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

[alsa-devel] ALSA porting to h/w having kernel version 2.6.18.5
by Rakesh Kumar 17 Jul '09

17 Jul '09

Hi I am new to ALSA and going to start porting of ALSA driver on out H/W box which is having 2.6.18.5 kernel (ARM), soc scenario. ALSA version supported on 2.6.18.5 is 1.0.12. I have following questions: 1. Shall I go ahead with ALSA 1.0.12 (which is already part of my kernel)? 2. or shall I download latest ALSA version 1.0.20? I can go with either with 1.0.12 or 1.0.20 versions, I want to know what is general trend followed in this scenario. I tried building my kernel with latest ALSA (as a part of kernel) but facing some compilation errors. Thanks Regards, Rakesh _________________________________________________________________ Stay updated! Add Facebook, LinkedIn, MySpace & Hi5 friends to your Windows Live network instantly. Add Now! http://profile.live.com/webactivities/?mkt=en-in

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Re: [alsa-devel] [PATCH 2/2] Add possibility to control the GPIO_STATUS shift
by Mark Brown 17 Jul '09

17 Jul '09

On Fri, Jul 17, 2009 at 10:01:16AM +0200, Marek Vasut wrote: > This patch allows tweaking the behaviour of GPIO_STATUS register > shift quirk that's in wm97xx-core. The problem with GPIO_STATUS > register being shifted by one doesn't appear on all hardware it > seems and causes problems with accelerated touchscreen drivers on > Palm hardware. Therefore an accelerated touchscreen driver can select > if the shift is/isn't happening on the hardware. Again, this isn't an ALSA patch. I've done some checking internally and I'm very suspicous that what's going on here is actually the masking of some other bug elsewhere in the system. Which CPU are the systems you're observing this on using? There is at least one special case in the current code for GPIO register readbacks in the PXA AC97 driver code and there have been a number of interoperabilty issues in that area...

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[alsa-devel] [PATCH 1/2] Add Palm support to Mainstone accelerated touch
by Marek Vasut 17 Jul '09

17 Jul '09

From f32fa9ef6476be0cf8106bc57a98724c48d4baa1 Mon Sep 17 00:00:00 2001 From: Marek Vasut <marek.vasut(a)gmail.com> Date: Fri, 17 Jul 2009 09:54:13 +0200 Subject: [PATCH 1/2] Add Palm support to Mainstone accelerated touch This patch refactors the Mainstone accelerated touch code a little and adds support for interrupt driven touchscreen on Palm LifeDrive, TX and Tungsten T5. Signed-off-by: Marek Vasut <marek.vasut(a)gmail.com> --- drivers/input/touchscreen/Kconfig | 4 +- drivers/input/touchscreen/mainstone-wm97xx.c | 52 +++++++++++++++++-------- 2 files changed, 37 insertions(+), 19 deletions(-) diff --git a/drivers/input/touchscreen/Kconfig b/drivers/input/touchscreen/Kconfig index 72e2712..0fdae1a 100644 --- a/drivers/input/touchscreen/Kconfig +++ b/drivers/input/touchscreen/Kconfig @@ -366,11 +366,11 @@ config TOUCHSCREEN_WM97XX_ATMEL be called atmel-wm97xx. config TOUCHSCREEN_WM97XX_MAINSTONE - tristate "WM97xx Mainstone accelerated touch" + tristate "WM97xx Mainstone/Palm accelerated touch" depends on TOUCHSCREEN_WM97XX && ARCH_PXA help Say Y here for support for streaming mode with WM97xx touchscreens - on Mainstone systems. + on Mainstone, Palm Tungsten T5, TX and LifeDrive systems. If unsure, say N. diff --git a/drivers/input/touchscreen/mainstone-wm97xx.c b/drivers/input/touchscreen/mainstone-wm97xx.c index 4cc047a..60bda9a 100644 --- a/drivers/input/touchscreen/mainstone-wm97xx.c +++ b/drivers/input/touchscreen/mainstone-wm97xx.c @@ -31,9 +31,13 @@ #include <linux/interrupt.h> #include <linux/wm97xx.h> #include <linux/io.h> +#include <linux/gpio.h> + #include <mach/regs-ac97.h> -#define VERSION "0.13" +#include <asm/mach-types.h> + +#define VERSION "0.14" struct continuous { u16 id; /* codec id */ @@ -62,6 +66,7 @@ static const struct continuous cinfo[] = { /* continuous speed index */ static int sp_idx; static u16 last, tries; +static int irq; /* * Pen sampling frequency (Hz) in continuous mode. @@ -171,7 +176,7 @@ up: static int wm97xx_acc_startup(struct wm97xx *wm) { - int idx = 0; + int idx = 0, ret = 0; /* check we have a codec */ if (wm->ac97 == NULL) @@ -191,18 +196,37 @@ static int wm97xx_acc_startup(struct wm97xx *wm) "mainstone accelerated touchscreen driver, %d samples/sec\n", cinfo[sp_idx].speed); + /* IRQ driven touchscreen is used on Palm hardware */ + if (machine_is_palmt5() || machine_is_palmtx() || machine_is_palmld()) { + pen_int = 1; + irq = 27; + } else if (machine_is_mainstone() && pen_int) + irq = 4; + + if (irq) { + ret = gpio_request(irq, "Touchscreen IRQ"); + if (ret) + goto out; + + ret = gpio_direction_input(irq); + if (ret) { + gpio_free(irq); + goto out; + } + + wm->pen_irq = gpio_to_irq(irq); + set_irq_type(wm->pen_irq, IRQ_TYPE_EDGE_BOTH); + } else /* pen irq not supported */ + pen_int = 0; + /* codec specific irq config */ if (pen_int) { switch (wm->id) { case WM9705_ID2: - wm->pen_irq = IRQ_GPIO(4); - set_irq_type(IRQ_GPIO(4), IRQ_TYPE_EDGE_BOTH); break; case WM9712_ID2: case WM9713_ID2: - /* enable pen down interrupt */ /* use PEN_DOWN GPIO 13 to assert IRQ on GPIO line 2 */ - wm->pen_irq = MAINSTONE_AC97_IRQ; wm97xx_config_gpio(wm, WM97XX_GPIO_13, WM97XX_GPIO_IN, WM97XX_GPIO_POL_HIGH, WM97XX_GPIO_STICKY, @@ -220,23 +244,17 @@ static int wm97xx_acc_startup(struct wm97xx *wm) } } - return 0; +out: + return ret; } static void wm97xx_acc_shutdown(struct wm97xx *wm) { /* codec specific deconfig */ if (pen_int) { - switch (wm->id & 0xffff) { - case WM9705_ID2: - wm->pen_irq = 0; - break; - case WM9712_ID2: - case WM9713_ID2: - /* disable interrupt */ - wm->pen_irq = 0; - break; - } + if (irq) + gpio_free(irq); + wm->pen_irq = 0; } } -- 1.6.3.3

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Re: [alsa-devel] [PATCH] hda: add bounds checking for the codec command fields v2
by Takashi Iwai 17 Jul '09

