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[alsa-devel] ASoC updates for 3.0
by Mark Brown 02 Jun '11

02 Jun '11

The following changes since commit 55922c9d1b84b89cb946c777fddccb3247e7df2c: Linux 3.0-rc1 (2011-05-29 17:43:36 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.40 Mark Brown (1): ASoC: Fix wm_hubs input PGA ZC bits Stephen Warren (1): ASoC: Fix dapm_is_shared_kcontrol so everything isn't shared sound/soc/codecs/wm_hubs.c | 8 ++++---- sound/soc/soc-dapm.c | 5 ++++- 2 files changed, 8 insertions(+), 5 deletions(-)

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[alsa-devel] [PATCH] ASoC: Use a lower detection rate when monitoring headphones on WM8915
by Mark Brown 02 Jun '11

02 Jun '11

We only need to increase the detection rate to maximum if we're monitoring for button presses as the response times needed for user interaction there are much lower. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8915.c | 20 ++++++++++++++------ 1 files changed, 14 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index 5ff6a77..5a59ef7 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -2288,6 +2288,12 @@ static void wm8915_micd(struct snd_soc_codec *codec) SND_JACK_HEADSET | SND_JACK_BTN_0); wm8915->jack_mic = true; wm8915->detecting = false; + + /* Increase poll rate to give better responsiveness + * for buttons */ + snd_soc_update_bits(codec, WM8915_MIC_DETECT_1, + WM8915_MICD_RATE_MASK, + 5 << WM8915_MICD_RATE_SHIFT); } /* If we detected a lower impedence during initial startup @@ -2328,15 +2334,17 @@ static void wm8915_micd(struct snd_soc_codec *codec) SND_JACK_HEADPHONE, SND_JACK_HEADSET | SND_JACK_BTN_0); + + /* Increase the detection rate a bit for + * responsiveness. + */ + snd_soc_update_bits(codec, WM8915_MIC_DETECT_1, + WM8915_MICD_RATE_MASK, + 7 << WM8915_MICD_RATE_SHIFT); + wm8915->detecting = false; } } - - /* Increase poll rate to give better responsiveness for buttons */ - if (!wm8915->detecting) - snd_soc_update_bits(codec, WM8915_MIC_DETECT_1, - WM8915_MICD_RATE_MASK, - 5 << WM8915_MICD_RATE_SHIFT); } static irqreturn_t wm8915_irq(int irq, void *data) -- 1.7.5.3

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Re: [alsa-devel] Question about adding support for new audio codecs from Conexant
by Jaroslav Kysela 02 Jun '11

02 Jun '11

Date 2.6.2011 07:04, Palash Bandyopadhyay wrote: > Dear Jaroslav, > > > > I am a group lead and a developer for Conexant Systems > (www.conexant.com <http://www.conexant.com>). Over the years, we have > been developing quite a range of audio devices. We have also written > ALSA drivers for a few of our I2S/PCM codecs that work with SoCs such as > NVidia’s Tegra, Samsung’s V210, Freescale’s iMx series. > > > > The question that I have is, what is the procedure of adding these > drivers to the main ALSA repository so that users can have access to > this directly rather than to have to contact us? Any suggestions/help in > this direction will be very helpful. Going through the wiki site, I see > that main stream developers might need write access to the git > repository. If that is true, what is the process that needs to be > followed in order to get write access to the repository? Hello, basically, send your patches (try to split changes to small pieces) to the alsa-devel mailing list. We review them and apply to our repository. If you have many patches available, it would be better to create your own kernel GIT tree with these patches and send a summary with the GIT URL to the alsa-devel mailing list. We pull your patches after review. > BTW, I am also a developer for the V4L/LinuxTV community as we > (Conexant) also make capture chips for PCTV and Surveillance. Nice to know. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH v5] [resend] radio-sf16fmr2: convert to generic TEA575x interface
by Ondrej Zary 01 Jun '11

01 Jun '11

Convert radio-sf16fmr2 to use generic TEA575x implementation. Most of the driver code goes away as SF16-FMR2 is basically just a TEA5757 tuner connected to ISA bus. The card can optionally be equipped with PT2254A volume control (equivalent of TC9154AP) - the volume setting is completely reworked (with balance control added) and tested. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-2.6.39-rc2-/sound/pci/Kconfig 2011-05-15 18:50:18.000000000 +0200 +++ linux-2.6.39-rc2/sound/pci/Kconfig 2011-05-17 23:35:30.000000000 +0200 @@ -565,8 +565,8 @@ config SND_FM801_TEA575X_BOOL config SND_TEA575X tristate - depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO - default SND_FM801 || SND_ES1968 + depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2 + default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2 source "sound/pci/hda/Kconfig" --- linux-2.6.39-rc2-/drivers/media/radio/radio-sf16fmr2.c 2011-04-06 03:30:43.000000000 +0200 +++ linux-2.6.39-rc2/drivers/media/radio/radio-sf16fmr2.c 2011-05-19 17:56:08.000000000 +0200 @@ -1,441 +1,209 @@ -/* SF16FMR2 radio driver for Linux radio support - * heavily based on fmi driver... - * (c) 2000-2002 Ziglio Frediano, freddy77(a)angelfire.com +/* SF16-FMR2 radio driver for Linux + * Copyright (c) 2011 Ondrej Zary * - * Notes on the hardware - * - * Frequency control is done digitally -- ie out(port,encodefreq(95.8)); - * No volume control - only mute/unmute - you have to use line volume - * - * For read stereo/mono you must wait 0.1 sec after set frequency and - * card unmuted so I set frequency on unmute - * Signal handling seem to work only on autoscanning (not implemented) - * - * Converted to V4L2 API by Mauro Carvalho Chehab <mchehab(a)infradead.org> + * Original driver was (c) 2000-2002 Ziglio Frediano, freddy77(a)angelfire.com + * but almost nothing remained here after conversion to generic TEA575x + * implementation */ +#include <linux/delay.h> #include <linux/module.h> /* Modules */ #include <linux/init.h> /* Initdata */ #include <linux/ioport.h> /* request_region */ -#include <linux/delay.h> /* udelay */ -#include <linux/videodev2.h> /* kernel radio structs */ -#include <linux/mutex.h> -#include <linux/version.h> /* for KERNEL_VERSION MACRO */ #include <linux/io.h> /* outb, outb_p */ -#include <media/v4l2-device.h> -#include <media/v4l2-ioctl.h> +#include <sound/tea575x-tuner.h> -MODULE_AUTHOR("Ziglio Frediano, freddy77(a)angelfire.com"); -MODULE_DESCRIPTION("A driver for the SF16FMR2 radio."); +MODULE_AUTHOR("Ondrej Zary"); +MODULE_DESCRIPTION("MediaForte SF16-FMR2 FM radio card driver"); MODULE_LICENSE("GPL"); -static int io = 0x384; -static int radio_nr = -1; - -module_param(io, int, 0); -MODULE_PARM_DESC(io, "I/O address of the SF16FMR2 card (should be 0x384, if do not work try 0x284)"); -module_param(radio_nr, int, 0); - -#define RADIO_VERSION KERNEL_VERSION(0,0,2) - -#define AUD_VOL_INDEX 1 - -#undef DEBUG -//#define DEBUG 1 - -#ifdef DEBUG -# define debug_print(s) printk s -#else -# define debug_print(s) -#endif - -/* this should be static vars for module size */ -struct fmr2 -{ - struct v4l2_device v4l2_dev; - struct video_device vdev; - struct mutex lock; +struct fmr2 { int io; - int curvol; /* 0-15 */ - int mute; - int stereo; /* card is producing stereo audio */ - unsigned long curfreq; /* freq in kHz */ - int card_type; + struct snd_tea575x tea; + struct v4l2_ctrl *volume; + struct v4l2_ctrl *balance; }; +/* the port is hardwired so no need to support multiple cards */ +#define FMR2_PORT 0x384 static struct fmr2 fmr2_card; -/* hw precision is 12.5 kHz - * It is only useful to give freq in interval of 200 (=0.0125Mhz), - * other bits will be truncated - */ -#define RSF16_ENCODE(x) ((x) / 200 + 856) -#define RSF16_MINFREQ (87 * 16000) -#define RSF16_MAXFREQ (108 * 16000) - -static inline void wait(int n, int io) -{ - for (; n; --n) - inb(io); -} - -static void outbits(int bits, unsigned int data, int nWait, int io) -{ - int bit; - - for (; --bits >= 0;) { - bit = (data >> bits) & 1; - outb(bit, io); - wait(nWait, io); - outb(bit | 2, io); - wait(nWait, io); - outb(bit, io); - wait(nWait, io); - } -} - -static inline void fmr2_mute(int io) -{ - outb(0x00, io); - wait(4, io); -} - -static inline void fmr2_unmute(int io) -{ - outb(0x04, io); - wait(4, io); -} - -static inline int fmr2_stereo_mode(int io) -{ - int n = inb(io); - - outb(6, io); - inb(io); - n = ((n >> 3) & 1) ^ 1; - debug_print((KERN_DEBUG "stereo: %d\n", n)); - return n; -} - -static int fmr2_product_info(struct fmr2 *dev) -{ - int n = inb(dev->io); - - n &= 0xC1; - if (n == 0) { - /* this should support volume set */ - dev->card_type = 12; - return 0; - } - /* not volume (mine is 11) */ - dev->card_type = (n == 128) ? 11 : 0; - return n; -} - -static inline int fmr2_getsigstr(struct fmr2 *dev) -{ - /* !!! works only if scanning freq */ - int res = 0xffff; - - outb(5, dev->io); - wait(4, dev->io); - if (!(inb(dev->io) & 1)) - res = 0; - debug_print((KERN_DEBUG "signal: %d\n", res)); - return res; -} - -/* set frequency and unmute card */ -static int fmr2_setfreq(struct fmr2 *dev) -{ - unsigned long freq = dev->curfreq; - - fmr2_mute(dev->io); - - /* 0x42 for mono output - * 0x102 forward scanning - * 0x182 scansione avanti - */ - outbits(9, 0x2, 3, dev->io); - outbits(16, RSF16_ENCODE(freq), 2, dev->io); - - fmr2_unmute(dev->io); - - /* wait 0.11 sec */ - msleep(110); - - /* NOTE if mute this stop radio - you must set freq on unmute */ - dev->stereo = fmr2_stereo_mode(dev->io); - return 0; -} - -/* !!! not tested, in my card this does't work !!! */ -static int fmr2_setvolume(struct fmr2 *dev) -{ - int vol[16] = { 0x021, 0x084, 0x090, 0x104, - 0x110, 0x204, 0x210, 0x402, - 0x404, 0x408, 0x410, 0x801, - 0x802, 0x804, 0x808, 0x810 }; - int i, a; - int n = vol[dev->curvol & 0x0f]; - - if (dev->card_type != 11) - return 1; - - for (i = 12; --i >= 0; ) { - a = ((n >> i) & 1) << 6; /* if (a==0) a = 0; else a = 0x40; */ - outb(a | 4, dev->io); - wait(4, dev->io); - outb(a | 0x24, dev->io); - wait(4, dev->io); - outb(a | 4, dev->io); - wait(4, dev->io); - } - for (i = 6; --i >= 0; ) { - a = ((0x18 >> i) & 1) << 6; - outb(a | 4, dev->io); - wait(4, dev->io); - outb(a | 0x24, dev->io); - wait(4, dev->io); - outb(a | 4, dev->io); - wait(4, dev->io); - } - wait(4, dev->io); - outb(0x14, dev->io); - return 0; -} - -static int vidioc_querycap(struct file *file, void *priv, - struct v4l2_capability *v) -{ - strlcpy(v->driver, "radio-sf16fmr2", sizeof(v->driver)); - strlcpy(v->card, "SF16-FMR2 radio", sizeof(v->card)); - strlcpy(v->bus_info, "ISA", sizeof(v->bus_info)); - v->version = RADIO_VERSION; - v->capabilities = V4L2_CAP_TUNER | V4L2_CAP_RADIO; - return 0; -} +/* TEA575x tuner pins */ +#define STR_DATA (1 << 0) +#define STR_CLK (1 << 1) +#define STR_WREN (1 << 2) +#define STR_MOST (1 << 3) +/* PT2254A/TC9154A volume control pins */ +#define PT_ST (1 << 4) +#define PT_CK (1 << 5) +#define PT_DATA (1 << 6) +/* volume control presence pin */ +#define FMR2_HASVOL (1 << 7) -static int vidioc_g_tuner(struct file *file, void *priv, - struct v4l2_tuner *v) +static void fmr2_tea575x_set_pins(struct snd_tea575x *tea, u8 pins) { - struct fmr2 *fmr2 = video_drvdata(file); - - if (v->index > 0) - return -EINVAL; + struct fmr2 *fmr2 = tea->private_data; + u8 bits = 0; - strlcpy(v->name, "FM", sizeof(v->name)); - v->type = V4L2_TUNER_RADIO; - - v->rangelow = RSF16_MINFREQ; - v->rangehigh = RSF16_MAXFREQ; - v->rxsubchans = fmr2->stereo ? V4L2_TUNER_SUB_STEREO : - V4L2_TUNER_SUB_MONO; - v->capability = V4L2_TUNER_CAP_STEREO | V4L2_TUNER_CAP_LOW; - v->audmode = V4L2_TUNER_MODE_STEREO; - mutex_lock(&fmr2->lock); - v->signal = fmr2_getsigstr(fmr2); - mutex_unlock(&fmr2->lock); - return 0; -} + bits |= (pins & TEA575X_DATA) ? STR_DATA : 0; + bits |= (pins & TEA575X_CLK) ? STR_CLK : 0; + /* WRITE_ENABLE is inverted, DATA must be high during read */ + bits |= (pins & TEA575X_WREN) ? 0 : STR_WREN | STR_DATA; -static int vidioc_s_tuner(struct file *file, void *priv, - struct v4l2_tuner *v) -{ - return v->index ? -EINVAL : 0; + outb(bits, fmr2->io); } -static int vidioc_s_frequency(struct file *file, void *priv, - struct v4l2_frequency *f) +static u8 fmr2_tea575x_get_pins(struct snd_tea575x *tea) { - struct fmr2 *fmr2 = video_drvdata(file); + struct fmr2 *fmr2 = tea->private_data; + u8 bits = inb(fmr2->io); - if (f->tuner != 0 || f->type != V4L2_TUNER_RADIO) - return -EINVAL; - if (f->frequency < RSF16_MINFREQ || - f->frequency > RSF16_MAXFREQ) - return -EINVAL; - /* rounding in steps of 200 to match the freq - that will be used */ - fmr2->curfreq = (f->frequency / 200) * 200; - - /* set card freq (if not muted) */ - if (fmr2->curvol && !fmr2->mute) { - mutex_lock(&fmr2->lock); - fmr2_setfreq(fmr2); - mutex_unlock(&fmr2->lock); - } - return 0; + return (bits & STR_DATA) ? TEA575X_DATA : 0 | + (bits & STR_MOST) ? TEA575X_MOST : 0; } -static int vidioc_g_frequency(struct file *file, void *priv, - struct v4l2_frequency *f) +static void fmr2_tea575x_set_direction(struct snd_tea575x *tea, bool output) { - struct fmr2 *fmr2 = video_drvdata(file); - - if (f->tuner != 0) - return -EINVAL; - f->type = V4L2_TUNER_RADIO; - f->frequency = fmr2->curfreq; - return 0; } -static int vidioc_queryctrl(struct file *file, void *priv, - struct v4l2_queryctrl *qc) -{ - struct fmr2 *fmr2 = video_drvdata(file); +static struct snd_tea575x_ops fmr2_tea_ops = { + .set_pins = fmr2_tea575x_set_pins, + .get_pins = fmr2_tea575x_get_pins, + .set_direction = fmr2_tea575x_set_direction, +}; - switch (qc->id) { - case V4L2_CID_AUDIO_MUTE: - return v4l2_ctrl_query_fill(qc, 0, 1, 1, 1); - case V4L2_CID_AUDIO_VOLUME: - /* Only card_type == 11 implements volume */ - if (fmr2->card_type == 11) - return v4l2_ctrl_query_fill(qc, 0, 15, 1, 0); - return v4l2_ctrl_query_fill(qc, 0, 1, 1, 0); - } - return -EINVAL; -} +/* TC9154A/PT2254A volume control */ -static int vidioc_g_ctrl(struct file *file, void *priv, - struct v4l2_control *ctrl) -{ - struct fmr2 *fmr2 = video_drvdata(file); +/* 18-bit shift register bit definitions */ +#define TC9154A_ATT_MAJ_0DB (1 << 0) +#define TC9154A_ATT_MAJ_10DB (1 << 1) +#define TC9154A_ATT_MAJ_20DB (1 << 2) +#define TC9154A_ATT_MAJ_30DB (1 << 3) +#define TC9154A_ATT_MAJ_40DB (1 << 4) +#define TC9154A_ATT_MAJ_50DB (1 << 5) +#define TC9154A_ATT_MAJ_60DB (1 << 6) + +#define TC9154A_ATT_MIN_0DB (1 << 7) +#define TC9154A_ATT_MIN_2DB (1 << 8) +#define TC9154A_ATT_MIN_4DB (1 << 9) +#define TC9154A_ATT_MIN_6DB (1 << 10) +#define TC9154A_ATT_MIN_8DB (1 << 11) +/* bit 12 is ignored */ +#define TC9154A_CHANNEL_LEFT (1 << 13) +#define TC9154A_CHANNEL_RIGHT (1 << 14) +/* bits 15, 16, 17 must be 0 */ + +#define TC9154A_ATT_MAJ(x) (1 << x) +#define TC9154A_ATT_MIN(x) (1 << (7 + x)) + +static void tc9154a_set_pins(struct fmr2 *fmr2, u8 pins) +{ + if (!fmr2->tea.mute) + pins |= STR_WREN; + + outb(pins, fmr2->io); +} + +static void tc9154a_set_attenuation(struct fmr2 *fmr2, int att, u32 channel) +{ + int i; + u32 reg; + u8 bit; + + reg = TC9154A_ATT_MAJ(att / 10) | TC9154A_ATT_MIN((att % 10) / 2); + reg |= channel; + /* write 18-bit shift register, LSB first */ + for (i = 0; i < 18; i++) { + bit = reg & (1 << i) ? PT_DATA : 0; + tc9154a_set_pins(fmr2, bit); + udelay(5); + tc9154a_set_pins(fmr2, bit | PT_CK); + udelay(5); + tc9154a_set_pins(fmr2, bit); + } + + /* latch register data */ + udelay(5); + tc9154a_set_pins(fmr2, PT_ST); + udelay(5); + tc9154a_set_pins(fmr2, 0); +} + +static int fmr2_s_ctrl(struct v4l2_ctrl *ctrl) +{ + struct snd_tea575x *tea = container_of(ctrl->handler, struct snd_tea575x, ctrl_handler); + struct fmr2 *fmr2 = tea->private_data; + int volume, balance, left, right; switch (ctrl->id) { - case V4L2_CID_AUDIO_MUTE: - ctrl->value = fmr2->mute; - return 0; case V4L2_CID_AUDIO_VOLUME: - ctrl->value = fmr2->curvol; - return 0; - } - return -EINVAL; -} - -static int vidioc_s_ctrl(struct file *file, void *priv, - struct v4l2_control *ctrl) -{ - struct fmr2 *fmr2 = video_drvdata(file); - - switch (ctrl->id) { - case V4L2_CID_AUDIO_MUTE: - fmr2->mute = ctrl->value; + volume = ctrl->val; + balance = fmr2->balance->cur.val; break; - case V4L2_CID_AUDIO_VOLUME: - fmr2->curvol = ctrl->value; + case V4L2_CID_AUDIO_BALANCE: + balance = ctrl->val; + volume = fmr2->volume->cur.val; break; default: return -EINVAL; } -#ifdef DEBUG - if (fmr2->curvol && !fmr2->mute) - printk(KERN_DEBUG "unmute\n"); - else - printk(KERN_DEBUG "mute\n"); -#endif - - mutex_lock(&fmr2->lock); - if (fmr2->curvol && !fmr2->mute) { - fmr2_setvolume(fmr2); - /* Set frequency and unmute card */ - fmr2_setfreq(fmr2); - } else - fmr2_mute(fmr2->io); - mutex_unlock(&fmr2->lock); - return 0; -} + left = right = volume; + if (balance < 0) + right = max(0, right + balance); + if (balance > 0) + left = max(0, left - balance); -static int vidioc_g_input(struct file *filp, void *priv, unsigned int *i) -{ - *i = 0; - return 0; -} - -static int vidioc_s_input(struct file *filp, void *priv, unsigned int i) -{ - return i ? -EINVAL : 0; -} + tc9154a_set_attenuation(fmr2, abs(left - 68), TC9154A_CHANNEL_LEFT); + tc9154a_set_attenuation(fmr2, abs(right - 68), TC9154A_CHANNEL_RIGHT); -static int vidioc_g_audio(struct file *file, void *priv, - struct v4l2_audio *a) -{ - a->index = 0; - strlcpy(a->name, "Radio", sizeof(a->name)); - a->capability = V4L2_AUDCAP_STEREO; return 0; } -static int vidioc_s_audio(struct file *file, void *priv, - struct v4l2_audio *a) +static const struct v4l2_ctrl_ops fmr2_ctrl_ops = { + .s_ctrl = fmr2_s_ctrl, +}; + +static int fmr2_tea_ext_init(struct snd_tea575x *tea) { - return a->index ? -EINVAL : 0; -} + struct fmr2 *fmr2 = tea->private_data; -static const struct v4l2_file_operations fmr2_fops = { - .owner = THIS_MODULE, - .unlocked_ioctl = video_ioctl2, -}; + if (inb(fmr2->io) & FMR2_HASVOL) { + fmr2->volume = v4l2_ctrl_new_std(&tea->ctrl_handler, &fmr2_ctrl_ops, V4L2_CID_AUDIO_VOLUME, 0, 68, 2, 56); + fmr2->balance = v4l2_ctrl_new_std(&tea->ctrl_handler, &fmr2_ctrl_ops, V4L2_CID_AUDIO_BALANCE, -68, 68, 2, 0); + if (tea->ctrl_handler.error) { + printk(KERN_ERR "radio-sf16fmr2: can't initialize controls\n"); + return tea->ctrl_handler.error; + } + } -static const struct v4l2_ioctl_ops fmr2_ioctl_ops = { - .vidioc_querycap = vidioc_querycap, - .vidioc_g_tuner = vidioc_g_tuner, - .vidioc_s_tuner = vidioc_s_tuner, - .vidioc_g_audio = vidioc_g_audio, - .vidioc_s_audio = vidioc_s_audio, - .vidioc_g_input = vidioc_g_input, - .vidioc_s_input = vidioc_s_input, - .vidioc_g_frequency = vidioc_g_frequency, - .vidioc_s_frequency = vidioc_s_frequency, - .vidioc_queryctrl = vidioc_queryctrl, - .vidioc_g_ctrl = vidioc_g_ctrl, - .vidioc_s_ctrl = vidioc_s_ctrl, -}; + return 0; +} static int __init fmr2_init(void) { struct fmr2 *fmr2 = &fmr2_card; - struct v4l2_device *v4l2_dev = &fmr2->v4l2_dev; - int res; - strlcpy(v4l2_dev->name, "sf16fmr2", sizeof(v4l2_dev->name)); - fmr2->io = io; - fmr2->stereo = 1; - mutex_init(&fmr2->lock); + fmr2->io = FMR2_PORT; - if (!request_region(fmr2->io, 2, "sf16fmr2")) { - v4l2_err(v4l2_dev, "request_region failed!\n"); + if (!request_region(fmr2->io, 2, "SF16-FMR2")) { + printk(KERN_ERR "radio-sf16fmr2: I/O port 0x%x already in use\n", fmr2->io); return -EBUSY; } - res = v4l2_device_register(NULL, v4l2_dev); - if (res < 0) { - release_region(fmr2->io, 2); - v4l2_err(v4l2_dev, "Could not register v4l2_device\n"); - return res; - } - - strlcpy(fmr2->vdev.name, v4l2_dev->name, sizeof(fmr2->vdev.name)); - fmr2->vdev.v4l2_dev = v4l2_dev; - fmr2->vdev.fops = &fmr2_fops; - fmr2->vdev.ioctl_ops = &fmr2_ioctl_ops; - fmr2->vdev.release = video_device_release_empty; - video_set_drvdata(&fmr2->vdev, fmr2); - - /* mute card - prevents noisy bootups */ - fmr2_mute(fmr2->io); - fmr2_product_info(fmr2); + fmr2->tea.private_data = fmr2; + fmr2->tea.ops = &fmr2_tea_ops; + fmr2->tea.ext_init = fmr2_tea_ext_init; + strlcpy(fmr2->tea.card, "SF16-FMR2", sizeof(fmr2->tea.card)); + strcpy(fmr2->tea.bus_info, "ISA"); - if (video_register_device(&fmr2->vdev, VFL_TYPE_RADIO, radio_nr) < 0) { - v4l2_device_unregister(v4l2_dev); + if (snd_tea575x_init(&fmr2->tea)) { + printk(KERN_ERR "radio-sf16fmr2: Unable to detect TEA575x tuner\n"); release_region(fmr2->io, 2); - return -EINVAL; + return -ENODEV; } - v4l2_info(v4l2_dev, "SF16FMR2 radio card driver at 0x%x.\n", fmr2->io); - debug_print((KERN_DEBUG "card_type %d\n", fmr2->card_type)); + printk(KERN_INFO "radio-sf16fmr2: SF16-FMR2 radio card at 0x%x.\n", fmr2->io); return 0; } @@ -443,22 +211,9 @@ static void __exit fmr2_exit(void) { struct fmr2 *fmr2 = &fmr2_card; - video_unregister_device(&fmr2->vdev); - v4l2_device_unregister(&fmr2->v4l2_dev); + snd_tea575x_exit(&fmr2->tea); release_region(fmr2->io, 2); } module_init(fmr2_init); module_exit(fmr2_exit); - -#ifndef MODULE - -static int __init fmr2_setup_io(char *str) -{ - get_option(&str, &io); - return 1; -} - -__setup("sf16fmr2=", fmr2_setup_io); - -#endif -- Ondrej Zary

3 5

[alsa-devel] [PATCH] ASoC: Update speyside audio driver for hardware revision 2
by Mark Brown 01 Jun '11

01 Jun '11

Revision 2 of the Speyside platform supplies a 32kHz clock on MCLK2 rather than MCLK1. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/speyside.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 360a333..93078b1 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -27,12 +27,12 @@ static int speyside_set_bias_level(struct snd_soc_card *card, switch (level) { case SND_SOC_BIAS_STANDBY: - ret = snd_soc_dai_set_sysclk(codec_dai, WM8915_SYSCLK_MCLK1, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8915_SYSCLK_MCLK2, 32768, SND_SOC_CLOCK_IN); if (ret < 0) return ret; - ret = snd_soc_dai_set_pll(codec_dai, WM8915_FLL_MCLK1, + ret = snd_soc_dai_set_pll(codec_dai, WM8915_FLL_MCLK2, 0, 0, 0); if (ret < 0) { pr_err("Failed to stop FLL\n"); @@ -66,7 +66,7 @@ static int speyside_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_pll(codec_dai, 0, WM8915_FLL_MCLK1, + ret = snd_soc_dai_set_pll(codec_dai, 0, WM8915_FLL_MCLK2, 32768, 256 * 48000); if (ret < 0) return ret; @@ -127,7 +127,7 @@ static int speyside_wm8915_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; int ret; - ret = snd_soc_dai_set_sysclk(dai, WM8915_SYSCLK_MCLK1, 32768, 0); + ret = snd_soc_dai_set_sysclk(dai, WM8915_SYSCLK_MCLK2, 32768, 0); if (ret < 0) return ret; -- 1.7.5.3

2 1

[alsa-devel] [PATCH] ASoC: Remove internally generated WM8915 supplies
by Mark Brown 01 Jun '11

01 Jun '11

DCVDD and MICVDD are intended to be (and almost always are) generated by on-board LDOs which are transparently controlled by the driver so we shouldn't really be requesting them from the regulator API. If the driver is updated to support external supply of these then we will need to change the way we handle this. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8915.c | 8 +------- 1 files changed, 1 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index bb1ff2c..5ff6a77 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -41,14 +41,12 @@ #define HPOUT2L 4 #define HPOUT2R 8 -#define WM8915_NUM_SUPPLIES 6 +#define WM8915_NUM_SUPPLIES 4 static const char *wm8915_supply_names[WM8915_NUM_SUPPLIES] = { - "DCVDD", "DBVDD", "AVDD1", "AVDD2", "CPVDD", - "MICVDD", }; struct wm8915_priv { @@ -113,8 +111,6 @@ WM8915_REGULATOR_EVENT(0) WM8915_REGULATOR_EVENT(1) WM8915_REGULATOR_EVENT(2) WM8915_REGULATOR_EVENT(3) -WM8915_REGULATOR_EVENT(4) -WM8915_REGULATOR_EVENT(5) static const u16 wm8915_reg[WM8915_MAX_REGISTER] = { [WM8915_SOFTWARE_RESET] = 0x8915, @@ -2495,8 +2491,6 @@ static int wm8915_probe(struct snd_soc_codec *codec) wm8915->disable_nb[1].notifier_call = wm8915_regulator_event_1; wm8915->disable_nb[2].notifier_call = wm8915_regulator_event_2; wm8915->disable_nb[3].notifier_call = wm8915_regulator_event_3; - wm8915->disable_nb[4].notifier_call = wm8915_regulator_event_4; - wm8915->disable_nb[5].notifier_call = wm8915_regulator_event_5; /* This should really be moved into the regulator core */ for (i = 0; i < ARRAY_SIZE(wm8915->supplies); i++) { -- 1.7.5.3

2 1

[alsa-devel] [PATCH 2/10] sound/soc/nuc900/nuc900-ac97.c: add missing clk_put
by Julia Lawall 01 Jun '11

01 Jun '11

From: Julia Lawall <julia(a)diku.dk> This goto is after the call to clk_get, so it should go to the label that includes a call to clk_put. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @r exists@ expression e1,e2; statement S; @@ e1 = clk_get@p1(...); ... when != e1 = e2 when != clk_put(e1) when any if (...) { ... when != clk_put(e1) when != if (...) { ... clk_put(e1) ... } * return@p3 ...; } else S // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/nuc900/nuc900-ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index dac6732..9c0edad 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -356,7 +356,7 @@ static int __devinit nuc900_ac97_drvprobe(struct platform_device *pdev) nuc900_audio->irq_num = platform_get_irq(pdev, 0); if (!nuc900_audio->irq_num) { ret = -EBUSY; - goto out2; + goto out3; } nuc900_ac97_data = nuc900_audio;

3 2

[alsa-devel] [PATCH] ASoC: Support edge triggered IRQs for WM8915
by Mark Brown 01 Jun '11

01 Jun '11

Really this should be something the IRQ core can cope with for us but since it doesn't currently do so (at least for threaded interrupts like this) do so in the driver. This allows us to run with interrupt controllers that only support edge triggered interrupts. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8915.c | 24 ++++++++++++++++++++++-- 1 files changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index a0b1a72..bb1ff2c 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -2382,6 +2382,20 @@ static irqreturn_t wm8915_irq(int irq, void *data) } } +static irqreturn_t wm8915_edge_irq(int irq, void *data) +{ + irqreturn_t ret = IRQ_NONE; + irqreturn_t val; + + do { + val = wm8915_irq(irq, data); + if (val != IRQ_NONE) + ret = val; + } while (val != IRQ_NONE); + + return ret; +} + static void wm8915_retune_mobile_pdata(struct snd_soc_codec *codec) { struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); @@ -2708,8 +2722,14 @@ static int wm8915_probe(struct snd_soc_codec *codec) irq_flags |= IRQF_ONESHOT; - ret = request_threaded_irq(i2c->irq, NULL, wm8915_irq, - irq_flags, "wm8915", codec); + if (irq_flags & (IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING)) + ret = request_threaded_irq(i2c->irq, NULL, + wm8915_edge_irq, + irq_flags, "wm8915", codec); + else + ret = request_threaded_irq(i2c->irq, NULL, wm8915_irq, + irq_flags, "wm8915", codec); + if (ret == 0) { /* Unmask the interrupt */ snd_soc_update_bits(codec, WM8915_INTERRUPT_CONTROL, -- 1.7.5.3

2 1

[alsa-devel] [PATCH] ASoC: Staticize ak4641_dai
by Mark Brown 01 Jun '11

01 Jun '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/ak4641.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index ed96f247c..7a64e58 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -457,7 +457,7 @@ static struct snd_soc_dai_ops ak4641_pcm_dai_ops = { .set_sysclk = ak4641_set_dai_sysclk, }; -struct snd_soc_dai_driver ak4641_dai[] = { +static struct snd_soc_dai_driver ak4641_dai[] = { { .name = "ak4641-hifi", .id = 1, -- 1.7.5.3

2 1

[alsa-devel] HDA Intel/ALC269 microphone problem
by Mathieu Bouffard 01 Jun '11

01 Jun '11

Read a few bug reports (going back to 2009!) around where users couldn't use their microphone with HDA Intel/ALC269. I had one set up JACK for 2 playback channels and 1 capture channel, make an .asoundrc to route everything to pulse which is started with a configuration only using module-jack-sink and module-jack-source, start it up and the microphone and the whole setup worked. But that's quite a hassle for a workaround! I had suspicions that it was the stereo mic causing problem and forcing mono would work and it seems to be the case. I attached a patch, hoping that the fix might be as simple as forcing mono at the ALSA level, but I don't think it'll be that simple... Can someone more experienced see if more is needed, and if not then commit it? (orig file is from 3.0-rc1 source) -- Mathieu Bouffard<mbouffard fedoraproject org>

1 0

[alsa-devel] hda: MacBookPro7, 1: no sound on left channel of 'Surround Speaker'
by Mourad De Clerck 01 Jun '11

01 Jun '11

Hi, My MacBookPro7,1 (nVidia MCP89) has a problem with the sound on its internal speakers. The 'Front Speaker' works properly, but is quite "thin" and quiet. The 'Surround Speaker' completes the volume level properly, but only the right channel works. Switching 'Surround Speaker' on/off for both left/right, or manipulating the 'Surround Speaker Playback Volume' only has an effect on the right channel, the left stays silent. It'd be nice if someone could provide me a patch or some guidance for patch_cirrus.c so we can fix this. Thanks, Mourad PS: here are the relevant mixer controls. (is there a reason why the surround switch and volume are separated?) -- Simple mixer control 'Front Speaker',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 115 Mono: Front Left: Playback 115 [100%] [0.00dB] [on] Front Right: Playback 115 [100%] [0.00dB] [on] Simple mixer control 'Surround Speaker',0 Capabilities: pswitch penum Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'Surround Speaker Playback Volum',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 115 Front Left: 115 [100%] Front Right: 115 [100%]

1 0

[alsa-devel] ASoC: Choppy audio on OMAP using McBSP3 interface
by Pedro Sanchez 31 May '11

31 May '11

Hello, I wrote a ASoC driver for a TI DM3730-based custom board interfacing a TI TAS5709 codec over the McBSP2 interface. I'm using the Linux kernel 2.6.32. The driver works very well, no problems with the sound quality at all. Unfortunately due to reasons alien to me I needed to move the driver to work using the McBSP3 interface instead. All I did was then to change the pointer in my machine code to point to the McBSP3 interface: static struct snd_soc_dai_link bp_audio_dai = { .name = "tas5709", .stream_name = "tas5709", .cpu_dai = &omap_mcbsp_dai[2], <-- from [1] to [2] .codec_dai = &tas5709_dai, .init = bp_audio_init, .ops = &bp_audio_ops, }; and in my machine init code: *(unsigned int *)bp_audio_dai.cpu_dai->private_data = 2; With this change the audio driver works well when playing local audio files via the aplay command. But it fails miserably when playing streaming audio from a VoIP connection, the audio is very choppy as it happens when there is a large packet loss. However audio is mono 16 bits @8 KHz, not really very demanding. I realize there is a big difference in the internal buffer size for the two interfaces, the McBSP2 having 10 times the buffer space of the McBSP3. I suspect this is somehow the source of my problems. I'm inclined to believe that there is some DMA-related optimization issue that has to be addressed somehow, but this is just a guess. I fail to understand why is it that I can play local audio files but not streaming audio. What makes the difference? Any suggestions on how to address this issue? Thank you, -- Pedro

1 1

[alsa-devel] [PATCH 2/2] ALSA: ASoc: putting together AT91SAM9260 and TLV320AIC3X
by Prchal Jiří 31 May '11

31 May '11

Hi, this patch is for example how to put together AT91SAM9260 and TLV320AIC3106 controlled via SPI. It tooks me a lot of time to make it working, so I think it could be helpfull for other people. In original source "snd-soc-afeb9260.c" which I use as example is BIG ERROR: In function "afeb9260_soc_init" is missing call of "atmel_ssc_set_audio(0);". It cause bug "PROBLEM: Asoc driver in 2.6.37.3 for AT91SAM9260 / TLV320AIC3X is broken" which I post some time ago. Depend on: [PATCH 1/2] ALSA: ASoc: TLV320AIC3X: ad SPI and clock on GPIO2 or BCLK Kernel version: 2.6.38 Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- diff -uprN -X linux-2.6.38-vanilla/Documentation/dontdiff linux-2.6.38-vanilla/arch/arm/mach-at91/board-cdu.c /home/prchal/arm/fw-cdu/linux/linux-2.6.38/arch/arm/mach-at91/board-cdu.c --- linux-2.6.38-vanilla/arch/arm/mach-at91/board-cdu.c 1970-01-01 01:00:00.000000000 +0100 +++ /home/prchal/arm/fw-cdu/linux/linux-2.6.38/arch/arm/mach-at91/board-cdu.c 2011-03-24 09:15:23.852370981 +0100 @@ -0,0 +1,345 @@ +/* + * linux/arch/arm/mach-at91/board-cdu.c + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2006 Atmel + * Copyright (C) 2011 AKsignal Brno + * jiri.prchal(a)aksignal.cz + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/types.h> +#include <linux/init.h> +#include <linux/mm.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/clk.h> +#include <linux/spi/mcp23s08.h> +#include <linux/dma-mapping.h> + +#include <mach/hardware.h> +#include <asm/setup.h> +#include <asm/mach-types.h> +#include <asm/irq.h> + +#include <asm/mach/arch.h> +#include <asm/mach/map.h> +#include <asm/mach/irq.h> + +#include <mach/board.h> +#include <mach/gpio.h> +#include <mach/at91sam9_smc.h> + +#include "sam9_smc.h" +#include "generic.h" + +static void __init cdu_map_io(void) +{ + /* Initialize processor: 6 MHz crystal */ + at91sam9260_initialize(6000000); + + /* DBGU on ttyS0. (Rx & Tx only) */ + at91_register_uart(0, 0, 0); + + /* USART0 on ttyS1. (Rx, Tx, CTS, RTS, DTR, DSR, DCD, RI) */ + at91_register_uart(AT91SAM9260_ID_US0, 1, ATMEL_UART_CTS | ATMEL_UART_RTS + | ATMEL_UART_DTR | ATMEL_UART_DSR | ATMEL_UART_DCD + | ATMEL_UART_RI); + + /* USART1 on ttyS2. (Rx, Tx, RTS) */ + at91_register_uart(AT91SAM9260_ID_US1, 2, ATMEL_UART_RTS); + + /* USART2 on ttyS3. (Rx, Tx, RTS) */ + at91_register_uart(AT91SAM9260_ID_US2, 3, ATMEL_UART_RTS); + + /* USART3 on ttyS4. (Rx, Tx, RTS) */ + at91_register_uart(AT91SAM9260_ID_US3, 4, ATMEL_UART_RTS); + + /* USART4 on ttyS5. (Tx) */ + /* EBUS - transmit audio, no rx */ + /*at91_register_uart(AT91SAM9260_ID_US4, 5, 0);*/ + + /* set serial console to ttyS0 (ie, DBGU) */ + at91_set_serial_console(0); +} + +static void __init cdu_init_irq(void) +{ + at91sam9260_init_interrupts(NULL); +} + +/* + * USB Host port + */ +static struct at91_usbh_data __initdata cdu_usbh_data = { + .ports = 2, +}; + +/* + * USB Device port + */ +static struct at91_udc_data __initdata cdu_udc_data = { + .vbus_pin = AT91_PIN_PC15, + .pullup_pin = 0, /* pull-up driven by UDC */ +}; + +/* + * SPI devices. + */ +#define MCP23S08_GPIO_BASE 128 + +static const struct mcp23s08_platform_data mcp23s08_gpio_info = { + .chip[0].is_present = true, + .chip[0].pullups = 0, + .base = MCP23S08_GPIO_BASE, +}; + +static struct spi_board_info cdu_spi_devices[] = { + { /* ADC LTC2488 */ + .modalias = "spidev", + .chip_select = 0, + .max_speed_hz = 1 * 1000 * 1000, + .bus_num = 1, + }, + { /* GPIO expander MCP23S08 */ + .modalias = "mcp23s08", + .chip_select = 1, + .max_speed_hz = 1000000, + .bus_num = 1, + .platform_data = &mcp23s08_gpio_info, + .mode = SPI_MODE_3, + }, + { /* non volatile memory (F-RAM) FM25VN10 */ + .modalias = "spidev", + .chip_select = 2, + .max_speed_hz = 1 * 1000 * 1000, + .bus_num = 1, + }, + { /* audiocodec TLV320AIC3106 */ + .modalias = "tlv320aic3x-codec", + .chip_select = 3, + .max_speed_hz = 1000000, + .bus_num = 1, + .mode = SPI_MODE_1, + }, +}; + +/* + * MACB Ethernet device + */ +static struct at91_eth_data __initdata cdu_macb_data = { + .phy_irq_pin = 0, //nc, + .is_rmii = 1, +}; + +/* + * NAND flash + */ +static struct mtd_partition __initdata cdu_nand_partition[] = { + { + .name = "bootstrap", + .offset = 0, + .size = 0x40000, + }, + { + .name = "uboot", + .offset = 0x40000, + .size = (0xc0000 - 0x40000), + }, + { + .name = "ubootenv", + .offset = 0xc0000, + .size = (0x100000 - 0xc0000), + }, + { + .name = "kernel", + .offset = 0x100000, + .size = (0x400000 - 0x100000), + }, + { + .name = "rootfs", + .offset = 0x400000, + .size = (0x8000000 - 0x400000), + }, +}; + +static struct mtd_partition * __init nand_partitions(int size, int *num_partitions) +{ + *num_partitions = ARRAY_SIZE(cdu_nand_partition); + return cdu_nand_partition; +} + +static struct atmel_nand_data __initdata cdu_nand_data = { + .ale = 21, + .cle = 22, +// .det_pin = ... not connected + .rdy_pin = AT91_PIN_PC13, + .enable_pin = AT91_PIN_PC14, + .partition_info = nand_partitions, + .bus_width_16 = 0, +}; + +static struct sam9_smc_config __initdata cdu_nand_smc_config = { + .ncs_read_setup = 0, + .nrd_setup = 1, + .ncs_write_setup = 0, + .nwe_setup = 1, + + .ncs_read_pulse = 3, + .nrd_pulse = 3, + .ncs_write_pulse = 3, + .nwe_pulse = 3, + + .read_cycle = 5, + .write_cycle = 5, + + .mode = AT91_SMC_READMODE | AT91_SMC_WRITEMODE | AT91_SMC_EXNWMODE_DISABLE, + .tdf_cycles = 2, +}; + +static void __init cdu_add_device_nand(void) +{ + /* setup bus-width (8 or 16) */ + if (cdu_nand_data.bus_width_16) + cdu_nand_smc_config.mode |= AT91_SMC_DBW_16; + else + cdu_nand_smc_config.mode |= AT91_SMC_DBW_8; + + /* configure chip-select 3 (NAND) */ + sam9_smc_configure(3, &cdu_nand_smc_config); + + at91_add_device_nand(&cdu_nand_data); +} + +/* +* MCI (SD/MMC) +*/ +static struct at91_mmc_data __initdata ek_mmc_data = { + .slot_b = 0, + .wire4 = 1, + // .det_pin = ... not connected + // .wp_pin = ... not connected + // .vcc_pin = ... not connected +}; + +/* + * LEDs + */ +static struct gpio_led cdu_leds[] = { + { + .name = "red", + .gpio = AT91_PIN_PC10, + .default_trigger = "timer", + }, + { + .name = "green", + .gpio = AT91_PIN_PA5, + .default_trigger = "heartbeat", + }, + { + .name = "yellow", + .gpio = AT91_PIN_PB20, + .active_low = 1, + .default_trigger = "mmc0", + }, + { + .name = "blue", + .gpio = AT91_PIN_PB21, + .active_low = 1, + .default_trigger = "nand-disk", + }, +}; + +/* + * GPIOs + */ +static struct gpio gpios[] = { + { AT91_PIN_PA0, GPIOF_OUT_INIT_LOW, "ebus_dir"}, + { AT91_PIN_PA1, GPIOF_OUT_INIT_LOW, "time_dir"}, + { AT91_PIN_PB12, GPIOF_OUT_INIT_LOW, "gsm_rst"}, + { AT91_PIN_PB13, GPIOF_OUT_INIT_LOW, "gsm_on"}, + { AT91_PIN_PC2, GPIOF_IN, "por"}, + { AT91_PIN_PC7, GPIOF_OUT_INIT_HIGH, "spk_shdn"}, + + { AT91_PIN_PC0, GPIOF_IN, "in13"}, + { AT91_PIN_PC1, GPIOF_IN, "in14"}, + { AT91_PIN_PA22, GPIOF_IN, "in17"}, + { AT91_PIN_PA23, GPIOF_IN, "in18"}, + { AT91_PIN_PA24, GPIOF_IN, "in19"}, + { AT91_PIN_PA25, GPIOF_IN, "in20"}, + + { AT91_PIN_PA26, GPIOF_OUT_INIT_LOW, "out15"}, + { AT91_PIN_PA27, GPIOF_OUT_INIT_LOW, "out16"}, + { AT91_PIN_PA28, GPIOF_OUT_INIT_LOW, "out17"}, + { AT91_PIN_PA29, GPIOF_OUT_INIT_LOW, "out18"}, + { AT91_PIN_PA30, GPIOF_OUT_INIT_LOW, "out19"}, + { AT91_PIN_PB29, GPIOF_OUT_INIT_LOW, "out20"}, + + { MCP23S08_GPIO_BASE + 0, GPIOF_IN, "busadrp"}, + { MCP23S08_GPIO_BASE + 2, GPIOF_IN, "busadr5"}, + { MCP23S08_GPIO_BASE + 3, GPIOF_IN, "busadr4"}, + { MCP23S08_GPIO_BASE + 4, GPIOF_IN, "busadr3"}, + { MCP23S08_GPIO_BASE + 5, GPIOF_IN, "busadr2"}, + { MCP23S08_GPIO_BASE + 6, GPIOF_IN, "busadr1"}, + { MCP23S08_GPIO_BASE + 7, GPIOF_IN, "busadr0"}, +}; + +static void __init cdu_add_gpio (void) +{ + int i; + + for (i = 0; i < ARRAY_SIZE (gpios); i++) { + gpio_request_one (gpios[i].gpio, gpios[i].flags, gpios[i].label); + } +} + +/* + * init + */ +static void __init cdu_board_init(void) +{ + /* Serial */ + at91_add_device_serial(); + /* USB Host */ + at91_add_device_usbh(&cdu_usbh_data); + /* USB Device */ + at91_add_device_udc(&cdu_udc_data); + /* SPI */ + at91_add_device_spi(cdu_spi_devices, ARRAY_SIZE(cdu_spi_devices)); + /* NAND */ + cdu_add_device_nand(); + /* Ethernet */ + at91_add_device_eth(&cdu_macb_data); + /* MMC */ + at91_add_device_mmc(0, &ek_mmc_data); + /* SSC */ + at91_add_device_ssc(AT91SAM9260_ID_SSC, (ATMEL_SSC_TF | ATMEL_SSC_TK | ATMEL_SSC_TD | ATMEL_SSC_RD)); + /* LEDs */ + at91_gpio_leds(cdu_leds, ARRAY_SIZE(cdu_leds)); + // GPIO + cdu_add_gpio (); +} + +MACHINE_START(AT91SAM9260EK, "CDU") + /* Maintainer: Atmel */ + .boot_params = AT91_SDRAM_BASE + 0x100, + .timer = &at91sam926x_timer, + .map_io = cdu_map_io, + .init_irq = cdu_init_irq, + .init_machine = cdu_board_init, +MACHINE_END diff -uprN -X linux-2.6.38-vanilla/Documentation/dontdiff linux-2.6.38-vanilla/arch/arm/mach-at91/Kconfig /home/prchal/arm/fw-cdu/linux/linux-2.6.38/arch/arm/mach-at91/Kconfig --- linux-2.6.38-vanilla/arch/arm/mach-at91/Kconfig 2011-03-15 02:20:32.000000000 +0100 +++ /home/prchal/arm/fw-cdu/linux/linux-2.6.38/arch/arm/mach-at91/Kconfig 2011-03-16 09:19:40.000000000 +0100 @@ -213,6 +213,12 @@ config MACH_AT91SAM9260EK Select this if you are using Atmel's AT91SAM9260-EK or AT91SAM9XE Evaluation Kit <http://www.atmel.com/dyn/products/tools_card.asp?tool_id=3933> +config MACH_CDU + bool "CDU AKsignal board" + help + Select this if you are using AKsignal CDU board + <http://www.aksignal.cz> + config MACH_CAM60 bool "KwikByte KB9260 (CAM60) board" help diff -uprN -X linux-2.6.38-vanilla/Documentation/dontdiff linux-2.6.38-vanilla/arch/arm/mach-at91/Makefile /home/prchal/arm/fw-cdu/linux/linux-2.6.38/arch/arm/mach-at91/Makefile --- linux-2.6.38-vanilla/arch/arm/mach-at91/Makefile 2011-03-15 02:20:32.000000000 +0100 +++ /home/prchal/arm/fw-cdu/linux/linux-2.6.38/arch/arm/mach-at91/Makefile 2011-03-16 09:19:40.000000000 +0100 @@ -40,6 +40,7 @@ obj-$(CONFIG_MACH_ECO920) += board-eco92 # AT91SAM9260 board-specific support obj-$(CONFIG_MACH_AT91SAM9260EK) += board-sam9260ek.o +obj-$(CONFIG_MACH_CDU) += board-cdu.o obj-$(CONFIG_MACH_CAM60) += board-cam60.o obj-$(CONFIG_MACH_SAM9_L9260) += board-sam9-l9260.o obj-$(CONFIG_MACH_USB_A9260) += board-usb-a9260.o diff -uprN -X linux-2.6.38-vanilla/Documentation/dontdiff linux-2.6.38-vanilla/sound/soc/atmel/Kconfig /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/atmel/Kconfig --- linux-2.6.38-vanilla/sound/soc/atmel/Kconfig 2011-03-15 02:20:32.000000000 +0100 +++ /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/atmel/Kconfig 2011-03-22 09:46:46.751566158 +0100 @@ -50,3 +50,11 @@ config SND_AT91_SOC_AFEB9260 select SND_SOC_TLV320AIC23 help Say Y here to support sound on AFEB9260 board. + +config SND_AT91_SOC_CDU + tristate "SoC Audio support for CDU board" + depends on ARCH_AT91 && MACH_CDU && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_SOC_TLV320AIC3X + help + Say Y here to support sound on CDU board. diff -uprN -X linux-2.6.38-vanilla/Documentation/dontdiff linux-2.6.38-vanilla/sound/soc/atmel/Makefile /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/atmel/Makefile --- linux-2.6.38-vanilla/sound/soc/atmel/Makefile 2011-03-15 02:20:32.000000000 +0100 +++ /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/atmel/Makefile 2011-03-16 09:26:17.000000000 +0100 @@ -14,3 +14,4 @@ snd-soc-playpaq-objs := playpaq_wm8510.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o +obj-$(CONFIG_SND_AT91_SOC_CDU) += snd-soc-cdu.o diff -uprN -X linux-2.6.38-vanilla/Documentation/dontdiff linux-2.6.38-vanilla/sound/soc/atmel/snd-soc-cdu.c /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/atmel/snd-soc-cdu.c --- linux-2.6.38-vanilla/sound/soc/atmel/snd-soc-cdu.c 1970-01-01 01:00:00.000000000 +0100 +++ /home/prchal/arm/fw-cdu/linux/linux-2.6.38/sound/soc/atmel/snd-soc-cdu.c 2011-03-24 09:27:55.404367652 +0100 @@ -0,0 +1,264 @@ +/* + * snd-soc-cdu -- SoC audio for AT91SAM9260-based + * AKsignal CDU board. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * Copyright (C) 2011 AK signal Brno + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * Jiri Prchal <jiri.prchal(a)aksignal.cz> + * + * Based on ati_b1_wm8731.c by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * Based on corgi.c by: + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> + +#include <linux/atmel-ssc.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> + +#include "../codecs/tlv320aic3x.h" +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + +struct { + unsigned int channels; + snd_pcm_format_t format; + unsigned int rate; + unsigned int codecclk; + unsigned int cmrdiv; + unsigned int period; +} cdu_audio[] = { + /* 16 bit stereo modes */ + {2, SNDRV_PCM_FORMAT_S16_LE, 8000, 2096000, 25, 130,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 16000, 2496000, 21, 77,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 32000, 2496000, 21, 38,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 48000, 2016000, 26, 20,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 96000, 4032000, 13, 20,}, + + {2, SNDRV_PCM_FORMAT_S16_LE, 11025, 2381400, 22, 107,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 22050, 2381400, 22, 53,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 44100, 2381400, 22, 26,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 88200, 4762800, 11, 26,}, + + {2, SNDRV_PCM_FORMAT_S16_LE, 11520, 2626560, 20, 113,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 23040, 2626560, 20, 56,}, + +}; + +static int cdu_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret; + int i, found = 0; + snd_pcm_format_t format = params_format(params); + unsigned int rate = params_rate(params); + unsigned int channels = params_channels(params); + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* find the correct audio parameters */ + for (i = 0; i < ARRAY_SIZE(cdu_audio); i++) { + if (rate == cdu_audio[i].rate && + format == cdu_audio[i].format && + channels == cdu_audio[i].channels) { + found = 1; + break; + } + } + if (!found) + return -EINVAL; + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, CLKIN_BCLK, cdu_audio[i].codecclk, SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + /* Set the cpu clock dividers to BCLK */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_CMR_DIV, cdu_audio[i].cmrdiv); + ret |= snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_TCMR_PERIOD, cdu_audio[i].period); + ret |= snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_RCMR_PERIOD, cdu_audio[i].period); + if (ret < 0) { + printk(KERN_ERR "can't set cpu system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops cdu_ops = { + .hw_params = cdu_hw_params, +}; + +static const struct snd_soc_dapm_widget cdu_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Headphone connected to HPLOUT, HPROUT */ + {"Headphone Jack", NULL, "HPLOUT"}, + {"Headphone Jack", NULL, "HPROUT"}, + + /* Line Out connected to LLOUT, RLOUT */ + {"Line Out", NULL, "LLOUT"}, + {"Line Out", NULL, "RLOUT"}, + + /* Mic connected to (MIC3L | MIC3R) */ + {"MIC3L", NULL, "Mic Bias 2V"}, + {"MIC3R", NULL, "Mic Bias 2V"}, + {"Mic Bias 2V", NULL, "Mic Jack"}, + + /* Line In connected to (LINE1L | LINE2L), (LINE1R | LINE2R) */ + {"LINE1L", NULL, "Line In"}, + {"LINE2L", NULL, "Line In"}, + {"LINE1R", NULL, "Line In"}, + {"LINE2R", NULL, "Line In"}, +}; + +/* + * Logic for a aic3x as connected on a cdu board. + */ +static int cdu_aic3x_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + printk(KERN_DEBUG "cdu_aic3x: cdu_aic3x_init called\n"); + + /* Add specific widgets */ + snd_soc_dapm_new_controls(dapm, cdu_dapm_widgets, + ARRAY_SIZE(cdu_dapm_widgets)); + /* Set up specific audio path interconnects */ + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); + + /* always connected */ + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Line Out"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); + snd_soc_dapm_enable_pin(dapm, "Line In"); + + snd_soc_dapm_sync(dapm); + + return 0; +} + +static struct snd_soc_dai_link cdu_dai = { + .name = "TLV320AIC3106", + .stream_name = "PCM", + .cpu_dai_name = "atmel-ssc-dai.0", + .codec_dai_name = "tlv320aic3x-hifi", + .init = cdu_aic3x_init, + .platform_name = "atmel-pcm-audio", +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + .codec_name = "tlv320aic3x-codec.0-001b", +#endif +#if defined(CONFIG_SPI_MASTER) + .codec_name = "spi1.3", +#endif + .ops = &cdu_ops, +}; + +static struct snd_soc_card snd_soc_cdu = { + .name = "TLV320AIC3106", + .dai_link = &cdu_dai, + .num_links = 1, +}; + +static struct platform_device *cdu_snd_device; + +static int __init cdu_init(void) +{ + struct clk *pllb; + int ret; + + ret = atmel_ssc_set_audio(0); + if (ret != 0) { + pr_err("Failed to set SSC 0 for audio: %d\n", ret); + return ret; + } + + cdu_snd_device = platform_device_alloc("soc-audio", -1); + if (!cdu_snd_device) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + ret = -ENOMEM; + } + + platform_set_drvdata(cdu_snd_device, &snd_soc_cdu); + + ret = platform_device_add(cdu_snd_device); + if (ret) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + goto err_device_add; + } + + return ret; + +err_device_add: + platform_device_put(cdu_snd_device); +err: + return ret; +} + +static void __exit cdu_exit(void) +{ + platform_device_unregister(cdu_snd_device); + cdu_snd_device = NULL; +} + +module_init(cdu_init); +module_exit(cdu_exit); + +/* Module information */ +MODULE_AUTHOR("Jiri Prchal <jiri.prchal(a)aksignal.cz>"); +MODULE_DESCRIPTION("ALSA SoC CDU_AIC3X"); +MODULE_LICENSE("GPL");

