cache_size = ARRAY_SIZE(wm8750->reg_cache) + 1; + codec->reg_cache = &wm8750->reg_cache; + codec->private_data = wm8750; + + memcpy(codec->reg_cache, wm8750_reg, sizeof(wm8750->reg_cache)); ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); if (ret < 0) { @@ -727,13 +794,6 @@ static int wm8750_init(struct snd_soc_device *socdev, goto err; } - /* register pcms */ - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if (ret < 0) { - printk(KERN_ERR "wm8750: failed to create pcms\n"); - goto err; - } - /* charge output caps */ wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -755,19 +815,37 @@ static int wm8750_init(struct snd_soc_device *socdev, reg = snd_soc_read(codec, WM8750_RINVOL); snd_soc_write(codec, WM8750_RINVOL, reg | 0x0100); - snd_soc_add_controls(codec, wm8750_snd_controls, - ARRAY_SIZE(wm8750_snd_controls)); - wm8750_add_widgets(codec); - return ret; + wm8750_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_dais(&wm8750_dai, 1); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); err: - kfree(codec->reg_cache); + kfree(wm8750); return ret; } -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ -static struct snd_soc_device *wm8750_socdev; +static void wm8750_unregister(struct wm8750_priv *wm8750) +{ + wm8750_set_bias_level(&wm8750->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dais(&wm8750_dai, 1); + snd_soc_unregister_codec(&wm8750->codec); + kfree(wm8750); + wm8750_codec = NULL; +} #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) @@ -781,24 +859,26 @@ static struct snd_soc_device *wm8750_socdev; static int wm8750_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct snd_soc_device *socdev = wm8750_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; + struct snd_soc_codec *codec; + struct wm8750_priv *wm8750; - i2c_set_clientdata(i2c, codec); - codec->control_data = i2c; + wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); + if (wm8750 == NULL) + return -ENOMEM; - ret = wm8750_init(socdev, SND_SOC_I2C); - if (ret < 0) - pr_err("failed to initialise WM8750\n"); + codec = &wm8750->codec; + codec->control_data = i2c; + i2c_set_clientdata(i2c, wm8750); - return ret; + codec->dev = &i2c->dev; + + return wm8750_register(wm8750, SND_SOC_I2C); } static int wm8750_i2c_remove(struct i2c_client *client) { - struct snd_soc_codec *codec = i2c_get_clientdata(client); - kfree(codec->reg_cache); + struct wm8750_priv *wm8750 = i2c_get_clientdata(client); + wm8750_unregister(wm8750); return 0; } @@ -817,66 +897,31 @@ static struct i2c_driver wm8750_i2c_driver = { .remove = wm8750_i2c_remove, .id_table = wm8750_i2c_id, }; - -static int wm8750_add_i2c_device(struct platform_device *pdev, - const struct wm8750_setup_data *setup) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - int ret; - - ret = i2c_add_driver(&wm8750_i2c_driver); - if (ret != 0) { - dev_err(&pdev->dev, "can't add i2c driver\n"); - return ret; - } - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = setup->i2c_address; - strlcpy(info.type, "wm8750", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(setup->i2c_bus); - if (!adapter) { - dev_err(&pdev->dev, "can't get i2c adapter %d\n", - setup->i2c_bus); - goto err_driver; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", - (unsigned int)info.addr); - goto err_driver; - } - - return 0; - -err_driver: - i2c_del_driver(&wm8750_i2c_driver); - return -ENODEV; -} #endif #if defined(CONFIG_SPI_MASTER) static int __devinit wm8750_spi_probe(struct spi_device *spi) { - struct snd_soc_device *socdev = wm8750_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; + struct snd_soc_codec *codec; + struct wm8750_priv *wm8750; + + wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); + if (wm8750 == NULL) + return -ENOMEM; + codec = &wm8750->codec; codec->control_data = spi; + codec->dev = &spi->dev; - ret = wm8750_init(socdev, SND_SOC_SPI); - if (ret < 0) - dev_err(&spi->dev, "failed to initialise WM8750\n"); + dev_set_drvdata(&spi->dev, wm8750); - return ret; + return wm8750_register(wm8750, SND_SOC_SPI); } static int __devexit wm8750_spi_remove(struct spi_device *spi) { + struct wm8750_priv *wm8750 = dev_get_drvdata(&spi->dev); + wm8750_unregister(wm8750); return 0; } @@ -891,93 +936,31 @@ static struct spi_driver wm8750_spi_driver = { }; #endif -static int wm8750_probe(struct platform_device *pdev) +static int __init wm8750_modinit(void) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct wm8750_setup_data *setup = socdev->codec_data; - struct snd_soc_codec *codec; - struct wm8750_priv *wm8750; int ret; - - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - - wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); - if (wm8750 == NULL) { - kfree(codec); - return -ENOMEM; - } - - codec->private_data = wm8750; - socdev->card->codec = codec; - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - wm8750_socdev = socdev; - - ret = -ENODEV; - #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - ret = wm8750_add_i2c_device(pdev, setup); - } + ret = i2c_add_driver(&wm8750_i2c_driver); + if (ret != 0) + pr_err("Failed to register WM8750 I2C driver: %d\n", ret); #endif #if defined(CONFIG_SPI_MASTER) - if (setup->spi) { - ret = spi_register_driver(&wm8750_spi_driver); - if (ret != 0) - printk(KERN_ERR "can't add spi driver"); - } + ret = spi_register_driver(&wm8750_spi_driver); + if (ret != 0) + pr_err("Failed to register WM8750 SPI driver: %d\n", ret); #endif - - if (ret != 0) { - kfree(codec->private_data); - kfree(codec); - } - return ret; + return 0; } +module_init(wm8750_modinit); -/* power down chip */ -static int wm8750_remove(struct platform_device *pdev) +static void __exit wm8750_exit(void) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - - if (codec->control_data) - wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8750_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) spi_unregister_driver(&wm8750_spi_driver); #endif - kfree(codec->private_data); - kfree(codec); - - return 0; -} - -struct snd_soc_codec_device soc_codec_dev_wm8750 = { - .probe = wm8750_probe, - .remove = wm8750_remove, - .suspend = wm8750_suspend, - .resume = wm8750_resume, -}; -EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); - -static int __init wm8750_modinit(void) -{ - return snd_soc_register_dai(&wm8750_dai); -} -module_init(wm8750_modinit); - -static void __exit wm8750_exit(void) -{ - snd_soc_unregister_dai(&wm8750_dai); } module_exit(wm8750_exit); -- 1.7.0

3 5

[alsa-devel] [PATCH 5/5] Echoaudio - add suspend/resume support
by Oliver Lupton 04 Apr '10

04 Apr '10

Just thought I'd mention that I've tested these five patches applied to a fedora 12 2.6.32.9 kernel with an Indigo IO card and it appears to now work fine when resuming from standby (which it didn't previously -- I had to remove/reinsert the card and close all applications with the device open) Cheers, Olli

2 1

[alsa-devel] [RFC][PATCH] ASoC: OMAP: Fix capture pointer handling for OMAP1510 to work correctly with recent ALSA PCM code
by Janusz Krzysztofik 02 Apr '10

02 Apr '10

With recent (2.6.34) chnages in PCM handling, capture stopped working on my OMAP1510 based Amstrad Delta videophone. Using 2.6.34-rc2, I was able to correct the problem in 3 different ways: 1. reverting commit 7b3a177b0d4f92b3431b8dca777313a07533a710, 2. enabling additional jiffies check with echo 4 >/proc/asound/card0/pcm0c0/xrun_debug 3. applying the patch below. Since I wasn't able to reproduce the problem on my i686 PC, I guess the problem is probably machine specific. The patch reuses the method for software emulation of missing hardware pointer, already implemented for playback on OMAP1510. It's possible that event if a hardware pointer is available for capture on this machine, its behaviour may be not compatible with what upper layer expects. If you think the problem may be more general and should be solved differently, on a higher level, I can try to work more on it if you give me a hint. If the patch gets accepted, I suggest it goes as a fix in the current release cycle. Created and tested against linux-2.6.34-rc2. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- git/sound/soc/omap/omap-pcm.c.orig 2010-03-25 16:00:13.000000000 +0100 +++ git/sound/soc/omap/omap-pcm.c 2010-03-28 05:16:42.000000000 +0200 @@ -60,12 +60,11 @@ static void omap_pcm_dma_irq(int ch, u16 struct omap_runtime_data *prtd = runtime->private_data; unsigned long flags; - if ((cpu_is_omap1510()) && - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) { + if ((cpu_is_omap1510())) { /* * OMAP1510 doesn't fully support DMA progress counter * and there is no software emulation implemented yet, - * so have to maintain our own playback progress counter + * so have to maintain our own progress counters * that can be used by omap_pcm_pointer() instead. */ spin_lock_irqsave(&prtd->lock, flags); @@ -189,8 +188,7 @@ static int omap_pcm_prepare(struct snd_p dma_params.frame_count = runtime->periods; omap_set_dma_params(prtd->dma_ch, &dma_params); - if ((cpu_is_omap1510()) && - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) + if ((cpu_is_omap1510())) omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ | OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ); else @@ -248,14 +246,15 @@ static snd_pcm_uframes_t omap_pcm_pointe dma_addr_t ptr; snd_pcm_uframes_t offset; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (cpu_is_omap1510()) { + offset = prtd->period_index * runtime->period_size; + } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { ptr = omap_get_dma_dst_pos(prtd->dma_ch); offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); - } else if (!(cpu_is_omap1510())) { + } else { ptr = omap_get_dma_src_pos(prtd->dma_ch); offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); - } else - offset = prtd->period_index * runtime->period_size; + } if (offset >= runtime->buffer_size) offset = 0;

3 8

[alsa-devel] [PATCH 4/4] ASoC: enable wm8753 in aspenite
by Haojian Zhuang 02 Apr '10

02 Apr '10

2 3

[alsa-devel] alsactl restore: unknown hardware: ymf724f
by Angel Tsankov 02 Apr '10

02 Apr '10

Hello, I run 'alsactl restore' on a machine with 2 sound cards -- a built-in Intel Corporation 82801EB/ER (ICH5/ICH5R) AC'97 Audio Controller (rev 02) and a non-built-in Yamaha Corporation YMF-724F [DS-1 Audio Controller] (rev 03) -- and get the following message: Unknown hardware: "YMF724F" "SigmaTel STAC9700,83,84" "AC97a:83847600" "0x1073" "0x000d" Hardware is initialized using a guess method As a consequence the volume levels of the Yamaha card do not get restored to the levels stored in /etc/asound.state. The volume levels of the built-in card however are properly restored. The asound.state file has been created by executing 'alsactl store'. The kernel has been built with support for ALSA. I've built and installed the kernel modules for both cards (not the ones in the alsa-driver package but those that come with kernel version 2.6.30.2). Any ideas why alsactl cannot find the hardware it has previously identified as "YMF724F", "SigmaTel STAC9700,83,84", and so on? Regards, Angel Tsankov

4 26

[alsa-devel] Memory corruption in ASoC
by Daniel Mack 01 Apr '10

01 Apr '10

Hi, I've been hunting a serious memory corruption in the ASoC core for some time now and finally managed to figure out where the actual problem lies. We're currently using a 2.6.32.10, on a PXA303 based series of devices (see sound/soc/pxa/raumfeld.c). So here's the demo program, reduced to the max: http://caiaq.de/download/tmp/alsa.c The thread launched to call getifaddrs() every 10ms is actually just there because a) this was the way the bug was originally triggered in our application and b) this seems to be a good way to put the kernel's memory management under fire, so you don't have to wait long for a crash. Important to mention is that the bug is only triggered when snd_pcm_set_params() is called for both streams, and only if more than one device is opened. This sources smash my kernel more or less instantly with a backtrace similar to the one at the end of this mail. You might also end up getting hung tasks, and the kernel detector which complains about them after awhile also points out code somewhere in the netlink area (which is of course innocent, we clearly deal with a memory corruption here). So - look at these simplified sniplets: // called for stream #0 pxa_ssp_hw_params(...) { // sound/soc/pxa/pxa-ssp.c ... cpu_dai->dma_data = ssp_get_dma_params(...) ... } // called for stream #1 pxa_ssp_startup(...) { // sound/soc/pxa/pxa-ssp.c ... if (cpu_dai->dma_data) { kfree(cpu_dai->dma_data); ... } } // called for stream #0 __pxa2xx_pcm_hw_free(...) { // sound/arm/pxa2xx-pcm-lib.c ... if (rtd && rtd->params && rtd->params->drcmr) *rtd->params->drcmr = 0; // <-- BOOM ... } So the first stream (PLAYBACK) already exported its dma_data which is now freed by the code initializing the second stream (CAPTURE). This corrupts all existing users of course, and in this particular case, the cleanup in __pxa2xx_pcm_hw_free() dereferences a pointer which is bogus. What I really don't understand is why this didn't crash a lot earlier for many more users. So how is this supposed to be fixed? Should dma_data become a member of some per-stream instance? I believe that also other platforms than PXA are actually affected - am I right? Daniel [ 42.158232] Unable to handle kernel NULL pointer dereference at virtual address 00000008 [ 42.166289] pgd = c7f14000 [ 42.168970] [00000008] *pgd=a7eff031, *pte=00000000, *ppte=00000000 [ 42.175187] Internal error: Oops: 17 [#1] [ 42.179163] last sysfs file: /sys/devices/virtual/vc/vcsa1/dev [ 42.184950] Modules linked in: [ 42.187983] CPU: 0 Not tainted (2.6.32.10 #99) [ 42.192769] PC is at netlink_dump_start+0x154/0x190 [ 42.197609] LR is at 0x306874 [ 42.200552] pc : [<c029dc38>] lr : [<00306874>] psr: 20000093 [ 42.200562] sp : c7ecdd10 ip : 00000000 fp : c053c648 [ 42.211954] r10: c7844000 r9 : c7ec8200 r8 : 0000045e [ 42.217141] r7 : c7edc140 r6 : c7ec8600 r5 : 00000000 r4 : 00000000 [ 42.223620] r3 : 20000093 r2 : 20000013 r1 : c79df600 r0 : 0000006c [ 42.230102] Flags: nzCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment user [ 42.237276] Control: 0000397f Table: a7f14018 DAC: 00000015 [ 42.242979] Process alsa (pid: 1118, stack limit = 0xc7ecc278) [ 42.248769] Stack: (0xc7ecdd10 to 0xc7ece000) [ 42.253099] dd00: c0296b94 c7ec8200 00000006 00000000 [ 42.261227] dd20: c7ec8600 c0298118 00000000 c7ecde40 c7ecdd5c c7ed8a80 c005030c c7ec8200 [ 42.269359] dd40: c7ed8a80 c0298040 c7ecdd7c c7ec8600 00000000 00000038 c053c648 c029ea10 [ 42.277493] dd60: c7ed8a80 00000014 c7ed8a80 c0298030 c7844000 c029e6e0 00000014 7fffffff [ 42.285625] dd80: 00000014 00000000 c7ed8a80 c7ec8600 c7ecdf5c c7672580 c7ecde80 00000000 [ 42.293748] dda0: 00000014 c029efb0 00000000 c005118c 00000000 00000000 0000045e 00000000 [ 42.301872] ddc0: 00000000 00000000 00000000 00000000 00000000 c7ecdf5c 00000014 be1ff9ec [ 42.309997] dde0: c7ecc000 0000000c be1ffa84 c027f730 c0069880 c7ecddf4 00000000 00000001 [ 42.318120] de00: ffffffff 00000000 00000000 00000000 00000000 00000000 c7a10f00 00000000 [ 42.326247] de20: 00000000 00000000 be1ff9ec c7a10f00 c0069880 c7ecde34 c7ecde34 00120000 [ 42.334379] de40: 00000014 be1ff9ec c7ecde80 00001000 c7672580 c0050518 c7ecdd68 c7ecdf5c [ 42.342503] de60: 00000000 be1ffa2c c7672580 be1ffa28 be1ffa44 be1ffa44 0000000c 00000000 [ 42.350628] de80: 0000000c c02802c4 c7672580 00000014 c7672580 c0280420 c7ecddb8 c7ecdf5c [ 42.358751] dea0: 00000000 00000000 c027ea8c 00000000 c7ecdedc 00000014 be1ff9ec c7ecdedc [ 42.366876] dec0: 0000000c c7672580 00000000 c7ecdedc 00000014 c027fa08 c00440c4 00000010 [ 42.375001] dee0: 00000000 00000000 0000000c c02802c4 0000000c c7672580 c7ecdf04 c0280690 [ 42.383124] df00: c7ecdf08 00000010 0000045e 00000000 00000000 c76747d8 c7674780 be1fea1c [ 42.391249] df20: 00000fec c7ecdf4c 00000005 c7ec9f80 c7672580 00000000 c03fd590 c027f3d8 [ 42.399373] df40: c03fd590 c0532ab0 c7e8ad80 00000000 00000000 c047c88f 00000005 c7ecdedc [ 42.407497] df60: 0000000c c7ecdf78 00000001 00000000 00000000 00000000 be1ff9ec 00000000 [ 42.415620] df80: 00000001 00000000 00000000 be1ff9ec 0000000c 0000000c 00000122 c00440c4 [ 42.423746] dfa0: 00000000 c0043f40 be1ff9ec 0000000c 00000005 be1ff9d8 00000014 00000000 [ 42.431870] dfc0: be1ff9ec 0000000c 0000000c 00000122 be1ff9ec 00001000 00000000 be1ffa84 [ 42.439995] dfe0: 00000000 be1ff9c0 4011e8a4 40123848 60000010 00000005 00000000 00000000 [ 42.448139] [<c029dc38>] (netlink_dump_start+0x154/0x190) from [<c0298118>] (rtnetlink_rcv_msg+0xd8/0x220) [ 42.457739] [<c0298118>] (rtnetlink_rcv_msg+0xd8/0x220) from [<c029ea10>] (netlink_rcv_skb+0x4c/0xb0) [ 42.466904] [<c029ea10>] (netlink_rcv_skb+0x4c/0xb0) from [<c0298030>] (rtnetlink_rcv+0x1c/0x2c) [ 42.475640] [<c0298030>] (rtnetlink_rcv+0x1c/0x2c) from [<c029e6e0>] (netlink_unicast+0x22c/0x2d0) [ 42.484550] [<c029e6e0>] (netlink_unicast+0x22c/0x2d0) from [<c029efb0>] (netlink_sendmsg+0x260/0x274) [ 42.493806] [<c029efb0>] (netlink_sendmsg+0x260/0x274) from [<c027f730>] (sock_sendmsg+0xa0/0xc4) [ 42.502630] [<c027f730>] (sock_sendmsg+0xa0/0xc4) from [<c027fa08>] (sys_sendto+0xb0/0xd4) [ 42.510866] [<c027fa08>] (sys_sendto+0xb0/0xd4) from [<c0043f40>] (ret_fast_syscall+0x0/0x28) [ 42.519344] Code: ebfffe2e e10f2000 e3823080 e121f003 (e5943008) [ 42.525679] ---[ end trace 3209e05a50ffc8c1 ]---