17 Jul '09

At Fri, 17 Jul 2009 12:46:01 +0200, I wrote: > > At Fri, 17 Jul 2009 18:41:05 +0800, > Guo, Chaohong wrote: > > > > Although it does address this issue, I am not comfortable with this fixing. > > It seems more like a workaround than fix. > > No, it's rather for catching a bug. This is definitely neither > "workaround" nor "fix". BTW, I fixed the patch again as below. Takashi --- From 82e1b804b03defe46fb69ffd2c8b19e6649bcb0d Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Fri, 17 Jul 2009 12:47:34 +0200 Subject: [PATCH] ALSA: hda - Fix the previous sanity check in make_codec_cmd() The newly added sanity-check for a codec verb can be better written with logical ORs. Also, the parameter can be more than 8bit. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d9d3262..35f0f22 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -150,8 +150,8 @@ make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, { u32 val; - if ((codec->addr & ~0xf) | (direct & ~1) | (nid & ~0x7f) | - (verb & ~0xfff) | (parm & ~0xff)) { + if ((codec->addr & ~0xf) || (direct & ~1) || (nid & ~0x7f) || + (verb & ~0xfff) || (parm & ~0xffff)) { printk(KERN_ERR "hda-codec: out of range cmd %x:%x:%x:%x:%x\n", codec->addr, direct, nid, verb, parm); return ~0; -- 1.6.3.2

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[PATCH] oss - Add missing initialization of fragments
by Takashi Iwai 17 Jul '09

17 Jul '09

The periods calculation was missing for initializing OSS fragments. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- oss/pcm_oss.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/oss/pcm_oss.c b/oss/pcm_oss.c index 8a2a672..d43b44c 100644 --- a/oss/pcm_oss.c +++ b/oss/pcm_oss.c @@ -181,6 +181,7 @@ static int oss_hw_params(snd_pcm_ioplug_t *io, fprintf(stderr, "*** OSS: invalid period size %d\n", (int)io->period_size); return -EINVAL; } + oss->periods = io->buffer_size / io->period_size; _retry: tmp = oss->period_shift | (oss->periods << 16); -- 1.6.3.2

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[alsa-devel] [PATCH] hda: add bounds checking for the codec command fields
by Wu Fengguang 17 Jul '09

17 Jul '09

A recent bug involves passing auto detected >0x7f NID to codec command, creating an invalid codec addr field, and finally lead to cmd timeout and fall back into single command mode. Jaroslav fixed that bug in alc880_parse_auto_config(). It would be safer to further check the bounds of all cmd fields. Cc: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 462e2ce..7d09650 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -150,6 +150,16 @@ make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, { u32 val; + if ((direct & ~1) || (nid & ~0x7f) || + (verb & ~0xfff) || (parm & ~0xff)) { + printk(KERN_ERR "hda-codec: out of range cmd %x:%x:%x:%x:%x\n", + codec->addr, direct, nid, verb, parm); + direct &= 1; + nid &= 0x7f; + verb &= 0xfff; + parm &= 0xff; + } + val = (u32)(codec->addr & 0x0f) << 28; val |= (u32)direct << 27; val |= (u32)nid << 20;

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Re: [alsa-devel] [PATCH]HDA: Add support for new AMD HD audio devices
by Takashi Iwai 17 Jul '09

17 Jul '09

At Fri, 17 Jul 2009 11:32:32 +0800, Andiry Brienza wrote: > > Add support for new AMD HD audio devices. Use generic driver to detect HD audio > devices with Vendor ID AMD. > > Signed-off-by: Andiry Xu <andiry.xu(a)amd.com> Thanks, applied now. Takashi > diff -ruN a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > --- a/sound/pci/hda/hda_intel.c 2009-07-17 10:46:24.000000000 +0800 > +++ b/sound/pci/hda/hda_intel.c 2009-07-17 11:04:45.000000000 +0800 > @@ -2604,11 +2604,15 @@ > /* this entry seems still valid -- i.e. without emu20kx chip */ > { PCI_DEVICE(0x1102, 0x0009), .driver_data = AZX_DRIVER_GENERIC }, > #endif > - /* AMD Generic, PCI class code and Vendor ID for HD Audio */ > + /* AMD/ATI Generic, PCI class code and Vendor ID for HD 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 }, > + { PCI_DEVICE(PCI_VENDOR_ID_AMD, PCI_ANY_ID), > + .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, > + .class_mask = 0xffffff, > + .driver_data = AZX_DRIVER_GENERIC }, > { 0, } > }; > MODULE_DEVICE_TABLE(pci, azx_ids); > >

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[alsa-devel] Callbacks from opening or closing alsa streams
by Baek Chang 17 Jul '09

17 Jul '09

Hi, I was wondering if it was possible to get callbacks from opening and closing of alsa streams with the alsa sink name that was opened. So if I have an asound.conf configured like so: pcm.music { type hw card 0 } Is it possible to get a callback saying that a stream on the "music" sink was opened? Thanks Baek -- Baek Chang baeksan(a)ccrma.stanford.edu

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[alsa-devel] ANYONE have spdif working with the ADI AD1986A chipset? (Asus M2NPV-VM mainboard)
by VDR User 17 Jul '09

17 Jul '09

This mainboard isn't that new but from what I've read spdif worked in alsa some time in the past. I get the IEC958/IEC958 D options in alsamixer but can not get any sound. Analog works fine. If anyone has spdif working on this mainboard then PLEASE post your exact setup & settings! The only other option would be to buy new hardware and that's not an option right now. Thanks for any/all help! -Derek

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[alsa-devel] While recording from mono codec twice as much data is received
by Louis Lee 16 Jul '09

16 Jul '09

Hi everyone, I am able to do playback from 8 kHz samples to 48 kHz samples properly. However, when recording using the exact same setup for the data interface I am not getting favorable results. For instance, if I record for 5 seconds I get output in a file for 10 seconds of audio (arecord or sox). This results in my recording being stretched. I change the DAI to stereo capture and record two channels and the recorded output comes back correctly (5 seconds recorded is 5 seconds). I'm using a wolfsom WM8974. I'm developing using ASoC framework (can I call it that?) reusing the WM8974 codec from the Linux GIT repository. The development is using a i.MX27. Data interface is a SSI bus in DSP mode. I believe my PCM code is correct since it works for playback. I'm stuck now. From my understanding the clocks are setup the same way for both playback and capture. Has anyone ran into this problem before? The WM8974 is a mono codec so I should not have to set the DAI to stereo in order to capture data properly. Any help is appreciated. Best regards,Louis _________________________________________________________________ Attention all humans. We are your photos. Free us. http://go.microsoft.com/?linkid=9666046

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[alsa-devel] [PATCH] ALSA: hda - Add CX20582 and OLPC XO-1.5 support
by Daniel Drake 16 Jul '09