2 4

Re: [alsa-devel] AudioArduino - an ALSA soundcard driver for FTDI-based Arduinos
by Smilen Dimitrov 31 May '11

31 May '11

Hi Sid, > Are there any possibilities of extending it beyond 8-bit / 44.1 kHz? The driver currently exposes a stereo, 16-bit, 44.1 kHz on the ALSA side, since that is the lowest quality with which Audacity works in the playback direction. However, Arduino Duemillanove's ATMega328 is not really capable of 16-bit reproduction in this context: it has one 16-bit and two 8-bit counters, and the 16-bit one is used to derive the 44.1 kHz (PCM) rate, so the only thing available for PWM (analog) reproduction are the 8-bit counters (there is a bit more on this in the paper on the webpage). So what happens during stereo, 16-bit, 44.1 kHz playback, is that the driver simply extracts the MSB of the left channel, and sends it as an 8-bit sample representation to the Arduino for PWM reproduction. So in brief - as far as ALSA interfaces are concerned, I believe it is possible to specify any rate you want (which is supported) - however, "by default", it would be easy to only get 8-bit resolution (hardware) reproduction on the Arduino (however, higher rates should in principle be possible; as I think there should still some of bandwidth left, since a duplex operation can be demonstrated at 2Mbps). Hope this helps, Cheers! > Regards > Sid. >

2 1

[alsa-devel] [PATCH] ASoC: Remove redundant freq assignment for max98095->sysclk/max98088->sysclk
by Axel Lin 28 May '11

28 May '11

Current implementation set max98095->sysclk/max98088->sysclk to freq twice. Set it once is enough, this patch removes the first assignment in case we may set invalid clock frequency to max98095->sysclk/max98088->sysclk. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- hi Peter, I don't have this hardware handy. Can you help to test it? Thanks, Axel sound/soc/codecs/max98088.c | 2 -- sound/soc/codecs/max98095.c | 2 -- 2 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 4173b67..ac65a2d 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1397,8 +1397,6 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai, if (freq == max98088->sysclk) return 0; - max98088->sysclk = freq; /* remember current sysclk */ - /* Setup clocks for slave mode, and using the PLL * PSCLK = 0x01 (when master clk is 10MHz to 20MHz) * 0x02 (when master clk is 20MHz to 30MHz).. diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index e1d282d..872a5fa 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1517,8 +1517,6 @@ static int max98095_dai_set_sysclk(struct snd_soc_dai *dai, if (freq == max98095->sysclk) return 0; - max98095->sysclk = freq; /* remember current sysclk */ - /* Setup clocks for slave mode, and using the PLL * PSCLK = 0x01 (when master clk is 10MHz to 20MHz) * 0x02 (when master clk is 20MHz to 40MHz).. -- 1.7.1

4 3

[alsa-devel] i8253 locking and PC speaker on non-{mips,x86}
by Ralf Baechle 28 May '11

28 May '11

No longer terribly relevant these days but ... drivers/input/misc/pcspkr.c: #if defined(CONFIG_MIPS) || defined(CONFIG_X86) /* Use the global PIT lock ! */ #include <asm/i8253.h> #else #include <asm/8253pit.h> static DEFINE_RAW_SPINLOCK(i8253_lock); #endif sound/drivers/pcsp/pcsp.h: #if defined(CONFIG_MIPS) || defined(CONFIG_X86) /* Use the global PIT lock ! */ #include <asm/i8253.h> #else #include <asm/8253pit.h> static DEFINE_RAW_SPINLOCK(i8253_lock); $ git grep -F pcsp.h sound/drivers/pcsp sound/drivers/pcsp/pcsp.c:#include "pcsp.h" sound/drivers/pcsp/pcsp_input.c:#include "pcsp.h" sound/drivers/pcsp/pcsp_lib.c:#include "pcsp.h" sound/drivers/pcsp/pcsp_mixer.c:#include "pcsp.h" $ git grep -w i8253_lock sound/drivers/pcsp/ sound/drivers/pcsp/pcsp.h:static DEFINE_RAW_SPINLOCK(i8253_lock); sound/drivers/pcsp/pcsp_input.c: raw_spin_lock_irqsave(&i8253_lock, flags sound/drivers/pcsp/pcsp_input.c: raw_spin_unlock_irqrestore(&i8253_lock, sound/drivers/pcsp/pcsp_lib.c: raw_spin_lock_irqsave(&i8253_lock, flags sound/drivers/pcsp/pcsp_lib.c: raw_spin_unlock_irqrestore(&i8253_lock, sound/drivers/pcsp/pcsp_lib.c: raw_spin_lock(&i8253_lock); sound/drivers/pcsp/pcsp_lib.c: raw_spin_unlock(&i8253_lock); sound/drivers/pcsp/pcsp_lib.c: raw_spin_lock(&i8253_lock); sound/drivers/pcsp/pcsp_lib.c: raw_spin_unlock(&i8253_lock); Locks are great, everybody should have their own lock! $ find . -name 8253pit.h ./arch/powerpc/include/asm/8253pit.h ./arch/alpha/include/asm/8253pit.h $ cat arch/*/include/asm/8253pit.h /* * 8253/8254 Programmable Interval Timer */ /* * 8253/8254 Programmable Interval Timer */ $ Eh... $ git grep -w PCSPKR_PLATFORM arch/mips/Kconfig: select PCSPKR_PLATFORM arch/mips/Kconfig: select PCSPKR_PLATFORM arch/mips/Kconfig: select PCSPKR_PLATFORM arch/powerpc/platforms/amigaone/Kconfig: select PCSPKR_PLATFORM drivers/input/misc/Kconfig: depends on PCSPKR_PLATFORM init/Kconfig:config PCSPKR_PLATFORM sound/drivers/Kconfig: depends on PCSPKR_PLATFORM && X86 && HIGH_RES_TIMERS So works on MIPS and x86. Compiles on PowerPC but locking is wrong. On Alpha can't even be selected though it sees otherwise right except the locking and even worse for the rest. Ralf

2 1

[alsa-devel] HDA HDMI pin to converter mapping
by Stephen Warren 27 May '11

27 May '11

Most cards supported by patch_hdmi.c have the property of a 1:1 mapping or connection between converter widgets and pin widgets. Hence, patch_hdmi.c sets up a PCM for each converter, and occasionally performs some operations on the pin widget as required. However, some new cards don't have this 1:1 mapping. In particular, the new GeForce GT 520 and many future NVIDIA cards, and at least the Intel Ibex Peak (in my wife's laptop), have more pin widgets than converters, and each pin widget has a mux to select which converter to take the audio from. For pretty pictures, see the 4th (and 3rd) diagrams at: ftp://download.nvidia.com/XFree86/gpu-hdmi-audio-document/gpu-hdmi-audio.ht… I'd like to discuss how best to handle such cards. I'd expect to see a PCM device created for each pin widget. This would create a 1:1 mapping between PCM devices and attached monitors and hence ELD data (with the possible exception of DisplayPort 1.2 daisy-chaining; I have no idea yet how that plays into this). Does this seem like a reasonable approach to everyone? The current situation is that we only create a PCM for each converter that is mux'd to a pin by default HW initialization. On both the GeForce 520 and Ibex Peak, this means that only 1 PCM gets created, since all pin widgets' muxes point at the first converter by default. This: a) Doesn't allow usage of both converters at once, since there's only 1 PCM object. b) Ends up with up to 4 (in NVIDIA's case) pin widgets (and hence ELD information) logically associated with the PCM object. Hence, it'll be confusing for application to know what features they can really use on the PCM. In fact, ALSA actually only enforces the ELD of the first pin associated with the converter, and ignores the rest. The basic mechanics of a-PCM-per-pin would require hdmi_pcm_open to look at the list of converters, find one not in use by a stream, and then dynamically associate the converter with the stream, or fail the open if a converter was not available. We'd also need a custom close or cleanup function to "unassign" the converter from the PCM, and set hda_pcm_stream.nid to an invalid value. That might be easiest to implement if hda_pcm_stream.nid were to refer to the pin widget rather than the converter widget. However, I notice that some core code in hda_codec.c would be confused by this, so perhaps we'd better just have have hdmi_pcm_open use some other way to determine which stream object is associated with the pin. I suppose that it'd be pretty easy to set up a mapping from hda_pcm_stream to pin nid as part of generic_hdmi_build_pcms. As an aside, this rather reminds me of the mux'ing that goes on inside codecs controlled by ASoC, although I don't think we need to expose this muxing to user-space in the case, but instead make it automatic as I outlined above. Thanks for any thoughts. -- nvpublic

3 8

[alsa-devel] [PATCH] ASoC: Fix wm_hubs input PGA ZC bits
by Mark Brown 27 May '11

27 May '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index e55b298..9e370d1 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -215,23 +215,23 @@ static const struct snd_kcontrol_new analogue_snd_controls[] = { SOC_SINGLE_TLV("IN1L Volume", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 0, 31, 0, inpga_tlv), SOC_SINGLE("IN1L Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 7, 1, 1), -SOC_SINGLE("IN1L ZC Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 7, 1, 0), +SOC_SINGLE("IN1L ZC Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 6, 1, 0), SOC_SINGLE_TLV("IN1R Volume", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 0, 31, 0, inpga_tlv), SOC_SINGLE("IN1R Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 7, 1, 1), -SOC_SINGLE("IN1R ZC Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 7, 1, 0), +SOC_SINGLE("IN1R ZC Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 6, 1, 0), SOC_SINGLE_TLV("IN2L Volume", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 0, 31, 0, inpga_tlv), SOC_SINGLE("IN2L Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 7, 1, 1), -SOC_SINGLE("IN2L ZC Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 7, 1, 0), +SOC_SINGLE("IN2L ZC Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 6, 1, 0), SOC_SINGLE_TLV("IN2R Volume", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 0, 31, 0, inpga_tlv), SOC_SINGLE("IN2R Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 7, 1, 1), -SOC_SINGLE("IN2R ZC Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 7, 1, 0), +SOC_SINGLE("IN2R ZC Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 6, 1, 0), SOC_SINGLE_TLV("MIXINL IN2L Volume", WM8993_INPUT_MIXER3, 7, 1, 0, inmix_sw_tlv), -- 1.7.5.1

2 1

[alsa-devel] [PATCH] ASoC: Tegra: Enable Kaen HP_MUTE at boot
by Stephen Warren 27 May '11

27 May '11

We want the default state of the HP_MUTE signal to be asserted, so that the headphones are muted before the first audio playback. Without this, the headphones are left unmuted until shortly after the first audio playback completes. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra_wm8903.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 0d6738a..a42e9ac 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -267,7 +267,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) } machine->gpio_requested |= GPIO_HP_MUTE; - gpio_direction_output(pdata->gpio_hp_mute, 0); + gpio_direction_output(pdata->gpio_hp_mute, 1); } if (gpio_is_valid(pdata->gpio_int_mic_en)) { -- 1.7.0.4

3 2

[alsa-devel] [PATCH] ASoC: Fix dapm_is_shared_kcontrol so everything isn't shared
by Stephen Warren 27 May '11

27 May '11

Commit af46800 ("ASoC: Implement mux control sharing") introduced function dapm_is_shared_kcontrol. When this function returns true, the naming of DAPM controls is derived from the kcontrol_new. Otherwise, the name comes from the widget (and possibly a widget's naming prefix). A bug in the implementation of dapm_is_shared_kcontrol made it return 1 in all cases. Hence, that commit caused a change in control naming for all controls instead of just shared controls. Specifically, a control is always considered shared because it is always compared against itself. Solve this by never comparing against the widget containing the control being created. Equally, controls should never be shared between DAPM contexts; when the same codec is instantiated multiple times, the same kcontrol_new will be used. However, the control should no be shared between the multiple instances. I tested that with the Tegra WM8903 driver: * Shared is now mostly 0 as expected, and sometimes 1. * The expected controls are still generated after this change. However, I don't have any systems that have a widget/control naming prefix, so I can't test that aspect. Thanks for Jarkko Nikula for pointing out how to fix this. v2: * Don't share across DAPM contexts. * Changed to Tested-by tag below for Jarkko. Reported-by: Liam Girdwood <lrg(a)ti.com> Tested-by: Jarkko Nikula <jhnikula(a)gmail.com> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/soc-dapm.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 456617e..10bb49b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -325,6 +325,7 @@ static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm, } static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *kcontrolw, const struct snd_kcontrol_new *kcontrol_new, struct snd_kcontrol **kcontrol) { @@ -334,6 +335,8 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, *kcontrol = NULL; list_for_each_entry(w, &dapm->card->widgets, list) { + if (w == kcontrolw || w->dapm != kcontrolw->dapm) + continue; for (i = 0; i < w->num_kcontrols; i++) { if (&w->kcontrol_news[i] == kcontrol_new) { if (w->kcontrols) @@ -468,7 +471,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, return -EINVAL; } - shared = dapm_is_shared_kcontrol(dapm, &w->kcontrol_news[0], + shared = dapm_is_shared_kcontrol(dapm, w, &w->kcontrol_news[0], &kcontrol); if (kcontrol) { wlist = kcontrol->private_data; -- 1.7.0.4

4 3

[alsa-devel] UCM representation questions
by Stephen Warren 27 May '11

27 May '11

Liam, I have a few more questions how to represent things in UCM. The Tegra boards I'm dealing with are set up like this: hw:0,0 >--wm8903--> headphones \---> speakers hw:0,1 >----------> hdmi The WM8903 allows the audio stream to be routed to neither, one, or both of headphones and speakers. hw:0,0 <--wm8903--< Analog Mic (AMIC) \----< Digital Mic (DMIC) The WM8903 can capture from one or the other or AMIC/DMIC, but not both. 1) How to indicate when certain devices can be used together, or are mutually exclusive? Note that Headphone/Speakers/HDMI can be used in any combination, but AMIC/DMIC are mutually exclusive. use_case.h indicates that neither Devices nor Modifiers are mutually exclusive, so neither alone can be used to represent this. I suppose you could just create a separate use-case for each legal set of combinations. Taking a capture-only case, we'd end up with separate "Capture AMIC" and "Capture DMIC" cases. However, a "HiFi" or "Music" case could simply list all 3 devices in a single case since they all can operate at once. 2) Let's then consider a "VoIP" use-case that wants to both play and capture audio. If we represent the exclusivity in the same way, we'd end up with 2 more cases "VoIP AMIC" and "VoIP DMIC", each listing 3 devices for playback, and 1 device for capture. It seems like switching between these two cases while a call was active would be problematic. In particular, in my current UCM files, there is e.g.: SectionDevice."Speakers".0 { Comment "Internal Speakers" EnableSequence [ cdev "hw:0" cset "name='Speaker Switch' on,on" cset "name='Int Spk Switch' on" ] DisableSequence [ cdev "hw:0" cset "name='Speaker Switch' off,off" cset "name='Int Spk Switch' off" ] ... Wouldn't the DisableSequence get run for this device when leaving the "VoIP AMIC" case, and the EnableSequence get run when entering the "VoIP DMIC" case; that would cause a glitch in the receive side of the call even though we were just switching the transmission side between mics. Perhaps the UCM library is intelligent enough to realize that the same device is active in both the old and new case, and then doesn't execute the Enable/DisableSequence in that case. But, that seems dangerous; the sequences might have required effects in some case. I can't omit the DisableSequence to make this work, or disabling the Speakers device (e.g. to switch to just headphones) wouldn't actually disable anything. I suppose I could put the Speaker disable logic in the Headphone EnableSequence, but then I wouldn't be able to use both together. Perhaps the answer is a "virtual" Device "Speakers and Headphones" which enables both. I couldn't correctly represent the device-specific volume controls or PCMs then. Plus, this approach would quickly lead to an explosion of Devices; 8 SectionDevices in my the case of my 3 physical IOs. 3) It seems like there's a lot of potential for duplication in the UCM file content. If I have "VoIP Low Quality", "VoIP High Quality", and "Hifi" cases, the latter with a modifier to allow capture of voice e.g. for voice commands, and each somehow supporting capture from either AMIC or DMIC (ignoring how to represent mutual exclusivity) then the UCM file for each case has to include the cset actions to switch between AMIC and DMIC: SectionDevice."Capture AMIC".0 { Comment "Analog Microphone Jack" EnableSequence [ cdev "hw:0" cset "name='ADC Input' 'ADC'" cset "name='Left Capture Mux' 'Right'" cset "name='Right Capture Mux' 'Right'" ] ... SectionDevice."Capture DMIC".0 { Comment "Internal Digital Microphone" EnableSequence [ cdev "hw:0" cset "name='ADC Input' 'DMIC'" cset "name='Left Capture Mux' 'Left'" cset "name='Right Capture Mux' 'Right'" ] ... Is there any way to avoid this cut/paste? I initially expected the UCM file format defining Devices outside of Verbs, and Verbs to list which devices were associated with the verb by reference rather than inclusion, and perhaps also configuring some additional properties of those devices, but not duplicating the raw "how to route this device to the PCM" instructions. 4) Is listing multiple CaptureVolumes in a single Device's Value section legal. For example, for a stereo analog mic: SectionModifier."Capture AMIC".0 { Comment "Analog Microphone Jack" Value { CaptureVolume "Left Input PGA Volume" CaptureVolume "Right Input PGA Volume" } } ... since the WM8903 exposes separate controls for the L/R channels, and I'd expect applications would want to expose a single control to users, and set them both to the same value all the time. Thanks again for any answers. -- nvpublic

3 6

[alsa-devel] [PATCH 1/4] ASoC: Convert 16x16 write to use cpu_to_be16()
by Mark Brown 27 May '11

27 May '11

Make it clear what we're doing. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 10 ++++------ 1 files changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 06b7b81..abdf8d1 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -277,14 +277,12 @@ static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec, static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - u8 data[4]; + u16 data[2]; - data[0] = (reg >> 8) & 0xff; - data[1] = reg & 0xff; - data[2] = (value >> 8) & 0xff; - data[3] = value & 0xff; + data[0] = cpu_to_be16(reg); + data[1] = cpu_to_be16(value); - return do_hw_write(codec, reg, value, data, 4); + return do_hw_write(codec, reg, value, data, sizeof(data)); } /* Primitive bulk write support for soc-cache. The data pointed to by -- 1.7.5.1

2 4

[alsa-devel] Suspecting a bug in plugin_ops.h
by Soulard, MathieuX 27 May '11

27 May '11

Hi all, I'm using alsalib and I checked its code because of one issue I have. I did not yet solved my issue, but I think I saw a bug in alsa-lib that I would like to share with you. It is located in [alsa-lib]/src/pcm/plugin_ops.h in the inline function sx24s(). static inline u_int32_t sx24s(u_int32_t x) { if(x&0x00008000) return x|0x000000FF; return x&0xFFFFFF00; } I think that 0x00008000 should be replaced by 0x00000800. Thanks Regards Mathieu

2 2

[alsa-devel] OMAP3530 -- Only One Sound Card allowed for Power Suspension process?
by Philip Chu 26 May '11

26 May '11

Hi all, I have a crash with power suspend to RAM using following command: echo mem > /sys/power/state It shows following message and hangs up: Freezing user space processes ... (elapsed 0.02 seconds) done. Freezing remaining freezable tasks ... (elapsed 0.01 seconds) And I found out the place to cause this trouble is, I have two sound cards registered in my module init function through: platform_device_alloc("soc-audio", -1); // Beagle board sound card and platform_device_alloc("soc-audio", 0); // my own sound card or platform_device_alloc("soc-audio", 0); and platform_device_alloc("soc-audio", 1); If I don't register the 2nd card or register the 2nd with another name, e.g., "soc-myaudio", it can pass power suspension, but it cannot be recognized by soc_driver in soc-core.c, which is using the name "soc-audio" . Anybody knows how I can register two sound cards and still pass power suspension? Philip Chu principal software engineer LogicPD www.logicpd.com 5 Clock Tower Place, Suite 215 Maynard, MA 01754 Phone: 978-243-2450 Fax: 978-461-1598

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Re: [alsa-devel] alsa patch for Fantom G
by Jaroslav Kysela 26 May '11

26 May '11

On Tue, 29 Mar 2011, Axel Schmidt wrote: > Hello, > > I do not know if I make this request to the right person. > > I've seen a simple patch that should make the Roland Fantom G working. > http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg23584.html > This patch is more then 2 years old. Is there any way to integrate this > patch into the official alsa? Clemens, does the patch looks reasonable? Anyway, the right list for patch posting is the alsa-devel mailing list. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

3 2

[alsa-devel] ASoC updates for 2.6.40
by Mark Brown 26 May '11

26 May '11

The following changes since commit de0853c000aa3b5c9983d066c137e5dc9c8355a0: Merge branch 'topic/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into for-2.6.40 (2011-05-22 10:31:51 +0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.40 These are all small, targetted fixes or just plain trivial. Axel Lin (1): ASoC: wm1250-ev1: Define "WM1250 Output" with SND_SOC_DAPM_OUTPUT Daniel Mack (1): ASoC: fix raumfeld platform Jarkko Nikula (1): ASoC: Fix power down for widgetless per-card DAPM context case Jesper Juhl (1): ASoC: Remove duplicate linux/delay.h inclusion. Liam Girdwood (2): ASoC: core - fix module reference counting for CPU DAIs ASoC: core - remove superfluous new line. Nicolas Ferre (2): ASoC: wm8731: fix wm8731_check_osc() connected condition ASoC: sam9g20_wm8731: use the proper SYSCKL value sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/codecs/wm1250-ev1.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8915.c | 1 - sound/soc/pxa/raumfeld.c | 92 ++++++++++++++++++++++++-------------- sound/soc/soc-core.c | 8 ++-- sound/soc/soc-dapm.c | 2 +- 7 files changed, 66 insertions(+), 43 deletions(-)

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[alsa-devel] No HDMI audio output, nVidia audio not listed in lspci
by Joost Verdoorn 26 May '11

26 May '11

Hi there, After asking around on the IRC it was recommended to me that I should take my problem to the alsa-devel mailing list. So here goes. When I ran Windows (1 year+ ago) I never tried HDMI audio output, or any HDMI output for that matter. Therefore I cannot verify that HDMI audio worked in the first place. However, my GPU, and more specifically, my laptop, is specced to be able to deliver audio through HDMI. I've been looking on and off for a solution to this problem for a couple of months now, but nowhere I found a solution to the specific symptoms that I'm having. My laptop in a nutshell: Machine: HP Compaq 8510w Mobile Workstation Chipset: Mobile Intel PM965 with 800-MHz front side bus GPU: NVIDIA Quadro FX 570M graphics with up to 256MB dedicated memory (512 MB TurboCache) CPU: Intel Core 2 Duo Processor T9300 (2.50 GHz, 6 MB L2 cache, 800 MHz FSB) OS: Linux x86-64 Ubuntu 11.04 Natty Narwhal nVidia Driver Version: 270.41.06 Output of alsa-info: http://paste.ubuntu.com/613101/ Output of lspci: http://paste.ubuntu.com/613123/ I really hope this is the right place for me to find a solution, with your help, and I thank you all for taking the time to read this email. Joost Verdoorn

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[alsa-devel] [PATCH] ASoC: Fix dapm_is_shared_kcontrol so everything isn't shared
by Stephen Warren 26 May '11

26 May '11

Commit af46800 ("ASoC: Implement mux control sharing") introduced function dapm_is_shared_kcontrol. When this function returns true, the naming of DAPM controls is derived from the kcontrol_new. Otherwise, the name comes from the widget (and possibly a widget's naming prefix). A bug in the implementation of dapm_is_shared_kcontrol made it return 1 in all cases. Hence, that commit caused a change in control naming for all controls instead of just shared controls. Specifically, a control is always considered shared because it is always compared against itself. Solve this by never comparing against the widget containing the control being created. I tested that with the Tegra WM8903 driver: * Shared is now mostly 0 as expected, and sometimes 1. * The expected controls are still generated after this change. Howwever, I don't have any systems that have a widget/control naming prefix, so I can't test that aspect. Reported-by: Liam Girdwood <lrg(a)ti.com> Root-caused-by: Jarkko Nikula <jhnikula(a)gmail.com> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/soc-dapm.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 456617e..5397699 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -325,6 +325,7 @@ static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm, } static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *kcontrolw, const struct snd_kcontrol_new *kcontrol_new, struct snd_kcontrol **kcontrol) { @@ -334,6 +335,8 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, *kcontrol = NULL; list_for_each_entry(w, &dapm->card->widgets, list) { + if (w == kcontrolw) + continue; for (i = 0; i < w->num_kcontrols; i++) { if (&w->kcontrol_news[i] == kcontrol_new) { if (w->kcontrols) @@ -468,7 +471,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, return -EINVAL; } - shared = dapm_is_shared_kcontrol(dapm, &w->kcontrol_news[0], + shared = dapm_is_shared_kcontrol(dapm, w, &w->kcontrol_news[0], &kcontrol); if (kcontrol) { wlist = kcontrol->private_data; -- 1.7.0.4 nvpublic

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[alsa-devel] [PATCH for 2.6.40] ASoC: Fix power down for widgetless per-card DAPM context case
by Jarkko Nikula 26 May '11

26 May '11

Commit 52ba67b ("ASoC: Force all DAPM contexts into the same bias state") powers up all the DAPM contexts in a card if any DAPM context becomes active. Unfortunately power down newer happens if per-card DAPM context doesn't have any widgets. Reason for this is that power state of per-card DAPM context without widgets is never cleared and thus all the DAPM contexts remain permanently active. Test for widgetless calling DAPM context in dapm_power_widgets() doesn't work for per-card DAPM context since power change is never originating from widgetless per-card DAPM context. Fix this by pre-clearing power state flag of non-codec DAPM context at the beginning of power sequence. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 456617e..999bb08 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1110,7 +1110,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) trace_snd_soc_dapm_start(card); list_for_each_entry(d, &card->dapm_list, list) - if (d->n_widgets) + if (d->n_widgets || d->codec == NULL) d->dev_power = 0; /* Check which widgets we need to power and store them in -- 1.7.4.4

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Re: [alsa-devel] [PATCH] ASoC: wm1250-ev1: Define "WM1250 Output" with SND_SOC_DAPM_OUTPUT
by Mark Brown 26 May '11

26 May '11

On Thu, May 26, 2011 at 10:54:12AM +0800, Axel Lin wrote: > Codec output pin should be defined with SND_SOC_DAPM_OUTPUT. Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: core: Don't schedule deferred_resume_work twice
by Stephen Warren 26 May '11

26 May '11

For cards that have two or more DAIs, snd_soc_resume's loop over all DAIs ends up calling schedule_work(deferred_resume_work) once per DAI. Since this is the same work item each time, the 2nd and subsequent calls return 0 (work item already queued), and trigger the dev_err message below stating that a work item may have been lost. Solve this by adjusting the loop to simply calculate whether to run the resume work immediately or defer it, and then call schedule work (or not) one time based on that. Note: This has not been tested in mainline, but only in chromeos-2.6.38; mainline doesn't support suspend/resume on Tegra, nor does the mainline Tegra ASoC driver contain multiple DAIs. It has been compile-checked in mainline. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/soc-core.c | 19 ++++++++++--------- 1 files changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a477e21..c261eeb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1256,7 +1256,7 @@ static void soc_resume_deferred(struct work_struct *work) int snd_soc_resume(struct device *dev) { struct snd_soc_card *card = dev_get_drvdata(dev); - int i; + int i, ac97_control = 0; /* AC97 devices might have other drivers hanging off them so * need to resume immediately. Other drivers don't have that @@ -1265,14 +1265,15 @@ int snd_soc_resume(struct device *dev) */ for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - if (cpu_dai->driver->ac97_control) { - dev_dbg(dev, "Resuming AC97 immediately\n"); - soc_resume_deferred(&card->deferred_resume_work); - } else { - dev_dbg(dev, "Scheduling resume work\n"); - if (!schedule_work(&card->deferred_resume_work)) - dev_err(dev, "resume work item may be lost\n"); - } + ac97_control |= cpu_dai->driver->ac97_control; + } + if (ac97_control) { + dev_dbg(dev, "Resuming AC97 immediately\n"); + soc_resume_deferred(&card->deferred_resume_work); + } else { + dev_dbg(dev, "Scheduling resume work\n"); + if (!schedule_work(&card->deferred_resume_work)) + dev_err(dev, "resume work item may be lost\n"); } return 0; -- 1.7.0.4

3 2

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Add correct hw registers to Line1 cross connect muxes
by Jarkko Nikula 26 May '11

26 May '11

Commit af46800 ("ASoC: Implement mux control sharing") revealed that "Left Line1[L | R] Mux" and "Right Line1[L | R] Mux" widgets were pointing to the same kcontrols and codec registers and thus soc-core falsely detected them as shared controls. This is actually wrong since there are separate registers in hardware that configure Line1L to RADC and Line1R to LADC cross connects so these muxes should not be shared. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/codecs/tlv320aic3x.c | 34 +++++++++++++++++++++------------- 1 files changed, 21 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 789453d..0963c4c 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -226,11 +226,13 @@ static const char *aic3x_adc_hpf[] = #define RDAC_ENUM 1 #define LHPCOM_ENUM 2 #define RHPCOM_ENUM 3 -#define LINE1L_ENUM 4 -#define LINE1R_ENUM 5 -#define LINE2L_ENUM 6 -#define LINE2R_ENUM 7 -#define ADC_HPF_ENUM 8 +#define LINE1L_2_L_ENUM 4 +#define LINE1L_2_R_ENUM 5 +#define LINE1R_2_L_ENUM 6 +#define LINE1R_2_R_ENUM 7 +#define LINE2L_ENUM 8 +#define LINE2R_ENUM 9 +#define ADC_HPF_ENUM 10 static const struct soc_enum aic3x_enum[] = { SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux), @@ -238,6 +240,8 @@ static const struct soc_enum aic3x_enum[] = { SOC_ENUM_SINGLE(HPLCOM_CFG, 4, 3, aic3x_left_hpcom_mux), SOC_ENUM_SINGLE(HPRCOM_CFG, 3, 5, aic3x_right_hpcom_mux), SOC_ENUM_SINGLE(LINE1L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), + SOC_ENUM_SINGLE(LINE1L_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), + SOC_ENUM_SINGLE(LINE1R_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), @@ -490,12 +494,16 @@ static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = { }; /* Left Line1 Mux */ -static const struct snd_kcontrol_new aic3x_left_line1_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_ENUM]); +static const struct snd_kcontrol_new aic3x_left_line1l_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_L_ENUM]); +static const struct snd_kcontrol_new aic3x_right_line1l_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_R_ENUM]); /* Right Line1 Mux */ -static const struct snd_kcontrol_new aic3x_right_line1_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_ENUM]); +static const struct snd_kcontrol_new aic3x_right_line1r_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_R_ENUM]); +static const struct snd_kcontrol_new aic3x_left_line1r_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_L_ENUM]); /* Left Line2 Mux */ static const struct snd_kcontrol_new aic3x_left_line2_mux_controls = @@ -535,9 +543,9 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { &aic3x_left_pga_mixer_controls[0], ARRAY_SIZE(aic3x_left_pga_mixer_controls)), SND_SOC_DAPM_MUX("Left Line1L Mux", SND_SOC_NOPM, 0, 0, - &aic3x_left_line1_mux_controls), + &aic3x_left_line1l_mux_controls), SND_SOC_DAPM_MUX("Left Line1R Mux", SND_SOC_NOPM, 0, 0, - &aic3x_left_line1_mux_controls), + &aic3x_left_line1r_mux_controls), SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0, &aic3x_left_line2_mux_controls), @@ -548,9 +556,9 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { &aic3x_right_pga_mixer_controls[0], ARRAY_SIZE(aic3x_right_pga_mixer_controls)), SND_SOC_DAPM_MUX("Right Line1L Mux", SND_SOC_NOPM, 0, 0, - &aic3x_right_line1_mux_controls), + &aic3x_right_line1l_mux_controls), SND_SOC_DAPM_MUX("Right Line1R Mux", SND_SOC_NOPM, 0, 0, - &aic3x_right_line1_mux_controls), + &aic3x_right_line1r_mux_controls), SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line2_mux_controls), -- 1.7.4.4

2 1

[alsa-devel] sound/trivial: some little typo and little fix
by Nicolas Ferre 26 May '11

26 May '11

Hi, This is a tiny patch series with two typos and one error path fix. More interesting stuff will come ;-) Best regards, -- Nicolas Ferre

3 5

[alsa-devel] Negative volume steps?
by Tanu Kaskinen 26 May '11

26 May '11

Hi, If a driver claims that the lowest volume step (integer step, not dB) of a volume element is -126 and the highest step is 0, is the driver broken? I'm asking, because apparently there are such drivers, but Pulseaudio currently assumes that the volume steps are never negative. -- Tanu

3 3

[alsa-devel] about ioctl calls
by Lorenzo Fundaró 26 May '11

26 May '11

Hi folks, I have a program that constantly make ioctl calls like this: ioctl(4, 0x4122, 0x8b703) ioctl(4, 0x4122, 0xbfb65768) I suspect that this call pertains to ALSA, since I have straced other programs that play music and they always use in the ioctl request field the 0x4122 code. I am interested on finding out what's in the third argument of the ioctl call, is it a sort of struct that contains the audio samples ? Thanks in advance, Lorenzo Fundaró García Université de Technologie de Compiègne Génie Informatique

2 2

Re: [alsa-devel] [PATCH] ASoC: wm1250-ev1: Define "WM1250 Output" with SND_SOC_DAPM_OUTPUT
by Liam Girdwood 26 May '11

26 May '11

On 26/05/11 03:54, Axel Lin wrote: > Codec output pin should be defined with SND_SOC_DAPM_OUTPUT. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > Note: > Just found this bug while reading the code. > I don't have this hardware handy. I appreciate if someone can test it. > > Axel > The fix does look correct. Acked-by: Liam Girdwood <lrg(a)ti.com>

1 0

[alsa-devel] Patch series for Fast Track Ultra mixer quirks
by Daniel Mack 26 May '11

26 May '11

Here's the set of patches to make Fast Track Ultra mixer quirks possible. It's just a split version of the patch we've discussed on the ALSA ML before. These patches can as well be pulled from git@github.com:zonque/linux-2.6.git usb-audio Daniel Mack (5): ALSA: usb-audio: move assignment of chip->ctrl_intf ALSA: usb-audio: rework add_control_to_empty() ALSA: usb-audio: export snd_usb_feature_unit_ctl ALSA: usb-audio: add new quirk type QUIRK_AUDIO_STANDARD_MIXER ALSA: usb-audio: more control quirks for M-Audio FastTrack devices sound/usb/card.c | 17 ++++++----- sound/usb/mixer.c | 32 ++++++++------------ sound/usb/mixer.h | 14 ++++++++- sound/usb/mixer_quirks.c | 70 ++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 4 +- sound/usb/quirks.c | 18 +++++++++++- sound/usb/usbaudio.h | 1 + 7 files changed, 125 insertions(+), 31 deletions(-)

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Re: [alsa-devel] [PATCH] sound core: remove unused variables.
by Takashi Iwai 26 May '11

26 May '11

At Wed, 25 May 2011 22:43:27 +0200, Luca Tettamanti wrote: > > Drop a few variables that are never read. > > Signed-off-by: Luca Tettamanti <kronos.it(a)gmail.com> Applied now. Thanks. Takashi > --- > sound/core/control.c | 3 +-- > sound/core/init.c | 3 --- > sound/core/oss/linear.c | 3 --- > sound/core/pcm_native.c | 8 -------- > sound/core/seq/seq_queue.c | 2 -- > 5 files changed, 1 insertions(+), 18 deletions(-) > > diff --git a/sound/core/control.c b/sound/core/control.c > index 5d98194..f8c5be4 100644 > --- a/sound/core/control.c > +++ b/sound/core/control.c > @@ -704,13 +704,12 @@ static int snd_ctl_elem_list(struct snd_card *card, > struct snd_ctl_elem_list list; > struct snd_kcontrol *kctl; > struct snd_ctl_elem_id *dst, *id; > - unsigned int offset, space, first, jidx; > + unsigned int offset, space, jidx; > > if (copy_from_user(&list, _list, sizeof(list))) > return -EFAULT; > offset = list.offset; > space = list.space; > - first = 0; > /* try limit maximum space */ > if (space > 16384) > return -ENOMEM; > diff --git a/sound/core/init.c b/sound/core/init.c > index 30ecad4..2c041bb 100644 > --- a/sound/core/init.c > +++ b/sound/core/init.c > @@ -342,7 +342,6 @@ static const struct file_operations snd_shutdown_f_ops = > int snd_card_disconnect(struct snd_card *card) > { > struct snd_monitor_file *mfile; > - struct file *file; > int err; > > if (!card) > @@ -366,8 +365,6 @@ int snd_card_disconnect(struct snd_card *card) > > spin_lock(&card->files_lock); > list_for_each_entry(mfile, &card->files_list, list) { > - file = mfile->file; > - > /* it's critical part, use endless loop */ > /* we have no room to fail */ > mfile->disconnected_f_op = mfile->file->f_op; > diff --git a/sound/core/oss/linear.c b/sound/core/oss/linear.c > index 13b3f6f..2045697 100644 > --- a/sound/core/oss/linear.c > +++ b/sound/core/oss/linear.c > @@ -90,11 +90,8 @@ static snd_pcm_sframes_t linear_transfer(struct snd_pcm_plugin *plugin, > struct snd_pcm_plugin_channel *dst_channels, > snd_pcm_uframes_t frames) > { > - struct linear_priv *data; > - > if (snd_BUG_ON(!plugin || !src_channels || !dst_channels)) > return -ENXIO; > - data = (struct linear_priv *)plugin->extra_data; > if (frames == 0) > return 0; > #ifdef CONFIG_SND_DEBUG > diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c > index 1a07750..3a5f86d 100644 > --- a/sound/core/pcm_native.c > +++ b/sound/core/pcm_native.c > @@ -1518,13 +1518,11 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, > static int snd_pcm_drop(struct snd_pcm_substream *substream) > { > struct snd_pcm_runtime *runtime; > - struct snd_card *card; > int result = 0; > > if (PCM_RUNTIME_CHECK(substream)) > return -ENXIO; > runtime = substream->runtime; > - card = substream->pcm->card; > > if (runtime->status->state == SNDRV_PCM_STATE_OPEN || > runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED || > @@ -2056,7 +2054,6 @@ static int snd_pcm_open_file(struct file *file, > { > struct snd_pcm_file *pcm_file; > struct snd_pcm_substream *substream; > - struct snd_pcm_str *str; > int err; > > if (rpcm_file) > @@ -2073,7 +2070,6 @@ static int snd_pcm_open_file(struct file *file, > } > pcm_file->substream = substream; > if (substream->ref_count == 1) { > - str = substream->pstr; > substream->file = pcm_file; > substream->pcm_release = pcm_release_private; > } > @@ -3015,11 +3011,9 @@ static const struct vm_operations_struct snd_pcm_vm_ops_status = > static int snd_pcm_mmap_status(struct snd_pcm_substream *substream, struct file *file, > struct vm_area_struct *area) > { > - struct snd_pcm_runtime *runtime; > long size; > if (!(area->vm_flags & VM_READ)) > return -EINVAL; > - runtime = substream->runtime; > size = area->vm_end - area->vm_start; > if (size != PAGE_ALIGN(sizeof(struct snd_pcm_mmap_status))) > return -EINVAL; > @@ -3054,11 +3048,9 @@ static const struct vm_operations_struct snd_pcm_vm_ops_control = > static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file *file, > struct vm_area_struct *area) > { > - struct snd_pcm_runtime *runtime; > long size; > if (!(area->vm_flags & VM_READ)) > return -EINVAL; > - runtime = substream->runtime; > size = area->vm_end - area->vm_start; > if (size != PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control))) > return -EINVAL; > diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c > index e7a8e9e..f907736 100644 > --- a/sound/core/seq/seq_queue.c > +++ b/sound/core/seq/seq_queue.c > @@ -467,13 +467,11 @@ int snd_seq_queue_timer_open(int queueid) > int snd_seq_queue_timer_close(int queueid) > { > struct snd_seq_queue *queue; > - struct snd_seq_timer *tmr; > int result = 0; > > queue = queueptr(queueid); > if (queue == NULL) > return -EINVAL; > - tmr = queue->timer; > snd_seq_timer_close(queue); > queuefree(queue); > return result; > -- > 1.7.5.1 >

1 0

[alsa-devel] [PATCH] ALSA: ASoC: Fix comment in cs4270 codec driver
by Daniel Mack 26 May '11

26 May '11

The comment does not reflect reality anymore since the multi-component monster patch landed. Things are matched by names now, and not by exporting and referencing a struct. Fix it to avoid confusion. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/codecs/cs4270.c | 5 +---- 1 files changed, 1 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 0206a17..6cc8678 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -636,10 +636,7 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec) #endif /* CONFIG_PM */ /* - * ASoC codec device structure - * - * Assign this variable to the codec_dev field of the machine driver's - * snd_soc_device structure. + * ASoC codec driver structure */ static const struct snd_soc_codec_driver soc_codec_device_cs4270 = { .probe = cs4270_probe, -- 1.7.5.1

4 4

[alsa-devel] Driver debugging help: OSS emulation and different results
by Mike Crowe 26 May '11

26 May '11

Hi folks, I know this is vague, and probably not worthy of this list. However, I'm struggling with where to go next. Here's the deal: - I have an embedded system where we have developed the soc driver, and it appears to be working fine - We have a VoIP library we are integrating, and only have object files and almost no support - The VoIP library uses OSS interface with our hardware - We validated our hardware using vrec/vplay, both OSS applications. They record and play fine (as does the ALSA arecord/aplay as well) - I have the output (playback) working on the VoIP library side, but the recording is choppy (though seems to be at the right frequency now) So, what I need to do next is try and understand how the library is opening OSS differently from vrec/vplay. However, I'm not sure where to start. Can somebody give me a high level pointer as to how to add some debugging output to my system to capture what is different between these two sessions? Where should I start? TIA Mike

1 0

[alsa-devel] EDID-like data for audio: Port correlation between ALSA and X
by Stephen Warren 26 May '11

26 May '11

(starting a new cross-mailing-list thread following some small discussion on this topic on alsa-devel) Background: Many recent graphics cards support sending an audio stream over HDMI or DisplayPort. This is supported by an HD-audio controller built into the graphics card. It'd be nice if configuration tools could easily determine which ALSA devices correspond to which X displays. Current implementation details: The X driver reads the EDID from the display, reformats it as an ELD ("EDID-Like Data"), as specified by the Intel HD-audio specification, and somehow communicates this into the HD-audio controller that's embedded into the graphics card, using a card-specific mechanism. The HD-audio controller then communicates the ELD to the driver in a standard-specified manner: The HW triggers an "unsolicited response" to notify the ALSA driver of a change in the data, and then the ALSA driver reads the ELD data using HD-audio commands (verbs). Finally, the eventual idea is that user-space applications will be able to read the ELD data from the ALSA driver for each PCM. In the HD-audio controller, there is a "pin widget" for each video output (HDMI port, DP port). ELD data is associated with a specific pin widget. ALSA creates a single "PCM" or audio device per pin widget. New discussion: We need some way of tying pin widgets or ALSA PCMs to the X display path that the audio data will be merged into. Option A: We can pass some X-specific data in the ELD. The ELD contains an 8-byte Port_ID field for this purpose. The display driver is supposed to fill this with some meaningful data, and ALSA (or an application) is supposed to interpret the field. Some possibilities for this field are: A1) An XRandR property, such as ConnectorNumber or Output. To me, this seems the simplest option overall; the X driver already has access to this information, so it should be fairly easy to place it into the ELD. The only downside here is that the NVIDIA binary driver doesn't currently implement recent XRandR versions, so we probably couldn't do this right now. A2) NVIDIA's graphics driver fills this field with an internal "display ID" value, which I believe is usable without our NV-CTRL X extension. This obviously isn't ideal for applications interpreting this data, since the format isn't cross-vendor. Applications don't want to interpret the ELD in N different ways. Option B: We can invent some X/XRandR property that refers to some ALSA HW or SW property. Applications can then enumerate this X/XRandR property for all outputs, and use that data to access the appropriate ALSA device. That data could be: B1) ALSA PCM name (e.g. "hw:1,3"). While perhaps technically feasible, it'd be pretty painful to determine this from the X driver. Sysfs parsing of ALSA-related data would probably work, but this kind of thing would be better done elsewhere than the X driver, especially if the ELD handoff is implemented in the kernel by a KMS driver. B2) HD-audio controller pin widget ID. This requires the X driver to know the internal architecture of the HD-audio controller. While technically perhaps not too hard, it'd most likely mean maintaining tables of audio-related data in the X driver, which is probably not something we'd want to do. Equally, the pin widget ID might not be the most useful thing to an application. AFAIK, ALSA doesn't have an API for exposing this information at present, except through user-oriented debug files in /proc. Does anyone see any other options? Do people agree that option A1 is the best? Related issues: There are issues correlating ALSA and X data such as XRandR connector numbers in the face of multiple X servers, remote X servers, VT-switching, etc. However, I believe most of these issues need to be solved outside the context of this discussion anyway, and hence probably don't affect this discussion much. Probably, PulseAudio already interfaces with DeviceKit or SomethingKit to solve some of these issues? Thanks for reading! -- nvpublic

1 0

[alsa-devel] [PATCH] ASoc: wm8731: fix wm8731_check_osc() connected condition
by Nicolas Ferre 25 May '11

25 May '11

The crystal oscillator is only enabled if the WM8731_SYSCLK_XTAL master clock is specified. Fix the connected() struct snd_soc_dapm_route function to take this into account. Oscillator is not enabled on machine that need it otherwise. Machine drivers have to make sure that they use the proper SYSCLK value. Signed-off-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/soc/codecs/wm8731.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 6dec7ce..2dc964b 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -198,7 +198,7 @@ static int wm8731_check_osc(struct snd_soc_dapm_widget *source, { struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(source->codec); - return wm8731->sysclk_type == WM8731_SYSCLK_MCLK; + return wm8731->sysclk_type == WM8731_SYSCLK_XTAL; } static const struct snd_soc_dapm_route wm8731_intercon[] = { -- 1.7.3

3 3

[alsa-devel] [PATCH] ASoc: sam9g20_wm8731: use the proper SYSCKL value
by Nicolas Ferre 25 May '11

25 May '11

at91sam9g20 is providing master clock to wm8731: not using a crystal but an external MCLK. We can avoid conflict and save power using WM8731_SYSCLK_MCLK as we do not need oscillator to be powered. Signed-off-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/soc/atmel/sam9g20_wm8731.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 28afbbf..95572d2 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -146,7 +146,7 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd) "at91sam9g20ek_wm8731 " ": at91sam9g20ek_wm8731_init() called\n"); - ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_MCLK, MCLK_RATE, SND_SOC_CLOCK_IN); if (ret < 0) { printk(KERN_ERR "Failed to set WM8731 SYSCLK: %d\n", ret); -- 1.7.3

3 3

[alsa-devel] [PATCH 0/6] ASoC: davinci-pcm: some fixes and convert to BATCH
by Ben Gardiner 25 May '11

25 May '11

This patchset is a collection of changes for the davinci-pcm driver that were accumulated during previous forays into the davinci-mcasp and davinci-pcm drivers [1]. The first three patches are minor changes that cleanup the ping-pong DMA params setup, expand the davinci-pcm reported format and also its maximum channels. The fourth patch in the series corrects an audible glitch that can be heard on da850evm's McASP when ping-pong buffers are enabled. The glitch occurs only on the second and susequent playbacks, not the first. Reversing the order in which the dma channels are started and moving the start to the trigger function from the prepare function fixes the glitch. The fifth and sixth patches in the series convert the davinci-pcm driver to BATCH mode. I noticed that the davinci-pcm pointer function is returning a value that is retrieved from within the running EDMA channel's state -- which is contrary to the warning in the comments of arch/arm/mach-davinci/dma.c:edma_get_position(): " * Returns current source and destination addresses for a particular * parameter RAM slot. Its channel should not be active when this is called. " To that end, we begin with some refactoring to centralize the modification of the davinci_runtime_data.period member. Followed by wholesale replacement of davinci_pcm_pointer() function so that it returns the position as reported by the last update of the period member -- along with corresponding changes to davinci_pcm_enqueue_dma() davinci_pcm_dma_irq() davinci_pcm_trigger() and reporting SNDRV_PCM_INFO_BATCH on playback and capture. The series has been tested with and without ping-pong buffers enabled [1]; furthermore, the pointer behaviour was inspected using both XRUN_DEBUG_PERIODUPDATE and XRUN_DEBUG_HWPTRUPDATE as well as the audio checked with headphones. Playback was tested with: echo 24 > /proc/asound/EVM/pcm0p/xrun_debug aplay -d 1 -t wav -D "default:CARD=EVM" test.wav echo 0 > /proc/asound/EVM/pcm0p/xrun_debug and capture was tested with (while playing test.wav on my PC into the capture input on the da850evm): #!/bin/bash echo 24 > /proc/asound/EVM/pcm0c/xrun_debug arecord -d 2 -t wav -D "default:CARD=EVM" -f dat - |aplay -t wav echo 0 > /proc/asound/EVM/pcm0c/xrun_debug Playback behaviour before (similar both with and without ping-pong buffers) ========================= hwptr_update: pcmC0D0p:0: pos=42/2048/32768, hwptr=42/0/42/0 hwptr_update: pcmC0D0p:0: pos=369/2048/32768, hwptr=327/42/369/0 hwptr_update: pcmC0D0p:0: pos=709/2048/32768, hwptr=340/369/709/0 hwptr_update: pcmC0D0p:0: pos=994/2048/32768, hwptr=285/709/994/0 hwptr_update: pcmC0D0p:0: pos=1367/2048/32768, hwptr=373/994/1367/0 hwptr_update: pcmC0D0p:0: pos=1729/2048/32768, hwptr=362/1367/1729/0 hwptr_update: pcmC0D0p:0: pos=2037/2048/32768, hwptr=308/1729/2037/0 period_update: pcmC0D0p:0: pos=2335/2048/32768, hwptr=298/2037/2335/0 hwptr_update: pcmC0D0p:0: pos=2762/2048/32768, hwptr=427/2335/2762/0 hwptr_update: pcmC0D0p:0: pos=3147/2048/32768, hwptr=385/2762/3147/0 hwptr_update: pcmC0D0p:0: pos=3456/2048/32768, hwptr=309/3147/3456/0 hwptr_update: pcmC0D0p:0: pos=3851/2048/32768, hwptr=395/3456/3851/0 period_update: pcmC0D0p:0: pos=4150/2048/32768, hwptr=299/3851/4150/0 [...] Playback behaviour after (identical both with and without ping-pong buffers) ======================== hwptr_update: pcmC0D0p:0: pos=0/2048/32768, hwptr=0/0/0/0 period_update: pcmC0D0p:0: pos=2048/2048/32768, hwptr=2048/0/2048/0 hwptr_update: pcmC0D0p:0: pos=2048/2048/32768, hwptr=0/2048/2048/0 period_update: pcmC0D0p:0: pos=4096/2048/32768, hwptr=2048/2048/4096/0 [...] [Capture behaviour was very similarly modified by this patchset] The playback behaviour before shows inconsistent increments of the pointers in syscall and interrupt context -- this is due to the pointer function peeking into the parameters of the dma channel. The playback behaviour afterwards shows regular increments from interrupt context. Which is to be expected since the driver now reports an incremented pointer after each dma completion. It seems that the conversion to BATCH mode is beneficial -- other than obeying the warning comment in dma.c -- since there appear to be fewer syscalls per execution of aplay. [1] http://article.gmane.org/gmane.linux.alsa.devel/85122 Ben Gardiner (6): ASoC: davinci-pcm: trivial: make ping-pong params setup symmetrical ASoC: davinci-pcm: expand the .formats ASoC: davinci-pcm: increase the maximum channels ASoC: davinci-pcm: fix audible glitch on 2nd ping-pong playback ASoC: davinci-pcm: extract period elapsed functions ASoC: davinci-pcm: convert to BATCH mode sound/soc/davinci/davinci-pcm.c | 127 ++++++++++++++++++++------------------- 1 files changed, 65 insertions(+), 62 deletions(-) -- 1.7.4.1