5 24

[alsa-devel] Regressions: MSI vs HDA-Intel
by Torsten Kaiser 01 Apr '10

01 Apr '10

I have two systems that use the alsa hda-intel driver, both show regressions in relation to MSI. System 1: Athlon(tm) X2 BE-2400 with an AMD RS690/SB600 chipset 00:14.2 Audio device [0403]: ATI Technologies Inc SBx00 Azalia (Intel HDA) [1002:4383] (Mainboard is a MSI MS-7368) After upgrading from 2.6.31 to 2.6.33 I get a delay during bootup: [ 1.155925] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 [ 1.159143] input: AT Translated Set 2 keyboard as /devices/platform/i8042/serio0/input/input2 [ 1.232595] firewire_core: created device fw0: GUID 00dc10000129c48f, S400 [ 1.252677] hda_codec: ALC888: BIOS auto-probing. [ 1.259226] HDA Intel 0000:01:05.2: PCI INT B -> GSI 19 (level, low) -> IRQ 19 [ 1.260293] HDA Intel 0000:01:05.2: irq 27 for MSI/MSI-X [ 1.314712] input: GenPS/2 Genius Mouse as /devices/platform/i8042/serio1/input/input3 [ 4.322508] hda-intel: azx_get_response timeout, switching to polling mode: last cmd=0x000f0000 [ 5.332508] hda-intel: No response from codec, disabling MSI: last cmd=0x000f0000 [ 6.342508] hda-intel: Codec #0 probe error; disabling it... [ 7.392508] hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x000f0000 [ 7.430464] ALSA device list: [ 7.431549] #0: HDA ATI SB at 0xfe7f4000 irq 16 [ 7.432656] #1: HDA ATI HDMI at 0xfe9e8000 irq 27 2.6.34-rc2 does not boot on this system, something related to the new bootmem allocator, so I can't say if this might already be fixed for 2.6.34. Should the system be blacklisted like the two Asus Athlon X2 boards, or is this some other bug? System 2: Opteron(tm) 2218 with an nVidia MCP55 chipset 00:06.1 Audio device [0403]: nVidia Corporation MCP55 High Definition Audio [10de:0371] (rev a2) (Mainboard is an ASUS KFN5-D SLI) Commit 80c43ed724797627d8f86855248c497a6161a214 disabled MSI for this system, but the MSI mode was working fine for all 2.6.33 versions I used. Should the generic blacklist disable MSI for all Athlon X2 instead of all nNvidia chipsets? If you need more information, please ask, I will try to provide anything you need. Thanks, Torsten

3 7

[alsa-devel] Handling of the DXS controls in the via82xx driver
by Soeren D. Schulze 01 Apr '10

01 Apr '10

Hi everyone, before my latest upgrade to Linux 2.6.32 (yes, sorry, it's Debian), I had four levers in my alsamixer: VIA DXS, VIA DXS 1, VIA DXS 2 and VIA DXS 3. They controlled the hardware mixing when accessing the sound device multiple times. Now they have disappeared. I think I can pretty certainly trace this down to commit 2fb930b53f513cbc4c102d415d2923a8a7091337 to Linux [http://git.kernel.org/?p=linux/kernel/git/stable/linux-2.6.32.y.git;a=commi…], which "move[s] DXS volume controls to PCM interface". As I'm not an ALSA developer, I don't realize the intended effect of this change, but the actual effect is that I cannot change the DXS volumes any more. mplayer still controls the global PCM volume. Well, in fact, there is one way. While an mplayer is running, I can do: amixer cset numid=47,iface=PCM,name='PCM Playback Volume' 24,24 When restarting mplayer, the volume is reset to 31,31 (maximum). Commit 3d00941371a765779c4e3509214c7e5793cce1fe [http://git.kernel.org/?p=linux/kernel/git/stable/linux-2.6.33.y.git;a=commi…] seems to be responsible for this. The bad thing about it is that 31,31 is a really bad setting for my poorly engineered sound card. It distorts a lot. I suspect that the real *bug* is that mplayer does not use the per-channel DXS controls even though it's supposed to (which seems to be why the DXS controls are called "PCM Playback Volume"). Personally, rather than fixing this bug, I would just revert to the former state. I was always happy with all the DXS levers set to 75%. Thank you Sören

