diff options
Diffstat (limited to 'sound')
114 files changed, 3783 insertions, 3603 deletions
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 8808b82311b1..76e0d5695075 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -140,6 +140,9 @@ int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) if (!prtd || !prtd->params) return 0; + if (prtd->dma_ch == -1) + return -EINVAL; + DCSR(prtd->dma_ch) &= ~DCSR_RUN; DCSR(prtd->dma_ch) = 0; DCMD(prtd->dma_ch) = 0; diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 917e4055ee30..150cb7edffee 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -253,7 +253,7 @@ static int snd_pcm_lib_preallocate_pages1(struct snd_pcm_substream *substream, * snd_pcm_lib_preallocate_pages - pre-allocation for the given DMA type * @substream: the pcm substream instance * @type: DMA type (SNDRV_DMA_TYPE_*) - * @data: DMA type dependant data + * @data: DMA type dependent data * @size: the requested pre-allocation size in bytes * @max: the max. allowed pre-allocation size * @@ -278,10 +278,10 @@ int snd_pcm_lib_preallocate_pages(struct snd_pcm_substream *substream, EXPORT_SYMBOL(snd_pcm_lib_preallocate_pages); /** - * snd_pcm_lib_preallocate_pages_for_all - pre-allocation for continous memory type (all substreams) + * snd_pcm_lib_preallocate_pages_for_all - pre-allocation for continuous memory type (all substreams) * @pcm: the pcm instance * @type: DMA type (SNDRV_DMA_TYPE_*) - * @data: DMA type dependant data + * @data: DMA type dependent data * @size: the requested pre-allocation size in bytes * @max: the max. allowed pre-allocation size * diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index fe5c8036beba..1a07750f3836 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -460,7 +460,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, PM_QOS_CPU_DMA_LATENCY, usecs); return 0; _error: - /* hardware might be unuseable from this time, + /* hardware might be unusable from this time, so we force application to retry to set the correct hardware parameter settings */ runtime->status->state = SNDRV_PCM_STATE_OPEN; diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c index f3bdc54b429a..1d7d90ca455e 100644 --- a/sound/core/seq/seq_dummy.c +++ b/sound/core/seq/seq_dummy.c @@ -50,7 +50,7 @@ option snd-seq-dummy ports=4 - The modle option "duplex=1" enables duplex operation to the port. + The model option "duplex=1" enables duplex operation to the port. In duplex mode, a pair of ports are created instead of single port, and events are tunneled between pair-ports. For example, input to port A is sent to output port of another port B and vice versa. diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index a89948ae9e8d..a39d3d8c2f9c 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -233,7 +233,7 @@ static void slave_free(struct snd_kcontrol *kcontrol) * Add a slave control to the group with the given master control * * All slaves must be the same type (returning the same information - * via info callback). The fucntion doesn't check it, so it's your + * via info callback). The function doesn't check it, so it's your * responsibility. * * Also, some additional limitations: diff --git a/sound/drivers/pcm-indirect2.c b/sound/drivers/pcm-indirect2.c index 3c93c23e4883..e73fafd761b3 100644 --- a/sound/drivers/pcm-indirect2.c +++ b/sound/drivers/pcm-indirect2.c @@ -264,7 +264,7 @@ snd_pcm_indirect2_playback_transfer(struct snd_pcm_substream *substream, if (diff < -(snd_pcm_sframes_t) (runtime->boundary / 2)) diff += runtime->boundary; /* number of bytes "added" by ALSA increases the number of - * bytes which are ready to "be transfered to HW"/"played" + * bytes which are ready to "be transferred to HW"/"played" * Then, set rec->appl_ptr to not count bytes twice next time. */ rec->sw_ready += (int)frames_to_bytes(runtime, diff); @@ -330,7 +330,7 @@ snd_pcm_indirect2_playback_transfer(struct snd_pcm_substream *substream, /* copy bytes from intermediate buffer position sw_data to the * HW and return number of bytes actually written * Furthermore, set hw_ready to 0, if the fifo isn't empty - * now => more could be transfered to fifo + * now => more could be transferred to fifo */ bytes = copy(substream, rec, bytes); rec->bytes2hw += bytes; diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index 35a2f71a6af5..5e897b236cec 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -1189,7 +1189,7 @@ void vx_pcm_update_intr(struct vx_core *chip, unsigned int events) /* - * vx_init_audio_io - check the availabe audio i/o and allocate pipe arrays + * vx_init_audio_io - check the available audio i/o and allocate pipe arrays */ static int vx_init_audio_io(struct vx_core *chip) { diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 0c40951b6523..5d61f5a29130 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -370,7 +370,7 @@ init_arrays(struct snd_emu8000 *emu) /* * Size the onboard memory. - * This is written so as not to need arbitary delays after the write. It + * This is written so as not to need arbitrary delays after the write. It * seems that the only way to do this is to use the one channel and keep * reallocating between read and write. */ diff --git a/sound/isa/wavefront/wavefront_midi.c b/sound/isa/wavefront/wavefront_midi.c index f14a7c0b6998..65329f3abc30 100644 --- a/sound/isa/wavefront/wavefront_midi.c +++ b/sound/isa/wavefront/wavefront_midi.c @@ -537,7 +537,7 @@ snd_wavefront_midi_start (snd_wavefront_card_t *card) } /* Turn on Virtual MIDI, but first *always* turn it off, - since otherwise consectutive reloads of the driver will + since otherwise consecutive reloads of the driver will never cause the hardware to generate the initial "internal" or "external" source bytes in the MIDI data stream. This is pretty important, since the internal hardware generally will diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 9191b32d9130..2a42cc377957 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -424,7 +424,7 @@ void snd_wss_mce_down(struct snd_wss *chip) /* * Wait for (possible -- during init auto-calibration may not be set) - * calibration process to start. Needs upto 5 sample periods on AD1848 + * calibration process to start. Needs up to 5 sample periods on AD1848 * which at the slowest possible rate of 5.5125 kHz means 907 us. */ msleep(1); diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index 76c090218073..6c93e051f9ae 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -22,10 +22,6 @@ config SOUND_VWSND <file:Documentation/sound/oss/vwsnd> for more info on this driver's capabilities. -config SOUND_AU1550_AC97 - tristate "Au1550/Au1200 AC97 Sound" - depends on SOC_AU1550 || SOC_AU1200 - config SOUND_MSNDCLAS tristate "Support for Turtle Beach MultiSound Classic, Tahiti, Monterey" depends on (m || !STANDALONE) && ISA diff --git a/sound/oss/Makefile b/sound/oss/Makefile index 90ffb99c6b17..77f21b68bf0f 100644 --- a/sound/oss/Makefile +++ b/sound/oss/Makefile @@ -25,7 +25,6 @@ obj-$(CONFIG_SOUND_WAVEARTIST) += waveartist.o obj-$(CONFIG_SOUND_MSNDCLAS) += msnd.o msnd_classic.o obj-$(CONFIG_SOUND_MSNDPIN) += msnd.o msnd_pinnacle.o obj-$(CONFIG_SOUND_VWSND) += vwsnd.o -obj-$(CONFIG_SOUND_AU1550_AC97) += au1550_ac97.o ac97_codec.o obj-$(CONFIG_SOUND_BCM_CS4297A) += swarm_cs4297a.o obj-$(CONFIG_DMASOUND) += dmasound/ diff --git a/sound/oss/ac97_codec.c b/sound/oss/ac97_codec.c deleted file mode 100644 index 854c303264dc..000000000000 --- a/sound/oss/ac97_codec.c +++ /dev/null @@ -1,1203 +0,0 @@ -/* - * ac97_codec.c: Generic AC97 mixer/modem module - * - * Derived from ac97 mixer in maestro and trident driver. - * - * Copyright 2000 Silicon Integrated System Corporation - * - * 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., 675 Mass Ave, Cambridge, MA 02139, USA. - * - ************************************************************************** - * - * The Intel Audio Codec '97 specification is available at: - * http://download.intel.com/support/motherboards/desktop/sb/ac97_r23.pdf - * - ************************************************************************** - * - * History - * May 02, 2003 Liam Girdwood <lrg@slimlogic.co.uk> - * Removed non existant WM9700 - * Added support for WM9705, WM9708, WM9709, WM9710, WM9711 - * WM9712 and WM9717 - * Mar 28, 2002 Randolph Bentson <bentson@holmsjoen.com> - * corrections to support WM9707 in ViewPad 1000 - * v0.4 Mar 15 2000 Ollie Lho - * dual codecs support verified with 4 channels output - * v0.3 Feb 22 2000 Ollie Lho - * bug fix for record mask setting - * v0.2 Feb 10 2000 Ollie Lho - * add ac97_read_proc for /proc/driver/{vendor}/ac97 - * v0.1 Jan 14 2000 Ollie Lho <ollie@sis.com.tw> - * Isolated from trident.c to support multiple ac97 codec - */ -#include <linux/module.h> -#include <linux/kernel.h> -#include <linux/slab.h> -#include <linux/string.h> -#include <linux/errno.h> -#include <linux/bitops.h> -#include <linux/delay.h> -#include <linux/pci.h> -#include <linux/ac97_codec.h> -#include <asm/uaccess.h> -#include <linux/mutex.h> - -#define CODEC_ID_BUFSZ 14 - -static int ac97_read_mixer(struct ac97_codec *codec, int oss_channel); -static void ac97_write_mixer(struct ac97_codec *codec, int oss_channel, - unsigned int left, unsigned int right); -static void ac97_set_mixer(struct ac97_codec *codec, unsigned int oss_mixer, unsigned int val ); -static int ac97_recmask_io(struct ac97_codec *codec, int rw, int mask); -static int ac97_mixer_ioctl(struct ac97_codec *codec, unsigned int cmd, unsigned long arg); - -static int ac97_init_mixer(struct ac97_codec *codec); - -static int wolfson_init03(struct ac97_codec * codec); -static int wolfson_init04(struct ac97_codec * codec); -static int wolfson_init05(struct ac97_codec * codec); -static int wolfson_init11(struct ac97_codec * codec); -static int wolfson_init13(struct ac97_codec * codec); -static int tritech_init(struct ac97_codec * codec); -static int tritech_maestro_init(struct ac97_codec * codec); -static int sigmatel_9708_init(struct ac97_codec *codec); -static int sigmatel_9721_init(struct ac97_codec *codec); -static int sigmatel_9744_init(struct ac97_codec *codec); -static int ad1886_init(struct ac97_codec *codec); -static int eapd_control(struct ac97_codec *codec, int); -static int crystal_digital_control(struct ac97_codec *codec, int slots, int rate, int mode); -static int cmedia_init(struct ac97_codec * codec); -static int cmedia_digital_control(struct ac97_codec *codec, int slots, int rate, int mode); -static int generic_digital_control(struct ac97_codec *codec, int slots, int rate, int mode); - - -/* - * AC97 operations. - * - * If you are adding a codec then you should be able to use - * eapd_ops - any codec that supports EAPD amp control (most) - * null_ops - any ancient codec that supports nothing - * - * The three functions are - * init - used for non AC97 standard initialisation - * amplifier - used to do amplifier control (1=on 0=off) - * digital - switch to digital modes (0 = analog) - * - * Not all codecs support all features, not all drivers use all the - * operations yet - */ - -static struct ac97_ops null_ops = { NULL, NULL, NULL }; -static struct ac97_ops default_ops = { NULL, eapd_control, NULL }; -static struct ac97_ops default_digital_ops = { NULL, eapd_control, generic_digital_control}; -static struct ac97_ops wolfson_ops03 = { wolfson_init03, NULL, NULL }; -static struct ac97_ops wolfson_ops04 = { wolfson_init04, NULL, NULL }; -static struct ac97_ops wolfson_ops05 = { wolfson_init05, NULL, NULL }; -static struct ac97_ops wolfson_ops11 = { wolfson_init11, NULL, NULL }; -static struct ac97_ops wolfson_ops13 = { wolfson_init13, NULL, NULL }; -static struct ac97_ops tritech_ops = { tritech_init, NULL, NULL }; -static struct ac97_ops tritech_m_ops = { tritech_maestro_init, NULL, NULL }; -static struct ac97_ops sigmatel_9708_ops = { sigmatel_9708_init, NULL, NULL }; -static struct ac97_ops sigmatel_9721_ops = { sigmatel_9721_init, NULL, NULL }; -static struct ac97_ops sigmatel_9744_ops = { sigmatel_9744_init, NULL, NULL }; -static struct ac97_ops crystal_digital_ops = { NULL, eapd_control, crystal_digital_control }; -static struct ac97_ops ad1886_ops = { ad1886_init, eapd_control, NULL }; -static struct ac97_ops cmedia_ops = { NULL, eapd_control, NULL}; -static struct ac97_ops cmedia_digital_ops = { cmedia_init, eapd_control, cmedia_digital_control}; - -/* sorted by vendor/device id */ -static const struct { - u32 id; - char *name; - struct ac97_ops *ops; - int flags; -} ac97_codec_ids[] = { - {0x41445303, "Analog Devices AD1819", &null_ops}, - {0x41445340, "Analog Devices AD1881", &null_ops}, - {0x41445348, "Analog Devices AD1881A", &null_ops}, - {0x41445360, "Analog Devices AD1885", &default_ops}, - {0x41445361, "Analog Devices AD1886", &ad1886_ops}, - {0x41445370, "Analog Devices AD1981", &null_ops}, - {0x41445372, "Analog Devices AD1981A", &null_ops}, - {0x41445374, "Analog Devices AD1981B", &null_ops}, - {0x41445460, "Analog Devices AD1885", &default_ops}, - {0x41445461, "Analog Devices AD1886", &ad1886_ops}, - {0x414B4D00, "Asahi Kasei AK4540", &null_ops}, - {0x414B4D01, "Asahi Kasei AK4542", &null_ops}, - {0x414B4D02, "Asahi Kasei AK4543", &null_ops}, - {0x414C4326, "ALC100P", &null_ops}, - {0x414C4710, "ALC200/200P", &null_ops}, - {0x414C4720, "ALC650", &default_digital_ops}, - {0x434D4941, "CMedia", &cmedia_ops, AC97_NO_PCM_VOLUME }, - {0x434D4942, "CMedia", &cmedia_ops, AC97_NO_PCM_VOLUME }, - {0x434D4961, "CMedia", &cmedia_digital_ops, AC97_NO_PCM_VOLUME }, - {0x43525900, "Cirrus Logic CS4297", &default_ops}, - {0x43525903, "Cirrus Logic CS4297", &default_ops}, - {0x43525913, "Cirrus Logic CS4297A rev A", &default_ops}, - {0x43525914, "Cirrus Logic CS4297A rev B", &default_ops}, - {0x43525923, "Cirrus Logic CS4298", &null_ops}, - {0x4352592B, "Cirrus Logic CS4294", &null_ops}, - {0x4352592D, "Cirrus Logic CS4294", &null_ops}, - {0x43525931, "Cirrus Logic CS4299 rev A", &crystal_digital_ops}, - {0x43525933, "Cirrus Logic CS4299 rev C", &crystal_digital_ops}, - {0x43525934, "Cirrus Logic CS4299 rev D", &crystal_digital_ops}, - {0x43585430, "CXT48", &default_ops, AC97_DELUDED_MODEM }, - {0x43585442, "CXT66", &default_ops, AC97_DELUDED_MODEM }, - {0x44543031, "Diamond Technology DT0893", &default_ops}, - {0x45838308, "ESS Allegro ES1988", &null_ops}, - {0x49434511, "ICE1232", &null_ops}, /* I hope --jk */ - {0x4e534331, "National Semiconductor LM4549", &null_ops}, - {0x53494c22, "Silicon Laboratory Si3036", &null_ops}, - {0x53494c23, "Silicon Laboratory Si3038", &null_ops}, - {0x545200FF, "TriTech TR?????", &tritech_m_ops}, - {0x54524102, "TriTech TR28022", &null_ops}, - {0x54524103, "TriTech TR28023", &null_ops}, - {0x54524106, "TriTech TR28026", &null_ops}, - {0x54524108, "TriTech TR28028", &tritech_ops}, - {0x54524123, "TriTech TR A5", &null_ops}, - {0x574D4C03, "Wolfson WM9703/07/08/17", &wolfson_ops03}, - {0x574D4C04, "Wolfson WM9704M/WM9704Q", &wolfson_ops04}, - {0x574D4C05, "Wolfson WM9705/WM9710", &wolfson_ops05}, - {0x574D4C09, "Wolfson WM9709", &null_ops}, - {0x574D4C12, "Wolfson WM9711/9712", &wolfson_ops11}, - {0x574D4C13, "Wolfson WM9713", &wolfson_ops13, AC97_DEFAULT_POWER_OFF}, - {0x83847600, "SigmaTel STAC????", &null_ops}, - {0x83847604, "SigmaTel STAC9701/3/4/5", &null_ops}, - {0x83847605, "SigmaTel STAC9704", &null_ops}, - {0x83847608, "SigmaTel STAC9708", &sigmatel_9708_ops}, - {0x83847609, "SigmaTel STAC9721/23", &sigmatel_9721_ops}, - {0x83847644, "SigmaTel STAC9744/45", &sigmatel_9744_ops}, - {0x83847652, "SigmaTel STAC9752/53", &default_ops}, - {0x83847656, "SigmaTel STAC9756/57", &sigmatel_9744_ops}, - {0x83847666, "SigmaTel STAC9750T", &sigmatel_9744_ops}, - {0x83847684, "SigmaTel STAC9783/84?", &null_ops}, - {0x57454301, "Winbond 83971D", &null_ops}, -}; - -/* this table has default mixer values for all OSS mixers. */ -static struct mixer_defaults { - int mixer; - unsigned int value; -} mixer_defaults[SOUND_MIXER_NRDEVICES] = { - /* all values 0 -> 100 in bytes */ - {SOUND_MIXER_VOLUME, 0x4343}, - {SOUND_MIXER_BASS, 0x4343}, - {SOUND_MIXER_TREBLE, 0x4343}, - {SOUND_MIXER_PCM, 0x4343}, - {SOUND_MIXER_SPEAKER, 0x4343}, - {SOUND_MIXER_LINE, 0x4343}, - {SOUND_MIXER_MIC, 0x0000}, - {SOUND_MIXER_CD, 0x4343}, - {SOUND_MIXER_ALTPCM, 0x4343}, - {SOUND_MIXER_IGAIN, 0x4343}, - {SOUND_MIXER_LINE1, 0x4343}, - {SOUND_MIXER_PHONEIN, 0x4343}, - {SOUND_MIXER_PHONEOUT, 0x4343}, - {SOUND_MIXER_VIDEO, 0x4343}, - {-1,0} -}; - -/* table to scale scale from OSS mixer value to AC97 mixer register value */ -static struct ac97_mixer_hw { - unsigned char offset; - int scale; -} ac97_hw[SOUND_MIXER_NRDEVICES]= { - [SOUND_MIXER_VOLUME] = {AC97_MASTER_VOL_STEREO,64}, - [SOUND_MIXER_BASS] = {AC97_MASTER_TONE, 16}, - [SOUND_MIXER_TREBLE] = {AC97_MASTER_TONE, 16}, - [SOUND_MIXER_PCM] = {AC97_PCMOUT_VOL, 32}, - [SOUND_MIXER_SPEAKER] = {AC97_PCBEEP_VOL, 16}, - [SOUND_MIXER_LINE] = {AC97_LINEIN_VOL, 32}, - [SOUND_MIXER_MIC] = {AC97_MIC_VOL, 32}, - [SOUND_MIXER_CD] = {AC97_CD_VOL, 32}, - [SOUND_MIXER_ALTPCM] = {AC97_HEADPHONE_VOL, 64}, - [SOUND_MIXER_IGAIN] = {AC97_RECORD_GAIN, 16}, - [SOUND_MIXER_LINE1] = {AC97_AUX_VOL, 32}, - [SOUND_MIXER_PHONEIN] = {AC97_PHONE_VOL, 32}, - [SOUND_MIXER_PHONEOUT] = {AC97_MASTER_VOL_MONO, 64}, - [SOUND_MIXER_VIDEO] = {AC97_VIDEO_VOL, 32}, -}; - -/* the following tables allow us to go from OSS <-> ac97 quickly. */ -enum ac97_recsettings { - AC97_REC_MIC=0, - AC97_REC_CD, - AC97_REC_VIDEO, - AC97_REC_AUX, - AC97_REC_LINE, - AC97_REC_STEREO, /* combination of all enabled outputs.. */ - AC97_REC_MONO, /*.. or the mono equivalent */ - AC97_REC_PHONE -}; - -static const unsigned int