3 13

[alsa-devel] [PATCH] ALSA: HDA: Use AZX_DRIVER_ATI for AMD/ATI platforms
by Andiry Xu 25 May '11

25 May '11

This commit makes AMD/ATI platforms use AZX_DRIVER_ATI for SB HD Audio and AZX_DRIVER_ATIHDMI for HDMI Audio, so some quirks can be applied to AMD/ATI platforms without checking the pci vendor ID. This also spearate SB and HDMI audio, since some quirks should only apply to SB audio. Signed-off-by: Andiry Xu <andiry.xu(a)amd.com> --- sound/pci/hda/hda_intel.c | 31 +++++++++++-------------------- 1 files changed, 11 insertions(+), 20 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 43a0367..05f551c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1062,7 +1062,9 @@ static void azx_init_pci(struct azx *chip) switch (chip->driver_type) { case AZX_DRIVER_ATI: - /* For ATI SB450 azalia HD audio, we need to enable snoop */ + /* For ATI SB450/600/700/800/900 and AMD Hudson azalia + * HD audio, we need to enable snoop. + */ update_pci_byte(chip->pci, ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, 0x07, ATI_SB450_HDAUDIO_ENABLE_SNOOP); @@ -1092,13 +1094,6 @@ static void azx_init_pci(struct azx *chip) ? "Failed" : "OK"); } break; - default: - /* AMD Hudson needs the similar snoop, as it seems... */ - if (chip->pci->vendor == PCI_VENDOR_ID_AMD) - update_pci_byte(chip->pci, - ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, - 0x07, ATI_SB450_HDAUDIO_ENABLE_SNOOP); - break; } } @@ -1457,8 +1452,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model) * sequence like the pin-detection. It seems that forcing the synced * access works around the stall. Grrr... */ - if (chip->pci->vendor == PCI_VENDOR_ID_AMD || - chip->pci->vendor == PCI_VENDOR_ID_ATI) { + if (chip->driver_type == AZX_DRIVER_ATI) { snd_printk(KERN_INFO SFX "Enable sync_write for AMD chipset\n"); chip->bus->sync_write = 1; chip->bus->allow_bus_reset = 1; @@ -2370,13 +2364,8 @@ static int __devinit check_position_fix(struct azx *chip, int fix) /* Use link position directly, avoid any transfer problem. */ return POS_FIX_VIACOMBO; case AZX_DRIVER_ATI: - /* ATI chipsets don't work well with position-buffer */ + /* ATI/AMD chipsets don't support position-buffer */ return POS_FIX_LPIB; - case AZX_DRIVER_GENERIC: - /* AMD chipsets also don't work with position-buffer */ - if (chip->pci->vendor == PCI_VENDOR_ID_AMD) - return POS_FIX_LPIB; - break; } return POS_FIX_AUTO; @@ -2800,9 +2789,11 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, .class_mask = 0xffffff, .driver_data = AZX_DRIVER_ICH }, - /* ATI SB 450/600 */ + /* ATI SB 450/600/700/800/900 */ { PCI_DEVICE(0x1002, 0x437b), .driver_data = AZX_DRIVER_ATI }, { PCI_DEVICE(0x1002, 0x4383), .driver_data = AZX_DRIVER_ATI }, + /* AMD Hudson */ + { PCI_DEVICE(0x1022, 0x780d), .driver_data = AZX_DRIVER_ATI }, /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x793b), .driver_data = AZX_DRIVER_ATIHDMI }, { PCI_DEVICE(0x1002, 0x7919), .driver_data = AZX_DRIVER_ATIHDMI }, @@ -2849,15 +2840,15 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x17f3, 0x3010), .driver_data = AZX_DRIVER_GENERIC }, /* VMware HDAudio */ { PCI_DEVICE(0x15ad, 0x1977), .driver_data = AZX_DRIVER_GENERIC }, - /* AMD/ATI Generic, PCI class code and Vendor ID for HD Audio */ + /* AMD/ATI Generic, PCI class code and Vendor ID for HDMI Audio */ { PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, .class_mask = 0xffffff, - .driver_data = AZX_DRIVER_GENERIC }, + .driver_data = AZX_DRIVER_ATIHDMI }, { PCI_DEVICE(PCI_VENDOR_ID_AMD, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, .class_mask = 0xffffff, - .driver_data = AZX_DRIVER_GENERIC }, + .driver_data = AZX_DRIVER_ATIHDMI }, { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); -- 1.7.1

3 3

[alsa-devel] [PATCH] Force AD1988_6STACK_DIG for Asus M3N-HT Deluxe
by Tony Vroon 25 May '11

25 May '11

The microphone input on the back panel (pink connector) stopped operating correctly after an upgrade from 2.6.35 to 2.6.38; the actual problem manifests itself as a lack of microphone bias voltage (VREF_HIZ) on node 0x17. With AD1988_6STACK_DIG the maximum bias voltage (VREF_80) is applied and the headset operates correctly. Signed-off-by: Tony Vroon <tony(a)linx.net> Tested-by: Doug Redlich <pbrigade(a)nxltech.net> diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index f1b3875..696ac25 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3159,6 +3159,7 @@ static const struct snd_pci_quirk ad1988_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG), + SND_PCI_QUIRK(0x1043, 0x82c0, "Asus M3N-HT Deluxe", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x8311, "Asus P5Q-Premium/Pro", AD1988_6STACK_DIG), {} };

2 1

[alsa-devel] [PATCH 0/9] ASoC: sh: fsi: tidyup and cleanup FSI
by Kuninori Morimoto 25 May '11

25 May '11

Dear Mark, Liam Cc Simon, yoshii-san, Magnus These are tidyup and cleanup patches for FSI Kuninori Morimoto (9): ASoC: sh: fsi: tidyup parameter of fsi_stream_push ASoC: sh: fsi: tidyup unclear variable naming ASoC: sh: fsi: remove pm_runtime from fsi_dai_set_fmt. ASoC: sh: fsi: make sure fsi_stream_push/pop access by spin lock ASoC: sh: fsi: Add fsi_set_master_clk ASoC: sh: fsi: irq control moves to fsi_port_start/stop ASoC: sh: fsi: Add fsi_hw_startup/shutdown ASoC: sh: fsi: remove fsi_module_init/kill ASoC: sh: fsi: cleanup suspend/resume #1 and #2 are tidyup for fsi_fifo_data_ctrl. The variables and sub-functions were unclear nameing. So, it was difficult to understand the detail of calculation. Thank you for pointing it yoshii-san. #3 - #9 are clean up for RuntimePM. Current FSI driver was wrong for RuntimePM. these patches fixup for RuntimePM and clean up for suspend/resume. last patch remove noisy saved_xxx variable. Best regards -- Kuninori Morimoto

3 13

[alsa-devel] A plea for help on mixer support for Fast Track Ultra (8R)
by Felix Homann 25 May '11

25 May '11

Hi, I need some help again by someone who's more familiar with Alsa's inner working than I am. This might be you, Daniel ;-) I've attached a patch series that would add support for the mixer in M-Audio's Fast Track Ultra (8R) devices. BUT, it will break others! It will break others in that mixer controls will show up multiple times. The reason for this seems to be my ambitiuos calling of snd_usb_create_mixer() for quirked devices in card.c. To circumvent the problem of a control showing up multiple times, I check for duplicates in add_unique_control_to_empty_mixer() and skip controls already present. This doesn't work for devices which are not using add_unique_control_to_empty_mixer(), in other words *all* devices not using my messy patch... So, probably the best thing would be to have some kind of entry in quirks-table.h to tell which device would need to be "mixerquirked" to prevent other devices to be disturbed. How can this be done? I would highly appreciate anybody coming up with a proper solution. Kind regards, Felix

5 33

[alsa-devel] [PATCH] ASoC: dapm - fix prefix for DAPM muxes
by Liam Girdwood 24 May '11

24 May '11

Make sure DAPM muxes have a valid kcontrol name instead of NULL. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 456617e..c5d98e3 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -489,10 +489,10 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, wlist->widgets[wlistentries - 1] = w; if (!kcontrol) { - if (dapm->codec) + if (dapm->codec && dapm->codec->name_prefix) prefix = dapm->codec->name_prefix; else - prefix = NULL; + prefix = w->name; if (shared) { name = w->kcontrol_news[0].name; -- 1.7.4.1

3 3

Re: [alsa-devel] GR-55 Driver
by Jeffrey Scott Flesher Gmail 24 May '11

24 May '11

I have read this article over a few times, trying to remove the kernel and Reiser4 instructions; and I got a little confused; although I did get it to compile and now it see's the GR-55 as a sound device; I tried to jack it to the output but got no audio; what should I do to test it? Do these instructions look correct? # Path: /home/username/dev/sound/alsa/sound-2.6 git clone git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git cd sound-2.6 git apply 0001-ALSA-snd-usb-add-quirks-for-Roland-GR-55.patch sudo cp -vi /boot/config-$(uname -r) .config sudo make oldconfig # a lot of questions, I hit enter for default make-kpkg clean # Since I'm not building the kernel I didn't do this: # But since this is the only command that will write the new kernel, # what do I do? nice fakeroot make-kpkg --initrd --append-to-version=-lockard --overlay-dir=/home/username/dev/kernel/source kernel-image kernel-headers If I was to guess; I'd change the overlay-dir to use the kernel in the sound-2.6 folder? nice fakeroot make-kpkg --initrd --append-to-version=-gr55 --overlay-dir=/home/username/dev/sound/alsa/sound-2.6/kernel kernel-image kernel-headers # is this right? cd .. ls -lh sudo dpkg -i linux-*.deb uname -r # 2.6.39-rc3-gr55+ One thing strange was that it locked up on reboot, showing rc3, but a reset with previous kernel, I saw the gr55 version so I picked it; it worked as stated above; just don't know why it didn't set this version as default, maybe a step I missed. Sorry it took so long to get back to this; I had a hard drive go bad; so I installed Ubuntu Studio 11.04; so I'm doing this against the newest kernel. Thanks From: Daniel Mack <zonque(a)gmail.com> To: Jeffrey Scott Flesher Gmail <jeffrey.scott.flesher(a)gmail.com> Subject: Re: [alsa-devel] GR-55 Driver Date: Fri, 13 May 2011 09:24:57 +0200 On Fri, May 13, 2011 at 4:30 AM, Jeffrey Scott Flesher Gmail <jeffrey.scott.flesher(a)gmail.com> wrote: > I'm running Ubuntu Studio 10.04 x64 with realtime kernel from > http://ppa.launchpad.net/abogani/ppa/ubuntu lucid (linux-realtime - > 2.6.33-29.1); my motherboard is ASUS M4A78T-E onboard VIA sound, Nvidia 460 > GTX Video card; AMD 955 X4 CPU. > Link to kernel used > https://launchpad.net/~abogani/+archive/ppa/+packages?field.name_filter=&fi… > > I searched for instructions on how to do this, with no good results; so I > must ask someone that knows how. Things are a little more complicated as modern Linux distributions use techniques like initramfs. For Ubuntu, you should follow one of the tutorials. Two I found are here: https://help.ubuntu.com/community/Kernel/Compile http://parabing.com/2011/04/28/ubuntu-natty-a-custom-kernel-is-what-you-wan… That looks comprehensive. Just don't download the ubuntu Kernel sources as described but clone the ALSA repository (which you already did). And instead of downloading the patches as described, just use "git apply" on the file I sent you. On case you want to revert the patching and return to the kernel you originally cloned, just call "git reset --hard". Hope this helps, looking forward to your results. Thanks, Daniel

3 13

[alsa-devel] aplay can't run in background since adding pause support
by Stephen Warren 24 May '11

24 May '11

Jaroslav, In recent versions of aplay, if I try to either run them in the background, or run them in the foreground, then ^Z and bg them, they hang indefinitely: aplay foo.wav & # doesn't play anything aplay foo.wav ... ^Z bg [1]+ Stopped aplay foo.wav bg [1]+ Stopped aplay foo.wav At least, this is true under ChromeOS on ARM with kernel 2.6.38 and ALSA 1.0.24.2, or an Ubuntu Natty rootstock on ARM with the same ChromeOS kernel. This was introduced by alsa-utils commit: 3bd65336222a4d00cefc4db5e74a7a96c07ab567 aplay/arecord: Added hardware pause support (press SPACE or Enter) What happens is that the call to tcsetattr causes the process to receive signal TTOU and be stopped. Running under strace yields an infinite loop with ioctl(TCSETS) returning -ERESTARTSYS, and being retried. I believe this is related to http://lkml.org/lkml/2008/6/2/157. I'm not exactly sure how to solve this though. Another post in that thread implied that adding a signal handler for TTOU and then adjusting the tcsetattr settings might help, but I'm not convinced about that, since I believe the -ERESTARTSYS handling loop is inside the call to tcsetattr, and hence tcsetattr's parameters can't be changed. Does anyone have any ideas? Thanks. -- nvpublic

2 2

[alsa-devel] [PATCH] ASoC: core - Allow some components to probe/remove later or earlier.
by Liam Girdwood 24 May '11

24 May '11

Some ASoC components depend on other ASoC components to provide clocks and power resources in order to probe(). Provide a method to allow some components to be probed() later and be also removed() earlier. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dai.h | 4 +++ include/sound/soc.h | 8 +++++++ sound/soc/soc-core.c | 49 ++++++++++++++++++++++++++++++++++------------ 3 files changed, 48 insertions(+), 13 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 1bafe95..ada1f24 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -209,6 +209,10 @@ struct snd_soc_dai_driver { struct snd_soc_pcm_stream capture; struct snd_soc_pcm_stream playback; unsigned int symmetric_rates:1; + + /* probe ordering - for components with runtime dependencies */ + bool late_probe; + bool early_remove; }; /* diff --git a/include/sound/soc.h b/include/sound/soc.h index f1de3e0..a2139d4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -613,6 +613,10 @@ struct snd_soc_codec_driver { void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); + + /* probe ordering - for components with runtime dependencies */ + bool late_probe; + bool early_remove; }; /* SoC platform interface */ @@ -637,6 +641,10 @@ struct snd_soc_platform_driver { /* platform stream ops */ struct snd_pcm_ops *ops; + + /* probe ordering - for components with runtime dependencies */ + bool late_probe; + bool early_remove; }; struct snd_soc_platform { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f9667f6..321182e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1393,7 +1393,7 @@ static void soc_remove_codec(struct snd_soc_codec *codec) module_put(codec->dev->driver->owner); } -static void soc_remove_dai_link(struct snd_soc_card *card, int num) +static void soc_remove_dai_link(struct snd_soc_card *card, int num, int early) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_codec *codec = rtd->codec; @@ -1410,7 +1410,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) } /* remove the CODEC DAI */ - if (codec_dai && codec_dai->probed) { + if (codec_dai && codec_dai->probed && + codec_dai->driver->early_remove == early) { if (codec_dai->driver->remove) { err = codec_dai->driver->remove(codec_dai); if (err < 0) @@ -1421,7 +1422,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) } /* remove the platform */ - if (platform && platform->probed) { + if (platform && platform->probed && + platform->driver->early_remove == early) { if (platform->driver->remove) { err = platform->driver->remove(platform); if (err < 0) @@ -1433,11 +1435,13 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) } /* remove the CODEC */ - if (codec && codec->probed) + if (codec && codec->probed && + codec->driver->early_remove == early) soc_remove_codec(codec); /* remove the cpu_dai */ - if (cpu_dai && cpu_dai->probed) { + if (cpu_dai && cpu_dai->probed && + cpu_dai->driver->early_remove == early) { if (cpu_dai->driver->remove) { err = cpu_dai->driver->remove(cpu_dai); if (err < 0) @@ -1454,7 +1458,9 @@ static void soc_remove_dai_links(struct snd_soc_card *card) int i; for (i = 0; i < card->num_rtd; i++) - soc_remove_dai_link(card, i); + soc_remove_dai_link(card, i, 1); /* early remove */ + for (i = 0; i < card->num_rtd; i++) + soc_remove_dai_link(card, i, 0); /* late remove */ card->num_rtd = 0; } @@ -1596,7 +1602,7 @@ static int soc_post_component_init(struct snd_soc_card *card, return 0; } -static int soc_probe_dai_link(struct snd_soc_card *card, int num) +static int soc_probe_dai_link(struct snd_soc_card *card, int num, int late) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; @@ -1605,7 +1611,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; int ret; - dev_dbg(card->dev, "probe %s dai link %d\n", card->name, num); + dev_dbg(card->dev, "probe %s dai link %d late %d\n", card->name, num, late); /* config components */ codec_dai->codec = codec; @@ -1617,7 +1623,8 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) rtd->pmdown_time = pmdown_time; /* probe the cpu_dai */ - if (!cpu_dai->probed) { + if (!cpu_dai->probed && + cpu_dai->driver->late_probe == late) { if (!try_module_get(cpu_dai->dev->driver->owner)) return -ENODEV; @@ -1636,14 +1643,16 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) } /* probe the CODEC */ - if (!codec->probed) { + if (!codec->probed && + codec->driver->late_probe == late) { ret = soc_probe_codec(card, codec); if (ret < 0) return ret; } /* probe the platform */ - if (!platform->probed) { + if (!platform->probed && + platform->driver->late_probe == late) { if (!try_module_get(platform->dev->driver->owner)) return -ENODEV; @@ -1662,7 +1671,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) } /* probe the CODEC DAI */ - if (!codec_dai->probed) { + if (!codec_dai->probed && codec_dai->driver->late_probe == late) { if (codec_dai->driver->probe) { ret = codec_dai->driver->probe(codec_dai); if (ret < 0) { @@ -1677,6 +1686,10 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) list_add(&codec_dai->card_list, &card->dai_dev_list); } + /* complete DAI probe during late probe */ + if (!late) + return 0; + /* DAPM dai link stream work */ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); @@ -1895,8 +1908,18 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) goto card_probe_error; } + /* early DAI link probe */ + for (i = 0; i < card->num_links; i++) { + ret = soc_probe_dai_link(card, i, 0); + if (ret < 0) { + pr_err("asoc: failed to instantiate card %s: %d\n", + card->name, ret); + goto probe_dai_err; + } + } + /* late DAI link probe */ for (i = 0; i < card->num_links; i++) { - ret = soc_probe_dai_link(card, i); + ret = soc_probe_dai_link(card, i, 1); if (ret < 0) { pr_err("asoc: failed to instantiate card %s: %d\n", card->name, ret); -- 1.7.4.1

1 0

[alsa-devel] [PATCH] ASoC: dapm - add method to get AIF widget from stream
by Liam Girdwood 24 May '11

24 May '11

In preparation for ASoC DSP support. Provide a method to get the AIF widget name from the stream name. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 2 ++ sound/soc/soc-dapm.c | 17 +++++++++++++++++ 2 files changed, 19 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index c46e7d8..f37c293 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -348,6 +348,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm); void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm); int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num); +const char *snd_soc_dapm_get_aif(struct snd_soc_dapm_context *dapm, + const char *stream_name, enum snd_soc_dapm_type type); /* dapm events */ int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c5d98e3..1b7f4ff 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1882,6 +1882,23 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); +const char *snd_soc_dapm_get_aif(struct snd_soc_dapm_context *dapm, + const char *stream_name, enum snd_soc_dapm_type type) +{ + struct snd_soc_dapm_widget *w; + + list_for_each_entry(w, &dapm->card->widgets, list) { + + if (!w->sname) + continue; + + if (w->id == type && strstr(w->sname, stream_name)) + return w->name; + } + return NULL; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_aif); + /** * snd_soc_dapm_get_volsw - dapm mixer get callback * @kcontrol: mixer control -- 1.7.4.1

1 0

[alsa-devel] [PATCH] ASoC: core - remove superfluous new line.
by Liam Girdwood 24 May '11

24 May '11

Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-core.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f051e85..f9667f6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1617,7 +1617,6 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) rtd->pmdown_time = pmdown_time; /* probe the cpu_dai */ - if (!cpu_dai->probed) { if (!try_module_get(cpu_dai->dev->driver->owner)) return -ENODEV; -- 1.7.4.1

1 0

[alsa-devel] Regression post-ALSA 1.0.22 for Lenovo G555 Internal mic
by Roo - Andrew Baines 24 May '11

24 May '11

Dear All, I have been told I should send an email to this list from this bug<https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5367> (0005367). There is a regression in ALSA 1.0.23 and above for Lenovo sound. The internal mic no longer works. A helpful guy 'Raymond' on the bug identified that pin-ctls is correct in ALSA 1.0.21 and 1.0.22, but incorrect in higher versions due to SND_PCI_QUIRK_VENDOR. This is based on alsa-info.sh run on systems using different versions of ALSA but on the same system. The jacksense has never actually worked but I have always managed to use hda-verb to do this for me - the main issue is the regression with the internal mic. Regards Andrew

2 13

[alsa-devel] [PATCH] ASoC: core - fix module reference counting for CPU DAIs
by Liam Girdwood 24 May '11

24 May '11

Currently CODEC and platform drivers have their module reference count incremented soc_probe_dai_link() whilst CPU DAI drivers have their reference count incremented in soc_bind_dai_link(). CPU DAIs should have their reference count incremented in soc_probe_dai_link() just like the CODEC and platform drivers. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-core.c | 9 +++++---- 1 files changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5968745..f051e85 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1306,10 +1306,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) /* no, then find CPU DAI from registered DAIs*/ list_for_each_entry(cpu_dai, &dai_list, list) { if (!strcmp(cpu_dai->name, dai_link->cpu_dai_name)) { - - if (!try_module_get(cpu_dai->dev->driver->owner)) - return -ENODEV; - rtd->cpu_dai = cpu_dai; goto find_codec; } @@ -1621,12 +1617,17 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) rtd->pmdown_time = pmdown_time; /* probe the cpu_dai */ + if (!cpu_dai->probed) { + if (!try_module_get(cpu_dai->dev->driver->owner)) + return -ENODEV; + if (cpu_dai->driver->probe) { ret = cpu_dai->driver->probe(cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: failed to probe CPU DAI %s\n", cpu_dai->name); + module_put(cpu_dai->dev->driver->owner); return ret; } } -- 1.7.4.1

2 2

[alsa-devel] [PATCH] ALSA: ASoC: fix raumfeld platform
by Daniel Mack 24 May '11

24 May '11

Commit f0fba2ad (ASoC: multi-component - ASoC Multi-Component Support) broke support for Raumfeld platforms as it didn't take into account the different hardware features on individual devices. In particular, Raumfeld speakers have no S/PDIF output, so the members of the snd_soc_card struct must be set dynamically. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/pxa/raumfeld.c | 92 +++++++++++++++++++++++++++++----------------- 1 files changed, 58 insertions(+), 34 deletions(-) diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 2afabaf..1a591f1 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -151,13 +151,13 @@ static struct snd_soc_ops raumfeld_cs4270_ops = { .hw_params = raumfeld_cs4270_hw_params, }; -static int raumfeld_line_suspend(struct snd_soc_card *card) +static int raumfeld_analog_suspend(struct snd_soc_card *card) { raumfeld_enable_audio(false); return 0; } -static int raumfeld_line_resume(struct snd_soc_card *card) +static int raumfeld_analog_resume(struct snd_soc_card *card) { raumfeld_enable_audio(true); return 0; @@ -225,32 +225,53 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .hw_params = raumfeld_ak4104_hw_params, }; -static struct snd_soc_dai_link raumfeld_dai[] = { +#define DAI_LINK_CS4270 \ +{ \ + .name = "CS4270", \ + .stream_name = "CS4270", \ + .cpu_dai_name = "pxa-ssp-dai.0", \ + .platform_name = "pxa-pcm-audio", \ + .codec_dai_name = "cs4270-hifi", \ + .codec_name = "cs4270-codec.0-0048", \ + .ops = &raumfeld_cs4270_ops, \ +} + +#define DAI_LINK_AK4104 \ +{ \ + .name = "ak4104", \ + .stream_name = "Playback", \ + .cpu_dai_name = "pxa-ssp-dai.1", \ + .codec_dai_name = "ak4104-hifi", \ + .platform_name = "pxa-pcm-audio", \ + .ops = &raumfeld_ak4104_ops, \ + .codec_name = "spi0.0", \ +} + +static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] = { - .name = "ak4104", - .stream_name = "Playback", - .cpu_dai_name = "pxa-ssp-dai.1", - .codec_dai_name = "ak4104-hifi", - .platform_name = "pxa-pcm-audio", - .ops = &raumfeld_ak4104_ops, - .codec_name = "ak4104-codec.0", -}, + DAI_LINK_CS4270, + DAI_LINK_AK4104, +}; + +static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] = { - .name = "CS4270", - .stream_name = "CS4270", - .cpu_dai_name = "pxa-ssp-dai.0", - .platform_name = "pxa-pcm-audio", - .codec_dai_name = "cs4270-hifi", - .codec_name = "cs4270-codec.0-0048", - .ops = &raumfeld_cs4270_ops, -},}; - -static struct snd_soc_card snd_soc_raumfeld = { - .name = "Raumfeld", - .dai_link = raumfeld_dai, - .suspend_post = raumfeld_line_suspend, - .resume_pre = raumfeld_line_resume, - .num_links = ARRAY_SIZE(raumfeld_dai), + DAI_LINK_CS4270, +}; + +static struct snd_soc_card snd_soc_raumfeld_connector = { + .name = "Raumfeld Connector", + .dai_link = snd_soc_raumfeld_connector_dai, + .num_links = ARRAY_SIZE(snd_soc_raumfeld_connector_dai), + .suspend_post = raumfeld_analog_suspend, + .resume_pre = raumfeld_analog_resume, +}; + +static struct snd_soc_card snd_soc_raumfeld_speaker = { + .name = "Raumfeld Speaker", + .dai_link = snd_soc_raumfeld_speaker_dai, + .num_links = ARRAY_SIZE(snd_soc_raumfeld_speaker_dai), + .suspend_post = raumfeld_analog_suspend, + .resume_pre = raumfeld_analog_resume, }; static struct platform_device *raumfeld_audio_device; @@ -271,22 +292,25 @@ static int __init raumfeld_audio_init(void) set_max9485_clk(MAX9485_MCLK_FREQ_122880); - /* Register LINE and SPDIF */ + /* Register analog device */ raumfeld_audio_device = platform_device_alloc("soc-audio", 0); if (!raumfeld_audio_device) return -ENOMEM; - platform_set_drvdata(raumfeld_audio_device, - &snd_soc_raumfeld); - ret = platform_device_add(raumfeld_audio_device); - - /* no S/PDIF on Speakers */ if (machine_is_raumfeld_speaker()) + platform_set_drvdata(raumfeld_audio_device, + &snd_soc_raumfeld_speaker); + + if (machine_is_raumfeld_connector()) + platform_set_drvdata(raumfeld_audio_device, + &snd_soc_raumfeld_connector); + + ret = platform_device_add(raumfeld_audio_device); + if (ret < 0) return ret; raumfeld_enable_audio(true); - - return ret; + return 0; } static void __exit raumfeld_audio_exit(void) -- 1.7.5.1

3 2

[alsa-devel] Clarrification about the alsa volume init framework.
by Luke Yelavich 24 May '11

24 May '11

Hi, So I am looking at cleaning up Ubuntu's collection of volume init db data originally set up in Ubuntu's/Debian old big init script. The switch has already been made to the new init system, and any existing settings were migrated from our old init script to the new system, however this is somewhat unclean. Is my understanding correct in that the new volume init system was designed to have quirks for hardware that needs particular volume settings? If so, how would upstream like these to be submitted, and would you want to have several dozen or more quirk files in the init db for different hardware? Thanks in advance. Luke

2 3

[alsa-devel] Problems with snd_hda_intel in Linux kernel 2.6.38
by Svante Signell 24 May '11

24 May '11

Now I'm subscribed to the list. It does not seem possible to get a message in without subscription. Would be nice if alsa had and used bug reporting pages outside the mailing list. > Trying again to get this message to the alsa-devel list, as recommended > by the Debian maintainer of the kernel, without being subscribed. Does > alsa not have a bug report page? > > Booting kernel 2.6.38 hangs at: > During boot of kernel 2.6.38 (and 2.6.37) udev bugs out: > Waiting for /dev to be fully populated > BUG: Unable to handle kernel paging request at ffffc90013cd8000 > axz_probe+ ... [snd_hda_intel] > ...lots of output lost... > udevadm timeout 180 sec ... > udevd[390]: worker [439] failed while handling > '/devices/pci0000:80/0000:80:01.0' > > No sound card is recognized after the timeout of the boot as shown above. > According to the /proc/asound/cards there are no sound cards > > See bug Debian bug #619034 for more information. > http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=619034 > > This problem was also reported as Debian bug #613979, as kernel ticket > > https://bugzilla.kernel.org/show_bug.cgi?id=30552 > > and sent to this mailing list in February 2011. > > snd_hda_intel works properly with 2.6.32-5-amd64 (and earlier kernels). >

4 15

[alsa-devel] [PATCH v3 0/3] ASoC: add support for audio on iPaq hx4700
by Dmitry Artamonow 24 May '11

24 May '11

Ok, here's another version of patches adding audio support for good ole iPAQ hx4700. It took a bit faster to make v3 than I expected, so here is it. Previous versions can be found here: https://lkml.org/lkml/2010/11/20/50 (v1) https://lkml.org/lkml/2011/5/16/462 (v2) This patchset is based on 'for-next' branch of Mark's ASoC git tree (fetched today), also applies cleanly on linux-next-20110518. Lightly tested on real hardware - playback/recording/mixer seem to work without any glitches. Changes v2 -> v3 * data based init is now used in codec and machine audio driver for DAPM and controls * machine audio driver now is a proper platform driver, instantiating sound card with snd_soc_register_card() * platform code registers corresponding platform_device for machine audio driver * ak4641_sync() is dropped in favour of snd_soc_cache_sync() * snd_soc_update_bits() is used instead of snd_soc_read()/update/snd_soc_write() sequences. Also return value of snd_soc_update_bits is propagated further for catching errors (previously result of snd_soc_write wasn't checked) * gpio_request_array is used now in machine audio driver * some other minor cleanup in the couple of places Dmitry Artamonow (3): ASoC: Asahi Kasei AK4641 codec driver ASoC: add iPAQ hx4700 machine driver pxa/hx4700: add platform device and I2C info for AK4641 codec arch/arm/mach-pxa/hx4700.c | 25 ++ arch/arm/mach-pxa/include/mach/hx4700.h | 4 +- include/sound/ak4641.h | 26 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4641.c | 664 +++++++++++++++++++++++++++++++ sound/soc/codecs/ak4641.h | 47 +++ sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/hx4700.c | 255 ++++++++++++ 10 files changed, 1036 insertions(+), 2 deletions(-) create mode 100644 include/sound/ak4641.h create mode 100644 sound/soc/codecs/ak4641.c create mode 100644 sound/soc/codecs/ak4641.h create mode 100644 sound/soc/pxa/hx4700.c -- 1.7.4.rc3

3 6

[alsa-devel] [PATCH] ASoC: Convert 16x16 write to use cpu_to_be16()
by Mark Brown 24 May '11

24 May '11

Make it clear what we're doing. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 10 ++++------ 1 files changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 06b7b81..abdf8d1 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -277,14 +277,12 @@ static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec, static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - u8 data[4]; + u16 data[2]; - data[0] = (reg >> 8) & 0xff; - data[1] = reg & 0xff; - data[2] = (value >> 8) & 0xff; - data[3] = value & 0xff; + data[0] = cpu_to_be16(reg); + data[1] = cpu_to_be16(value); - return do_hw_write(codec, reg, value, data, 4); + return do_hw_write(codec, reg, value, data, sizeof(data)); } /* Primitive bulk write support for soc-cache. The data pointed to by -- 1.7.5.1

2 1

[alsa-devel] ALSA and MediaController
by pl bossart 23 May '11

23 May '11

During the ALSA/Embedded audio workshop two weeks ago we talked about using the MediaController API (merged in 2.6.39) to provide userspace with a graph-like representation of the transforms implemented in firmware or hardware. Two main applications were mentioned: - volume control, at the moment in PulseAudio we get a flat list of volume controls and we use a convoluted mixer logic to figure out if 'pcm playback volume' is done before or after 'master playback volume'. - tuning applications. With literally dozens of controls exposed in embedded solutions, a flat list is going to be very hard to use. It would make sense during the device tuning steps to figure out which controls is located where to better understand its impacts. This would help create UCM configuration files as well. At the moment the Media Controller does not provide means to set any values/parameters in a media_entity. We would need to cross-reference each media_entity with ALSA controls and use the existing control get/set interfaces. This raises 2 main questions: - Is the list of ALSA controls is completely defined after the card initialization step? Or do we have cases where ALSA controls are created dynamically after the init? I tend to believe the former is true but want to verify this with others on the mailing list. It's my understanding that only connections may be changed in such a graph. - Laurent suggested a new ioctl to export media_entity information to userspace, as a binary buffer containing a list of TLV (Type-Length-Value) field. However it looks like there's already an ELEM_BYTES control type. It might make more sense to expose this information using native ALSA controls, but again feedback from smart people on this list would be welcome. Thanks, -Pierre

4 8

[alsa-devel] [PATCH v5] [resend] tea575x: convert to control framework
by Ondrej Zary 23 May '11

23 May '11

Convert tea575x-tuner to use the new V4L2 control framework. Also add ext_init() callback that can be used by a card driver for additional initialization right before registering the video device (for SF16-FMR2). Also embed struct video_device to struct snd_tea575x to simplify the code. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-2.6.39-rc2-/include/sound/tea575x-tuner.h 2011-05-13 19:39:23.000000000 +0200 +++ linux-2.6.39-rc2/include/sound/tea575x-tuner.h 2011-05-19 17:27:45.000000000 +0200 @@ -23,8 +23,8 @@ */ #include <linux/videodev2.h> +#include <media/v4l2-ctrls.h> #include <media/v4l2-dev.h> -#include <media/v4l2-ioctl.h> #define TEA575X_FMIF 10700 @@ -42,7 +42,7 @@ struct snd_tea575x_ops { }; struct snd_tea575x { - struct video_device *vd; /* video device */ + struct video_device vd; /* video device */ bool tea5759; /* 5759 chip is present */ bool mute; /* Device is muted? */ bool stereo; /* receiving stereo */ @@ -54,6 +54,8 @@ struct snd_tea575x { void *private_data; u8 card[32]; u8 bus_info[32]; + struct v4l2_ctrl_handler ctrl_handler; + int (*ext_init)(struct snd_tea575x *tea); }; int snd_tea575x_init(struct snd_tea575x *tea); --- linux-2.6.39-rc2-/sound/i2c/other/tea575x-tuner.c 2011-05-13 19:39:23.000000000 +0200 +++ linux-2.6.39-rc2/sound/i2c/other/tea575x-tuner.c 2011-05-19 17:31:45.000000000 +0200 @@ -22,11 +22,11 @@ #include <asm/io.h> #include <linux/delay.h> -#include <linux/interrupt.h> #include <linux/init.h> #include <linux/slab.h> #include <linux/version.h> -#include <sound/core.h> +#include <media/v4l2-dev.h> +#include <media/v4l2-ioctl.h> #include <sound/tea575x-tuner.h> MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); @@ -62,17 +62,6 @@ module_param(radio_nr, int, 0); #define TEA575X_BIT_DUMMY (1<<15) /* buffer */ #define TEA575X_BIT_FREQ_MASK 0x7fff -static struct v4l2_queryctrl radio_qctrl[] = { - { - .id = V4L2_CID_AUDIO_MUTE, - .name = "Mute", - .minimum = 0, - .maximum = 1, - .default_value = 1, - .type = V4L2_CTRL_TYPE_BOOLEAN, - } -}; - /* * lowlevel part */ @@ -266,47 +255,17 @@ static int vidioc_s_audio(struct file *f return 0; } -static int vidioc_queryctrl(struct file *file, void *priv, - struct v4l2_queryctrl *qc) +static int tea575x_s_ctrl(struct v4l2_ctrl *ctrl) { - int i; - - for (i = 0; i < ARRAY_SIZE(radio_qctrl); i++) { - if (qc->id && qc->id == radio_qctrl[i].id) { - memcpy(qc, &(radio_qctrl[i]), - sizeof(*qc)); - return 0; - } - } - return -EINVAL; -} - -static int vidioc_g_ctrl(struct file *file, void *priv, - struct v4l2_control *ctrl) -{ - struct snd_tea575x *tea = video_drvdata(file); + struct snd_tea575x *tea = container_of(ctrl->handler, struct snd_tea575x, ctrl_handler); switch (ctrl->id) { case V4L2_CID_AUDIO_MUTE: - ctrl->value = tea->mute; + tea->mute = ctrl->val; + snd_tea575x_set_freq(tea); return 0; } - return -EINVAL; -} -static int vidioc_s_ctrl(struct file *file, void *priv, - struct v4l2_control *ctrl) -{ - struct snd_tea575x *tea = video_drvdata(file); - - switch (ctrl->id) { - case V4L2_CID_AUDIO_MUTE: - if (tea->mute != ctrl->value) { - tea->mute = ctrl->value; - snd_tea575x_set_freq(tea); - } - return 0; - } return -EINVAL; } @@ -355,16 +314,17 @@ static const struct v4l2_ioctl_ops tea57 .vidioc_s_input = vidioc_s_input, .vidioc_g_frequency = vidioc_g_frequency, .vidioc_s_frequency = vidioc_s_frequency, - .vidioc_queryctrl = vidioc_queryctrl, - .vidioc_g_ctrl = vidioc_g_ctrl, - .vidioc_s_ctrl = vidioc_s_ctrl, }; static struct video_device tea575x_radio = { .name = "tea575x-tuner", .fops = &tea575x_fops, .ioctl_ops = &tea575x_ioctl_ops, - .release = video_device_release, + .release = video_device_release_empty, +}; + +static const struct v4l2_ctrl_ops tea575x_ctrl_ops = { + .s_ctrl = tea575x_s_ctrl, }; /* @@ -373,7 +333,6 @@ static struct video_device tea575x_radio int snd_tea575x_init(struct snd_tea575x *tea) { int retval; - struct video_device *tea575x_radio_inst; tea->mute = 1; @@ -384,40 +343,45 @@ int snd_tea575x_init(struct snd_tea575x tea->in_use = 0; tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_10_40; tea->freq = 90500 * 16; /* 90.5Mhz default */ + snd_tea575x_set_freq(tea); - tea575x_radio_inst = video_device_alloc(); - if (tea575x_radio_inst == NULL) { - printk(KERN_ERR "tea575x-tuner: not enough memory\n"); - return -ENOMEM; - } + tea->vd = tea575x_radio; + video_set_drvdata(&tea->vd, tea); - memcpy(tea575x_radio_inst, &tea575x_radio, sizeof(tea575x_radio)); + v4l2_ctrl_handler_init(&tea->ctrl_handler, 1); + tea->vd.ctrl_handler = &tea->ctrl_handler; + v4l2_ctrl_new_std(&tea->ctrl_handler, &tea575x_ctrl_ops, V4L2_CID_AUDIO_MUTE, 0, 1, 1, 1); + retval = tea->ctrl_handler.error; + if (retval) { + printk(KERN_ERR "tea575x-tuner: can't initialize controls\n"); + v4l2_ctrl_handler_free(&tea->ctrl_handler); + return retval; + } - strcpy(tea575x_radio.name, tea->tea5759 ? - "TEA5759 radio" : "TEA5757 radio"); + if (tea->ext_init) { + retval = tea->ext_init(tea); + if (retval) { + v4l2_ctrl_handler_free(&tea->ctrl_handler); + return retval; + } + } - video_set_drvdata(tea575x_radio_inst, tea); + v4l2_ctrl_handler_setup(&tea->ctrl_handler); - retval = video_register_device(tea575x_radio_inst, - VFL_TYPE_RADIO, radio_nr); + retval = video_register_device(&tea->vd, VFL_TYPE_RADIO, radio_nr); if (retval) { printk(KERN_ERR "tea575x-tuner: can't register video device!\n"); - kfree(tea575x_radio_inst); + v4l2_ctrl_handler_free(&tea->ctrl_handler); return retval; } - snd_tea575x_set_freq(tea); - tea->vd = tea575x_radio_inst; - return 0; } void snd_tea575x_exit(struct snd_tea575x *tea) { - if (tea->vd) { - video_unregister_device(tea->vd); - tea->vd = NULL; - } + video_unregister_device(&tea->vd); + v4l2_ctrl_handler_free(&tea->ctrl_handler); } static int __init alsa_tea575x_module_init(void) -- Ondrej Zary

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[alsa-devel] [PATCH 1/2] ASoC: tlv320aic3x: Don't sync first two registers from register cache
by Jarkko Nikula 23 May '11

23 May '11

There is no need to sync first two registers from cache to hw after a reset. First one is used to select page for register access and this driver is normally accessing page 0 only. Second one does a software reset which is obviously unneeded after hardware or previous software reset command. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/codecs/tlv320aic3x.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index c3d96fc..9047bb1 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1114,7 +1114,7 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power) /* Sync reg_cache with the hardware */ codec->cache_only = 0; - for (i = 0; i < ARRAY_SIZE(aic3x_reg); i++) + for (i = AIC3X_SAMPLE_RATE_SEL_REG; i < ARRAY_SIZE(aic3x_reg); i++) snd_soc_write(codec, i, cache[i]); if (aic3x->model == AIC3X_MODEL_3007) aic3x_init_3007(codec); -- 1.7.4.4

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[alsa-devel] [PATCH] ALSA - HDA - Change "Mic" to "Rear Mic" in ad1988_6stack_capture_source
by Raymond Yau 23 May '11

23 May '11

Change "Mic" to "Rear Mic" in ad1988_6stack_capture_source

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[alsa-devel] [PATCH] ALSA: HDA: Add quirk for Lenovo U350
by David Henningsson 23 May '11

23 May '11

Since the auto-parser for Conexant 5066 is yet not activated, I guess it makes sense to keep adding quirks? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[PATCH] ALSA: HDA: Add quirk for Lenovo U350
by David Henningsson 23 May '11

23 May '11

Add model=asus quirk for Lenovo Ideapad U350 to make internal mic work correctly. Cc: stable(a)kernel.org (2.6.38+) BugLink: http://bugs.launchpad.net/bugs/751681 Reported-by: Kent Baxley <kent.baxley(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4f37477..e8725d2 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3098,6 +3098,7 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ {} -- 1.7.4.1 --------------090201020202020607060101--

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[alsa-devel] ALSA pcm_lib.c:1789: playback write error (DMA or IRQ trouble?) ALSA pcm_native.c:1531: playback drain error
by Raymond Yau 23 May '11

23 May '11

For those HDA Codec which support rate lower than 44100 Hz with prealloc = 4096 ALSA pcm_lib.c:1789: playback write error (DMA or IRQ trouble?) ALSA hda_intel.c:1678: azx_pcm_prepare: bufsize=0xd7500, format=0x4311 This error occr when aplay using a period time longer than 10 seconds ( usually occur above 12 seconds) aplay -v -Dhw:1 --period-time=12000000 11025.wav Playing WAVE '11025.wav' : Signed 16 bit Little Endian, Rate 11025 Hz, Stereo Hardware PCM card 1 'HDA Intel' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 11025 exact rate : 11025 (11025/1) msbits : 16 buffer_size : 264576 period_size : 132288 period_time : 11998911 tstamp_mode : NONE period_step : 1 avail_min : 132288 period_event : 0 start_threshold : 264576 stop_threshold : 264576 silence_threshold: 0 silence_size : 0 boundary : 1083703296 appl_ptr : 0 hw_ptr : 0 ALSA pcm_native.c:1531: playback drain error (DMA or IRQ trouble?) ALSA hda_intel.c:1678: azx_pcm_prepare: bufsize=0x400000, format=0x511 This error occur with speaker-test which use maximum buffer size speaker-test -d -c 2 -t sine --rate=8000 -Dhw:CARD=Intel speaker-test 1.0.24.2 Playback device is hw:CARD=Intel Stream parameters are 8000Hz, S16_LE, 2 channels Sine wave rate is 440.0000Hz Rate set to 8000Hz (requested 8000Hz) Buffer size range from 64 to 1048576 Period size range from 32 to 524288 Using max buffer size 1048576 Periods = 4 was set period_size = 262144 was set buffer_size = 1048576 Hardware PCM card 1 'HDA Intel' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 1048576 period_size : 262144 period_time : 32768000 tstamp_mode : NONE period_step : 1 avail_min : 262144 period_event : 0 start_threshold : 1048576 stop_threshold : 1048576 silence_threshold: 0 silence_size : 0 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 0 - Front Left 1 - Front Right Time per period = 20.075443

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[alsa-devel] Does amixer has to be used to make a complete path in ASoC?
by Feng Ye 23 May '11

23 May '11

Hello, I wrote ASoC driver with interacts with WM8750 codec chip. It does not work, no sound at all. Some codec registers are not programmed right, and the check_power callback (in soc-dapm.c) always return 0. I think the problem may not be in the driver, instead, amixer should be used to select the codec mixer settings to build a complete audio path (like the LEFT MIXER in wm8750). I have following questions: 1. Is my understanding correct? amixer has to be used before aplay? 2. Is there way to do it in the code instead of using amixer? 3. If amixer has to be used, where I can find sample wm8750 amixer setting usage for a simple playback from I2S in to Lout1 & Rout1 out? Here's the simple code in my driver, just FYI: static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Speaker", NULL), }; static const struct snd_soc_dapm_route wm8750_audio_map[] = { { "Ext Speaker", NULL, "LOUT1" }, { "Ext Speaker", NULL, "ROUT1" }, }; ... (in dai_link init function): err = snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, ARRAY_SIZE(wm8750_dapm_widgets)); ... snd_soc_dapm_add_routes(codec, wm8750_audio_map, ARRAY_SIZE(wm8750_audio_map)); snd_soc_dapm_sync(codec); ... (in snd_soc_ops startup function): snd_soc_dapm_enable_pin(codec, "Ext Speaker"); Thanks, Feng

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[alsa-devel] ask about audio latency
by 周勇发 22 May '11

22 May '11

hi I want to get the latency between record and playback, so i use following solution to calculate it. #audio.sh arecord -d 3 -r 96000 in.wav & sleep 1 aplay mode.wav model.wav is a sine wave. this file is used for calculate the delta timestamp of record and playback. I have append do_gettimeofday() function in azx_stream_start() in hda_intel.c file. it will printk the dma trigger time after trigger dma. the result is about 12ms. as far as i known. fifo size is 192, sample rate is 96000, so it will take about 2ms latency, my question is where does the other 10ms latency come from? Thanks -- Best Regards！

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[alsa-devel] ASoC updates for 2.6.40
by Mark Brown 22 May '11

22 May '11

The following changes since commit 4a787a3ff3f419c23ab0a5cef677fa441356b818: Merge branch 'for-2.6.40' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc (2011-05-20 11:25:32 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.40 A few more small fixes. Jarkko Nikula (1): ASoC: core: Don't set "(null)" as a driver name Mark Brown (1): Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-2.6.40 Stephen Warren (2): ASoC: spdif-dit: Add missing MODULE_* ASoC: Tegra: Fix compile when debugfs not enabled sound/soc/codecs/spdif_transciever.c | 8 ++++++-- sound/soc/soc-core.c | 2 +- sound/soc/tegra/tegra_i2s.c | 2 +- 3 files changed, 8 insertions(+), 4 deletions(-)

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Re: [alsa-devel] playback write error(DMA or IRQ trouble?)
by innoway＠mymelody.com 22 May '11

22 May '11

I'll try to fix the programs from this Monday and I think I have to learn alsa system more deeply because I 'm a newcomer about it. Thank you 2011/5/20 <innoway(a)mymelody.com>: > I develop an embedded system on the Linux board with Freescale i.MX27 > (ARM processor). I succeed on sound, which is transferred using > Synchronous Serial Interface (SSI). It,s fine and then I try to do test > of inserting or pulling out a USB memory stick in the USB host connector > while sounding. I get the following error message from > snd_pcm_lib_write1() of <linux src>/sound /core/pcm_lib.c after waiting > for a while (about 10sec). playback write error (DMA or IRQ > trouble?) and then from snd_pcm_drain() of <linux > src>/sound/core/pcm_native.c playback drain error (DMA or IRQ > trouble?) (No sound during this period) > > and then it is sounding again. > > I want to prevent the sound from stopping even if something happens > suddenly such as the USB memory stick is plugged in or out. > > I try to fix the above programs, but I don't succeed. > > Can someone give me the clue? [13:21:38]:/proc# cat interrupts > CPU0 > 12: 3456 - mxc_i2c > 16: 141 - CSPI_IRQ > 18: 952 - mxcintuart > 26: 1051255 - MXC Timer Tick > 28: 29446 - PCMCIA IRQ, ide0 > 50: 33 - fec > 54: 793 - ehci_hcd:usb1 > 56: 1 - arc_udc > 61: 0 - LCDC > 117: 0 - m66291_udc > 147: 0 - PMIC_IRQ > 157: 337179 - SMSC_LAN911x_ISR > 176: 0 - fec(MII) > > [13:35:03]:/proc# cat dma > dma channel 0: unused > dma channel 1: unused > dma channel 2: unused > dma channel 3: unused > dma channel 4: unused > dma channel 5: unused > dma channel 6: unused > dma channel 7: unused > dma channel 8: unused > dma channel 9: unused > dma channel 10: unused > dma channel 11: unused > dma channel 12: unused > dma channel 13: unused > dma channel 14: unused > dma channel 15: unused Thank you > > Masanori Inoue > This error also occur in snd-hda-intel echo 32768 > /proc/asound/Intel/pcm0p/sub0/prealloc speaker-test which use maximum buffer size also have (DMA or IRQ error in system log THis seem to be related to the schedule_timeout tout = schedule_timeout(msecs_to_jiffies(10000)); I guess this is why PA server treat avail > 10 seconds is invalid _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] UCM files for tegra-wm8903 on seaboard
by Stephen Warren 21 May '11

21 May '11

Liam, I've come up with a basic set of UCM files for Tegra Seaboard, included below. I briefly tested parts of them with alsaucm. I do have a few questions on UCM and how to write these files before I go too far: Some of the answers may depend on, or be impacted by, policy decisions by the user of UCM; general desktop PulseAudio may have different requirements to a more tightly controlled ChromeOS configuration. I'm not sure how much that impacts the UCM files themselves, or whether the UCM files are supposed to be fully general and hence the same for all OSs and policy setups. 1) Can multiple verbs be active at once? In other words, I have cases defined for HiFi (playback) and Record (capture). Should these be merged together such that both playback and capture can operate at once, or can an application just active both verbs at once when it wants to do multiple things? 2) In the ChromeOS kernel, there's a separate PCM for playback to WM8903 and playback to the SPDIF controller, and hence on to HDMI. Since these use different values for PlaybackPCM, I defined separate use-cases for analog and HDMI/SPDIF playback. Is this the typical way of splitting things up, or would one typically have a unified use-case with the PlaybackPCM Value being part of the SectionDevice? 3) For the HiFi case, there are a couple of volume controls in serial; i.e. "Digital Playback Volume" affects both speaker/headphone output whereas "Speaker Volume" and "Headphone Volume" affect just one of the devices. I've defined a PlaybackVolume Value for the overall verb and the same Value for each device. Are applications able to query the Value separately for verb and devices, or will using the same name cause conflicts? Is there any kind of (de-facto) standard for how to represent this situation? Thanks very much for any feedback. ------------------ tegra-wm8903-seaboard.conf: # UCM configuration for Tegra WM8903 driver for Seaboard SectionUseCase."Record" { File "record" Comment "Record" } SectionUseCase."HiFi" { File "hifi" Comment "Play HiFi quality Music" } SectionUseCase."HiFi HDMI" { File "hdmi" Comment "Play HiFi quality Music over HDMI" } SectionDefaults [ cdev "hw:0" cset "name='Headphone Switch' off,off" cset "name='Left Output Mixer DACL Switch' on" cset "name='Left Output Mixer DACR Switch' off" cset "name='Left Output Mixer Left Bypass Switch' off" cset "name='Left Output Mixer Right Bypass Switch' off" cset "name='Right Output Mixer DACL Switch' off" cset "name='Right Output Mixer DACR Switch' on" cset "name='Right Output Mixer Left Bypass Switch' off" cset "name='Right Output Mixer Right Bypass Switch' off" cset "name='Int Spk Switch' off" cset "name='Speaker Switch' off,off" cset "name='Left Speaker Mixer DACL Switch' on" cset "name='Left Speaker Mixer DACR Switch' off" cset "name='Left Speaker Mixer Left Bypass Switch' off" cset "name='Left Speaker Mixer Right Bypass Switch' off" cset "name='Right Speaker Mixer DACL Switch' off" cset "name='Right Speaker Mixer DACR Switch' on" cset "name='Right Speaker Mixer Left Bypass Switch' off" cset "name='Right Speaker Mixer Right Bypass Switch' off" cset "name='Left Input PGA Switch' off" cset "name='Right Input PGA Switch' off" cset "name='Right Inverting Input Mux' 'IN2R'" ] ------------------ hifi: SectionVerb { Value { PlaybackPCM "hw:0,0" PlaybackVolume "Digital Playback Volume" } } SectionDevice."Speakers".0 { Comment "Internal Speakers" EnableSequence [ cdev "hw:0" cset "name='Speaker Switch' on,on" cset "name='Int Spk Switch' on" ] DisableSequence [ cdev "hw:0" cset "name='Speaker Switch' off,off" cset "name='Int Spk Switch' off" ] Value { PlaybackVolume "Speaker Volume" PlaybackSwitch "Speaker Switch" } } SectionDevice."Headphones".0 { Comment "Headphone Jack" EnableSequence [ cdev "hw:0" cset "name='Headphone Switch' on,on" ] DisableSequence [ cdev "hw:0" cset "name='Headphone Switch' off,off" ] Value { PlaybackVolume "Headphone Volume" PlaybackSwitch "Headphone Switch" } } ------------------ hifi-hdmi: SectionVerb { Value { PlaybackPCM "hw:0,1" } } SectionDevice."HDMI".0 { Comment "HDMI" } ------------------ record: SectionVerb { Value { CapturePCM "hw:0,0" CaptureVolume "Digital Capture Volume" } } SectionDevice."AMIC".0 { Comment "Analog Microphone Jack" EnableSequence [ cdev "hw:0" cset "name='ADC Input' 'ADC'" cset "name='Left Capture Mux' 'Right'" cset "name='Right Capture Mux' 'Right'" ] Value { CaptureVolume "Right Input PGA Volume" } } SectionDevice."DMIC".0 { Comment "Internal Digital Microphone" EnableSequence [ cdev "hw:0" cset "name='ADC Input' 'DMIC'" cset "name='Left Capture Mux' 'Left'" cset "name='Right Capture Mux' 'Right'" ] Value { } } -- nvpublic