16 Jul '09

This adds support for the Conexant CX20582 codec, based on code from http://www.linuxant.com/alsa-driver/alsa-driver-linuxant-1.0.19ppch12-1.noa… This is the codec to be shipped in the OLPC XO-1.5, so this patch also includes an XO-specific profile. Resultant configuration: http://dev.laptop.org/~dsd/20090713/codec0.txt http://dev.laptop.org/~dsd/20090713/codec0.svg As the Linuxant code is structured differently to the other codecs, I was unable to cleanly reimplement everything in the generic and Dell profiles as more info is needed (e.g. codec graphs). I simplified those profiles so that hopefully it will not break anyone's audio. If it does, it may be worth returning -ENODEV from patch_cx5066 on non-OLPC systems, and then fixing snd_hda_codec_configure() to fall back on the generic parser, at least until support for other systems is figured out. --- sound/pci/hda/patch_conexant.c | 479 ++++++++++++++++++++++++++++++++++++++++ 1 files changed, 479 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4fcbe21..77b4f28 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -108,6 +108,8 @@ struct conexant_spec { struct hda_input_mux private_imux; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; + unsigned int dell_automute; + unsigned int port_d_mode; }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -1908,6 +1910,480 @@ static int patch_cxt5051(struct hda_codec *codec) return 0; } +/* Conexant 5066 specific */ + +static hda_nid_t cxt5066_dac_nids[1] = { 0x10 }; +static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 }; +static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 }; +#define CXT5066_SPDIF_OUT 0x21 + +static struct hda_channel_mode cxt5066_modes[1] = { + { 2, NULL }, +}; + +static void cxt5066_update_speaker(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + unsigned int pinctl; + + snd_printdd("CXT5066: update speaker, hp_present=%d\n", + spec->hp_present); + + /* Port A (HP) */ + pinctl = ((spec->hp_present & 1) && spec->cur_eapd) ? PIN_HP : 0; + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl); + + /* Port D (HP/LO) */ + pinctl = ((spec->hp_present & 2) && spec->cur_eapd) + ? spec->port_d_mode : 0; + snd_hda_codec_write(codec, 0x1c, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl); + + /* CLASS_D AMP */ + pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl); + + if (spec->dell_automute) { + /* DELL AIO Port Rule: PortA > PortD > IntSpk */ + pinctl = (!(spec->hp_present & 1) && spec->cur_eapd) + ? PIN_OUT : 0; + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); + } +} + +/* turn on/off EAPD (+ mute HP) as a master switch */ +static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + if (!cxt_eapd_put(kcontrol, ucontrol)) + return 0; + + cxt5066_update_speaker(codec); + return 1; +} + +/* toggle input of built-in and mic jack appropriately */ +static void cxt5066_automic(struct hda_codec *codec) +{ + static struct hda_verb ext_mic_present[] = { + /* enable external mic, port B */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + + /* switch to external mic input */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0}, + + /* disable internal mic, port C */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {} + }; + static struct hda_verb ext_mic_absent[] = { + /* enable internal mic, port C */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + + /* switch to internal mic input */ + {0x17, AC_VERB_SET_CONNECT_SEL, 1}, + + /* disable external mic, port B */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {} + }; + unsigned int present; + + present = snd_hda_codec_read(codec, 0x1a, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + if (present) { + snd_printdd("CXT5066: external microphone detected\n"); + snd_hda_sequence_write(codec, ext_mic_present); + } else { + snd_printdd("CXT5066: external microphone absent\n"); + snd_hda_sequence_write(codec, ext_mic_absent); + } +} + +/* mute internal speaker if HP is plugged */ +static void cxt5066_hp_automute(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + unsigned int portA, portD; + + /* Port A */ + portA = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + + /* Port D */ + portD = (snd_hda_codec_read(codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE) << 1; + + spec->hp_present = !!(portA | portD); + snd_printdd("CXT5066: hp automute portA=%x portD=%x present=%d\n", + portA, portD, spec->hp_present); + cxt5066_update_speaker(codec); +} + +/* unsolicited event for jack sensing */ +static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) +{ + snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); + switch (res >> 26) { + case CONEXANT_HP_EVENT: + cxt5066_hp_automute(codec); + break; + case CONEXANT_MIC_EVENT: + cxt5066_automic(codec); + break; + } +} + +static const struct hda_input_mux cxt5066_analog_mic_boost = { + .num_items = 5, + .items = { + { "0dB", 0 }, + { "10dB", 1 }, + { "20dB", 2 }, + { "30dB", 3 }, + { "40dB", 4 }, + }, +}; + +static int cxt5066_mic_boost_mux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return snd_hda_input_mux_info(&cxt5066_analog_mic_boost, uinfo); +} + +static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + int val; + + val = snd_hda_codec_read(codec, 0x17, 0, + AC_VERB_GET_AMP_GAIN_MUTE, AC_AMP_GET_OUTPUT); + + ucontrol->value.enumerated.item[0] = val & AC_AMP_GAIN; + return 0; +} + +static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + const struct hda_input_mux *imux = &cxt5066_analog_mic_boost; + unsigned int idx; + + if (!imux->num_items) + return 0; + idx = ucontrol->value.enumerated.item[0]; + if (idx >= imux->num_items) + idx = imux->num_items - 1; + + snd_hda_codec_write_cache(codec, 0x17, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT | + imux->items[idx].index); + + return 1; +} + +static struct hda_input_mux cxt5066_capture_source = { + .num_items = 4, + .items = { + { "Mic B", 0 }, + { "Mic C", 1 }, + { "Mic E", 2 }, + { "Mic F", 3 }, + }, +}; + +static struct hda_bind_ctls cxt5066_bind_capture_vol_others = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT), + 0 + }, +}; + +static struct hda_bind_ctls cxt5066_bind_capture_sw_others = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT), + 0 + }, +}; + +static struct snd_kcontrol_new cxt5066_mixer_master[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), + {} +}; + +static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, + .info = snd_hda_mixer_amp_volume_info, + .get = snd_hda_mixer_amp_volume_get, + .put = snd_hda_mixer_amp_volume_put, + .tlv = { .c = snd_hda_mixer_amp_tlv }, + /* offset by 28 volume steps to limit minimum gain to -46dB */ + .private_value = + HDA_COMPOSE_AMP_VAL_OFS(0x10, 3, 0, HDA_OUTPUT, 28), + }, + {} +}; + +static struct snd_kcontrol_new cxt5066_mixers[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5066_hp_master_sw_put, + .private_value = 0x1d, + }, + + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Mic Boost Capture Enum", + .info = cxt5066_mic_boost_mux_enum_info, + .get = cxt5066_mic_boost_mux_enum_get, + .put = cxt5066_mic_boost_mux_enum_put, + }, + + HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others), + HDA_BIND_SW("Capture Switch", &cxt5066_bind_capture_sw_others), + {} +}; + +static struct hda_verb cxt5066_init_verbs[] = { + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */ + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */ + + /* Speakers */ + {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* HP, Amp */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* no digital microphone support yet */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Audio input selector */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3}, + + /* SPDIF route: PCM */ + {0x20, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x22, AC_VERB_SET_CONNECT_SEL, 0x0}, + + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* EAPD */ + {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + + /* not handling these yet */ + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + { } /* end */ +}; + +static struct hda_verb cxt5066_init_verbs_olpc[] = { + /* Port A: headphones */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* Port B: external microphone */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + + /* Port C: internal microphone */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + + /* Port D: unused */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Port E: unused, but has primary EAPD */ + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + + /* Port F: unused */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Port G: internal speakers */ + {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* DAC2: unused */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + + /* Disable digital microphone port */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Audio input selectors */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + + /* Disable SPDIF */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* enable unsolicited events for Port A and B */ + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + { } /* end */ +}; + +static struct hda_verb cxt5066_init_verbs_portd_lo[] = { + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { } /* end */ +}; + +/* initialize jack-sensing, too */ +static int cxt5066_init(struct hda_codec *codec) +{ + snd_printdd("CXT5066: init\n"); + conexant_init(codec); + if (codec->patch_ops.unsol_event) { + cxt5066_hp_automute(codec); + cxt5066_automic(codec); + } + return 0; +} + +enum { + CXT5066_LAPTOP, /* Laptops w/ EAPD support */ + CXT5066_DELL_LAPTOP, /* Dell Laptop */ + CXT5066_OLPC_XO_1_5, /* OLPC XO 1.5 */ + CXT5066_MODELS +}; + +static const char *cxt5066_models[CXT5066_MODELS] = { + [CXT5066_LAPTOP] = "laptop", + [CXT5066_DELL_LAPTOP] = "dell-laptop", + [CXT5066_OLPC_XO_1_5] = "olpc-xo-1_5", +}; + +static struct snd_pci_quirk cxt5066_cfg_tbl[] = { + SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", + CXT5066_LAPTOP), + SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", + CXT5066_DELL_LAPTOP), + {} +}; + +static int patch_cxt5066(struct hda_codec *codec) +{ + struct conexant_spec *spec; + int board_config; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + codec->spec = spec; + + codec->patch_ops = conexant_patch_ops; + codec->patch_ops.init = cxt5066_init; + + spec->dell_automute = 0; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = ARRAY_SIZE(cxt5066_dac_nids); + spec->multiout.dac_nids = cxt5066_dac_nids; + spec->multiout.dig_out_nid = CXT5066_SPDIF_OUT; + spec->num_adc_nids = 1; + spec->adc_nids = cxt5066_adc_nids; + spec->capsrc_nids = cxt5066_capsrc_nids; + spec->input_mux = &cxt5066_capture_source; + + spec->port_d_mode = PIN_HP; + + spec->num_init_verbs = 1; + spec->init_verbs[0] = cxt5066_init_verbs; + spec->num_channel_mode = ARRAY_SIZE(cxt5066_modes); + spec->channel_mode = cxt5066_modes; + spec->cur_adc = 0; + spec->cur_adc_idx = 0; + + board_config = snd_hda_check_board_config(codec, CXT5066_MODELS, + cxt5066_models, cxt5066_cfg_tbl); + switch (board_config) { + default: + case CXT5066_LAPTOP: + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + break; + case CXT5066_DELL_LAPTOP: + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + + spec->port_d_mode = PIN_OUT; + spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_portd_lo; + spec->num_init_verbs++; + spec->dell_automute = 1; + break; + case CXT5066_OLPC_XO_1_5: + codec->patch_ops.unsol_event = cxt5066_unsol_event; + spec->init_verbs[0] = cxt5066_init_verbs_olpc; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + spec->port_d_mode = 0; + + /* no S/PDIF out */ + spec->multiout.dig_out_nid = 0; + + /* input source automatically selected */ + spec->input_mux = NULL; + break; + } + + return 0; +} /* */ @@ -1919,12 +2395,15 @@ static struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_cxt5047 }, { .id = 0x14f15051, .name = "CX20561 (Hermosa)", .patch = patch_cxt5051 }, + { .id = 0x14f15066, .name = "CX20582 (Pebble)", + .patch = patch_cxt5066 }, {} /* terminator */ }; MODULE_ALIAS("snd-hda-codec-id:14f15045"); MODULE_ALIAS("snd-hda-codec-id:14f15047"); MODULE_ALIAS("snd-hda-codec-id:14f15051"); +MODULE_ALIAS("snd-hda-codec-id:14f15066"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); -- 1.6.2.5