3 6

[PATCH] ASoC: split pxa ssp for reusing code
by Haojian Zhuang 31 Mar '10

31 Mar '10

Since basic SSP features are shared between PXA2xx and PXA168, the difference is focused on clock generating. Now split ssp code into two parts. One is for general ssp feature. The other is for pxa2xx parts. Update the SSP timing configuration in hw_params(). The SSP timing is verified on I2S and Left J interface of PXA168 with 16-bit sample stream. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/Kconfig | 8 +- sound/soc/pxa/Makefile | 4 +- sound/soc/pxa/magician.c | 38 ++-- sound/soc/pxa/pxa-ssp.c | 728 +++++++++++--------------------------------- sound/soc/pxa/pxa-ssp.h | 52 ++-- sound/soc/pxa/pxa2xx-ssp.c | 318 +++++++++++++++++++ sound/soc/pxa/pxa2xx-ssp.h | 48 +++ sound/soc/pxa/raumfeld.c | 14 +- sound/soc/pxa/zylonite.c | 6 +- 9 files changed, 603 insertions(+), 613 deletions(-) create mode 100644 sound/soc/pxa/pxa2xx-ssp.c create mode 100644 sound/soc/pxa/pxa2xx-ssp.h diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 78e6121..7be1d5f 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -21,7 +21,7 @@ config SND_PXA2XX_SOC_AC97 config SND_PXA2XX_SOC_I2S tristate -config SND_PXA_SOC_SSP +config SND_PXA2XX_SOC_SSP tristate select PXA_SSP @@ -113,7 +113,7 @@ config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE select SND_PXA2XX_SOC_AC97 - select SND_PXA_SOC_SSP + select SND_PXA2XX_SOC_SSP select SND_SOC_WM9713 help Say Y if you want to add support for SoC audio on the @@ -122,7 +122,7 @@ config SND_SOC_ZYLONITE config SND_SOC_RAUMFELD tristate "SoC Audio support Raumfeld audio adapter" depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) - select SND_PXA_SOC_SSP + select SND_PXA2XX_SOC_SSP select SND_SOC_CS4270 select SND_SOC_AK4104 help @@ -132,7 +132,7 @@ config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" depends on SND_PXA2XX_SOC && MACH_MAGICIAN select SND_PXA2XX_SOC_I2S - select SND_PXA_SOC_SSP + select SND_PXA2XX_SOC_SSP select SND_SOC_UDA1380 help Say Y if you want to add support for SoC audio on the diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index f3e08fd..33c1579 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -2,12 +2,12 @@ snd-soc-pxa2xx-objs := pxa2xx-pcm.o snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o -snd-soc-pxa-ssp-objs := pxa-ssp.o +snd-soc-pxa2xx-ssp-objs := pxa-ssp.o pxa2xx-ssp.o obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o -obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o +obj-$(CONFIG_SND_PXA2XX_SOC_SSP) += snd-soc-pxa2xx-ssp.o # PXA Machine Support snd-soc-corgi-objs := corgi.o diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 4c8d99a..da6ff83 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -34,7 +34,7 @@ #include "../codecs/uda1380.h" #include "pxa2xx-pcm.h" #include "pxa2xx-i2s.h" -#include "pxa-ssp.h" +#include "pxa2xx-ssp.h" #define MAGICIAN_MIC 0 #define MAGICIAN_MIC_EXT 1 @@ -89,7 +89,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int acps, acds, width, rate; - unsigned int div4 = PXA_SSP_CLK_SCDB_4; + unsigned int div4 = PXA2XX_SSP_CLK_SCDB_4; int ret = 0; rate = params_rate(params); @@ -106,11 +106,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_16; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_16; break; default: /* 32 */ /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_8; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_8; } break; case 11025: @@ -118,11 +118,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_4; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_4; break; default: /* 32 */ /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; } break; case 22050: @@ -130,11 +130,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; break; default: /* 32 */ /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_1; } break; case 44100: @@ -142,11 +142,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; break; default: /* 32 */ /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_1; } break; case 48000: @@ -154,11 +154,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; break; default: /* 32 */ /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_1; } break; case 96000: @@ -167,12 +167,12 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_1; break; default: /* 32 */ /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - div4 = PXA_SSP_CLK_SCDB_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; + div4 = PXA2XX_SSP_CLK_SCDB_1; break; } break; @@ -195,20 +195,20 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, return ret; /* set audio clock as clock source */ - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_SSP_CLK_AUDIO, 0, SND_SOC_CLOCK_OUT); if (ret < 0) return ret; /* set the SSP audio system clock ACDS divider */ ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_ACDS, acds); + PXA2XX_SSP_AUDIO_DIV_ACDS, acds); if (ret < 0) return ret; /* set the SSP audio system clock SCDB divider4 */ ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_SCDB, div4); + PXA2XX_SSP_AUDIO_DIV_SCDB, div4); if (ret < 0) return ret; @@ -427,7 +427,7 @@ static struct snd_soc_dai_link magician_dai[] = { { .name = "uda1380", .stream_name = "UDA1380 Playback", - .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .cpu_dai = &pxa2xx_ssp_dai[PXA2XX_DAI_SSP1], .codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK], .init = magician_uda1380_init, .ops = &magician_playback_ops, diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index ed54bef..bd5e800 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -5,6 +5,9 @@ * Author: Liam Girdwood * Mark Brown <broonie(a)opensource.wolfsonmicro.com> * + * Copyright 2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.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 @@ -14,44 +17,20 @@ * o Test network mode for > 16bit sample size */ -#include <linux/init.h> -#include <linux/module.h> -#include <linux/platform_device.h> #include <linux/clk.h> -#include <linux/io.h> - -#include <asm/irq.h> - #include <sound/core.h> +#include <sound/soc.h> #include <sound/pcm.h> -#include <sound/initval.h> #include <sound/pcm_params.h> -#include <sound/soc.h> #include <sound/pxa2xx-lib.h> #include <mach/hardware.h> #include <mach/dma.h> -#include <mach/audio.h> #include <plat/ssp.h> #include "pxa2xx-pcm.h" #include "pxa-ssp.h" -/* - * SSP audio private data - */ -struct ssp_priv { - struct ssp_device *ssp; - unsigned int sysclk; - int dai_fmt; -#ifdef CONFIG_PM - uint32_t cr0; - uint32_t cr1; - uint32_t to; - uint32_t psp; -#endif -}; - static void dump_registers(struct ssp_device *ssp) { dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n", @@ -63,7 +42,7 @@ static void dump_registers(struct ssp_device *ssp) pxa_ssp_read_reg(ssp, SSACD)); } -static void pxa_ssp_enable(struct ssp_device *ssp) +void pxa_ssp_enable(struct ssp_device *ssp) { uint32_t sscr0; @@ -71,7 +50,7 @@ static void pxa_ssp_enable(struct ssp_device *ssp) __raw_writel(sscr0, ssp->mmio_base + SSCR0); } -static void pxa_ssp_disable(struct ssp_device *ssp) +void pxa_ssp_disable(struct ssp_device *ssp) { uint32_t sscr0; @@ -84,7 +63,7 @@ struct pxa2xx_pcm_dma_data { char name[20]; }; -static struct pxa2xx_pcm_dma_params * +struct pxa2xx_pcm_dma_params * pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) { struct pxa2xx_pcm_dma_data *dma; @@ -105,6 +84,102 @@ pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) return &dma->params; } +EXPORT_SYMBOL_GPL(pxa_ssp_get_dma_params); + +/* + * Set up the SSP DAI format. + * The SSP Port must be inactive before calling this function as the + * physical interface format is changed. + */ +static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + u32 sscr0; + u32 sscr1; + u32 sspsp; + + /* check if we need to change anything at all */ + if (priv->dai_fmt == fmt) + return 0; + + pxa_ssp_disable(ssp); + + /* reset port settings */ + sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & + (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); + sspsp = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + sscr1 |= SSCR1_SCLKDIR; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + sspsp |= SSPSP_SFRMP; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + case SND_SOC_DAIFMT_IB_IF: + sspsp |= SSPSP_SCMODE(2); + break; + case SND_SOC_DAIFMT_IB_NF: + sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + sscr0 |= SSCR0_PSP; + sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; + /* See hw_params() */ + break; + + case SND_SOC_DAIFMT_LEFT_J: + sscr0 |= SSCR0_PSP; + sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; + /* See hw_params() */ + break; + + case SND_SOC_DAIFMT_DSP_A: + sspsp |= SSPSP_FSRT; + case SND_SOC_DAIFMT_DSP_B: + sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; + break; + + default: + return -EINVAL; + } + + pxa_ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR1, sscr1); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); + pxa_ssp_enable(ssp); + + /* Since we are configuring the timings for the format by hand + * we have to defer some things until hw_params() where we + * know parameters like the sample size. + */ + priv->dai_fmt = fmt; + + dump_registers(ssp); + + return 0; +} static int pxa_ssp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) @@ -193,212 +268,88 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) #define pxa_ssp_resume NULL #endif -/** - * ssp_set_clkdiv - set SSP clock divider - * @div: serial clock rate divider - */ -static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) -{ - u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { - sscr0 &= ~0x0000ff00; - sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ - } else { - sscr0 &= ~0x000fff00; - sscr0 |= (div - 1) << 8; /* 1..4096 */ - } - pxa_ssp_write_reg(ssp, SSCR0, sscr0); -} - -/** - * pxa_ssp_get_clkdiv - get SSP clock divider - */ -static u32 pxa_ssp_get_scr(struct ssp_device *ssp) -{ - u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - u32 div; - - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) - div = ((sscr0 >> 8) & 0xff) * 2 + 2; - else - div = ((sscr0 >> 8) & 0xfff) + 1; - return div; -} - /* - * Set the SSP ports SYSCLK. + * While configuring SSP timing, two formula on SSP clocks should be followed. + * + * 1) (frame delay + frame width) <= (start delay + dummy start + * + data size + dummy stop) + * + * 2) 1 <= frame width < (dummy start + data size + dummy stop) */ -static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) +static int pxa_ssp_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_dai *cpu_dai = rtd->dai->cpu_dai; struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->ssp; - int val; + int width = snd_pcm_format_physical_width(params_format(params)); + int channels = params_channels(params); + int dma_32b = 0, stream_out = 0, data_size; + u32 sscr0, sspsp; - u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & - ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + /* generate correct DMA params */ + kfree(cpu_dai->dma_data); - dev_dbg(&ssp->pdev->dev, - "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", - cpu_dai->id, clk_id, freq); + if ((width == 32) || (channels == 1)) + dma_32b = 1; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + stream_out = 1; + cpu_dai->dma_data = pxa_ssp_get_dma_params(ssp, dma_32b, stream_out); - switch (clk_id) { - case PXA_SSP_CLK_NET_PLL: - sscr0 |= SSCR0_MOD; - break; - case PXA_SSP_CLK_PLL: - /* Internal PLL is fixed */ - if (cpu_is_pxa25x()) - priv->sysclk = 1843200; - else - priv->sysclk = 13000000; - break; - case PXA_SSP_CLK_EXT: - priv->sysclk = freq; - sscr0 |= SSCR0_ECS; - break; - case PXA_SSP_CLK_NET: - priv->sysclk = freq; - sscr0 |= SSCR0_NCS | SSCR0_MOD; + /* clear selected SSP bits */ + sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); + + /* data_size should only be 16-bit or 32-bit because of DMA. + * 24-bit data is treated as 32-bit data. + */ + data_size = width * channels; + switch (data_size) { + case 16: + sscr0 |= SSCR0_DataSize(16); break; - case PXA_SSP_CLK_AUDIO: - priv->sysclk = 0; - pxa_ssp_set_scr(ssp, 1); - sscr0 |= SSCR0_ACS; + case 32: + sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); break; default: - return -ENODEV; + dev_err(&ssp->pdev->dev, "Invalid data size:%d\n", data_size); + return -EINVAL; } - /* The SSP clock must be disabled when changing SSP clock mode - * on PXA2xx. On PXA3xx it must be enabled when doing so. */ - if (!cpu_is_pxa3xx()) - clk_disable(ssp->clk); - val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; - pxa_ssp_write_reg(ssp, SSCR0, val); - if (!cpu_is_pxa3xx()) - clk_enable(ssp->clk); - - return 0; -} - -/* - * Set the SSP clock dividers. - */ -static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->ssp; - int val; - - switch (div_id) { - case PXA_SSP_AUDIO_DIV_ACDS: - val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); - pxa_ssp_write_reg(ssp, SSACD, val); - break; - case PXA_SSP_AUDIO_DIV_SCDB: - val = pxa_ssp_read_reg(ssp, SSACD); - val &= ~SSACD_SCDB; -#if defined(CONFIG_PXA3xx) - if (cpu_is_pxa3xx()) - val &= ~SSACD_SCDX8; -#endif - switch (div) { - case PXA_SSP_CLK_SCDB_1: - val |= SSACD_SCDB; - break; - case PXA_SSP_CLK_SCDB_4: - break; -#if defined(CONFIG_PXA3xx) - case PXA_SSP_CLK_SCDB_8: - if (cpu_is_pxa3xx()) - val |= SSACD_SCDX8; - else - return -EINVAL; - break; -#endif - default: - return -EINVAL; + pxa_ssp_disable(ssp); + sspsp = pxa_ssp_read_reg(ssp, SSPSP); + sspsp &= ~SSPSP_TIMING_MASK; + switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + if (channels == 1) { + sspsp |= SSPSP_DMYSTRT(1); + sspsp |= SSPSP_EDMYSTOP(((width - 1) >> 2) & 7); + sspsp |= SSPSP_DMYSTOP((width - 1) & 3); + sspsp |= SSPSP_SFRMDLY(width << 1); + sspsp |= SSPSP_SFRMWDTH(width); + } else { + /* channels == 2 */ + sspsp |= SSPSP_SFRMWDTH(width); + /* Don't set any timing while FSRT is set */ + sspsp |= SSPSP_FSRT; } - pxa_ssp_write_reg(ssp, SSACD, val); break; - case PXA_SSP_DIV_SCR: - pxa_ssp_set_scr(ssp, div); + case SND_SOC_DAIFMT_LEFT_J: + if (channels == 1) { + sspsp |= SSPSP_EDMYSTOP((width >> 2) & 7); + sspsp |= SSPSP_DMYSTOP(width & 3); + } + sspsp |= SSPSP_SFRMWDTH(width); break; - default: - return -ENODEV; } - return 0; -} - -/* - * Configure the PLL frequency pxa27x and (afaik - pxa320 only) - */ -static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) -{ - struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->ssp; - u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; - -#if defined(CONFIG_PXA3xx) - if (cpu_is_pxa3xx()) - pxa_ssp_write_reg(ssp, SSACDD, 0); -#endif - - switch (freq_out) { - case 5622000: - break; - case 11345000: - ssacd |= (0x1 << 4); - break; - case 12235000: - ssacd |= (0x2 << 4); - break; - case 14857000: - ssacd |= (0x3 << 4); - break; - case 32842000: - ssacd |= (0x4 << 4); - break; - case 48000000: - ssacd |= (0x5 << 4); - break; - case 0: - /* Disable */ - break; - - default: -#ifdef CONFIG_PXA3xx - /* PXA3xx has a clock ditherer which can be used to generate - * a wider range of frequencies - calculate a value for it. - */ - if (cpu_is_pxa3xx()) { - u32 val; - u64 tmp = 19968; - tmp *= 1000000; - do_div(tmp, freq_out); - val = tmp; - - val = (val << 16) | 64; - pxa_ssp_write_reg(ssp, SSACDD, val); - - ssacd |= (0x6 << 4); - - dev_dbg(&ssp->pdev->dev, - "Using SSACDD %x to supply %uHz\n", - val, freq_out); - break; - } -#endif - - return -EINVAL; - } + /* update SSP register at the same time */ + pxa_ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); + pxa_ssp_enable(ssp); - pxa_ssp_write_reg(ssp, SSACD, ssacd); + dump_registers(ssp); return 0; } @@ -458,213 +409,6 @@ static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai, return 0; } -/* - * Set up the SSP DAI format. - * The SSP Port must be inactive before calling this function as the - * physical interface format is changed. - */ -static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->ssp; - u32 sscr0; - u32 sscr1; - u32 sspsp; - - /* check if we need to change anything at all */ - if (priv->dai_fmt == fmt) - return 0; - - /* we can only change the settings if the port is not in use */ - if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) { - dev_err(&ssp->pdev->dev, - "can't change hardware dai format: stream is in use"); - return -EINVAL; - } - - /* reset port settings */ - sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & - (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); - sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); - sspsp = 0; - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR; - break; - case SND_SOC_DAIFMT_CBM_CFS: - sscr1 |= SSCR1_SCLKDIR; - break; - case SND_SOC_DAIFMT_CBS_CFS: - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - sspsp |= SSPSP_SFRMP; - break; - case SND_SOC_DAIFMT_NB_IF: - break; - case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SCMODE(2); - break; - case SND_SOC_DAIFMT_IB_NF: - sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - sscr0 |= SSCR0_PSP; - sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; - /* See hw_params() */ - break; - - case SND_SOC_DAIFMT_DSP_A: - sspsp |= SSPSP_FSRT; - case SND_SOC_DAIFMT_DSP_B: - sscr0 |= SSCR0_MOD | SSCR0_PSP; - sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; - break; - - default: - return -EINVAL; - } - - pxa_ssp_write_reg(ssp, SSCR0, sscr0); - pxa_ssp_write_reg(ssp, SSCR1, sscr1); - pxa_ssp_write_reg(ssp, SSPSP, sspsp); - - dump_registers(ssp); - - /* Since we are configuring the timings for the format by hand - * we have to defer some things until hw_params() where we - * know parameters like the sample size. - */ - priv->dai_fmt = fmt; - - return 0; -} - -/* - * Set the SSP audio DMA parameters and sample size. - * Can be called multiple times by oss emulation. - */ -static int pxa_ssp_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_dai *cpu_dai = rtd->dai->cpu_dai; - struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->ssp; - int chn = params_channels(params); - u32 sscr0; - u32 sspsp; - int width = snd_pcm_format_physical_width(params_format(params)); - int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf; - - /* generate correct DMA params */ - if (cpu_dai->dma_data) - kfree(cpu_dai->dma_data); - - /* Network mode with one active slot (ttsa == 1) can be used - * to force 16-bit frame width on the wire (for S16_LE), even - * with two channels. Use 16-bit DMA transfers for this case. - */ - cpu_dai->dma_data = pxa_ssp_get_dma_params(ssp, - ((chn == 2) && (ttsa != 1)) || (width == 32), - substream->stream == SNDRV_PCM_STREAM_PLAYBACK); - - /* we can only change the settings if the port is not in use */ - if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) - return 0; - - /* clear selected SSP bits */ - sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); - pxa_ssp_write_reg(ssp, SSCR0, sscr0); - - /* bit size */ - sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: -#ifdef CONFIG_PXA3xx - if (cpu_is_pxa3xx()) - sscr0 |= SSCR0_FPCKE; -#endif - sscr0 |= SSCR0_DataSize(16); - break; - case SNDRV_PCM_FORMAT_S24_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); - break; - case SNDRV_PCM_FORMAT_S32_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); - break; - } - pxa_ssp_write_reg(ssp, SSCR0, sscr0); - - switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - sspsp = pxa_ssp_read_reg(ssp, SSPSP); - - if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) { - /* This is a special case where the bitclk is 64fs - * and we're not dealing with 2*32 bits of audio - * samples. - * - * The SSP values used for that are all found out by - * trying and failing a lot; some of the registers - * needed for that mode are only available on PXA3xx. - */ - -#ifdef CONFIG_PXA3xx - if (!cpu_is_pxa3xx()) - return -EINVAL; - - sspsp |= SSPSP_SFRMWDTH(width * 2); - sspsp |= SSPSP_SFRMDLY(width * 4); - sspsp |= SSPSP_EDMYSTOP(3); - sspsp |= SSPSP_DMYSTOP(3); - sspsp |= SSPSP_DMYSTRT(1); -#else - return -EINVAL; -#endif - } else { - /* The frame width is the width the LRCLK is - * asserted for; the delay is expressed in - * half cycle units. We need the extra cycle - * because the data starts clocking out one BCLK - * after LRCLK changes polarity. - */ - sspsp |= SSPSP_SFRMWDTH(width + 1); - sspsp |= SSPSP_SFRMDLY((width + 1) * 2); - sspsp |= SSPSP_DMYSTRT(1); - } - - pxa_ssp_write_reg(ssp, SSPSP, sspsp); - break; - default: - break; - } - - /* When we use a network mode, we always require TDM slots - * - complain loudly and fail if they've not been set up yet. - */ - if ((sscr0 & SSCR0_MOD) && !ttsa) { - dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); - return -EINVAL; - } - - dump_registers(ssp); - - return 0; -} - static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -760,128 +504,22 @@ static void pxa_ssp_remove(struct platform_device *pdev, pxa_ssp_free(priv->ssp); } -#define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ - SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) - -#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) - -static struct snd_soc_dai_ops pxa_ssp_dai_ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, -}; - -struct snd_soc_dai pxa_ssp_dai[] = { - { - .name = "pxa2xx-ssp1", - .id = 0, - .probe = pxa_ssp_probe, - .remove = pxa_ssp_remove, - .suspend = pxa_ssp_suspend, - .resume = pxa_ssp_resume, - .playback = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .ops = &pxa_ssp_dai_ops, - }, - { .name = "pxa2xx-ssp2", - .id = 1, - .probe = pxa_ssp_probe, - .remove = pxa_ssp_remove, - .suspend = pxa_ssp_suspend, - .resume = pxa_ssp_resume, - .playback = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .ops = &pxa_ssp_dai_ops, - }, - { - .name = "pxa2xx-ssp3", - .id = 2, - .probe = pxa_ssp_probe, - .remove = pxa_ssp_remove, - .suspend = pxa_ssp_suspend, - .resume = pxa_ssp_resume, - .playback = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .ops = &pxa_ssp_dai_ops, - }, - { - .name = "pxa2xx-ssp4", - .id = 3, - .probe = pxa_ssp_probe, - .remove = pxa_ssp_remove, - .suspend = pxa_ssp_suspend, - .resume = pxa_ssp_resume, - .playback = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .ops = &pxa_ssp_dai_ops, - }, -}; -EXPORT_SYMBOL_GPL(pxa_ssp_dai); - -static int __init pxa_ssp_init(void) -{ - return snd_soc_register_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai)); -} -module_init(pxa_ssp_init); - -static void __exit pxa_ssp_exit(void) +int pxa_ssp_register_dai(struct snd_soc_dai *dai) { - snd_soc_unregister_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai)); + struct snd_soc_dai_ops *ops = dai->ops; + + ops->startup = pxa_ssp_startup; + ops->shutdown = pxa_ssp_shutdown; + ops->trigger = pxa_ssp_trigger; + ops->hw_params = pxa_ssp_hw_params; + ops->set_fmt = pxa_ssp_set_dai_fmt; + ops->set_tdm_slot = pxa_ssp_set_dai_tdm_slot; + ops->set_tristate = pxa_ssp_set_dai_tristate; + + dai->probe = pxa_ssp_probe; + dai->remove = pxa_ssp_remove; + dai->suspend = pxa_ssp_suspend; + dai->resume = pxa_ssp_resume; + + return snd_soc_register_dai(dai); } -module_exit(pxa_ssp_exit); - -/* Module information */ -MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); -MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa-ssp.h b/sound/soc/pxa/pxa-ssp.h index 91deadd..af4a09b 100644 --- a/sound/soc/pxa/pxa-ssp.h +++ b/sound/soc/pxa/pxa-ssp.h @@ -9,39 +9,25 @@ #ifndef _PXA_SSP_H #define _PXA_SSP_H -/* pxa DAI SSP IDs */ -#define PXA_DAI_SSP1 0 -#define PXA_DAI_SSP2 1 -#define PXA_DAI_SSP3 2 -#define PXA_DAI_SSP4 3 - -/* SSP clock sources */ -#define PXA_SSP_CLK_PLL 0 -#define PXA_SSP_CLK_EXT 1 -#define PXA_SSP_CLK_NET 2 -#define PXA_SSP_CLK_AUDIO 3 -#define PXA_SSP_CLK_NET_PLL 4 - -/* SSP audio dividers */ -#define PXA_SSP_AUDIO_DIV_ACDS 0 -#define PXA_SSP_AUDIO_DIV_SCDB 1 -#define PXA_SSP_DIV_SCR 2 - -/* SSP ACDS audio dividers values */ -#define PXA_SSP_CLK_AUDIO_DIV_1 0 -#define PXA_SSP_CLK_AUDIO_DIV_2 1 -#define PXA_SSP_CLK_AUDIO_DIV_4 2 -#define PXA_SSP_CLK_AUDIO_DIV_8 3 -#define PXA_SSP_CLK_AUDIO_DIV_16 4 -#define PXA_SSP_CLK_AUDIO_DIV_32 5 - -/* SSP divider bypass */ -#define PXA_SSP_CLK_SCDB_4 0 -#define PXA_SSP_CLK_SCDB_1 1 -#define PXA_SSP_CLK_SCDB_8 2 - -#define PXA_SSP_PLL_OUT 0 +/* + * SSP audio data + */ +struct ssp_priv { + struct ssp_device *ssp; + unsigned int sysclk; + int dai_fmt; +#ifdef CONFIG_PM + uint32_t cr0; + uint32_t cr1; + uint32_t to; + uint32_t psp; +#endif +}; -extern struct snd_soc_dai pxa_ssp_dai[4]; +extern void pxa_ssp_enable(struct ssp_device *ssp); +extern void pxa_ssp_disable(struct ssp_device *ssp); +extern struct pxa2xx_pcm_dma_params * +pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out); +extern int pxa_ssp_register_dai(struct snd_soc_dai *dai); #endif diff --git a/sound/soc/pxa/pxa2xx-ssp.c b/sound/soc/pxa/pxa2xx-ssp.c new file mode 100644 index 0000000..eff5334 --- /dev/null +++ b/sound/soc/pxa/pxa2xx-ssp.c @@ -0,0 +1,318 @@ +/* + * pxa2xx-ssp.c -- ALSA Soc Audio Layer + * + * Copyright 2005,2008 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * Copyright 2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.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. + * + * TODO: + * o Test network mode for > 16bit sample size + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/pxa2xx-lib.h> + +#include <mach/hardware.h> +#include <mach/dma.h> +#include <plat/ssp.h> + +#include "pxa2xx-pcm.h" +#include "pxa2xx-ssp.h" +#include "pxa-ssp.h" + +/** + * pxa_ssp_set_clkdiv - set SSP clock divider + * @div: serial clock rate divider + */ +static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) +{ + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { + sscr0 &= ~0x0000ff00; + sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ + } else { + sscr0 &= ~0x000fff00; + sscr0 |= (div - 1) << 8; /* 1..4096 */ + } + pxa_ssp_write_reg(ssp, SSCR0, sscr0); +} + +/** + * pxa_ssp_get_clkdiv - get SSP clock divider + */ +static u32 pxa_ssp_get_scr(struct ssp_device *ssp) +{ + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); + u32 div; + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) + div = ((sscr0 >> 8) & 0xff) * 2 + 2; + else + div = ((sscr0 >> 8) & 0xfff) + 1; + return div; +} + +/* + * Set the SSP ports SYSCLK. + */ +static int pxa2xx_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + int val; + + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & + ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + + dev_dbg(&ssp->pdev->dev, + "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", + cpu_dai->id, clk_id, freq); + + switch (clk_id) { + case PXA2XX_SSP_CLK_NET_PLL: + sscr0 |= SSCR0_MOD; + break; + case PXA2XX_SSP_CLK_PLL: + /* Internal PLL is fixed */ + if (cpu_is_pxa25x()) + priv->sysclk = 1843200; + else + priv->sysclk = 13000000; + break; + case PXA2XX_SSP_CLK_EXT: + priv->sysclk = freq; + sscr0 |= SSCR0_ECS; + break; + case PXA2XX_SSP_CLK_NET: + priv->sysclk = freq; + sscr0 |= SSCR0_NCS | SSCR0_MOD; + break; + case PXA2XX_SSP_CLK_AUDIO: + priv->sysclk = 0; + pxa_ssp_set_scr(ssp, 1); + sscr0 |= SSCR0_ACS; + break; + default: + return -ENODEV; + } + + /* The SSP clock must be disabled when changing SSP clock mode + * on PXA2xx. On PXA3xx it must be enabled when doing so. */ + if (!cpu_is_pxa3xx()) + clk_disable(ssp->clk); + val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; + pxa_ssp_write_reg(ssp, SSCR0, val); + if (!cpu_is_pxa3xx()) + clk_enable(ssp->clk); + + return 0; +} + +/* + * Set the SSP clock dividers. + */ +static int pxa2xx_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + int val; + + switch (div_id) { + case PXA2XX_SSP_AUDIO_DIV_ACDS: + val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); + pxa_ssp_write_reg(ssp, SSACD, val); + break; + case PXA2XX_SSP_AUDIO_DIV_SCDB: + val = pxa_ssp_read_reg(ssp, SSACD); + val &= ~SSACD_SCDB; +#if defined(CONFIG_PXA3xx) + if (cpu_is_pxa3xx()) + val &= ~SSACD_SCDX8; +#endif + switch (div) { + case PXA2XX_SSP_CLK_SCDB_1: + val |= SSACD_SCDB; + break; + case PXA2XX_SSP_CLK_SCDB_4: + break; +#if defined(CONFIG_PXA3xx) + case PXA2XX_SSP_CLK_SCDB_8: + if (cpu_is_pxa3xx()) + val |= SSACD_SCDX8; + else + return -EINVAL; + break; +#endif + default: + return -EINVAL; + } + pxa_ssp_write_reg(ssp, SSACD, val); + break; + case PXA2XX_SSP_DIV_SCR: + pxa_ssp_set_scr(ssp, div); + break; + default: + return -ENODEV; + } + + return 0; +} + +/* + * Configure the PLL frequency pxa27x and (afaik - pxa320 only) + */ +static int pxa2xx_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; + +#if defined(CONFIG_PXA3xx) + if (cpu_is_pxa3xx()) + pxa_ssp_write_reg(ssp, SSACDD, 0); +#endif + + switch (freq_out) { + case 5622000: + break; + case 11345000: + ssacd |= (0x1 << 4); + break; + case 12235000: + ssacd |= (0x2 << 4); + break; + case 14857000: + ssacd |= (0x3 << 4); + break; + case 32842000: + ssacd |= (0x4 << 4); + break; + case 48000000: + ssacd |= (0x5 << 4); + break; + case 0: + /* Disable */ + break; + + default: +#ifdef CONFIG_PXA3xx + /* PXA3xx has a clock ditherer which can be used to generate + * a wider range of frequencies - calculate a value for it. + */ + if (cpu_is_pxa3xx()) { + u32 val; + u64 tmp = 19968; + tmp *= 1000000; + do_div(tmp, freq_out); + val = tmp; + + val = (val << 16) | 64; + pxa_ssp_write_reg(ssp, SSACDD, val); + + ssacd |= (0x6 << 4); + + dev_dbg(&ssp->pdev->dev, + "Using SSACDD %x to supply %uHz\n", + val, freq_out); + break; + } +#endif + + return -EINVAL; + } + + pxa_ssp_write_reg(ssp, SSACD, ssacd); + + return 0; +} + +static struct snd_soc_dai_ops pxa2xx_ssp_dai_ops = { + .set_sysclk = pxa2xx_ssp_set_dai_sysclk, + .set_clkdiv = pxa2xx_ssp_set_dai_clkdiv, + .set_pll = pxa2xx_ssp_set_dai_pll, +}; + +#define PXA2XX_SSP_RATES SNDRV_PCM_RATE_8000_96000 +#define PXA2XX_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define PXA2XX_SSP_DAI(_id) \ +{ \ + .name = "pxa2xx-ssp", \ + .id = _id, \ + .playback = { \ + .channels_min = 1, \ + .channels_max = 8, \ + .rates = PXA2XX_SSP_RATES, \ + .formats = PXA2XX_SSP_FORMATS, \ + }, \ + .capture = { \ + .channels_min = 1, \ + .channels_max = 8, \ + .rates = PXA2XX_SSP_RATES, \ + .formats = PXA2XX_SSP_FORMATS, \ + }, \ + .ops = &pxa2xx_ssp_dai_ops, \ +} + +struct snd_soc_dai pxa2xx_ssp_dai[] = { + PXA2XX_SSP_DAI(PXA2XX_DAI_SSP1), + PXA2XX_SSP_DAI(PXA2XX_DAI_SSP2), + PXA2XX_SSP_DAI(PXA2XX_DAI_SSP3), + PXA2XX_SSP_DAI(PXA2XX_DAI_SSP4), +}; +EXPORT_SYMBOL_GPL(pxa2xx_ssp_dai); + +static int __init pxa2xx_ssp_init(void) +{ + struct snd_soc_dai *dai; + int i, ret; + + for (i = 0; i < ARRAY_SIZE(pxa2xx_ssp_dai); i++) { + dai = &pxa2xx_ssp_dai[i]; + ret = pxa_ssp_register_dai(dai); + if (ret) + return ret; + } + return ret; +} +module_init(pxa2xx_ssp_init); + +static void __exit pxa2xx_ssp_exit(void) +{ + struct snd_soc_dai *dai = NULL; + int i; + + for (i = 0; i < ARRAY_SIZE(pxa2xx_ssp_dai); i++) { + dai = &pxa2xx_ssp_dai[i]; + snd_soc_unregister_dai(dai); + } +} +module_exit(pxa2xx_ssp_exit); + +/* Module information */ +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_DESCRIPTION("PXA2xx SSP/PCM SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa2xx-ssp.h b/sound/soc/pxa/pxa2xx-ssp.h new file mode 100644 index 0000000..c24e0b5 --- /dev/null +++ b/sound/soc/pxa/pxa2xx-ssp.h @@ -0,0 +1,48 @@ + +/* + * ASoC PXA SSP port support + * + * 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 __PXA2XX_SOC_SSP_H +#define __PXA2XX_SOC_SSP_H + +/* pxa DAI SSP IDs */ +#define PXA2XX_DAI_SSP1 0 +#define PXA2XX_DAI_SSP2 1 +#define PXA2XX_DAI_SSP3 2 +#define PXA2XX_DAI_SSP4 3 + +/* SSP clock sources */ +#define PXA2XX_SSP_CLK_PLL 0 +#define PXA2XX_SSP_CLK_EXT 1 +#define PXA2XX_SSP_CLK_NET 2 +#define PXA2XX_SSP_CLK_AUDIO 3 +#define PXA2XX_SSP_CLK_NET_PLL 4 + +/* SSP audio dividers */ +#define PXA2XX_SSP_AUDIO_DIV_ACDS 0 +#define PXA2XX_SSP_AUDIO_DIV_SCDB 1 +#define PXA2XX_SSP_DIV_SCR 2 + +/* SSP ACDS audio dividers values */ +#define PXA2XX_SSP_CLK_AUDIO_DIV_1 0 +#define PXA2XX_SSP_CLK_AUDIO_DIV_2 1 +#define PXA2XX_SSP_CLK_AUDIO_DIV_4 2 +#define PXA2XX_SSP_CLK_AUDIO_DIV_8 3 +#define PXA2XX_SSP_CLK_AUDIO_DIV_16 4 +#define PXA2XX_SSP_CLK_AUDIO_DIV_32 5 + +/* SSP divider bypass */ +#define PXA2XX_SSP_CLK_SCDB_4 0 +#define PXA2XX_SSP_CLK_SCDB_1 1 +#define PXA2XX_SSP_CLK_SCDB_8 2 + +#define PXA2XX_SSP_PLL_OUT 0 + +extern struct snd_soc_dai pxa2xx_ssp_dai[]; + +#endif /* __PXA2XX_SOC_SSP_H */ diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 7e3f416..e66dc12 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -29,7 +29,7 @@ #include "../codecs/cs4270.h" #include "../codecs/ak4104.h" #include "pxa2xx-pcm.h" -#include "pxa-ssp.h" +#include "pxa2xx-ssp.h" #define GPIO_SPDIF_RESET (38) #define GPIO_MCLK_RESET (111) @@ -138,11 +138,11 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA2XX_SSP_DIV_SCR, 4); if (ret < 0) return ret; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_SSP_CLK_EXT, clk, 1); if (ret < 0) return ret; @@ -170,7 +170,7 @@ static int raumfeld_line_resume(struct platform_device *pdev) static struct snd_soc_dai_link raumfeld_line_dai = { .name = "CS4270", .stream_name = "CS4270", - .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .cpu_dai = &pxa2xx_ssp_dai[PXA2XX_DAI_SSP1], .codec_dai = &cs4270_dai, .ops = &raumfeld_cs4270_ops, }; @@ -232,11 +232,11 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA2XX_SSP_DIV_SCR, 4); if (ret < 0) return ret; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_SSP_CLK_EXT, clk, 1); if (ret < 0) return ret; @@ -250,7 +250,7 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { static struct snd_soc_dai_link raumfeld_spdif_dai = { .name = "ak4104", .stream_name = "Playback", - .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP2], + .cpu_dai = &pxa2xx_ssp_dai[PXA2XX_DAI_SSP2], .codec_dai = &ak4104_dai, .ops = &raumfeld_ak4104_ops, }; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index dd678ae..c65e0db 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -25,7 +25,7 @@ #include "../codecs/wm9713.h" #include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" -#include "pxa-ssp.h" +#include "pxa2xx-ssp.h" /* * There is a physical switch SW15 on the board which changes the MCLK @@ -125,7 +125,7 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, /* Add 1 to the width for the leading clock cycle */ pll_out = rate * (width + 1) * 8; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_SSP_CLK_AUDIO, 0, 1); if (ret < 0) return ret; @@ -176,7 +176,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { { .name = "WM9713 Voice", .stream_name = "WM9713 Voice", - .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP3], + .cpu_dai = &pxa2xx_ssp_dai[PXA2XX_DAI_SSP3], .codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE], .ops = &zylonite_voice_ops, }, -- 1.5.6.5