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[alsa-devel] [PATCH] ASoC: spdif-dit: Add missing MODULE_*
by Stephen Warren 21 May '11

21 May '11

MODULE_ALIAS is required so that the module will auto-load based on a platform_device registration in the board file. While we're at it, add some other MODULE_*. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/spdif_transciever.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c index 4c32b54..6a1a7e7 100644 --- a/sound/soc/codecs/spdif_transciever.c +++ b/sound/soc/codecs/spdif_transciever.c @@ -21,7 +21,7 @@ #include <sound/pcm.h> #include <sound/initval.h> -MODULE_LICENSE("GPL"); +#define DRV_NAME "spdif-dit" #define STUB_RATES SNDRV_PCM_RATE_8000_96000 #define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE @@ -56,7 +56,7 @@ static struct platform_driver spdif_dit_driver = { .probe = spdif_dit_probe, .remove = spdif_dit_remove, .driver = { - .name = "spdif-dit", + .name = DRV_NAME, .owner = THIS_MODULE, }, }; @@ -74,3 +74,7 @@ static void __exit dit_exit(void) module_init(dit_modinit); module_exit(dit_exit); +MODULE_AUTHOR("Steve Chen <schen(a)mvista.com>"); +MODULE_DESCRIPTION("SPDIF dummy codec driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); -- 1.7.0.4

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[alsa-devel] [PATCH v1 RESEND] audio: tlv320aic26: fix PLL register configuration
by Michael Williamson 21 May '11

21 May '11

The current PLL configuration code for the tlc320aic26 codec appears to assume a hardcoded system clock of 12 MHz. Use the clock value provided by the DAI_OPS API for the calculation. Tested using a MityDSP-L138 platform providing a 24.576 MHz clock. Signed-off-by: Michael Williamson <michael.williamson(a)criticallink.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- This got bounced by the alsa-devel list (I wasn't on list). I'm not sure whose tree this needs to go through, but given the lack of response I'm guessing alsa-devel. If I'm missing a list, any advice would be appreciated. Changes from V0: - Added some parens for clarification - Added some comments to describe what the computations were attempting to do sound/soc/codecs/tlv320aic26.c | 14 +++++++++++--- 1 files changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index e2a7608..7859bdc 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -161,10 +161,18 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL; } - /* Configure PLL */ + /** + * Configure PLL + * fsref = (mclk * PLLM) / 2048 + * where PLLM = J.DDDD (DDDD register ranges from 0 to 9999, decimal) + */ pval = 1; - jval = (fsref == 44100) ? 7 : 8; - dval = (fsref == 44100) ? 5264 : 1920; + /* compute J portion of multiplier */ + jval = fsref / (aic26->mclk / 2048); + /* compute fractional DDDD component of multiplier */ + dval = fsref - (jval * (aic26->mclk / 2048)); + dval = (10000 * dval) / (aic26->mclk / 2048); + dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval); qval = 0; reg = 0x8000 | qval << 11 | pval << 8 | jval << 2; aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg); -- 1.7.0.4

2 1

[alsa-devel] [PATCHv2] ASoC: core: Don't set "(null)" as a driver name
by Jarkko Nikula 21 May '11

21 May '11

Commit 22de71b ("ASoC: core - allow ASoC more flexible machine name") writes "(null)" to driver name string in struct snd_card if card->driver_name is NULL. This causes segmentation faults with some user space ALSA utilities like aplay and arecord. Fix this by using the card->name if no driver name is specified. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/soc-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5968745..bb7cd58 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1930,7 +1930,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), "%s", card->long_name ? card->long_name : card->name); snprintf(card->snd_card->driver, sizeof(card->snd_card->driver), - "%s", card->driver_name); + "%s", card->driver_name ? card->driver_name : card->name); if (card->late_probe) { ret = card->late_probe(card); -- 1.7.4.4

3 2

Re: [alsa-devel] [PATCH] ALSA: hda: delay EAPD de-assert for alex system
by Takashi Iwai 21 May '11

21 May '11

At Fri, 20 May 2011 16:12:51 -0700, Todd Broch wrote: > > With power savings (CONFIG_SND_HDA_POWER_SAVE) enabled the EAPD GPIO > de-asserts prior to proper biasing of the speaker inputs into the > external amplifier causing a 'pop'. Delaying EAPD by 25ms provides > necessary setup time to avoid the pop. > > Signed-off-by: Todd Broch <tbroch(a)chromium.org> Could you test the latest sound git tree (topic/hda or master branch) if not done yet? There was a similar fix (adding some delay in power-up/down path), thanks, Takashi > --- > sound/pci/hda/patch_realtek.c | 5 +++++ > 1 files changed, 5 insertions(+), 0 deletions(-) > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index c82979a..5b70dc1 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -1262,6 +1262,11 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) > case 0x10ec0269: > case 0x10ec0270: > case 0x10ec0272: > + /* delay de-assert of eapd to allow biasing of amp > + inputs to settle avoiding an audible 'pop' > + */ > + if (codec->subsystem_id == 0x144dc0a7) > + msleep(25); > case 0x10ec0660: > case 0x10ec0662: > case 0x10ec0663: > -- > 1.7.3.1 >

1 0

[alsa-devel] Should we really use POS_FIX_VIACOMBO for AMD/ATI chipsets?
by David Henningsson 20 May '11

20 May '11

I'm far from sure, but I could be on to something... Looking at the code for azx_via_get_position, it seems that for recording streams, it accesses a register at position VIA_IN_STREAM0_FIFO_SIZE_OFFSET = 0x90. Looking at the datasheet for SB700, that register is not even present. This does not seem right to me. Could it be that we should use POS_FIX_LPIB by default for AMD/ATI chipsets instead? It seems more reasonable to me. In addition, I looked through some of the commits for quirking to POS_FIX_LPIB and the four I checked were all ATI chipsets. So; I've tried adding: options snd-hda-intel position_fix=1 to /etc/modprobe.d/alsa-base.conf ...to my machine with the infamous [1002:4383] controller (and rebooted), and the few times I've tested, recording worked successfully with pulseaudio. So could you please follow this up with testing on your machines to see if we have actually managed to solve this long-standing bug? The only annoying thing is that I didn't realise this earlier :-/ -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 5

[alsa-devel] [HEADSUP] Please submit something for the Audio track at LPC!
by Lennart Poettering 20 May '11

20 May '11

Heya, The Linux Plumbers Conference 2011 is coming soon. Mark Brown and I are running the Audio track at LPC, and we are looking for more proposals! So, please consider submitting something if you haven't done so yet. We are looking for all kinds of technical talks covering everything audio plumbing related: audio drivers, audio APIs, sound servers, pro audio, consumer audio. If you can propose something audio related -- like talks on media controller routing, on audio for ASOC/Embedded, submit something! If you care for low-latency audio, submit something. If you care about the Linux audio stack in general, submit something. LPC is probably the most relevant technical conference on the general Linux platform, so be sure that if you want your project, your work, your ideas to be heard then this is the right forum for everything related to the Linux stack. And the Audio track covers everything in our Audio Stack, regardless whether it is pro or consumer audio. So, submit! submit! submit! (And contrary to what the web site might suggest regarding deadlines we'll still consider your submission, if you post it quickly!) http://www.linuxplumbersconf.org/2011/ Lennart -- Lennart Poettering - Red Hat, Inc.

1 0

[alsa-devel] [PATCH] ASoC: core: Don't set "(null)" as a driver name
by Jarkko Nikula 20 May '11

20 May '11

Commit 22de71b ("ASoC: core - allow ASoC more flexible machine name") writes "(null)" to driver name string in struct snd_card if card->driver_name is NULL. This causes segmentation faults with some user space ALSA utilities like aplay and arecord. Fix this by using null terminated string instead as this was the case before. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/soc-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5968745..ab05f42 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1930,7 +1930,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), "%s", card->long_name ? card->long_name : card->name); snprintf(card->snd_card->driver, sizeof(card->snd_card->driver), - "%s", card->driver_name); + "%s", card->driver_name ? card->driver_name : ""); if (card->late_probe) { ret = card->late_probe(card); -- 1.7.4.4

2 2

[alsa-devel] [PATCH 1/2 v3] ASoC: soc-cache: Block based rbtree compression
by Dimitris Papastamos 20 May '11

20 May '11

This patch prepares the ground for the actual rbtree optimization patch which will save a pointer to the last accessed rbnode that was used in either the read() or write() functions. Each rbnode manages a variable length block of registers. There can be no two nodes with overlapping blocks. Each block has a base register and a currently top register, all the other registers, if any, lie in between these two and in ascending order. The reasoning behind the construction of this rbtree is simple. In the snd_soc_rbtree_cache_init() function, we iterate over the register defaults provided by the driver. For each register value that is non-zero we insert it in the rbtree. In order to determine in which rbnode we need to add the register, we first look if there is another register already added that is adjacent to the one we are about to add. If that is the case we append it in that rbnode block, otherwise we create a new rbnode with a single register in its block and add it to the tree. In the next patch, where a cached rbnode is used by both the write() and the read() functions, we also check if the register we are about to add is in the cached rbnode (the least recently accessed one) and if so we append it in that rbnode block. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 232 +++++++++++++++++++++++++++++++++++++------------ 1 files changed, 177 insertions(+), 55 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 06b7b81..b72f591 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -483,31 +483,85 @@ static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx, } struct snd_soc_rbtree_node { - struct rb_node node; - unsigned int reg; - unsigned int value; - unsigned int defval; + struct rb_node node; /* the actual rbtree node holding this block */ + unsigned int base_reg; /* base register handled by this block */ + unsigned int word_size; /* number of bytes needed to represent the register index */ + void *block; /* block of adjacent registers */ + unsigned int blklen; /* number of registers available in the block */ } __attribute__ ((packed)); struct snd_soc_rbtree_ctx { struct rb_root root; }; +static inline void snd_soc_rbtree_get_base_top_reg( + struct snd_soc_rbtree_node *rbnode, + unsigned int *base, unsigned int *top) +{ + *base = rbnode->base_reg; + *top = rbnode->base_reg + rbnode->blklen - 1; +} + +static unsigned int snd_soc_rbtree_get_register( + struct snd_soc_rbtree_node *rbnode, unsigned int idx) +{ + unsigned int val; + + switch (rbnode->word_size) { + case 1: { + u8 *p = rbnode->block; + val = p[idx]; + return val; + } + case 2: { + u16 *p = rbnode->block; + val = p[idx]; + return val; + } + default: + BUG(); + break; + } + return -1; +} + +static void snd_soc_rbtree_set_register(struct snd_soc_rbtree_node *rbnode, + unsigned int idx, unsigned int val) +{ + switch (rbnode->word_size) { + case 1: { + u8 *p = rbnode->block; + p[idx] = val; + break; + } + case 2: { + u16 *p = rbnode->block; + p[idx] = val; + break; + } + default: + BUG(); + break; + } +} + static struct snd_soc_rbtree_node *snd_soc_rbtree_lookup( struct rb_root *root, unsigned int reg) { struct rb_node *node; struct snd_soc_rbtree_node *rbnode; + unsigned int base_reg, top_reg; node = root->rb_node; while (node) { rbnode = container_of(node, struct snd_soc_rbtree_node, node); - if (rbnode->reg < reg) - node = node->rb_left; - else if (rbnode->reg > reg) - node = node->rb_right; - else + snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg); + if (reg >= base_reg && reg <= top_reg) return rbnode; + else if (reg > top_reg) + node = node->rb_right; + else if (reg < base_reg) + node = node->rb_left; } return NULL; @@ -518,19 +572,28 @@ static int snd_soc_rbtree_insert(struct rb_root *root, { struct rb_node **new, *parent; struct snd_soc_rbtree_node *rbnode_tmp; + unsigned int base_reg_tmp, top_reg_tmp; + unsigned int base_reg; parent = NULL; new = &root->rb_node; while (*new) { rbnode_tmp = container_of(*new, struct snd_soc_rbtree_node, node); + /* base and top registers of the current rbnode */ + snd_soc_rbtree_get_base_top_reg(rbnode_tmp, &base_reg_tmp, + &top_reg_tmp); + /* base register of the rbnode to be added */ + base_reg = rbnode->base_reg; parent = *new; - if (rbnode_tmp->reg < rbnode->reg) - new = &((*new)->rb_left); - else if (rbnode_tmp->reg > rbnode->reg) - new = &((*new)->rb_right); - else + /* if this register has already been inserted, just return */ + if (base_reg >= base_reg_tmp && + base_reg <= top_reg_tmp) return 0; + else if (base_reg > top_reg_tmp) + new = &((*new)->rb_right); + else if (base_reg < base_reg_tmp) + new = &((*new)->rb_left); } /* insert the node into the rbtree */ @@ -545,57 +608,120 @@ static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec) struct snd_soc_rbtree_ctx *rbtree_ctx; struct rb_node *node; struct snd_soc_rbtree_node *rbnode; + unsigned int regtmp; unsigned int val; int ret; + int i; rbtree_ctx = codec->reg_cache; for (node = rb_first(&rbtree_ctx->root); node; node = rb_next(node)) { rbnode = rb_entry(node, struct snd_soc_rbtree_node, node); - if (rbnode->value == rbnode->defval) - continue; - WARN_ON(codec->writable_register && - codec->writable_register(codec, rbnode->reg)); - ret = snd_soc_cache_read(codec, rbnode->reg, &val); - if (ret) - return ret; - codec->cache_bypass = 1; - ret = snd_soc_write(codec, rbnode->reg, val); - codec->cache_bypass = 0; - if (ret) - return ret; - dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", - rbnode->reg, val); + for (i = 0; i < rbnode->blklen; ++i) { + regtmp = rbnode->base_reg + i; + WARN_ON(codec->writable_register && + codec->writable_register(codec, regtmp)); + val = snd_soc_rbtree_get_register(rbnode, i); + codec->cache_bypass = 1; + ret = snd_soc_write(codec, regtmp, val); + codec->cache_bypass = 0; + if (ret) + return ret; + dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", + regtmp, val); + } } return 0; } +static int snd_soc_rbtree_insert_to_block(struct snd_soc_rbtree_node *rbnode, + unsigned int pos, unsigned int reg, + unsigned int value) +{ + u8 *blk; + + blk = krealloc(rbnode->block, + (rbnode->blklen + 1) * rbnode->word_size, GFP_KERNEL); + if (!blk) + return -ENOMEM; + + /* insert the register value in the correct place in the rbnode block */ + memmove(blk + (pos + 1) * rbnode->word_size, + blk + pos * rbnode->word_size, + (rbnode->blklen - pos) * rbnode->word_size); + + /* update the rbnode block, its size and the base register */ + rbnode->block = blk; + rbnode->blklen++; + if (!pos) + rbnode->base_reg = reg; + + snd_soc_rbtree_set_register(rbnode, pos, value); + return 0; +} + static int snd_soc_rbtree_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { struct snd_soc_rbtree_ctx *rbtree_ctx; - struct snd_soc_rbtree_node *rbnode; + struct snd_soc_rbtree_node *rbnode, *rbnode_tmp; + struct rb_node *node; + unsigned int val; + unsigned int reg_tmp; + unsigned int pos; + int i; + int ret; rbtree_ctx = codec->reg_cache; rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg); if (rbnode) { - if (rbnode->value == value) + reg_tmp = reg - rbnode->base_reg; + val = snd_soc_rbtree_get_register(rbnode, reg_tmp); + if (val == value) return 0; - rbnode->value = value; + snd_soc_rbtree_set_register(rbnode, reg_tmp, value); } else { /* bail out early, no need to create the rbnode yet */ if (!value) return 0; - /* - * for uninitialized registers whose value is changed - * from the default zero, create an rbnode and insert - * it into the tree. + /* look for an adjacent register to the one we are about to add */ + for (node = rb_first(&rbtree_ctx->root); node; + node = rb_next(node)) { + rbnode_tmp = rb_entry(node, struct snd_soc_rbtree_node, node); + for (i = 0; i < rbnode_tmp->blklen; ++i) { + reg_tmp = rbnode_tmp->base_reg + i; + if (abs(reg_tmp - reg) != 1) + continue; + /* decide where in the block to place our register */ + if (reg_tmp + 1 == reg) + pos = i + 1; + else + pos = i; + ret = snd_soc_rbtree_insert_to_block(rbnode_tmp, pos, + reg, value); + if (ret) + return ret; + return 0; + } + } + /* we did not manage to find a place to insert it in an existing + * block so create a new rbnode with a single register in its block. + * This block will get populated further if any other adjacent + * registers get modified in the future. */ rbnode = kzalloc(sizeof *rbnode, GFP_KERNEL); if (!rbnode) return -ENOMEM; - rbnode->reg = reg; - rbnode->value = value; + rbnode->blklen = 1; + rbnode->base_reg = reg; + rbnode->word_size = codec->driver->reg_word_size; + rbnode->block = kmalloc(rbnode->blklen * rbnode->word_size, + GFP_KERNEL); + if (!rbnode->block) { + kfree(rbnode); + return -ENOMEM; + } + snd_soc_rbtree_set_register(rbnode, 0, value); snd_soc_rbtree_insert(&rbtree_ctx->root, rbnode); } @@ -607,11 +733,13 @@ static int snd_soc_rbtree_cache_read(struct snd_soc_codec *codec, { struct snd_soc_rbtree_ctx *rbtree_ctx; struct snd_soc_rbtree_node *rbnode; + unsigned int reg_tmp; rbtree_ctx = codec->reg_cache; rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg); if (rbnode) { - *value = rbnode->value; + reg_tmp = reg - rbnode->base_reg; + *value = snd_soc_rbtree_get_register(rbnode, reg_tmp); } else { /* uninitialized registers default to 0 */ *value = 0; @@ -637,6 +765,7 @@ static int snd_soc_rbtree_cache_exit(struct snd_soc_codec *codec) rbtree_node = rb_entry(next, struct snd_soc_rbtree_node, node); next = rb_next(&rbtree_node->node); rb_erase(&rbtree_node->node, &rbtree_ctx->root); + kfree(rbtree_node->block); kfree(rbtree_node); } @@ -649,10 +778,9 @@ static int snd_soc_rbtree_cache_exit(struct snd_soc_codec *codec) static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec) { - struct snd_soc_rbtree_node *rbtree_node; struct snd_soc_rbtree_ctx *rbtree_ctx; - unsigned int val; unsigned int word_size; + unsigned int val; int i; int ret; @@ -666,28 +794,22 @@ static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec) if (!codec->reg_def_copy) return 0; - /* - * populate the rbtree with the initialized registers. All other - * registers will be inserted when they are first modified. - */ word_size = codec->driver->reg_word_size; for (i = 0; i < codec->driver->reg_cache_size; ++i) { - val = snd_soc_get_cache_val(codec->reg_def_copy, i, word_size); + val = snd_soc_get_cache_val(codec->reg_def_copy, i, + word_size); if (!val) continue; - rbtree_node = kzalloc(sizeof *rbtree_node, GFP_KERNEL); - if (!rbtree_node) { - ret = -ENOMEM; - snd_soc_cache_exit(codec); - break; - } - rbtree_node->reg = i; - rbtree_node->value = val; - rbtree_node->defval = val; - snd_soc_rbtree_insert(&rbtree_ctx->root, rbtree_node); + ret = snd_soc_rbtree_cache_write(codec, i, val); + if (ret) + goto err; } return 0; + +err: + snd_soc_cache_exit(codec); + return ret; } #ifdef CONFIG_SND_SOC_CACHE_LZO -- 1.7.5.1

3 3

[alsa-devel] [PATCH] ASoC: Tegra: Fix compile when debugfs not enabled
by Stephen Warren 20 May '11

20 May '11

The prototype of the inline dummy version of tegra_i2s_debug_add was not consistent with the real version. Reported-by: Rhyland-Klein < rklein(a)nvidia.com> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra_i2s.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 4f5e2c9..6b817e2 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -114,7 +114,7 @@ static void tegra_i2s_debug_remove(struct tegra_i2s *i2s) debugfs_remove(i2s->debug); } #else -static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s) +static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id) { } -- 1.7.0.4

2 1

[alsa-devel] [GIT PULL] ASoC updates for 2.6.40
by Liam Girdwood 20 May '11

20 May '11

Hi Takashi, Final updates for 2.6.40 Liam --- The following changes since commit 1209842af4db98ffd7364ec9cf7d1a59293aa74c: Merge branch 'for-2.6.40' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2011-05-10 09:24:50 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.40 Liam Girdwood (2): ASoC: twl6040 - fix LINEGAIN volume control MAINTAINERS: ASoC and Regulator email address change. Peter Ujfalusi (7): ASoC: tlv320dac33: Update e-mail address ASoC: tpa6130a2: Update e-mail address ASoC: omap-mcbsp: Update e-mail address ASoC: omap-pcm: Update e-mail address ASoC: RX51: Update e-mail address MAINTAINERS: Update e-mail address for asoc/twl4030 ASoC: omap-pcm: Period wakeup disabling on OMAP2+ Sanjeev Premi (1): ASoC: omap-mcbsp: Remove restrictive checks for cpu type MAINTAINERS | 6 +++--- include/sound/tlv320dac33-plat.h | 2 +- include/sound/tpa6130a2-plat.h | 2 +- sound/soc/codecs/tlv320dac33.c | 4 ++-- sound/soc/codecs/tlv320dac33.h | 2 +- sound/soc/codecs/tpa6130a2.c | 4 ++-- sound/soc/codecs/tpa6130a2.h | 2 +- sound/soc/codecs/twl6040.c | 6 +++--- sound/soc/omap/omap-mcbsp.c | 6 +++--- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/omap-pcm.c | 7 ++++--- sound/soc/omap/omap-pcm.h | 2 +- sound/soc/omap/rx51.c | 2 +- 13 files changed, 24 insertions(+), 23 deletions(-)

2 1

Re: [alsa-devel] [PATCH 2/2] davinci: da850evm: enable mcasp ping-pong bufs and evq's
by Liam Girdwood 20 May '11

20 May '11

On 20/05/11 08:08, Nori, Sekhar wrote: > Hi Mark, > > On Fri, May 20, 2011 at 02:44:06, Mark Brown wrote: >> On Wed, May 18, 2011 at 09:27:46AM -0400, Ben Gardiner wrote: >>> The davinci-mcasp driver will copy the platform data specified sram sizes and >>> eventq's. >>> >>> Set the event queues and sram sizes for da850. 8K SRAM buffers are selected >>> because it is the minimum that resulted in the same period size when testing >>> 48KHz S16_LE stereo. Event queues 0 and 1 are assigned to match those >>> specified by Troy Kisky in his introduction of ping-pong buffers for dm644x. >>> >>> Signed-off-by: Ben Gardiner <bengardiner(a)nanometrics.ca> >> >> Davinci folks, are you OK with this? > > No, please don't merge this just yet. As Ben mentioned in his cover > letter, the default SRAM pool on DA850 is not suitable for Audio > transfers. This patch depends on SRAM not being available for audio > usage in spite of the platform asking for it. This is liable to break. > > The SRAM allocation on DaVinci and OMAP is under a bit of flux and > once that is sorted out, this patch should be safe to merge. So, > this probably needs to wait for 2.6.41. > Ok. Sekhar, Ben, please resend with Acks when the DaVinci depedencies have been merged. Thanks Liam

1 0

[alsa-devel] [PATCH] hda - add Intel Panther Point HDMI codec id
by Wu Fengguang 20 May '11

20 May '11

Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) --- sound-2.6.orig/sound/pci/hda/patch_hdmi.c 2011-05-20 14:29:50.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_hdmi.c 2011-05-20 14:31:23.000000000 +0800 @@ -1659,6 +1659,7 @@ static struct hda_codec_preset snd_hda_p { .id = 0x80862803, .name = "Eaglelake HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862804, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862805, .name = "CougarPoint HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi }, { .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi }, {} /* terminator */ }; @@ -1704,6 +1705,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862802" MODULE_ALIAS("snd-hda-codec-id:80862803"); MODULE_ALIAS("snd-hda-codec-id:80862804"); MODULE_ALIAS("snd-hda-codec-id:80862805"); +MODULE_ALIAS("snd-hda-codec-id:80862806"); MODULE_ALIAS("snd-hda-codec-id:808629fb"); MODULE_LICENSE("GPL");

2 2

[alsa-devel] [RFC] Remove Alchemy OSS drivers?
by Ralf Baechle 20 May '11

20 May '11

OSS has very little if any useful life in it left for MIPS, at least from my perspective. So I propose to remove the Au1550 driver - there is an ALSA replacement for it available. Anybody against it? Are there still any remaining features or platforms that are only supported by the OSS driver? The OSS code is so backwards that I'd like to retire the possible remaining MIPS users and then make OSS unavailable in Kconfig for MIPS. Ralf

3 4

[alsa-devel] [PATCH] ctlparse: Respect softfloat configure option
by Alexander Stein 20 May '11

20 May '11

If we want softlfoat we can't use ceil which uses libm. Signed-off-by: Alexander Stein <a.stein(a)systec-electronic.com> --- This patch relates to http://mailman.alsa-project.org/pipermail/alsa-devel/2011-May/039693.html and following. src/control/ctlparse.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/src/control/ctlparse.c b/src/control/ctlparse.c index a929816..a16f96a 100644 --- a/src/control/ctlparse.c +++ b/src/control/ctlparse.c @@ -33,8 +33,17 @@ /* Function to convert from percentage to volume. val = percentage */ +#ifdef HAVE_SOFT_FLOAT +static inline long int convert_prange1(long val, long min, long max) +{ + long temp = val * (max - min); + return temp / 100 + min + ((temp % 100) == 0 ? 0 : 1); +} +#else + #define convert_prange1(val, min, max) \ ceil((val) * ((max) - (min)) * 0.01 + (min)) +#endif #define check_range(val, min, max) \ ((val < min) ? (min) : ((val > max) ? (max) : (val))) -- 1.7.3.4

2 1

[alsa-devel] [PATCH - ALSA - emu10k1] Add dB range to Bass and Treble controls for SB Live!
by Raymond Yau 20 May '11

20 May '11

As the "Wave", "Wave Surround" or "Front" Playback Volume must be changed to 70% (i.e. -12 dB) so that distortion won't occur when increase Bass and Treble from 50% to 100%, so the maximum gain in Bass and Treble are +12 dB

2 1

[alsa-devel] Debugging custom driver: BUG: pcmC0D0c:0 error
by Mike Crowe 20 May '11

20 May '11

Hi folks, I'm digging into a custom, embedded ALSA driver for some hardware we are developing. The basic functionality (interfacing with an soc/wm8990 codec) is working fine. We have a new library that we are adding which interfaces with the system via OSS. In preparation, I built the vrec/vplay functions, which also use OSS, and they work fine with the OSS Emulation layer of ALSA. I have enabled verbose printk and debug messages, and am seeing the following entries in the log file. Since I did not have these enabled before I started integrating, I'm not sure if this is new, or if they have been there for a while. Are these a problem, or can I ignore these? ALSA sound/core/pcm_lib.c:318: BUG: pcmC0D0c:0, pos = 230989824, buffer size = 12304, period size = 769 [ 2018.944000] ALSA sound/core/pcm_lib.c:318: BUG: pcmC0D0c:0, pos = 65536, buffer size = 65536, period size = 4096 [ 2020.304000] ALSA sound/core/pcm_lib.c:318: BUG: pcmC0D0c:0, pos = 65536, buffer size = 65536, period size = 4096 [ 2021.660000] ALSA sound/core/pcm_lib.c:318: BUG: pcmC0D0c:0, pos = 65536, buffer size = 65536, period size = 4096 TIA Mike

2 1

[alsa-devel] [PATCH] - ALSA - HDA - Remove "PCM Playback Volume" and "PCM Playback Switch" from Virtual Master of realtek codec
by Raymond Yau 20 May '11

20 May '11

Afer commit aa202455eec51699e44f658530728162cefa1307 , none of realtek codec has hardware volume control "PCM Playback Volume" and "PCM Playback Switch" As Virtual Master require all slave controls must have same number of step and dB range

2 1

[alsa-devel] [PATCH - hda / AD1988 ; 1/1] Compare pin type enum to the pin type and not the array index
by Adrian 20 May '11

20 May '11

Fixes bug#0005368

2 1

[alsa-devel] playback write error(DMA or IRQ trouble?)
by innoway＠mymelody.com 20 May '11

20 May '11

I develop an embedded system on the Linux board with Freescale i.MX27 (ARM processor). � I succeed on sound, which is transferred using Synchronous Serial Interface (SSI). It,s fine and then I try to do test of inserting or pulling out a USB memory stick in the USB host connector while sounding. � I get the following error message from snd_pcm_lib_write1() of <linux src>/sound /core/pcm_lib.c after waiting for a while (about 10sec). � �� playback� write error (DMA or IRQ trouble?) � and then from snd_pcm_drain() of <linux src>/sound/core/pcm_native.c � �� playback drain error (DMA or IRQ trouble?) � (No sound during this period) and then it is sounding again. I want to prevent the sound from stopping even if something happens suddenly such as the USB memory stick is plugged in or out. I try to fix the above programs, but I don't succeed. Can someone give me the clue? � [13:21:38]:/proc# cat interrupts �� CPU0 �12:�� 3456�� -� mxc_i2c �16:�� 141 � � � �� -� CSPI_IRQ �18:�� 952�� -� mxcintuart �26:�� 1051255�� -� MXC Timer Tick �28:�� 29446�� -� PCMCIA IRQ, ide0 �50:�� 33�� -� fec �54:�� 793�� -� ehci_hcd:usb1 �56:�� 1�� -� arc_udc �61:�� 0�� -� LCDC 117:�� 0�� -� m66291_udc 147:�� 0�� -� PMIC_IRQ 157:�� 337179�� -� SMSC_LAN911x_ISR 176:�� 0�� -� fec(MII) [13:35:03]:/proc# cat dma dma channel� 0: unused dma channel� 1: unused dma channel� 2: unused dma channel� 3: unused dma channel� 4: unused dma channel� 5: unused dma channel� 6: unused dma channel� 7: unused dma channel� 8: unused dma channel� 9: unused dma channel 10: unused dma channel 11: unused dma channel 12: unused dma channel 13: unused dma channel 14: unused dma channel 15: unused � Thank you Masanori Inoue

2 1

[alsa-devel] [RFC] Remove Alchemy OSS drivers?
by Ralf Baechle 20 May '11

20 May '11

OSS has very little if any useful life in it left for MIPS, at least from my perspective. So I propose to remove the Au1550 driver - there is an ALSA replacement for it available. Anybody against it? Are there still any remaining features or platforms that are only supported by the OSS driver? The OSS code is so backwards that I'd like to retire the possible remaining MIPS users and then make OSS unavailable in Kconfig for MIPS. Ralf

1 0

[alsa-devel] ASoC updates for 2.6.40
by Mark Brown 20 May '11

20 May '11

The following changes since commit 0f3c6af92100193cf9a86bdd0ee35e6da3e0c56e: Merge branch 'for-2.6.39' into for-2.6.40 (2011-05-10 15:58:17 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.40 Ben Gardiner (1): ASoC: davinci-mcasp: enable ping-pong SRAM buffers Dmitry Artamonow (2): ASoC: Asahi Kasei AK4641 codec driver ASoC: add iPAQ hx4700 machine driver Jarkko Nikula (1): ASoC: Fix wrong data type access in a few codec drivers Jin Park (3): ASoC: codecs: max98088: Fixed invalid register definitions in mixer controls ASoC: codecs: max98088: Moved the EX Limiter Mode from dapm widget to control ASoC: codecs: max98088: Added digital mute function in DAI1 and DAI2 Liam Girdwood (1): ASoC: core - allow ASoC more flexible machine name Mark Brown (10): ASoC: Fix NULL vs. 0 warning in SSM2602 ASoC: Don't squash 16x8 registers down to 8 bits ASoC: Remove byte swap in 4x12 SPI write ASoC: Use spi_write() for SPI writes ASoC: Disable WM8994/58 microphone detection over suspend ASoC: Reintroduce do_spi_write() ASoC: Ensure output PGA is enabled for line outputs in wm_hubs ASoC: Add some missing volume update bit sets for wm_hubs devices ASoC: Skip noop reconfiguration of WM8958 DSP2 algorithms ASoC: Don't restart an already running WM8958 DSP2 Misael Lopez Cruz (1): ASoC: DMIC codec - Add input widget Randy Dunlap (1): ASoC: fix wm8958-dsp2 printk format warnings Stephen Boyd (1): ASoC: Silence DEBUG_STRICT_USER_COPY_CHECKS=y warning Taylor Hutt (1): ASoC: Max98095: Move existing NULL check before pointer dereference. include/sound/ak4641.h | 26 ++ include/sound/soc.h | 2 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4641.c | 664 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4641.h | 47 +++ sound/soc/codecs/dmic.c | 26 ++- sound/soc/codecs/max98088.c | 62 +++- sound/soc/codecs/max98088.h | 13 + sound/soc/codecs/max98095.c | 16 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 3 +- sound/soc/codecs/wm8903.c | 3 +- sound/soc/codecs/wm8958-dsp2.c | 29 ++- sound/soc/codecs/wm8993.c | 3 +- sound/soc/codecs/wm8994.c | 31 ++- sound/soc/codecs/wm8995.c | 4 +- sound/soc/codecs/wm_hubs.c | 24 +- sound/soc/davinci/davinci-mcasp.c | 2 + sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/hx4700.c | 255 ++++++++++++++ sound/soc/soc-cache.c | 140 +------- sound/soc/soc-core.c | 8 +- 24 files changed, 1199 insertions(+), 178 deletions(-) create mode 100644 include/sound/ak4641.h create mode 100644 sound/soc/codecs/ak4641.c create mode 100644 sound/soc/codecs/ak4641.h create mode 100644 sound/soc/pxa/hx4700.c

1 0

[alsa-devel] [PATCH 0/2] ASoC: davinci: ping-pong buffers for mcasp on da850evm
by Ben Gardiner 20 May '11

20 May '11

davinci-i2s enables the use of ping-pong buffers by copying the sram sizes specified by platform data into the dma params used by davinci-pcm. This patch series first implements that same behaviour in davinic-mcasp and then specified sram sizes and eventq's for da850evm. To achieve the use of ping-pong buffers at runtime this patch depends also on the conversion of da850's SRAM allocator to the "Share RAM" region from the "ARM Local RAM" region -- originally proposed by Subhasish Ghosh[1] and recently reposted by myself [2]. This is because the "ARM Local RAM" region currently used by the da850 SRAM allocator is not addressable by the EDMA. The resulting behaviour, when ping-pong buffers are used by davinci-pcm, is that playback produces silence and capture acquires silence. However, there is no change in behaviour for da850evm build with the in-tree defconfig since suspend allocates SRAM which prevents a successful allocation of 8K by davinci-pcm and thus the behaviour falls back to the usual scheme. In the case of da850evm builds where CONFIG_SUSPEND is not set, playback will produce silence and capture will acquire silence until such time as the patches to change the da850 SRAM allocator to the "Shared RAM" region are merged. [1] http://article.gmane.org/gmane.linux.kernel/1098928 [2] http://article.gmane.org/gmane.linux.ports.arm.kernel/117261 Ben Gardiner (2): ASoC: davinci-mcasp: enable ping-pong SRAM buffers davinci: da850evm: enable mcasp ping-pong buffers and eventq's arch/arm/mach-davinci/board-da850-evm.c | 5 ++++- sound/soc/davinci/davinci-mcasp.c | 2 ++ 2 files changed, 6 insertions(+), 1 deletions(-) -- 1.7.4.1

3 5

[alsa-devel] [PATCH RFC] radio-sf16fmr2: convert to generic TEA575x interface
by Ondrej Zary 19 May '11

19 May '11

Hello, the SF16-FMR2 radio card is basically a TEA5757 tuner connected to ISA bus but the driver currently uses its own implementation. I converted the driver to use generic tea575x-tuner implementation (see patch below) and it works fine. But the tea575x-tuner module is located in /sound/i2c/other and the corresponding Kconfig in /sound/pci. Should it be moved to /drivers/media/radio or better leave it as is? The card can optionally be equipped with PT2254A volume control (equivalent of TC9154AP) so I need to extend tea575x-tuner to allow adding additional V4L2 controls (and add one if the chip is present). Is this OK or should be something like v4l2_subdev used somehow? (My card does not have the PT2254A chip so that code is untested.) --- linux-2.6.39-rc2-/include/sound/tea575x-tuner.h 2011-05-13 19:39:23.000000000 +0200 +++ linux-2.6.39-rc2/include/sound/tea575x-tuner.h 2011-05-14 00:14:23.000000000 +0200 @@ -54,6 +54,11 @@ struct snd_tea575x { void *private_data; u8 card[32]; u8 bus_info[32]; + /* card driver can provide additional controls (e.g. volume) */ + struct v4l2_queryctrl *extra_ctrls; /* array of controls */ + int num_extra_ctrls; /* number of additional controls */ + int (*g_extra_ctrl)(struct file *file, void *fh, struct v4l2_control *a); + int (*s_extra_ctrl)(struct file *file, void *fh, struct v4l2_control *a); }; int snd_tea575x_init(struct snd_tea575x *tea); --- linux-2.6.39-rc2-/sound/i2c/other/tea575x-tuner.c 2011-05-13 19:39:23.000000000 +0200 +++ linux-2.6.39-rc2/sound/i2c/other/tea575x-tuner.c 2011-05-13 21:35:57.000000000 +0200 @@ -269,6 +269,7 @@ static int vidioc_s_audio(struct file *f static int vidioc_queryctrl(struct file *file, void *priv, struct v4l2_queryctrl *qc) { + struct snd_tea575x *tea = video_drvdata(file); int i; for (i = 0; i < ARRAY_SIZE(radio_qctrl); i++) { @@ -278,6 +279,13 @@ static int vidioc_queryctrl(struct file return 0; } } + for (i = 0; i < tea->num_extra_ctrls; i++) { + if (qc->id && qc->id == tea->extra_ctrls[i].id) { + memcpy(qc, &(tea->extra_ctrls[i]), + sizeof(*qc)); + return 0; + } + } return -EINVAL; } @@ -291,6 +299,10 @@ static int vidioc_g_ctrl(struct file *fi ctrl->value = tea->mute; return 0; } + + if (tea->g_extra_ctrl) + return tea->g_extra_ctrl(file, priv, ctrl); + return -EINVAL; } @@ -307,6 +319,10 @@ static int vidioc_s_ctrl(struct file *fi } return 0; } + + if (tea->s_extra_ctrl) + return tea->s_extra_ctrl(file, priv, ctrl); + return -EINVAL; } --- linux-2.6.39-rc2-/drivers/media/radio/radio-sf16fmr2.c 2011-04-06 03:30:43.000000000 +0200 +++ linux-2.6.39-rc2/drivers/media/radio/radio-sf16fmr2.c 2011-05-14 00:15:05.000000000 +0200 @@ -1,441 +1,206 @@ -/* SF16FMR2 radio driver for Linux radio support - * heavily based on fmi driver... - * (c) 2000-2002 Ziglio Frediano, freddy77(a)angelfire.com +/* SF16-FMR2 radio driver for Linux + * Copyright (c) 2011 Ondrej Zary * - * Notes on the hardware - * - * Frequency control is done digitally -- ie out(port,encodefreq(95.8)); - * No volume control - only mute/unmute - you have to use line volume - * - * For read stereo/mono you must wait 0.1 sec after set frequency and - * card unmuted so I set frequency on unmute - * Signal handling seem to work only on autoscanning (not implemented) - * - * Converted to V4L2 API by Mauro Carvalho Chehab <mchehab(a)infradead.org> + * Original driver was (c) 2000-2002 Ziglio Frediano, freddy77(a)angelfire.com + * but almost nothing remained here after conversion to generic TEA575x + * interface */ +#include <linux/delay.h> #include <linux/module.h> /* Modules */ #include <linux/init.h> /* Initdata */ #include <linux/ioport.h> /* request_region */ -#include <linux/delay.h> /* udelay */ -#include <linux/videodev2.h> /* kernel radio structs */ -#include <linux/mutex.h> -#include <linux/version.h> /* for KERNEL_VERSION MACRO */ #include <linux/io.h> /* outb, outb_p */ -#include <media/v4l2-device.h> -#include <media/v4l2-ioctl.h> +#include <sound/tea575x-tuner.h> -MODULE_AUTHOR("Ziglio Frediano, freddy77(a)angelfire.com"); -MODULE_DESCRIPTION("A driver for the SF16FMR2 radio."); +MODULE_AUTHOR("Ondrej Zary"); +MODULE_DESCRIPTION("MediaForte SF16-FMR2 FM radio card driver"); MODULE_LICENSE("GPL"); -static int io = 0x384; -static int radio_nr = -1; - -module_param(io, int, 0); -MODULE_PARM_DESC(io, "I/O address of the SF16FMR2 card (should be 0x384, if do not work try 0x284)"); -module_param(radio_nr, int, 0); - -#define RADIO_VERSION KERNEL_VERSION(0,0,2) - -#define AUD_VOL_INDEX 1 - -#undef DEBUG -//#define DEBUG 1 - -#ifdef DEBUG -# define debug_print(s) printk s -#else -# define debug_print(s) -#endif - -/* this should be static vars for module size */ -struct fmr2 -{ - struct v4l2_device v4l2_dev; - struct video_device vdev; - struct mutex lock; +struct fmr2 { int io; - int curvol; /* 0-15 */ - int mute; - int stereo; /* card is producing stereo audio */ - unsigned long curfreq; /* freq in kHz */ - int card_type; + int volume; + struct snd_tea575x tea; }; +/* the port is hardwired so no need to support multiple cards */ +#define FMR2_PORT 0x384 static struct fmr2 fmr2_card; -/* hw precision is 12.5 kHz - * It is only useful to give freq in interval of 200 (=0.0125Mhz), - * other bits will be truncated - */ -#define RSF16_ENCODE(x) ((x) / 200 + 856) -#define RSF16_MINFREQ (87 * 16000) -#define RSF16_MAXFREQ (108 * 16000) - -static inline void wait(int n, int io) -{ - for (; n; --n) - inb(io); -} - -static void outbits(int bits, unsigned int data, int nWait, int io) -{ - int bit; - - for (; --bits >= 0;) { - bit = (data >> bits) & 1; - outb(bit, io); - wait(nWait, io); - outb(bit | 2, io); - wait(nWait, io); - outb(bit, io); - wait(nWait, io); - } -} - -static inline void fmr2_mute(int io) -{ - outb(0x00, io); - wait(4, io); -} - -static inline void fmr2_unmute(int io) -{ - outb(0x04, io); - wait(4, io); -} - -static inline int fmr2_stereo_mode(int io) -{ - int n = inb(io); - - outb(6, io); - inb(io); - n = ((n >> 3) & 1) ^ 1; - debug_print((KERN_DEBUG "stereo: %d\n", n)); - return n; -} - -static int fmr2_product_info(struct fmr2 *dev) -{ - int n = inb(dev->io); - - n &= 0xC1; - if (n == 0) { - /* this should support volume set */ - dev->card_type = 12; - return 0; - } - /* not volume (mine is 11) */ - dev->card_type = (n == 128) ? 11 : 0; - return n; -} - -static inline int fmr2_getsigstr(struct fmr2 *dev) -{ - /* !!! works only if scanning freq */ - int res = 0xffff; - - outb(5, dev->io); - wait(4, dev->io); - if (!(inb(dev->io) & 1)) - res = 0; - debug_print((KERN_DEBUG "signal: %d\n", res)); - return res; -} +/* TEA575x tuner pins */ +#define STR_DATA (1 << 0) +#define STR_CLK (1 << 1) +#define STR_WREN (1 << 2) +#define STR_MOST (1 << 3) +/* PT2254A/TC9154A volume control pins */ +#define PT_ST (1 << 4) +#define PT_CK (1 << 5) +#define PT_DATA (1 << 6) +/* volume control presence pin */ +#define FMR2_HASVOL (1 << 7) -/* set frequency and unmute card */ -static int fmr2_setfreq(struct fmr2 *dev) +static void fmr2_tea575x_set_pins(struct snd_tea575x *tea, u8 pins) { - unsigned long freq = dev->curfreq; + struct fmr2 *fmr2 = tea->private_data; + u8 bits = 0; - fmr2_mute(dev->io); + bits |= (pins & TEA575X_DATA) ? STR_DATA : 0; + bits |= (pins & TEA575X_CLK) ? STR_CLK : 0; + /* WRITE_ENABLE is inverted, DATA must be high during read */ + bits |= (pins & TEA575X_WREN) ? 0 : STR_WREN | STR_DATA; - /* 0x42 for mono output - * 0x102 forward scanning - * 0x182 scansione avanti - */ - outbits(9, 0x2, 3, dev->io); - outbits(16, RSF16_ENCODE(freq), 2, dev->io); - - fmr2_unmute(dev->io); - - /* wait 0.11 sec */ - msleep(110); - - /* NOTE if mute this stop radio - you must set freq on unmute */ - dev->stereo = fmr2_stereo_mode(dev->io); - return 0; -} - -/* !!! not tested, in my card this does't work !!! */ -static int fmr2_setvolume(struct fmr2 *dev) -{ - int vol[16] = { 0x021, 0x084, 0x090, 0x104, - 0x110, 0x204, 0x210, 0x402, - 0x404, 0x408, 0x410, 0x801, - 0x802, 0x804, 0x808, 0x810 }; - int i, a; - int n = vol[dev->curvol & 0x0f]; - - if (dev->card_type != 11) - return 1; - - for (i = 12; --i >= 0; ) { - a = ((n >> i) & 1) << 6; /* if (a==0) a = 0; else a = 0x40; */ - outb(a | 4, dev->io); - wait(4, dev->io); - outb(a | 0x24, dev->io); - wait(4, dev->io); - outb(a | 4, dev->io); - wait(4, dev->io); - } - for (i = 6; --i >= 0; ) { - a = ((0x18 >> i) & 1) << 6; - outb(a | 4, dev->io); - wait(4, dev->io); - outb(a | 0x24, dev->io); - wait(4, dev->io); - outb(a | 4, dev->io); - wait(4, dev->io); - } - wait(4, dev->io); - outb(0x14, dev->io); - return 0; -} - -static int vidioc_querycap(struct file *file, void *priv, - struct v4l2_capability *v) -{ - strlcpy(v->driver, "radio-sf16fmr2", sizeof(v->driver)); - strlcpy(v->card, "SF16-FMR2 radio", sizeof(v->card)); - strlcpy(v->bus_info, "ISA", sizeof(v->bus_info)); - v->version = RADIO_VERSION; - v->capabilities = V4L2_CAP_TUNER | V4L2_CAP_RADIO; - return 0; + outb(bits, fmr2->io); } -static int vidioc_g_tuner(struct file *file, void *priv, - struct v4l2_tuner *v) +static u8 fmr2_tea575x_get_pins(struct snd_tea575x *tea) { - struct fmr2 *fmr2 = video_drvdata(file); - - if (v->index > 0) - return -EINVAL; + struct fmr2 *fmr2 = tea->private_data; + u8 bits = inb(fmr2->io); - strlcpy(v->name, "FM", sizeof(v->name)); - v->type = V4L2_TUNER_RADIO; - - v->rangelow = RSF16_MINFREQ; - v->rangehigh = RSF16_MAXFREQ; - v->rxsubchans = fmr2->stereo ? V4L2_TUNER_SUB_STEREO : - V4L2_TUNER_SUB_MONO; - v->capability = V4L2_TUNER_CAP_STEREO | V4L2_TUNER_CAP_LOW; - v->audmode = V4L2_TUNER_MODE_STEREO; - mutex_lock(&fmr2->lock); - v->signal = fmr2_getsigstr(fmr2); - mutex_unlock(&fmr2->lock); - return 0; + return (bits & STR_DATA) ? TEA575X_DATA : 0 | + (bits & STR_MOST) ? TEA575X_MOST : 0; } -static int vidioc_s_tuner(struct file *file, void *priv, - struct v4l2_tuner *v) +static void fmr2_tea575x_set_direction(struct snd_tea575x *tea, bool output) { - return v->index ? -EINVAL : 0; } -static int vidioc_s_frequency(struct file *file, void *priv, - struct v4l2_frequency *f) -{ - struct fmr2 *fmr2 = video_drvdata(file); - - if (f->tuner != 0 || f->type != V4L2_TUNER_RADIO) - return -EINVAL; - if (f->frequency < RSF16_MINFREQ || - f->frequency > RSF16_MAXFREQ) - return -EINVAL; - /* rounding in steps of 200 to match the freq - that will be used */ - fmr2->curfreq = (f->frequency / 200) * 200; - - /* set card freq (if not muted) */ - if (fmr2->curvol && !fmr2->mute) { - mutex_lock(&fmr2->lock); - fmr2_setfreq(fmr2); - mutex_unlock(&fmr2->lock); - } - return 0; -} - -static int vidioc_g_frequency(struct file *file, void *priv, - struct v4l2_frequency *f) -{ - struct fmr2 *fmr2 = video_drvdata(file); - - if (f->tuner != 0) - return -EINVAL; - f->type = V4L2_TUNER_RADIO; - f->frequency = fmr2->curfreq; - return 0; -} +static struct snd_tea575x_ops fmr2_tea_ops = { + .set_pins = fmr2_tea575x_set_pins, + .get_pins = fmr2_tea575x_get_pins, + .set_direction = fmr2_tea575x_set_direction, +}; -static int vidioc_queryctrl(struct file *file, void *priv, - struct v4l2_queryctrl *qc) -{ - struct fmr2 *fmr2 = video_drvdata(file); +/* TC9154A/PT2254A volume control */ - switch (qc->id) { - case V4L2_CID_AUDIO_MUTE: - return v4l2_ctrl_query_fill(qc, 0, 1, 1, 1); - case V4L2_CID_AUDIO_VOLUME: - /* Only card_type == 11 implements volume */ - if (fmr2->card_type == 11) - return v4l2_ctrl_query_fill(qc, 0, 15, 1, 0); - return v4l2_ctrl_query_fill(qc, 0, 1, 1, 0); +/* 18-bit shift register bit definitions */ +#define TC9154A_ATT_MAJ_0DB (1 << 0) +#define TC9154A_ATT_MAJ_10DB (1 << 1) +#define TC9154A_ATT_MAJ_20DB (1 << 2) +#define TC9154A_ATT_MAJ_30DB (1 << 3) +#define TC9154A_ATT_MAJ_40DB (1 << 4) +#define TC9154A_ATT_MAJ_50DB (1 << 5) +#define TC9154A_ATT_MAJ_60DB (1 << 6) + +#define TC9154A_ATT_MIN_0DB (1 << 7) +#define TC9154A_ATT_MIN_2DB (1 << 8) +#define TC9154A_ATT_MIN_4DB (1 << 9) +#define TC9154A_ATT_MIN_6DB (1 << 10) +#define TC9154A_ATT_MIN_8DB (1 << 11) +/* bit 12 is ignored */ +#define TC9154A_CHANNEL_LEFT (1 << 13) +#define TC9154A_CHANNEL_RIGHT (1 << 14) +/* bits 15, 16, 17 must be 0 */ + +#define TC9154A_ATT_MAJ(x) (1 << x) +#define TC9154A_ATT_MIN(x) (1 << (7 + x)) + +static void tc9154a_set_pins(struct fmr2 *fmr2, u8 pins) +{ + if (!fmr2->tea.mute) + pins |= STR_WREN; + + outb(pins, fmr2->io); +} + +static void tc9154a_set_volume(struct fmr2 *fmr2, int volume) +{ + int i; + u32 reg; + u8 bit; + + reg = TC9154A_ATT_MAJ(volume / 10) | TC9154A_ATT_MIN((volume % 10) / 2); + reg |= TC9154A_CHANNEL_LEFT | TC9154A_CHANNEL_RIGHT; + /* write 18-bit shift register, LSB first */ + for (i = 0; i < 18; i++) { + bit = reg & (1 << i) ? PT_DATA : 0; + tc9154a_set_pins(fmr2, bit); + udelay(5); + tc9154a_set_pins(fmr2, bit | PT_CK); + udelay(5); + tc9154a_set_pins(fmr2, bit); + } + + /* latch register data */ + udelay(5); + tc9154a_set_pins(fmr2, PT_ST); + udelay(5); + tc9154a_set_pins(fmr2, 0); +} + +static struct v4l2_queryctrl fmr2_volume_qctrl[] = { + { + .id = V4L2_CID_AUDIO_VOLUME, + .name = "Volume", + .minimum = -68, + .maximum = 0, + .step = 2, + .default_value = -68, + .type = V4L2_CTRL_TYPE_INTEGER, } - return -EINVAL; -} +}; -static int vidioc_g_ctrl(struct file *file, void *priv, +static int fmr2_g_extra_ctrl(struct file *file, void *priv, struct v4l2_control *ctrl) { - struct fmr2 *fmr2 = video_drvdata(file); + struct snd_tea575x *tea = video_drvdata(file); + struct fmr2 *fmr2 = tea->private_data; switch (ctrl->id) { - case V4L2_CID_AUDIO_MUTE: - ctrl->value = fmr2->mute; - return 0; case V4L2_CID_AUDIO_VOLUME: - ctrl->value = fmr2->curvol; + ctrl->value = fmr2->volume; return 0; } + return -EINVAL; } -static int vidioc_s_ctrl(struct file *file, void *priv, +static int fmr2_s_extra_ctrl(struct file *file, void *priv, struct v4l2_control *ctrl) { - struct fmr2 *fmr2 = video_drvdata(file); + struct snd_tea575x *tea = video_drvdata(file); + struct fmr2 *fmr2 = tea->private_data; switch (ctrl->id) { - case V4L2_CID_AUDIO_MUTE: - fmr2->mute = ctrl->value; - break; case V4L2_CID_AUDIO_VOLUME: - fmr2->curvol = ctrl->value; - break; - default: - return -EINVAL; + fmr2->volume = ctrl->value; + tc9154a_set_volume(fmr2, fmr2->volume); + return 0; } -#ifdef DEBUG - if (fmr2->curvol && !fmr2->mute) - printk(KERN_DEBUG "unmute\n"); - else - printk(KERN_DEBUG "mute\n"); -#endif - - mutex_lock(&fmr2->lock); - if (fmr2->curvol && !fmr2->mute) { - fmr2_setvolume(fmr2); - /* Set frequency and unmute card */ - fmr2_setfreq(fmr2); - } else - fmr2_mute(fmr2->io); - mutex_unlock(&fmr2->lock); - return 0; -} - -static int vidioc_g_input(struct file *filp, void *priv, unsigned int *i) -{ - *i = 0; - return 0; -} - -static int vidioc_s_input(struct file *filp, void *priv, unsigned int i) -{ - return i ? -EINVAL : 0; -} - -static int vidioc_g_audio(struct file *file, void *priv, - struct v4l2_audio *a) -{ - a->index = 0; - strlcpy(a->name, "Radio", sizeof(a->name)); - a->capability = V4L2_AUDCAP_STEREO; - return 0; -} - -static int vidioc_s_audio(struct file *file, void *priv, - struct v4l2_audio *a) -{ - return a->index ? -EINVAL : 0; + return -EINVAL; } -static const struct v4l2_file_operations fmr2_fops = { - .owner = THIS_MODULE, - .unlocked_ioctl = video_ioctl2, -}; - -static const struct v4l2_ioctl_ops fmr2_ioctl_ops = { - .vidioc_querycap = vidioc_querycap, - .vidioc_g_tuner = vidioc_g_tuner, - .vidioc_s_tuner = vidioc_s_tuner, - .vidioc_g_audio = vidioc_g_audio, - .vidioc_s_audio = vidioc_s_audio, - .vidioc_g_input = vidioc_g_input, - .vidioc_s_input = vidioc_s_input, - .vidioc_g_frequency = vidioc_g_frequency, - .vidioc_s_frequency = vidioc_s_frequency, - .vidioc_queryctrl = vidioc_queryctrl, - .vidioc_g_ctrl = vidioc_g_ctrl, - .vidioc_s_ctrl = vidioc_s_ctrl, -}; - static int __init fmr2_init(void) { struct fmr2 *fmr2 = &fmr2_card; - struct v4l2_device *v4l2_dev = &fmr2->v4l2_dev; - int res; - strlcpy(v4l2_dev->name, "sf16fmr2", sizeof(v4l2_dev->name)); - fmr2->io = io; - fmr2->stereo = 1; - mutex_init(&fmr2->lock); + fmr2->io = FMR2_PORT; - if (!request_region(fmr2->io, 2, "sf16fmr2")) { - v4l2_err(v4l2_dev, "request_region failed!\n"); + if (!request_region(fmr2->io, 2, "SF16-FMR2")) { + printk(KERN_ERR "I/O port 0x%x already in use\n", fmr2->io); return -EBUSY; } - res = v4l2_device_register(NULL, v4l2_dev); - if (res < 0) { + fmr2->tea.private_data = fmr2; + fmr2->tea.ops = &fmr2_tea_ops; + strlcpy(fmr2->tea.card, "SF16-FMR2", sizeof(fmr2->tea.card)); + strcpy(fmr2->tea.bus_info, "ISA"); + + if (snd_tea575x_init(&fmr2->tea)) { + printk(KERN_ERR "Unable to detect TEA575x tuner\n"); release_region(fmr2->io, 2); - v4l2_err(v4l2_dev, "Could not register v4l2_device\n"); - return res; + return -ENODEV; } - strlcpy(fmr2->vdev.name, v4l2_dev->name, sizeof(fmr2->vdev.name)); - fmr2->vdev.v4l2_dev = v4l2_dev; - fmr2->vdev.fops = &fmr2_fops; - fmr2->vdev.ioctl_ops = &fmr2_ioctl_ops; - fmr2->vdev.release = video_device_release_empty; - video_set_drvdata(&fmr2->vdev, fmr2); - - /* mute card - prevents noisy bootups */ - fmr2_mute(fmr2->io); - fmr2_product_info(fmr2); - - if (video_register_device(&fmr2->vdev, VFL_TYPE_RADIO, radio_nr) < 0) { - v4l2_device_unregister(v4l2_dev); - release_region(fmr2->io, 2); - return -EINVAL; + if (inb(fmr2->io) & FMR2_HASVOL) { + fmr2->tea.extra_ctrls = fmr2_volume_qctrl; + fmr2->tea.num_extra_ctrls = 1; + fmr2->tea.g_extra_ctrl = fmr2_g_extra_ctrl; + fmr2->tea.s_extra_ctrl = fmr2_s_extra_ctrl; } - v4l2_info(v4l2_dev, "SF16FMR2 radio card driver at 0x%x.\n", fmr2->io); - debug_print((KERN_DEBUG "card_type %d\n", fmr2->card_type)); + printk(KERN_INFO "SF16-FMR2 radio card at 0x%x.\n", fmr2->io); return 0; } @@ -443,22 +208,9 @@ static void __exit fmr2_exit(void) { struct fmr2 *fmr2 = &fmr2_card; - video_unregister_device(&fmr2->vdev); - v4l2_device_unregister(&fmr2->v4l2_dev); + snd_tea575x_exit(&fmr2->tea); release_region(fmr2->io, 2); } module_init(fmr2_init); module_exit(fmr2_exit); - -#ifndef MODULE - -static int __init fmr2_setup_io(char *str) -{ - get_option(&str, &io); - return 1; -} - -__setup("sf16fmr2=", fmr2_setup_io); - -#endif -- Ondrej Zary