ac97_rm2oss[] = { - [AC97_REC_MIC] = SOUND_MIXER_MIC, - [AC97_REC_CD] = SOUND_MIXER_CD, - [AC97_REC_VIDEO] = SOUND_MIXER_VIDEO, - [AC97_REC_AUX] = SOUND_MIXER_LINE1, - [AC97_REC_LINE] = SOUND_MIXER_LINE, - [AC97_REC_STEREO]= SOUND_MIXER_IGAIN, - [AC97_REC_PHONE] = SOUND_MIXER_PHONEIN -}; - -/* indexed by bit position */ -static const unsigned int ac97_oss_rm[] = { - [SOUND_MIXER_MIC] = AC97_REC_MIC, - [SOUND_MIXER_CD] = AC97_REC_CD, - [SOUND_MIXER_VIDEO] = AC97_REC_VIDEO, - [SOUND_MIXER_LINE1] = AC97_REC_AUX, - [SOUND_MIXER_LINE] = AC97_REC_LINE, - [SOUND_MIXER_IGAIN] = AC97_REC_STEREO, - [SOUND_MIXER_PHONEIN] = AC97_REC_PHONE -}; - -static LIST_HEAD(codecs); -static LIST_HEAD(codec_drivers); -static DEFINE_MUTEX(codec_mutex); - -/* reads the given OSS mixer from the ac97 the caller must have insured that the ac97 knows - about that given mixer, and should be holding a spinlock for the card */ -static int ac97_read_mixer(struct ac97_codec *codec, int oss_channel) -{ - u16 val; - int ret = 0; - int scale; - struct ac97_mixer_hw *mh = &ac97_hw[oss_channel]; - - val = codec->codec_read(codec , mh->offset); - - if (val & AC97_MUTE) { - ret = 0; - } else if (AC97_STEREO_MASK & (1 << oss_channel)) { - /* nice stereo mixers .. */ - int left,right; - - left = (val >> 8) & 0x7f; - right = val & 0x7f; - - if (oss_channel == SOUND_MIXER_IGAIN) { - right = (right * 100) / mh->scale; - left = (left * 100) / mh->scale; - } else { - /* these may have 5 or 6 bit resolution */ - if(oss_channel == SOUND_MIXER_VOLUME || oss_channel == SOUND_MIXER_ALTPCM) - scale = (1 << codec->bit_resolution); - else - scale = mh->scale; - - right = 100 - ((right * 100) / scale); - left = 100 - ((left * 100) / scale); - } - ret = left | (right << 8); - } else if (oss_channel == SOUND_MIXER_SPEAKER) { - ret = 100 - ((((val & 0x1e)>>1) * 100) / mh->scale); - } else if (oss_channel == SOUND_MIXER_PHONEIN) { - ret = 100 - (((val & 0x1f) * 100) / mh->scale); - } else if (oss_channel == SOUND_MIXER_PHONEOUT) { - scale = (1 << codec->bit_resolution); - ret = 100 - (((val & 0x1f) * 100) / scale); - } else if (oss_channel == SOUND_MIXER_MIC) { - ret = 100 - (((val & 0x1f) * 100) / mh->scale); - /* the low bit is optional in the tone sliders and masking - it lets us avoid the 0xf 'bypass'.. */ - } else if (oss_channel == SOUND_MIXER_BASS) { - ret = 100 - ((((val >> 8) & 0xe) * 100) / mh->scale); - } else if (oss_channel == SOUND_MIXER_TREBLE) { - ret = 100 - (((val & 0xe) * 100) / mh->scale); - } - -#ifdef DEBUG - printk("ac97_codec: read OSS mixer %2d (%s ac97 register 0x%02x), " - "0x%04x -> 0x%04x\n", - oss_channel, codec->id ? "Secondary" : "Primary", - mh->offset, val, ret); -#endif - - return ret; -} - -/* write the OSS encoded volume to the given OSS encoded mixer, again caller's job to - make sure all is well in arg land, call with spinlock held */ -static void ac97_write_mixer(struct ac97_codec *codec, int oss_channel, - unsigned int left, unsigned int right) -{ - u16 val = 0; - int scale; - struct ac97_mixer_hw *mh = &ac97_hw[oss_channel]; - -#ifdef DEBUG - printk("ac97_codec: wrote OSS mixer %2d (%s ac97 register 0x%02x), " - "left vol:%2d, right vol:%2d:", - oss_channel, codec->id ? "Secondary" : "Primary", - mh->offset, left, right); -#endif - - if (AC97_STEREO_MASK & (1 << oss_channel)) { - /* stereo mixers */ - if (left == 0 && right == 0) { - val = AC97_MUTE; - } else { - if (oss_channel == SOUND_MIXER_IGAIN) { - right = (right * mh->scale) / 100; - left = (left * mh->scale) / 100; - if (right >= mh->scale) - right = mh->scale-1; - if (left >= mh->scale) - left = mh->scale-1; - } else { - /* these may have 5 or 6 bit resolution */ - if (oss_channel == SOUND_MIXER_VOLUME || - oss_channel == SOUND_MIXER_ALTPCM) - scale = (1 << codec->bit_resolution); - else - scale = mh->scale; - - right = ((100 - right) * scale) / 100; - left = ((100 - left) * scale) / 100; - if (right >= scale) - right = scale-1; - if (left >= scale) - left = scale-1; - } - val = (left << 8) | right; - } - } else if (oss_channel == SOUND_MIXER_BASS) { - val = codec->codec_read(codec , mh->offset) & ~0x0f00; - left = ((100 - left) * mh->scale) / 100; - if (left >= mh->scale) - left = mh->scale-1; - val |= (left << 8) & 0x0e00; - } else if (oss_channel == SOUND_MIXER_TREBLE) { - val = codec->codec_read(codec , mh->offset) & ~0x000f; - left = ((100 - left) * mh->scale) / 100; - if (left >= mh->scale) - left = mh->scale-1; - val |= left & 0x000e; - } else if(left == 0) { - val = AC97_MUTE; - } else if (oss_channel == SOUND_MIXER_SPEAKER) { - left = ((100 - left) * mh->scale) / 100; - if (left >= mh->scale) - left = mh->scale-1; - val = left << 1; - } else if (oss_channel == SOUND_MIXER_PHONEIN) { - left = ((100 - left) * mh->scale) / 100; - if (left >= mh->scale) - left = mh->scale-1; - val = left; - } else if (oss_channel == SOUND_MIXER_PHONEOUT) { - scale = (1 << codec->bit_resolution); - left = ((100 - left) * scale) / 100; - if (left >= mh->scale) - left = mh->scale-1; - val = left; - } else if (oss_channel == SOUND_MIXER_MIC) { - val = codec->codec_read(codec , mh->offset) & ~0x801f; - left = ((100 - left) * mh->scale) / 100; - if (left >= mh->scale) - left = mh->scale-1; - val |= left; - /* the low bit is optional in the tone sliders and masking - it lets us avoid the 0xf 'bypass'.. */ - } -#ifdef DEBUG - printk(" 0x%04x", val); -#endif - - codec->codec_write(codec, mh->offset, val); - -#ifdef DEBUG - val = codec->codec_read(codec, mh->offset); - printk(" -> 0x%04x\n", val); -#endif -} - -/* a thin wrapper for write_mixer */ -static void ac97_set_mixer(struct ac97_codec *codec, unsigned int oss_mixer, unsigned int val ) -{ - unsigned int left,right; - - /* cleanse input a little */ - right = ((val >> 8) & 0xff) ; - left = (val & 0xff) ; - - if (right > 100) right = 100; - if (left > 100) left = 100; - - codec->mixer_state[oss_mixer] = (right << 8) | left; - codec->write_mixer(codec, oss_mixer, left, right); -} - -/* read or write the recmask, the ac97 can really have left and right recording - inputs independantly set, but OSS doesn't seem to want us to express that to - the user. the caller guarantees that we have a supported bit set, and they - must be holding the card's spinlock */ -static int ac97_recmask_io(struct ac97_codec *codec, int rw, int mask) -{ - unsigned int val; - - if (rw) { - /* read it from the card */ - val = codec->codec_read(codec, AC97_RECORD_SELECT); -#ifdef DEBUG - printk("ac97_codec: ac97 recmask to set to 0x%04x\n", val); -#endif - return (1 << ac97_rm2oss[val & 0x07]); - } - - /* else, write the first set in the mask as the - output */ - /* clear out current set value first (AC97 supports only 1 input!) */ - val = (1 << ac97_rm2oss[codec->codec_read(codec, AC97_RECORD_SELECT) & 0x07]); - if (mask != val) - mask &= ~val; - - val = ffs(mask); - val = ac97_oss_rm[val-1]; - val |= val << 8; /* set both channels */ - -#ifdef DEBUG - printk("ac97_codec: setting ac97 recmask to 0x%04x\n", val); -#endif - - codec->codec_write(codec, AC97_RECORD_SELECT, val); - - return 0; -}; - -static int ac97_mixer_ioctl(struct ac97_codec *codec, unsigned int cmd, unsigned long arg) -{ - int i, val = 0; - - if (cmd == SOUND_MIXER_INFO) { - mixer_info info; - memset(&info, 0, sizeof(info)); - strlcpy(info.id, codec->name, sizeof(info.id)); - strlcpy(info.name, codec->name, sizeof(info.name)); - info.modify_counter = codec->modcnt; - if (copy_to_user((void __user *)arg, &info, sizeof(info))) - return -EFAULT; - return 0; - } - if (cmd == SOUND_OLD_MIXER_INFO) { - _old_mixer_info info; - memset(&info, 0, sizeof(info)); - strlcpy(info.id, codec->name, sizeof(info.id)); - strlcpy(info.name, codec->name, sizeof(info.name)); - if (copy_to_user((void __user *)arg, &info, sizeof(info))) - return -EFAULT; - return 0; - } - - if (_IOC_TYPE(cmd) != 'M' || _SIOC_SIZE(cmd) != sizeof(int)) - return -EINVAL; - - if (cmd == OSS_GETVERSION) - return put_user(SOUND_VERSION, (int __user *)arg); - - if (_SIOC_DIR(cmd) == _SIOC_READ) { - switch (_IOC_NR(cmd)) { - case SOUND_MIXER_RECSRC: /* give them the current record source */ - if (!codec->recmask_io) { - val = 0; - } else { - val = codec->recmask_io(codec, 1, 0); - } - break; - - case SOUND_MIXER_DEVMASK: /* give them the supported mixers */ - val = codec->supported_mixers; - break; - - case SOUND_MIXER_RECMASK: /* Arg contains a bit for each supported recording source */ - val = codec->record_sources; - break; - - case SOUND_MIXER_STEREODEVS: /* Mixer channels supporting stereo */ - val = codec->stereo_mixers; - break; - - case SOUND_MIXER_CAPS: - val = SOUND_CAP_EXCL_INPUT; - break; - - default: /* read a specific mixer */ - i = _IOC_NR(cmd); - - if (!supported_mixer(codec, i)) - return -EINVAL; - - /* do we ever want to touch the hardware? */ - /* val = codec->read_mixer(codec, i); */ - val = codec->mixer_state[i]; - break; - } - return put_user(val, (int __user *)arg); - } - - if (_SIOC_DIR(cmd) == (_SIOC_WRITE|_SIOC_READ)) { - codec->modcnt++; - if (get_user(val, (int __user *)arg)) - return -EFAULT; - - switch (_IOC_NR(cmd)) { - case SOUND_MIXER_RECSRC: /* Arg contains a bit for each recording source */ - if (!codec->recmask_io) return -EINVAL; - if (!val) return 0; - if (!