1 0

[PATCH] ASoC: split pxa ssp for reusing code
by Haojian Zhuang 31 Mar '10

31 Mar '10

Since basic SSP features are shared between PXA2xx and PXA168, the difference is focused on clock generating. Now split ssp code into two parts. One is for general ssp feature. The other is for pxa2xx parts. Update the SSP timing configuration in hw_params(). The SSP timing is verified on I2S and Left J interface of PXA168 with 16-bit sample stream. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/Kconfig | 8 +- sound/soc/pxa/Makefile | 4 +- sound/soc/pxa/magician.c | 38 ++-- sound/soc/pxa/pxa-ssp.c | 728 +++++++++++--------------------------------- sound/soc/pxa/pxa-ssp.h | 52 ++-- sound/soc/pxa/pxa2xx-ssp.c | 318 +++++++++++++++++++ sound/soc/pxa/pxa2xx-ssp.h | 48 +++ sound/soc/pxa/raumfeld.c | 14 +- sound/soc/pxa/zylonite.c | 6 +- 9 files changed, 603 insertions(+), 613 deletions(-) create mode 100644 sound/soc/pxa/pxa2xx-ssp.c create mode 100644 sound/soc/pxa/pxa2xx-ssp.h diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 78e6121..7be1d5f 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -21,7 +21,7 @@ config SND_PXA2XX_SOC_AC97 config SND_PXA2XX_SOC_I2S tristate -config SND_PXA_SOC_SSP +config SND_PXA2XX_SOC_SSP tristate select PXA_SSP @@ -113,7 +113,7 @@ config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE select SND_PXA2XX_SOC_AC97 - select SND_PXA_SOC_SSP + select SND_PXA2XX_SOC_SSP select SND_SOC_WM9713 help Say Y if you want to add support for SoC audio on the @@ -122,7 +122,7 @@ config SND_SOC_ZYLONITE config SND_SOC_RAUMFELD tristate "SoC Audio support Raumfeld audio adapter" depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) - select SND_PXA_SOC_SSP + select SND_PXA2XX_SOC_SSP select SND_SOC_CS4270 select SND_SOC_AK4104 help @@ -132,7 +132,7 @@ config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" depends on SND_PXA2XX_SOC && MACH_MAGICIAN select SND_PXA2XX_SOC_I2S - select SND_PXA_SOC_SSP + select SND_PXA2XX_SOC_SSP select SND_SOC_UDA1380 help Say Y if you want to add support for SoC audio on the diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index f3e08fd..33c1579 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -2,12 +2,12 @@ snd-soc-pxa2xx-objs := pxa2xx-pcm.o snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o -snd-soc-pxa-ssp-objs := pxa-ssp.o +snd-soc-pxa2xx-ssp-objs := pxa-ssp.o pxa2xx-ssp.o obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o -obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o +obj-$(CONFIG_SND_PXA2XX_SOC_SSP) += snd-soc-pxa2xx-ssp.o # PXA Machine Support snd-soc-corgi-objs := corgi.o diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 4c8d99a..da6ff83 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -34,7 +34,7 @@ #include "../codecs/uda1380.h" #include "pxa2xx-pcm.h" #include "pxa2xx-i2s.h" -#include "pxa-ssp.h" +#include "pxa2xx-ssp.h" #define MAGICIAN_MIC 0 #define MAGICIAN_MIC_EXT 1 @@ -89,7 +89,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int acps, acds, width, rate; - unsigned int div4 = PXA_SSP_CLK_SCDB_4; + unsigned int div4 = PXA2XX_SSP_CLK_SCDB_4; int ret = 0; rate = params_rate(params); @@ -106,11 +106,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_16; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_16; break; default: /* 32 */ /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_8; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_8; } break; case 11025: @@ -118,11 +118,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_4; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_4; break; default: /* 32 */ /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; } break; case 22050: @@ -130,11 +130,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; break; default: /* 32 */ /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_1; } break; case 44100: @@ -142,11 +142,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; break; default: /* 32 */ /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_1; } break; case 48000: @@ -154,11 +154,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; break; default: /* 32 */ /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_1; } break; case 96000: @@ -167,12 +167,12 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_1; break; default: /* 32 */ /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - div4 = PXA_SSP_CLK_SCDB_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; + div4 = PXA2XX_SSP_CLK_SCDB_1; break; } break; @@ -195,20 +195,20 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, return ret; /* set audio clock as clock source */ - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_SSP_CLK_AUDIO, 0, SND_SOC_CLOCK_OUT); if (ret < 0) return ret; /* set the SSP audio system clock ACDS divider */ ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_ACDS, acds); + PXA2XX_SSP_AUDIO_DIV_ACDS, acds); if (ret < 0) return ret; /* set the SSP audio system clock SCDB divider4 */ ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_SCDB, div4); + PXA2XX_SSP_AUDIO_DIV_SCDB, div4); if (ret < 0) return ret; @@ -427,7 +427,7 @@ static struct snd_soc_dai_link magician_dai[] = { { .name = "uda1380", .stream_name = "UDA1380 Playback", - .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .cpu_dai = &pxa2xx_ssp_dai[PXA2XX_DAI_SSP1], .codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK], .init = magician_uda1380_init, .ops = &magician_playback_ops, diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index ed54bef..bd5e800 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -5,6 +5,9 @@ * Author: Liam Girdwood * Mark Brown <broonie(a)opensource.wolfsonmicro.com> * + * Copyright 2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.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 @@ -14,44 +17,20 @@ * o Test network mode for > 16bit sample size */ -#include <linux/init.h> -#include <linux/module.h> -#include <linux/platform_device.h> #include <linux/clk.h> -#include <linux/io.h> - -#include <asm/irq.h> - #include <sound/core.h> +#include <sound/soc.h> #include <sound/pcm.h> -#include <sound/initval.h> #include <sound/pcm_params.h> -#include <sound/soc.h> #include <sound/pxa2xx-lib.h> #include <mach/hardware.h> #include <mach/dma.h> -#include <mach/audio.h> #include <plat/ssp.h> #include "pxa2xx-pcm.h" #include "pxa-ssp.h" -/* - * SSP audio private data - */ -struct ssp_priv { - struct ssp_device *ssp; - unsigned int sysclk; - int dai_fmt; -#ifdef CONFIG_PM - uint32_t cr0; - uint32_t cr1; - uint32_t to; - uint32_t psp; -#endif -}; - static void dump_registers(struct ssp_device *ssp) { dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n", @@ -63,7 +42,7 @@ static void dump_registers(struct ssp_device *ssp) pxa_ssp_read_reg(ssp, SSACD)); } -static void pxa_ssp_enable(struct ssp_device *ssp) +void pxa_ssp_enable(struct ssp_device *ssp) { uint32_t sscr0; @@ -71,7 +50,7 @@ static void pxa_ssp_enable(struct ssp_device *ssp) __raw_writel(sscr0, ssp->mmio_base + SSCR0); } -static void pxa_ssp_disable(struct ssp_device *ssp) +void pxa_ssp_disable(struct ssp_device *ssp) { uint32_t sscr0; @@ -84,7 +63,7 @@ struct pxa2xx_pcm_dma_data { char name[20]; }; -static struct pxa2xx_pcm_dma_params * +struct pxa2xx_pcm_dma_params * pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) { struct pxa2xx_pcm_dma_data *dma; @@ -105,6 +84,102 @@ pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) return &dma->params; } +EXPORT_SYMBOL_GPL(pxa_ssp_get_dma_params); + +/* + * Set up the SSP DAI format. + * The SSP Port must be inactive before calling this function as the + * physical interface format is changed. + */ +static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + u32 sscr0; + u32 sscr1; + u32 sspsp; + + /* check if we need to change anything at all */ + if (priv->dai_fmt == fmt) + return 0; + + pxa_ssp_disable(ssp); + + /* reset port settings */ + sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & + (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); + sspsp = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + sscr1 |= SSCR1_SCLKDIR; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + sspsp |= SSPSP_SFRMP; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + case SND_SOC_DAIFMT_IB_IF: + sspsp |= SSPSP_SCMODE(2); + break; + case SND_SOC_DAIFMT_IB_NF: + sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + sscr0 |= SSCR0_PSP; + sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; + /* See hw_params() */ + break; + + case SND_SOC_DAIFMT_LEFT_J: + sscr0 |= SSCR0_PSP; + sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; + /* See hw_params() */ + break; + + case SND_SOC_DAIFMT_DSP_A: + sspsp |= SSPSP_FSRT; + case SND_SOC_DAIFMT_DSP_B: + sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; + break; + + default: + return -EINVAL; + } + + pxa_ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR1, sscr1); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); + pxa_ssp_enable(ssp); + + /* Since we are configuring the timings for the format by hand + * we have to defer some things until hw_params() where we + * know parameters like the sample size. + */ + priv->dai_fmt = fmt; + + dump_registers(ssp); + + return 0; +} static int pxa_ssp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) @@ -193,212 +268,88 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) #define pxa_ssp_resume NULL #endif -/** - * ssp_set_clkdiv - set SSP clock divider - * @div: serial clock rate divider - */ -static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) -{ - u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { - sscr0 &= ~0x0000ff00; - sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ - } else { - sscr0 &= ~0x000fff00; - sscr0 |= (div - 1) << 8; /* 1..4096 */ - } - pxa_ssp_write_reg(ssp, SSCR0, sscr0); -} - -/** - * pxa_ssp_get_clkdiv - get SSP clock divider - */ -static u32 pxa_ssp_get_scr(struct ssp_device *ssp) -{ - u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - u32 div; - - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) - div = ((sscr0 >> 8) & 0xff) * 2 + 2; - else - div = ((sscr0 >> 8) & 0xfff) + 1; - return div; -} - /* - * Set the SSP ports SYSCLK. + * While configuring SSP timing, two formula on SSP clocks should be followed. + * + * 1) (frame delay + frame width) <= (start delay + dummy start + * + data size + dummy stop) + * + * 2) 1 <= frame width < (dummy start + data size + dummy stop) */ -static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) +static int pxa_ssp_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_dai *cpu_dai = rtd->dai->cpu_dai; struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->ssp; - int val; + int width = snd_pcm_format_physical_width(params_format(params)); + int channels = params_channels(params); + int dma_32b = 0, stream_out = 0, data_size; + u32 sscr0, sspsp; - u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & - ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + /* generate correct DMA params */ + kfree(cpu_dai->dma_data); - dev_dbg(&ssp->pdev->dev, - "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", - cpu_dai->id, clk_id, freq); + if ((width == 32) || (channels == 1)) + dma_32b = 1; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + stream_out = 1; + cpu_dai->dma_data = pxa_ssp_get_dma_params(ssp, dma_32b, stream_out); - switch (clk_id) { - case PXA_SSP_CLK_NET_PLL: - sscr0 |= SSCR0_MOD; - break; - case PXA_SSP_CLK_PLL: - /* Internal PLL is fixed */ - if (cpu_is_pxa25x()) - priv->sysclk = 1843200; - else - priv->sysclk = 13000000; - break; - case PXA_SSP_CLK_EXT: - priv->sysclk = freq; - sscr0 |= SSCR0_ECS; - break; - case PXA_SSP_CLK_NET: - priv->sysclk = freq; - sscr0 |= SSCR0_NCS | SSCR0_MOD; + /* clear selected SSP bits */ + sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); + + /* data_size should only be 16-bit or 32-bit because of DMA. + * 24-bit data is treated as 32-bit data. + */ + data_size = width * channels; + switch (data_size) { + case 16: + sscr0 |= SSCR0_DataSize(16); break; - case PXA_SSP_CLK_AUDIO: - priv->sysclk = 0; - pxa_ssp_set_scr(ssp, 1); - sscr0 |= SSCR0_ACS; + case 32: + sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); break; default: - return -ENODEV; + dev_err(&ssp->pdev->dev, "Invalid data size:%d\n", data_size); + return -EINVAL; } - /* The SSP clock must be disabled when changing SSP clock mode - * on PXA2xx. On PXA3xx it must be enabled when doing so. */ - if (!cpu_is_pxa3xx()) - clk_disable(ssp->clk); - val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; - pxa_ssp_write_reg(ssp, SSCR0, val); - if (!cpu_is_pxa3xx()) - clk_enable(ssp->clk); - - return 0; -} - -/* - * Set the SSP clock dividers. - */ -static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->ssp; - int val; - - switch (div_id) { - case PXA_SSP_AUDIO_DIV_ACDS: - val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); - pxa_ssp_write_reg(ssp, SSACD, val); - break; - case PXA_SSP_AUDIO_DIV_SCDB: - val = pxa_ssp_read_reg(ssp, SSACD); - val &= ~SSACD_SCDB; -#if defined(CONFIG_PXA3xx) - if (cpu_is_pxa3xx()) - val &= ~SSACD_SCDX8; -#endif - switch (div) { - case PXA_SSP_CLK_SCDB_1: - val |= SSACD_SCDB; - break; - case PXA_SSP_CLK_SCDB_4: - break; -#if defined(CONFIG_PXA3xx) - case PXA_SSP_CLK_SCDB_8: - if (cpu_is_pxa3xx()) - val |= SSACD_SCDX8; - else - return -EINVAL; - break; -#endif - default: - return -EINVAL; + pxa_ssp_disable(ssp); + sspsp = pxa_ssp_read_reg(ssp, SSPSP); + sspsp &= ~SSPSP_TIMING_MASK; + switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + if (channels == 1) { + sspsp |= SSPSP_DMYSTRT(1); + sspsp |= SSPSP_EDMYSTOP(((width - 1) >> 2) & 7); + sspsp |= SSPSP_DMYSTOP((width - 1) & 3); + sspsp |= SSPSP_SFRMDLY(width << 1); + sspsp |= SSPSP_SFRMWDTH(width); + } else { + /* channels == 2 */ + sspsp |= SSPSP_SFRMWDTH(width); + /* Don't set any timing while FSRT is set */ + sspsp |= SSPSP_FSRT; } - pxa_ssp_write_reg(ssp, SSACD, val); break; - case PXA_SSP_DIV_SCR: - pxa_ssp_set_scr(ssp, div); + case SND_SOC_DAIFMT_LEFT_J: + if (channels == 1) { + sspsp |= SSPSP_EDMYSTOP((width >> 2) & 7); + sspsp |= SSPSP_DMYSTOP(width & 3); + } + sspsp |= SSPSP_SFRMWDTH(width); break; - default: - return -ENODEV; } - return 0; -} - -/* - * Configure the PLL frequency pxa27x and (afaik - pxa320 only) - */ -static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) -{ - struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->ssp; - u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; - -#if defined(CONFIG_PXA3xx) - if (cpu_is_pxa3xx()) - pxa_ssp_write_reg(ssp, SSACDD, 0); -#endif - - switch (freq_out) { - case 5622000: - break; - case 11345000: - ssacd |= (0x1 << 4); - break; - case 12235000: - ssacd |= (0x2 << 4); - break; - case 14857000: - ssacd |= (0x3 << 4); - break; - case 32842000: - ssacd |= (0x4 << 4); - break; - case 48000000: - ssacd |= (0x5 << 4); - break; - case 0: - /* Disable */ - break; - - default: -#ifdef CONFIG_PXA3xx - /* PXA3xx has a clock ditherer which can be used to generate - * a wider range of frequencies - calculate a value for it. - */ - if (cpu_is_pxa3xx()) { - u32 val; - u64 tmp = 19968; - tmp *= 1000000; - do_div(tmp, freq_out); - val = tmp; - - val = (val << 16) | 64; - pxa_ssp_write_reg(ssp, SSACDD, val); - - ssacd |= (0x6 << 4); - - dev_dbg(&ssp->pdev->dev, - "Using SSACDD %x to supply %uHz\n", - val, freq_out); - break; - } -#endif - - return -EINVAL; - } + /* update SSP register at the same time */ + pxa_ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); + pxa_ssp_enable(ssp); - pxa_ssp_write_reg(ssp, SSACD, ssacd); + dump_registers(ssp); return 0; } @@ -458,213 +409,6 @@ static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai, return 0; } -/* - * Set up the SSP DAI format. - * The SSP Port must be inactive before calling this function as the - * physical interface format is changed. - */ -static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->ssp; - u32 sscr0; - u32 sscr1; - u32 sspsp; - - /* check if we need to change anything at all */ - if (priv->dai_fmt == fmt) - return 0; - - /* we can only change the settings if the port is not in use */ - if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) { - dev_err(&ssp->pdev->dev, - "can't change hardware dai format: stream is in use"); - return -EINVAL; - } - - /* reset port settings */ - sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & - (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); - sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); - sspsp = 0; - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR; - break; - case SND_SOC_DAIFMT_CBM_CFS: - sscr1 |= SSCR1_SCLKDIR; - break; - case SND_SOC_DAIFMT_CBS_CFS: - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - sspsp |= SSPSP_SFRMP; - break; - case SND_SOC_DAIFMT_NB_IF: - break; - case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SCMODE(2); - break; - case SND_SOC_DAIFMT_IB_NF: - sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - sscr0 |= SSCR0_PSP; - sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; - /* See hw_params() */ - break; - - case SND_SOC_DAIFMT_DSP_A: - sspsp |= SSPSP_FSRT; - case SND_SOC_DAIFMT_DSP_B: - sscr0 |= SSCR0_MOD | SSCR0_PSP; - sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; - break; - - default: - return -EINVAL; - } - - pxa_ssp_write_reg(ssp, SSCR0, sscr0); - pxa_ssp_write_reg(ssp, SSCR1, sscr1); - pxa_ssp_write_reg(ssp, SSPSP, sspsp); - - dump_registers(ssp); - - /* Since we are configuring the timings for the format by hand - * we have to defer some things until hw_params() where we - * know parameters like the sample size. - */ - priv->dai_fmt = fmt; - - return 0; -} - -/* - * Set the SSP audio DMA parameters and sample size. - * Can be called multiple times by oss emulation. - */ -static int pxa_ssp_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_dai *cpu_dai = rtd->dai->cpu_dai; - struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->ssp; - int chn = params_channels(params); - u32 sscr0; - u32 sspsp; - int width = snd_pcm_format_physical_width(params_format(params)); - int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf; - - /* generate correct DMA params */ - if (cpu_dai->dma_data) - kfree(cpu_dai->dma_data); - - /* Network mode with one active slot (ttsa == 1) can be used - * to force 16-bit frame width on the wire (for S16_LE), even - * with two channels. Use 16-bit DMA transfers for this case. - */ - cpu_dai->dma_data = pxa_ssp_get_dma_params(ssp, - ((chn == 2) && (ttsa != 1)) || (width == 32), - substream->stream == SNDRV_PCM_STREAM_PLAYBACK); - - /* we can only change the settings if the port is not in use */ - if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) - return 0; - - /* clear selected SSP bits */ - sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); - pxa_ssp_write_reg(ssp, SSCR0, sscr0); - - /* bit size */ - sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: -#ifdef CONFIG_PXA3xx - if (cpu_is_pxa3xx()) - sscr0 |= SSCR0_FPCKE; -#endif - sscr0 |= SSCR0_DataSize(16); - break; - case SNDRV_PCM_FORMAT_S24_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); - break; - case SNDRV_PCM_FORMAT_S32_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); - break; - } - pxa_ssp_write_reg(ssp, SSCR0, sscr0); - - switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - sspsp = pxa_ssp_read_reg(ssp, SSPSP); - - if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) { - /* This is a special case where the bitclk is 64fs - * and we're not dealing with 2*32 bits of audio - * samples. - * - * The SSP values used for that are all found out by - * trying and failing a lot; some of the registers - * needed for that mode are only available on PXA3xx. - */ - -#ifdef CONFIG_PXA3xx - if (!cpu_is_pxa3xx()) - return -EINVAL; - - sspsp |= SSPSP_SFRMWDTH(width * 2); - sspsp |= SSPSP_SFRMDLY(width * 4); - sspsp |= SSPSP_EDMYSTOP(3); - sspsp |= SSPSP_DMYSTOP(3); - sspsp |= SSPSP_DMYSTRT(1); -#else - return -EINVAL; -#endif - } else { - /* The frame width is the width the LRCLK is - * asserted for; the delay is expressed in - * half cycle units. We need the extra cycle - * because the data starts clocking out one BCLK - * after LRCLK changes polarity. - */ - sspsp |= SSPSP_SFRMWDTH(width + 1); - sspsp |= SSPSP_SFRMDLY((width + 1) * 2); - sspsp |= SSPSP_DMYSTRT(1); - } - - pxa_ssp_write_reg(ssp, SSPSP, sspsp); - break; - default: - break; - } - - /* When we use a network mode, we always require TDM slots - * - complain loudly and fail if they've not been set up yet. - */ - if ((sscr0 & SSCR0_MOD) && !ttsa) { - dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); - return -EINVAL; - } - - dump_registers(ssp); - - return 0; -} - static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -760,128 +504,22 @@ static void pxa_ssp_remove(struct platform_device *pdev, pxa_ssp_free(priv->ssp); } -#define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ - SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) - -#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) - -static struct snd_soc_dai_ops pxa_ssp_dai_ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, -}; - -struct snd_soc_dai pxa_ssp_dai[] = { - { - .name = "pxa2xx-ssp1", - .id = 0, - .probe = pxa_ssp_probe, - .remove = pxa_ssp_remove, - .suspend = pxa_ssp_suspend, - .resume = pxa_ssp_resume, - .playback = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .ops = &pxa_ssp_dai_ops, - }, - { .name = "pxa2xx-ssp2", - .id = 1, - .probe = pxa_ssp_probe, - .remove = pxa_ssp_remove, - .suspend = pxa_ssp_suspend, - .resume = pxa_ssp_resume, - .playback = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .ops = &pxa_ssp_dai_ops, - }, - { - .name = "pxa2xx-ssp3", - .id = 2, - .probe = pxa_ssp_probe, - .remove = pxa_ssp_remove, - .suspend = pxa_ssp_suspend, - .resume = pxa_ssp_resume, - .playback = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .ops = &pxa_ssp_dai_ops, - }, - { - .name = "pxa2xx-ssp4", - .id = 3, - .probe = pxa_ssp_probe, - .remove = pxa_ssp_remove, - .suspend = pxa_ssp_suspend, - .resume = pxa_ssp_resume, - .playback = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .ops = &pxa_ssp_dai_ops, - }, -}; -EXPORT_SYMBOL_GPL(pxa_ssp_dai); - -static int __init pxa_ssp_init(void) -{ - return snd_soc_register_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai)); -} -module_init(pxa_ssp_init); - -static void __exit pxa_ssp_exit(void) +int pxa_ssp_register_dai(struct snd_soc_dai *dai) { - snd_soc_unregister_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai)); + struct snd_soc_dai_ops *ops = dai->ops; + + ops->startup = pxa_ssp_startup; + ops->shutdown = pxa_ssp_shutdown; + ops->trigger = pxa_ssp_trigger; + ops->hw_params = pxa_ssp_hw_params; + ops->set_fmt = pxa_ssp_set_dai_fmt; + ops->set_tdm_slot = pxa_ssp_set_dai_tdm_slot; + ops->set_tristate = pxa_ssp_set_dai_tristate; + + dai->probe = pxa_ssp_probe; + dai->remove = pxa_ssp_remove; + dai->suspend = pxa_ssp_suspend; + dai->resume = pxa_ssp_resume; + + return snd_soc_register_dai(dai); } -module_exit(pxa_ssp_exit); - -/* Module information */ -MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); -MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa-ssp.h b/sound/soc/pxa/pxa-ssp.h index 91deadd..af4a09b 100644 --- a/sound/soc/pxa/pxa-ssp.h +++ b/sound/soc/pxa/pxa-ssp.h @@ -9,39 +9,25 @@ #ifndef _PXA_SSP_H #define _PXA_SSP_H -/* pxa DAI SSP IDs */ -#define PXA_DAI_SSP1 0 -#define PXA_DAI_SSP2 1 -#define PXA_DAI_SSP3 2 -#define PXA_DAI_SSP4 3 - -/* SSP clock sources */ -#define PXA_SSP_CLK_PLL 0 -#define PXA_SSP_CLK_EXT 1 -#define PXA_SSP_CLK_NET 2 -#define PXA_SSP_CLK_AUDIO 3 -#define PXA_SSP_CLK_NET_PLL 4 - -/* SSP audio dividers */ -#define PXA_SSP_AUDIO_DIV_ACDS 0 -#define PXA_SSP_AUDIO_DIV_SCDB 1 -#define PXA_SSP_DIV_SCR 2 - -/* SSP ACDS audio dividers values */ -#define PXA_SSP_CLK_AUDIO_DIV_1 0 -#define PXA_SSP_CLK_AUDIO_DIV_2 1 -#define PXA_SSP_CLK_AUDIO_DIV_4 2 -#define PXA_SSP_CLK_AUDIO_DIV_8 3 -#define PXA_SSP_CLK_AUDIO_DIV_16 4 -#define PXA_SSP_CLK_AUDIO_DIV_32 5 - -/* SSP divider bypass */ -#define PXA_SSP_CLK_SCDB_4 0 -#define PXA_SSP_CLK_SCDB_1 1 -#define PXA_SSP_CLK_SCDB_8 2 - -#define PXA_SSP_PLL_OUT 0 +/* + * SSP audio data + */ +struct ssp_priv { + struct ssp_device *ssp; + unsigned int sysclk; + int dai_fmt; +#ifdef CONFIG_PM + uint32_t cr0; + uint32_t cr1; + uint32_t to; + uint32_t psp; +#endif +}; -extern struct snd_soc_dai pxa_ssp_dai[4]; +extern void pxa_ssp_enable(struct ssp_device *ssp); +extern void pxa_ssp_disable(struct ssp_device *ssp); +extern struct pxa2xx_pcm_dma_params * +pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out); +extern int pxa_ssp_register_dai(struct snd_soc_dai *dai); #endif diff --git a/sound/soc/pxa/pxa2xx-ssp.c b/sound/soc/pxa/pxa2xx-ssp.c new file mode 100644 index 0000000..eff5334 --- /dev/null +++ b/sound/soc/pxa/pxa2xx-ssp.c @@ -0,0 +1,318 @@ +/* + * pxa2xx-ssp.c -- ALSA Soc Audio Layer + * + * Copyright 2005,2008 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * Copyright 2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.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. + * + * TODO: + * o Test network mode for > 16bit sample size + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/pxa2xx-lib.h> + +#include <mach/hardware.h> +#include <mach/dma.h> +#include <plat/ssp.h> + +#include "pxa2xx-pcm.h" +#include "pxa2xx-ssp.h" +#include "pxa-ssp.h" + +/** + * pxa_ssp_set_clkdiv - set SSP clock divider + * @div: serial clock rate divider + */ +static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) +{ + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { + sscr0 &= ~0x0000ff00; + sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ + } else { + sscr0 &= ~0x000fff00; + sscr0 |= (div - 1) << 8; /* 1..4096 */ + } + pxa_ssp_write_reg(ssp, SSCR0, sscr0); +} + +/** + * pxa_ssp_get_clkdiv - get SSP clock divider + */ +static u32 pxa_ssp_get_scr(struct ssp_device *ssp) +{ + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); + u32 div; + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) + div = ((sscr0 >> 8) & 0xff) * 2 + 2; + else + div = ((sscr0 >> 8) & 0xfff) + 1; + return div; +} + +/* + * Set the SSP ports SYSCLK. + */ +static int pxa2xx_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + int val; + + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & + ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + + dev_dbg(&ssp->pdev->dev, + "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", + cpu_dai->id, clk_id, freq); + + switch (clk_id) { + case PXA2XX_SSP_CLK_NET_PLL: + sscr0 |= SSCR0_MOD; + break; + case PXA2XX_SSP_CLK_PLL: + /* Internal PLL is fixed */ + if (cpu_is_pxa25x()) + priv->sysclk = 1843200; + else + priv->sysclk = 13000000; + break; + case PXA2XX_SSP_CLK_EXT: + priv->sysclk = freq; + sscr0 |= SSCR0_ECS; + break; + case PXA2XX_SSP_CLK_NET: + priv->sysclk = freq; + sscr0 |= SSCR0_NCS | SSCR0_MOD; + break; + case PXA2XX_SSP_CLK_AUDIO: + priv->sysclk = 0; + pxa_ssp_set_scr(ssp, 1); + sscr0 |= SSCR0_ACS; + break; + default: + return -ENODEV; + } + + /* The SSP clock must be disabled when changing SSP clock mode + * on PXA2xx. On PXA3xx it must be enabled when doing so. */ + if (!cpu_is_pxa3xx()) + clk_disable(ssp->clk); + val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; + pxa_ssp_write_reg(ssp, SSCR0, val); + if (!cpu_is_pxa3xx()) + clk_enable(ssp->clk); + + return 0; +} + +/* + * Set the SSP clock dividers. + */ +static int pxa2xx_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + int val; + + switch (div_id) { + case PXA2XX_SSP_AUDIO_DIV_ACDS: + val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); + pxa_ssp_write_reg(ssp, SSACD, val); + break; + case PXA2XX_SSP_AUDIO_DIV_SCDB: + val = pxa_ssp_read_reg(ssp, SSACD); + val &= ~SSACD_SCDB; +#if defined(CONFIG_PXA3xx) + if (cpu_is_pxa3xx()) + val &= ~SSACD_SCDX8; +#endif + switch (div) { + case PXA2XX_SSP_CLK_SCDB_1: + val |= SSACD_SCDB; + break; + case PXA2XX_SSP_CLK_SCDB_4: + break; +#if defined(CONFIG_PXA3xx) + case PXA2XX_SSP_CLK_SCDB_8: + if (cpu_is_pxa3xx()) + val |= SSACD_SCDX8; + else + return -EINVAL; + break; +#endif + default: + return -EINVAL; + } + pxa_ssp_write_reg(ssp, SSACD, val); + break; + case PXA2XX_SSP_DIV_SCR: + pxa_ssp_set_scr(ssp, div); + break; + default: + return -ENODEV; + } + + return 0; +} + +/* + * Configure the PLL frequency pxa27x and (afaik - pxa320 only) + */ +static int pxa2xx_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; + +#if defined(CONFIG_PXA3xx) + if (cpu_is_pxa3xx()) + pxa_ssp_write_reg(ssp, SSACDD, 0); +#endif + + switch (freq_out) { + case 5622000: + break; + case 11345000: + ssacd |= (0x1 << 4); + break; + case 12235000: + ssacd |= (0x2 << 4); + break; + case 14857000: + ssacd |= (0x3 << 4); + break; + case 32842000: + ssacd |= (0x4 << 4); + break; + case 48000000: + ssacd |= (0x5 << 4); + break; + case 0: + /* Disable */ + break; + + default: +#ifdef CONFIG_PXA3xx + /* PXA3xx has a clock ditherer which can be used to generate + * a wider range of frequencies - calculate a value for it. + */ + if (cpu_is_pxa3xx()) { + u32 val; + u64 tmp = 19968; + tmp *= 1000000; + do_div(tmp, freq_out); + val = tmp; + + val = (val << 16) | 64; + pxa_ssp_write_reg(ssp, SSACDD, val); + + ssacd |= (0x6 << 4); + + dev_dbg(&ssp->pdev->dev, + "Using SSACDD %x to supply %uHz\n", + val, freq_out); + break; + } +#endif + + return -EINVAL; + } + + pxa_ssp_write_reg(ssp, SSACD, ssacd); + + return 0; +} + +static struct snd_soc_dai_ops pxa2xx_ssp_dai_ops = { + .set_sysclk = pxa2xx_ssp_set_dai_sysclk, + .set_clkdiv = pxa2xx_ssp_set_dai_clkdiv, + .set_pll = pxa2xx_ssp_set_dai_pll, +}; + +#define PXA2XX_SSP_RATES SNDRV_PCM_RATE_8000_96000 +#define PXA2XX_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define PXA2XX_SSP_DAI(_id) \ +{ \ + .name = "pxa2xx-ssp", \ + .id = _id, \ + .playback = { \ + .channels_min = 1, \ + .channels_max = 8, \ + .rates = PXA2XX_SSP_RATES, \ + .formats = PXA2XX_SSP_FORMATS, \ + }, \ + .capture = { \ + .channels_min = 1, \ + .channels_max = 8, \ + .rates = PXA2XX_SSP_RATES, \ + .formats = PXA2XX_SSP_FORMATS, \ + }, \ + .ops = &pxa2xx_ssp_dai_ops, \ +} + +struct snd_soc_dai pxa2xx_ssp_dai[] = { + PXA2XX_SSP_DAI(PXA2XX_DAI_SSP1), + PXA2XX_SSP_DAI(PXA2XX_DAI_SSP2), + PXA2XX_SSP_DAI(PXA2XX_DAI_SSP3), + PXA2XX_SSP_DAI(PXA2XX_DAI_SSP4), +}; +EXPORT_SYMBOL_GPL(pxa2xx_ssp_dai); + +static int __init pxa2xx_ssp_init(void) +{ + struct snd_soc_dai *dai; + int i, ret; + + for (i = 0; i < ARRAY_SIZE(pxa2xx_ssp_dai); i++) { + dai = &pxa2xx_ssp_dai[i]; + ret = pxa_ssp_register_dai(dai); + if (ret) + return ret; + } + return ret; +} +module_init(pxa2xx_ssp_init); + +static void __exit pxa2xx_ssp_exit(void) +{ + struct snd_soc_dai *dai = NULL; + int i; + + for (i = 0; i < ARRAY_SIZE(pxa2xx_ssp_dai); i++) { + dai = &pxa2xx_ssp_dai[i]; + snd_soc_unregister_dai(dai); + } +} +module_exit(pxa2xx_ssp_exit); + +/* Module information */ +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_DESCRIPTION("PXA2xx SSP/PCM SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa2xx-ssp.h b/sound/soc/pxa/pxa2xx-ssp.h new file mode 100644 index 0000000..c24e0b5 --- /dev/null +++ b/sound/soc/pxa/pxa2xx-ssp.h @@ -0,0 +1,48 @@ + +/* + * ASoC PXA SSP port support + * + * 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 __PXA2XX_SOC_SSP_H +#define __PXA2XX_SOC_SSP_H + +/* pxa DAI SSP IDs */ +#define PXA2XX_DAI_SSP1 0 +#define PXA2XX_DAI_SSP2 1 +#define PXA2XX_DAI_SSP3 2 +#define PXA2XX_DAI_SSP4 3 + +/* SSP clock sources */ +#define PXA2XX_SSP_CLK_PLL 0 +#define PXA2XX_SSP_CLK_EXT 1 +#define PXA2XX_SSP_CLK_NET 2 +#define PXA2XX_SSP_CLK_AUDIO 3 +#define PXA2XX_SSP_CLK_NET_PLL 4 + +/* SSP audio dividers */ +#define PXA2XX_SSP_AUDIO_DIV_ACDS 0 +#define PXA2XX_SSP_AUDIO_DIV_SCDB 1 +#define PXA2XX_SSP_DIV_SCR 2 + +/* SSP ACDS audio dividers values */ +#define PXA2XX_SSP_CLK_AUDIO_DIV_1 0 +#define PXA2XX_SSP_CLK_AUDIO_DIV_2 1 +#define PXA2XX_SSP_CLK_AUDIO_DIV_4 2 +#define PXA2XX_SSP_CLK_AUDIO_DIV_8 3 +#define PXA2XX_SSP_CLK_AUDIO_DIV_16 4 +#define PXA2XX_SSP_CLK_AUDIO_DIV_32 5 + +/* SSP divider bypass */ +#define PXA2XX_SSP_CLK_SCDB_4 0 +#define PXA2XX_SSP_CLK_SCDB_1 1 +#define PXA2XX_SSP_CLK_SCDB_8 2 + +#define PXA2XX_SSP_PLL_OUT 0 + +extern struct snd_soc_dai pxa2xx_ssp_dai[]; + +#endif /* __PXA2XX_SOC_SSP_H */ diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 7e3f416..e66dc12 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -29,7 +29,7 @@ #include "../codecs/cs4270.h" #include "../codecs/ak4104.h" #include "pxa2xx-pcm.h" -#include "pxa-ssp.h" +#include "pxa2xx-ssp.h" #define GPIO_SPDIF_RESET (38) #define GPIO_MCLK_RESET (111) @@ -138,11 +138,11 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA2XX_SSP_DIV_SCR, 4); if (ret < 0) return ret; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_SSP_CLK_EXT, clk, 1); if (ret < 0) return ret; @@ -170,7 +170,7 @@ static int raumfeld_line_resume(struct platform_device *pdev) static struct snd_soc_dai_link raumfeld_line_dai = { .name = "CS4270", .stream_name = "CS4270", - .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .cpu_dai = &pxa2xx_ssp_dai[PXA2XX_DAI_SSP1], .codec_dai = &cs4270_dai, .ops = &raumfeld_cs4270_ops, }; @@ -232,11 +232,11 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA2XX_SSP_DIV_SCR, 4); if (ret < 0) return ret; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_SSP_CLK_EXT, clk, 1); if (ret < 0) return ret; @@ -250,7 +250,7 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { static struct snd_soc_dai_link raumfeld_spdif_dai = { .name = "ak4104", .stream_name = "Playback", - .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP2], + .cpu_dai = &pxa2xx_ssp_dai[PXA2XX_DAI_SSP2], .codec_dai = &ak4104_dai, .ops = &raumfeld_ak4104_ops, }; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index dd678ae..c65e0db 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -25,7 +25,7 @@ #include "../codecs/wm9713.h" #include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" -#include "pxa-ssp.h" +#include "pxa2xx-ssp.h" /* * There is a physical switch SW15 on the board which changes the MCLK @@ -125,7 +125,7 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, /* Add 1 to the width for the leading clock cycle */ pll_out = rate * (width + 1) * 8; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_SSP_CLK_AUDIO, 0, 1); if (ret < 0) return ret; @@ -176,7 +176,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { { .name = "WM9713 Voice", .stream_name = "WM9713 Voice", - .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP3], + .cpu_dai = &pxa2xx_ssp_dai[PXA2XX_DAI_SSP3], .codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE], .ops = &zylonite_voice_ops, }, -- 1.5.6.5