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[alsa-devel] Better Fast Track Pro support (24-bit/96kHz)
by jack.schnippes＠arcor.de 19 May '11

19 May '11

Dear alsa-developers, Many thanks for your continuous work to make sound hardware work on Linux!! :) Right now the Fast Track Pro works for Linux, but not with 24bit/96khz support. There's a patch for kernel version 2.6.31.4 that works (see [1]), but it's kinda complicated to get it working (at least for me). I would love to see this patch integrated in future versions of ALSA, so that other Linux newcomers and audio enthusiasts can fully use this device out of the box. This is not just a personal plea, since there are over 180 responses on the topic at [1]. Thank you for reading this and thank you for all the contributions to ALSA and the Linux world. [1] http://www.joegiampaoli.com/blog/?p=462 With best regards, Jack Schnippes

1 0

[alsa-devel] [PATCH] intel8x0m: enable AMD8111 modem
by Dmitry Eremin-Solenikov 19 May '11

19 May '11

AMD 8111 southbridges contain a controller for MC'97 modem. Enable support for this controller in intel8x0m driver. Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> --- sound/pci/intel8x0m.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 27709f0..f3353b4 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -235,8 +235,8 @@ static DEFINE_PCI_DEVICE_TABLE(snd_intel8x0m_ids) = { { PCI_VDEVICE(NVIDIA, 0x0069), DEVICE_NFORCE }, /* NFORCE2 */ { PCI_VDEVICE(NVIDIA, 0x0089), DEVICE_NFORCE }, /* NFORCE2s */ { PCI_VDEVICE(NVIDIA, 0x00d9), DEVICE_NFORCE }, /* NFORCE3 */ + { PCI_VDEVICE(AMD, 0x746e), DEVICE_INTEL }, /* AMD8111 */ #if 0 - { PCI_VDEVICE(AMD, 0x746d), DEVICE_INTEL }, /* AMD8111 */ { PCI_VDEVICE(AL, 0x5455), DEVICE_ALI }, /* Ali5455 */ #endif { 0, } @@ -1261,9 +1261,9 @@ static struct shortname_table { { PCI_DEVICE_ID_NVIDIA_MCP2_MODEM, "NVidia nForce2" }, { PCI_DEVICE_ID_NVIDIA_MCP2S_MODEM, "NVidia nForce2s" }, { PCI_DEVICE_ID_NVIDIA_MCP3_MODEM, "NVidia nForce3" }, + { 0x746e, "AMD AMD8111" }, #if 0 { 0x5455, "ALi M5455" }, - { 0x746d, "AMD AMD8111" }, #endif { 0 }, }; -- 1.7.4.4

2 3

[alsa-devel] Is the latency call thread safe ?
by Werner Van Belle 19 May '11

19 May '11

Hello, Can anybody chime in and tell me whether the latency call for a dsp device is thread safe or not? If not, what would be the possible outcome due to race conditions ? wkr, -- http://werner.yellowcouch.org/

1 0

[alsa-devel] Application: Regd ALSA playback
by Thilaga 19 May '11

19 May '11

Hi, I am new to ALSA and trying to play a wav file through ALSA by referring the sample code. It is a stereo, 44100Hz, 16 bit little endian file. It is not properly playing the file. I suspect that the frames parameter might be wrong. Below is the source code. Could any one give pointers on this. long loops; int rc; int size; snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int val; int dir; snd_pcm_uframes_t frames; char *buffer; /* Open PCM device for playback. */ rc = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0); if (rc < 0) { fprintf(stderr, "unable to open pcm device: %s\n", snd_strerror(rc)); exit(1); } /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&params); /* Fill it in with default values. */ snd_pcm_hw_params_any(handle, params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Signed 16-bit little-endian format */ snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(handle, params, 2); /* 44100 bits/second sampling rate (CD quality) */ val = 44100; snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir); /* Set period size to 32 frames. */ frames = 44100; snd_pcm_hw_params_set_period_size_near(handle, params, &frames, &dir); /* Write the parameters to the driver */ rc = snd_pcm_hw_params(handle, params); if (rc < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(rc)); exit(1); } /* Use a buffer large enough to hold one period */ snd_pcm_hw_params_get_period_size(params, &frames, &dir); printf("get period_size: %d\n",frames); size = frames * 4 ; /* 2 bytes/sample, 2 channels */ printf("Size:%d\n",size); buffer = (char *) malloc(size); /* We want to loop for 2 seconds */ snd_pcm_hw_params_get_period_time(params, &val, &dir); printf("period time: %d\n",val); /* 2 seconds in microseconds divided by * period time */ loops = 3000000 / val; printf("loops: %d\n",loops); FILE *fp; int file; fp = fopen("ring-1.wav", "rb"); if (!fp) { printf("file open failure\n"); return NULL; } printf("file open success\n"); while (loops > 0) { loops--; //rc = read(0, buffer, size); rc = fread(buffer, sizeof(unsigned char), frames, fp); printf("bytes read: %d\n", rc); if (rc == 0) { printf("end of file on input\n"); break; } else if (rc != size) { fprintf(stderr, "short read: read %d bytes\n", rc); } rc = snd_pcm_writei(handle, buffer, frames); if (rc == -EPIPE) { /* EPIPE means underrun */ fprintf(stderr, "underrun occurred\n"); snd_pcm_prepare(handle); } else if (rc < 0) { fprintf(stderr, "error from writei: %s\n", snd_strerror(rc)); } else if (rc != (int)frames) { fprintf(stderr, "short write, write %d frames\n", rc); } } snd_pcm_drain(handle); snd_pcm_close(handle); free(buffer); fclose(fp); return 0; } Thanks & Regards, Thilaga

1 0

Re: [alsa-devel] Audio Video synchronization for data received from a HDMI receiver chip
by Hans Verkuil 19 May '11

19 May '11

>> > Hi Hans, >> > >> > I have another doubt regarding the framework choice for the entire >> > system that I have, especially the video part of the system. The >> overall >> > system is similar to the one depicted below: >> > >> > HDMI data --> HDMI receiver chip --> Video Port IP on SoC --> System >> DDR >> > >> > HDMI data is received from external world (from say a set-up box or >> dvd player), >> > which is fed to the HDMI receiver chip on-board and then parallel >> data lines feed >> > this data to a Video Port IP on the SoC which has a DMA master >> interface and >> > hence can push the data thus received directly on system DDR. >> > >> > Now, I can figure out that there will be two drivers required here: >> > # HDMI receiver chip driver (which is essentially a v4l2 subdev being >> controller via I2C) >> > # Video Port driver (which is a v4l2 bridge driver) >> > >> > Is my understanding correct? >> >> Yes. > > Thanks for clarifying this. > >> > Are there any HDMI receiver subdev driver and video bridge driver >> already available which I can >> > use for reference? >> >> Video bridge drivers are easier: examples are in >> drivers/media/video/s5p-fimc >> or in drivers/media/video/davinci. Note that you should use the new >> videobuf2 >> framework instead of the older videobuf framework. s5p-fimc is using >> vb2 already. >> but the vpif capture and display drivers in the davinci directory do >> not. > > I quickly had a look at the s5p-fimc and davinci approaches, but I found > that > the video bridge drivers supported in both the Samsung and TI SoCs, > support video post-processing operations whereas in our case the Video > Port > IP performs almost no additional processing and only passes the unpacked > RBG > raw data received from HDMI bus to system DDR via a DMA master interface. This is similar to the davinci vpif-capture/vpif-display drivers for the TI dm646x SoCs. Those videoports are also just a DMA engine without video processing. > So as such these are no format conversion operations(rgb-to-yuv or > vice-versa), > image resizing operations (cropping, scaling..) and image quality > operations > (filtering, distortion removal) available in the H/W block. > > What should be the correct choice in such a case? vpif-capture/display drivers are identical in that respect. Regards, Hans

1 0

[alsa-devel] ASoC "long name" generation, without enumerating all machine_is_xxx() calls
by Stephen Warren 19 May '11

19 May '11

Liam, I have a question about your recent "long name" change: commit 22de71ba03311cdc1063757c50a1488cb90a1fca ASoC: core - allow ASoC more flexible machine name The question is: How to generate the long name. The only example I could find of this was: + if (machine_is_omap_4430sdp()) + snd_soc_sdp4430.name = "SDP4430"; + else if (machine_is_omap4_panda()) + snd_soc_sdp4430.name = "Panda" (from http://patchwork.ozlabs.org/patch/67831/) But, I don't really want to call machine_is_xxx for every single machine the tegra-wm8903 driver can support; upstream there are 5 so far, and more similar derivatives downstream. That said, I suppose the code is already switching on machine_is_xxx to pick the correct dapm_route map, so I could just assign the name there, and fold multiple identical machines onto a single name. It looks like the kernel doesn't have any kind of machine_name() API; the code in e.g. arch/arm/kernel/setup.c that prints the machine name at boot stores it in a static that's only exposed through syss/similar file. Thanks for any hints! -- nvpublic

2 1

[PATCH] ALSA: HDA: Add jack detection for HDMI
by David Henningsson 19 May '11

19 May '11

Just as for headphones and microphone jacks, this patch adds reporting of HDMI jack status through the input layer. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.c | 2 ++ sound/pci/hda/patch_hdmi.c | 11 +++++++++++ 2 files changed, 13 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c63f376..8edd998 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5055,6 +5055,8 @@ static const char *get_jack_default_name(struct hda_codec *codec, hda_nid_t nid, return "Line-out"; case SND_JACK_HEADSET: return "Headset"; + case SND_JACK_VIDEOOUT: + return "HDMI/DP"; default: return "Misc"; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 6eb209d..7348296 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -33,6 +33,7 @@ #include <linux/slab.h> #include <linux/moduleparam.h> #include <sound/core.h> +#include <sound/jack.h> #include "hda_codec.h" #include "hda_local.h" @@ -720,6 +721,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) &spec->sink_eld[index]); /* TODO: do real things about ELD */ } + + snd_hda_input_jack_report(codec, tag); } static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) @@ -912,6 +915,7 @@ static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid, static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) { struct hdmi_spec *spec = codec->spec; + int err; if (spec->num_pins >= MAX_HDMI_PINS) { snd_printk(KERN_WARNING @@ -919,6 +923,12 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) return -E2BIG; } + err = snd_hda_input_jack_add(codec, pin_nid, + SND_JACK_VIDEOOUT, NULL); + if (err < 0) + return err; + snd_hda_input_jack_report(codec, pin_nid); + hdmi_present_sense(codec, pin_nid, &spec->sink_eld[spec->num_pins]); spec->pin[spec->num_pins] = pin_nid; @@ -1120,6 +1130,7 @@ static void generic_hdmi_free(struct hda_codec *codec) for (i = 0; i < spec->num_pins; i++) snd_hda_eld_proc_free(codec, &spec->sink_eld[i]); + snd_hda_input_jack_free(codec); kfree(spec); } -- 1.7.4.1 --------------040806040506060705070005--

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Re: [alsa-devel] Audio Video synchronization for data received from a HDMI receiver chip
by Hans Verkuil 19 May '11

19 May '11

On Thursday, May 19, 2011 07:39:32 Bhupesh SHARMA wrote: > > On Wednesday, May 18, 2011 06:10:43 Bhupesh SHARMA wrote: > > > Hi, > > > > > > (adding alsa mailing list in cc) > > > > > > > On Thursday, May 12, 2011 18:59:33 Charlie X. Liu wrote: > > > > > Which HDMI receiver chip? > > > > > > > > Indeed, that's my question as well :-) > > > > > > We use Sil 9135 receiver chip which is provided by Silicon Image. > > > Please see details here: > > http://www.siliconimage.com/products/product.aspx?pid=109 > > > > > > > Anyway, this question comes up regularly. V4L2 provides timestamps > > for > > > > each > > > > frame, so that's no problem. But my understanding is that ALSA does > > not > > > > give > > > > you timestamps, so if there are processing delays between audio and > > > > video, then > > > > you have no way of knowing. The obvious solution is to talk to the > > ALSA > > > > people > > > > to see if some sort of timestamping is possible, but nobody has > > done > > > > that. > > > > > > I am aware of the time stamping feature provided by V4L2, but I am > > also > > > not sure whether the same feature is supported by ALSA. I have > > included > > > alsa-mailing list also in copy of this mail. Let's see if we can get > > > some sort of confirmation on this from them. > > > > > > > This is either because everyone that needs it hacks around it > > instead > > > > of trying > > > > to really solve it, or because it is never a problem in practice. > > > > > > What should be the proper solution according to you to solve this > > issue. > > > Do we require a Audio-Video Bridge kind of utility/mechanism? > > > > I don't believe so. All you need is reliable time stamping for your > > audio > > and video streams. That's enough for userspace to detect AV sync > > issues. > > > > Hi Hans, > > I have another doubt regarding the framework choice for the entire > system that I have, especially the video part of the system. The overall > system is similar to the one depicted below: > > HDMI data --> HDMI receiver chip --> Video Port IP on SoC --> System DDR > > HDMI data is received from external world (from say a set-up box or dvd player), > which is fed to the HDMI receiver chip on-board and then parallel data lines feed > this data to a Video Port IP on the SoC which has a DMA master interface and > hence can push the data thus received directly on system DDR. > > Now, I can figure out that there will be two drivers required here: > # HDMI receiver chip driver (which is essentially a v4l2 subdev being controller via I2C) > # Video Port driver (which is a v4l2 bridge driver) > > Is my understanding correct? Yes. > Are there any HDMI receiver subdev driver and video bridge driver already available which I can > use for reference? Video bridge drivers are easier: examples are in drivers/media/video/s5p-fimc or in drivers/media/video/davinci. Note that you should use the new videobuf2 framework instead of the older videobuf framework. s5p-fimc is using vb2 already. but the vpif capture and display drivers in the davinci directory do not. With regards to HDMI receivers: these are still under development. One example is here: http://git.linuxtv.org/hverkuil/cisco.git?a=shortlog;h=refs/heads/cobalt This tree contains a driver for the adv7604 HDMI/Graphics receiver. It is fairly simplistic at the moment, our internal driver is developed a lot further but I haven't had the chance yet to update the git tree with our latest code (Cisco is developing this driver). In addition the HDMI API for V4L2 is still under development. It requires some V4L2 core support to be merged first (control events) before we can continue with that. > > Also will the audio ALSA driver fit as a subdev driver in the entire system? No, although I have heard that the ALSA developers are looking at a subdev-like approach. There are several V4L drivers that support ALSA as a separate driver (cx18 for example). This is usually not a problem. Regards, Hans

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[alsa-devel] [PATCH] alsa-1.0.24 realtek hda codec patch
by Zbigniew Luszpinski 19 May '11

19 May '11

Hello, maybe you do not know but Realtek has in download section its own alsa 1.0.24 source package with additional code for Realtek codecs. Because this code has almost zero chance to reach end users because everyone uses vanilla alsa from alsa site or from kernel.org or alsa from distro package I thought it would be good to exctract it and put it here for inclusion in mainstream/kernel alsa. This way all users will benefit from it and code will not be wasted. The attached patch contains cleaned up diff between: alsa-driver-1.0.24.tar.bz2 from alsa-project site and LinuxPkg_5.16rc12.tar.bz2 dated 2011/4/28 from realtek.com.tw site. Its strange that Realtek cooks this in silence instead posting it to alsa for everyone. I hope Realtek will have nothing against merging these changes mainstream. This way Realtek hardware will be better supported by Linux distros. have a nice day, Zbigniew Luszpinski

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[alsa-devel] [PATCH] ALSA: sound, core, pcm_lib: fix xrun_log
by Ben Gardiner 19 May '11

19 May '11

The xrun_log function was augmented with the in_interrupt parameter whereas the empty macro definition used when xrun logging is disabled was not. Add a third parameter to the empty macro definition so as to not cause compiler errors when xrun logging (CONFIG_SND_PCM_XRUN_DEBUG) is disabled. Signed-off-by: Ben Gardiner <bengardiner(a)nanometrics.ca> --- This fixes the build error caused by ec08b14483de ("ALSA: sound, core, pcm_lib: xrun_log: log also in_interrupt") -- which was my patch. Sorry for the breakage. --- sound/core/pcm_lib.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index c8c8091..abfeff16 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -266,7 +266,7 @@ static void xrun_log_show(struct snd_pcm_substream *substream) #else /* ! CONFIG_SND_PCM_XRUN_DEBUG */ #define hw_ptr_error(substream, fmt, args...) do { } while (0) -#define xrun_log(substream, pos) do { } while (0) +#define xrun_log(substream, pos, in_interrupt) do { } while (0) #define xrun_log_show(substream) do { } while (0) #endif -- 1.7.4.1

2 1

[alsa-devel] Multiple codecs on one card (SOC)
by Caleb Crome 19 May '11

19 May '11

Hi again, I have a multi-codec board up and running, and need a little guidance. I've got 8 I2S codecs connected to a beagleboard. I have the first codec up and running, but I don't know how to add more codecs to the same alsa card. I have it set for multiple DAI links. Is that the correct way to have multiple codecs bound to the same card, and bound to eachother? The struct is below. When I do this, I have the problem, that the codec driver, which registers its mixer controls using macros like SOC_DOUBLE_R_TLV("PCM Playback Volume, LDAC_VOL, RDAC_VOL,...), they fail when the 2nd codec is getting probed. Am I supposed to use multiple snd_soc_dai_link declarations? How do I end up with a single 16-channel sound card built from multiple 2 channel devices? Do I use subdevices? Substreams? multiple dai_links? How do i instantiate them? Any pointers would be greatly appreciated. Thanks, -Caleb static struct snd_soc_dai_link puppy_dai[] = { { .name = "tlv320aic33-se-a", .stream_name = "tlv320aic33-se", .cpu_dai_name = "omap-mcbsp-dai.2", .platform_name = "omap-pcm-audio", .codec_dai_name = "tlv320aic3x-se-hifi", .codec_name = "tlv320aic3x-se-codec.2-0018", .ops = &puppy_ops, .symmetric_rates = 1, // I think this requires send and receive be the same rate. .init = puppy_dai_link_init, }, { .name = "tlv320aic33-se-b", .stream_name = "tlv320aic33-se", .cpu_dai_name = "omap-mcbsp-dai.2", .platform_name = "omap-pcm-audio", .codec_dai_name = "tlv320aic3x-se-hifi", .codec_name = "tlv320aic3x-se-codec.2-0019", .ops = &puppy_ops, .symmetric_rates = 1, // I think this requires send and receive be the same rate. .init = puppy_dai_link_init, }, };

1 0

Re: [alsa-devel] [PATCH] Max98095: Move existing NULL check before pointer dereference.
by Mark Brown 18 May '11

18 May '11

On Tue, May 17, 2011 at 06:03:54PM -0700, Taylor Hutt wrote: > Visual inspection shows that max98095_put_eq_enum() and > max98095_put_bq_enum() each have a possible NULL deref of 'pdata'. > > This change moves the NULL check above the use. Applied, thanks. Please try to keep your subject line consistent with the rest of the subsystem.

1 0

[alsa-devel] hdspm.c - confused about SNDRV_HDSPM_IOCTL_GET_[CONFIG|STATUS]
by Jörn Nettingsmeier 18 May '11

18 May '11

hi *! i'm trying to extend hdspconf to work with the rme hdspm madi card, and got confused over the snd_hdspm_hwdep_ioctl: apparently, the hdspm.c driver differentiates between settable controls (CONFIG) and non-settable status indicators (STATUS), unlike the earlier hdspm.c. however, usage seems a bit inconsistent: struct hdspm_config contains a member madi_sync_check, which is apparently never set. instead, struct hdspm_status card_specific.sync_madi is used, same for tco and sync in. but there is a wordclock_sync_check in struct hdspm_config which is in use. am i guessing right that the latter is somewhat deprecated and that i should be taking all status values from struct hdspm_status instead? thanks, jörn please cc: me, i'm not on alsa.devel. -- Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister für Veranstaltungstechnik (Bühne/Studio) Tonmeister (VDT) http://stackingdwarves.net

2 1

[alsa-devel] [PATCH] sound, core, pcm_lib: xrun_log: log also in_interrupt
by Ben Gardiner 18 May '11

18 May '11

When debugging pcm drivers I found the "period" or "hw" prefix printed by either XRUN_DEBUG_PERIODUPDATE or XRUN_DEBUG_PERIODUPDATE events, respectively to be very useful is observing the interplay between interrupt-context updates and syscall-context updates. Similarly, when debugging overruns with XRUN_DEBUG_LOG it is useful to see the context of the last 10 positions. Add an in_interrupt member to hwptr_log_entry which stores the value of the in_interrupt parameter of snd_pcm_update_hw_ptr0 when the log entry is created. Print a "[Q]" prefix when dumping the log entries if in_interrupt was true. Signed-off-by: Ben Gardiner <bengardiner(a)nanometrics.ca> --- sound/core/pcm_lib.c | 12 ++++++++---- 1 files changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 64449cb..c8c8091 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -189,6 +189,7 @@ static void xrun(struct snd_pcm_substream *substream) #define XRUN_LOG_CNT 10 struct hwptr_log_entry { + unsigned int in_interrupt; unsigned long jiffies; snd_pcm_uframes_t pos; snd_pcm_uframes_t period_size; @@ -204,7 +205,7 @@ struct snd_pcm_hwptr_log { }; static void xrun_log(struct snd_pcm_substream *substream, - snd_pcm_uframes_t pos) + snd_pcm_uframes_t pos, int in_interrupt) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_pcm_hwptr_log *log = runtime->hwptr_log; @@ -220,6 +221,7 @@ static void xrun_log(struct snd_pcm_substream *substream, return; } entry = &log->entries[log->idx]; + entry->in_interrupt = in_interrupt; entry->jiffies = jiffies; entry->pos = pos; entry->period_size = runtime->period_size; @@ -246,9 +248,11 @@ static void xrun_log_show(struct snd_pcm_substream *substream) entry = &log->entries[idx]; if (entry->period_size == 0) break; - snd_printd("hwptr log: %s: j=%lu, pos=%ld/%ld/%ld, " + snd_printd("hwptr log: %s: %sj=%lu, pos=%ld/%ld/%ld, " "hwptr=%ld/%ld\n", - name, entry->jiffies, (unsigned long)entry->pos, + name, entry->in_interrupt ? "[Q] " : "", + entry->jiffies, + (unsigned long)entry->pos, (unsigned long)entry->period_size, (unsigned long)entry->buffer_size, (unsigned long)entry->old_hw_ptr, @@ -326,7 +330,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, } pos -= pos % runtime->min_align; if (xrun_debug(substream, XRUN_DEBUG_LOG)) - xrun_log(substream, pos); + xrun_log(substream, pos, in_interrupt); hw_base = runtime->hw_ptr_base; new_hw_ptr = hw_base + pos; if (in_interrupt) { -- 1.7.4.1

2 1

Re: [alsa-devel] [PATCH] Max98095: Move existing NULL check before pointer dereference.
by Peter Hsiang 18 May '11

18 May '11

On Tue, May 17, 2011, Taylor Hutt wrote: > Visual inspection shows that max98095_put_eq_enum() and > max98095_put_bq_enum() each have a possible NULL deref of 'pdata'. > > This change moves the NULL check above the use. > > BUG=none > TEST=Visual inspection Mark, this patch looks good. Thank you Taylor.

3 3

[alsa-devel] Reported latency > buffersize and sometimes <0
by Werner Van Belle 18 May '11

18 May '11

Hello, I recently started to check whether the new alsa drivers still work with BpmDj (http://bpmdj.yellowcouch.org/). Thereby I encountered the problem that the reported latency might be larger than the buffersize itself. I guess this has to do with the asynchronity between asking the question and retrieving the answer (e.g: a thread switch might come in between and the actual latency is in that case bumped beyond the buffersize.. or so .. Does this reasoning make sense ? Can I assume that such weird values are indeed correct ? Another problem I sometimes encounter is that the returned latency is -1; a very annoying number since the stream was truly honoustly playing, so how can the latency be so low then ? With kind regards, Werner,- -- http://werner.yellowcouch.org/

1 0

[alsa-devel] Assorted usb-audio patches
by Daniel Mack 18 May '11

18 May '11

Hi, here are some patches I collected for the usb-audio driver. Most of them have been discussed on this list already, and all are rather small and shouldn't cause any problems. FWIW, they can also be pulled from git://github.com/zonque/linux-2.6.git usb-audio Daniel Daniel Mack (8): MAINTAINERS: Add entry for Native Instruments sound driver ALSA: usb-audio: make hwc_debug a noop in case HW_CONST_DEBUG is not set ALSA: usb-audio: include format.h in format.c ALSA: usb-audio: remove invalid extra mixers for Komplete Audio 6 ALSA: usb-audio: Add quirk for KORG PANDORA PX5D MIDI interface ALSA: usb-audio: add quirks for Roland GR-55 ALSA: usb-audio: assume valid clock ALSA: usb-audio: handle "Fast Track Ultra" with USB_DEVICE_VENDOR_SPEC() MAINTAINERS | 7 +++++++ sound/usb/clock.c | 11 ++++++++++- sound/usb/debug.h | 2 +- sound/usb/format.c | 1 + sound/usb/mixer_quirks.c | 17 ----------------- sound/usb/quirks-table.h | 43 ++++++++++++++++++++++++++++++++++++++++--- sound/usb/quirks.c | 2 +- 7 files changed, 60 insertions(+), 23 deletions(-)

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[PATCH - Basic mixer support for Fast Track Ultra 5/6]Try to create a mixer for quirked devices, too
by Felix Homann 18 May '11

18 May '11

Signed-off-by: Felix Homann <linuxaudio(a)showlabor.de> diff --git a/sound/usb/card.c b/sound/usb/card.c index 40722f8..52ad7b0 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -493,6 +493,11 @@ static void *snd_usb_audio_probe(struct usb_device *dev, */ if (!chip->ctrl_intf) chip->ctrl_intf = alts; + + /* try to create a mixer for quirked devices, too */ + if (snd_usb_create_mixer(chip, ifnum, ignore_ctl_error) < 0) { + goto __error; + } if (err > 0) { /* create normal USB audio interfaces */ -- 1.7.4.1 --------------000101040900080309070209--

1 0

[PATCH - Basic mixer support for Fast Track Ultra 3/6]Add mixer quirk for Fast Track Ultra devices
by Felix Homann 18 May '11

18 May '11

Signed-off-by: Felix Homann <linuxaudio(a)showlabor.de> diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ea95ce2..1c37792 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -542,6 +542,31 @@ static int add_control_to_empty(struct mixer_build *state, struct snd_kcontrol * return 0; } +/* + * This is almost the same as 'add_control_to_empty' but using a mixer instead of a mixer_build + * it simplifies building a mixer for devices without proper descriptors, + * e.g. the Fast Track Ultra devices + */ +int add_unique_control_to_empty_mixer(struct usb_mixer_interface *mixer, struct snd_kcontrol *kctl) +{ + struct usb_mixer_elem_info *cval = kctl->private_data; + int err; + + /* No duplicates! */ + if (snd_ctl_find_id(mixer->chip->card, &kctl->id)) + return 0; + + if ((err = snd_ctl_add(mixer->chip->card, kctl)) < 0) { + snd_printd(KERN_ERR "cannot add control (err = %d)\n", err); + return err; + } + + cval->elem_id = &kctl->id; + cval->next_id_elem = mixer->id_elems[cval->id]; + mixer->id_elems[cval->id] = cval; + + return 0; +} /* * get a terminal name string @@ -1331,6 +1356,57 @@ static void build_mixer_unit_ctl(struct mixer_build *state, add_control_to_empty(state, kctl); } +/* + * build a mixer unit control and give it a name + * + * this is essentially identical to build_mixer_unit_ctl but doesn't + * rely on proper descriptors to get a name for a control. + * At least, useful for Fast Track Ultra devices. + */ + +int build_named_mixer_unit_ctl(struct usb_mixer_interface *mixer, + int in_pin, int in_ch, int unitid, int val_type, unsigned int cmask, char* ctrl_name, int channels) +{ + struct usb_mixer_elem_info *cval; + struct snd_kcontrol *kctl; + int err; + + cval = kzalloc(sizeof(*cval), GFP_KERNEL); + if (! cval) + return -1; + + cval->mixer = mixer; + cval->id = unitid; + cval->control = in_ch + 1; /* based on 1 */ + cval->val_type = val_type; /* like USB_MIXER_S16 etc. */ + if (!((0 < channels) && (channels <= MAX_CHANNELS))) + return -1; + cval->channels = 1; + cval->cmask = cmask; + /* get min/max values */ + get_min_max(cval, 0); + + kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); + if (! kctl) { + snd_printk(KERN_ERR "cannot malloc kcontrol\n"); + kfree(cval); + return -1; + } + + kctl->tlv.c = mixer_vol_tlv; + kctl->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + kctl->private_free = usb_mixer_elem_free; + + sprintf(kctl->id.name, ctrl_name); + + if (err = add_unique_control_to_empty_mixer(mixer, kctl)) + return err; + + return 0; +} + /* * parse a mixer unit diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 86af922..6fe7bea 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -61,6 +61,9 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid); int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set); +int build_named_mixer_unit_ctl(struct usb_mixer_interface *mixer, + int in_pin, int in_ch, int unitid, int val_type, + unsigned int cmask, char* ctrl_name, int channels); void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer); int snd_usb_mixer_activate(struct usb_mixer_interface *mixer); diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 73dcc82..e87f359 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -500,6 +500,36 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, } } +/* + * Create mixer controls for Fast Track Ultra (8R) + */ +int snd_ftu_controls_create(struct usb_mixer_interface *mixer) +{ + char name[44]; + int out, in_ch, unitid, err; + unsigned int cmask; + int val_type = USB_MIXER_S16; + int channels = 1; /* mono channels */ + + unitid = 5; // FTU's unit id + for (out = 0; out < 8; ++out) { + for (in_ch = 0; in_ch < 16; ++in_ch) { + cmask = (1 << in_ch); + if (in_ch < 8) + sprintf(name, "AIn%d - Out%d Capture Volume", in_ch + 1, out + 1); + else if (in_ch >= 8) + sprintf(name, "DIn%d - Out%d Playback Volume", in_ch - 7, out + 1); + if ((err = build_named_mixer_unit_ctl(mixer, in_ch, out, unitid, + val_type, cmask, name, channels))) { + return err; + } + } + } + + return 0; +} + + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -537,6 +567,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) snd_nativeinstruments_ta10_mixers, ARRAY_SIZE(snd_nativeinstruments_ta10_mixers)); break; + case USB_ID(0x0763, 0x2080):/* M-Audio Fast Track Ultra */ + case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ + err = snd_ftu_controls_create(mixer); + break; } return err; -- 1.7.4.1 --------------000101040900080309070209 Content-Type: text/x-patch; name="0004-Try-to-create-a-mixer-for-quirked-devices-too.patch" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename*0="0004-Try-to-create-a-mixer-for-quirked-devices-too.patch"

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[PATCH - Basic mixer support for Fast Track Ultra 2/6]Move USB_MIXER_ enum to mixer.h This facilitates using it in mixer_quirks.c
by Felix Homann 18 May '11

18 May '11

Signed-off-by: Felix Homann <linuxaudio(a)showlabor.de> diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 5e47757..ea95ce2 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -86,16 +86,6 @@ struct mixer_build { const struct usbmix_selector_map *selector_map; }; -enum { - USB_MIXER_BOOLEAN, - USB_MIXER_INV_BOOLEAN, - USB_MIXER_S8, - USB_MIXER_U8, - USB_MIXER_S16, - USB_MIXER_U16, -}; - - /*E-mu 0202/0404/0204 eXtension Unit(XU) control*/ enum { USB_XU_CLOCK_RATE = 0xe301, diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 06d035f..86af922 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -1,6 +1,15 @@ #ifndef __USBMIXER_H #define __USBMIXER_H +enum { + USB_MIXER_BOOLEAN, + USB_MIXER_INV_BOOLEAN, + USB_MIXER_S8, + USB_MIXER_U8, + USB_MIXER_S16, + USB_MIXER_U16, +}; + struct usb_mixer_interface { struct snd_usb_audio *chip; struct list_head list; -- 1.7.4.1 --------------000101040900080309070209 Content-Type: text/x-patch; name="0003-Add-mixer-quirk-for-Fast-Track-Ultra-devices.patch" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename*0="0003-Add-mixer-quirk-for-Fast-Track-Ultra-devices.patch"

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[PATCH - Basic mixer support for Fast Track Ultra 1/6]Raise MAX_CHANNELS to 16 as needed by Fast TRack Ultra (8R)
by Felix Homann 18 May '11

18 May '11

Signed-off-by: Felix Homann <linuxaudio(a)showlabor.de> diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index b4a2c81..06d035f 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -24,7 +24,7 @@ struct usb_mixer_interface { u8 xonar_u1_status; }; -#define MAX_CHANNELS 10 /* max logical channels */ +#define MAX_CHANNELS 16 /* max logical channels */ struct usb_mixer_elem_info { struct usb_mixer_interface *mixer; -- 1.7.4.1 --------------000101040900080309070209 Content-Type: text/x-patch; name="0002-Move-USB_MIXER_-enum-to-mixer.h.patch" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename="0002-Move-USB_MIXER_-enum-to-mixer.h.patch"

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Re: [alsa-devel] Audio Video synchronization for data received from a HDMI receiver chip
by Hans Verkuil 18 May '11

18 May '11

On Wednesday, May 18, 2011 06:10:43 Bhupesh SHARMA wrote: > Hi, > > (adding alsa mailing list in cc) > > > On Thursday, May 12, 2011 18:59:33 Charlie X. Liu wrote: > > > Which HDMI receiver chip? > > > > Indeed, that's my question as well :-) > > We use Sil 9135 receiver chip which is provided by Silicon Image. > Please see details here: http://www.siliconimage.com/products/product.aspx?pid=109 > > > Anyway, this question comes up regularly. V4L2 provides timestamps for > > each > > frame, so that's no problem. But my understanding is that ALSA does not > > give > > you timestamps, so if there are processing delays between audio and > > video, then > > you have no way of knowing. The obvious solution is to talk to the ALSA > > people > > to see if some sort of timestamping is possible, but nobody has done > > that. > > I am aware of the time stamping feature provided by V4L2, but I am also > not sure whether the same feature is supported by ALSA. I have included > alsa-mailing list also in copy of this mail. Let's see if we can get > some sort of confirmation on this from them. > > > This is either because everyone that needs it hacks around it instead > > of trying > > to really solve it, or because it is never a problem in practice. > > What should be the proper solution according to you to solve this issue. > Do we require a Audio-Video Bridge kind of utility/mechanism? I don't believe so. All you need is reliable time stamping for your audio and video streams. That's enough for userspace to detect AV sync issues. Regards, Hans > > Regards, > Bhupesh > > > > > > > > > -----Original Message----- > > > From: linux-media-owner(a)vger.kernel.org > > > [mailto:linux-media-owner@vger.kernel.org] On Behalf Of Bhupesh > > SHARMA > > > Sent: Wednesday, May 11, 2011 10:49 PM > > > To: linux-media(a)vger.kernel.org > > > Cc: Laurent Pinchart; Guennadi Liakhovetski; Hans Verkuil > > > Subject: Audio Video synchronization for data received from a HDMI > > receiver > > > chip > > > > > > Hi Linux media folks, > > > > > > We are considering putting an advanced HDMI receiver chip on our SoC, > > > to allow reception of HDMI audio and video. The chip receives HDMI > > data > > > from a host like a set-up box or DVD player. It provides a video data > > > interface > > > and SPDIF/I2S audio data interface. > > > > > > We plan to support the HDMI video using the V4L2 framework and the > > HDMI > > > audio using ALSA framework. > > > > > > Now, what seems to be intriguing us is how the audio-video > > synchronization > > > will be maintained? Will a separate bridging entity required to > > ensure the > > > same > > > or whether this can be left upon a user space application like > > mplayer or > > > gstreamer. > > > > > > Also is there a existing interface between the V4L2 and ALSA > > frameworks and > > > the same > > > can be used in our design? > > > > > > Regards, > > > Bhupesh > > > ST Microelectronics > > > -- > > > To unsubscribe from this list: send the line "unsubscribe linux- > > media" in > > > the body of a message to majordomo(a)vger.kernel.org > > > More majordomo info at http://vger.kernel.org/majordomo-info.html > > > > > > > > -- > > To unsubscribe from this list: send the line "unsubscribe linux-media" > > in > > the body of a message to majordomo(a)vger.kernel.org > > More majordomo info at http://vger.kernel.org/majordomo-info.html > > > > > > > > -- > > regards > > Shiraz Hashim > >

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Re: [alsa-devel] ALC892 Support (Intel HDA) - HDMI
by Raymond Yau 18 May '11

18 May '11

2011/5/18 Jeremy Bush <jbush(a)sittercity.com>: > > On May 15, 2011, at 10:50 PM, Jeremy Bush wrote: > >> >> On May 15, 2011, at 10:45 PM, Raymond Yau wrote: >> >>> 2011/5/16 Jeremy Bush <contractfrombelow(a)gmail.com> >>> >>>>>> However, it seems that the ALSA drivers (even the ones from the realtek >>>>>> website) limit the output to 2 channels, 16bit, 88.2KHz. >>>>>> >>>>>> Here is my system information that you can see these results: >>>>>> >>>>>> >>>> http://www.alsa-project.org/db/?f=5b55a63fea6580dfcbf1a85c7c734f977df798f1 >>>>>> >>>>>> I see the ALC892 as being supported in the ALSA changelog, but I don't >>>> see >>>>>> any model options in Documentation/HD-Audio-Models.txt. I'm currently >>>> using >>>>>> "options snd-hda-intel model=6stack-digout" (as I have 6-jack with a >>>> SPDIF >>>>>> out on the motherboard) but I'm not sure that is actually doing >>>> anything. >>>>>> >>>>>> >>>>> As your codec info does not have "Side Playback Channel" and "Side >>>> Playback >>>>> Switch", it look like the model is used for 5stack motherboard which >>>> support >>>>> 5.1 >>>>> >>>>> Do you hear any noise since Valerio reported noise appear when using the >>>>> model=auto in >>>>> >>>>> http://thread.gmane.org/gmane.linux.alsa.devel/84653/focus=84724 >>>> >>>> I'm not actually trying to get analog working or the front-panel output. >>>> Only surround sound over HDMI. I have tried model=auto and didn't notice >>>> anything different than specifying the model. >>>> >>>>>> Is what I'm getting now the best I should expect over HDMI? I saw >>>> support >>>>>> in ALSA and bought the motherboard expecting it to be fully supported >>>> since >>>>>> the changelogs say support was added. >>>>> >>>>> Codec: Intel CougarPoint HDMI >>>>> >>>>> Address: 3 >>>>> AFG Function Id: 0x1 (unsol 0) >>>>> Vendor Id: 0x80862805 >>>>> Subsystem Id: 0x80862805 >>>>> Revision Id: 0x100000 >>>>> No Modem Function Group found >>>>> >>>>> Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital >>>>> >>>>> Control: name="IEC958 Playback Con Mask", index=1, device=0 >>>>> Control: name="IEC958 Playback Pro Mask", index=1, device=0 >>>>> Control: name="IEC958 Playback Default", index=1, device=0 >>>>> Control: name="IEC958 Playback Switch", index=1, device=0 >>>>> Device: name="HDMI 0", type="HDMI", device=3 >>>>> Converter: stream=8, channel=0 >>>>> Digital: Enabled >>>>> Digital category: 0x0 >>>>> PCM: >>>>> rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 >>>>> bits [0x1e]: 16 20 24 32 >>>>> formats [0x5]: PCM AC3 >>>>> Power states: D0 D3 EPSS >>>>> Power: setting=D0, actual=D0 >>>>> >>>>> Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP >>>>> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >>>>> Amp-Out vals: [0x00 0x00] >>>>> Pincap 0x09000094: OUT Detect HBR HDMI DP >>>>> Pin Default 0x18560020: [Jack] Digital Out at Int HDMI >>>>> Conn = Digital, Color = Unknown >>>>> DefAssociation = 0x2, Sequence = 0x0 >>>>> Pin-ctls: 0x40: OUT >>>>> Unsolicited: tag=06, enabled=1 >>>>> Power states: D0 D3 EPSS >>>>> Power: setting=D0, actual=D0 >>>>> Connection: 1 >>>>> 0x03 >>>> >>>> So this seems to say I should be getting 7.1 over the HDMI output. What do >>>> I need to change in my config to actually get it? I tried messing around >>>> with /sys/class/sound/hwC1D3/user_pin_configs but I'm not sure what to >>>> change. >>>> >>>> >>> Do your hdmi monitor support 7.1 ? >>> >>> [ 2.090473] HDMI: detected monitor at connection type HDMI >>> [ 2.090475] HDMI: available speakers: FL/FR >>> [ 2.090477] HDMI: supports coding type LPCM: channels = 2, rates = >>> 44100 48000 88200, bits = 16 >>> [ 2.090480] HDMI hot plug event: Pin=6 Presence_Detect=1 ELD_Valid=1 >> >> Yes, it's connected to an Onkyo TX-SR607 7.2-Channel receiver. > > > I've also upgraded to the latest nightly build of alsa, and still experience the same behavior. > > Jeremy you have to check the video driver since it is the video driver query hdmi monitor and pass back ELD to audio driver >>> [ 2.090475] HDMI: available speakers: FL/FR >>> [ 2.090477] HDMI: supports coding type LPCM: channels = 2, rates = >>> 44100 48000 88200, bits = 16

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[alsa-devel] snd jack report and unsolicited event ad1988
by Raymond Yau 18 May '11

18 May '11

2011/4/28 Takashi Iwai <tiwai(a)suse.de> > At Thu, 28 Apr 2011 21:10:23 +0800, > Raymond Yau wrote: > > > > > Still unable to enable the unsolicited event for jack sense even if I add > > the unsol_event and verb for the audio jacks at rear panel since I don't > > have the HDA compilant front audio panel > > > > SoundMax automatically popup immediately when jack is plug into the rear > > panel at other OS, so the hardware is capable of jack sense at rear panel > > > > AFG Function Id: 0x1 (unsol 0) > > > > Is there any trick to enable/debug the unsolicited event ? > > seem unsol event is disabled in the HDA controller > > No, there shouldn't be big differences. Otherwise it won't work on > any other machines. > > There might be other way to trigger the jack detection, e.g. GPIO > unsolicited event, depending on the codec chip. I don't remember how > is AD1988, though... > > > OK, after a few experiments, unsolicited events seem to work for those rear panel jacks 1) for the unsolicited event , different tags have to be assigned for six different jacks This mean that the driver need to define FRONT_MIC_EVENT, REAR_MIC_EVENT, LINE_IN_EVENT in addition to HP_EVENT Can snd_jack_report used to report the mic event of two pink jacks (front panel and rear panel) of the desktop to the user space program? 2) when the jack is plug in or out. , snd_hda_jack_detect() work with no_trigger_sense=1 to detect jack presence but the driver get several unsolicited responses with no_trigger_sense=0 3) snd_hda_pin_sense() need trigger sense to get the Impedance it usually get the value of 0x7FFF,FFFF and need to wait for a while to get the sense measurement (all1’s) indicates that a valid sense reading is not available, or the sense measurement is busy if it has been recently triggered

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[alsa-devel] [PATCH 0/3] ASoC: codecs: max98088: bug fixes and changes
by jinyoungp＠nvidia.com 18 May '11

18 May '11

From: Jin Park <jinyoungp(a)nvidia.com> Recently, I working on max98088 and changed something in max98088 asoc driver. The MAX98088 driver had some bug and needed modifies. The changes was confirmed by Peter Hsiang who author of max98088 driver and maxim engineer. Please review my changes. Thanks, Jin Park. Jin Park (3): ASoC: codecs: max98088: Fixed invalid register definitions in mixer controls ASoC: codecs: max98088: Moved the EX Limiter Mode from dapm widget to control ASoC: codecs: max98088: Added digital mute function in DAI1 and DAI2 sound/soc/codecs/max98088.c | 62 +++++++++++++++++++++++++++++++------------ sound/soc/codecs/max98088.h | 13 +++++++++ 2 files changed, 58 insertions(+), 17 deletions(-) ----------------------------------------------------------------------------------- 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] ALC892 Support (Intel HDA) - HDMI
by Jeremy Bush 18 May '11

18 May '11

Hello- I'm trying to get my Realtek ALC892 chip to work according to it's specifications over HDMI. The manufacturers site say it should be able to do 7.1+2 channels, 24bit audio and 192Khz sample rate. However, it seems that the ALSA drivers (even the ones from the realtek website) limit the output to 2 channels, 16bit, 88.2KHz. Here is my system information that you can see these results: http://www.alsa-project.org/db/?f=5b55a63fea6580dfcbf1a85c7c734f977df798f1 I see the ALC892 as being supported in the ALSA changelog, but I don't see any model options in Documentation/HD-Audio-Models.txt. I'm currently using "options snd-hda-intel model=6stack-digout" (as I have 6-jack with a SPDIF out on the motherboard) but I'm not sure that is actually doing anything. What's the *actual* support of this chipset? Is what I'm getting now the best I should expect over HDMI? I saw support in ALSA and bought the motherboard expecting it to be fully supported since the changelogs say support was added. I also tried with an ALC887 chipset as well, and got basically the same results. Any help getting this to work would be appreciated. Let me know if there's any other information required. Thank you- Jeremy

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[alsa-devel] Alsa Support for Korg Pandora Mini, Vox ToneLab ST
by Pawel Siemienski 17 May '11

17 May '11

Hi, I am Ubuntu Studio user and I use also ALSA for driving my audio for recording etc... I have tested two devices with ALSA: 1. Vox Tonelab ST - works with ALSA out of the box as USB audio interface with no need of modifications, provides complete functionalities needed for recording and playing audio 2. Korg Pandora Mini - is detected as audio interface, but when selecting it information that no controls are available is displayed, this device can not work as audio interface in fact with Alsa. Could you please advise what modifications in ALSA should be made to enable Korg Pandora Mini as audio interface with ALSA, is there any patch for this device? If you would need any information on Korg Pandora Mini please let know. Thanks Pawel Siemienski

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[PATCH] hda-emu: update jack.h to match latest kernel version
by David Henningsson 17 May '11

17 May '11

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- include/sound/jack.h | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 77247d0..a6a4f21 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -12,6 +12,9 @@ enum snd_jack_types { SND_JACK_MICROPHONE = 0x0002, SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE, SND_JACK_LINEOUT = 0x0004, + SND_JACK_MECHANICAL = 0x0008, /* If detected separately */ + SND_JACK_VIDEOOUT = 0x0010, + SND_JACK_AVOUT = SND_JACK_LINEOUT | SND_JACK_VIDEOOUT, }; struct snd_jack { -- 1.7.4.1 --------------030905020802080904050601--

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[alsa-devel] error: implicit declaration of function ‘fw_iso_resource_manage’
by Felix Homann 17 May '11

17 May '11

I've just tried building Alsa from git (alsa-driver/alsa-kernel). It won't build resulting in this error message: ~/workspace/Alsa/alsa-driver/firewire/../alsa-kernel/firewire/iso-resources.c: In function ‘fw_iso_resources_allocate’: ~/workspace/Alsa/alsa-driver/firewire/../alsa-kernel/firewire/iso-resources.c:135:2: error: implicit declaration of function ‘fw_iso_resource_manage’ Are there any patches waiting to resolve this issue? Regards, Felix

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[PATCH] ALSA: HDA: Add jack detection for HDMI
by David Henningsson 17 May '11

17 May '11

Just as for headphones and microphone jacks, this patch adds reporting of HDMI jack status through the input layer. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.c | 2 ++ sound/pci/hda/patch_hdmi.c | 15 +++++++++++++++ 2 files changed, 17 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c63f376..8edd998 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5055,6 +5055,8 @@ static const char *get_jack_default_name(struct hda_codec *codec, hda_nid_t nid, return "Line-out"; case SND_JACK_HEADSET: return "Headset"; + case SND_JACK_VIDEOOUT: + return "HDMI/DP"; default: return "Misc"; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 6eb209d..16c1505 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -33,6 +33,7 @@ #include <linux/slab.h> #include <linux/moduleparam.h> #include <sound/core.h> +#include <sound/jack.h> #include "hda_codec.h" #include "hda_local.h" @@ -720,6 +721,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) &spec->sink_eld[index]); /* TODO: do real things about ELD */ } + + snd_hda_input_jack_report(codec, tag); } static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) @@ -919,6 +922,17 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) return -E2BIG; } +#ifdef CONFIG_SND_HDA_INPUT_JACK + { + int err; + err = snd_hda_input_jack_add(codec, pin_nid, + SND_JACK_VIDEOOUT, NULL); + if (err < 0) + return err; + snd_hda_input_jack_report(codec, pin_nid); + } +#endif + hdmi_present_sense(codec, pin_nid, &spec->sink_eld[spec->num_pins]); spec->pin[spec->num_pins] = pin_nid; @@ -1120,6 +1134,7 @@ static void generic_hdmi_free(struct hda_codec *codec) for (i = 0; i < spec->num_pins; i++) snd_hda_eld_proc_free(codec, &spec->sink_eld[i]); + snd_hda_input_jack_free(codec); kfree(spec); } -- 1.7.4.1 --------------030905020802080904050601 Content-Type: text/x-patch; name="0001-hda-emu-update-jack.h-to-match-latest-kernel-version.patch" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename*0="0001-hda-emu-update-jack.h-to-match-latest-kernel-version.pa"; filename*1="tch"