(val &= codec->record_sources)) return -EINVAL; - - codec->recmask_io(codec, 0, val); - - return 0; - default: /* write a specific mixer */ - i = _IOC_NR(cmd); - - if (!supported_mixer(codec, i)) - return -EINVAL; - - ac97_set_mixer(codec, i, val); - - return 0; - } - } - return -EINVAL; -} - -/** - * codec_id - Turn id1/id2 into a PnP string - * @id1: Vendor ID1 - * @id2: Vendor ID2 - * @buf: CODEC_ID_BUFSZ byte buffer - * - * Fills buf with a zero terminated PnP ident string for the id1/id2 - * pair. For convenience the return is the passed in buffer pointer. - */ - -static char *codec_id(u16 id1, u16 id2, char *buf) -{ - if(id1&0x8080) { - snprintf(buf, CODEC_ID_BUFSZ, "0x%04x:0x%04x", id1, id2); - } else { - buf[0] = (id1 >> 8); - buf[1] = (id1 & 0xFF); - buf[2] = (id2 >> 8); - snprintf(buf+3, CODEC_ID_BUFSZ - 3, "%d", id2&0xFF); - } - return buf; -} - -/** - * ac97_check_modem - Check if the Codec is a modem - * @codec: codec to check - * - * Return true if the device is an AC97 1.0 or AC97 2.0 modem - */ - -static int ac97_check_modem(struct ac97_codec *codec) -{ - /* Check for an AC97 1.0 soft modem (ID1) */ - if(codec->codec_read(codec, AC97_RESET) & 2) - return 1; - /* Check for an AC97 2.x soft modem */ - codec->codec_write(codec, AC97_EXTENDED_MODEM_ID, 0L); - if(codec->codec_read(codec, AC97_EXTENDED_MODEM_ID) & 1) - return 1; - return 0; -} - - -/** - * ac97_alloc_codec - Allocate an AC97 codec - * - * Returns a new AC97 codec structure. AC97 codecs may become - * refcounted soon so this interface is needed. Returns with - * one reference taken. - */ - -struct ac97_codec *ac97_alloc_codec(void) -{ - struct ac97_codec *codec = kzalloc(sizeof(struct ac97_codec), GFP_KERNEL); - if(!codec) - return NULL; - - spin_lock_init(&codec->lock); - INIT_LIST_HEAD(&codec->list); - return codec; -} - -EXPORT_SYMBOL(ac97_alloc_codec); - -/** - * ac97_release_codec - Release an AC97 codec - * @codec: codec to release - * - * Release an allocated AC97 codec. This will be refcounted in - * time but for the moment is trivial. Calls the unregister - * handler if the codec is now defunct. - */ - -void ac97_release_codec(struct ac97_codec *codec) -{ - /* Remove from the list first, we don't want to be - "rediscovered" */ - mutex_lock(&codec_mutex); - list_del(&codec->list); - mutex_unlock(&codec_mutex); - /* - * The driver needs to deal with internal - * locking to avoid accidents here. - */ - if(codec->driver) - codec->driver->remove(codec, codec->driver); - kfree(codec); -} - -EXPORT_SYMBOL(ac97_release_codec); - -/** - * ac97_probe_codec - Initialize and setup AC97-compatible codec - * @codec: (in/out) Kernel info for a single AC97 codec - * - * Reset the AC97 codec, then initialize the mixer and - * the rest of the @codec structure. - * - * The codec_read and codec_write fields of @codec are - * required to be setup and working when this function - * is called. All other fields are set by this function. - * - * codec_wait field of @codec can optionally be provided - * when calling this function. If codec_wait is not %NULL, - * this function will call codec_wait any time it is - * necessary to wait for the audio chip to reach the - * codec-ready state. If codec_wait is %NULL, then - * the default behavior is to call schedule_timeout. - * Currently codec_wait is used to wait for AC97 codec - * reset to complete. - * - * Some codecs will power down when a register reset is - * performed. We now check for such codecs. - * - * Returns 1 (true) on success, or 0 (false) on failure. - */ - -int ac97_probe_codec(struct ac97_codec *codec) -{ - u16 id1, id2; - u16 audio; - int i; - char cidbuf[CODEC_ID_BUFSZ]; - u16 f; - struct list_head *l; - struct ac97_driver *d; - - /* wait for codec-ready state */ - if (codec->codec_wait) - codec->codec_wait(codec); - else - udelay(10); - - /* will the codec power down if register reset ? */ - id1 = codec->codec_read(codec, AC97_VENDOR_ID1); - id2 = codec->codec_read(codec, AC97_VENDOR_ID2); - codec->name = NULL; - codec->codec_ops = &null_ops; - for (i = 0; i < ARRAY_SIZE(ac97_codec_ids); i++) { - if (ac97_codec_ids[i].id == ((id1 << 16) | id2)) { - codec->type = ac97_codec_ids[i].id; - codec->name = ac97_codec_ids[i].name; - codec->codec_ops = ac97_codec_ids[i].ops; - codec->flags = ac97_codec_ids[i].flags; - break; - } - } - - codec->model = (id1 << 16) | id2; - if ((codec->flags & AC97_DEFAULT_POWER_OFF) == 0) { - /* reset codec and wait for the ready bit before we continue */ - codec->codec_write(codec, AC97_RESET, 0L); - if (codec->codec_wait) - codec->codec_wait(codec); - else - udelay(10); - } - - /* probing AC97 codec, AC97 2.0 says that bit 15 of register 0x00 (reset) should - * be read zero. - * - * FIXME: is the following comment outdated? -jgarzik - * Probing of AC97 in this way is not reliable, it is not even SAFE !! - */ - if ((audio = codec->codec_read(codec, AC97_RESET)) & 0x8000) { - printk(KERN_ERR "ac97_codec: %s ac97 codec not present\n", - (codec->id & 0x2) ? (codec->id&1 ? "4th" : "Tertiary") - : (codec->id&1 ? "Secondary": "Primary")); - return 0; - } - - /* probe for Modem Codec */ - codec->modem = ac97_check_modem(codec); - - /* enable SPDIF */ - f = codec->codec_read(codec, AC97_EXTENDED_STATUS); - if((codec->codec_ops == &null_ops) && (f & 4)) - codec->codec_ops = &default_digital_ops; - - /* A device which thinks its a modem but isnt */ - if(codec->flags & AC97_DELUDED_MODEM) - codec->modem = 0; - - if (codec->name == NULL) - codec->name = "Unknown"; - printk(KERN_INFO "ac97_codec: AC97 %s codec, id: %s (%s)\n", - codec->modem ? "Modem" : (audio ? "Audio" : ""), - codec_id(id1, id2, cidbuf), codec->name); - - if(!ac97_init_mixer(codec)) - return 0; - - /* - * Attach last so the caller can override the mixer - * callbacks. - */ - - mutex_lock(&codec_mutex); - list_add(&codec->list, &codecs); - - list_for_each(l, &codec_drivers) { - d = list_entry(l, struct ac97_driver, list); - if ((codec->model ^ d->codec_id) & d->codec_mask) - continue; - if(d->probe(codec, d) == 0) - { - codec->driver = d; - break; - } - } - - mutex_unlock(&codec_mutex); - return 1; -} - -static int ac97_init_mixer(struct ac97_codec *codec) -{ - u16 cap; - int i; - - cap = codec->codec_read(codec, AC97_RESET); - - /* mixer masks */ - codec->supported_mixers = AC97_SUPPORTED_MASK; - codec->stereo_mixers = AC97_STEREO_MASK; - codec->record_sources = AC97_RECORD_MASK; - if (!(cap & 0x04)) - codec->supported_mixers &= ~(SOUND_MASK_BASS|SOUND_MASK_TREBLE); - if (!(cap & 0x10)) - codec->supported_mixers &= ~SOUND_MASK_ALTPCM; - - - /* detect bit resolution */ - codec->codec_write(codec, AC97_MASTER_VOL_STEREO, 0x2020); - if(codec->codec_read(codec, AC97_MASTER_VOL_STEREO) == 0x2020) - codec->bit_resolution = 6; - else - codec->bit_resolution = 5; - - /* generic OSS to AC97 wrapper */ - codec->read_mixer = ac97_read_mixer; - codec->write_mixer = ac97_write_mixer; - codec->recmask_io = ac97_recmask_io; - codec->mixer_ioctl = ac97_mixer_ioctl; - - /* initialize mixer channel volumes */ - for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) { - struct mixer_defaults *md = &mixer_defaults[i]; - if (md->mixer == -1) - break; - if (!supported_mixer(codec, md->mixer)) - continue; - ac97_set_mixer(codec, md->mixer, md->value); - } - - /* codec specific initialization for 4-6 channel output or secondary codec stuff */ - if (codec->codec_ops->init != NULL) { - codec->codec_ops->init(codec); - } - - /* - * Volume is MUTE only on this device. We have to initialise - * it but its useless beyond that. - */ - if(codec->flags & AC97_NO_PCM_VOLUME) - { - codec->supported_mixers &= ~SOUND_MASK_PCM; - printk(KERN_WARNING "AC97 codec does not have proper volume support.