1 0

[PATCH] [ARM] pxa: add namespace on ssp
by Haojian Zhuang 31 Mar '10

31 Mar '10

In order to prevent code ambiguous, add namespace on functions in ssp driver. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- arch/arm/plat-pxa/include/plat/ssp.h | 12 ++-- arch/arm/plat-pxa/ssp.c | 22 +++--- drivers/spi/pxa2xx_spi.c | 8 +- sound/soc/pxa/pxa-ssp.c | 132 +++++++++++++++++----------------- 4 files changed, 87 insertions(+), 87 deletions(-) diff --git a/arch/arm/plat-pxa/include/plat/ssp.h b/arch/arm/plat-pxa/include/plat/ssp.h index d16d79a..fe43150 100644 --- a/arch/arm/plat-pxa/include/plat/ssp.h +++ b/arch/arm/plat-pxa/include/plat/ssp.h @@ -159,28 +159,28 @@ struct ssp_device { }; /** - * ssp_write_reg - Write to a SSP register + * pxa_ssp_write_reg - Write to a SSP register * * @dev: SSP device to access * @reg: Register to write to * @val: Value to be written. */ -static inline void ssp_write_reg(struct ssp_device *dev, u32 reg, u32 val) +static inline void pxa_ssp_write_reg(struct ssp_device *dev, u32 reg, u32 val) { __raw_writel(val, dev->mmio_base + reg); } /** - * ssp_read_reg - Read from a SSP register + * pxa_ssp_read_reg - Read from a SSP register * * @dev: SSP device to access * @reg: Register to read from */ -static inline u32 ssp_read_reg(struct ssp_device *dev, u32 reg) +static inline u32 pxa_ssp_read_reg(struct ssp_device *dev, u32 reg) { return __raw_readl(dev->mmio_base + reg); } -struct ssp_device *ssp_request(int port, const char *label); -void ssp_free(struct ssp_device *); +struct ssp_device *pxa_ssp_request(int port, const char *label); +void pxa_ssp_free(struct ssp_device *); #endif /* __ASM_ARCH_SSP_H */ diff --git a/arch/arm/plat-pxa/ssp.c b/arch/arm/plat-pxa/ssp.c index cfebcd8..511805e 100644 --- a/arch/arm/plat-pxa/ssp.c +++ b/arch/arm/plat-pxa/ssp.c @@ -37,7 +37,7 @@ static DEFINE_MUTEX(ssp_lock); static LIST_HEAD(ssp_list); -struct ssp_device *ssp_request(int port, const char *label) +struct ssp_device *pxa_ssp_request(int port, const char *label) { struct ssp_device *ssp = NULL; @@ -58,9 +58,9 @@ struct ssp_device *ssp_request(int port, const char *label) return ssp; } -EXPORT_SYMBOL(ssp_request); +EXPORT_SYMBOL(pxa_ssp_request); -void ssp_free(struct ssp_device *ssp) +void pxa_ssp_free(struct ssp_device *ssp) { mutex_lock(&ssp_lock); if (ssp->use_count) { @@ -70,9 +70,9 @@ void ssp_free(struct ssp_device *ssp) dev_err(&ssp->pdev->dev, "device already free\n"); mutex_unlock(&ssp_lock); } -EXPORT_SYMBOL(ssp_free); +EXPORT_SYMBOL(pxa_ssp_free); -static int __devinit ssp_probe(struct platform_device *pdev) +static int __devinit pxa_ssp_probe(struct platform_device *pdev) { const struct platform_device_id *id = platform_get_device_id(pdev); struct resource *res; @@ -164,7 +164,7 @@ err_free: return ret; } -static int __devexit ssp_remove(struct platform_device *pdev) +static int __devexit pxa_ssp_remove(struct platform_device *pdev) { struct resource *res; struct ssp_device *ssp; @@ -196,9 +196,9 @@ static const struct platform_device_id ssp_id_table[] = { { }, }; -static struct platform_driver ssp_driver = { - .probe = ssp_probe, - .remove = __devexit_p(ssp_remove), +static struct platform_driver pxa_ssp_driver = { + .probe = pxa_ssp_probe, + .remove = __devexit_p(pxa_ssp_remove), .driver = { .owner = THIS_MODULE, .name = "pxa2xx-ssp", @@ -208,12 +208,12 @@ static struct platform_driver ssp_driver = { static int __init pxa_ssp_init(void) { - return platform_driver_register(&ssp_driver); + return platform_driver_register(&pxa_ssp_driver); } static void __exit pxa_ssp_exit(void) { - platform_driver_unregister(&ssp_driver); + platform_driver_unregister(&pxa_ssp_driver); } arch_initcall(pxa_ssp_init); diff --git a/drivers/spi/pxa2xx_spi.c b/drivers/spi/pxa2xx_spi.c index 4a64da7..2fd7d5a 100644 --- a/drivers/spi/pxa2xx_spi.c +++ b/drivers/spi/pxa2xx_spi.c @@ -1464,7 +1464,7 @@ static int __init pxa2xx_spi_probe(struct platform_device *pdev) platform_info = dev->platform_data; - ssp = ssp_request(pdev->id, pdev->name); + ssp = pxa_ssp_request(pdev->id, pdev->name); if (ssp == NULL) { dev_err(&pdev->dev, "failed to request SSP%d\n", pdev->id); return -ENODEV; @@ -1474,7 +1474,7 @@ static int __init pxa2xx_spi_probe(struct platform_device *pdev) master = spi_alloc_master(dev, sizeof(struct driver_data) + 16); if (!master) { dev_err(&pdev->dev, "cannot alloc spi_master\n"); - ssp_free(ssp); + pxa_ssp_free(ssp); return -ENOMEM; } drv_data = spi_master_get_devdata(master); @@ -1603,7 +1603,7 @@ out_error_irq_alloc: out_error_master_alloc: spi_master_put(master); - ssp_free(ssp); + pxa_ssp_free(ssp); return status; } @@ -1647,7 +1647,7 @@ static int pxa2xx_spi_remove(struct platform_device *pdev) free_irq(ssp->irq, drv_data); /* Release SSP */ - ssp_free(ssp); + pxa_ssp_free(ssp); /* Disconnect from the SPI framework */ spi_unregister_master(drv_data->master); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 4ca9245..ed54bef 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -55,15 +55,15 @@ struct ssp_priv { static void dump_registers(struct ssp_device *ssp) { dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n", - ssp_read_reg(ssp, SSCR0), ssp_read_reg(ssp, SSCR1), - ssp_read_reg(ssp, SSTO)); + pxa_ssp_read_reg(ssp, SSCR0), pxa_ssp_read_reg(ssp, SSCR1), + pxa_ssp_read_reg(ssp, SSTO)); dev_dbg(&ssp->pdev->dev, "SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x\n", - ssp_read_reg(ssp, SSPSP), ssp_read_reg(ssp, SSSR), - ssp_read_reg(ssp, SSACD)); + pxa_ssp_read_reg(ssp, SSPSP), pxa_ssp_read_reg(ssp, SSSR), + pxa_ssp_read_reg(ssp, SSACD)); } -static void ssp_enable(struct ssp_device *ssp) +static void pxa_ssp_enable(struct ssp_device *ssp) { uint32_t sscr0; @@ -71,7 +71,7 @@ static void ssp_enable(struct ssp_device *ssp) __raw_writel(sscr0, ssp->mmio_base + SSCR0); } -static void ssp_disable(struct ssp_device *ssp) +static void pxa_ssp_disable(struct ssp_device *ssp) { uint32_t sscr0; @@ -85,7 +85,7 @@ struct pxa2xx_pcm_dma_data { }; static struct pxa2xx_pcm_dma_params * -ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) +pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) { struct pxa2xx_pcm_dma_data *dma; @@ -117,7 +117,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, if (!cpu_dai->active) { clk_enable(ssp->clk); - ssp_disable(ssp); + pxa_ssp_disable(ssp); } if (cpu_dai->dma_data) { @@ -136,7 +136,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, struct ssp_device *ssp = priv->ssp; if (!cpu_dai->active) { - ssp_disable(ssp); + pxa_ssp_disable(ssp); clk_disable(ssp->clk); } @@ -161,7 +161,7 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) priv->to = __raw_readl(ssp->mmio_base + SSTO); priv->psp = __raw_readl(ssp->mmio_base + SSPSP); - ssp_disable(ssp); + pxa_ssp_disable(ssp); clk_disable(ssp->clk); return 0; } @@ -181,7 +181,7 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) __raw_writel(priv->psp, ssp->mmio_base + SSPSP); if (cpu_dai->active) - ssp_enable(ssp); + pxa_ssp_enable(ssp); else clk_disable(ssp->clk); @@ -197,9 +197,9 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) * ssp_set_clkdiv - set SSP clock divider * @div: serial clock rate divider */ -static void ssp_set_scr(struct ssp_device *ssp, u32 div) +static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) { - u32 sscr0 = ssp_read_reg(ssp, SSCR0); + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { sscr0 &= ~0x0000ff00; @@ -208,15 +208,15 @@ static void ssp_set_scr(struct ssp_device *ssp, u32 div) sscr0 &= ~0x000fff00; sscr0 |= (div - 1) << 8; /* 1..4096 */ } - ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); } /** - * ssp_get_clkdiv - get SSP clock divider + * pxa_ssp_get_clkdiv - get SSP clock divider */ -static u32 ssp_get_scr(struct ssp_device *ssp) +static u32 pxa_ssp_get_scr(struct ssp_device *ssp) { - u32 sscr0 = ssp_read_reg(ssp, SSCR0); + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); u32 div; if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) @@ -236,7 +236,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, struct ssp_device *ssp = priv->ssp; int val; - u32 sscr0 = ssp_read_reg(ssp, SSCR0) & + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); dev_dbg(&ssp->pdev->dev, @@ -264,7 +264,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, break; case PXA_SSP_CLK_AUDIO: priv->sysclk = 0; - ssp_set_scr(ssp, 1); + pxa_ssp_set_scr(ssp, 1); sscr0 |= SSCR0_ACS; break; default: @@ -275,8 +275,8 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, * on PXA2xx. On PXA3xx it must be enabled when doing so. */ if (!cpu_is_pxa3xx()) clk_disable(ssp->clk); - val = ssp_read_reg(ssp, SSCR0) | sscr0; - ssp_write_reg(ssp, SSCR0, val); + val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; + pxa_ssp_write_reg(ssp, SSCR0, val); if (!cpu_is_pxa3xx()) clk_enable(ssp->clk); @@ -295,11 +295,11 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, switch (div_id) { case PXA_SSP_AUDIO_DIV_ACDS: - val = (ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); - ssp_write_reg(ssp, SSACD, val); + val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); + pxa_ssp_write_reg(ssp, SSACD, val); break; case PXA_SSP_AUDIO_DIV_SCDB: - val = ssp_read_reg(ssp, SSACD); + val = pxa_ssp_read_reg(ssp, SSACD); val &= ~SSACD_SCDB; #if defined(CONFIG_PXA3xx) if (cpu_is_pxa3xx()) @@ -322,10 +322,10 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, default: return -EINVAL; } - ssp_write_reg(ssp, SSACD, val); + pxa_ssp_write_reg(ssp, SSACD, val); break; case PXA_SSP_DIV_SCR: - ssp_set_scr(ssp, div); + pxa_ssp_set_scr(ssp, div); break; default: return -ENODEV; @@ -342,11 +342,11 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, { struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->ssp; - u32 ssacd = ssp_read_reg(ssp, SSACD) & ~0x70; + u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; #if defined(CONFIG_PXA3xx) if (cpu_is_pxa3xx()) - ssp_write_reg(ssp, SSACDD, 0); + pxa_ssp_write_reg(ssp, SSACDD, 0); #endif switch (freq_out) { @@ -384,7 +384,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, val = tmp; val = (val << 16) | 64; - ssp_write_reg(ssp, SSACDD, val); + pxa_ssp_write_reg(ssp, SSACDD, val); ssacd |= (0x6 << 4); @@ -398,7 +398,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, return -EINVAL; } - ssp_write_reg(ssp, SSACD, ssacd); + pxa_ssp_write_reg(ssp, SSACD, ssacd); return 0; } @@ -413,7 +413,7 @@ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, struct ssp_device *ssp = priv->ssp; u32 sscr0; - sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 = pxa_ssp_read_reg(ssp, SSCR0); sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(8) | SSCR0_EDSS | SSCR0_DSS); /* set slot width */ @@ -430,10 +430,10 @@ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, sscr0 |= SSCR0_SlotsPerFrm(slots); /* set active slot mask */ - ssp_write_reg(ssp, SSTSA, tx_mask); - ssp_write_reg(ssp, SSRSA, rx_mask); + pxa_ssp_write_reg(ssp, SSTSA, tx_mask); + pxa_ssp_write_reg(ssp, SSRSA, rx_mask); } - ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); return 0; } @@ -448,12 +448,12 @@ static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai, struct ssp_device *ssp = priv->ssp; u32 sscr1; - sscr1 = ssp_read_reg(ssp, SSCR1); + sscr1 = pxa_ssp_read_reg(ssp, SSCR1); if (tristate) sscr1 &= ~SSCR1_TTE; else sscr1 |= SSCR1_TTE; - ssp_write_reg(ssp, SSCR1, sscr1); + pxa_ssp_write_reg(ssp, SSCR1, sscr1); return 0; } @@ -477,14 +477,14 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, return 0; /* we can only change the settings if the port is not in use */ - if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) { + if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) { dev_err(&ssp->pdev->dev, "can't change hardware dai format: stream is in use"); return -EINVAL; } /* reset port settings */ - sscr0 = ssp_read_reg(ssp, SSCR0) & + sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); sspsp = 0; @@ -536,9 +536,9 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } - ssp_write_reg(ssp, SSCR0, sscr0); - ssp_write_reg(ssp, SSCR1, sscr1); - ssp_write_reg(ssp, SSPSP, sspsp); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR1, sscr1); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); dump_registers(ssp); @@ -567,7 +567,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, u32 sscr0; u32 sspsp; int width = snd_pcm_format_physical_width(params_format(params)); - int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; + int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf; /* generate correct DMA params */ if (cpu_dai->dma_data) @@ -577,20 +577,20 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, * to force 16-bit frame width on the wire (for S16_LE), even * with two channels. Use 16-bit DMA transfers for this case. */ - cpu_dai->dma_data = ssp_get_dma_params(ssp, + cpu_dai->dma_data = pxa_ssp_get_dma_params(ssp, ((chn == 2) && (ttsa != 1)) || (width == 32), substream->stream == SNDRV_PCM_STREAM_PLAYBACK); /* we can only change the settings if the port is not in use */ - if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) + if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) return 0; /* clear selected SSP bits */ - sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); - ssp_write_reg(ssp, SSCR0, sscr0); + sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); /* bit size */ - sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 = pxa_ssp_read_reg(ssp, SSCR0); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: #ifdef CONFIG_PXA3xx @@ -606,13 +606,13 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); break; } - ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - sspsp = ssp_read_reg(ssp, SSPSP); + sspsp = pxa_ssp_read_reg(ssp, SSPSP); - if ((ssp_get_scr(ssp) == 4) && (width == 16)) { + if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) { /* This is a special case where the bitclk is 64fs * and we're not dealing with 2*32 bits of audio * samples. @@ -646,7 +646,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sspsp |= SSPSP_DMYSTRT(1); } - ssp_write_reg(ssp, SSPSP, sspsp); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); break; default: break; @@ -677,45 +677,45 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: - ssp_enable(ssp); + pxa_ssp_enable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - val = ssp_read_reg(ssp, SSCR1); + val = pxa_ssp_read_reg(ssp, SSCR1); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) val |= SSCR1_TSRE; else val |= SSCR1_RSRE; - ssp_write_reg(ssp, SSCR1, val); - val = ssp_read_reg(ssp, SSSR); - ssp_write_reg(ssp, SSSR, val); + pxa_ssp_write_reg(ssp, SSCR1, val); + val = pxa_ssp_read_reg(ssp, SSSR); + pxa_ssp_write_reg(ssp, SSSR, val); break; case SNDRV_PCM_TRIGGER_START: - val = ssp_read_reg(ssp, SSCR1); + val = pxa_ssp_read_reg(ssp, SSCR1); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) val |= SSCR1_TSRE; else val |= SSCR1_RSRE; - ssp_write_reg(ssp, SSCR1, val); - ssp_enable(ssp); + pxa_ssp_write_reg(ssp, SSCR1, val); + pxa_ssp_enable(ssp); break; case SNDRV_PCM_TRIGGER_STOP: - val = ssp_read_reg(ssp, SSCR1); + val = pxa_ssp_read_reg(ssp, SSCR1); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) val &= ~SSCR1_TSRE; else val &= ~SSCR1_RSRE; - ssp_write_reg(ssp, SSCR1, val); + pxa_ssp_write_reg(ssp, SSCR1, val); break; case SNDRV_PCM_TRIGGER_SUSPEND: - ssp_disable(ssp); + pxa_ssp_disable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - val = ssp_read_reg(ssp, SSCR1); + val = pxa_ssp_read_reg(ssp, SSCR1); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) val &= ~SSCR1_TSRE; else val &= ~SSCR1_RSRE; - ssp_write_reg(ssp, SSCR1, val); + pxa_ssp_write_reg(ssp, SSCR1, val); break; default: @@ -737,7 +737,7 @@ static int pxa_ssp_probe(struct platform_device *pdev, if (!priv) return -ENOMEM; - priv->ssp = ssp_request(dai->id + 1, "SoC audio"); + priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio"); if (priv->ssp == NULL) { ret = -ENODEV; goto err_priv; @@ -757,7 +757,7 @@ static void pxa_ssp_remove(struct platform_device *pdev, struct snd_soc_dai *dai) { struct ssp_priv *priv = dai->private_data; - ssp_free(priv->ssp); + pxa_ssp_free(priv->ssp); } #define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ -- 1.5.6.5