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[alsa-devel] [PATCH v2 0/3] ASoC: add support for audio on iPaq hx4700
by Dmitry Artamonow 17 May '11

17 May '11

Ok, here's patches adding audio support for good ole iPAQ hx4700: ak4641 codec, hx4700 machine sound driver and some glue platform code. It took quite longer to make v2 than I expected, but it's finally here. Sorry for such late submission in this release cycle, but if it's not ready yet, no big deal - it certainly can wait for 2.6.41. v1 of these patches can be found here: http://lkml.org/lkml/2010/11/20/50 Current patchset is based on next-20110513 (it also can be applied to 2.6.39-rc7 after slight adjustment of Kconfig/Makefile chunks) Latest version of these patches were only slightly tested on top of 2.6.39-rc7 (-next is tested compile-only, as it doesn't boot due to some unrelated bugs), but all basic operations like playback, recording, EQ, Speaker, Mic Sidetone seem to work. Changes from v1: AK4641 codec: * '-codec' is removed from 'ak4641-codec' platform device name * SNDRV_PCM_RATE_8000_48000 macro now used instead of explicit freqs listing * Unneeded ifdef's on CONFIG_I2C are removed * Missing "Volume" added to names of volume controls, ditto for switches * TLV information is added to various volume controls * Enum control doesn't use indexed array of strings anymore * Enum controls like 'Mic Boost +20dB/0dB' are turned into volume controls * Deemphasis control is converted from enum to switch with automatic detection of optimal deemphasis value depending on current playback frequency * Check for wrong fs value is added to hw_params for HiFi DAI (that actually uncovered bug in hx4700 machine driver - it never set sysclk for the codec. Sound worked properly this way due to strange coincidence: both default fs in codec driver and divider in PXA i2s hardware were equal 256) * empty hw_params() function for Voice DAI is removed. Looks like that samplerate of Voice ADC/DAC can't be controlled by software at all, only format of samples. This Voice DAC/ADC are designed to be connected to Bluetooth chip, so I suspect codec gets sampling frequency directly from it. * Names of mic sidetone controls are adjusted * 'Left/Right Out Enable' DAPM controls are turned into ordinary ones * Mute code readability is slightly improved (based on Axel Lin's patch for ak4535 - commit cca67a3668) * My copyright is added to file, so people won't blame Harald for all the bugs I added during multicomponent conversion and other cleanup :) * Power/reset GPIOs are now handled in set_bias_level(), so codec is powered only when needed. HX4700 sound: * Setting of codec's sysclk is added to hw_params() HX4700 platform: * No changes Dmitry Artamonow (2): ASoC: Asahi Kasei AK4641 codec driver ASoC: add iPAQ hx4700 machine driver Philipp Zabel (1): pxa/hx4700: add I2C board info for AK4641 codec arch/arm/mach-pxa/hx4700.c | 18 + arch/arm/mach-pxa/include/mach/hx4700.h | 4 +- include/sound/ak4641.h | 26 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4641.c | 687 +++++++++++++++++++++++++++++++ sound/soc/codecs/ak4641.h | 47 +++ sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/hx4700.c | 260 ++++++++++++ 10 files changed, 1057 insertions(+), 2 deletions(-) create mode 100644 include/sound/ak4641.h create mode 100644 sound/soc/codecs/ak4641.c create mode 100644 sound/soc/codecs/ak4641.h create mode 100644 sound/soc/pxa/hx4700.c -- 1.7.4.rc1

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[alsa-devel] [PATCH RFC 0/2] da850 mcasp playback ping-pong buffers don't work
by Ben Gardiner 16 May '11

16 May '11

Hi All, I've been testing playback on the da850evm. The good news is on 2.6.39-rc7 playback works fine. The bad news is that if the patches in the series are applied -- playback does not work. The first patch copies the sram-size platform data to the dma parameters used by davinci-pcm the same way that the existing davinci-i2s driver does. The second patch assigns a buffer size to the playback sram platform data and sets the queues as per Troy's introduction of ping-pong buffers. I was hoping this would result in playback of audio; but instead there is soft click and then silence. I added pointer debug and observed that the period_update and hwptr_update events progress; but that's as far as I have probed. Does anyone have any clues why ping-pong buffers on da850 to the McASP would work? Or any clues why the patches are broken? Best Regards, Ben Gardiner Ben Gardiner (2): [RFC] ASoC: davinci-mcasp: enable ping-pong SRAM buffers [RFC] da850evm: enable mcasp ping-pong buffers arch/arm/configs/da8xx_omapl_defconfig | 1 + arch/arm/mach-davinci/board-da850-evm.c | 4 +++- sound/soc/davinci/davinci-mcasp.c | 2 ++ 3 files changed, 6 insertions(+), 1 deletions(-) -- 1.7.4.1

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Re: [alsa-devel] [PATCH 7/9] ASoC: Silence DEBUG_STRICT_USER_COPY_CHECKS=y warning
by Liam Girdwood 16 May '11

16 May '11

On Thu, 2011-05-12 at 16:50 -0700, Stephen Boyd wrote: > Enabling DEBUG_STRICT_USER_COPY_CHECKS causes the following > warning: > > In file included from arch/x86/include/asm/uaccess.h:573, > from include/linux/poll.h:14, > from include/sound/pcm.h:29, > from include/sound/ac97_codec.h:31, > from sound/soc/soc-core.c:34: > In function 'copy_from_user', > inlined from 'codec_reg_write_file' at > sound/soc/soc-core.c:252: > arch/x86/include/asm/uaccess_64.h:65: > warning: call to 'copy_from_user_overflow' declared with > attribute warning: copy_from_user() buffer size is not provably > correct > > presumably due to buf_size being signed causing GCC to fail to > see that buf_size can't become negative. > > Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> > Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > Signed-off-by: Stephen Boyd <sboyd(a)codeaurora.org> Acked-by: Liam Girdwood <lrg(a)ti.com>

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[alsa-devel] [PATCHv2] ASoC: Fix wrong data type access in a few codec drivers
by Jarkko Nikula 16 May '11

16 May '11

Commit fafd217 ("ASoC: Store a list of widgets in a DAPM mux/mixer kcontrol") changed the control private data type that is passed to snd_soc_cnew when creating dapm mixer and mux controls. Commit did not update a few codec drivers that are using their own put callbacks and thus are accessing a wrong data type. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Tested-by: Stephen Warren <swarren(a)nvidia.com> --- v2: - wm8994 and wm8995 needs to be changed too. Thanks to Stephen Warren <swarren(a)nvidia.com> for noticing. - "Otherwise, on Tegra+WM8903, I repro'd the bug, and tested the fix, so: Tested-by: Stephen Warren <swarren(a)nvidia.com>" --- sound/soc/codecs/tlv320aic3x.c | 3 ++- sound/soc/codecs/wm8903.c | 3 ++- sound/soc/codecs/wm8993.c | 3 ++- sound/soc/codecs/wm8994.c | 6 ++++-- sound/soc/codecs/wm8995.c | 4 ++-- 5 files changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6c43c13..c3d96fc 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -157,7 +157,8 @@ static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg, static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 957cd66..43e3d76 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -382,7 +382,8 @@ static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, static int wm8903_class_w_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); u16 reg; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 056aef9..9e5ff78 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -718,7 +718,8 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w, static int class_w_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); int ret; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e6dfa10..970a95c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -877,7 +877,8 @@ static const char *hp_mux_text[] = { static int wm8994_put_hp_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *w = wlist->widgets[0]; struct snd_soc_codec *codec = w->codec; int ret; @@ -1004,7 +1005,8 @@ SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, static int wm8994_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *w = wlist->widgets[0]; struct snd_soc_codec *codec = w->codec; int ret; diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 67eaaec..5ad873f 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -305,11 +305,11 @@ static int check_clk_sys(struct snd_soc_dapm_widget *source, static int wm8995_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *w = wlist->widgets[0]; struct snd_soc_codec *codec; int ret; - w = snd_kcontrol_chip(kcontrol); codec = w->codec; ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); wm8995_update_class_w(codec); -- 1.7.0.4

4 3

[alsa-devel] hda-jack-sense-test
by David Henningsson 16 May '11

16 May '11

Just a little something I wrote on the train back home yesterday. I hope someone finds it a nice way to figure out if your BIOS is lying to you about what pin is used for what. It (re)uses some of the classes in hda-analyzer. For Ubuntu 11.04 users, this small tool is now packaged together with codecgraph, hda-verb, hda-emu and hda-analyzer in my snd-hda-tools package, which is available in ppa:diwic/ppa. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

1 0

[alsa-devel] [PATCH 1/2] ASoC: Skip noop reconfiguration of WM8958 DSP2 algorithms
by Mark Brown 16 May '11

16 May '11

If we're setting the currently applied value for one of the DSP algorithm configurations we can just skip all the handling as the control set is a noop. This ensures we do not disrupt a running DSP. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8958-dsp2.c | 19 +++++++++++++++++++ 1 files changed, 19 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 5d4bc7a..ca26779 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -508,6 +508,9 @@ static int wm8958_mbc_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + if (wm8994->mbc_ena[mbc] == ucontrol->value.integer.value[0]) + return 0; + if (ucontrol->value.integer.value[0] > 1) return -EINVAL; @@ -628,6 +631,9 @@ static int wm8958_vss_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + if (wm8994->vss_ena[vss] == ucontrol->value.integer.value[0]) + return 0; + if (ucontrol->value.integer.value[0] > 1) return -EINVAL; @@ -689,6 +695,16 @@ static int wm8958_hpf_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + if (hpf < 3) { + if (wm8994->hpf1_ena[hpf % 3] == + ucontrol->value.integer.value[0]) + return 0; + } else { + if (wm8994->hpf2_ena[hpf % 3] == + ucontrol->value.integer.value[0]) + return 0; + } + if (ucontrol->value.integer.value[0] > 1) return -EINVAL; @@ -782,6 +798,9 @@ static int wm8958_enh_eq_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + if (wm8994->enh_eq_ena[eq] == ucontrol->value.integer.value[0]) + return 0; + if (ucontrol->value.integer.value[0] > 1) return -EINVAL; -- 1.7.5.1

2 2

[alsa-devel] [PATCH] ASoC: Add some missing volume update bit sets for wm_hubs devices
by Mark Brown 16 May '11

16 May '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index c7c2fec..e55b298 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -836,17 +836,21 @@ int wm_hubs_add_analogue_controls(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, WM8993_IN2_VU, WM8993_IN2_VU); + snd_soc_update_bits(codec, WM8993_SPEAKER_VOLUME_LEFT, + WM8993_SPKOUT_VU, WM8993_SPKOUT_VU); snd_soc_update_bits(codec, WM8993_SPEAKER_VOLUME_RIGHT, WM8993_SPKOUT_VU, WM8993_SPKOUT_VU); snd_soc_update_bits(codec, WM8993_LEFT_OUTPUT_VOLUME, - WM8993_HPOUT1L_ZC, WM8993_HPOUT1L_ZC); + WM8993_HPOUT1_VU | WM8993_HPOUT1L_ZC, + WM8993_HPOUT1_VU | WM8993_HPOUT1L_ZC); snd_soc_update_bits(codec, WM8993_RIGHT_OUTPUT_VOLUME, WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC, WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC); snd_soc_update_bits(codec, WM8993_LEFT_OPGA_VOLUME, - WM8993_MIXOUTL_ZC, WM8993_MIXOUTL_ZC); + WM8993_MIXOUTL_ZC | WM8993_MIXOUT_VU, + WM8993_MIXOUTL_ZC | WM8993_MIXOUT_VU); snd_soc_update_bits(codec, WM8993_RIGHT_OPGA_VOLUME, WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU, WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU); -- 1.7.5.1

2 1

[alsa-devel] [PATCH] alsa-lib - emu10k1.conf - Fix no sound problem when using SB Live! Platinum
by Raymond Yau 16 May '11

16 May '11

This patch allow pulseaudio to use "front" device for playback and capture , also allow user to use "rear" device of SB Live! for playback by adding type asym to front, rear and center_lfe After commit 16f716eadd61be80b28839396105c020e401294c which remove the lock If application using "hw", "front" , "rear" , "surround40" and "surround51" for playback , right channel or both left and right channels suddenly lost sound beause the route of the first stereo channel is changed when other application using "front", "rear" or "center_lfe" to record After applying this patch, "rear" , "center_lfe", "surround40" and "surround51" can no longer be used to record arecord -v -Drear:CARD=Live -f cd /dev/null ALSA lib pcm_asym.c:106:(_snd_pcm_asym_open) capture slave is not defined arecord: main:660: audio open error: Invalid argument

2 1

[alsa-devel] [PATCH] ALSA: HDA: Use one dmic only for Dell Studio 1558
by David Henningsson 16 May '11

16 May '11

There are no signs of a dmic at node 0x0b, so the user is left with an additional internal mic which does not exist. This commit removes that non-existing mic. Codec info at http://bugs.launchpad.net/bugs/731706. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[PATCH] ALSA: HDA: Use one dmic only for Dell Studio 1558
by David Henningsson 16 May '11

16 May '11

There are no signs of a dmic at node 0x0b, so the user is left with an additional internal mic which does not exist. This commit removes that non-existing mic. Cc: stable(a)kernel.org (2.6.32+) BugLink: http://bugs.launchpad.net/bugs/731706 Reported-by: James Page <james.page(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a2884bb..38f4317 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1604,7 +1604,7 @@ static const struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02fe, "Dell Studio XPS 1645", STAC_DELL_M6_BOTH), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0413, - "Dell Studio 1558", STAC_DELL_M6_BOTH), + "Dell Studio 1558", STAC_DELL_M6_DMIC), {} /* terminator */ }; -- 1.7.4.1 --------------080400010105080501090504--

1 0

Re: [alsa-devel] iMac12,2 cirrus_patch.c model detection
by Jérémy Lal 16 May '11

16 May '11

On 15/05/2011 11:53, Takashi Iwai wrote: > At Sun, 15 May 2011 10:20:10 +0200, > Jérémy Lal wrote: >> >> Hi, >> the imac12,2 27" (model name iMac12,2 from may 2011), >> has some headphones bug with your latest patch (the one that adds model=imac27). >> More precisions at : >> https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5364 >> >> I cooked a dirty patch (that fixes all controls that i can test, that is >> all analog in/out) and attached it in that bug report. >> However, since lspci reports subsystem 0x8086, 0x7270 too, >> i can't make that model auto-detected without killing your imac27 >> patch, see bug report. > > The bug tracking system is almost broken and actually no developer > is using it. I proposed to shut it down many times, but it's still > present, unfortunately. > > Could you post your results to alsa-devel ML and continue discussing > there? cc-ing alsa-devel, then. >> Would you have some advice to make it properly detect "my" model ? > > Create a table of codec SSID (not PCI SSID) and check the model via > snd_hda_check_board_codec_sid_config() first, then use > snd_hda_check_board_config() when no matching entry is found, as > fallback. That makes sense, and i'll use patch_realtek.c as an example. I'm inexperimented in C or alsa, though, so i might fail doing this. > Of course, this assumes that the codec SSID is unique. > Hopefully Apple does something right, at least. I don't get what you mean. Anyway bootcamp ships with the cirrus driver for windows, which lists all vendor/dev/subsys supported ids, with all configurations for pins, gpio, and other stuff i don't understand. It's tempting to use that to cover all actual models... I'd need to get a minimal knowledge of how intel hda and associated chips like cirrus work together. Any advice, beginner's guide ? Jérémy Lal

2 1

[alsa-devel] [PATCH v2 0/5] ASoC: OMAP4: Add support for HDMI audio
by Ricardo Neri 15 May '11

15 May '11

This is to add support for HDMI audio on OMAP4 chips. This work has two parts: DSS implementation and ASoC implementation. This set of patches presents the ASoC implementation. The approach is to utilize the DSS HDMI driver to act as an ASoC codec. This set of patches adds the ASoC CPU DAI and machine drivers. These drivers will utilize the ASoC HDMI codec provided in the DSS HDMI driver. DSS part of the implementation is currently under review in the linux-omap list. The DSS HDMI driver controls the availability of HDMI output and audio can be played only when the HDMI video device is enabled and the video timing in use is one of the CEA-861-D video timings supported by the OMAP4 HDMI IP. This implementation is for Basic Audio as defined in CEA-861-D: 16 bit/sample linear PCM 2-channel audio with sample rates of 32, 44.1 and 48kHz. As described in the HDMI specification, support for Basic Audio is mandatory for HDMI sinks and does not require of EDID parsing. Additionally, this implementation supports 24-bit samples in 32-bit words. Further audio capabilities such as more sample rates and multichannel audio will be added in the future when EDID parsing is available for HDMI driver. This implementation is based on on: * OMAP4: DSS2: Add dss_dss_clk opt clock for OMAP4, by Sumit Semwal (http://gitorious.org/linux-omap-dss2/linux/commit/0da5637569bb9bf93e6930cd1…) * Kernel 2.6.39-rc7 (git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux-2.6.git v2.6.39-rc7) * v1 of OMAP4: DSS2: HDMI: Add support for audio http://www.mail-archive.com/linux-omap@vger.kernel.org/msg49011.html In order to validate functionality, it is necessary to utilize the DSS HDMI ASoC codec. The complete implementation, DSS and ASoC parts, is available at: git://gitorious.org/omap-audio/linux-audio.git ricardon/topic/hdmi-audio-v2 Validation was performed on HDMI TV. Penguins were present on the display and audio playback was performed with aplay at 32, 44.1 and 48kHz with S16_LE and S24_LE sample formats. Validation was performed on SDP4430 ES2.1 and ES2.2 and Panda ES2.1 and ES2.0. Changes from V1: 1. Moved check of HDMI power state from CPU DAI start-up to machine hw_params. 2. Improved error handling when checking HDMI power-state. 3. Removed build-time conditionals for HDMI devices. 4. Added MODULE_ALIASes to drivers. 5. Merged CPU DAI header patch with its driver patch. 6. Use dev_err to print error messages. 7. Removed unnecessary calls to set_drvdata. 8. Corrected e-mail addresses of authors. 9. Corrected indentation. Ricardo Neri (5): OMAP4: HDMI: Add OMAP device for HDMI audio CPU DAI OMAP4: Add device for HDMI OMAP4 audio for ASoC machine driver ASoC: OMAP: Add CPU DAI driver for HDMI ASoC: OMAP4: Add HDMI Audio machine driver for OMAP4 boards ASoC: OMAP: Update Makefile and Kconfig for HDMI audio arch/arm/mach-omap2/devices.c | 26 ++++++ sound/soc/omap/Kconfig | 11 +++ sound/soc/omap/Makefile | 4 + sound/soc/omap/omap-hdmi.c | 158 ++++++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-hdmi.h | 36 +++++++++ sound/soc/omap/omap4-hdmi-card.c | 129 +++++++++++++++++++++++++++++++ 6 files changed, 364 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap-hdmi.c create mode 100644 sound/soc/omap/omap-hdmi.h create mode 100644 sound/soc/omap/omap4-hdmi-card.c

2 8

[alsa-devel] [PATCH] ASoC: Ensure output PGA is enabled for line outputs in wm_hubs
by Mark Brown 15 May '11

15 May '11

Also fix a left/right typo while we're at it. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 4005e9a..c7c2fec 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -787,17 +787,17 @@ static const struct snd_soc_dapm_route analogue_routes[] = { static const struct snd_soc_dapm_route lineout1_diff_routes[] = { { "LINEOUT1 Mixer", "IN1L Switch", "IN1L PGA" }, { "LINEOUT1 Mixer", "IN1R Switch", "IN1R PGA" }, - { "LINEOUT1 Mixer", "Output Switch", "Left Output Mixer" }, + { "LINEOUT1 Mixer", "Output Switch", "Left Output PGA" }, { "LINEOUT1N Driver", NULL, "LINEOUT1 Mixer" }, { "LINEOUT1P Driver", NULL, "LINEOUT1 Mixer" }, }; static const struct snd_soc_dapm_route lineout1_se_routes[] = { - { "LINEOUT1N Mixer", "Left Output Switch", "Left Output Mixer" }, - { "LINEOUT1N Mixer", "Right Output Switch", "Left Output Mixer" }, + { "LINEOUT1N Mixer", "Left Output Switch", "Left Output PGA" }, + { "LINEOUT1N Mixer", "Right Output Switch", "Right Output PGA" }, - { "LINEOUT1P Mixer", "Left Output Switch", "Left Output Mixer" }, + { "LINEOUT1P Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" }, { "LINEOUT1P Driver", NULL, "LINEOUT1P Mixer" }, @@ -806,17 +806,17 @@ static const struct snd_soc_dapm_route lineout1_se_routes[] = { static const struct snd_soc_dapm_route lineout2_diff_routes[] = { { "LINEOUT2 Mixer", "IN2L Switch", "IN2L PGA" }, { "LINEOUT2 Mixer", "IN2R Switch", "IN2R PGA" }, - { "LINEOUT2 Mixer", "Output Switch", "Right Output Mixer" }, + { "LINEOUT2 Mixer", "Output Switch", "Right Output PGA" }, { "LINEOUT2N Driver", NULL, "LINEOUT2 Mixer" }, { "LINEOUT2P Driver", NULL, "LINEOUT2 Mixer" }, }; static const struct snd_soc_dapm_route lineout2_se_routes[] = { - { "LINEOUT2N Mixer", "Left Output Switch", "Left Output Mixer" }, - { "LINEOUT2N Mixer", "Right Output Switch", "Left Output Mixer" }, + { "LINEOUT2N Mixer", "Left Output Switch", "Left Output PGA" }, + { "LINEOUT2N Mixer", "Right Output Switch", "Right Output PGA" }, - { "LINEOUT2P Mixer", "Right Output Switch", "Right Output Mixer" }, + { "LINEOUT2P Mixer", "Right Output Switch", "Right Output PGA" }, { "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" }, { "LINEOUT2P Driver", NULL, "LINEOUT2P Mixer" }, -- 1.7.5.1

1 0

[alsa-devel] [PATCH] ASoC: Ensure output PGA is enabled for line outputs in wm_hubs
by Mark Brown 15 May '11

15 May '11

Also fix a left/right typo while we're at it. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 4005e9a..7d674ee 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -787,17 +787,17 @@ static const struct snd_soc_dapm_route analogue_routes[] = { static const struct snd_soc_dapm_route lineout1_diff_routes[] = { { "LINEOUT1 Mixer", "IN1L Switch", "IN1L PGA" }, { "LINEOUT1 Mixer", "IN1R Switch", "IN1R PGA" }, - { "LINEOUT1 Mixer", "Output Switch", "Left Output Mixer" }, + { "LINEOUT1 Mixer", "Output Switch", "Left Output PGA" }, { "LINEOUT1N Driver", NULL, "LINEOUT1 Mixer" }, { "LINEOUT1P Driver", NULL, "LINEOUT1 Mixer" }, }; static const struct snd_soc_dapm_route lineout1_se_routes[] = { - { "LINEOUT1N Mixer", "Left Output Switch", "Left Output Mixer" }, - { "LINEOUT1N Mixer", "Right Output Switch", "Left Output Mixer" }, + { "LINEOUT1N Mixer", "Left Output Switch", "Left Output PGA" }, + { "LINEOUT1N Mixer", "Right Output Switch", "Right Output PGA" }, - { "LINEOUT1P Mixer", "Left Output Switch", "Left Output Mixer" }, + { "LINEOUT1P Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" }, { "LINEOUT1P Driver", NULL, "LINEOUT1P Mixer" }, @@ -806,17 +806,17 @@ static const struct snd_soc_dapm_route lineout1_se_routes[] = { static const struct snd_soc_dapm_route lineout2_diff_routes[] = { { "LINEOUT2 Mixer", "IN2L Switch", "IN2L PGA" }, { "LINEOUT2 Mixer", "IN2R Switch", "IN2R PGA" }, - { "LINEOUT2 Mixer", "Output Switch", "Right Output Mixer" }, + { "LINEOUT2 Mixer", "Output Switch", "Right Output PGA" }, { "LINEOUT2N Driver", NULL, "LINEOUT2 Mixer" }, { "LINEOUT2P Driver", NULL, "LINEOUT2 Mixer" }, }; static const struct snd_soc_dapm_route lineout2_se_routes[] = { - { "LINEOUT2N Mixer", "Left Output Switch", "Left Output Mixer" }, - { "LINEOUT2N Mixer", "Right Output Switch", "Left Output Mixer" }, + { "LINEOUT2N Mixer", "Left Output Switch", "Left Output PGA" }, + { "LINEOUT2N Mixer", "Right Output Switch", "Left Output PGA" }, - { "LINEOUT2P Mixer", "Right Output Switch", "Right Output Mixer" }, + { "LINEOUT2P Mixer", "Right Output Switch", "Right Output PGA" }, { "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" }, { "LINEOUT2P Driver", NULL, "LINEOUT2P Mixer" }, -- 1.7.5.1

2 2

[alsa-devel] X-Fi Surround 5.1 USB Remote Control LIRC config file
by roiman＠rambler.ru 15 May '11

15 May '11

Hi, all. Here is content of LIRC config file lircd.conf with IR-codes for X-Fi Surround 5.1 USB remote control. All buttons (include volume knob) works fine with Alsa driver 1.0.24 and LIRC 0.9.0. Wiki-page (http://alsa.opensrc.org/Usb-audio#Creative_USB_X-Fi_Surround_5.1) for this card presents information about button codes only for volume knob on this card. Because of this information may be useful, I think. Thanks, guys, for your work. Roman. ----------------------------------------------- ----------------------------------------------- begin remote name X-F.Surround-5.1.Remote.Control bits 8 gap 300000 begin codes knob-mute 0x0d knob-vol- 0x0f knob-vol+ 0x10 power 0x29 mute 0x2a vol+ 0x2b vol- 0x2c menu-back 0x2d display 0x2e play-pause 0x2f ok 0x30 up 0x31 down 0x32 prev 0x33 next 0x34 cmss3d-on 0x37 cmss3d-off 0x38 cryst-on 0x39 cryst-off 0x3a shuffle 0x3b repeat 0x3c end codes end remote ----------------------------------------------- -----------------------------------------------

1 0

[alsa-devel] [PATCH] fm801: clean-up radio-related Kconfig
by Ondrej Zary 15 May '11

15 May '11

Change the weird SND_FM801_TEA575X_BOOL define in Kconfig to SND_FM801_RADIO and remove TEA575X_RADIO define from fm801.c. Also update help text to include all supported cards. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-2.6.39-rc2-/sound/pci/Kconfig 2011-05-13 19:36:27.000000000 +0200 +++ linux-2.6.39-rc2/sound/pci/Kconfig 2011-05-13 19:23:00.000000000 +0200 @@ -554,18 +554,18 @@ config SND_FM801 To compile this driver as a module, choose M here: the module will be called snd-fm801. -config SND_FM801_TEA575X_BOOL +config SND_FM801_RADIO bool "ForteMedia FM801 + TEA5757 tuner" depends on SND_FM801 depends on VIDEO_V4L2=y || VIDEO_V4L2=SND_FM801 help Say Y here to include support for soundcards based on the ForteMedia - FM801 chip with a TEA5757 tuner connected to GPIO1-3 pins (Media - Forte SF256-PCS-02) into the snd-fm801 driver. + FM801 chip with a TEA5757 tuner (MediaForte SF256-PCS, SF256-PCP and + SF64-PCR) into the snd-fm801 driver. config SND_TEA575X tristate - depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO + depends on SND_FM801_RADIO || SND_ES1968_RADIO default SND_FM801 || SND_ES1968 source "sound/pci/hda/Kconfig" --- linux-2.6.39-rc2-/sound/pci/fm801.c 2011-05-13 19:39:23.000000000 +0200 +++ linux-2.6.39-rc2/sound/pci/fm801.c 2011-05-13 19:22:20.000000000 +0200 @@ -36,9 +36,8 @@ #include <asm/io.h> -#ifdef CONFIG_SND_FM801_TEA575X_BOOL +#ifdef CONFIG_SND_FM801_RADIO #include <sound/tea575x-tuner.h> -#define TEA575X_RADIO 1 #endif MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); @@ -196,7 +195,7 @@ struct fm801 { spinlock_t reg_lock; struct snd_info_entry *proc_entry; -#ifdef TEA575X_RADIO +#ifdef CONFIG_SND_FM801_RADIO struct snd_tea575x tea; #endif @@ -715,7 +714,7 @@ static int __devinit snd_fm801_pcm(struc * TEA5757 radio */ -#ifdef TEA575X_RADIO +#ifdef CONFIG_SND_FM801_RADIO /* GPIO to TEA575x maps */ struct snd_fm801_tea575x_gpio { @@ -1150,7 +1149,7 @@ static int snd_fm801_free(struct fm801 * outw(cmdw, FM801_REG(chip, IRQ_MASK)); __end_hw: -#ifdef TEA575X_RADIO +#ifdef CONFIG_SND_FM801_RADIO snd_tea575x_exit(&chip->tea); #endif if (chip->irq >= 0) @@ -1229,7 +1228,7 @@ static int __devinit snd_fm801_create(st snd_card_set_dev(card, &pci->dev); -#ifdef TEA575X_RADIO +#ifdef CONFIG_SND_FM801_RADIO chip->tea.private_data = chip; chip->tea.ops = &snd_fm801_tea_ops; sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); -- Ondrej Zary

2 3

[alsa-devel] Multistreaming Playback using Front Panel Headphone with realtek codec
by Raymond Yau 15 May '11

15 May '11

Request for testers to test the patch Multistreaming allows you to listen to one audio source through the back panel speakers and a second audio source through front panel headphones or speakers. Hardware requirement 1) 10 channels realtek codec (e.g. alc892) which already work with model=auto 2) Headphone at front panel and Line out at rear panel "hw:0,0" is used for the rear panel audio jack "hw:0,2" is used for the front panel headphone when "Independent HP" is switched ON Add an "Independent HP" switch to turn this feature on/off for desktop for those HDA codec Test: 1) speaker-test -c 8 -t wav -Dhw:0,0 2) speaker-test -c 2 -t wav -Dhw:0,2 It will need modification to support this feature for those desktop with 8 channels codec

3 13

[alsa-devel] [PATCH 1/3] tea575x: remove freq_fixup from struct
by Ondrej Zary 14 May '11

14 May '11

freq_fixup is a constant, no need to hold it in struct snd_tea575x and set in each driver. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-2.6.39-rc2-/include/sound/tea575x-tuner.h 2011-05-10 22:31:40.000000000 +0200 +++ linux-2.6.39-rc2/include/sound/tea575x-tuner.h 2011-05-12 21:00:50.000000000 +0200 @@ -26,6 +26,8 @@ #include <media/v4l2-dev.h> #include <media/v4l2-ioctl.h> +#define TEA575X_FMIF 10700 + #define TEA575X_DATA (1 << 0) #define TEA575X_CLK (1 << 1) #define TEA575X_WREN (1 << 2) @@ -46,7 +48,6 @@ struct snd_tea575x { bool mute; /* Device is muted? */ bool stereo; /* receiving stereo */ bool tuned; /* tuned to a station */ - unsigned int freq_fixup; /* crystal onboard */ unsigned int val; /* hw value */ unsigned long freq; /* frequency */ unsigned long in_use; /* set if the device is in use */ --- linux-2.6.39-rc2-/sound/i2c/other/tea575x-tuner.c 2011-05-10 22:31:40.000000000 +0200 +++ linux-2.6.39-rc2/sound/i2c/other/tea575x-tuner.c 2011-05-12 21:00:37.000000000 +0200 @@ -141,9 +141,9 @@ static void snd_tea575x_get_freq(struct freq /= 10; /* crystal fixup */ if (tea->tea5759) - freq += tea->freq_fixup; + freq += TEA575X_FMIF; else - freq -= tea->freq_fixup; + freq -= TEA575X_FMIF; tea->freq = freq * 16; /* from kHz */ } @@ -156,9 +156,9 @@ static void snd_tea575x_set_freq(struct freq /= 16; /* to kHz */ /* crystal fixup */ if (tea->tea5759) - freq -= tea->freq_fixup; + freq -= TEA575X_FMIF; else - freq += tea->freq_fixup; + freq += TEA575X_FMIF; /* freq /= 12.5 */ freq *= 10; freq /= 125; --- linux-2.6.39-rc2-/sound/pci/es1968.c 2011-05-10 22:31:43.000000000 +0200 +++ linux-2.6.39-rc2/sound/pci/es1968.c 2011-05-10 23:47:32.000000000 +0200 @@ -2794,7 +2794,6 @@ static int __devinit snd_es1968_create(s #ifdef CONFIG_SND_ES1968_RADIO chip->tea.card = card; - chip->tea.freq_fixup = 10700; chip->tea.private_data = chip; chip->tea.ops = &snd_es1968_tea_ops; if (!snd_tea575x_init(&chip->tea)) --- linux-2.6.39-rc2-/sound/pci/fm801.c 2011-05-10 23:24:39.000000000 +0200 +++ linux-2.6.39-rc2/sound/pci/fm801.c 2011-05-10 23:47:43.000000000 +0200 @@ -1231,7 +1231,6 @@ static int __devinit snd_fm801_create(st #ifdef TEA575X_RADIO chip->tea.card = card; - chip->tea.freq_fixup = 10700; chip->tea.private_data = chip; chip->tea.ops = &snd_fm801_tea_ops; if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 && -- Ondrej Zary

2 1

[alsa-devel] [PATCH v2] MFD: twl4030-codec: Update e-mail address
by Peter Ujfalusi 13 May '11

13 May '11

Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hello, Instead of keeping the old Nokia mail address, just change it to my new email address in the author line. drivers/mfd/twl4030-codec.c | 4 ++-- include/linux/mfd/twl4030-codec.h | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/mfd/twl4030-codec.c b/drivers/mfd/twl4030-codec.c index c02fded..534cc2e 100644 --- a/drivers/mfd/twl4030-codec.c +++ b/drivers/mfd/twl4030-codec.c @@ -1,7 +1,7 @@ /* * MFD driver for twl4030 codec submodule * - * Author: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> + * Author: Peter Ujfalusi <peter.ujfalusi(a)ti.com> * * Copyright: (C) 2009 Nokia Corporation * @@ -270,6 +270,6 @@ static void __devexit twl4030_codec_exit(void) } module_exit(twl4030_codec_exit); -MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi(a)nokia.com>"); +MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi(a)ti.com>"); MODULE_LICENSE("GPL"); diff --git a/include/linux/mfd/twl4030-codec.h b/include/linux/mfd/twl4030-codec.h index 2ec317c..5cc16bb 100644 --- a/include/linux/mfd/twl4030-codec.h +++ b/include/linux/mfd/twl4030-codec.h @@ -1,7 +1,7 @@ /* * MFD driver for twl4030 codec submodule * - * Author: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> + * Author: Peter Ujfalusi <peter.ujfalusi(a)ti.com> * * Copyright: (C) 2009 Nokia Corporation * -- 1.7.5.rc3

2 1

[alsa-devel] [PATCH] ASoC: Fix wrong data type access in a few codec drivers
by Jarkko Nikula 13 May '11

13 May '11

Commit fafd217 ("ASoC: Store a list of widgets in a DAPM mux/mixer kcontrol") changed the control private data type that is passed to snd_soc_cnew when creating dapm mixer and mux controls. Commit did not update a few codec drivers that are using their own put callbacks and thus are accessing a wrong data type. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Stephen Warren <swarren(a)nvidia.com> --- This was noted as a null pointer deference when trying to do mutex_lock in tlv320aic3x: snd_soc_dapm_put_volsw_aic3x but I believe something similar could orrur also in wm8993 and wm8903 too. --- sound/soc/codecs/tlv320aic3x.c | 3 ++- sound/soc/codecs/wm8903.c | 3 ++- sound/soc/codecs/wm8993.c | 3 ++- 3 files changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6c43c13..c3d96fc 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -157,7 +157,8 @@ static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg, static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 957cd66..43e3d76 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -382,7 +382,8 @@ static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, static int wm8903_class_w_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); u16 reg; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 056aef9..9e5ff78 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -718,7 +718,8 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w, static int class_w_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); int ret; -- 1.7.0.4

2 2

Re: [alsa-devel] Updated mach-types update
by Tony Lindgren 13 May '11

13 May '11

* Tony Lindgren <tony(a)atomide.com> [110324 10:57]: > * Russell King - ARM Linux <linux(a)arm.linux.org.uk> [110320 04:39]: > > On Sun, Mar 20, 2011 at 11:03:22AM +0000, Russell King - ARM Linux wrote: > > > > Note that OMAP references machine_is_omap2evm() yet this has never been > > merged. > > > > arch/arm/plat-omap/include/plat/uncompress.h: > > > > DEBUG_LL_OMAP2(1, omap2evm); > > > > so this entry needs to be removed. > > Hmm looks like we have LCD support and ASoC support for it, but no > board :) > > Anybody working with this board? > > If so, please send patches for the related board-*.c support ASAP. > Otherwise let's plan on deleting the related support. No patches, so let's remove it. Here's the patch to do that, I'm planning to add this to my devel-cleanup branch. Regards, Tony From: Tony Lindgren <tony(a)atomide.com> Date: Fri, 13 May 2011 04:41:32 -0700 Subject: [PATCH] omap: Remove support for omap2evm The board support has never been merged for it as noticed by Russell King <linux(a)arm.linux.org.uk>. So let's remove the related dead code. Cc: linux-fbdev(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Tony Lindgren <tony(a)atomide.com> diff --git a/arch/arm/plat-omap/include/plat/uncompress.h b/arch/arm/plat-omap/include/plat/uncompress.h index 30b891c..2b576f1 100644 --- a/arch/arm/plat-omap/include/plat/uncompress.h +++ b/arch/arm/plat-omap/include/plat/uncompress.h @@ -129,7 +129,6 @@ static inline void __arch_decomp_setup(unsigned long arch_id) DEBUG_LL_OMAP1(3, sx1); /* omap2 based boards using UART1 */ - DEBUG_LL_OMAP2(1, omap2evm); DEBUG_LL_OMAP2(1, omap_2430sdp); DEBUG_LL_OMAP2(1, omap_apollon); DEBUG_LL_OMAP2(1, omap_h4); diff --git a/drivers/video/omap/Makefile b/drivers/video/omap/Makefile index 49226a1..25db556 100644 --- a/drivers/video/omap/Makefile +++ b/drivers/video/omap/Makefile @@ -30,7 +30,6 @@ objs-y$(CONFIG_MACH_OMAP_APOLLON) += lcd_apollon.o objs-y$(CONFIG_MACH_OMAP_2430SDP) += lcd_2430sdp.o objs-y$(CONFIG_MACH_OMAP_3430SDP) += lcd_2430sdp.o objs-y$(CONFIG_MACH_OMAP_LDP) += lcd_ldp.o -objs-y$(CONFIG_MACH_OMAP2EVM) += lcd_omap2evm.o objs-y$(CONFIG_MACH_OMAP3EVM) += lcd_omap3evm.o objs-y$(CONFIG_MACH_OMAP3_BEAGLE) += lcd_omap3beagle.o objs-y$(CONFIG_FB_OMAP_LCD_MIPID) += lcd_mipid.o diff --git a/drivers/video/omap/lcd_omap2evm.c b/drivers/video/omap/lcd_omap2evm.c deleted file mode 100644 index 7e7a65c..0000000 --- a/drivers/video/omap/lcd_omap2evm.c +++ /dev/null @@ -1,192 +0,0 @@ -/* - * LCD panel support for the MISTRAL OMAP2EVM board - * - * Author: Arun C <arunedarath(a)mistralsolutions.com> - * - * Derived from drivers/video/omap/lcd_omap3evm.c - * Derived from drivers/video/omap/lcd-apollon.c - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. - */ - -#include <linux/module.h> -#include <linux/platform_device.h> -#include <linux/gpio.h> -#include <linux/i2c/twl.h> - -#include <plat/mux.h> -#include <asm/mach-types.h> - -#include "omapfb.h" - -#define LCD_PANEL_ENABLE_GPIO 154 -#define LCD_PANEL_LR 128 -#define LCD_PANEL_UD 129 -#define LCD_PANEL_INI 152 -#define LCD_PANEL_QVGA 148 -#define LCD_PANEL_RESB 153 - -#define TWL_LED_LEDEN 0x00 -#define TWL_PWMA_PWMAON 0x00 -#define TWL_PWMA_PWMAOFF 0x01 - -static unsigned int bklight_level; - -static int omap2evm_panel_init(struct lcd_panel *panel, - struct omapfb_device *fbdev) -{ - gpio_request(LCD_PANEL_ENABLE_GPIO, "LCD enable"); - gpio_request(LCD_PANEL_LR, "LCD lr"); - gpio_request(LCD_PANEL_UD, "LCD ud"); - gpio_request(LCD_PANEL_INI, "LCD ini"); - gpio_request(LCD_PANEL_QVGA, "LCD qvga"); - gpio_request(LCD_PANEL_RESB, "LCD resb"); - - gpio_direction_output(LCD_PANEL_ENABLE_GPIO, 1); - gpio_direction_output(LCD_PANEL_RESB, 1); - gpio_direction_output(LCD_PANEL_INI, 1); - gpio_direction_output(LCD_PANEL_QVGA, 0); - gpio_direction_output(LCD_PANEL_LR, 1); - gpio_direction_output(LCD_PANEL_UD, 1); - - twl_i2c_write_u8(TWL4030_MODULE_LED, 0x11, TWL_LED_LEDEN); - twl_i2c_write_u8(TWL4030_MODULE_PWMA, 0x01, TWL_PWMA_PWMAON); - twl_i2c_write_u8(TWL4030_MODULE_PWMA, 0x02, TWL_PWMA_PWMAOFF); - bklight_level = 100; - - return 0; -} - -static void omap2evm_panel_cleanup(struct lcd_panel *panel) -{ - gpio_free(LCD_PANEL_RESB); - gpio_free(LCD_PANEL_QVGA); - gpio_free(LCD_PANEL_INI); - gpio_free(LCD_PANEL_UD); - gpio_free(LCD_PANEL_LR); - gpio_free(LCD_PANEL_ENABLE_GPIO); -} - -static int omap2evm_panel_enable(struct lcd_panel *panel) -{ - gpio_set_value(LCD_PANEL_ENABLE_GPIO, 0); - return 0; -} - -static void omap2evm_panel_disable(struct lcd_panel *panel) -{ - gpio_set_value(LCD_PANEL_ENABLE_GPIO, 1); -} - -static unsigned long omap2evm_panel_get_caps(struct lcd_panel *panel) -{ - return 0; -} - -static int omap2evm_bklight_setlevel(struct lcd_panel *panel, - unsigned int level) -{ - u8 c; - if ((level >= 0) && (level <= 100)) { - c = (125 * (100 - level)) / 100 + 2; - twl_i2c_write_u8(TWL4030_MODULE_PWMA, c, TWL_PWMA_PWMAOFF); - bklight_level = level; - } - return 0; -} - -static unsigned int omap2evm_bklight_getlevel(struct lcd_panel *panel) -{ - return bklight_level; -} - -static unsigned int omap2evm_bklight_getmaxlevel(struct lcd_panel *panel) -{ - return 100; -} - -struct lcd_panel omap2evm_panel = { - .name = "omap2evm", - .config = OMAP_LCDC_PANEL_TFT | OMAP_LCDC_INV_VSYNC | - OMAP_LCDC_INV_HSYNC, - - .bpp = 16, - .data_lines = 18, - .x_res = 480, - .y_res = 640, - .hsw = 3, - .hfp = 0, - .hbp = 28, - .vsw = 2, - .vfp = 1, - .vbp = 0, - - .pixel_clock = 20000, - - .init = omap2evm_panel_init, - .cleanup = omap2evm_panel_cleanup, - .enable = omap2evm_panel_enable, - .disable = omap2evm_panel_disable, - .get_caps = omap2evm_panel_get_caps, - .set_bklight_level = omap2evm_bklight_setlevel, - .get_bklight_level = omap2evm_bklight_getlevel, - .get_bklight_max = omap2evm_bklight_getmaxlevel, -}; - -static int omap2evm_panel_probe(struct platform_device *pdev) -{ - omapfb_register_panel(&omap2evm_panel); - return 0; -} - -static int omap2evm_panel_remove(struct platform_device *pdev) -{ - return 0; -} - -static int omap2evm_panel_suspend(struct platform_device *pdev, - pm_message_t mesg) -{ - return 0; -} - -static int omap2evm_panel_resume(struct platform_device *pdev) -{ - return 0; -} - -struct platform_driver omap2evm_panel_driver = { - .probe = omap2evm_panel_probe, - .remove = omap2evm_panel_remove, - .suspend = omap2evm_panel_suspend, - .resume = omap2evm_panel_resume, - .driver = { - .name = "omap2evm_lcd", - .owner = THIS_MODULE, - }, -}; - -static int __init omap2evm_panel_drv_init(void) -{ - return platform_driver_register(&omap2evm_panel_driver); -} - -static void __exit omap2evm_panel_drv_exit(void) -{ - platform_driver_unregister(&omap2evm_panel_driver); -} - -module_init(omap2evm_panel_drv_init); -module_exit(omap2evm_panel_drv_exit); diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index ba9fc65..6c2c87e 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -13,7 +13,6 @@ snd-soc-rx51-objs := rx51.o snd-soc-ams-delta-objs := ams-delta.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o -snd-soc-omap2evm-objs := omap2evm.o snd-soc-omap3evm-objs := omap3evm.o snd-soc-am3517evm-objs := am3517evm.o snd-soc-sdp3430-objs := sdp3430.o diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c deleted file mode 100644 index 29b60d6..0000000 --- a/sound/soc/omap/omap2evm.c +++ /dev/null @@ -1,139 +0,0 @@ -/* - * omap2evm.c -- SoC audio machine driver for omap2evm board - * - * Author: Arun KS <arunks(a)mistralsolutions.com> - * - * Based on sound/soc/omap/overo.c by Steve Sakoman - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include <linux/clk.h> -#include <linux/platform_device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> -#include <mach/hardware.h> -#include <mach/gpio.h> -#include <plat/mcbsp.h> - -#include "omap-mcbsp.h" -#include "omap-pcm.h" - -static int omap2evm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret; - - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return ret; - } - - /* Set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, - SND_SOC_CLOCK_IN); - if (ret < 0) { - printk(KERN_ERR "can't set codec system clock\n"); - return ret; - } - - return 0; -} - -static struct snd_soc_ops omap2evm_ops = { - .hw_params = omap2evm_hw_params, -}; - -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link omap2evm_dai = { - .name = "TWL4030", - .stream_name = "TWL4030", - .cpu_dai_name = "omap-mcbsp-dai.1", - .codec_dai_name = "twl4030-hifi", - .platform_name = "omap-pcm-audio", - .codec_name = "twl4030-codec", - .ops = &omap2evm_ops, -}; - -/* Audio machine driver */ -static struct snd_soc_card snd_soc_omap2evm = { - .name = "omap2evm", - .dai_link = &omap2evm_dai, - .num_links = 1, -}; - -static struct platform_device *omap2evm_snd_device; - -static int __init omap2evm_soc_init(void) -{ - int ret; - - if (!machine_is_omap2evm()) - return -ENODEV; - printk(KERN_INFO "omap2evm SoC init\n"); - - omap2evm_snd_device = platform_device_alloc("soc-audio", -1); - if (!omap2evm_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(omap2evm_snd_device, &snd_soc_omap2evm); - - ret = platform_device_add(omap2evm_snd_device); - if (ret) - goto err1; - - return 0; - -err1: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(omap2evm_snd_device); - - return ret; -} -module_init(omap2evm_soc_init); - -static void __exit omap2evm_soc_exit(void) -{ - platform_device_unregister(omap2evm_snd_device); -} -module_exit(omap2evm_soc_exit); - -MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); -MODULE_DESCRIPTION("ALSA SoC omap2evm"); -MODULE_LICENSE("GPL");

5 5

[alsa-devel] [PATCH] ASoC: omap-pcm: Period wakeup disabling on OMAP2+
by Peter Ujfalusi 13 May '11

13 May '11

Allow disabling ALSA period wakeup interrupts. This can only be done on OMAP2+ (2/3/4), since there we can chain the DMA. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hello, with this patch we can dramatically decrease the number of wake ups for user space during audio playback on OMAP2/3/4 platforms. Currently PulseAudio is the only one supporting this mode (using timer instead of pcm wakeup). If for example PA is configured to use ~2sec audio buffer, we do not need to handle the DMA interrupts - they are coming in ~10ms (McBSP2), or ~1ms intervals. Internally we still going to have triggers (from McBSP to the sDMA), but we are not going to bother to let SW stack know about this. Output from pulseaudio, without this patch: I: alsa-util.c: cannot disable ALSA period wakeups I: alsa-util.c: ALSA period wakeups were not disabled with this patch in the kernel: I: alsa-util.c: Trying to disable ALSA period wakeups, using timers only I: alsa-util.c: ALSA period wakeups disabled And in fact the amount of interrupts decreased with the patch. Peter sound/soc/omap/omap-pcm.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 551b7f5..3a1e395 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -37,7 +37,8 @@ static const struct snd_pcm_hardware omap_pcm_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = 32, @@ -195,7 +196,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) if ((cpu_is_omap1510())) omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ | OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ); - else + else if (!substream->runtime->no_period_wakeup) omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ); if (!(cpu_class_is_omap1())) { -- 1.7.5.rc3

4 3

[alsa-devel] [PATCH] ASoC: omap-mcbsp: Remove restrictive checks for cpu type
by Sanjeev Premi 13 May '11

13 May '11

Current checks for cpu type were too restrictive leading to failures for other silicons in same family. The problem was found while testing audio playback on AM37x and AM35x processors. But should exist on OMAP36xx as well. Signed-off-by: Sanjeev Premi <premi(a)ti.com> cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> cc: Liam Girdwood <lrg(a)ti.com> cc: Jarkko Nikula <jhnikula(a)gmail.com> --- Changes in v4: Reposting after validating that patch still applies cleanly on linux-omap master [46966f1] Changed the subject as suggested by Jarkko. Since the subject has changed, i have omitted v4. Changes in v3:: None. Reposting[1] after subscribing to alsa-devel list Changes in v2: 1) Added one more check to the changes. 2) Updated commit message to use plurals. [1] http://marc.info/?l=linux-omap&m=130026430110592&w=2 sound/soc/omap/omap-mcbsp.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 2175f09..a88a0bf 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -146,7 +146,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) */ - if (cpu_is_omap343x() || cpu_is_omap44xx()) { + if (cpu_is_omap34xx() || cpu_is_omap44xx()) { /* * Rule for the buffer size. We should not allow * smaller buffer than the FIFO size to avoid underruns @@ -258,7 +258,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } - if (cpu_is_omap343x()) { + if (cpu_is_omap34xx()) { dma_data->set_threshold = omap_mcbsp_set_threshold; /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */ if (omap_mcbsp_get_dma_op_mode(bus_id) == -- 1.7.2.2

5 4

[alsa-devel] [PATCH] MAINTAINERS: ASoC and Regulator email address change.
by Liam Girdwood 13 May '11

13 May '11

Update my email address to new employer. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- MAINTAINERS | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 03cfc1c..75318bf 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5840,7 +5840,7 @@ F: include/sound/ F: sound/ SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) -M: Liam Girdwood <lrg(a)slimlogic.co.uk> +M: Liam Girdwood <lrg(a)ti.com> M: Mark Brown <broonie(a)opensource.wolfsonmicro.com> T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) @@ -6736,7 +6736,7 @@ F: drivers/scsi/vmw_pvscsi.c F: drivers/scsi/vmw_pvscsi.h VOLTAGE AND CURRENT REGULATOR FRAMEWORK -M: Liam Girdwood <lrg(a)slimlogic.co.uk> +M: Liam Girdwood <lrg(a)ti.com> M: Mark Brown <broonie(a)opensource.wolfsonmicro.com> W: http://opensource.wolfsonmicro.com/node/15 W: http://www.slimlogic.co.uk/?p=48 -- 1.7.4.1

3 4

Re: [alsa-devel] GR-55 Driver
by Daniel Mack 13 May '11

13 May '11

Hi Jeff, please, bear in mind two rules: 1. don't just reply to me, but always use the "reply-to-all" feature in your mail client. This way, all people on the ALSA list can follow the discussion and chime in. 2. don't top-post, always reply using proper quoting. On Fri, May 13, 2011 at 4:30 AM, Jeffrey Scott Flesher Gmail <jeffrey.scott.flesher(a)gmail.com> wrote: > Thanks. > > I'm running Ubuntu Studio 10.04 x64 with realtime kernel from > http://ppa.launchpad.net/abogani/ppa/ubuntu lucid (linux-realtime - > 2.6.33-29.1); my motherboard is ASUS M4A78T-E onboard VIA sound, Nvidia 460 > GTX Video card; AMD 955 X4 CPU. > Link to kernel used > https://launchpad.net/~abogani/+archive/ppa/+packages?field.name_filter=&fi… > > I searched for instructions on how to do this, with no good results; so I > must ask someone that knows how. Things are a little more complicated as modern Linux distributions use techniques like initramfs. For Ubuntu, you should follow one of the tutorials. Two I found are here: https://help.ubuntu.com/community/Kernel/Compile http://parabing.com/2011/04/28/ubuntu-natty-a-custom-kernel-is-what-you-wan… That looks comprehensive. Just don't download the ubuntu Kernel sources as described but clone the ALSA repository (which you already did). And instead of downloading the patches as described, just use "git apply" on the file I sent you. On case you want to revert the patching and return to the kernel you originally cloned, just call "git reset --hard". > Just to make sure I do this right, since I've never compiled ALSA driver > before; these are the instructions I used: > > 1. git clone > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > 2. cp 0001-ALSA-snd-usb-add-quirks-for-Roland-GR-55.patch /sound-2.6 > 3. cd sound-2.6 > 4. patch -p1 < ./0001-ALSA-snd-usb-add-quirks-for-Roland-GR-55.patch > 5. make x86_64_defconfig > 6. make > 7. make install > > This worked, not tested; I still need to get the kernel to see it; I did > this in my home directory; where should I move it? > > I see a file called vmlinux, 19.7 MB executable; I take it this is the new > ALSA Driver. > > I do see vmlinux files that are new in these folders; which would seem to me > that they got where they need to go with the make install command. > /proc/4055/cwd > /proc/4055/cwd/arch/x86/boot/compressed > > Now do I need to compile this into the Kernel, if so, what kernel do you > recommend; realtime, low latency or normal; and what version? I would first go for a normal kernel configuration, and once the driver supports the device, care for optimisation. HTH, Daniel