\n"); - } - return 1; -} - -#define AC97_SIGMATEL_ANALOG 0x6c /* Analog Special */ -#define AC97_SIGMATEL_DAC2INVERT 0x6e -#define AC97_SIGMATEL_BIAS1 0x70 -#define AC97_SIGMATEL_BIAS2 0x72 -#define AC97_SIGMATEL_MULTICHN 0x74 /* Multi-Channel programming */ -#define AC97_SIGMATEL_CIC1 0x76 -#define AC97_SIGMATEL_CIC2 0x78 - - -static int sigmatel_9708_init(struct ac97_codec * codec) -{ - u16 codec72, codec6c; - - codec72 = codec->codec_read(codec, AC97_SIGMATEL_BIAS2) & 0x8000; - codec6c = codec->codec_read(codec, AC97_SIGMATEL_ANALOG); - - if ((codec72==0) && (codec6c==0)) { - codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba); - codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x1000); - codec->codec_write(codec, AC97_SIGMATEL_BIAS1, 0xabba); - codec->codec_write(codec, AC97_SIGMATEL_BIAS2, 0x0007); - } else if ((codec72==0x8000) && (codec6c==0)) { - codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba); - codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x1001); - codec->codec_write(codec, AC97_SIGMATEL_DAC2INVERT, 0x0008); - } else if ((codec72==0x8000) && (codec6c==0x0080)) { - /* nothing */ - } - codec->codec_write(codec, AC97_SIGMATEL_MULTICHN, 0x0000); - return 0; -} - - -static int sigmatel_9721_init(struct ac97_codec * codec) -{ - /* Only set up secondary codec */ - if (codec->id == 0) - return 0; - - codec->codec_write(codec, AC97_SURROUND_MASTER, 0L); - - /* initialize SigmaTel STAC9721/23 as secondary codec, decoding AC link - sloc 3,4 = 0x01, slot 7,8 = 0x00, */ - codec->codec_write(codec, AC97_SIGMATEL_MULTICHN, 0x00); - - /* we don't have the crystal when we are on an AMR card, so use - BIT_CLK as our clock source. Write the magic word ABBA and read - back to enable register 0x78 */ - codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba); - codec->codec_read(codec, AC97_SIGMATEL_CIC1); - - /* sync all the clocks*/ - codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x3802); - - return 0; -} - - -static int sigmatel_9744_init(struct ac97_codec * codec) -{ - // patch for SigmaTel - codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba); - codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x0000); // is this correct? --jk - codec->codec_write(codec, AC97_SIGMATEL_BIAS1, 0xabba); - codec->codec_write(codec, AC97_SIGMATEL_BIAS2, 0x0002); - codec->codec_write(codec, AC97_SIGMATEL_MULTICHN, 0x0000); - return 0; -} - -static int cmedia_init(struct ac97_codec *codec) -{ - /* Initialise the CMedia 9739 */ - /* - We could set various options here - Register 0x20 bit 0x100 sets mic as center bass - Also do multi_channel_ctrl &=~0x3000 |=0x1000 - - For now we set up the GPIO and PC beep - */ - - u16 v; - - /* MIC */ - codec->codec_write(codec, 0x64, 0x3000); - v = codec->codec_read(codec, 0x64); - v &= ~0x8000; - codec->codec_write(codec, 0x64, v); - codec->codec_write(codec, 0x70, 0x0100); - codec->codec_write(codec, 0x72, 0x0020); - return 0; -} - -#define AC97_WM97XX_FMIXER_VOL 0x72 -#define AC97_WM97XX_RMIXER_VOL 0x74 -#define AC97_WM97XX_TEST 0x5a -#define AC97_WM9704_RPCM_VOL 0x70 -#define AC97_WM9711_OUT3VOL 0x16 - -static int wolfson_init03(struct ac97_codec * codec) -{ - /* this is known to work for the ViewSonic ViewPad 1000 */ - codec->codec_write(codec, AC97_WM97XX_FMIXER_VOL, 0x0808); - codec->codec_write(codec, AC97_GENERAL_PURPOSE, 0x8000); - return 0; -} - -static int wolfson_init04(struct ac97_codec * codec) -{ - codec->codec_write(codec, AC97_WM97XX_FMIXER_VOL, 0x0808); - codec->codec_write(codec, AC97_WM97XX_RMIXER_VOL, 0x0808); - - // patch for DVD noise - codec->codec_write(codec, AC97_WM97XX_TEST, 0x0200); - - // init vol as PCM vol - codec->codec_write(codec, AC97_WM9704_RPCM_VOL, - codec->codec_read(codec, AC97_PCMOUT_VOL)); - - /* set rear surround volume */ - codec->codec_write(codec, AC97_SURROUND_MASTER, 0x0000); - return 0; -} - -/* WM9705, WM9710 */ -static int wolfson_init05(struct ac97_codec * codec) -{ - /* set front mixer volume */ - codec->codec_write(codec, AC97_WM97XX_FMIXER_VOL, 0x0808); - return 0; -} - -/* WM9711, WM9712 */ -static int wolfson_init11(struct ac97_codec * codec) -{ - /* stop pop's during suspend/resume */ - codec->codec_write(codec, AC97_WM97XX_TEST, - codec->codec_read(codec, AC97_WM97XX_TEST) & 0xffbf); - - /* set out3 volume */ - codec->codec_write(codec, AC97_WM9711_OUT3VOL, 0x0808); - return 0; -} - -/* WM9713 */ -static int wolfson_init13(struct ac97_codec * codec) -{ - codec->codec_write(codec, AC97_RECORD_GAIN, 0x00a0); - codec->codec_write(codec, AC97_POWER_CONTROL, 0x0000); - codec->codec_write(codec, AC97_EXTENDED_MODEM_ID, 0xDA00); - codec->codec_write(codec, AC97_EXTEND_MODEM_STAT, 0x3810); - codec->codec_write(codec, AC97_PHONE_VOL, 0x0808); - codec->codec_write(codec, AC97_PCBEEP_VOL, 0x0808); - - return 0; -} - -static int tritech_init(struct ac97_codec * codec) -{ - codec->codec_write(codec, 0x26, 0x0300); - codec->codec_write(codec, 0x26, 0x0000); - codec->codec_write(codec, AC97_SURROUND_MASTER, 0x0000); - codec->codec_write(codec, AC97_RESERVED_3A, 0x0000); - return 0; -} - - -/* copied from drivers/sound/maestro.c */ -static int tritech_maestro_init(struct ac97_codec * codec) -{ - /* no idea what this does */ - codec->codec_write(codec, 0x2A, 0x0001); - codec->codec_write(codec, 0x2C, 0x0000); - codec->codec_write(codec, 0x2C, 0XFFFF); - return 0; -} - - - -/* - * Presario700 workaround - * for Jack Sense/SPDIF Register mis-setting causing - * no audible output - * by Santiago Nullo 04/05/2002 - */ - -#define AC97_AD1886_JACK_SENSE 0x72 - -static int ad1886_init(struct ac97_codec * codec) -{ - /* from AD1886 Specs */ - codec->codec_write(codec, AC97_AD1886_JACK_SENSE, 0x0010); - return 0; -} - - - - -/* - * This is basically standard AC97. It should work as a default for - * almost all modern codecs. Note that some cards wire EAPD *backwards* - * That side of it is up to the card driver not us to cope with. - * - */ - -static int eapd_control(struct ac97_codec * codec, int on) -{ - if(on) - codec->codec_write(codec, AC97_POWER_CONTROL, - codec->codec_read(codec, AC97_POWER_CONTROL)|0x8000); - else - codec->codec_write(codec, AC97_POWER_CONTROL, - codec->codec_read(codec, AC97_POWER_CONTROL)&~0x8000); - return 0; -} - -static int generic_digital_control(struct ac97_codec *codec, int slots, int rate, int mode) -{ - u16 reg; - - reg = codec->codec_read(codec, AC97_SPDIF_CONTROL); - - switch(rate) - { - /* Off by default */ - default: - case 0: - reg = codec->codec_read(codec, AC97_EXTENDED_STATUS); - codec->codec_write(codec, AC97_EXTENDED_STATUS, (reg & ~AC97_EA_SPDIF)); - if(rate == 0) - return 0; - return -EINVAL; - case 1: - reg = (reg & AC97_SC_SPSR_MASK) | AC97_SC_SPSR_48K; - break; - case 2: - reg = (reg & AC97_SC_SPSR_MASK) | AC97_SC_SPSR_44K; - break; - case 3: - reg = (reg & AC97_SC_SPSR_MASK) | AC97_SC_SPSR_32K; - break; - } - - reg &= ~AC97_SC_CC_MASK; - reg |= (mode & AUDIO_CCMASK) << 6; - - if(mode & AUDIO_DIGITAL) - reg |= 2; - if(mode & AUDIO_PRO) - reg |= 1; - if(mode & AUDIO_DRS) - reg |= 0x4000; - - codec->codec_write(codec, AC97_SPDIF_CONTROL, reg); - - reg = codec->codec_read(codec, AC97_EXTENDED_STATUS); - reg &= (AC97_EA_SLOT_MASK); - reg |= AC97_EA_VRA | AC97_EA_SPDIF | slots; - codec->codec_write(codec, AC97_EXTENDED_STATUS, reg); - - reg = codec->codec_read(codec, AC97_EXTENDED_STATUS); - if(!(reg & 0x0400)) - { - codec->codec_write(codec, AC97_EXTENDED_STATUS, reg & ~ AC97_EA_SPDIF); - return -EINVAL; - } - return 0; -} - -/* - * Crystal digital audio control (CS4299) - */ - -static int crystal_digital_control(struct ac97_codec *codec, int slots, int rate, int mode) -{ - u16 cv; - - if(mode & AUDIO_DIGITAL) - return -EINVAL; - - switch(rate) - { - case 0: cv = 0x0; break; /* SPEN off */ - case 48000: cv = 0x8004; break; /* 48KHz digital */ - case 44100: cv = 0x8104; break; /* 44.1KHz digital */ - case 32768: /* 32Khz */ - default: - return -EINVAL; - } - codec->codec_write(codec, 0x68, cv); - return 0; -} - -/* - * CMedia digital audio control - * Needs more work. - */ - -static int cmedia_digital_control(struct ac97_codec *codec, int slots, int rate, int mode) -{ - u16 cv; - - if(mode & AUDIO_DIGITAL) - return -EINVAL; - - switch(rate) - { - case 0: cv = 0x0001; break; /* SPEN off */ - case 48000: cv = 0x0009; break; /* 48KHz digital */ - default: - return -EINVAL; - } - codec->codec_write(codec, 0x2A, 0x05c4); - codec->codec_write(codec, 0x6C, cv); - - /* Switch on mix to surround */ - cv = codec->codec_read(codec, 0x64); - cv &= ~0x0200; - if(mode) - cv |= 0x0200; - codec->codec_write(codec, 0x64, cv); - return 0; -} - - -/* copied from drivers/sound/maestro.c */ -#if 0 /* there has been 1 person on the planet with a pt101 that we - know of. If they care, they can put this back in :) */ -static int pt101_init(struct ac97_codec * codec) -{ - printk(KERN_INFO "ac97_codec: PT101 Codec detected, initializing but _not_ installing mixer device.