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Re: [alsa-devel] [PATCH 01/06] Fix compilation warning for sound/soc/codecs/wm8400.c
by Mark Brown 16 Jul '09

16 Jul '09

On Thu, Jul 16, 2009 at 06:40:01PM +0530, Subrata Modak wrote: > How about the following brutal shutdown ? No, this is the patch you originally submitted. You are missing the point here - your patches are all just papering over the warning and won't help at all if there is an actual problem. > 'factors' will get initialized here, as, 'freq_out' will probably never have > a '0' value. So, 'fll_factors()' will actually overwrite values in 'factors' > even after the initial brutal assignment: > "struct fll_factors factors = {}" This is pretty much the point of the warning when it's valid - it's trying to catch situations where there is a code path where the variable is used without being initialised. If you just blindly initialise the variable as you are doing then the warning goes away but any problematic code remains so the situation is worse. In this case I suspect that whatever compiler you are using (none of those I've tried with seem to be doing this) isn't able to figure out that if we skip initialising the variable then we use exactly the same condition to return from the function before we try to use the variable. For something like this the warning can normally be worked around by changing the control flow so that the compiler is able to figure out that things are safe.

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Re: [alsa-devel] No sound with nVidia Corporation MCP79 High Definition Audio (rev b1) (Macbook Pro 5, 5)
by Takashi Iwai 16 Jul '09