1 0

Re: [alsa-devel] [PATCH] ASoC: twl6040 - fix LINEGAIN volume control
by Mark Brown 13 May '11

13 May '11

On Thu, May 12, 2011 at 01:54:21PM +0100, Liam Girdwood wrote: > Fix the TWL6040 LINEGAIN volume control to match the TRM. > > Signed-off-by: Liam Girdwood <lrg(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com>

1 0

[alsa-devel] [PATCH alsa-lib emu10k1.conf 1/1] Fix "front" device of SB Live! Platinum
by Raymond Yau 13 May '11

13 May '11

Since commit 148c2b8e7c12b4ab8a039995fa9904e7e1300cc4. Front channels via fxbus 8 and 9 No sound from the front jack of SB Live! platinum CT4760P when the application using "front" device but sound from front jack when application using hw:0,0 and /dev/dsp Error message " I: (alsa-lib)setup.c: Cannot lock ctl elem*"* when Pulseaudio server open "front" device for playback and capture at the same time because of locked "EMU10K1 Send PCM Volume"

2 5

Re: [alsa-devel] GR-55 Driver
by Daniel Mack 13 May '11

13 May '11

On Thu, May 12, 2011 at 10:53 PM, Jeffrey Scott Flesher Gmail <jeffrey.scott.flesher(a)gmail.com> wrote: > Output from lsusb -v for Roland GR-55 > ****** > > Bus 001 Device 008: ID 0582:0127 Roland Corp. > Device Descriptor: > bLength 18 > bDescriptorType 1 > bcdUSB 2.00 > bDeviceClass 255 Vendor Specific Class > bDeviceSubClass 0 > bDeviceProtocol 255 > bMaxPacketSize0 64 > idVendor 0x0582 Roland Corp. > idProduct 0x0127 > bcdDevice 0.01 > iManufacturer 1 > iProduct 2 > iSerial 0 > bNumConfigurations 1 > Configuration Descriptor: > bLength 9 > bDescriptorType 2 > wTotalLength 167 > bNumInterfaces 3 > bConfigurationValue 1 > iConfiguration 0 > bmAttributes 0xc0 > Self Powered > MaxPower 0mA > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 0 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 2 > iInterface 0 > ** UNRECOGNIZED: 06 24 f1 01 00 00 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 0 > bAlternateSetting 1 > bNumEndpoints 1 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 2 > iInterface 0 > ** UNRECOGNIZED: 07 24 01 01 00 01 00 > ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x05 EP 5 OUT > bmAttributes 5 > Transfer Type Isochronous > Synch Type Asynchronous > Usage Type Data > wMaxPacketSize 0x0070 1x 112 bytes > bInterval 1 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 1 > iInterface 0 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 1 > bNumEndpoints 1 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 1 > iInterface 0 > ** UNRECOGNIZED: 07 24 01 07 00 01 00 > ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x84 EP 4 IN > bmAttributes 37 > Transfer Type Isochronous > Synch Type Asynchronous > Usage Type Implicit feedback Data > wMaxPacketSize 0x0070 1x 112 bytes > bInterval 1 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 0 > bNumEndpoints 2 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 3 > bInterfaceProtocol 0 > iInterface 0 > ** UNRECOGNIZED: 06 24 f1 02 02 02 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x03 EP 3 OUT > bmAttributes 2 > Transfer Type Bulk > Synch Type None > Usage Type Data > wMaxPacketSize 0x0200 1x 512 bytes > bInterval 1 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x82 EP 2 IN > bmAttributes 2 > Transfer Type Bulk > Synch Type None > Usage Type Data > wMaxPacketSize 0x0200 1x 512 bytes > bInterval 0 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 1 > bNumEndpoints 2 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 3 > bInterfaceProtocol 0 > iInterface 0 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x03 EP 3 OUT > bmAttributes 2 > Transfer Type Bulk > Synch Type None > Usage Type Data > wMaxPacketSize 0x0200 1x 512 bytes > bInterval 1 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x81 EP 1 IN > bmAttributes 3 > Transfer Type Interrupt > Synch Type None > Usage Type Data > wMaxPacketSize 0x0040 1x 64 bytes > bInterval 1 > Ok, that seems to be identical to the BOSS JS-8 layout, so it can be handled by the generic audio driver. Can you try the attached patch? Make sure you apply it against the most recent ALSA tree found in the git repository at git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git. Thanks, Daniel

1 0

[alsa-devel] Output from lsusb -v for Roland GR-55
by Jeffrey Scott Flesher Gmail 13 May '11

13 May '11

Output from lsusb -v for Roland GR-55 ****** Bus 001 Device 008: ID 0582:0127 Roland Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x0582 Roland Corp. idProduct 0x0127 bcdDevice 0.01 iManufacturer 1 iProduct 2 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 167 bNumInterfaces 3 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 06 24 f1 01 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x05 EP 5 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 02 02 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 1

1 0

[alsa-devel] plughw and hw
by hong zhang 13 May '11

13 May '11

Takashi, Last e-mail, you mention do not use plughw but hw. I compare both of them. And wonder which way is correct to capture multiple pins. my HDMI has 4 pins 1. HW failed with "-c 8" # arecord -f dat -d 5 -D hw:0,1 -c 8 05-12-new-hw-3pins.wav Recording WAVE '05-12-new-hw-3pins.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Channels 8 arecord: set_params:1059: Channels count non available 2. HW collect data for all 4 pins without "-c 8" 3. plughw with "-c 8" but only first pin has data inside 2 channels. The rest pins collect 0 data. 4. plughw collects data for all pins without "-c 8" why "-c 8" would not work right? Thanks. ---henry

1 0

[alsa-devel] [PATCH 3/3] tea575x: use better card and bus names
by Ondrej Zary 12 May '11

12 May '11

Provide real card and bus_info instead of hardcoded values. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-2.6.39-rc2-/include/sound/tea575x-tuner.h 2011-05-12 21:53:43.000000000 +0200 +++ linux-2.6.39-rc2/include/sound/tea575x-tuner.h 2011-05-12 21:37:40.000000000 +0200 @@ -52,6 +52,8 @@ struct snd_tea575x { unsigned long in_use; /* set if the device is in use */ struct snd_tea575x_ops *ops; void *private_data; + u8 card[32]; + u8 bus_info[32]; }; int snd_tea575x_init(struct snd_tea575x *tea); --- linux-2.6.39-rc2-/sound/i2c/other/tea575x-tuner.c 2011-05-12 21:22:35.000000000 +0200 +++ linux-2.6.39-rc2/sound/i2c/other/tea575x-tuner.c 2011-05-12 21:41:11.000000000 +0200 @@ -178,8 +178,9 @@ static int vidioc_querycap(struct file * struct snd_tea575x *tea = video_drvdata(file); strlcpy(v->driver, "tea575x-tuner", sizeof(v->driver)); - strlcpy(v->card, tea->tea5759 ? "TEA5759" : "TEA5757", sizeof(v->card)); - sprintf(v->bus_info, "PCI"); + strlcpy(v->card, tea->card, sizeof(v->card)); + strlcat(v->card, tea->tea5759 ? " TEA5759" : " TEA5757", sizeof(v->card)); + strlcpy(v->bus_info, tea->bus_info, sizeof(v->bus_info)); v->version = RADIO_VERSION; v->capabilities = V4L2_CAP_TUNER | V4L2_CAP_RADIO; return 0; --- linux-2.6.39-rc2-/sound/pci/es1968.c 2011-05-12 21:53:43.000000000 +0200 +++ linux-2.6.39-rc2/sound/pci/es1968.c 2011-05-12 21:45:59.000000000 +0200 @@ -2795,6 +2795,8 @@ static int __devinit snd_es1968_create(s #ifdef CONFIG_SND_ES1968_RADIO chip->tea.private_data = chip; chip->tea.ops = &snd_es1968_tea_ops; + strlcpy(chip->tea.card, "SF64-PCE2", sizeof(chip->tea.card)); + sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); if (!snd_tea575x_init(&chip->tea)) printk(KERN_INFO "es1968: detected TEA575x radio\n"); #endif --- linux-2.6.39-rc2-/sound/pci/fm801.c 2011-05-12 21:53:43.000000000 +0200 +++ linux-2.6.39-rc2/sound/pci/fm801.c 2011-05-12 21:50:19.000000000 +0200 @@ -1232,6 +1232,7 @@ static int __devinit snd_fm801_create(st #ifdef TEA575X_RADIO chip->tea.private_data = chip; chip->tea.ops = &snd_fm801_tea_ops; + sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 && (tea575x_tuner & TUNER_TYPE_MASK) < 4) { if (snd_tea575x_init(&chip->tea)) @@ -1246,6 +1247,7 @@ static int __devinit snd_fm801_create(st break; } } + strlcpy(chip->tea.card, snd_fm801_tea575x_gpios[(tea575x_tuner & TUNER_TYPE_MASK) - 1].name, sizeof(chip->tea.card)); #endif *rchip = chip; -- Ondrej Zary

1 0

[alsa-devel] [PATCH 2/3] tea575x: remove unused card from struct
by Ondrej Zary 12 May '11

12 May '11

struct snd_card *card is present in struct snd_tea575x but never used. Remove it. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-2.6.39-rc2-/include/sound/tea575x-tuner.h 2011-05-12 21:22:35.000000000 +0200 +++ linux-2.6.39-rc2/include/sound/tea575x-tuner.h 2011-05-12 21:21:37.000000000 +0200 @@ -42,7 +42,6 @@ struct snd_tea575x_ops { }; struct snd_tea575x { - struct snd_card *card; struct video_device *vd; /* video device */ bool tea5759; /* 5759 chip is present */ bool mute; /* Device is muted? */ --- linux-2.6.39-rc2-/sound/pci/es1968.c 2011-05-12 21:22:35.000000000 +0200 +++ linux-2.6.39-rc2/sound/pci/es1968.c 2011-05-12 21:22:09.000000000 +0200 @@ -2793,7 +2793,6 @@ static int __devinit snd_es1968_create(s snd_card_set_dev(card, &pci->dev); #ifdef CONFIG_SND_ES1968_RADIO - chip->tea.card = card; chip->tea.private_data = chip; chip->tea.ops = &snd_es1968_tea_ops; if (!snd_tea575x_init(&chip->tea)) --- linux-2.6.39-rc2-/sound/pci/fm801.c 2011-05-12 21:22:35.000000000 +0200 +++ linux-2.6.39-rc2/sound/pci/fm801.c 2011-05-12 21:22:05.000000000 +0200 @@ -1230,7 +1230,6 @@ static int __devinit snd_fm801_create(st snd_card_set_dev(card, &pci->dev); #ifdef TEA575X_RADIO - chip->tea.card = card; chip->tea.private_data = chip; chip->tea.ops = &snd_fm801_tea_ops; if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 && -- Ondrej Zary

1 0

[alsa-devel] GR-55 Driver
by Jeffrey Scott Flesher Gmail 12 May '11

12 May '11

I need a Driver for the Roland GR-55 Guitar Synthesizer, not sure how to get started, so I thought this would be a good place to ask. The GR-55 has MIDI and AUDIO KSNAME_Wo11="WaveOut1-1" WDM.Wo11.szPname="OUT (GR-55)" KSNAME_Wi11="WaveIn1-1" WDM.Wi11.szPname="IN (GR-55)" KSNAME_Md11="Midi1-1" WDM.Md11.szPname="GR-55" My thinking is that there may be a driver that is similar, or someone could already be working on a driver, if so, I would like to find out if I could work on that project or if I could start a new project here. I did contact Roland about getting help in the way of Documentation; but was told not to count on it since most of this documentation is proprietary; so I might end up having to snoop the information to reverse engineer it. I have been a programmer for over 30 years, and have written Unix drivers and interfaces back in the 80's; things have changed since then and Linux is Unix like, but ALSA seems a lot easier than what we did back in those days. I can get up to speed with just a point in the right direction. Things like do I do this as a driver or a module? Are just a few questions that I need answers to, just to get started. Can you recommend a driver that is close enough to use as a template to start a new driver from? Thanks. Jeff Flesher

2 1

[alsa-devel] Small Regression in midi for M-Audio Fast Track Ultra.
by Grant Diffey 12 May '11

12 May '11

In 2.6.33/34 when support originally was put in alsa mainline the MIDI ports were functional today they're not for me and I get the following in the kernel log when using this card. (Audio all works acceptably.) [ 1846.736062] usb 2-1: new high speed USB device using ehci_hcd and address 4 [ 1846.869790] usb 2-1: config 1 interface 3 altsetting 0 bulk endpoint 0x7 has invalid maxpacket 8 [ 1846.869802] usb 2-1: config 1 interface 3 altsetting 0 bulk endpoint 0x87 has invalid maxpacket 8 [ 1846.870897] usb 2-1: New USB device found, idVendor=0763, idProduct=2080 [ 1846.870905] usb 2-1: New USB device strings: Mfr=1, Product=2, SerialNumber=0 [ 1846.870912] usb 2-1: Product: Fast Track Ultra [ 1846.870917] usb 2-1: Manufacturer: M-Audio This issue may be unrelated to the data below but it's included in case it's useful. The midi port is interface 3 and has a comment in quirk-table.h like this /* interface 3 (MIDI) is standard compliant */ { .ifnum = -1 } Which is as I understand it a noop to the quirk stuff so that it just get's processed normally. Which could also be completely wrong. I've verified in lsusb -v of the midi interface (3) of the card is below Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 5 MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 63 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 2 iJack 0 MIDIStreaming Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) Warning: Descriptor too short bJackType 1 Embedded bJackID 3 bNrInputPins 1 baSourceID( 0) 2 BaSourcePin( 0) 1 iJack 8 MIDIStreaming Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) Warning: Descriptor too short bJackType 2 External bJackID 4 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 97 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x07 EP 7 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0008 1x 8 bytes bInterval 8 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x87 EP 7 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0008 1x 8 bytes bInterval 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 3

4 9

[alsa-devel] [PATCH] ASoC: core - allow ASoC more flexible machine name
by Liam Girdwood 12 May '11

12 May '11

Allow ASoC machine drivers to register a driver name and a longname. This allows user space to determine the flavour of machine driver. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 6 ++++-- 2 files changed, 6 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index b27c7a2..f1de3e0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -702,6 +702,8 @@ struct snd_soc_aux_dev { /* SoC card */ struct snd_soc_card { const char *name; + const char *long_name; + const char *driver_name; struct device *dev; struct snd_card *snd_card; struct module *owner; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a477e21..c1a4cf4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1926,9 +1926,11 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) card->num_dapm_routes); snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), - "%s", card->name); - snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), "%s", card->name); + snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), + "%s", card->long_name ? card->long_name : card->name); + snprintf(card->snd_card->driver, sizeof(card->snd_card->driver), + "%s", card->driver_name); if (card->late_probe) { ret = card->late_probe(card); -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: DMIC codec - Add input widget
by Liam Girdwood 12 May '11

12 May '11

From: Misael Lopez Cruz <misael.lopez(a)ti.com> Digital microphones can have some additional elements in their audio path (like microphone bias). An input widget is required for digital microphone CODEC driver to allow external connections in machine drivers. Change-Id: Ice9fcb2dbe9acccd859f90ae875d2280926bf08a Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/codecs/dmic.c | 26 +++++++++++++++++++++++++- 1 files changed, 25 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index 57e9dac..f9a8773 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -39,7 +39,31 @@ static struct snd_soc_dai_driver dmic_dai = { }, }; -static struct snd_soc_codec_driver soc_dmic = {}; +static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = { + SND_SOC_DAPM_AIF_OUT("DMIC AIF", "Capture", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_INPUT("DMic"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"DMIC AIF", NULL, "DMic"}, +}; + +static int dmic_probe(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets, + ARRAY_SIZE(dmic_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); + snd_soc_dapm_new_widgets(dapm); + + return 0; +} + +static struct snd_soc_codec_driver soc_dmic = { + .probe = dmic_probe, +}; static int __devinit dmic_dev_probe(struct platform_device *pdev) { -- 1.7.4.1

2 1

[alsa-devel] [PATCH] Always link to libm
by Alexander Stein 12 May '11

12 May '11

From: Alexander Stein <alexander.stein(a)systec-electronic.com> Starting with commit 9b6df1cf64ea7ccb329b4922d138c1f36ace00c0 control: add ASCII parsers from amixer ctlparse.c uses ceil() and so libm is need unrelated to soft- or hardfloat. Signed-off-by: Alexander Stein <alexander.stein(a)systec-electronic.com> --- configure.in | 5 +---- 1 files changed, 1 insertions(+), 4 deletions(-) diff --git a/configure.in b/configure.in index 7ee0ccc..687e5eb 100644 --- a/configure.in +++ b/configure.in @@ -228,10 +228,7 @@ else AC_MSG_RESULT(no) fi -ALSA_DEPLIBS="" -if test "$softfloat" != "yes"; then - ALSA_DEPLIBS="-lm" -fi +ALSA_DEPLIBS="-lm" dnl Check for libdl AC_MSG_CHECKING(for libdl) -- 1.7.3.4

2 1

[alsa-devel] [PATCH 0/6] ASoC: OMAP4: Add support for HDMI audio
by Ricardo Neri 12 May '11

12 May '11

This is to add support for HDMI audio on OMAP4 chips. This work has two parts: DSS implementation and ASoC implementation. This set of patches presents the ASoC implementation. The approach is to utilize the DSS HDMI driver to act as an ASoC codec. This set of patches adds the ASoC CPU DAI and machine drivers. These drivers will utilize the ASoC HDMI codec provided in the DSS HDMI driver. DSS part of the implementation is currently under review in the linux-omap list. The DSS HDMI driver controls the availability of HDMI output and audio can be played only when the HDMI video device is enabled and the video timing in use is one of the CEA-861-D video timings supported by the OMAP4 HDMI IP. This implementation is for Basic Audio as defined in CEA-861-D: 16 bit/sample linear PCM 2-channel audio with sample rates of 32, 44.1 and 48kHz. As described in the HDMI specification, support for Basic Audio is mandatory for HDMI sinks and does not require of EDID parsing. Additionally, this implementation supports 24-bit samples in 32-bit words. Further audio capabilities such as more sample rates and multichannel audio will be added in the future when EDID parsing is available for HDMI driver. This implementation is based on on: * OMAP4: DSS2: Add dss_dss_clk opt clock for OMAP4, by Sumit Semwal (http://gitorious.org/linux-omap-dss2/linux/commit/0da5637569bb9bf93e6930cd1…) * Kernel 2.6.39-rc6 (git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux-2.6.git v2.6.39-rc6) * v4 of OMAP4: DSS2: HDMI: Add support for audio git://gitorious.org/omap-audio/linux-audio.git ricardon/topic/hdmi-audio-v1 In order to validate functionality, it is necessary to utilize the DSS HDMI ASoC codec. The complete implementation, DSS and ASoC parts, is available at: git://gitorious.org/omap-audio/linux-audio.git ricardon/topic/hdmi-audio-v1 Validation was performed on HDMI TV. Penguins were present on the display and audio playback was performed with aplay at 32, 44.1 and 48kHz with S16_LE and S24_LE sample formats. Validation was performed on SDP4430 ES2.1 and Panda ES2.1 and ES2.0. Ricardo Neri (6): OMAP4: HDMI: Add OMAP device for HDMI audio CPU DAI OMAP4: Add device for HDMI OMAP4 audio for ASoC machine driver ASoC: OMAP: Add header for HDMI CPU DAI ASoC: OMAP: Add CPU DAI driver for HDMI ASoC: OMAP4: Add HDMI Audio machine driver for OMAP4 boards ASoC: OMAP: Update Makefile and Kconfig for HDMI audio arch/arm/mach-omap2/devices.c | 32 +++++++ sound/soc/omap/Kconfig | 11 +++ sound/soc/omap/Makefile | 4 + sound/soc/omap/omap-hdmi.c | 169 ++++++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-hdmi.h | 36 ++++++++ sound/soc/omap/omap4-hdmi-card.c | 100 ++++++++++++++++++++++ 6 files changed, 352 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap-hdmi.c create mode 100644 sound/soc/omap/omap-hdmi.h create mode 100644 sound/soc/omap/omap4-hdmi-card.c

4 27

[alsa-devel] [PATCH 1/4] Blackfin: Use 8bit spi transfers for the ad1836
by Lars-Peter Clausen 12 May '11

12 May '11

Currently there is a special case for the ad1836 in the ASoC generic spi write functions, which swaps the upper and the lower byte for 4/12 transfers. This was done, because the 4/12 spi write function was added for the ad1836 for which all of the users are configured to use use 16-bit transfers. In order to be able to get rid of this special casing switch all users of ad1836 to 8-bit transfers. Since 8-bit, no-dma transfers is the default configuration we can remove the bfin5xx_spi_chip struct for the ad1836 altogether. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- arch/blackfin/mach-bf527/boards/ad7160eval.c | 9 --------- arch/blackfin/mach-bf527/boards/cm_bf527.c | 9 --------- arch/blackfin/mach-bf527/boards/ezkit.c | 9 --------- arch/blackfin/mach-bf533/boards/H8606.c | 8 -------- arch/blackfin/mach-bf533/boards/cm_bf533.c | 8 -------- arch/blackfin/mach-bf533/boards/ezkit.c | 8 -------- arch/blackfin/mach-bf533/boards/stamp.c | 8 -------- arch/blackfin/mach-bf537/boards/cm_bf537e.c | 8 -------- arch/blackfin/mach-bf537/boards/cm_bf537u.c | 8 -------- arch/blackfin/mach-bf537/boards/pnav10.c | 9 --------- arch/blackfin/mach-bf537/boards/stamp.c | 9 --------- arch/blackfin/mach-bf537/boards/tcm_bf537.c | 8 -------- arch/blackfin/mach-bf548/boards/ezkit.c | 9 --------- arch/blackfin/mach-bf561/boards/cm_bf561.c | 8 -------- arch/blackfin/mach-bf561/boards/ezkit.c | 9 --------- 15 files changed, 0 insertions(+), 127 deletions(-) diff --git a/arch/blackfin/mach-bf527/boards/ad7160eval.c b/arch/blackfin/mach-bf527/boards/ad7160eval.c index ccab4c6..eb712ce 100644 --- a/arch/blackfin/mach-bf527/boards/ad7160eval.c +++ b/arch/blackfin/mach-bf527/boards/ad7160eval.c @@ -269,14 +269,6 @@ static struct bfin5xx_spi_chip spi_flash_chip_info = { }; #endif -#if defined(CONFIG_SND_BF5XX_SOC_AD183X) \ - || defined(CONFIG_SND_BF5XX_SOC_AD183X_MODULE) -static struct bfin5xx_spi_chip ad1836_spi_chip_info = { - .enable_dma = 0, - .bits_per_word = 16, -}; -#endif - #if defined(CONFIG_MMC_SPI) || defined(CONFIG_MMC_SPI_MODULE) static struct bfin5xx_spi_chip mmc_spi_chip_info = { .enable_dma = 0, @@ -328,7 +320,6 @@ static struct spi_board_info bfin_spi_board_info[] __initdata = { .max_speed_hz = 3125000, /* max spi clock (SCK) speed in HZ */ .bus_num = 0, .chip_select = 4, - .controller_data = &ad1836_spi_chip_info, }, #endif #if defined(CONFIG_MMC_SPI) || defined(CONFIG_MMC_SPI_MODULE) diff --git a/arch/blackfin/mach-bf527/boards/cm_bf527.c b/arch/blackfin/mach-bf527/boards/cm_bf527.c index c9d6dc8..ccdf2e7 100644 --- a/arch/blackfin/mach-bf527/boards/cm_bf527.c +++ b/arch/blackfin/mach-bf527/boards/cm_bf527.c @@ -367,14 +367,6 @@ static struct bfin5xx_spi_chip spi_adc_chip_info = { }; #endif -#if defined(CONFIG_SND_BF5XX_SOC_AD183X) \ - || defined(CONFIG_SND_BF5XX_SOC_AD183X_MODULE) -static struct bfin5xx_spi_chip ad1836_spi_chip_info = { - .enable_dma = 0, - .bits_per_word = 16, -}; -#endif - #if defined(CONFIG_MMC_SPI) || defined(CONFIG_MMC_SPI_MODULE) static struct bfin5xx_spi_chip mmc_spi_chip_info = { .enable_dma = 0, @@ -452,7 +444,6 @@ static struct spi_board_info bfin_spi_board_info[] __initdata = { .max_speed_hz = 3125000, /* max spi clock (SCK) speed in HZ */ .bus_num = 0, .chip_select = 4, - .controller_data = &ad1836_spi_chip_info, }, #endif #if defined(CONFIG_MMC_SPI) || defined(CONFIG_MMC_SPI_MODULE) diff --git a/arch/blackfin/mach-bf527/boards/ezkit.c b/arch/blackfin/mach-bf527/boards/ezkit.c index 2cd2ff6..503837b 100644 --- a/arch/blackfin/mach-bf527/boards/ezkit.c +++ b/arch/blackfin/mach-bf527/boards/ezkit.c @@ -460,14 +460,6 @@ static struct bfin5xx_spi_chip spi_adc_chip_info = { }; #endif -#if defined(CONFIG_SND_BF5XX_SOC_AD183X) \ - || defined(CONFIG_SND_BF5XX_SOC_AD183X_MODULE) -static struct bfin5xx_spi_chip ad1836_spi_chip_info = { - .enable_dma = 0, - .bits_per_word = 16, -}; -#endif - #if defined(CONFIG_MMC_SPI) || defined(CONFIG_MMC_SPI_MODULE) static struct bfin5xx_spi_chip mmc_spi_chip_info = { .enable_dma = 0, @@ -583,7 +575,6 @@ static struct spi_board_info bfin_spi_board_info[] __initdata = { .max_speed_hz = 3125000, /* max spi clock (SCK) speed in HZ */ .bus_num = 0, .chip_select = 4, - .controller_data = &ad1836_spi_chip_info, }, #endif #if defined(CONFIG_MMC_SPI) || defined(CONFIG_MMC_SPI_MODULE) diff --git a/arch/blackfin/mach-bf533/boards/H8606.c b/arch/blackfin/mach-bf533/boards/H8606.c index d4bfcea..41c9329 100644 --- a/arch/blackfin/mach-bf533/boards/H8606.c +++ b/arch/blackfin/mach-bf533/boards/H8606.c @@ -171,13 +171,6 @@ static struct bfin5xx_spi_chip spi_adc_chip_info = { }; #endif -#if defined(CONFIG_SND_BF5XX_SOC_AD183X) || defined(CONFIG_SND_BF5XX_SOC_AD183X_MODULE) -static struct bfin5xx_spi_chip ad1836_spi_chip_info = { - .enable_dma = 0, - .bits_per_word = 16, -}; -#endif - /* Notice: for blackfin, the speed_hz is the value of register * SPI_BAUD, not the real baudrate */ static struct spi_board_info bfin_spi_board_info[] __initdata = { @@ -212,7 +205,6 @@ static struct spi_board_info bfin_spi_board_info[] __initdata = { .max_speed_hz = 16, .bus_num = 1, .chip_select = 4, - .controller_data = &ad1836_spi_chip_info, }, #endif diff --git a/arch/blackfin/mach-bf533/boards/cm_bf533.c b/arch/blackfin/mach-bf533/boards/cm_bf533.c index 4d5604e..2506703 100644 --- a/arch/blackfin/mach-bf533/boards/cm_bf533.c +++ b/arch/blackfin/mach-bf533/boards/cm_bf533.c @@ -71,13 +71,6 @@ static struct bfin5xx_spi_chip spi_adc_chip_info = { }; #endif -#if defined(CONFIG_SND_BF5XX_SOC_AD183X) || defined(CONFIG_SND_BF5XX_SOC_AD183X_MODULE) -static struct bfin5xx_spi_chip ad1836_spi_chip_info = { - .enable_dma = 0, - .bits_per_word = 16, -}; -#endif - #if defined(CONFIG_MMC_SPI) || defined(CONFIG_MMC_SPI_MODULE) static struct bfin5xx_spi_chip mmc_spi_chip_info = { .enable_dma = 0, @@ -116,7 +109,6 @@ static struct spi_board_info bfin_spi_board_info[] __initdata = { .max_speed_hz = 3125000, /* max spi clock (SCK) speed in HZ */ .bus_num = 0, .chip_select = 4, - .controller_data = &ad1836_spi_chip_info, }, #endif diff --git a/arch/blackfin/mach-bf533/boards/ezkit.c b/arch/blackfin/mach-bf533/boards/ezkit.c index b67b91d..7d82463 100644 --- a/arch/blackfin/mach-bf533/boards/ezkit.c +++ b/arch/blackfin/mach-bf533/boards/ezkit.c @@ -222,13 +222,6 @@ static struct bfin5xx_spi_chip spi_adc_chip_info = { }; #endif -#if defined(CONFIG_SND_BF5XX_SOC_AD183X) || defined(CONFIG_SND_BF5XX_SOC_AD183X_MODULE) -static struct bfin5xx_spi_chip ad1836_spi_chip_info = { - .enable_dma = 0, - .bits_per_word = 16, -}; -#endif - #if defined(CONFIG_SPI_SPIDEV) || defined(CONFIG_SPI_SPIDEV_MODULE) static struct bfin5xx_spi_chip spidev_chip_info = { .enable_dma = 0, @@ -267,7 +260,6 @@ static struct spi_board_info bfin_spi_board_info[] __initdata = { .max_speed_hz = 3125000, /* max spi clock (SCK) speed in HZ */ .bus_num = 0, .chip_select = 4, - .controller_data = &ad1836_spi_chip_info, }, #endif #if defined(CONFIG_SPI_SPIDEV) || defined(CONFIG_SPI_SPIDEV_MODULE) diff --git a/arch/blackfin/mach-bf533/boards/stamp.c b/arch/blackfin/mach-bf533/boards/stamp.c index 43224ef..32091b8 100644 --- a/arch/blackfin/mach-bf533/boards/stamp.c +++ b/arch/blackfin/mach-bf533/boards/stamp.c @@ -184,13 +184,6 @@ static struct bfin5xx_spi_chip spi_adc_chip_info = { }; #endif -#if defined(CONFIG_SND_BF5XX_SOC_AD183X) || defined(CONFIG_SND_BF5XX_SOC_AD183X_MODULE) -static struct bfin5xx_spi_chip ad1836_spi_chip_info = { - .enable_dma = 0, - .bits_per_word = 16, -}; -#endif - #if defined(CONFIG_SPI_SPIDEV) || defined(CONFIG_SPI_SPIDEV_MODULE) static struct bfin5xx_spi_chip spidev_chip_info = { .enable_dma = 0, @@ -258,7 +251,6 @@ static struct spi_board_info bfin_spi_board_info[] __initdata = { .bus_num = 0, .chip_select = 4, .platform_data = "ad1836", /* only includes chip name for the moment */ - .controller_data = &ad1836_spi_chip_info, .mode = SPI_MODE_3, }, #endif diff --git a/arch/blackfin/mach-bf537/boards/cm_bf537e.c b/arch/blackfin/mach-bf537/boards/cm_bf537e.c index d582b81..cb0a34e 100644 --- a/arch/blackfin/mach-bf537/boards/cm_bf537e.c +++ b/arch/blackfin/mach-bf537/boards/cm_bf537e.c @@ -73,13 +73,6 @@ static struct bfin5xx_spi_chip spi_adc_chip_info = { }; #endif -#if defined(CONFIG_SND_BF5XX_SOC_AD183X) || defined(CONFIG_SND_BF5XX_SOC_AD183X_MODULE) -static struct bfin5xx_spi_chip ad1836_spi_chip_info = { - .enable_dma = 0, - .bits_per_word = 16, -}; -#endif - #if defined(CONFIG_MMC_SPI) || defined(CONFIG_MMC_SPI_MODULE) static struct bfin5xx_spi_chip mmc_spi_chip_info = { .enable_dma = 0, @@ -118,7 +111,6 @@ static struct spi_board_info bfin_spi_board_info[] __initdata = { .max_speed_hz = 3125000, /* max spi clock (SCK) speed in HZ */ .bus_num = 0, .chip_select = 4, - .controller_data = &ad1836_spi_chip_info, }, #endif diff --git a/arch/blackfin/mach-bf537/boards/cm_bf537u.c b/arch/blackfin/mach-bf537/boards/cm_bf537u.c index cbb8098..197a77c 100644 --- a/arch/blackfin/mach-bf537/boards/cm_bf537u.c +++ b/arch/blackfin/mach-bf537/boards/cm_bf537u.c @@ -74,13 +74,6 @@ static struct bfin5xx_spi_chip spi_adc_chip_info = { }; #endif -#if defined(CONFIG_SND_BF5XX_SOC_AD183X) || defined(CONFIG_SND_BF5XX_SOC_AD183X_MODULE) -static struct bfin5xx_spi_chip ad1836_spi_chip_info = { - .enable_dma = 0, - .bits_per_word = 16, -}; -#endif - #if defined(CONFIG_MMC_SPI) || defined(CONFIG_MMC_SPI_MODULE) static struct bfin5xx_spi_chip mmc_spi_chip_info = { .enable_dma = 0, @@ -119,7 +112,6 @@ static struct spi_board_info bfin_spi_board_info[] __initdata = { .max_speed_hz = 3125000, /* max spi clock (SCK) speed in HZ */ .bus_num = 0, .chip_select = 4, - .controller_data = &ad1836_spi_chip_info, }, #endif diff --git a/arch/blackfin/mach-bf537/boards/pnav10.c b/arch/blackfin/mach-bf537/boards/pnav10.c index 9389f03..7037128 100644 --- a/arch/blackfin/mach-bf537/boards/pnav10.c +++ b/arch/blackfin/mach-bf537/boards/pnav10.c @@ -197,14 +197,6 @@ static struct bfin5xx_spi_chip spi_adc_chip_info = { }; #endif -#if defined(CONFIG_SND_BF5XX_SOC_AD183X) \ - || defined(CONFIG_SND_BF5XX_SOC_AD183X_MODULE) -static struct bfin5xx_spi_chip ad1836_spi_chip_info = { - .enable_dma = 0, - .bits_per_word = 16, -}; -#endif - #if defined(CONFIG_MMC_SPI) || defined(CONFIG_MMC_SPI_MODULE) static struct bfin5xx_spi_chip mmc_spi_chip_info = { .enable_dma = 0, @@ -267,7 +259,6 @@ static struct spi_board_info bfin_spi_board_info[] __initdata = { .max_speed_hz = 3125000, /* max spi clock (SCK) speed in HZ */ .bus_num = 0, .chip_select = 4, - .controller_data = &ad1836_spi_chip_info, }, #endif #if defined(CONFIG_MMC_SPI) || defined(CONFIG_MMC_SPI_MODULE) diff --git a/arch/blackfin/mach-bf537/boards/stamp.c b/arch/blackfin/mach-bf537/boards/stamp.c index 3fa3354..343f969 100644 --- a/arch/blackfin/mach-bf537/boards/stamp.c +++ b/arch/blackfin/mach-bf537/boards/stamp.c @@ -549,14 +549,6 @@ static struct bfin5xx_spi_chip spi_adc_chip_info = { }; #endif -#if defined(CONFIG_SND_BF5XX_SOC_AD183X) \ - || defined(CONFIG_SND_BF5XX_SOC_AD183X_MODULE) -static struct bfin5xx_spi_chip ad1836_spi_chip_info = { - .enable_dma = 0, - .bits_per_word = 16, -}; -#endif - #if defined(CONFIG_SND_BF5XX_SOC_AD193X) \ || defined(CONFIG_SND_BF5XX_SOC_AD193X_MODULE) static struct bfin5xx_spi_chip ad1938_spi_chip_info = { @@ -1076,7 +1068,6 @@ static struct spi_board_info bfin_spi_board_info[] __initdata = { .bus_num = 0, .chip_select = 4, .platform_data = "ad1836", /* only includes chip name for the moment */ - .controller_data = &ad1836_spi_chip_info, .mode = SPI_MODE_3, }, #endif diff --git a/arch/blackfin/mach-bf537/boards/tcm_bf537.c b/arch/blackfin/mach-bf537/boards/tcm_bf537.c index 164a7e0..b720fec 100644 --- a/arch/blackfin/mach-bf537/boards/tcm_bf537.c +++ b/arch/blackfin/mach-bf537/boards/tcm_bf537.c @@ -74,13 +74,6 @@ static struct bfin5xx_spi_chip spi_adc_chip_info = { }; #endif -#if defined(CONFIG_SND_BF5XX_SOC_AD183X) || defined(CONFIG_SND_BF5XX_SOC_AD183X_MODULE) -static struct bfin5xx_spi_chip ad1836_spi_chip_info = { - .enable_dma = 0, - .bits_per_word = 16, -}; -#endif - #if defined(CONFIG_MMC_SPI) || defined(CONFIG_MMC_SPI_MODULE) static struct bfin5xx_spi_chip mmc_spi_chip_info = { .enable_dma = 0, @@ -119,7 +112,6 @@ static struct spi_board_info bfin_spi_board_info[] __initdata = { .max_speed_hz = 3125000, /* max spi clock (SCK) speed in HZ */ .bus_num = 0, .chip_select = 4, - .controller_data = &ad1836_spi_chip_info, }, #endif diff --git a/arch/blackfin/mach-bf548/boards/ezkit.c b/arch/blackfin/mach-bf548/boards/ezkit.c index 93e19a5..002e0f9 100644 --- a/arch/blackfin/mach-bf548/boards/ezkit.c +++ b/arch/blackfin/mach-bf548/boards/ezkit.c @@ -1013,14 +1013,6 @@ static struct bfin5xx_spi_chip spi_flash_chip_info = { }; #endif -#if defined(CONFIG_SND_BF5XX_SOC_AD183X) \ - || defined(CONFIG_SND_BF5XX_SOC_AD183X_MODULE) -static struct bfin5xx_spi_chip ad1836_spi_chip_info = { - .enable_dma = 0, - .bits_per_word = 16, -}; -#endif - #if defined(CONFIG_TOUCHSCREEN_AD7877) || defined(CONFIG_TOUCHSCREEN_AD7877_MODULE) static struct bfin5xx_spi_chip spi_ad7877_chip_info = { .enable_dma = 0, @@ -1077,7 +1069,6 @@ static struct spi_board_info bfin_spi_board_info[] __initdata = { .max_speed_hz = 3125000, /* max spi clock (SCK) speed in HZ */ .bus_num = 1, .chip_select = 4, - .controller_data = &ad1836_spi_chip_info, }, #endif #if defined(CONFIG_TOUCHSCREEN_AD7877) || defined(CONFIG_TOUCHSCREEN_AD7877_MODULE) diff --git a/arch/blackfin/mach-bf561/boards/cm_bf561.c b/arch/blackfin/mach-bf561/boards/cm_bf561.c index 87595cd..1418053 100644 --- a/arch/blackfin/mach-bf561/boards/cm_bf561.c +++ b/arch/blackfin/mach-bf561/boards/cm_bf561.c @@ -72,13 +72,6 @@ static struct bfin5xx_spi_chip spi_adc_chip_info = { }; #endif -#if defined(CONFIG_SND_BF5XX_SOC_AD183X) || defined(CONFIG_SND_BF5XX_SOC_AD183X_MODULE) -static struct bfin5xx_spi_chip ad1836_spi_chip_info = { - .enable_dma = 0, - .bits_per_word = 16, -}; -#endif - #if defined(CONFIG_MMC_SPI) || defined(CONFIG_MMC_SPI_MODULE) static struct bfin5xx_spi_chip mmc_spi_chip_info = { .enable_dma = 0, @@ -117,7 +110,6 @@ static struct spi_board_info bfin_spi_board_info[] __initdata = { .max_speed_hz = 3125000, /* max spi clock (SCK) speed in HZ */ .bus_num = 0, .chip_select = 4, - .controller_data = &ad1836_spi_chip_info, }, #endif #if defined(CONFIG_MMC_SPI) || defined(CONFIG_MMC_SPI_MODULE) diff --git a/arch/blackfin/mach-bf561/boards/ezkit.c b/arch/blackfin/mach-bf561/boards/ezkit.c index f667e77..9f58605 100644 --- a/arch/blackfin/mach-bf561/boards/ezkit.c +++ b/arch/blackfin/mach-bf561/boards/ezkit.c @@ -275,14 +275,6 @@ static struct platform_device ezkit_flash_device = { }; #endif -#if defined(CONFIG_SND_BF5XX_SOC_AD183X) \ - || defined(CONFIG_SND_BF5XX_SOC_AD183X_MODULE) -static struct bfin5xx_spi_chip ad1836_spi_chip_info = { - .enable_dma = 0, - .bits_per_word = 16, -}; -#endif - #if defined(CONFIG_SPI_SPIDEV) || defined(CONFIG_SPI_SPIDEV_MODULE) static struct bfin5xx_spi_chip spidev_chip_info = { .enable_dma = 0, @@ -337,7 +329,6 @@ static struct spi_board_info bfin_spi_board_info[] __initdata = { .bus_num = 0, .chip_select = 4, .platform_data = "ad1836", /* only includes chip name for the moment */ - .controller_data = &ad1836_spi_chip_info, .mode = SPI_MODE_3, }, #endif -- 1.7.2.5

4 11

[alsa-devel] hdspmixer/ubuntu
by Tim Blechmann 12 May '11

12 May '11

hi all, the current version of ubuntu ships alsa-tools without hdspmixer [1], probably because of the problems of older versions with newer kernels (iirc a hardcoded preprocessor definition made the old version unusable). while it is still possible to compile a working hdspmixer from sources, this is quite inconvenient (especially for newbies). questions: - is there a plan to release a 1.0.24.2, including the hdspmixer fixes? - gitcompile currently does not work on ubuntu, but requires this patch [2] in order to run the autotools. according to the name, it is already required for older versions - compiling ld10k1 fails with: make[1]: Entering directory `/home/tim/workspace/alsa-tools/ld10k1' Making all in setup make[2]: Entering directory `/home/tim/workspace/alsa-tools/ld10k1/setup' Making all in effects make[3]: Entering directory `/home/tim/workspace/alsa- tools/ld10k1/setup/effects' make[3]: *** No rule to make target `simple.emu10k1', needed by `all-am'. Stop. is this a problem on my side? thanks, tim [1] https://bugs.launchpad.net/ubuntu/+source/alsa-tools/+bug/723276 [2] http://svnweb.mageia.org/packages/cauldron/alsa-tools/current/SOURCES/alsa- tools-1.0.11-gtk-buildfix.patch?view=log&pathrev=1611 -- tim(a)klingt.org http://tim.klingt.org A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? A: Top-posting. Q: What is the most annoying thing in e-mail?

2 2

[alsa-devel] [PATCH] ASoC: Reintroduce do_spi_write()
by Mark Brown 12 May '11

12 May '11

There is an unfortunate difference in return values between spi_write() and i2c_master_send() so we need an adaptor function to translate. Reported-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 16 +++++++++++++++- 1 files changed, 15 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 04861c4..dee86fc 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -20,6 +20,20 @@ #include <trace/events/asoc.h> +#ifdef CONFIG_SPI_MASTER +static int do_spi_write(void *control, const char *data, int len) +{ + struct spi_device *spi = control; + int ret; + + ret = spi_write(spi, data, len); + if (ret < 0) + return ret; + + return len; +} +#endif + static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value, const void *data, int len) { @@ -409,7 +423,7 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, case SND_SOC_SPI: #ifdef CONFIG_SPI_MASTER - codec->hw_write = (hw_write_t)spi_write; + codec->hw_write = do_spi_write; #endif codec->control_data = container_of(codec->dev, -- 1.7.5.1

3 3

[alsa-devel] [PATCH] Enable E-mu 0204
by Joseph Teichman 12 May '11

12 May '11

This is a trivial patch to enable the E-mu 0204, which is the newest device in the series 0202USB, 0404USB and Tracker Pre. Signed-off-by: Joseph Teichman <josteich at gmail.com> --- alsa-kmirror/usb/mixer.c 2011-01-27 07:09:20.000000000 -0500 +++ alsa-kmirror-jt/usb/mixer.c 2011-02-05 20:42:00.000000000 -0500 @@ -1566,7 +1566,7 @@ cval->initialized = 1; } else { if (type == USB_XU_CLOCK_RATE) { - /* E-Mu USB 0404/0202/TrackerPre + /* E-Mu USB 0404/0202/TrackerPre/0204 * samplerate control quirk */ cval->min = 0; diff -u ./quirks.c /home/joe/alsa-driver-1.0.24-jt/sound/usb/quirks.c --- ./quirks.c 2011-01-27 07:09:20.000000000 -0500 +++ /home/joe/alsa-driver-1.0.24-jt/sound/usb/quirks.c 2011-02-05 20:40:28.000000000 -0500 @@ -532,7 +532,7 @@ } /* - * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device, + * For E-Mu 0404USB/0202USB/TrackerPre/0204USB sample rate should be set for device, * not for interface. */ @@ -589,6 +589,7 @@ case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ + case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; } diff -u ./quirks-table.h /home/joe/alsa-driver-1.0.24-jt/sound/usb/quirks-table.h --- ./quirks-table.h 2011-01-27 07:09:20.000000000 -0500 +++ /home/joe/alsa-driver-1.0.24-jt/sound/usb/quirks-table.h 2011-02-05 20:41:02.000000000 -0500 @@ -79,6 +79,13 @@ .idProduct = 0x3f0a, .bInterfaceClass = USB_CLASS_AUDIO, }, +{ + /* E-Mu 0204 USB */ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x041e, + .idProduct = 0x3f19, + .bInterfaceClass = USB_CLASS_AUDIO, +}, /* * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface

3 2

[alsa-devel] Has hdspmixer been dropped from alsa-tools?
by Lars Zobbe 12 May '11

12 May '11

Hi. Just upgraded to ubuntu 11.04 and went through the usual process with getting HDSP 9632 to work. Everything seems to be fine, modules loaded and stuff. But I can't find the hdspmixer. Normally it's included in the alsa-tools and alsa-tools-gui packages. And actually the mixer GUI does get an entry in the menu. But there is no command for it to execute. It is supposed to run hdspmixer - but it's not there. If I write "hdsp" and try to autocomplete with tab in the terminal, I can now only se hdsploader and hdspconf (both seem to run just fine btw.). Without the alsamixer I won't get any sound. Is the error in the ubuntu natty packaging or did hdspmixer get lost? Is there any way I can install the hdspmixer tool? Would it break anything if I just snatched an old version fx from ubuntu 10.04 repositories (where it IS present)? Some info: 00-00: RME Hammerfall HDSP 9632 : RME Hammerfall HDSP 9632 : playback 1 : capt _From *lspci* 02:09.0 Multimedia audio controller: Xilinx Corporation RME Hammerfall DSP (rev 9a) _From *lshw*: description: Multimedia audio controller product: RME Hammerfall DSP vendor: Xilinx Corporation physical id: 9 bus info: pci@0000:02:09.0 version: 9a width: 32 bits clock: 33MHz capabilities: bus_master configuration: driver=RME Hammerfall DSP latency=255 resources: irq:19 memory:fdff0000-fdffffff *uname -a:* Linux lars-desktop 2.6.38-8-generic #42-Ubuntu SMP Mon Apr 11 03:31:50 UTC 2011 i686 athlon i386 GNU/Linux *Version of alsa-tools*: *** 1.0.24.1-0ubuntu1 0 *Version of alsa-firmware-loaders*: *** 1.0.24.1-0ubuntu1 0 *cat /proc/asound/version * Advanced Linux Sound Architecture Driver Version 1.0.23. *cat /proc/asound/cards* 0 [DSP ]: H-DSP - Hammerfall DSP RME Hammerfall HDSP 9632 at 0xfdff0000, irq 19 *cat /proc/asound/modules * 0 snd_hdsp *cat /proc/asound/DSP/hdsp * RME Hammerfall HDSP 9632 (Card #1) Buffers: capture f2400000 playback f2600000 IRQ: 19 Registers bus: 0xfdff0000 VM: 0xf819c000 Control register: 0x300409a Control2 register: 0x0 Status register: 0x32000000 Status2 register: 0xffff8701 FIFO status: 0 MIDI1 Output status: 0xffffff00 MIDI1 Input status: 0xffffff00 MIDI2 Output status: 0xffffff00 MIDI2 Input status: 0xffffff00 Use Midi Tasklet: on Buffer Size (Latency): 2048 samples (2 periods of 8192 bytes) Hardware pointer (frames): 0 Precise pointer: off Line out: on Firmware version: 1 Sample Clock Source: Internal 44.1 kHz Preferred Sync Reference: ADAT1 AutoSync Reference: None AutoSync Frequency: 0 System Clock Mode: Master System Clock Frequency: 44100 System Clock Locked: No IEC958 input: Coaxial IEC958 output: Coaxial only IEC958 quality: Consumer IEC958 emphasis: off IEC958 NonAudio: off IEC958 sample rate: Error flag set ADAT1: No Lock SPDIF: No Lock Word Clock: No Lock ADAT Sync: No Lock AD Gain : +4 dBu DA Gain : -10 dBV Phones Gain : -12 dB XLR Breakout Cable : no AEB : off (ADAT1 external) *modinfo soundcore* filename: /lib/modules/2.6.38-8-generic/kernel/sound/soundcore.ko license: GPL author: Alan Cox description: Core sound module srcversion: 982D4ED4F640D0B9C43B130 depends: vermagic: 2.6.38-8-generic SMP mod_unload modversions 686 *lsmod | grep snd* snd_hdsp 53247 0 snd_pcm 80244 1 snd_hdsp snd_page_alloc 14073 2 snd_hdsp,snd_pcm snd_hwdep 13274 1 snd_hdsp snd_seq_midi 13132 0 snd_rawmidi 25269 2 snd_hdsp,snd_seq_midi snd_seq_midi_event 14475 1 snd_seq_midi snd_seq 51291 2 snd_seq_midi,snd_seq_midi_event snd_timer 28659 2 snd_pcm,snd_seq snd_seq_device 14110 3 snd_seq_midi,snd_rawmidi,snd_seq snd 55295 7 snd_hdsp,snd_pcm,snd_hwdep,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 12600 1 snd Sincerly Lars Zobbe.