\n"); - /* who knows.. */ - codec->codec_write(codec, 0x2A, 0x0001); - codec->codec_write(codec, 0x2C, 0x0000); - codec->codec_write(codec, 0x2C, 0xFFFF); - codec->codec_write(codec, 0x10, 0x9F1F); - codec->codec_write(codec, 0x12, 0x0808); - codec->codec_write(codec, 0x14, 0x9F1F); - codec->codec_write(codec, 0x16, 0x9F1F); - codec->codec_write(codec, 0x18, 0x0404); - codec->codec_write(codec, 0x1A, 0x0000); - codec->codec_write(codec, 0x1C, 0x0000); - codec->codec_write(codec, 0x02, 0x0404); - codec->codec_write(codec, 0x04, 0x0808); - codec->codec_write(codec, 0x0C, 0x801F); - codec->codec_write(codec, 0x0E, 0x801F); - return 0; -} -#endif - - -EXPORT_SYMBOL(ac97_probe_codec); - -MODULE_LICENSE("GPL"); - diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c deleted file mode 100644 index a8f626d99c5b..000000000000 --- a/sound/oss/au1550_ac97.c +++ /dev/null @@ -1,2147 +0,0 @@ -/* - * au1550_ac97.c -- Sound driver for Alchemy Au1550 MIPS Internet Edge - * Processor. - * - * Copyright 2004 Embedded Edge, LLC - * dan@embeddededge.com - * - * Mostly copied from the au1000.c driver and some from the - * PowerMac dbdma driver. - * We assume the processor can do memory coherent DMA. - * - * Ported to 2.6 by Matt Porter <mporter@kernel.crashing.org> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN - * NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, - * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT - * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF - * USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON - * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT - * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF - * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - * - * 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., - * 675 Mass Ave, Cambridge, MA 02139, USA. - * - */ - -#undef DEBUG - -#include <linux/module.h> -#include <linux/string.h> -#include <linux/ioport.h> -#include <linux/sched.h> -#include <linux/delay.h> -#include <linux/sound.h> -#include <linux/slab.h> -#include <linux/soundcard.h> -#include <linux/init.h> -#include <linux/interrupt.h> -#include <linux/kernel.h> -#include <linux/poll.h> -#include <linux/bitops.h> -#include <linux/spinlock.h> -#include <linux/ac97_codec.h> -#include <linux/mutex.h> - -#include <asm/io.h> -#include <asm/uaccess.h> -#include <asm/hardirq.h> -#include <asm/mach-au1x00/au1xxx_psc.h> -#include <asm/mach-au1x00/au1xxx_dbdma.h> -#include <asm/mach-au1x00/au1xxx.h> - -#undef OSS_DOCUMENTED_MIXER_SEMANTICS - -/* misc stuff */ -#define POLL_COUNT 0x50000 -#define AC97_EXT_DACS (AC97_EXTID_SDAC | AC97_EXTID_CDAC | AC97_EXTID_LDAC) - -/* The number of DBDMA ring descriptors to allocate. No sense making - * this too large....if you can't keep up with a few you aren't likely - * to be able to with lots of them, either. - */ -#define NUM_DBDMA_DESCRIPTORS 4 - -#define err(format, arg...) printk(KERN_ERR format "\n" , ## arg) - -/* Boot options - * 0 = no VRA, 1 = use VRA if codec supports it - */ -static DEFINE_MUTEX(au1550_ac97_mutex); -static int vra = 1; -module_param(vra, bool, 0); -MODULE_PARM_DESC(vra, "if 1 use VRA if codec supports it"); - -static struct au1550_state { - /* soundcore stuff */ - int dev_audio; - - struct ac97_codec *codec; - unsigned codec_base_caps; /* AC'97 reg 00h, "Reset Register" */ - unsigned codec_ext_caps; /* AC'97 reg 28h, "Extended Audio ID" */ - int no_vra; /* do not use VRA */ - - spinlock_t lock; - struct mutex open_mutex; - struct mutex sem; - fmode_t open_mode; - wait_queue_head_t open_wait; - - struct dmabuf { - u32 dmanr; - unsigned sample_rate; - unsigned src_factor; - unsigned sample_size; - int num_channels; - int dma_bytes_per_sample; - int user_bytes_per_sample; - int cnt_factor; - - void *rawbuf; - unsigned buforder; - unsigned numfrag; - unsigned fragshift; - void *nextIn; - void *nextOut; - int count; - unsigned total_bytes; - unsigned error; - wait_queue_head_t wait; - - /* redundant, but makes calculations easier */ - unsigned fragsize; - unsigned dma_fragsize; - unsigned dmasize; - unsigned dma_qcount; - - /* OSS stuff */ - unsigned mapped:1; - unsigned ready:1; - unsigned stopped:1; - unsigned ossfragshift; - int ossmaxfrags; - unsigned subdivision; - } dma_dac, dma_adc; -} au1550_state; - -static unsigned -ld2(unsigned int x) -{ - unsigned r = 0; - - if (x >= 0x10000) { - x >>= 16; - r += 16; - } - if (x >= 0x100) { - x >>= 8; - r += 8; - } - if (x >= 0x10) { - x >>= 4; - r += 4; - } - if (x >= 4) { - x >>= 2; - r += 2; - } - if (x >= 2) - r++; - return r; -} - -static void -au1550_delay(int msec) -{ - if (in_interrupt()) - return; - - schedule_timeout_uninterruptible(msecs_to_jiffies(msec)); -} - -static u16 -rdcodec(struct ac97_codec *codec, u8 addr) -{ - struct au1550_state *s = codec->private_data; - unsigned long flags; - u32 cmd, val; - u16 data; - int i; - - spin_lock_irqsave(&s->lock, flags); - - for (i = 0; i < POLL_COUNT; i++) { - val = au_readl(PSC_AC97STAT); - au_sync(); - if (!(val & PSC_AC97STAT_CP)) - break; - } - if (i == POLL_COUNT) - err("rdcodec: codec cmd pending expired!"); - - cmd = (u32)PSC_AC97CDC_INDX(addr); - cmd |= PSC_AC97CDC_RD; /* read command */ - au_writel(cmd, PSC_AC97CDC); - au_sync(); - - /* now wait for the data - */ - for (i = 0; i < POLL_COUNT; i++) { - val = au_readl(PSC_AC97STAT); - au_sync(); - if (!(val & PSC_AC97STAT_CP)) - break; - } - if (i == POLL_COUNT) { - err("rdcodec: read poll expired!"); - data = 0; - goto out; - } - - /* wait for command done? - */ - for (i = 0; i < POLL_COUNT; i++) { - val = au_readl(PSC_AC97EVNT); - au_sync(); - if (val & PSC_AC97EVNT_CD) - break; - } - if (i == POLL_COUNT) { - err("rdcodec: read cmdwait expired!"); - data = 0; - goto out; - } - - data = au_readl(PSC_AC97CDC) & 0xffff; - au_sync(); - - /* Clear command done event. - */ - au_writel(PSC_AC97EVNT_CD, PSC_AC97EVNT); - au_sync(); - - out: - spin_unlock_irqrestore(&s->lock, flags); - - return data; -} - - -static void -wrcodec(struct ac97_codec *codec, u8 addr, u16 data) -{ - struct au1550_state *s = codec->private_data; - unsigned long flags; - u32 cmd, val; - int i; - - spin_lock_irqsave(&s->lock, flags); - - for (i = 0; i < POLL_COUNT; i++) { - val = au_readl(PSC_AC97STAT); - au_sync(); - if (!(val & PSC_AC97STAT_CP)) - break; - } - if (i == POLL_COUNT) - err("wrcodec: codec cmd pending expired!"); - - cmd = (u32)PSC_AC97CDC_INDX(addr); - cmd |= (u32)data; - au_writel(cmd, PSC_AC97CDC); - au_sync(); - - for (i = 0; i < POLL_COUNT; i++) { - val = au_readl(PSC_AC97STAT); - au_sync(); - if (!(val & PSC_AC97STAT_CP)) - break; - } - if (i == POLL_COUNT) - err("wrcodec: codec cmd pending expired!"); - - for (i = 0; i < POLL_COUNT; i++) { - val = au_readl(PSC_AC97EVNT); - au_sync(); - if (val & PSC_AC97EVNT_CD) - break; - } - if (i == POLL_COUNT) - err("wrcodec: read cmdwait expired!"); - - /* Clear command done event. - */ - au_writel(PSC_AC97EVNT_CD, PSC_AC97EVNT); - au_sync(); - - spin_unlock_irqrestore(&s->lock, flags); -} - -static void -waitcodec(struct ac97_codec *codec) -{ - u16 temp; - u32 val; - int i; - - /* codec_wait is used to wait for a ready state after - * an AC97C_RESET. - */ - au1550_delay(10); - - /* first poll the CODEC_READY tag bit - */ - for (i = 0; i < POLL_COUNT; i++) { - val = au_readl(PSC_AC97STAT); - au_sync(); - if (val & PSC_AC97STAT_CR) - break; - } - if (i == POLL_COUNT) { - err("waitcodec: CODEC_READY poll expired!"); - return; - } - - /* get AC'97 powerdown control/status register - */ - temp = rdcodec(codec, AC97_POWER_CONTROL); - - /* If anything is powered down, power'em up - */ - if (temp & 0x7f00) { - /* Power on - */ - wrcodec(codec, AC97_POWER_CONTROL, 0); - au1550_delay(100); - - /* Reread - */ - temp = rdcodec(codec, AC97_POWER_CONTROL); - } - - /* Check if Codec REF,ANL,DAC,ADC ready - */ - if ((temp & 0x7f0f) != 0x000f) - err("codec reg 26 status (0x%x) not ready!!", temp); -} - -/* stop the ADC before calling */ -static void -set_adc_rate(struct au1550_state *s, unsigned rate) -{ - struct dmabuf *adc = &s->dma_adc; - struct dmabuf *dac = &s->dma_dac; - unsigned adc_rate, dac_rate; - u16 ac97_extstat; - - if (s->no_vra) { - /* calc SRC factor - */ - adc->src_factor = ((96000 / rate) + 1) >> 1; - adc->sample_rate = 48000 / adc->src_factor; - return; - } - - adc->src_factor = 1; - - ac97_extstat = rdcodec(s->codec, AC97_EXTENDED_STATUS); - - rate = rate > 48000 ? 48000 : rate; - - /* enable VRA - */ - wrcodec(s->codec, AC97_EXTENDED_STATUS, - ac97_extstat | AC97_EXTSTAT_VRA); - - /* now write the sample rate - */ - wrcodec(s->codec, AC97_PCM_LR_ADC_RATE, (u16) rate); - - /* read it back for actual supported rate - */ - adc_rate = rdcodec(s->codec, AC97_PCM_LR_ADC_RATE); - - pr_debug("set_adc_rate: set to %d Hz\n", adc_rate); - - /* some codec's don't allow unequal DAC and ADC rates, in which case - * writing one rate reg actually changes both. - */ - dac_rate = rdcodec(s->codec, AC97_PCM_FRONT_DAC_RATE); - if (dac->num_channels > 2) - wrcodec(s->codec, AC97_PCM_SURR_DAC_RATE, dac_rate); - if (dac->num_channels > 4) - wrcodec(s->codec, AC97_PCM_LFE_DAC_RATE, dac_rate); - - adc->sample_rate = adc_rate; - dac->sample_rate = dac_rate; -} - -/* stop the DAC before calling */ -static void -set_dac_rate(struct au1550_state *s, unsigned rate) -{ - struct dmabuf *dac = &s->dma_dac; - struct dmabuf *adc = &s->dma_adc; - unsigned adc_rate, dac_rate; - u16 ac97_extstat; - - if (s->no_vra) { - /* calc SRC factor - */ - dac->src_factor = ((96000 / rate) + 1) >> 1; - dac->sample_rate = 48000 / dac->src_factor; - return; - } - - dac->src_factor = 1; - - ac97_extstat = rdcodec(s->codec, AC97_EXTENDED_STATUS); - - rate = rate > 48000 ? 