1 0

[PATCH] [ARM] pxa: add namespace on ssp
by Haojian Zhuang 31 Mar '10

31 Mar '10

In order to prevent code ambiguous, add namespace on functions in ssp driver. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- arch/arm/plat-pxa/include/plat/ssp.h | 12 ++-- arch/arm/plat-pxa/ssp.c | 22 +++--- drivers/spi/pxa2xx_spi.c | 8 +- sound/soc/pxa/pxa-ssp.c | 132 +++++++++++++++++----------------- 4 files changed, 87 insertions(+), 87 deletions(-) diff --git a/arch/arm/plat-pxa/include/plat/ssp.h b/arch/arm/plat-pxa/include/plat/ssp.h index d16d79a..fe43150 100644 --- a/arch/arm/plat-pxa/include/plat/ssp.h +++ b/arch/arm/plat-pxa/include/plat/ssp.h @@ -159,28 +159,28 @@ struct ssp_device { }; /** - * ssp_write_reg - Write to a SSP register + * pxa_ssp_write_reg - Write to a SSP register * * @dev: SSP device to access * @reg: Register to write to * @val: Value to be written. */ -static inline void ssp_write_reg(struct ssp_device *dev, u32 reg, u32 val) +static inline void pxa_ssp_write_reg(struct ssp_device *dev, u32 reg, u32 val) { __raw_writel(val, dev->mmio_base + reg); } /** - * ssp_read_reg - Read from a SSP register + * pxa_ssp_read_reg - Read from a SSP register * * @dev: SSP device to access * @reg: Register to read from */ -static inline u32 ssp_read_reg(struct ssp_device *dev, u32 reg) +static inline u32 pxa_ssp_read_reg(struct ssp_device *dev, u32 reg) { return __raw_readl(dev->mmio_base + reg); } -struct ssp_device *ssp_request(int port, const char *label); -void ssp_free(struct ssp_device *); +struct ssp_device *pxa_ssp_request(int port, const char *label); +void pxa_ssp_free(struct ssp_device *); #endif /* __ASM_ARCH_SSP_H */ diff --git a/arch/arm/plat-pxa/ssp.c b/arch/arm/plat-pxa/ssp.c index cfebcd8..511805e 100644 --- a/arch/arm/plat-pxa/ssp.c +++ b/arch/arm/plat-pxa/ssp.c @@ -37,7 +37,7 @@ static DEFINE_MUTEX(ssp_lock); static LIST_HEAD(ssp_list); -struct ssp_device *ssp_request(int port, const char *label) +struct ssp_device *pxa_ssp_request(int port, const char *label) { struct ssp_device *ssp = NULL; @@ -58,9 +58,9 @@ struct ssp_device *ssp_request(int port, const char *label) return ssp; } -EXPORT_SYMBOL(ssp_request); +EXPORT_SYMBOL(pxa_ssp_request); -void ssp_free(struct ssp_device *ssp) +void pxa_ssp_free(struct ssp_device *ssp) { mutex_lock(&ssp_lock); if (ssp->use_count) { @@ -70,9 +70,9 @@ void ssp_free(struct ssp_device *ssp) dev_err(&ssp->pdev->dev, "device already free\n"); mutex_unlock(&ssp_lock); } -EXPORT_SYMBOL(ssp_free); +EXPORT_SYMBOL(pxa_ssp_free); -static int __devinit ssp_probe(struct platform_device *pdev) +static int __devinit pxa_ssp_probe(struct platform_device *pdev) { const struct platform_device_id *id = platform_get_device_id(pdev); struct resource *res; @@ -164,7 +164,7 @@ err_free: return ret; } -static int __devexit ssp_remove(struct platform_device *pdev) +static int __devexit pxa_ssp_remove(struct platform_device *pdev) { struct resource *res; struct ssp_device *ssp; @@ -196,9 +196,9 @@ static const struct platform_device_id ssp_id_table[] = { { }, }; -static struct platform_driver ssp_driver = { - .probe = ssp_probe, - .remove = __devexit_p(ssp_remove), +static struct platform_driver pxa_ssp_driver = { + .probe = pxa_ssp_probe, + .remove = __devexit_p(pxa_ssp_remove), .driver = { .owner = THIS_MODULE, .name = "pxa2xx-ssp", @@ -208,12 +208,12 @@ static struct platform_driver ssp_driver = { static int __init pxa_ssp_init(void) { - return platform_driver_register(&ssp_driver); + return platform_driver_register(&pxa_ssp_driver); } static void __exit pxa_ssp_exit(void) { - platform_driver_unregister(&ssp_driver); + platform_driver_unregister(&pxa_ssp_driver); } arch_initcall(pxa_ssp_init); diff --git a/drivers/spi/pxa2xx_spi.c b/drivers/spi/pxa2xx_spi.c index 4a64da7..2fd7d5a 100644 --- a/drivers/spi/pxa2xx_spi.c +++ b/drivers/spi/pxa2xx_spi.c @@ -1464,7 +1464,7 @@ static int __init pxa2xx_spi_probe(struct platform_device *pdev) platform_info = dev->platform_data; - ssp = ssp_request(pdev->id, pdev->name); + ssp = pxa_ssp_request(pdev->id, pdev->name); if (ssp == NULL) { dev_err(&pdev->dev, "failed to request SSP%d\n", pdev->id); return -ENODEV; @@ -1474,7 +1474,7 @@ static int __init pxa2xx_spi_probe(struct platform_device *pdev) master = spi_alloc_master(dev, sizeof(struct driver_data) + 16); if (!master) { dev_err(&pdev->dev, "cannot alloc spi_master\n"); - ssp_free(ssp); + pxa_ssp_free(ssp); return -ENOMEM; } drv_data = spi_master_get_devdata(master); @@ -1603,7 +1603,7 @@ out_error_irq_alloc: out_error_master_alloc: spi_master_put(master); - ssp_free(ssp); + pxa_ssp_free(ssp); return status; } @@ -1647,7 +1647,7 @@ static int pxa2xx_spi_remove(struct platform_device *pdev) free_irq(ssp->irq, drv_data); /* Release SSP */ - ssp_free(ssp); + pxa_ssp_free(ssp); /* Disconnect from the SPI framework */ spi_unregister_master(drv_data->master); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 4ca9245..ed54bef 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -55,15 +55,15 @@ struct ssp_priv { static void dump_registers(struct ssp_device *ssp) { dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n", - ssp_read_reg(ssp, SSCR0), ssp_read_reg(ssp, SSCR1), - ssp_read_reg(ssp, SSTO)); + pxa_ssp_read_reg(ssp, SSCR0), pxa_ssp_read_reg(ssp, SSCR1), + pxa_ssp_read_reg(ssp, SSTO)); dev_dbg(&ssp->pdev->dev, "SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x\n", - ssp_read_reg(ssp, SSPSP), ssp_read_reg(ssp, SSSR), - ssp_read_reg(ssp, SSACD)); + pxa_ssp_read_reg(ssp, SSPSP), pxa_ssp_read_reg(ssp, SSSR), + pxa_ssp_read_reg(ssp, SSACD)); } -static void ssp_enable(struct ssp_device *ssp) +static void pxa_ssp_enable(struct ssp_device *ssp) { uint32_t sscr0; @@ -71,7 +71,7 @@ static void ssp_enable(struct ssp_device *ssp) __raw_writel(sscr0, ssp->mmio_base + SSCR0); } -static void ssp_disable(struct ssp_device *ssp) +static void pxa_ssp_disable(struct ssp_device *ssp) { uint32_t sscr0; @@ -85,7 +85,7 @@ struct pxa2xx_pcm_dma_data { }; static struct pxa2xx_pcm_dma_params * -ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) +pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) { struct pxa2xx_pcm_dma_data *dma; @@ -117,7 +117,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, if (!cpu_dai->active) { clk_enable(ssp->clk); - ssp_disable(ssp); + pxa_ssp_disable(ssp); } if (cpu_dai->dma_data) { @@ -136,7 +136,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, struct ssp_device *ssp = priv->ssp; if (!cpu_dai->active) { - ssp_disable(ssp); + pxa_ssp_disable(ssp); clk_disable(ssp->clk); } @@ -161,7 +161,7 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) priv->to = __raw_readl(ssp->mmio_base + SSTO); priv->psp = __raw_readl(ssp->mmio_base + SSPSP); - ssp_disable(ssp); + pxa_ssp_disable(ssp); clk_disable(ssp->clk); return 0; } @@ -181,7 +181,7 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) __raw_writel(priv->psp, ssp->mmio_base + SSPSP); if (cpu_dai->active) - ssp_enable(ssp); + pxa_ssp_enable(ssp); else clk_disable(ssp->clk); @@ -197,9 +197,9 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) * ssp_set_clkdiv - set SSP clock divider * @div: serial clock rate divider */ -static void ssp_set_scr(struct ssp_device *ssp, u32 div) +static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) { - u32 sscr0 = ssp_read_reg(ssp, SSCR0); + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { sscr0 &= ~0x0000ff00; @@ -208,15 +208,15 @@ static void ssp_set_scr(struct ssp_device *ssp, u32 div) sscr0 &= ~0x000fff00; sscr0 |= (div - 1) << 8; /* 1..4096 */ } - ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); } /** - * ssp_get_clkdiv - get SSP clock divider + * pxa_ssp_get_clkdiv - get SSP clock divider */ -static u32 ssp_get_scr(struct ssp_device *ssp) +static u32 pxa_ssp_get_scr(struct ssp_device *ssp) { - u32 sscr0 = ssp_read_reg(ssp, SSCR0); + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); u32 div; if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) @@ -236,7 +236,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, struct ssp_device *ssp = priv->ssp; int val; - u32 sscr0 = ssp_read_reg(ssp, SSCR0) & + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); dev_dbg(&ssp->pdev->dev, @@ -264,7 +264,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, break; case PXA_SSP_CLK_AUDIO: priv->sysclk = 0; - ssp_set_scr(ssp, 1); + pxa_ssp_set_scr(ssp, 1); sscr0 |= SSCR0_ACS; break; default: @@ -275,8 +275,8 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, * on PXA2xx. On PXA3xx it must be enabled when doing so. */ if (!cpu_is_pxa3xx()) clk_disable(ssp->clk); - val = ssp_read_reg(ssp, SSCR0) | sscr0; - ssp_write_reg(ssp, SSCR0, val); + val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; + pxa_ssp_write_reg(ssp, SSCR0, val); if (!cpu_is_pxa3xx()) clk_enable(ssp->clk); @@ -295,11 +295,11 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, switch (div_id) { case PXA_SSP_AUDIO_DIV_ACDS: - val = (ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); - ssp_write_reg(ssp, SSACD, val); + val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); + pxa_ssp_write_reg(ssp, SSACD, val); break; case PXA_SSP_AUDIO_DIV_SCDB: - val = ssp_read_reg(ssp, SSACD); + val = pxa_ssp_read_reg(ssp, SSACD); val &= ~SSACD_SCDB; #if defined(CONFIG_PXA3xx) if (cpu_is_pxa3xx()) @@ -322,10 +322,10 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, default: return -EINVAL; } - ssp_write_reg(ssp, SSACD, val); + pxa_ssp_write_reg(ssp, SSACD, val); break; case PXA_SSP_DIV_SCR: - ssp_set_scr(ssp, div); + pxa_ssp_set_scr(ssp, div); break; default: return -ENODEV; @@ -342,11 +342,11 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, { struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->ssp; - u32 ssacd = ssp_read_reg(ssp, SSACD) & ~0x70; + u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; #if defined(CONFIG_PXA3xx) if (cpu_is_pxa3xx()) - ssp_write_reg(ssp, SSACDD, 0); + pxa_ssp_write_reg(ssp, SSACDD, 0); #endif switch (freq_out) { @@ -384,7 +384,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, val = tmp; val = (val << 16) | 64; - ssp_write_reg(ssp, SSACDD, val); + pxa_ssp_write_reg(ssp, SSACDD, val); ssacd |= (0x6 << 4); @@ -398,7 +398,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, return -EINVAL; } - ssp_write_reg(ssp, SSACD, ssacd); + pxa_ssp_write_reg(ssp, SSACD, ssacd); return 0; } @@ -413,7 +413,7 @@ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, struct ssp_device *ssp = priv->ssp; u32 sscr0; - sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 = pxa_ssp_read_reg(ssp, SSCR0); sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(8) | SSCR0_EDSS | SSCR0_DSS); /* set slot width */ @@ -430,10 +430,10 @@ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, sscr0 |= SSCR0_SlotsPerFrm(slots); /* set active slot mask */ - ssp_write_reg(ssp, SSTSA, tx_mask); - ssp_write_reg(ssp, SSRSA, rx_mask); + pxa_ssp_write_reg(ssp, SSTSA, tx_mask); + pxa_ssp_write_reg(ssp, SSRSA, rx_mask); } - ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); return 0; } @@ -448,12 +448,12 @@ static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai, struct ssp_device *ssp = priv->ssp; u32 sscr1; - sscr1 = ssp_read_reg(ssp, SSCR1); + sscr1 = pxa_ssp_read_reg(ssp, SSCR1); if (tristate) sscr1 &= ~SSCR1_TTE; else sscr1 |= SSCR1_TTE; - ssp_write_reg(ssp, SSCR1, sscr1); + pxa_ssp_write_reg(ssp, SSCR1, sscr1); return 0; } @@ -477,14 +477,14 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, return 0; /* we can only change the settings if the port is not in use */ - if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) { + if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) { dev_err(&ssp->pdev->dev, "can't change hardware dai format: stream is in use"); return -EINVAL; } /* reset port settings */ - sscr0 = ssp_read_reg(ssp, SSCR0) & + sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); sspsp = 0; @@ -536,9 +536,9 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } - ssp_write_reg(ssp, SSCR0, sscr0); - ssp_write_reg(ssp, SSCR1, sscr1); - ssp_write_reg(ssp, SSPSP, sspsp); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR1, sscr1); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); dump_registers(ssp); @@ -567,7 +567,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, u32 sscr0; u32 sspsp; int width = snd_pcm_format_physical_width(params_format(params)); - int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; + int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf; /* generate correct DMA params */ if (cpu_dai->dma_data) @@ -577,20 +577,20 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, * to force 16-bit frame width on the wire (for S16_LE), even * with two channels. Use 16-bit DMA transfers for this case. */ - cpu_dai->dma_data = ssp_get_dma_params(ssp, + cpu_dai->dma_data = pxa_ssp_get_dma_params(ssp, ((chn == 2) && (ttsa != 1)) || (width == 32), substream->stream == SNDRV_PCM_STREAM_PLAYBACK); /* we can only change the settings if the port is not in use */ - if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) + if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) return 0; /* clear selected SSP bits */ - sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); - ssp_write_reg(ssp, SSCR0, sscr0); + sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); /* bit size */ - sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 = pxa_ssp_read_reg(ssp, SSCR0); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: #ifdef CONFIG_PXA3xx @@ -606,13 +606,13 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); break; } - ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - sspsp = ssp_read_reg(ssp, SSPSP); + sspsp = pxa_ssp_read_reg(ssp, SSPSP); - if ((ssp_get_scr(ssp) == 4) && (width == 16)) { + if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) { /* This is a special case where the bitclk is 64fs * and we're not dealing with 2*32 bits of audio * samples. @@ -646,7 +646,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sspsp |= SSPSP_DMYSTRT(1); } - ssp_write_reg(ssp, SSPSP, sspsp); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); break; default: break; @@ -677,45 +677,45 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: - ssp_enable(ssp); + pxa_ssp_enable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - val = ssp_read_reg(ssp, SSCR1); + val = pxa_ssp_read_reg(ssp, SSCR1); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) val |= SSCR1_TSRE; else val |= SSCR1_RSRE; - ssp_write_reg(ssp, SSCR1, val); - val = ssp_read_reg(ssp, SSSR); - ssp_write_reg(ssp, SSSR, val); + pxa_ssp_write_reg(ssp, SSCR1, val); + val = pxa_ssp_read_reg(ssp, SSSR); + pxa_ssp_write_reg(ssp, SSSR, val); break; case SNDRV_PCM_TRIGGER_START: - val = ssp_read_reg(ssp, SSCR1); + val = pxa_ssp_read_reg(ssp, SSCR1); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) val |= SSCR1_TSRE; else val |= SSCR1_RSRE; - ssp_write_reg(ssp, SSCR1, val); - ssp_enable(ssp); + pxa_ssp_write_reg(ssp, SSCR1, val); + pxa_ssp_enable(ssp); break; case SNDRV_PCM_TRIGGER_STOP: - val = ssp_read_reg(ssp, SSCR1); + val = pxa_ssp_read_reg(ssp, SSCR1); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) val &= ~SSCR1_TSRE; else val &= ~SSCR1_RSRE; - ssp_write_reg(ssp, SSCR1, val); + pxa_ssp_write_reg(ssp, SSCR1, val); break; case SNDRV_PCM_TRIGGER_SUSPEND: - ssp_disable(ssp); + pxa_ssp_disable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - val = ssp_read_reg(ssp, SSCR1); + val = pxa_ssp_read_reg(ssp, SSCR1); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) val &= ~SSCR1_TSRE; else val &= ~SSCR1_RSRE; - ssp_write_reg(ssp, SSCR1, val); + pxa_ssp_write_reg(ssp, SSCR1, val); break; default: @@ -737,7 +737,7 @@ static int pxa_ssp_probe(struct platform_device *pdev, if (!priv) return -ENOMEM; - priv->ssp = ssp_request(dai->id + 1, "SoC audio"); + priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio"); if (priv->ssp == NULL) { ret = -ENODEV; goto err_priv; @@ -757,7 +757,7 @@ static void pxa_ssp_remove(struct platform_device *pdev, struct snd_soc_dai *dai) { struct ssp_priv *priv = dai->private_data; - ssp_free(priv->ssp); + pxa_ssp_free(priv->ssp); } #define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ -- 1.5.6.5

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[alsa-devel] [PATCH 0/4] pxa ssp patches
by Haojian Zhuang 31 Mar '10

31 Mar '10

These patches are based on ssp_cleanup of pxa git tree. They're verified on pxa168 aspenite platform. Support I2S and Left J interface.