16 Jul '09

At Wed, 8 Jul 2009 18:11:29 +0200, Andreas Nüßlein wrote: > > On Wednesday 08 July 2009 16:49:34 you wrote: > > At Tue, 07 Jul 2009 12:53:48 +0200, > > > > I wrote: > > > At Tue, 7 Jul 2009 11:49:57 +0200, > > > > > > Andreas Nüßlein wrote: > > > > On Tuesday 07 July 2009 08:00:19 you wrote: > > > > > At Mon, 6 Jul 2009 22:14:56 +0200, > > > > > > > > > > Andreas Nüßlein wrote: > > > > > > On Monday 06 July 2009 21:26:18 you wrote: > > > > > > > At Mon, 06 Jul 2009 11:16:35 -0600, > > > > > > > > > > > > > > Sean Burke wrote: > > > > > > > > Scríobh Takashi Iwai: > > > > > > > > > At Mon, 6 Jul 2009 17:53:46 +0200, > > > > > > > > > > > > > > > > > > Andreas Nüßlein wrote: > > > > > > > > >>> The missing pin configuration initialization was already > > > > > > > > >>> fixed by the driver overriding it after checking PCI SSID > > > > > > > > >>> (which is different from the codec SSID). So, this should > > > > > > > > >>> be no problem. > > > > > > > > >>> > > > > > > > > >>> However, the reason why the analog output doesn't work > > > > > > > > >>> might be different from that. There might be something > > > > > > > > >>> else missing, but I don't know. > > > > > > > > >>> > > > > > > > > >>> > > > > > > > > >>> Takashi > > > > > > > > >> > > > > > > > > >> oh =( > > > > > > > > >> > > > > > > > > >> > > > > > > > > >> hmm.. anything i can do? would it help if i tried changing > > > > > > > > >> values randomly with hda-analyzer.py? > > > > > > > > > > > > > > > > > > Well, did the driver without my change work more or less with > > > > > > > > > the analog audio, or have you never gotten the analog output? > > > > > > > > > You can use the generic parser (i.e. the state without cirrus > > > > > > > > > patch) by passing model=generic option to snd-hda-intel. > > > > > > > > > > > > > > > > For my part, nothing worked with the generic driver. I can't > > > > > > > > confirm digital out, but I can confirm that the kfree error is > > > > > > > > gone. What options are open for figuring out what remains? > > > > > > > > > > > > > > Easy things to test are GPIO bits. Run hda-verb like > > > > > > > > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > > > > or > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > > > > > > > > > > > etc. CS4206 seems to have 4 GPIO lines, and each bit (0-3) > > > > > > > corresponds to each GPIO. In many case, GPIO0 or GPIO1 > > > > > > > corresponds to the amplifier (EAPD) bit. > > > > > > > Define the GPIO direction of each GPIO bit by SET_GPIO_DIR, and > > > > > > > turn on/off the GPIO bits by SET_GPIO_DATA. Running > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 GET_GPIO_DATA 0 > > > > > > > will show the current GPIO data bits. Or you can check it in > > > > > > > codec#* proc file. > > > > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > > > w000000000000000000000000000000000t! =) > > > > > > > > > > > > takashi, thank you _so_ much! > > > > > > > > > > > > after running all 4 of those: > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x0f > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > > > > or > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x0f > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x0f > > > > > > > > > > > > i suddendly had sound!! :D :D :D :D :D :D :D > > > > > > only via speakers though - there is no sound via headphones right > > > > > > now. > > > > > > > > > > > > mixer channels: > > > > > > - Master (with Mutebutton), PCM and Front (also with Mute) all work > > > > > > =) - i don't know what surround would do (or it's extra switch) - > > > > > > headphones-volumes and mute button don't affect the speakers, which > > > > > > is good =) > > > > > > > > > > > > > > > > > > is there a way to reset what i did with hda-verb, so that i can > > > > > > figure out which combination it was exactly? > > > > > > > > > > You can just change the value 0x0f to a different value. > > > > > At least, you can try commands like > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x04 > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x08 > > > > > and check the speaker output at each time. > > > > > Also, check the GPIO direction, > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x01 > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x02 > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 > > > > > ... > > > > > > > > > > Regarding the headphone: is the speaker muted when you plug in the > > > > > headphone? If not, it's likely an issue of the jack detection. If > > > > > the speaker is muted but no headphone output, it's a missing > > > > > initialization (or wrong GPIO setup). > > > > > > > > > > > > > > > Takashi > > > > > > > > so here are my findings so far: > > > > > > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x08 > > > > > > > > > > > > is sufficient to turn on the right functionality for the speakers; > > > > meaning: - 'front speaker'-channel actually controls the two front > > > > speakers within the macbookpro - even controlling left and right > > > > channel seperately works =) - 'surround' mute-switch and > > > > volume-control (why are those not within on thing btw?) also work and > > > > control a third speaker in the mac, which really seems responsibly for > > > > surround :) > > > > > > > > headphones do not yet work, however i made a diff of > > > > "/proc/asound/card0/codec#0" with and without something plugged in: > > > > > > OK, so actually the driver works as expected, but the hardware doesn't > > > behave as expected. For muting the speaker, something else (like > > > toggling that GPIO) is needed. > > > > > > The mystery about the silent HP output still remains, though. > > > > One another thing we can try is to use only one PCM stream instead of > > multi streams. For example, try the patch below. This will enable > > only the pin for the headphone, thus only one DAC is used. > > > > > > Takashi > > > > --- > > diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c > > index 57251d7..d7f52b1 100644 > > --- a/sound/pci/hda/patch_cirrus.c > > +++ b/sound/pci/hda/patch_cirrus.c > > @@ -1086,13 +1086,17 @@ struct cs_pincfg { > > > > static struct cs_pincfg mbp55_pincfgs[] = { > > { 0x09, 0x012b4030 }, > > - { 0x0a, 0x90100121 }, > > - { 0x0b, 0x90100120 }, > > + // { 0x0a, 0x90100121 }, > > + { 0x0a, 0x400000f0 }, > > + // { 0x0b, 0x90100120 }, > > + { 0x0b, 0x400000f0 }, > > { 0x0c, 0x400000f0 }, > > - { 0x0d, 0x90a00110 }, > > + // { 0x0d, 0x90a00110 }, > > + { 0x0d, 0x400000f0 }, > > { 0x0e, 0x400000f0 }, > > { 0x0f, 0x400000f0 }, > > - { 0x10, 0x014be040 }, > > + // { 0x10, 0x014be040 }, > > + { 0x10, 0x400000f0 }, > > { 0x12, 0x400000f0 }, > > { 0x15, 0x400000f0 }, > > {} /* terminator */ > > > =( no - that doesn't work either > > (there is obivously no sound via built-in speakers either anymore) Hmm, then I have really no clue. At best, you can try again GPIO pins and reverting COEF setup by my earlier patch, and try some more adjustments of widgets amp or pinctl via hda-verb or hda-analyzer... Takashi

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 16 Jul '09

16 Jul '09

The following changes since commit 4b75e9476737dffef44a66818a4a481edd2512a1: Mark Brown (1): ASoC: Error out if we can't determine a suitable WM9081 sysclk are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (1): ASoC: new ad1938 codec driver based on asoc Joonyoung Shim (3): ASoC: add SOC_DOUBLE_EXT_TLV control type ASoC: add SOC_DOUBLE_R_EXT_TLV control type ASoC: MAX9877: add MAX9877 amp driver Kevin Hilman (1): ASoC: davinci: don't use clock names include/sound/soc.h | 22 ++ sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 8 + sound/soc/codecs/ad1938.c | 652 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ad1938.h | 100 ++++++ sound/soc/codecs/max9877.c | 270 +++++++++++++++ sound/soc/codecs/max9877.h | 37 ++ sound/soc/davinci/davinci-i2s.c | 2 +- sound/soc/davinci/davinci-mcasp.c | 2 +- 9 files changed, 1100 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/ad1938.c create mode 100644 sound/soc/codecs/ad1938.h create mode 100644 sound/soc/codecs/max9877.c create mode 100644 sound/soc/codecs/max9877.h

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[alsa-devel] [PATCH 0/4] alsa-info.sh updates
by Wu Fengguang 16 Jul '09

16 Jul '09

Hi Takashi, This patchset mainly disables the auto-upload of alsa info by default. Comments are welcome. Thanks, Fengguang --

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[alsa-devel] New Audio Driver Design
by jassi brar 15 Jul '09

15 Jul '09

Hello, I am to write an ALSA driver for my SOC which has a dedicated DSP subsystem which can be loaded with decoder instructions and parameteres, and then we load it with encoded data. The dsp then decodes the data and sends it to the PCM controller. Could someone please refer me to some similar driver/codec which already does that? If not, how should i design the driver? :- Should I load the encoder inst/params using firmware API? Then, which API should I use to submit encoded data to the DSP? Any pointers please? Thanks.

2 1

[alsa-devel] [PATCH 3/3] ASoC: MAX9877: add MAX9877 amp driver
by Joonyoung Shim 15 Jul '09