2 2

[alsa-devel] [PATCH 0/3] ASoC: codecs: max98088: bug fixes and changes
by jinyoungp＠nvidia.com 12 May '11

12 May '11

From: Jin Park <jinyoungp(a)nvidia.com> Recently, I working on max98088 and changed something in max98088 asoc driver. The MAX98088 driver had some bug and needed modifies. The changes was confirmed by Peter Hsiang who author of max98088 driver and maxim engineer. Please review my changes. Thanks, Jin Park. Jin Park (3): ASoC: codecs: max98088: Fixed invalid register definitions in mixer controls ASoC: codecs: max98088: Moved the EX Limiter Mode from dapm widget to control ASoC: codecs: max98088: Added digital mute function in DAI1 and DAI2 sound/soc/codecs/max98088.c | 62 +++++++++++++++++++++++++++++++------------ sound/soc/codecs/max98088.h | 13 +++++++++ 2 files changed, 58 insertions(+), 17 deletions(-) ----------------------------------------------------------------------------------- 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 1/2] ASoC: Remove byte swap in 4x12 SPI write
by Mark Brown 11 May '11

11 May '11

snd_soc_4_12_spi_write() contains a byte swap. Since this code was written for an Analog CODEC on a Blackfin reference board it appears that this is done because while Blackfin is little endian the CODEC is big endian (as are most CODECs). Push this up into the generic 4x12 write function and use cpu_to_be16() to do the byte swap so things are more regular and things work on both CPU endiannesses. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 11 +++++------ 1 files changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 6e5e30a..73907e5 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -101,12 +101,11 @@ static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - u8 data[2]; + u16 data; - data[0] = (reg << 4) | ((value >> 8) & 0x000f); - data[1] = value & 0x00ff; + data = cpu_to_be16((reg << 12) | (value & 0xffffff)); - return do_hw_write(codec, reg, value, data, 2); + return do_hw_write(codec, reg, value, &data, 2); } #if defined(CONFIG_SPI_MASTER) @@ -115,8 +114,8 @@ static int snd_soc_4_12_spi_write(void *control_data, const char *data, { u8 msg[2]; - msg[0] = data[1]; - msg[1] = data[0]; + msg[0] = data[0]; + msg[1] = data[1]; return do_spi_write(control_data, msg, len); } -- 1.7.5.1

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[alsa-devel] how to link multi plugin to dsnoop ?
by Vincent Meserette 11 May '11

11 May '11

Hi all, I would like to allow multiple applications to access the audio inputs and outputs of my system at same time. I succeded to make it for the audio outputs by using the dmix plugin, but I don't know how to do it with the dsnoop plugin because it seems I cannot link directly the multi plugin to dsnoop. Here is the error message I have with the /etc/asound.conf file presented below : arecord -D"alduplex" -c4 -r48000 -fS16_LE /tmp/test.wav ALSA lib pcm_direct.c:1490:(_snd_pcm_direct_get_slave_ipc_offset) Invalid type 'multi' for slave PCM arecord: main:654: audio open error: Invalid argument Does someone know how can I route the multi plugin to dsnoop ? Thanks in advance My /etc/asound.conf file : #output device pcm.AD1989A_outputs{ type hw card 0 } pcm.aldmix{ type dmix ipc_key 1024 ipc_perm 0666 slave{ pcm AD1989A_outputs } } #input device pcm.4channelsDeinterleaved { type multi slaves.a.pcm "hw:0,0,0" slaves.a.channels 2 slaves.b.pcm "hw:0,0,2" slaves.b.channels 2 bindings.0.slave a bindings.0.channel 0 bindings.1.slave a bindings.1.channel 1 bindings.2.slave b bindings.2.channel 0 bindings.3.slave b bindings.3.channel 1 } pcm.AD1989A_inputs { type route slave.pcm "4channelsDeinterleaved" ttable.0.0 1 ttable.1.1 1 ttable.2.2 1 ttable.3.3 1 } #4 channels dsnoop (plug-in for capture) pcm.aldsnoop{ type dsnoop ipc_key 1025 ipc_perm 0666 slave{ pcm AD1989A_inputs rate 48000 channels 4 } } pcm.alduplex { type asym capture.pcm "aldsnoop" playback.pcm "aldmix" } ### CTL DEVICES (controls) ctl.mixer0 { type hw card 0 }

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[alsa-devel] ASoC update for 2.6.39
by Mark Brown 11 May '11

11 May '11

The following changes since commit 1209842af4db98ffd7364ec9cf7d1a59293aa74c: Merge branch 'for-2.6.40' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2011-05-10 09:24:50 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 Stephen Warren (1): ASoC: WM8903: Fix Digital Capture Volume range sound/soc/codecs/wm8903.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

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[alsa-devel] ASoC update for 2.6.40
by Mark Brown 11 May '11

11 May '11

The following changes since commit 1209842af4db98ffd7364ec9cf7d1a59293aa74c: Merge branch 'for-2.6.40' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2011-05-10 09:24:50 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.40 Mark Brown (1): Merge branch 'for-2.6.39' into for-2.6.40 Stephen Warren (1): ASoC: WM8903: Fix Digital Capture Volume range sound/soc/codecs/wm8903.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

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[alsa-devel] [PATCH] ASoC: Disable WM8994/58 microphone detection over suspend
by Mark Brown 11 May '11

11 May '11

It will be non-functional with the basises and clocks off anyway, if the system needs microphone detection enabled over suspend then it should be causing the CODEC to ignore suspend using the APIs for that to prevent the biases being disabled. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 25 +++++++++++++++++++++++++ 1 files changed, 25 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b6d47e7..e6dfa10 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2416,8 +2416,19 @@ static struct snd_soc_dai_driver wm8994_dai[] = { static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = codec->control_data; int i, ret; + switch (control->type) { + case WM8994: + snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0); + break; + case WM8958: + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, + WM8958_MICD_ENA, 0); + break; + } + for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i], sizeof(struct wm8994_fll_config)); @@ -2435,6 +2446,7 @@ static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8994_resume(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = codec->control_data; int i, ret; unsigned int val, mask; @@ -2473,6 +2485,19 @@ static int wm8994_resume(struct snd_soc_codec *codec) i + 1, ret); } + switch (control->type) { + case WM8994: + if (wm8994->micdet[0].jack || wm8994->micdet[1].jack) + snd_soc_update_bits(codec, WM8994_MICBIAS, + WM8994_MICD_ENA, WM8994_MICD_ENA); + break; + case WM8958: + if (wm8994->jack_cb) + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, + WM8958_MICD_ENA, WM8958_MICD_ENA); + break; + } + return 0; } #else -- 1.7.5.1

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[alsa-devel] [PATCH] ASoC: Don't squash 16x8 registers down to 8 bits
by Mark Brown 11 May '11

11 May '11

Currently we'll force all registers to fit in 8 bits before passing down to the I/O function. Looks like a cut'n'paste bug. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index bb8e446..40548ce 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -232,7 +232,6 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, data[0] = (reg >> 8) & 0xff; data[1] = reg & 0xff; data[2] = value; - reg &= 0xff; return do_hw_write(codec, reg, value, data, 3); } -- 1.7.5.1

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[alsa-devel] [PATCH] ASoC: Fix NULL vs. 0 warning in SSM2602
by Mark Brown 11 May '11

11 May '11

sparse complains if 0 is used as a NULL pointer constant. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/ssm2602.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 70099c9..84f4ad5 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -137,7 +137,7 @@ SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1), SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1), SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0), -SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, 0, 0), +SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("LOUT"), SND_SOC_DAPM_OUTPUT("ROUT"), -- 1.7.5.1

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[alsa-devel] a xrun() query
by Santosh Sivaraj 11 May '11

11 May '11

Hi, I am getting glitches while playing audio with aplay. With little debugging I figured out that the state changes from RUNNING to XRUN at which time the trigger gets called. I also figured out that XRUN state is set in the function: static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime) { <snip> } else { if (avail >= runtime->stop_threshold) { xrun(substream); return -EPIPE; } } </snip> } I am not able to find out why the condition (avail > runtime->stop_threshold) becomes true. Can the experts here tell during what use cases/scenarios this happens? Thanks, Santosh -- http://fossix.org, http://dot.slashb.in Luck is a matter of preparation meeting the opportunity. -- Anon

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[alsa-devel] [PATCH 1/7] ASoC: Add control_type in snd_soc_codec
by Dimitris Papastamos 11 May '11

11 May '11

This is mainly used by the soc-cache code to easily determine the currently used underlying serial bus. Set SND_SOC_CUSTOM to 1 so we can distinguish it if it is not initialized or set. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index bfa4836..abb56ef 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -248,7 +248,7 @@ typedef int (*hw_write_t)(void *,const char* ,int); extern struct snd_ac97_bus_ops soc_ac97_ops; enum snd_soc_control_type { - SND_SOC_CUSTOM, + SND_SOC_CUSTOM = 1, SND_SOC_I2C, SND_SOC_SPI, }; @@ -539,6 +539,7 @@ struct snd_soc_codec { /* codec IO */ void *control_data; /* codec control (i2c/3wire) data */ + enum snd_soc_control_type control_type; hw_write_t hw_write; unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); -- 1.7.4.1

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[alsa-devel] fm801: implement TEA575x tuner autodetection
by Ondrej Zary 11 May '11

11 May '11

Autodetect TEA575x tuner connection type during init. This allows tuner to work out-of-the box. tea575x_tuner module parameter remains functional to force tuner type. Tested with SF256-PCP and SF64-PCR. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-2.6.39-rc2-/sound/pci/fm801.c 2011-05-10 22:31:45.000000000 +0200 +++ linux-2.6.39-rc2/sound/pci/fm801.c 2011-05-10 23:21:42.000000000 +0200 @@ -53,7 +53,7 @@ static int enable[SNDRV_CARDS] = SNDRV_D /* * Enable TEA575x tuner * 1 = MediaForte 256-PCS - * 2 = MediaForte 256-PCPR + * 2 = MediaForte 256-PCP * 3 = MediaForte 64-PCR * 16 = setup tuner only (this is additional bit), i.e. SF64-PCR FM card * High 16-bits are video (radio) device number + 1 @@ -67,7 +67,7 @@ MODULE_PARM_DESC(id, "ID string for the module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable FM801 soundcard."); module_param_array(tea575x_tuner, int, NULL, 0444); -MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (1 = SF256-PCS, 2=SF256-PCPR, 3=SF64-PCR, +16=tuner-only)."); +MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (0 = auto, 1 = SF256-PCS, 2=SF256-PCP, 3=SF64-PCR, 8=disable, +16=tuner-only)."); #define TUNER_ONLY (1<<4) #define TUNER_TYPE_MASK (~TUNER_ONLY & 0xFFFF) @@ -720,12 +720,13 @@ static int __devinit snd_fm801_pcm(struc /* GPIO to TEA575x maps */ struct snd_fm801_tea575x_gpio { u8 data, clk, wren, most; + char *name; }; static struct snd_fm801_tea575x_gpio snd_fm801_tea575x_gpios[] = { - { .data = 1, .clk = 3, .wren = 2, .most = 0 }, /* SF256-PCS */ - { .data = 1, .clk = 0, .wren = 2, .most = 3 }, /* SF256-PCP */ - { .data = 2, .clk = 0, .wren = 1, .most = 3 }, /* SF64-PCR */ + { .data = 1, .clk = 3, .wren = 2, .most = 0, .name = "SF256-PCS" }, + { .data = 1, .clk = 0, .wren = 2, .most = 3, .name = "SF256-PCP" }, + { .data = 2, .clk = 0, .wren = 1, .most = 3, .name = "SF64-PCR" }, }; static void snd_fm801_tea575x_set_pins(struct snd_tea575x *tea, u8 pins) @@ -1229,14 +1230,24 @@ static int __devinit snd_fm801_create(st snd_card_set_dev(card, &pci->dev); #ifdef TEA575X_RADIO + chip->tea.card = card; + chip->tea.freq_fixup = 10700; + chip->tea.private_data = chip; + chip->tea.ops = &snd_fm801_tea_ops; if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 && (tea575x_tuner & TUNER_TYPE_MASK) < 4) { - chip->tea.card = card; - chip->tea.freq_fixup = 10700; - chip->tea.private_data = chip; - chip->tea.ops = &snd_fm801_tea_ops; - snd_tea575x_init(&chip->tea); - } + if (snd_tea575x_init(&chip->tea)) + snd_printk(KERN_ERR "TEA575x radio not found\n"); + } else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) + /* autodetect tuner connection */ + for (tea575x_tuner = 1; tea575x_tuner <= 3; tea575x_tuner++) { + chip->tea575x_tuner = tea575x_tuner; + if (!snd_tea575x_init(&chip->tea)) { + snd_printk(KERN_INFO "detected TEA575x radio type %s\n", + snd_fm801_tea575x_gpios[tea575x_tuner - 1].name); + break; + } + } #endif *rchip = chip; -- Ondrej Zary

2 1

[alsa-devel] [PATCH] ASoC: Update cx20442 for TTY API change
by Mark Brown 11 May '11

11 May '11

receive_buf() was recently changed to return the number of bytes received but the cx20442 driver wasn't updated to match the new API. I don't have any hardware but since we don't actually appears to be listening to the data at all just report that we accepted all the data that was offered to us. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cx20442.c | 8 +++++--- 1 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index d68ea53..f8c663d 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -262,14 +262,14 @@ static int v253_hangup(struct tty_struct *tty) } /* Line discipline .receive_buf() */ -static void v253_receive(struct tty_struct *tty, - const unsigned char *cp, char *fp, int count) +static unsigned int v253_receive(struct tty_struct *tty, + const unsigned char *cp, char *fp, int count) { struct snd_soc_codec *codec = tty->disc_data; struct cx20442_priv *cx20442; if (!codec) - return; + return count; cx20442 = snd_soc_codec_get_drvdata(codec); @@ -281,6 +281,8 @@ static void v253_receive(struct tty_struct *tty, codec->hw_write = (hw_write_t)tty->ops->write; codec->card->pop_time = 1; } + + return count; } /* Line discipline .write_wakeup() */ -- 1.7.5.1

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[alsa-devel] ASoC: codecs: max98088: bug fixes and changes
by Jinyoung Park 11 May '11

11 May '11

Dear Mark, Recently, I working on max98088 and changed something in max98088 asoc driver. I included three changes for max98088 in this mail. - 1/3 Fixed invalid register definitions in mixer controls * This change was confirmed by Peter Hsiang who author of max98088 driver and maxim engineer. - 2/3 Moved the EX Limiter Mode from dapm widget to control - 3/3 Added digital mute function in DAI1 and DAI2 Please review below my changes. ------------------------------------------------------------------------------------------------------------------------------

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[alsa-devel] [PATCH v2 0/5] ASoC: Update e-mail address
by Peter Ujfalusi 11 May '11

11 May '11

Hello, Update my e-mail address due to change in employment. Changes since v1: - Replace the no longer valid Nokia mail address Peter --- *** BLURB HERE *** Peter Ujfalusi (5): ASoC: tlv320dac33: Update e-mail address ASoC: tpa6130a2: Update e-mail address ASoC: omap-mcbsp: Update e-mail address ASoC: omap-pcm: Update e-mail address ASoC: RX51: Update e-mail address include/sound/tlv320dac33-plat.h | 2 +- include/sound/tpa6130a2-plat.h | 2 +- sound/soc/codecs/tlv320dac33.c | 4 ++-- sound/soc/codecs/tlv320dac33.h | 2 +- sound/soc/codecs/tpa6130a2.c | 4 ++-- sound/soc/codecs/tpa6130a2.h | 2 +- sound/soc/omap/omap-mcbsp.c | 2 +- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/omap-pcm.c | 2 +- sound/soc/omap/omap-pcm.h | 2 +- sound/soc/omap/rx51.c | 2 +- 11 files changed, 13 insertions(+), 13 deletions(-) -- 1.7.5.rc3

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[PATCH 3/3] ASoC: codecs: max98088: Added digital mute function =
by Jin Park 11 May '11

11 May '11

in DAI1 and DAI2 Added digital mute function in DAI1 and DAI2. Signed-off-by: Jin Park <jinyoungp(a)nvidia.com> --- sound/soc/codecs/max98088.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/codecs/max98088.h | 13 +++++++++++++ 2 files changed, 45 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index c520093..b5ccf37 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1582,6 +1582,36 @@ static int max98088_dai2_set_fmt(struct snd_soc_da= i *codec_dai, =20 return 0; } +static int max98088_dai1_digital_mute(struct snd_soc_dai *codec_dai, int= =20mute) +{ + struct snd_soc_codec *codec =3D codec_dai->codec; + int reg; + + if (mute) + reg =3D M98088_DAI_MUTE; + else + reg =3D 0; + + snd_soc_update_bits(codec, M98088_REG_2F_LVL_DAI1_PLAY, + M98088_DAI_MUTE_MASK, reg); + return 0; +} + +static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int= =20mute) +{ + struct snd_soc_codec *codec =3D codec_dai->codec; + int reg; + + if (mute) + reg =3D M98088_DAI_MUTE; + else + reg =3D 0; + + snd_soc_update_bits(codec, M98088_REG_31_LVL_DAI2_PLAY, + M98088_DAI_MUTE_MASK, reg); + return 0; +} + static void max98088_sync_cache(struct snd_soc_codec *codec) { =20 u16 *reg_cache =3D codec->reg_cache; @@ -1643,12 +1673,14 @@ static struct snd_soc_dai_ops max98088_dai1_ops =3D= =20{ =20 .set_sysclk =3D max98088_dai_set_sysclk, =20 .set_fmt =3D max98088_dai1_set_fmt, =20 .hw_params =3D max98088_dai1_hw_params, + .digital_mute =3D max98088_dai1_digital_mute, }; =20static struct snd_soc_dai_ops max98088_dai2_ops =3D { =20 .set_sysclk =3D max98088_dai_set_sysclk, =20 .set_fmt =3D max98088_dai2_set_fmt, =20 .hw_params =3D max98088_dai2_hw_params, + .digital_mute =3D max98088_dai2_digital_mute, }; =20static struct snd_soc_dai_driver max98088_dai[] =3D { diff --git a/sound/soc/codecs/max98088.h b/sound/soc/codecs/max98088.h index 56554c7..be89a4f 100644 --- a/sound/soc/codecs/max98088.h +++ b/sound/soc/codecs/max98088.h @@ -133,6 +133,19 @@ =20 #define M98088_REC_LINEMODE (1<<7) =20 #define M98088_REC_LINEMODE_MASK (1<<7) +/* M98088_REG_2D_MIX_SPK_CNTL */ + #define M98088_MIX_SPKR_GAIN_MASK (3<<2) + #define M98088_MIX_SPKR_GAIN_SHIFT 2 + #define M98088_MIX_SPKL_GAIN_MASK (3<<0) + #define M98088_MIX_SPKL_GAIN_SHIFT 0 + +/* M98088_REG_2F_LVL_DAI1_PLAY, M98088_REG_31_LVL_DAI2_PLAY */ + #define M98088_DAI_MUTE (1<<7) + #define M98088_DAI_MUTE_MASK (1<<7) + #define M98088_DAI_VOICE_GAIN_MASK (3<<4) + #define M98088_DAI_ATTENUATION_MASK (0xF<<0) + #define M98088_DAI_ATTENUATION_SHIFT 0 + /* M98088_REG_35_LVL_MIC1, M98088_REG_36_LVL_MIC2 */ =20 #define M98088_MICPRE_MASK (3<<5) =20 #define M98088_MICPRE_SHIFT 5 -- 1.7.1 Best regards, Jinyoung. -------------------------------------------------------------------------= ---------- This email message is for the sole use of the intended recipient(s) and m= ay contain confidential information. Any unauthorized review, use, disclosure or di= stribution is prohibited. If you are not the intended recipient, please contact the= =20sender by reply email and destroy all copies of the original message. -------------------------------------------------------------------------= ----------

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[PATCH 3/3] ASoC: codecs: max98088: Added digital mute function =
by Jin Park 11 May '11

11 May '11

in DAI1 and DAI2 Added digital mute function in DAI1 and DAI2. Signed-off-by: Jin Park <jinyoungp(a)nvidia.com<mailto:jinyoungp@nvidia.com= >> --- sound/soc/codecs/max98088.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/codecs/max98088.h | 13 +++++++++++++ 2 files changed, 45 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index c520093..b5ccf37 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1582,6 +1582,36 @@ static int max98088_dai2_set_fmt(struct snd_soc_da= i *codec_dai, =20 return 0; } +static int max98088_dai1_digital_mute(struct snd_soc_dai *codec_dai, int= =20mute) +{ + struct snd_soc_codec *codec =3D codec_dai->codec; + int reg; + + if (mute) + reg =3D M98088_DAI_MUTE; + else + reg =3D 0; + + snd_soc_update_bits(codec, M98088_REG_2F_LVL_DAI1_PLAY, + M98088_DAI_MUTE_MASK, reg); + return 0; +} + +static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int= =20mute) +{ + struct snd_soc_codec *codec =3D codec_dai->codec; + int reg; + + if (mute) + reg =3D M98088_DAI_MUTE; + else + reg =3D 0; + + snd_soc_update_bits(codec, M98088_REG_31_LVL_DAI2_PLAY, + M98088_DAI_MUTE_MASK, reg); + return 0; +} + static void max98088_sync_cache(struct snd_soc_codec *codec) { =20 u16 *reg_cache =3D codec->reg_cache; @@ -1643,12 +1673,14 @@ static struct snd_soc_dai_ops max98088_dai1_ops =3D= =20{ =20 .set_sysclk =3D max98088_dai_set_sysclk, =20 .set_fmt =3D max98088_dai1_set_fmt, =20 .hw_params =3D max98088_dai1_hw_params, + .digital_mute =3D max98088_dai1_digital_mute, }; =20static struct snd_soc_dai_ops max98088_dai2_ops =3D { =20 .set_sysclk =3D max98088_dai_set_sysclk, =20 .set_fmt =3D max98088_dai2_set_fmt, =20 .hw_params =3D max98088_dai2_hw_params, + .digital_mute =3D max98088_dai2_digital_mute, }; =20static struct snd_soc_dai_driver max98088_dai[] =3D { diff --git a/sound/soc/codecs/max98088.h b/sound/soc/codecs/max98088.h index 56554c7..be89a4f 100644 --- a/sound/soc/codecs/max98088.h +++ b/sound/soc/codecs/max98088.h @@ -133,6 +133,19 @@ =20 #define M98088_REC_LINEMODE (1<<7) =20 #define M98088_REC_LINEMODE_MASK (1<<7) +/* M98088_REG_2D_MIX_SPK_CNTL */ + #define M98088_MIX_SPKR_GAIN_MASK (3<<2) + #define M98088_MIX_SPKR_GAIN_SHIFT 2 + #define M98088_MIX_SPKL_GAIN_MASK (3<<0) + #define M98088_MIX_SPKL_GAIN_SHIFT 0 + +/* M98088_REG_2F_LVL_DAI1_PLAY, M98088_REG_31_LVL_DAI2_PLAY */ + #define M98088_DAI_MUTE (1<<7) + #define M98088_DAI_MUTE_MASK (1<<7) + #define M98088_DAI_VOICE_GAIN_MASK (3<<4) + #define M98088_DAI_ATTENUATION_MASK (0xF<<0) + #define M98088_DAI_ATTENUATION_SHIFT 0 + /* M98088_REG_35_LVL_MIC1, M98088_REG_36_LVL_MIC2 */ =20 #define M98088_MICPRE_MASK (3<<5) =20 #define M98088_MICPRE_SHIFT 5 -- 1.7.1 Best regards, Jinyoung. -------------------------------------------------------------------------= ---------- This email message is for the sole use of the intended recipient(s) and m= ay contain confidential information. Any unauthorized review, use, disclosure or di= stribution is prohibited. If you are not the intended recipient, please contact the= =20sender by reply email and destroy all copies of the original message. -------------------------------------------------------------------------= ----------

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[PATCH 3/3] ASoC: codecs: max98088: Added digital mute function =
by Jin Park 11 May '11

11 May '11

in DAI1 and DAI2 Added digital mute function in DAI1 and DAI2. Signed-off-by: Jin Park <jinyoungp(a)nvidia.com<mailto:jinyoungp@nvidia.com= >> --- sound/soc/codecs/max98088.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/codecs/max98088.h | 13 +++++++++++++ 2 files changed, 45 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index c520093..b5ccf37 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1582,6 +1582,36 @@ static int max98088_dai2_set_fmt(struct snd_soc_da= i *codec_dai, =20 return 0; } +static int max98088_dai1_digital_mute(struct snd_soc_dai *codec_dai, int= =20mute) +{ + struct snd_soc_codec *codec =3D codec_dai->codec; + int reg; + + if (mute) + reg =3D M98088_DAI_MUTE; + else + reg =3D 0; + + snd_soc_update_bits(codec, M98088_REG_2F_LVL_DAI1_PLAY, + M98088_DAI_MUTE_MASK, reg); + return 0; +} + +static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int= =20mute) +{ + struct snd_soc_codec *codec =3D codec_dai->codec; + int reg; + + if (mute) + reg =3D M98088_DAI_MUTE; + else + reg =3D 0; + + snd_soc_update_bits(codec, M98088_REG_31_LVL_DAI2_PLAY, + M98088_DAI_MUTE_MASK, reg); + return 0; +} + static void max98088_sync_cache(struct snd_soc_codec *codec) { =20 u16 *reg_cache =3D codec->reg_cache; @@ -1643,12 +1673,14 @@ static struct snd_soc_dai_ops max98088_dai1_ops =3D= =20{ =20 .set_sysclk =3D max98088_dai_set_sysclk, =20 .set_fmt =3D max98088_dai1_set_fmt, =20 .hw_params =3D max98088_dai1_hw_params, + .digital_mute =3D max98088_dai1_digital_mute, }; =20static struct snd_soc_dai_ops max98088_dai2_ops =3D { =20 .set_sysclk =3D max98088_dai_set_sysclk, =20 .set_fmt =3D max98088_dai2_set_fmt, =20 .hw_params =3D max98088_dai2_hw_params, + .digital_mute =3D max98088_dai2_digital_mute, }; =20static struct snd_soc_dai_driver max98088_dai[] =3D { diff --git a/sound/soc/codecs/max98088.h b/sound/soc/codecs/max98088.h index 56554c7..be89a4f 100644 --- a/sound/soc/codecs/max98088.h +++ b/sound/soc/codecs/max98088.h @@ -133,6 +133,19 @@ =20 #define M98088_REC_LINEMODE (1<<7) =20 #define M98088_REC_LINEMODE_MASK (1<<7) +/* M98088_REG_2D_MIX_SPK_CNTL */ + #define M98088_MIX_SPKR_GAIN_MASK (3<<2) + #define M98088_MIX_SPKR_GAIN_SHIFT 2 + #define M98088_MIX_SPKL_GAIN_MASK (3<<0) + #define M98088_MIX_SPKL_GAIN_SHIFT 0 + +/* M98088_REG_2F_LVL_DAI1_PLAY, M98088_REG_31_LVL_DAI2_PLAY */ + #define M98088_DAI_MUTE (1<<7) + #define M98088_DAI_MUTE_MASK (1<<7) + #define M98088_DAI_VOICE_GAIN_MASK (3<<4) + #define M98088_DAI_ATTENUATION_MASK (0xF<<0) + #define M98088_DAI_ATTENUATION_SHIFT 0 + /* M98088_REG_35_LVL_MIC1, M98088_REG_36_LVL_MIC2 */ =20 #define M98088_MICPRE_MASK (3<<5) =20 #define M98088_MICPRE_SHIFT 5 -- 1.7.1 Best regards, Jinyoung. -------------------------------------------------------------------------= ---------- This email message is for the sole use of the intended recipient(s) and m= ay contain confidential information. Any unauthorized review, use, disclosure or di= stribution is prohibited. If you are not the intended recipient, please contact the= =20sender by reply email and destroy all copies of the original message. -------------------------------------------------------------------------= ----------

1 0

[PATCH 2/3] ASoC: codecs: max98088: Moved the EX Limiter Mode fr=
by Jin Park 11 May '11

11 May '11

om dapm widget to control Moved the EX Limiter Mode from dapm widget to control, because it was not= required DAPM route. Signed-off-by: Jin Park <jinyoungp(a)nvidia.com> --- sound/soc/codecs/max98088.c | 6 +----- 1 files changed, 1 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 3bdaa04..c520093 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -656,8 +656,6 @@ static const struct soc_enum max98088_exmode_enum =3D= =20 ARRAY_SIZE(max98088_exmode_texts), =20 max98088_exmode_texts, =20 max98088_exmode_values); -static const struct snd_kcontrol_new max98088_exmode_controls =3D - SOC_DAPM_VALUE_ENUM("Route", max98088_exmode_enum); =20static const char *max98088_ex_thresh[] =3D { /* volts PP */ =20 "0.6", "1.2", "1.8", "2.4", "3.0", "3.6", "4.2", "4.8"}; @@ -783,6 +781,7 @@ static const struct snd_kcontrol_new max98088_snd_con= trols[] =3D { =20 SOC_SINGLE("EQ1 Switch", M98088_REG_49_CFG_LEVEL, 0, 1, 0), =20 SOC_SINGLE("EQ2 Switch", M98088_REG_49_CFG_LEVEL, 1, 1, 0), + SOC_ENUM("EX Limiter Mode", max98088_exmode_enum), =20 SOC_ENUM("EX Limiter Threshold", max98088_ex_thresh_enum), =20 SOC_ENUM("DAI1 Filter Mode", max98088_filter_mode_enum), @@ -1094,9 +1093,6 @@ static const struct snd_soc_dapm_widget max98088_da= pm_widgets[] =3D { =20 SND_SOC_DAPM_MICBIAS("MICBIAS", M98088_REG_4C_PWR_EN_IN, 3, 0),= - SND_SOC_DAPM_MUX("EX Limiter Mode", SND_SOC_NOPM, 0, 0, - &max98088_exmode_controls), - =20 SND_SOC_DAPM_OUTPUT("HPL"), =20 SND_SOC_DAPM_OUTPUT("HPR"), =20 SND_SOC_DAPM_OUTPUT("SPKL"), -- 1.7.1 -------------------------------------------------------------------------= -----------------------------------------------------

1 0

[PATCH 2/3] ASoC: codecs: max98088: Moved the EX Limiter Mode fr=
by Jin Park 11 May '11

11 May '11

om dapm widget to control Moved the EX Limiter Mode from dapm widget to control, because it was not= required DAPM route. Signed-off-by: Jin Park <jinyoungp(a)nvidia.com<mailto:jinyoungp@nvidia.com= >> --- sound/soc/codecs/max98088.c | 6 +----- 1 files changed, 1 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 3bdaa04..c520093 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -656,8 +656,6 @@ static const struct soc_enum max98088_exmode_enum =3D= =20 ARRAY_SIZE(max98088_exmode_texts), =20 max98088_exmode_texts, =20 max98088_exmode_values); -static const struct snd_kcontrol_new max98088_exmode_controls =3D - SOC_DAPM_VALUE_ENUM("Route", max98088_exmode_enum); =20static const char *max98088_ex_thresh[] =3D { /* volts PP */ =20 "0.6", "1.2", "1.8", "2.4", "3.0", "3.6", "4.2", "4.8"}; @@ -783,6 +781,7 @@ static const struct snd_kcontrol_new max98088_snd_con= trols[] =3D { =20 SOC_SINGLE("EQ1 Switch", M98088_REG_49_CFG_LEVEL, 0, 1, 0), =20 SOC_SINGLE("EQ2 Switch", M98088_REG_49_CFG_LEVEL, 1, 1, 0), + SOC_ENUM("EX Limiter Mode", max98088_exmode_enum), =20 SOC_ENUM("EX Limiter Threshold", max98088_ex_thresh_enum), =20 SOC_ENUM("DAI1 Filter Mode", max98088_filter_mode_enum), @@ -1094,9 +1093,6 @@ static const struct snd_soc_dapm_widget max98088_da= pm_widgets[] =3D { =20 SND_SOC_DAPM_MICBIAS("MICBIAS", M98088_REG_4C_PWR_EN_IN, 3, 0),= - SND_SOC_DAPM_MUX("EX Limiter Mode", SND_SOC_NOPM, 0, 0, - &max98088_exmode_controls), - =20 SND_SOC_DAPM_OUTPUT("HPL"), =20 SND_SOC_DAPM_OUTPUT("HPR"), =20 SND_SOC_DAPM_OUTPUT("SPKL"), -- 1.7.1 -------------------------------------------------------------------------= -----------------------------------------------------

1 0

[PATCH 2/3] ASoC: codecs: max98088: Moved the EX Limiter Mode fr=
by Jin Park 11 May '11

11 May '11

om dapm widget to control Moved the EX Limiter Mode from dapm widget to control, because it was not= required DAPM route. Signed-off-by: Jin Park <jinyoungp(a)nvidia.com<mailto:jinyoungp@nvidia.com= >> --- sound/soc/codecs/max98088.c | 6 +----- 1 files changed, 1 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 3bdaa04..c520093 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -656,8 +656,6 @@ static const struct soc_enum max98088_exmode_enum =3D= =20 ARRAY_SIZE(max98088_exmode_texts), =20 max98088_exmode_texts, =20 max98088_exmode_values); -static const struct snd_kcontrol_new max98088_exmode_controls =3D - SOC_DAPM_VALUE_ENUM("Route", max98088_exmode_enum); =20static const char *max98088_ex_thresh[] =3D { /* volts PP */ =20 "0.6", "1.2", "1.8", "2.4", "3.0", "3.6", "4.2", "4.8"}; @@ -783,6 +781,7 @@ static const struct snd_kcontrol_new max98088_snd_con= trols[] =3D { =20 SOC_SINGLE("EQ1 Switch", M98088_REG_49_CFG_LEVEL, 0, 1, 0), =20 SOC_SINGLE("EQ2 Switch", M98088_REG_49_CFG_LEVEL, 1, 1, 0), + SOC_ENUM("EX Limiter Mode", max98088_exmode_enum), =20 SOC_ENUM("EX Limiter Threshold", max98088_ex_thresh_enum), =20 SOC_ENUM("DAI1 Filter Mode", max98088_filter_mode_enum), @@ -1094,9 +1093,6 @@ static const struct snd_soc_dapm_widget max98088_da= pm_widgets[] =3D { =20 SND_SOC_DAPM_MICBIAS("MICBIAS", M98088_REG_4C_PWR_EN_IN, 3, 0),= - SND_SOC_DAPM_MUX("EX Limiter Mode", SND_SOC_NOPM, 0, 0, - &max98088_exmode_controls), - =20 SND_SOC_DAPM_OUTPUT("HPL"), =20 SND_SOC_DAPM_OUTPUT("HPR"), =20 SND_SOC_DAPM_OUTPUT("SPKL"), -- 1.7.1 -------------------------------------------------------------------------= -----------------------------------------------------

1 0

[PATCH 1/3] ASoC: codecs: max98088: Fixed invalid register defin=
by Jin Park 11 May '11

11 May '11

itions in mixer controls Fixed invalid register definitions in mixer controls such as left speaker mixer, left hp mixer and left rec mixer. Signed-off-by: Jin Park <jinyoungp(a)nvidia.com> --- sound/soc/codecs/max98088.c | 24 ++++++++++++------------ 1 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 89498f9..3bdaa04 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -808,10 +808,10 @@ static const struct snd_kcontrol_new max98088_snd_c= ontrols[] =3D { =20/* Left speaker mixer switch */ static const struct snd_kcontrol_new max98088_left_speaker_mixer_controls= [] =3D { - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7= , 1, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, = 0, 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7= , 1, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, = 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0= , 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, = 7, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0= , 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, = 7, 1, 0), =20 SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 5, 1= , 0), =20 SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 6, 1= , 0), =20 SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 1, 1= , 0), @@ -836,10 +836,10 @@ static const struct snd_kcontrol_new max98088_right= _speaker_mixer_controls[] =3D { =20/* Left headphone mixer switch */ static const struct snd_kcontrol_new max98088_left_hp_mixer_controls[] =3D= =20{ - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 7,= =201, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 0= , 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 7,= =201, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 0= , 1, 0), + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 0,= =201, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 7= , 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 0,= =201, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 7= , 1, 0), =20 SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_25_MIX_HP_LEFT, 5, 1,= =200), =20 SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_25_MIX_HP_LEFT, 6, 1,= =200), =20 SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_25_MIX_HP_LEFT, 1, 1,= =200), @@ -864,10 +864,10 @@ static const struct snd_kcontrol_new max98088_right= _hp_mixer_controls[] =3D { =20/* Left earpiece/receiver mixer switch */ static const struct snd_kcontrol_new max98088_left_rec_mixer_controls[] =3D= =20{ - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 7= , 1, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, = 0, 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 7= , 1, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, = 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 0= , 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, = 7, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 0= , 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, = 7, 1, 0), =20 SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_28_MIX_REC_LEFT, 5, 1= , 0), =20 SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_28_MIX_REC_LEFT, 6, 1= , 0), =20 SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_28_MIX_REC_LEFT, 1, 1= , 0), -- 1.7.1 -------------------------------------------------------------------------= -----------------------------------------------------

1 0

[PATCH 1/3] ASoC: codecs: max98088: Fixed invalid register defin=
by Jin Park 11 May '11

11 May '11

itions in mixer controls Fixed invalid register definitions in mixer controls such as left speaker mixer, left hp mixer and left rec mixer. Signed-off-by: Jin Park <jinyoungp(a)nvidia.com<mailto:jinyoungp@nvidia.com= >> --- sound/soc/codecs/max98088.c | 24 ++++++++++++------------ 1 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 89498f9..3bdaa04 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -808,10 +808,10 @@ static const struct snd_kcontrol_new max98088_snd_c= ontrols[] =3D { =20/* Left speaker mixer switch */ static const struct snd_kcontrol_new max98088_left_speaker_mixer_controls= [] =3D { - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7= , 1, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, = 0, 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7= , 1, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, = 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0= , 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, = 7, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0= , 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, = 7, 1, 0), =20 SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 5, 1= , 0), =20 SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 6, 1= , 0), =20 SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 1, 1= , 0), @@ -836,10 +836,10 @@ static const struct snd_kcontrol_new max98088_right= _speaker_mixer_controls[] =3D { =20/* Left headphone mixer switch */ static const struct snd_kcontrol_new max98088_left_hp_mixer_controls[] =3D= =20{ - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 7,= =201, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 0= , 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 7,= =201, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 0= , 1, 0), + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 0,= =201, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 7= , 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 0,= =201, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 7= , 1, 0), =20 SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_25_MIX_HP_LEFT, 5, 1,= =200), =20 SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_25_MIX_HP_LEFT, 6, 1,= =200), =20 SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_25_MIX_HP_LEFT, 1, 1,= =200), @@ -864,10 +864,10 @@ static const struct snd_kcontrol_new max98088_right= _hp_mixer_controls[] =3D { =20/* Left earpiece/receiver mixer switch */ static const struct snd_kcontrol_new max98088_left_rec_mixer_controls[] =3D= =20{ - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 7= , 1, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, = 0, 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 7= , 1, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, = 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 0= , 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, = 7, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 0= , 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, = 7, 1, 0), =20 SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_28_MIX_REC_LEFT, 5, 1= , 0), =20 SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_28_MIX_REC_LEFT, 6, 1= , 0), =20 SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_28_MIX_REC_LEFT, 1, 1= , 0), -- 1.7.1 -------------------------------------------------------------------------= -----------------------------------------------------

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[PATCH 1/3] ASoC: codecs: max98088: Fixed invalid register defin=
by Jin Park 11 May '11

11 May '11

itions in mixer controls Fixed invalid register definitions in mixer controls such as left speaker mixer, left hp mixer and left rec mixer. Signed-off-by: Jin Park <jinyoungp(a)nvidia.com<mailto:jinyoungp@nvidia.com= >> --- sound/soc/codecs/max98088.c | 24 ++++++++++++------------ 1 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 89498f9..3bdaa04 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -808,10 +808,10 @@ static const struct snd_kcontrol_new max98088_snd_c= ontrols[] =3D { =20/* Left speaker mixer switch */ static const struct snd_kcontrol_new max98088_left_speaker_mixer_controls= [] =3D { - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7= , 1, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, = 0, 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7= , 1, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, = 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0= , 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, = 7, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0= , 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, = 7, 1, 0), =20 SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 5, 1= , 0), =20 SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 6, 1= , 0), =20 SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 1, 1= , 0), @@ -836,10 +836,10 @@ static const struct snd_kcontrol_new max98088_right= _speaker_mixer_controls[] =3D { =20/* Left headphone mixer switch */ static const struct snd_kcontrol_new max98088_left_hp_mixer_controls[] =3D= =20{ - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 7,= =201, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 0= , 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 7,= =201, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 0= , 1, 0), + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 0,= =201, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 7= , 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 0,= =201, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 7= , 1, 0), =20 SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_25_MIX_HP_LEFT, 5, 1,= =200), =20 SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_25_MIX_HP_LEFT, 6, 1,= =200), =20 SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_25_MIX_HP_LEFT, 1, 1,= =200), @@ -864,10 +864,10 @@ static const struct snd_kcontrol_new max98088_right= _hp_mixer_controls[] =3D { =20/* Left earpiece/receiver mixer switch */ static const struct snd_kcontrol_new max98088_left_rec_mixer_controls[] =3D= =20{ - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 7= , 1, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, = 0, 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 7= , 1, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, = 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 0= , 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, = 7, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 0= , 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, = 7, 1, 0), =20 SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_28_MIX_REC_LEFT, 5, 1= , 0), =20 SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_28_MIX_REC_LEFT, 6, 1= , 0), =20 SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_28_MIX_REC_LEFT, 1, 1= , 0), -- 1.7.1 -------------------------------------------------------------------------= -----------------------------------------------------

1 0

[alsa-devel] [PATCH] CHROMIUM: ASoC: Remove 'const' modifiers which cause warnings during compilation.
by Taylor Hutt 11 May '11

11 May '11

This change removes a warning produced when 'const * const' values are assigned to a 'const **'. These code being modified originally came from these two commits: http://git.kernel.org/?p=linux/kernel/git/broonie/sound-2.6.git;a=commit;h=… http://git.kernel.org/?p=linux/kernel/git/broonie/sound-2.6.git;a=commit;h=… Change-Id: I964ff6d24aafe78c2db7002abfd03b9069db9687 Signed-off-by: Taylor Hutt <thutt(a)chromium.org> --- sound/soc/codecs/max98095.c | 12 +++++------- 1 files changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index a6cc94e..55f5d0f 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -678,7 +678,7 @@ static void m98095_biquad_band(struct snd_soc_codec *codec, unsigned int dai, } } -static const char * const max98095_fltr_mode[] = { "Voice", "Music" }; +static const char *max98095_fltr_mode[] = { "Voice", "Music" }; static const struct soc_enum max98095_dai1_filter_mode_enum[] = { SOC_ENUM_SINGLE(M98095_02E_DAI1_FILTERS, 7, 2, max98095_fltr_mode), }; @@ -686,7 +686,7 @@ static const struct soc_enum max98095_dai2_filter_mode_enum[] = { SOC_ENUM_SINGLE(M98095_038_DAI2_FILTERS, 7, 2, max98095_fltr_mode), }; -static const char * const max98095_extmic_text[] = { "None", "MIC1", "MIC2" }; +static const char *max98095_extmic_text[] = { "None", "MIC1", "MIC2" }; static const struct soc_enum max98095_extmic_enum = SOC_ENUM_SINGLE(M98095_087_CFG_MIC, 0, 3, max98095_extmic_text); @@ -694,24 +694,22 @@ static const struct soc_enum max98095_extmic_enum = static const struct snd_kcontrol_new max98095_extmic_mux = SOC_DAPM_ENUM("External MIC Mux", max98095_extmic_enum); -static const char * const max98095_linein_text[] = { "INA", "INB" }; - +static const char *max98095_linein_text[] = { "INA", "INB" }; static const struct soc_enum max98095_linein_enum = SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 6, 2, max98095_linein_text); static const struct snd_kcontrol_new max98095_linein_mux = SOC_DAPM_ENUM("Linein Input Mux", max98095_linein_enum); -static const char * const max98095_line_mode_text[] = { +static const char *max98095_line_mode_text[] = { "Stereo", "Differential"}; - static const struct soc_enum max98095_linein_mode_enum = SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 7, 2, max98095_line_mode_text); static const struct soc_enum max98095_lineout_mode_enum = SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 4, 2, max98095_line_mode_text); -static const char * const max98095_dai_fltr[] = { +static const char *max98095_dai_fltr[] = { "Off", "Elliptical-HPF-16k", "Butterworth-HPF-16k", "Elliptical-HPF-8k", "Butterworth-HPF-8k", "Butterworth-HPF-Fs/240"}; static const struct soc_enum max98095_dai1_dac_filter_enum[] = { -- 1.7.3.1

2 1

[alsa-devel] --separate-channels issues
by hong zhang 10 May '11

10 May '11

Takashi, When I run following commands with 2 pins inputs. $ arecord -f S16_LE -r 48000 -c 4 -D plughw:0,1 --separate-channels c1.wav c2.wav c3.wav c4.wav hd c3.wav and c4.wav returns 0 but c1.wav and c2.wav got right data. Such as $ hd c3.wav 00000000 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 |................| * 005ad000 Also, I see trigger Rx start command and then stop command immediately. Any hint to look into? Thanks. ---henry

2 3

[alsa-devel] [PATCH] ASoC: WM8903: Fix Digital Capture Volume range
by Stephen Warren 10 May '11

10 May '11

Increase the range of the Digital Capture Volume control to be 120 steps. Each step is 0.75dB, and the range starts at -72dB, giving a max setting of 18dB, which matches the latest datasheet, to the precision of the step size. Tested-on: Seaboard non-clamshell. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/wm8903.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index f3cab84..957cd66 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -699,7 +699,7 @@ SOC_ENUM("DRC Smoothing Threshold", drc_smoothing), SOC_SINGLE_TLV("DRC Startup Volume", WM8903_DRC_0, 6, 18, 0, drc_tlv_startup), SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8903_ADC_DIGITAL_VOLUME_LEFT, - WM8903_ADC_DIGITAL_VOLUME_RIGHT, 1, 96, 0, digital_tlv), + WM8903_ADC_DIGITAL_VOLUME_RIGHT, 1, 120, 0, digital_tlv), SOC_ENUM("ADC Companding Mode", adc_companding), SOC_SINGLE("ADC Companding Switch", WM8903_AUDIO_INTERFACE_0, 3, 1, 0), -- 1.7.0.4

3 2

[alsa-devel] [PATCH] pulseaudio - Add pa_alsa_dump when alsa-sink/alsa-source undderrun
by Raymond Yau 10 May '11

10 May '11

Add pa_alsa_dump when alsa-sink/alsa-source undderrun

3 2

[alsa-devel] [PATCH 1/3] tea575x: unify read/write functions
by Ondrej Zary 10 May '11

10 May '11

Implement generic read/write functions to access TEA575x tuners. They're now implemented 4 times (once in es1968 and 3 times in fm801). This also allows mute to work on all cards. Also improve tuner detection/initialization. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-2.6.39-rc2-/include/sound/tea575x-tuner.h 2011-04-29 22:48:34.000000000 +0200 +++ linux-2.6.39-rc2/include/sound/tea575x-tuner.h 2011-05-06 22:20:46.000000000 +0200 @@ -26,12 +26,17 @@ #include <media/v4l2-dev.h> #include <media/v4l2-ioctl.h> +#define TEA575X_DATA (1 << 0) +#define TEA575X_CLK (1 << 1) +#define TEA575X_WREN (1 << 2) +#define TEA575X_MOST (1 << 3) + struct snd_tea575x; struct snd_tea575x_ops { - void (*write)(struct snd_tea575x *tea, unsigned int val); - unsigned int (*read)(struct snd_tea575x *tea); - void (*mute)(struct snd_tea575x *tea, unsigned int mute); + void (*set_pins)(struct snd_tea575x *tea, u8 pins); + u8 (*get_pins)(struct snd_tea575x *tea); + void (*set_direction)(struct snd_tea575x *tea, bool output); }; struct snd_tea575x { @@ -49,7 +54,7 @@ struct snd_tea575x { void *private_data; }; -void snd_tea575x_init(struct snd_tea575x *tea); +int snd_tea575x_init(struct snd_tea575x *tea); void snd_tea575x_exit(struct snd_tea575x *tea); #endif /* __SOUND_TEA575X_TUNER_H */ --- linux-2.6.39-rc2-/sound/i2c/other/tea575x-tuner.c 2011-05-06 22:44:14.000000000 +0200 +++ linux-2.6.39-rc2/sound/i2c/other/tea575x-tuner.c 2011-05-06 22:21:09.000000000 +0200 @@ -77,11 +77,65 @@ static struct v4l2_queryctrl radio_qctrl * lowlevel part */ +static void snd_tea575x_write(struct snd_tea575x *tea, unsigned int val) +{ + u16 l; + u8 data; + + tea->ops->set_direction(tea, 1); + udelay(16); + + for (l = 25; l > 0; l--) { + data = (val >> 24) & TEA575X_DATA; + val <<= 1; /* shift data */ + tea->ops->set_pins(tea, data | TEA575X_WREN); + udelay(2); + tea->ops->set_pins(tea, data | TEA575X_WREN | TEA575X_CLK); + udelay(2); + tea->ops->set_pins(tea, data | TEA575X_WREN); + udelay(2); + } + + if (!tea->mute) + tea->ops->set_pins(tea, 0); +} + +static unsigned int snd_tea575x_read(struct snd_tea575x *tea) +{ + u16 l, rdata; + u32 data = 0; + + tea->ops->set_direction(tea, 0); + tea->ops->set_pins(tea, 0); + udelay(16); + + for (l = 24; l--;) { + tea->ops->set_pins(tea, TEA575X_CLK); + udelay(2); + if (!l) + tea->tuned = tea->ops->get_pins(tea) & TEA575X_MOST ? 0 : 1; + tea->ops->set_pins(tea, 0); + udelay(2); + data <<= 1; /* shift data */ + rdata = tea->ops->get_pins(tea); + if (!l) + tea->stereo = (rdata & TEA575X_MOST) ? 0 : 1; + if (rdata & TEA575X_DATA) + data++; + udelay(2); + } + + if (tea->mute) + tea->ops->set_pins(tea, TEA575X_WREN); + + return data; +} + static void snd_tea575x_get_freq(struct snd_tea575x *tea) { unsigned long freq; - freq = tea->ops->read(tea) & TEA575X_BIT_FREQ_MASK; + freq = snd_tea575x_read(tea) & TEA575X_BIT_FREQ_MASK; /* freq *= 12.5 */ freq *= 125; freq /= 10; @@ -111,7 +165,7 @@ static void snd_tea575x_set_freq(struct tea->val &= ~TEA575X_BIT_FREQ_MASK; tea->val |= freq & TEA575X_BIT_FREQ_MASK; - tea->ops->write(tea, tea->val); + snd_tea575x_write(tea, tea->val); } /* @@ -139,7 +193,7 @@ static int vidioc_g_tuner(struct file *f if (v->index > 0) return -EINVAL; - tea->ops->read(tea); + snd_tea575x_read(tea); strcpy(v->name, "FM"); v->type = V4L2_TUNER_RADIO; @@ -233,10 +287,8 @@ static int vidioc_g_ctrl(struct file *fi switch (ctrl->id) { case V4L2_CID_AUDIO_MUTE: - if (tea->ops->mute) { - ctrl->value = tea->mute; - return 0; - } + ctrl->value = tea->mute; + return 0; } return -EINVAL; } @@ -248,11 +300,11 @@ static int vidioc_s_ctrl(struct file *fi switch (ctrl->id) { case V4L2_CID_AUDIO_MUTE: - if (tea->ops->mute) { - tea->ops->mute(tea, ctrl->value); + if (tea->mute != ctrl->value) { tea->mute = ctrl->value; - return 0; + snd_tea575x_set_freq(tea); } + return 0; } return -EINVAL; } @@ -317,18 +369,16 @@ static struct video_device tea575x_radio /* * initialize all the tea575x chips */ -void snd_tea575x_init(struct snd_tea575x *tea) +int snd_tea575x_init(struct snd_tea575x *tea) { int retval; - unsigned int val; struct video_device *tea575x_radio_inst; - val = tea->ops->read(tea); - if (val == 0x1ffffff || val == 0) { - snd_printk(KERN_ERR - "tea575x-tuner: Cannot find TEA575x chip\n"); - return; - } + tea->mute = 1; + + snd_tea575x_write(tea, 0x55AA); + if (snd_tea575x_read(tea) != 0x55AA) + return -ENODEV; tea->in_use = 0; tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_10_40; @@ -337,7 +387,7 @@ void snd_tea575x_init(struct snd_tea575x tea575x_radio_inst = video_device_alloc(); if (tea575x_radio_inst == NULL) { printk(KERN_ERR "tea575x-tuner: not enough memory\n"); - return; + return -ENOMEM; } memcpy(tea575x_radio_inst, &tea575x_radio, sizeof(tea575x_radio)); @@ -352,17 +402,13 @@ void snd_tea575x_init(struct snd_tea575x if (retval) { printk(KERN_ERR "tea575x-tuner: can't register video device!\n"); kfree(tea575x_radio_inst); - return; + return retval; } snd_tea575x_set_freq(tea); - - /* mute on init */ - if (tea->ops->mute) { - tea->ops->mute(tea, 1); - tea->mute = 1; - } tea->vd = tea575x_radio_inst; + + return 0; } void snd_tea575x_exit(struct snd_tea575x *tea) -- Ondrej Zary

2 1

[alsa-devel] ASoC updates for 2.6.40
by Mark Brown 10 May '11

10 May '11

The following changes since commit 87023ff74b2358b5e51d3c790704f786e89ff769: ASoC: Declare const properly for enum texts (2011-05-03 12:51:59 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.40 Dimitris Papastamos (1): ASoC: soc-cache: Allow codec->cache_bypass to be used with snd_soc_hw_bulk_write_raw() Lars-Peter Clausen (18): ASoC: Free the card's DAPM context ASoC: Move DAPM debugfs directory creation to snd_soc_dapm_debugfs_init ASoC: Move DAPM widget debugfs entry creation to snd_soc_dapm_new_widgets ASoC: Remove DAPM debugfs entries before freeing widgets ASoC: JZ4740: Fix i2s shutdown ASoC: Create codec DAPM widgets before calling the codecs probe function ASoC: SSM2602: Properly annotate i2c probe and remove functions ASoC: SSM2602: Fix 'Mic Boost2' control ASoC: SSM2602: Fix reg_cache_size ASoC: SSM2602: Cleanup coeff handling ASoC: SSM2602: Remove duplicate control ASoC: SSM2602: Remove unused struct and define ASoC: SSM2602: Fix default register cache ASoC: SSM2602: Do not power the codec up in probe ASoC: SSM2602: Add SSM2604 support ASoC: SSM2602: Add entry for the ssm2603 to the device id table ASoC: SSM2602: Model power supply for the digital core as a DAPM widget ASoC: SSM2602: Provide dB ranges for the volume controls Marek Belisko (1): ASoC: UDA134x: Remove POWER_OFF_ON_STANDBY define. Mark Brown (8): Merge branch 'for-2.6.39' into for-2.6.40 ASoC: Don't crash on PM operations Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-2.6.40 ASoC: Use shared controls for input signal path in WM8915 ASoC: Remove outdated FIXME from WM8915 Merge branch 'for-2.6.39' into for-2.6.40 Merge branch 'for-2.6.39' into for-2.6.40 Merge branch 'for-2.6.39' into for-2.6.40 Mike Rapoport (1): ASoC: tegra: TrimSlice machine support Stephen Warren (4): ASoC: s/w->kcontrols/w->kcontrol_news/g ASoC: Add w->kcontrols, and populate it ASoC: Store a list of widgets in a DAPM mux/mixer kcontrol ASoC: Implement mux control sharing xingchao (1): ASoC: sst_platform: add hw_free callback to fix resource leak include/sound/soc-dapm.h | 78 +++++---- sound/soc/codecs/88pm860x-codec.c | 2 +- sound/soc/codecs/ssm2602.c | 302 ++++++++++++++++++------------- sound/soc/codecs/ssm2602.h | 6 - sound/soc/codecs/uda134x.c | 2 - sound/soc/codecs/wm8915.c | 46 +++-- sound/soc/jz4740/jz4740-i2s.c | 2 +- sound/soc/mid-x86/sst_platform.c | 6 + sound/soc/soc-cache.c | 11 +- sound/soc/soc-core.c | 38 ++--- sound/soc/soc-dapm.c | 373 ++++++++++++++++++++++++++----------- sound/soc/tegra/Kconfig | 8 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/trimslice.c | 228 ++++++++++++++++++++++ 14 files changed, 780 insertions(+), 324 deletions(-) create mode 100644 sound/soc/tegra/trimslice.c

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[alsa-devel] ASoC updates for 2.6.39
by Mark Brown 10 May '11

10 May '11

The following changes since commit 69b91bc1551a2fc746a01fea9d3291e60be3780d: ASoC: Fix CODEC DAI names for Goni (2011-04-28 12:09:53 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 Lars-Peter Clausen (4): ASoC: JZ4740: Fix i2s shutdown ASoC: SSM2602: Properly annotate i2c probe and remove functions ASoC: SSM2602: Fix 'Mic Boost2' control ASoC: SSM2602: Fix reg_cache_size Marek Belisko (1): ASoC: UDA134x: Remove POWER_OFF_ON_STANDBY define. Mark Brown (1): ASoC: Don't crash on PM operations xingchao (1): ASoC: sst_platform: add hw_free callback to fix resource leak sound/soc/codecs/ssm2602.c | 10 +++++----- sound/soc/codecs/uda134x.c | 2 -- sound/soc/jz4740/jz4740-i2s.c | 2 +- sound/soc/mid-x86/sst_platform.c | 6 ++++++ sound/soc/soc-core.c | 2 ++ 5 files changed, 14 insertions(+), 8 deletions(-)

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[alsa-devel] [GIT PULL] ASoC updates for 2.6.40
by Liam Girdwood 10 May '11

10 May '11

The following changes since commit 333802e90d3f0366c4a1cb767e2783d2e1df73a8: ASoC: Support !REGULATOR build for sgtl5000 (2011-03-22 18:26:30 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.40 Peter Ujfalusi (2): ASoC: tlv320dac33: Move codec power up to DAPM ASoC: tlv320dac33: Lower the OSC calibration time sound/soc/codecs/tlv320dac33.c | 13 ++++++++----- 1 files changed, 8 insertions(+), 5 deletions(-)

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