1 0

[alsa-devel] [PATCH 2/4] ASoC: split pxa ssp for reusing code
by Haojian Zhuang 31 Mar '10

31 Mar '10

1 0

[alsa-devel] pulseaudio module-udev-detect failure on 2.6.34-rc1
by Wu Fengguang 31 Mar '10

31 Mar '10

Greetings, Pulseaudio can no longer detect sound cards on kernel 2.6.34-rc1. Note that 2.6.33 is OK. I noticed that the related sysfs directories/files changed. Note that the card0/ directory is gone in new kernel: 2.6.33 wfg@hp ~% ll /sys/class/sound total 0 lrwxrwxrwx 1 root root 0 2010-03-31 12:23 timer -> ../../devices/virtual/sound/timer lrwxrwxrwx 1 root root 0 2010-03-31 12:23 pcmC0D1p -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0/pcmC0D1p lrwxrwxrwx 1 root root 0 2010-03-31 12:23 pcmC0D0p -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0/pcmC0D0p lrwxrwxrwx 1 root root 0 2010-03-31 12:23 pcmC0D0c -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0/pcmC0D0c lrwxrwxrwx 1 root root 0 2010-03-31 12:23 mixer5 -> ../../devices/platform/thinkpad_acpi/sound/card5/mixer5 lrwxrwxrwx 1 root root 0 2010-03-31 12:23 mixer -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0/mixer lrwxrwxrwx 1 root root 0 2010-03-31 12:23 hwC0D0 -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0 lrwxrwxrwx 1 root root 0 2010-03-31 12:23 dsp -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0/dsp lrwxrwxrwx 1 root root 0 2010-03-31 12:23 controlC5 -> ../../devices/platform/thinkpad_acpi/sound/card5/controlC5 lrwxrwxrwx 1 root root 0 2010-03-31 12:23 controlC0 -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0/controlC0 lrwxrwxrwx 1 root root 0 2010-03-31 12:23 card5 -> ../../devices/platform/thinkpad_acpi/sound/card5 lrwxrwxrwx 1 root root 0 2010-03-31 12:23 card0 -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0 lrwxrwxrwx 1 root root 0 2010-03-31 12:23 audio -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0/audio lrwxrwxrwx 1 root root 0 2010-03-31 12:23 adsp -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0/adsp 2.6.34-rc1 wfg@hp ~% ll /sys/class/sound total 0 drwxr-xr-x 3 root root 0 2010-03-31 19:27 timer drwxr-xr-x 3 root root 0 2010-03-31 19:27 pcmC0D7p drwxr-xr-x 3 root root 0 2010-03-31 19:27 pcmC0D3p drwxr-xr-x 3 root root 0 2010-03-31 19:27 pcmC0D0p drwxr-xr-x 3 root root 0 2010-03-31 19:27 pcmC0D0c drwxr-xr-x 3 root root 0 2010-03-31 19:27 hwC0D3 drwxr-xr-x 3 root root 0 2010-03-31 19:27 hwC0D0 drwxr-xr-x 3 root root 0 2010-03-31 19:27 controlC0 Any ideas? Thanks, Fengguang

2 2

[alsa-devel] [PATCH] ALSA: hda: Fix 0 dB offset for Lenovo Thinkpad models using AD1981
by Daniel Chen 31 Mar '10

31 Mar '10

Hi, Attached is a patch against sound-2.6 master HEAD (applicable to 2.6.32.y and 2.6.33.y) that clamps the 0 dB offset accordingly. Best, -Dan

2 1

Re: [alsa-devel] mx31 snd and mc13783 codec status
by Mark Brown 31 Mar '10

31 Mar '10

On Tue, Mar 30, 2010 at 06:23:18PM +0200, Valentin Longchamp wrote: > I have quickly hacked something based on what you had done for > pcm037 with the MC13783 codec in some of your first sound patches. I > know that this is not ready for mainline and I'm still struggling to > debug my code, but I would like to have results quite soon so that > my colleague could see what sounds (especially regarding to the > volume) our system is capable to produce. If you're struggling to find time to do full support for the CODEC a good technique is often to get just very basic fixed function support (say just DAC to headphone support) implemented and merged, then go back and implement I'd also suggest looking at the current Freescale BSPs - obviously this device is used on a lot of Freescale reference boards and I believe there's a reasonably mainlineish driver in the BSPs they have for current devices.

2 1

Re: [alsa-devel] mx31 snd and mc13783 codec status
by Sascha Hauer 31 Mar '10

31 Mar '10

Hi Valentin, On Tue, Mar 30, 2010 at 06:23:18PM +0200, Valentin Longchamp wrote: > Hi Sascha, > > For one of our experiments where our robots need to produce sound, I > would need a sound driver for mx31moboard platform. > > I have quickly hacked something based on what you had done for pcm037 > with the MC13783 codec in some of your first sound patches. I know that > this is not ready for mainline and I'm still struggling to debug my > code, but I would like to have results quite soon so that my colleague > could see what sounds (especially regarding to the volume) our system is > capable to produce. > > I would like to know if by any chance you have some updates on this code > ? Find the latest version of my code here: The following changes since commit 01e77706cdde7c0b47e5ca1f4284a795504c7c40: Linus Torvalds (1): Merge branch 'for-linus' of git://gitorious.org/linux-omap-dss2/linux are available in the git repository at: git://git.pengutronix.de/git/sha/linux-2.6.git mc13783 Sascha Hauer (3): add a mc13783 codec driver add phycore-mc13783 sound support pcm038: add sound support arch/arm/mach-mx2/mach-pcm038.c | 23 ++- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mc13783.c | 727 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/mc13783.h | 32 ++ sound/soc/imx/Kconfig | 9 + sound/soc/imx/Makefile | 3 + sound/soc/imx/phycore-mc13783.c | 160 +++++++++ 8 files changed, 959 insertions(+), 1 deletions(-) create mode 100644 sound/soc/codecs/mc13783.c create mode 100644 sound/soc/codecs/mc13783.h create mode 100644 sound/soc/imx/phycore-mc13783.c > And do you know if your initial mc13783 codec support coupled with > mx31 had some limitations ? Our setup is quite straightforward, we have > direct connection from the mx31 to the mc13783 on a single SSI. Our board uses both SSI channels of the MC13783 which we then mux into one channel in the DAM unit. I don't know how this affects you. Sascha -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

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[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 30 Mar '10

30 Mar '10

The following changes since commit 923a00427a141a6d1c83f1b151221a2458b2de8d: Tejun Heo (1): ASoC: update gfp/slab.h includes are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Barry Song (1): ASoC: ad193x: move codec register/unregister to bus probe/remove Dan Carpenter (1): ASoC: wm8994: playback => capture Graham Gower (1): ASoC: Fix passing platform_data to ac97 bus users and fix a leak Kuninori Morimoto (4): ASoC: fsi: ensures process inside master lock ASoC: fsi: IRQ related process had be united ASoC: fsi: Add FIFO size calculate ASoC: fsi: Add FSI2 device support Mark Brown (2): Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-2.6.35 Merge branch 'for-2.6.34' into for-2.6.35 sound/soc/codecs/ac97.c | 15 ++-- sound/soc/codecs/ad193x.c | 102 ++++++++++-------------- sound/soc/codecs/wm8994.c | 2 +- sound/soc/sh/Kconfig | 3 +- sound/soc/sh/fsi.c | 191 ++++++++++++++++++++++++++++++++------------- sound/soc/soc-core.c | 3 +- 6 files changed, 192 insertions(+), 124 deletions(-)

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[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 30 Mar '10

30 Mar '10

The following changes since commit 3cc4e53f86dab635166929bfa47cc68d59b28c26: Mark Brown (1): ASoC: Remove BROKEN from i.MX audio after dependencies merged are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 A couple of fairly straightforward bug fixes. The ac97 patch is mildly invasive but it tested OK for me (and should be very obvious if it fails). Dan Carpenter (1): ASoC: wm8994: playback => capture Graham Gower (1): ASoC: Fix passing platform_data to ac97 bus users and fix a leak sound/soc/codecs/ac97.c | 15 +++++++++------ sound/soc/codecs/wm8994.c | 2 +- sound/soc/soc-core.c | 3 ++- 3 files changed, 12 insertions(+), 8 deletions(-)

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[alsa-devel] digiface vs. multiface vs. jack vs. other cards
by Fernando Lopez-Lezcano 30 Mar '10

30 Mar '10

At CCRMA we have several studios with linux computers in them (in addition to computers sprinkled throughout the building). Some of the studios have digiface interfaces, some multiface. All of them are connected to the sound system or mixer digitally through adat lightpipes. Workstations have Delta 66 or Gina3G soundcards. It would be nice for a user to be able to play, say, a stereo Ardour session in any of them without any changes to the session or routing. The digifaces send channels 1-24 out the adat ports. The multifaces on the other hand send 9-16 on the adat port. Deltas and Ginas send out stuff on channels 1-4 or 1-6. You can see the multiface interfaces are a problem. To get sound out of them you have to send on channels 9-16 which is different from _all_ others! I tried several workarounds without much success: a) fix this at the alsa level. Should be the way to go, right? Create an /etc/asound.conf that uses ttable to translate channels for the multifaces so that the digital out is actually sent through channels 1-8 (and create the same name plug interface in all other computers that does nothing so that the user can set that name as the default in qjackctl and will get the same behavior in all computers). Works, but not quite. When I try to use that "fake" plug interface with jack2 I get a warning that I'm using a plug interface (sort of fair) and jack only opens it with _2_ i/o channels (even though the ttable is 18 channels wide!). If I _force_ input and output channels in jack to 18 it does work fine. But now it will _not_ work on other computers that have less (or more) than 18 channels. Argh. b) fix it in hdspmixer. Well, it turns out that hdspmixer does not read the default matrix gains from a file so I can't really override the default. AFAICT the gains are calculated algorithmically in the code and there is no command line option to load presets from a file. Or I could write a small script that will set the right gains for 1-8 to go out through 9-16 but if a user starts hdspmixer that gets undone with no warning. And, of course: c) give up and connect an 8 channel analog snake to the analog outputs of the multifaces and do away with a digital connection to the mixer... Anyone has any other suggestions?? This should be easy to fix, right? :-) -- Fernando

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[alsa-devel] [PATCH] ad193x: move codec register/unregister to bus probe/remove
by Barry Song 29 Mar '10

29 Mar '10

The way i've factored out the bus probe and removal functions so that there's no code in the individual I2C and SPI functions means that the register() and unregister() functions could just be squashed into the bus_probe() and bus_remove() functions. Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/codecs/ad193x.c | 102 +++++++++++++++++++------------------------- 1 files changed, 44 insertions(+), 58 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index d034464..e6fe8ac 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -34,8 +34,6 @@ static const u8 ad193x_reg[AD193X_NUM_REGS] = { static struct snd_soc_codec *ad193x_codec; struct snd_soc_codec_device soc_codec_dev_ad193x; -static int ad193x_register(struct ad193x_priv *ad193x, int bus_type); -static void ad193x_unregister(struct ad193x_priv *ad193x); /* * AD193X volume/mute/de-emphasis etc. controls @@ -289,69 +287,23 @@ static int ad193x_bus_probe(struct device *dev, void *ctrl_data, int bus_type) { struct snd_soc_codec *codec; struct ad193x_priv *ad193x; + int ret; + + if (ad193x_codec) { + dev_err(dev, "Another ad193x is registered\n"); + return -EINVAL; + } ad193x = kzalloc(sizeof(struct ad193x_priv), GFP_KERNEL); if (ad193x == NULL) return -ENOMEM; - codec = &ad193x->codec; - codec->control_data = ctrl_data; - codec->dev = dev; - dev_set_drvdata(dev, ad193x); - return ad193x_register(ad193x, bus_type); -} - -static int ad193x_bus_remove(struct device *dev) -{ - struct ad193x_priv *ad193x = dev_get_drvdata(dev); - - ad193x_unregister(ad193x); - return 0; -} - -static struct snd_soc_dai_ops ad193x_dai_ops = { - .hw_params = ad193x_hw_params, - .digital_mute = ad193x_mute, - .set_tdm_slot = ad193x_set_tdm_slot, - .set_fmt = ad193x_set_dai_fmt, -}; - -/* codec DAI instance */ -struct snd_soc_dai ad193x_dai = { - .name = "AD193X", - .playback = { - .stream_name = "Playback", - .channels_min = 2, - .channels_max = 8, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, - }, - .capture = { - .stream_name = "Capture", - .channels_min = 2, - .channels_max = 4, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, - }, - .ops = &ad193x_dai_ops, -}; -EXPORT_SYMBOL_GPL(ad193x_dai); - -static int ad193x_register(struct ad193x_priv *ad193x, int bus_type) -{ - int ret; - struct snd_soc_codec *codec = &ad193x->codec; - - if (ad193x_codec) { - dev_err(codec->dev, "Another ad193x is registered\n"); - return -EINVAL; - } - + codec = &ad193x->codec; mutex_init(&codec->mutex); + codec->control_data = ctrl_data; + codec->dev = dev; codec->private_data = ad193x; codec->reg_cache = ad193x->reg_cache; codec->reg_cache_size = AD193X_NUM_REGS; @@ -412,14 +364,48 @@ static int ad193x_register(struct ad193x_priv *ad193x, int bus_type) return 0; } -static void ad193x_unregister(struct ad193x_priv *ad193x) +static int ad193x_bus_remove(struct device *dev) { + struct ad193x_priv *ad193x = dev_get_drvdata(dev); + snd_soc_unregister_dai(&ad193x_dai); snd_soc_unregister_codec(&ad193x->codec); kfree(ad193x); ad193x_codec = NULL; + + return 0; } +static struct snd_soc_dai_ops ad193x_dai_ops = { + .hw_params = ad193x_hw_params, + .digital_mute = ad193x_mute, + .set_tdm_slot = ad193x_set_tdm_slot, + .set_fmt = ad193x_set_dai_fmt, +}; + +/* codec DAI instance */ +struct snd_soc_dai ad193x_dai = { + .name = "AD193X", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &ad193x_dai_ops, +}; +EXPORT_SYMBOL_GPL(ad193x_dai); + static int ad193x_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); -- 1.5.6.3

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[alsa-devel] Volume levels with Geode LX/CS5536
by Valentin 29 Mar '10

29 Mar '10

Hi, the maximum volume level (100%) on my soundcard for the PCM mixer is 12.0dB instead of 0.0dB. Is there a way for me to fix this, or help fixing it? crimsun on freenode/#alsa told me to ask you guys :) My alsa-info.sh output is at http://0au.de/~apo/p/ai.txt Cheers, Valentin

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[alsa-devel] bug in spi and i2c dual buses codecs
by Barry Song 29 Mar '10

29 Mar '10

Hi Mark/Liam, I think there is really a bug in spi and i2c dual buses codecs drivers, for example, for any codec with two buses, if you config kernel like this: SPI MASTER built-in I2C module Codec built-in You will get a link error like this: sound/built-in.o: In function `i2c_add_driver': include/linux/i2c.h:423: undefined reference to `i2c_register_driver' sound/built-in.o: In function `ad193x_modexit': sound/soc/codecs/ad193x.c:537: undefined reference to `i2c_del_driver' make: *** [.tmp_vmlinux1] Error 1 I remember that is also one reason why we have splitted this kind of driver into three parts(codec core, i2c bus, spi bus). In fact, if any bus is compiled as module and code is built-in, link error will be reported. Thanks Barry

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[alsa-devel] Soundcard Detection problem (ALSA regression)
by Gregory T Flynn 29 Mar '10

29 Mar '10

So far I have tested the current stable drive and now the current snapshot, both to no avail. The last ALSA driver version that worked on this sound card was 1.0.18. General Info from the 1.0.18 driver: !!PCI Soundcards installed in the system !!-------------------------------------- 00:10.1 Audio device: nVidia Corporation MCP51 High Definition Audio (rev a2) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:10.1 0403: 10de:026c (rev a2) Subsystem: 1462:373d Currently I'm running Ubuntu 10.04 beta1 and this sounds regression (which was also in 9.10) is my only problem with the new system. So it would be awesome to get solved and packed into the beta before release. more details in bug report: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4967 -- - Greg Flynn

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[alsa-devel] sound: patch_realtek: truncating bits
by Dan Carpenter 29 Mar '10

29 Mar '10

Smatch complains that we are truncating bits here: sound/pci/hda/patch_realtek.c +12459 alc268_aspire_one_speaker_automute(6) warn: value 28800 can't fit into 255 'bits' sound/pci/hda/patch_realtek.c +13482 alc269_quanta_fl1_speaker_automute(6) warn: value 28800 can't fit into 255 'bits' sound/pci/hda/patch_realtek.c +13511 alc269_lifebook_speaker_automute(11) warn: value 28800 can't fit into 255 'bits' sound/pci/hda/patch_realtek.c +13646 alc269_speaker_automute(8) warn: value 28800 can't fit into 255 'bits' The code looks like this: 13643 unsigned char bits; 13644 13645 present = snd_hda_jack_detect(codec, nid); 13646 bits = present ? AMP_IN_MUTE(0) : 0; "bits" is declared as an unsigned char but AMP_IN_MUTE(0) is 0x7080 so we are only using the last 0x80. I couldn't figure out if it was intended. This was first introduced by this commit which change "bits" from unsigned int to unsigned char. commit 60db6b53fb43421beb2ff3fe3e63412bf81620aa Author: Kailang Yang <kailang(a)realtek.com> Date: Tue Aug 26 13:13:00 2008 +0200 ALSA: hda - Add support of Quanta FL1 Added the support of Quanta FL1 with ALC269 code chip. Also a bit space clean-ups. If this is intentional, it would be cleaner to explicitly cast or mask away the bits we don't care about. regards, dan carpenter

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[alsa-devel] [PATCH] ALSA: hda: Use LPIB for ga-ma770-ud3 board
by Daniel Chen 29 Mar '10

29 Mar '10

Hi, Attached is a patch against current sound-2.6 master HEAD (also applicable to 2.6.32.y and 2.6.33.y) that makes the OR's audio more tolerable with PulseAudio. Best, -Dan

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