15 Jul '09

The MAX9877 combines a high-efficiency Class D audio power amplifier with a stereo Class AB capacitor-less DirectDrive headphone amplifier. The max9877_add_controls() is called to register the MAX9877 specific controls on machine specific init() of the machine driver. The datasheet for the MAX9877 can find at the following url: http://datasheets.maxim-ic.com/en/ds/MAX9877.pdf Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- This patch is version 2. I updated from review of version 1. Change log: v2: - add SND_SOC_MAX9877 to SND_SOC_ALL_CODECS on Kconfig. - use SOC_DOUBLE_R_EXT_TLV instead of SOC_SINGLE_EXT_TLV for HP volume control. - modify max9877_get_reg() and max9877_set_reg for double reg controls. - add EXPORT_SYMBOL_GPL(max9877_add_controls). sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/max9877.c | 270 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max9877.h | 37 ++++++ 4 files changed, 318 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/max9877.c create mode 100644 sound/soc/codecs/max9877.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b674d3a..16a710f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -49,6 +49,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS select SND_SOC_WM9713 if SND_SOC_AC97_BUS + select SND_SOC_MAX9877 if I2C help Normally ASoC codec drivers are only built if a machine driver which uses them is also built since they are only usable with a machine @@ -188,3 +189,7 @@ config SND_SOC_WM9712 config SND_SOC_WM9713 tristate + +# Amp +config SND_SOC_MAX9877 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 301eb08..cf111c9 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -38,6 +38,9 @@ snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o +# Amp +snd-soc-max9877-objs := max9877.o + obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o @@ -77,3 +80,6 @@ obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o + +# Amp +obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c new file mode 100644 index 0000000..7df9a7c --- /dev/null +++ b/sound/soc/codecs/max9877.c @@ -0,0 +1,270 @@ +/* + * max9877.c -- amp driver for max9877 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "max9877.h" + +static struct i2c_client *i2c; + +static u8 max9877_regs[5] = { 0x40, 0x00, 0x00, 0x00, 0x49 }; + +static void max9877_write_regs(void) +{ + if (i2c_master_send(i2c, max9877_regs, 5) != 5) + dev_err(&i2c->dev, "i2c write failed\n"); +} + +static int max9877_get_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; + int reg2 = mc->rreg; + int shift = mc->shift; + int mask = mc->max; + + ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; + + if (reg2) + ucontrol->value.integer.value[1] = + (max9877_regs[reg2] >> shift) & mask; + + return 0; +} + +static int max9877_set_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; + int reg2 = mc->rreg; + int shift = mc->shift; + int mask = mc->max; + int change = 1; + int change2 = 1; + int ret = 0; + + if (((max9877_regs[reg] >> shift) & mask) == + ucontrol->value.integer.value[0]) + change = 0; + + if (reg2) + if (((max9877_regs[reg2] >> shift) & mask) == + ucontrol->value.integer.value[1]) + change2 = 0; + + if (change) { + max9877_regs[reg] &= ~(mask << shift); + max9877_regs[reg] |= ucontrol->value.integer.value[0] << shift; + ret = change; + } + + if (reg2 && change2) { + max9877_regs[reg2] &= ~(mask << shift); + max9877_regs[reg2] |= ucontrol->value.integer.value[1] << shift; + ret = change2; + } + + if (ret) + max9877_write_regs(); + + return ret; +} + +static int max9877_get_out_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK; + + if (value) + value -= 1; + + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int max9877_set_out_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = ucontrol->value.integer.value[0]; + + if (value) + value += 1; + + if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK) == value) + return 0; + + max9877_regs[MAX9877_OUTPUT_MODE] |= value; + max9877_write_regs(); + return 1; +} + +static int max9877_get_osc_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = (max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK); + + value = value >> MAX9877_OSC_OFFSET; + + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int max9877_set_osc_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = ucontrol->value.integer.value[0]; + + value = value << MAX9877_OSC_OFFSET; + if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK) == value) + return 0; + + max9877_regs[MAX9877_OUTPUT_MODE] |= value; + max9877_write_regs(); + return 1; +} + +static const unsigned int max9877_pgain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(0, 900, 0), + 2, 2, TLV_DB_SCALE_ITEM(2000, 0, 0), +}; + +static const unsigned int max9877_output_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 7, TLV_DB_SCALE_ITEM(-7900, 400, 1), + 8, 15, TLV_DB_SCALE_ITEM(-4700, 300, 0), + 16, 23, TLV_DB_SCALE_ITEM(-2300, 200, 0), + 24, 31, TLV_DB_SCALE_ITEM(-700, 100, 0), +}; + +static const char *max9877_out_mode[] = { + "INA -> SPK", + "INA -> HP", + "INA -> SPK and HP", + "INB -> SPK", + "INB -> HP", + "INB -> SPK and HP", + "INA + INB -> SPK", + "INA + INB -> HP", + "INA + INB -> SPK and HP", +}; + +static const char *max9877_osc_mode[] = { + "1176KHz", + "1100KHz", + "700KHz", +}; + +static const struct soc_enum max9877_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_out_mode), max9877_out_mode), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_osc_mode), max9877_osc_mode), +}; + +static const struct snd_kcontrol_new max9877_controls[] = { + SOC_SINGLE_EXT_TLV("MAX9877 PGAINA Playback Volume", + MAX9877_INPUT_MODE, 0, 2, 0, + max9877_get_reg, max9877_set_reg, max9877_pgain_tlv), + SOC_SINGLE_EXT_TLV("MAX9877 PGAINB Playback Volume", + MAX9877_INPUT_MODE, 2, 2, 0, + max9877_get_reg, max9877_set_reg, max9877_pgain_tlv), + SOC_SINGLE_EXT_TLV("MAX9877 Amp Speaker Playback Volume", + MAX9877_SPK_VOLUME, 0, 31, 0, + max9877_get_reg, max9877_set_reg, max9877_output_tlv), + SOC_DOUBLE_R_EXT_TLV("MAX9877 Amp HP Playback Volume", + MAX9877_HPL_VOLUME, MAX9877_HPR_VOLUME, 0, 31, 0, + max9877_get_reg, max9877_set_reg, max9877_output_tlv), + SOC_SINGLE_EXT("MAX9877 INB Stereo Switch", + MAX9877_INPUT_MODE, 4, 1, 1, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 INA Stereo Switch", + MAX9877_INPUT_MODE, 5, 1, 1, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 Zero-crossing detection Switch", + MAX9877_INPUT_MODE, 6, 1, 0, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 Bypass Mode Switch", + MAX9877_OUTPUT_MODE, 6, 1, 0, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 Shutdown Mode Switch", + MAX9877_OUTPUT_MODE, 7, 1, 1, + max9877_get_reg, max9877_set_reg), + SOC_ENUM_EXT("MAX9877 Output Mode", max9877_enum[0], + max9877_get_out_mode, max9877_set_out_mode), + SOC_ENUM_EXT("MAX9877 Oscillator Mode", max9877_enum[1], + max9877_get_osc_mode, max9877_set_osc_mode), +}; + +/* This function is called from ASoC machine driver */ +int max9877_add_controls(struct snd_soc_codec *codec) +{ + return snd_soc_add_controls(codec, max9877_controls, + ARRAY_SIZE(max9877_controls)); +} +EXPORT_SYMBOL_GPL(max9877_add_controls); + +static int __devinit max9877_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + i2c = client; + + max9877_write_regs(); + + return 0; +} + +static __devexit int max9877_i2c_remove(struct i2c_client *client) +{ + i2c = NULL; + + return 0; +} + +static const struct i2c_device_id max9877_i2c_id[] = { + { "max9877", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max9877_i2c_id); + +static struct i2c_driver max9877_i2c_driver = { + .driver = { + .name = "max9877", + .owner = THIS_MODULE, + }, + .probe = max9877_i2c_probe, + .remove = __devexit_p(max9877_i2c_remove), + .id_table = max9877_i2c_id, +}; + +static int __init max9877_init(void) +{ + return i2c_add_driver(&max9877_i2c_driver); +} +module_init(max9877_init); + +static void __exit max9877_exit(void) +{ + i2c_del_driver(&max9877_i2c_driver); +} +module_exit(max9877_exit); + +MODULE_DESCRIPTION("ASoC MAX9877 amp driver"); +MODULE_AUTHOR("Joonyoung Shim <jy0922.shim(a)samsung.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max9877.h b/sound/soc/codecs/max9877.h new file mode 100644 index 0000000..6da7229 --- /dev/null +++ b/sound/soc/codecs/max9877.h @@ -0,0 +1,37 @@ +/* + * max9877.h -- amp driver for max9877 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef _MAX9877_H +#define _MAX9877_H + +#define MAX9877_INPUT_MODE 0x00 +#define MAX9877_SPK_VOLUME 0x01 +#define MAX9877_HPL_VOLUME 0x02 +#define MAX9877_HPR_VOLUME 0x03 +#define MAX9877_OUTPUT_MODE 0x04 + +/* MAX9877_INPUT_MODE */ +#define MAX9877_INB (1 << 4) +#define MAX9877_INA (1 << 5) +#define MAX9877_ZCD (1 << 6) + +/* MAX9877_OUTPUT_MODE */ +#define MAX9877_OUTMODE_MASK (15 << 0) +#define MAX9877_OSC_MASK (3 << 4) +#define MAX9877_OSC_OFFSET 4 +#define MAX9877_BYPASS (1 << 6) +#define MAX9877_SHDN (1 << 7) + +extern int max9877_add_controls(struct snd_soc_codec *codec); + +#endif -- 1.6.0.4

2 1

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 15 Jul '09

15 Jul '09

The following changes since commit ba3b64b976f91067710ef1a87b9b445406910e8f: Mark Brown (1): Merge branch 'for-2.6.31' into for-2.6.32 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Cliff Cai (2): ASoC: Blackfin AC97: fix resume handling ASoC: Blackfin I2S: fix resume handling Mark Brown (2): ASoC: Fix sample rate lookup in WM8993 ASoC: Error out if we can't determine a suitable WM9081 sysclk sound/soc/blackfin/bf5xx-ac97.c | 10 +++------- sound/soc/blackfin/bf5xx-i2s.c | 8 ++------ sound/soc/codecs/wm8993.c | 3 ++- sound/soc/codecs/wm9081.c | 8 ++++++++ 4 files changed, 15 insertions(+), 14 deletions(-)

2 1

[alsa-devel] [PATCH 2/3] ASoC: add SOC_DOUBLE_R_EXT_TLV control type
by Joonyoung Shim 15 Jul '09

15 Jul '09

This is a macro for double controls with special callback function and TLV. The SOC_DOUBLE_R_EXT_TLV needs two registers and one shift for double controls. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- include/sound/soc.h | 11 +++++++++++ 1 files changed, 11 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 795a6b4..756fb59 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -146,6 +146,17 @@ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ .max = xmax, .invert = xinvert} } +#define SOC_DOUBLE_R_EXT_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert,\ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_2r, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ + .max = xmax, .invert = xinvert} } #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_bool_ext, \ -- 1.6.0.4

1 0

[alsa-devel] [PATCH 1/3] ASoC: add SOC_DOUBLE_EXT_TLV control type
by Joonyoung Shim 15 Jul '09

15 Jul '09

This is a macro for double controls with special callback function and TLV. The SOC_DOUBLE_EXT_TLV needs one register and two shifts for double controls. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- include/sound/soc.h | 11 +++++++++++ 1 files changed, 11 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 94fcc65..795a6b4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -135,6 +135,17 @@ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } +#define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ + .max = xmax, .invert = xinvert} } #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_bool_ext, \ -- 1.6.0.4

1 0

[alsa-devel] [PATCH] ASoC: MAX9877: add MAX9877 amp driver
by Joonyoung Shim 15 Jul '09

15 Jul '09

The MAX9877 combines a high-efficiency Class D audio power amplifier with a stereo Class AB capacitor-less DirectDrive headphone amplifier. The max9877_add_controls() is called to register the MAX9877 specific controls on machine specific init() of the machine driver. The datasheet for the MAX9877 can find at the following url: http://datasheets.maxim-ic.com/en/ds/MAX9877.pdf Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- Hi, I added amp driver in codec directory of ASoC. This is because of mail about amp driver the past. http://www.spinics.net/lists/alsa-devel/msg24876.html Please review, thank you. sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/max9877.c | 246 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max9877.h | 37 +++++++ 4 files changed, 293 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/max9877.c create mode 100644 sound/soc/codecs/max9877.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b674d3a..8930557 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -188,3 +188,7 @@ config SND_SOC_WM9712 config SND_SOC_WM9713 tristate + +# Amp +config SND_SOC_MAX9877 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 301eb08..cf111c9 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -38,6 +38,9 @@ snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o +# Amp +snd-soc-max9877-objs := max9877.o + obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o @@ -77,3 +80,6 @@ obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o + +# Amp +obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c new file mode 100644 index 0000000..cb79fe8 --- /dev/null +++ b/sound/soc/codecs/max9877.c @@ -0,0 +1,246 @@ +/* + * max9877.c -- amp driver for max9877 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "max9877.h" + +static struct i2c_client *i2c; + +static u8 max9877_regs[5] = { 0x40, 0x00, 0x00, 0x00, 0x49 }; + +static void max9877_write_regs(void) +{ + if (i2c_master_send(i2c, max9877_regs, 5) != 5) + dev_err(&i2c->dev, "i2c write failed\n"); +} + +static int max9877_get_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; + int shift = mc->shift; + int mask = mc->max; + + ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; + + return 0; +} + +static int max9877_set_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; + int shift = mc->shift; + int mask = mc->max; + + if (((max9877_regs[reg] >> shift) & mask) == + ucontrol->value.integer.value[0]) + return 0; + + max9877_regs[reg] &= ~(mask << shift); + max9877_regs[reg] |= ucontrol->value.integer.value[0] << shift; + max9877_write_regs(); + return 1; +} + +static int max9877_get_out_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK; + + if (value) + value -= 1; + + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int max9877_set_out_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = ucontrol->value.integer.value[0]; + + if (value) + value += 1; + + if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK) == value) + return 0; + + max9877_regs[MAX9877_OUTPUT_MODE] |= value; + max9877_write_regs(); + return 1; +} + +static int max9877_get_osc_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = (max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK); + + value = value >> MAX9877_OSC_OFFSET; + + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int max9877_set_osc_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = ucontrol->value.integer.value[0]; + + value = value << MAX9877_OSC_OFFSET; + if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK) == value) + return 0; + + max9877_regs[MAX9877_OUTPUT_MODE] |= value; + max9877_write_regs(); + return 1; +} + +static const unsigned int max9877_pgain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(0, 900, 0), + 2, 2, TLV_DB_SCALE_ITEM(2000, 0, 0), +}; + +static const unsigned int max9877_output_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 7, TLV_DB_SCALE_ITEM(-7900, 400, 1), + 8, 15, TLV_DB_SCALE_ITEM(-4700, 300, 0), + 16, 23, TLV_DB_SCALE_ITEM(-2300, 200, 0), + 24, 31, TLV_DB_SCALE_ITEM(-700, 100, 0), +}; + +static const char *max9877_out_mode[] = { + "INA -> SPK", + "INA -> HP", + "INA -> SPK and HP", + "INB -> SPK", + "INB -> HP", + "INB -> SPK and HP", + "INA + INB -> SPK", + "INA + INB -> HP", + "INA + INB -> SPK and HP", +}; + +static const char *max9877_osc_mode[] = { + "1176KHz", + "1100KHz", + "700KHz", +}; + +static const struct soc_enum max9877_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_out_mode), max9877_out_mode), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_osc_mode), max9877_osc_mode), +}; + +static const struct snd_kcontrol_new max9877_controls[] = { + SOC_SINGLE_EXT_TLV("MAX9877 PGAINA Playback Volume", + MAX9877_INPUT_MODE, 0, 2, 0, + max9877_get_reg, max9877_set_reg, max9877_pgain_tlv), + SOC_SINGLE_EXT_TLV("MAX9877 PGAINB Playback Volume", + MAX9877_INPUT_MODE, 2, 2, 0, + max9877_get_reg, max9877_set_reg, max9877_pgain_tlv), + SOC_SINGLE_EXT_TLV("MAX9877 Amp Speaker Playback Volume", + MAX9877_SPK_VOLUME, 0, 31, 0, + max9877_get_reg, max9877_set_reg, max9877_output_tlv), + SOC_SINGLE_EXT_TLV("MAX9877 Amp HP Left Playback Volume", + MAX9877_HPL_VOLUME, 0, 31, 0, + max9877_get_reg, max9877_set_reg, max9877_output_tlv), + SOC_SINGLE_EXT_TLV("MAX9877 Amp HP Right Playback Volume", + MAX9877_HPR_VOLUME, 0, 31, 0, + max9877_get_reg, max9877_set_reg, max9877_output_tlv), + SOC_SINGLE_EXT("MAX9877 INB Stereo Switch", + MAX9877_INPUT_MODE, 4, 1, 1, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 INA Stereo Switch", + MAX9877_INPUT_MODE, 5, 1, 1, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 Zero-crossing detection Switch", + MAX9877_INPUT_MODE, 6, 1, 0, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 Bypass Mode Switch", + MAX9877_OUTPUT_MODE, 6, 1, 0, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 Shutdown Mode Switch", + MAX9877_OUTPUT_MODE, 7, 1, 1, + max9877_get_reg, max9877_set_reg), + SOC_ENUM_EXT("MAX9877 Output Mode", max9877_enum[0], + max9877_get_out_mode, max9877_set_out_mode), + SOC_ENUM_EXT("MAX9877 Oscillator Mode", max9877_enum[1], + max9877_get_osc_mode, max9877_set_osc_mode), +}; + +/* This function is called from ASoC machine driver */ +int max9877_add_controls(struct snd_soc_codec *codec) +{ + return snd_soc_add_controls(codec, max9877_controls, + ARRAY_SIZE(max9877_controls)); +} + +static int __devinit max9877_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + i2c = client; + + max9877_write_regs(); + + return 0; +} + +static __devexit int max9877_i2c_remove(struct i2c_client *client) +{ + i2c = NULL; + + return 0; +} + +static const struct i2c_device_id max9877_i2c_id[] = { + { "max9877", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max9877_i2c_id); + +static struct i2c_driver max9877_i2c_driver = { + .driver = { + .name = "max9877", + .owner = THIS_MODULE, + }, + .probe = max9877_i2c_probe, + .remove = __devexit_p(max9877_i2c_remove), + .id_table = max9877_i2c_id, +}; + +static int __init max9877_init(void) +{ + return i2c_add_driver(&max9877_i2c_driver); +} +module_init(max9877_init); + +static void __exit max9877_exit(void) +{ + i2c_del_driver(&max9877_i2c_driver); +} +module_exit(max9877_exit); + +MODULE_DESCRIPTION("ASoC MAX9877 amp driver"); +MODULE_AUTHOR("Joonyoung Shim <jy0922.shim(a)samsung.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max9877.h b/sound/soc/codecs/max9877.h new file mode 100644 index 0000000..6da7229 --- /dev/null +++ b/sound/soc/codecs/max9877.h @@ -0,0 +1,37 @@ +/* + * max9877.h -- amp driver for max9877 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef _MAX9877_H +#define _MAX9877_H + +#define MAX9877_INPUT_MODE 0x00 +#define MAX9877_SPK_VOLUME 0x01 +#define MAX9877_HPL_VOLUME 0x02 +#define MAX9877_HPR_VOLUME 0x03 +#define MAX9877_OUTPUT_MODE 0x04 + +/* MAX9877_INPUT_MODE */ +#define MAX9877_INB (1 << 4) +#define MAX9877_INA (1 << 5) +#define MAX9877_ZCD (1 << 6) + +/* MAX9877_OUTPUT_MODE */ +#define MAX9877_OUTMODE_MASK (15 << 0) +#define MAX9877_OSC_MASK (3 << 4) +#define MAX9877_OSC_OFFSET 4 +#define MAX9877_BYPASS (1 << 6) +#define MAX9877_SHDN (1 << 7) + +extern int max9877_add_controls(struct snd_soc_codec *codec); + +#endif -- 1.6.0.4

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Re: [alsa-devel] [PATCH 01/06] Fix compilation warning for sound/soc/codecs/wm8400.c
by Mark Brown 15 Jul '09

15 Jul '09

On Wed, Jul 15, 2009 at 10:13:38AM +0530, Jaswinder Singh Rajput wrote: > Hello Subrata, > This kind of warnings born due to not handling the variables properly. > Please try to understand the root cause of this warning, just shutting > this warning is of no use. ...as I said last time you submitted rougly the same patch. Please also remember that sometimes these problems are false positives caused by issues with the compiler flow analysis (as appears to be the case here).

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[alsa-devel] [PATCH 06/06] Fix compilation warning for sound/drivers/serial/u16550.c
by Subrata Modak 15 Jul '09

15 Jul '09

Following fix is inspired by David Howells fix few days back: http://lkml.org/lkml/2009/7/9/109, Signed-off-by: Subrata Modak<subrata(a)linux.vnet.ibm.com>, --- --- a/sound/drivers/serial-u16550.c 2009-05-20 10:48:23.000000000 +0530 +++ b/sound/drivers/serial-u16550.c 2009-07-15 06:46:48.000000000 +0530 @@ -902,7 +902,7 @@ static int __devinit snd_uart16550_rmidi static int __devinit snd_serial_probe(struct platform_device *devptr) { struct snd_card *card; - struct snd_uart16550 *uart; + struct snd_uart16550 *uninitialized_var(uart); int err; int dev = devptr->id; --- Regards-- Subrata

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[alsa-devel] [PATCH 01/06] Fix compilation warning for sound/soc/codecs/wm8400.c
by Subrata Modak 15 Jul '09

15 Jul '09

Following fix is inspired by David Howells fix few days back: http://lkml.org/lkml/2009/7/9/109, Signed-off-by: Subrata Modak<subrata(a)linux.vnet.ibm.com>, --- --- a/sound/soc/codecs/wm8400.c 2009-06-15 07:52:31.000000000 +0530 +++ b/sound/soc/codecs/wm8400.c 2009-07-15 05:35:00.000000000 +0530 @@ -1015,7 +1015,7 @@ static int wm8400_set_dai_pll(struct snd { struct snd_soc_codec *codec = codec_dai->codec; struct wm8400_priv *wm8400 = codec->private_data; - struct fll_factors factors; + struct fll_factors uninitialized_var(factors); int ret; u16 reg; --- Regards-- Subrata

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[alsa-devel] Alsa buffer
by arif setiawan 14 Jul '09

14 Jul '09

Hi, I'm trying to get meaningful voice from ALSA buffer. I read tutorial on http://www.suse.de/~mana/alsa090_howto.html<http://www.suse.de/%7Emana/alsa090_howto.html>and realize that if alsa use 16 bit format value then I have to byte shift to store it in 8 bit unsigned char buffer. I see this in sine wave generator in tutorial. To confirm it, I try to save the output into file and open it in Matlab. I want to know if this is correct, because when I try saving alsa buffer on a from recording I still get noise not the voice. Thank you. Best regards, Arif This is my implementation : int main() { int nchannel = 1; int periodsize = 8192; int frame_size = nchannel * (16 / 8); int num_frames = periodsize / frame_size; unsigned char *data; int l2; short s; data = (unsigned char *)malloc(periodsize); // fill the buffer for(l2 = 0; l2 < num_frames; l2++) { s1 = (l2 % 128) * 100 - 5000; data[2*l2] = (unsigned char)s1; data[2*l2+1] = s1 >> 8; } // directly save FILE *fp1 = fopen("sineout1", "w"); for (int i = 0; i < periodsize; i++) { fprintf_s(fp1, "%d\n", data[i]); } fclose(fp1); // with bit shifting from char to short FILE *fp2 = fopen("sineout2", "w"); for (int i = 0; i < num_frames; i++) { short newval = data[2*i+1] << 8; newval += data[2*i]; fprintf_s(fp2, "%d\n", newval); } fclose(fp2); return 0; }

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