48000 : rate; - - /* enable VRA - */ - wrcodec(s->codec, AC97_EXTENDED_STATUS, - ac97_extstat | AC97_EXTSTAT_VRA); - - /* now write the sample rate - */ - wrcodec(s->codec, AC97_PCM_FRONT_DAC_RATE, (u16) rate); - - /* I don't support different sample rates for multichannel, - * so make these channels the same. - */ - if (dac->num_channels > 2) - wrcodec(s->codec, AC97_PCM_SURR_DAC_RATE, (u16) rate); - if (dac->num_channels > 4) - wrcodec(s->codec, AC97_PCM_LFE_DAC_RATE, (u16) rate); - /* read it back for actual supported rate - */ - dac_rate = rdcodec(s->codec, AC97_PCM_FRONT_DAC_RATE); - - pr_debug("set_dac_rate: set to %d Hz\n", dac_rate); - - /* some codec's don't allow unequal DAC and ADC rates, in which case - * writing one rate reg actually changes both. - */ - adc_rate = rdcodec(s->codec, AC97_PCM_LR_ADC_RATE); - - dac->sample_rate = dac_rate; - adc->sample_rate = adc_rate; -} - -static void -stop_dac(struct au1550_state *s) -{ - struct dmabuf *db = &s->dma_dac; - u32 stat; - unsigned long flags; - - if (db->stopped) - return; - - spin_lock_irqsave(&s->lock, flags); - - au_writel(PSC_AC97PCR_TP, PSC_AC97PCR); - au_sync(); - - /* Wait for Transmit Busy to show disabled. - */ - do { - stat = au_readl(PSC_AC97STAT); - au_sync(); - } while ((stat & PSC_AC97STAT_TB) != 0); - - au1xxx_dbdma_reset(db->dmanr); - - db->stopped = 1; - - spin_unlock_irqrestore(&s->lock, flags); -} - -static void -stop_adc(struct au1550_state *s) -{ - struct dmabuf *db = &s->dma_adc; - unsigned long flags; - u32 stat; - - if (db->stopped) - return; - - spin_lock_irqsave(&s->lock, flags); - - au_writel(PSC_AC97PCR_RP, PSC_AC97PCR); - au_sync(); - - /* Wait for Receive Busy to show disabled. - */ - do { - stat = au_readl(PSC_AC97STAT); - au_sync(); - } while ((stat & PSC_AC97STAT_RB) != 0); - - au1xxx_dbdma_reset(db->dmanr); - - db->stopped = 1; - - spin_unlock_irqrestore(&s->lock, flags); -} - - -static void -set_xmit_slots(int num_channels) -{ - u32 ac97_config, stat; - - ac97_config = au_readl(PSC_AC97CFG); - au_sync(); - ac97_config &= ~(PSC_AC97CFG_TXSLOT_MASK | PSC_AC97CFG_DE_ENABLE); - au_writel(ac97_config, PSC_AC97CFG); - au_sync(); - - switch (num_channels) { - case 6: /* stereo with surround and center/LFE, - * slots 3,4,6,7,8,9 - */ - ac97_config |= PSC_AC97CFG_TXSLOT_ENA(6); - ac97_config |= PSC_AC97CFG_TXSLOT_ENA(9); - - case 4: /* stereo with surround, slots 3,4,7,8 */ - ac97_config |= PSC_AC97CFG_TXSLOT_ENA(7); - ac97_config |= PSC_AC97CFG_TXSLOT_ENA(8); - - case 2: /* stereo, slots 3,4 */ - case 1: /* mono */ - ac97_config |= PSC_AC97CFG_TXSLOT_ENA(3); - ac97_config |= PSC_AC97CFG_TXSLOT_ENA(4); - } - - au_writel(ac97_config, PSC_AC97CFG); - au_sync(); - - ac97_config |= PSC_AC97CFG_DE_ENABLE; - au_writel(ac97_config, PSC_AC97CFG); - au_sync(); - - /* Wait for Device ready. - */ - do { - stat = au_readl(PSC_AC97STAT); - au_sync(); - } while ((stat & PSC_AC97STAT_DR) == 0); -} - -static void -set_recv_slots(int num_channels) -{ - u32 ac97_config, stat; - - ac97_config = au_readl(PSC_AC97CFG); - au_sync(); - ac97_config &= ~(PSC_AC97CFG_RXSLOT_MASK | PSC_AC97CFG_DE_ENABLE); - au_writel(ac97_config, PSC_AC97CFG); - au_sync(); - - /* Always enable slots 3 and 4 (stereo). Slot 6 is - * optional Mic ADC, which we don't support yet. - */ - ac97_config |= PSC_AC97CFG_RXSLOT_ENA(3); - ac97_config |= PSC_AC97CFG_RXSLOT_ENA(4); - - au_writel(ac97_config, PSC_AC97CFG); - au_sync(); - - ac97_config |= PSC_AC97CFG_DE_ENABLE; - au_writel(ac97_config, PSC_AC97CFG); - au_sync(); - - /* Wait for Device ready. - */ - do { - stat = au_readl(PSC_AC97STAT); - au_sync(); - } while ((stat & PSC_AC97STAT_DR) == 0); -} - -/* Hold spinlock for both start_dac() and start_adc() calls */ -static void -start_dac(struct au1550_state *s) -{ - struct dmabuf *db = &s->dma_dac; - - if (!db->stopped) - return; - - set_xmit_slots(db->num_channels); - au_writel(PSC_AC97PCR_TC, PSC_AC97PCR); - au_sync(); - au_writel(PSC_AC97PCR_TS, PSC_AC97PCR); - au_sync(); - - au1xxx_dbdma_start(db->dmanr); - - db->stopped = 0; -} - -static void -start_adc(struct au1550_state *s) -{ - struct dmabuf *db = &s->dma_adc; - int i; - - if (!db->stopped) - return; - - /* Put two buffers on the ring to get things started. - */ - for (i=0; i<2; i++) { - au1xxx_dbdma_put_dest(db->dmanr, virt_to_phys(db->nextIn), - db->dma_fragsize, DDMA_FLAGS_IE); - - db->nextIn += db->dma_fragsize; - if (db->nextIn >= db->rawbuf + db->dmasize) - db->nextIn -= db->dmasize; - } - - set_recv_slots(db->num_channels); - au1xxx_dbdma_start(db->dmanr); - au_writel(PSC_AC97PCR_RC, PSC_AC97PCR); - au_sync(); - au_writel(PSC_AC97PCR_RS, PSC_AC97PCR); - au_sync(); - - db->stopped = 0; -} - -static int -prog_dmabuf(struct au1550_state *s, struct dmabuf *db) -{ - unsigned user_bytes_per_sec; - unsigned bufs; - unsigned rate = db->sample_rate; - - if (!db->rawbuf) { - db->ready = db->mapped = 0; - db->buforder = 5; /* 32 * PAGE_SIZE */ - db->rawbuf = kmalloc((PAGE_SIZE << db->buforder), GFP_KERNEL); - if (!db->rawbuf) - return -ENOMEM; - } - - db->cnt_factor = 1; - if (db->sample_size == 8) - db->cnt_factor *= 2; - if (db->num_channels == 1) - db->cnt_factor *= 2; - db->cnt_factor *= db->src_factor; - - db->count = 0; - db->dma_qcount = 0; - db->nextIn = db->nextOut = db->rawbuf; - - db->user_bytes_per_sample = (db->sample_size>>3) * db->num_channels; - db->dma_bytes_per_sample = 2 * ((db->num_channels == 1) ? - 2 : db->num_channels); - - user_bytes_per_sec = rate * db->user_bytes_per_sample; - bufs = PAGE_SIZE << db->buforder; - if (db->ossfragshift) { - if ((1000 << db->ossfragshift) < user_bytes_per_sec) - db->fragshift = ld2(user_bytes_per_sec/1000); - else - db->fragshift = db->ossfragshift; - } else { - db->fragshift = ld2(user_bytes_per_sec / 100 / - (db->subdivision ? db->subdivision : 1)); - if (db->fragshift < 3) - db->fragshift = 3; - } - - db->fragsize = 1 << db->fragshift; - db->dma_fragsize = db->fragsize * db->cnt_factor; - db->numfrag = bufs / db->dma_fragsize; - - while (db->numfrag < 4 && db->fragshift > 3) { - db->fragshift--; - db->fragsize = 1 << db->fragshift; - db->dma_fragsize = db->fragsize * db->cnt_factor; - db->numfrag = bufs / db->dma_fragsize; - } - - if (db->ossmaxfrags >= 4 && db->ossmaxfrags < db->numfrag) - db->numfrag = db->ossmaxfrags; - - db->dmasize = db->dma_fragsize * db->numfrag; - memset(db->rawbuf, 0, bufs); - - pr_debug("prog_dmabuf: rate=%d, samplesize=%d, channels=%d\n", - rate, db->sample_size, db->num_channels); - pr_debug("prog_dmabuf: fragsize=%d, cnt_factor=%d, dma_fragsize=%d\n", - db->fragsize, db->cnt_factor, db->dma_fragsize); - pr_debug("prog_dmabuf: numfrag=%d, dmasize=%d\n", db->numfrag, db->dmasize); - - db->ready = 1; - return 0; -} - -static int -prog_dmabuf_adc(struct au1550_state *s) -{ - stop_adc(s); - return prog_dmabuf(s, &s->dma_adc); - -} - -static int -prog_dmabuf_dac(struct au1550_state *s) -{ - stop_dac(s); - return prog_dmabuf(s, &s->dma_dac); -} - - -static void dac_dma_interrupt(int irq, void *dev_id) -{ - struct au1550_state *s = (struct au1550_state *) dev_id; - struct dmabuf *db = &s->dma_dac; - u32 ac97c_stat; - - spin_lock(&s->lock); - - ac97c_stat = au_readl(PSC_AC97STAT); - if (ac97c_stat & (AC97C_XU | AC97C_XO | AC97C_TE)) - pr_debug("AC97C status = 0x%08x\n", ac97c_stat); - db->dma_qcount--; - - if (db->count >= db->fragsize) { - if (au1xxx_dbdma_put_source(db->dmanr, - virt_to_phys(db->nextOut), db->fragsize, - DDMA_FLAGS_IE) == 0) { - err("qcount < 2 and no ring room!"); - } - db->nextOut += db->fragsize; - if (db->nextOut >= db->rawbuf + db->dmasize) - db->nextOut -= db->dmasize; - db->count -= db->fragsize; - db->total_bytes += db->dma_fragsize; - db->dma_qcount++; - } - - /* wake up anybody listening */ - if (waitqueue_active(&db->wait)) - wake_up(&db->wait); - - spin_unlock(&s->lock); -} - - -static void adc_dma_interrupt(int irq, void *dev_id) -{ - struct au1550_state *s = (struct au1550_state *)dev_id; - struct dmabuf *dp = &s->dma_adc; - u32 obytes; - char *obuf; - - spin_lock(&s->lock); - - /* Pull the buffer from the dma queue. - */ - au1xxx_dbdma_get_dest(dp->dmanr, (void *)(&obuf), &obytes); - - if ((dp->count + obytes) > dp->dmasize) { - /* Overrun. Stop ADC and log the error - */ - spin_unlock(&s->lock); - stop_adc(s); - dp->error++; - err("adc overrun"); - return; - } - - /* Put a new empty buffer on the destination DMA. - */ - au1xxx_dbdma_put_dest(dp->dmanr, virt_to_phys(dp->nextIn), - dp->dma_fragsize, DDMA_FLAGS_IE); - - dp->nextIn += dp->dma_fragsize; - if (dp->nextIn >= dp->rawbuf + dp->dmasize) - dp->nextIn -= dp->dmasize; - - dp->count += obytes; - dp->total_bytes += obytes; - - /* wake up anybody listening - */ - if (waitqueue_active(&dp->wait)) - wake_up(&dp->wait); - - spin_unlock(&s->lock); -} - -static loff_t -au1550_llseek(struct file *file, loff_t offset, int origin) -{ - return -ESPIPE; -} - - -static int -au1550_open_mixdev(struct inode *inode, struct file *file) -{ - mutex_lock(&au1550_ac97_mutex); - file->private_data = &au1550_state; - mutex_unlock(&au1550_ac97_mutex); - return 0; -} - -static int -au1550_release_mixdev(struct inode *inode, struct file *file) -{ - return 0; -} - -static int -mixdev_ioctl(struct ac97_codec *codec, unsigned int cmd, - unsigned long arg) -{ - return codec->mixer_ioctl(codec, cmd, arg); -} - -static long -au1550_ioctl_mixdev(struct file *file, unsigned int cmd, unsigned long arg) -{ - struct au1550_state *s = file->private_data; - struct ac97_codec *codec = s->codec; - int ret; - - mutex_lock(&au1550_ac97_mutex); - ret = mixdev_ioctl(codec, cmd, arg); - mutex_unlock(&au1550_ac97_mutex); - - return ret; -} - -static /*const */ struct file_operations au1550_mixer_fops = { - .owner = THIS_MODULE, - .llseek = au1550_llseek, - .unlocked_ioctl = au1550_ioctl_mixdev, - .open = au1550_open_mixdev, - .release = au1550_release_mixdev, -}; - -static int -drain_dac(struct au1550_state *s, int nonblock) -{ - unsigned long flags; - int count, tmo; - - if (s->dma_dac.mapped || !s->dma_dac.ready || s->dma_dac.stopped) - return 0; - - for (;;) { - spin_lock_irqsave(&s->lock, flags); - count = s->dma_dac.count; - spin_unlock_irqrestore(&s->lock, flags); - if (count <= s->dma_dac.fragsize) - break; - if (signal_pending(current)) - break; - if (nonblock) - return -EBUSY; - tmo = 1000 * count / (s->no_vra ? - 48000 : s->dma_dac.sample_rate); - tmo /= s->dma_dac.dma_bytes_per_sample; - au1550_delay(tmo); - } - if (signal_pending(current)) - return -ERESTARTSYS; - return 0; -} - -static inline u8 S16_TO_U8(s16 ch) -{ - return (u8) (ch >> 8) + 0x80; -} -static inline s16 U8_TO_S16(u8 ch) -{ - return (s16) (ch - 0x80) << 8; -} - -/* - * Translates user samples to dma buffer suitable for AC'97 DAC data: - * If mono, copy left channel to right channel in dma buffer. - * If 8 bit samples, cvt to 16-bit before writing to dma buffer. - * If interpolating (no VRA), duplicate every audio frame src_factor times. - */ -static int -translate_from_user(struct dmabuf *db, char* dmabuf, char* userbuf, - int dmacount) -{ - int sample, i; - int interp_bytes_per_sample; - int num_samples; - int mono = (db->num_channels == 1); - char usersample[12]; - s16 ch, dmasample[6]; - - if (db->sample_size == 16 && !mono && db->src_factor == 1) { - /* no translation necessary, just copy - */ - if (copy_from_user(dmabuf, userbuf, dmacount)) - return -EFAULT; - return dmacount; - } - - interp_bytes_per_sample = db->dma_bytes_per_sample * db->src_factor; - num_samples = dmacount / interp_bytes_per_sample; - - for (sample = 0; sample < num_samples; sample++) { - if (copy_from_user(usersample, userbuf, - db->user_bytes_per_sample)) { - return -EFAULT; - } - - for (i = 0; i < db->num_channels; i++) { - if (db->sample_size == 8) - ch = U8_TO_S16(usersample[i]); - else - ch = *((s16 *) (&usersample[i * 2])); - dmasample[i] = ch; - if (mono) - dmasample[i + 1] = ch; /* right channel */ - } - - /* duplicate every audio frame src_factor times - */ - for (i = 0; i < db->src_factor; i++) - memcpy(dmabuf, dmasample, db->dma_bytes_per_sample); - - userbuf += db->user_bytes_per_sample; - dmabuf += interp_bytes_per_sample; - } - - return num_samples * interp_bytes_per_sample; -} - -/* - * Translates AC'97 ADC samples to user buffer: - * If mono, send only left channel to user buffer. - * If 8 bit samples, cvt from 16 to 8 bit before writing to user buffer. - * If decimating (no VRA), skip over src_factor audio frames. - */ -static int -translate_to_user(struct dmabuf *db, char* userbuf, char* dmabuf, - int dmacount) -{ - int sample, i; - int interp_bytes_per_sample; - int num_samples; - int mono = (db->num_channels == 1); - char usersample[12]; - - if (db->sample_size == 16 && !mono && db->src_factor == 1) { - /* no translation necessary, just copy - */ - if (copy_to_user(userbuf, dmabuf, dmacount)) - return -EFAULT; - return dmacount; - } - - interp_bytes_per_sample = db->dma_bytes_per_sample * db->src_factor; - num_samples = dmacount / interp_bytes_per_sample; - - for (sample = 0; sample < num_samples; sample++) { - for (i = 0; i < db->num_channels; i++) { - if (db->sample_size == 8) - usersample[i] = - S16_TO_U8(*((s16 *) (&dmabuf[i * 2]))); - else - *((s16 *) (&usersample[i * 2])) = - *((s16 *) (&dmabuf[i * 2])); - } - - if (copy_to_user(userbuf, usersample, - db->user_bytes_per_sample)) { - return -EFAULT; - } - - userbuf += db->user_bytes_per_sample; - dmabuf += interp_bytes_per_sample; - } - - return num_samples * interp_bytes_per_sample; -} - -/* - * Copy audio data to/from user buffer from/to dma buffer, taking care - * that we wrap when reading/writing the dma buffer. Returns actual byte - * count written to or read from the dma buffer. - */ -static int -copy_dmabuf_user(struct dmabuf *db, char* userbuf, int count, int to_user) -{ - char *bufptr = to_user ? db->nextOut : db->nextIn; - char *bufend = db->rawbuf + db->dmasize; - int cnt, ret; - - if (bufptr + count > bufend) { - int partial = (int) (bufend - bufptr); - if (to_user) { - if ((cnt = translate_to_user(db, userbuf, - bufptr, partial)) < 0) - return cnt; - ret = cnt; - if ((cnt = translate_to_user(db, userbuf + partial, - db->rawbuf, - count - partial)) < 0) - return cnt; - ret += cnt; - } else { - if ((cnt = translate_from_user(db, bufptr, userbuf, - partial)) < 0) - return cnt; - ret = cnt; - if ((cnt = translate_from_user(db, db->rawbuf, - userbuf + partial, - count - partial)) < 0) - return cnt; - ret += cnt; - } - } else { - if (to_user) - ret = translate_to_user(db, userbuf, bufptr, count); - else - ret = translate_from_user(db, bufptr, userbuf, count); - } - - return ret; -} - - -static ssize_t -au1550_read(struct file *file, char *buffer, size_t count, loff_t *ppos) -{ - struct au1550_state *s = file->private_data; - struct dmabuf *db = &s->dma_adc; - DECLARE_WAITQUEUE(wait, current); - ssize_t ret; - unsigned long flags; - int cnt, usercnt, avail; - - if (db->mapped) - return -ENXIO; - if (!access_ok(VERIFY_WRITE, buffer, count)) - return -EFAULT; - ret = 0; - - count *= db->cnt_factor; - - mutex_lock(&s->sem); - add_wait_queue(&db->wait, &wait); - - while (count > 0) { - /* wait for samples in ADC dma buffer - */ - do { - spin_lock_irqsave(&s->lock, flags); - if (db->stopped) - start_adc(s); - avail = db->count; - if (avail <= 0) - __set_current_state(TASK_INTERRUPTIBLE); - spin_unlock_irqrestore(&s->lock, flags); - if (avail <= 0) { - if (file->f_flags & O_NONBLOCK) { - if (!ret) - ret = -EAGAIN; - goto out; - } - mutex_unlock(&s->sem); - schedule(); - if (signal_pending(current)) { - if (!ret) - ret = -ERESTARTSYS; - goto out2; - } - mutex_lock(&s->sem); - } - } while (avail <= 0); - - /* copy from nextOut to user - */ - if ((cnt = copy_dmabuf_user(db, buffer, - count > avail ? - avail : count, 1)) < 0) { - if (!ret) - ret = -EFAULT; - goto out; - } - - spin_lock_irqsave(&s->lock, flags); - db->count -= cnt; - db->nextOut += cnt; - if (db->nextOut >= db->rawbuf + db->dmasize) - db->nextOut -= db->dmasize; - spin_unlock_irqrestore(&s->lock, flags); - - count -= cnt; - usercnt = cnt / db->cnt_factor; - buffer += usercnt; - ret += usercnt; - } /* while (count > 0) */ - -out: - mutex_unlock(&s->sem); -out2: - remove_wait_queue(&db->wait, &wait); - set_current_state(TASK_RUNNING); - return ret; -} - -static ssize_t -au1550_write(struct file *file, const char *buffer, size_t count, loff_t * ppos) -{ - struct au1550_state *s = file->private_data; - struct dmabuf *db = &s->dma_dac; - DECLARE_WAITQUEUE(wait, current); - ssize_t ret = 0; - unsigned long flags; - int cnt, usercnt, avail; - - pr_debug("write: count=%d\n", count); - - if (db->mapped) - return -ENXIO; - if (!access_ok(VERIFY_READ, buffer, count)) - return -EFAULT; - - count *= db->cnt_factor; - - mutex_lock(&s->sem); - add_wait_queue(&db->wait, &wait); - - while (count > 0) { - /* wait for space in playback buffer - */ - do { - spin_lock_irqsave(&s->lock, flags); - avail = (int) db->dmasize - db->count; - if (avail <= 0) - __set_current_state(TASK_INTERRUPTIBLE); - spin_unlock_irqrestore(&s->lock, flags); - if (avail <= 0) { - if (file->f_flags & O_NONBLOCK) { - if (!ret) - ret = -EAGAIN; - goto out; - } - mutex_unlock(&s->sem); - schedule(); - if (signal_pending(current)) { - if (!ret) - ret = -ERESTARTSYS; - goto out2; - } - mutex_lock(&s->sem); - } - } while (avail <= 0); - - /* copy from user to nextIn - */ - if ((cnt = copy_dmabuf_user(db, (char *) buffer, - count > avail ? - avail : count, 0)) < 0) { - if (!ret) - ret = -EFAULT; - goto out; - } - - spin_lock_irqsave(&s->lock, flags); - db->count += cnt; - db->nextIn += cnt; - if (db->nextIn >= db->rawbuf + db->dmasize) - db->nextIn -= db->dmasize; - - /* If the data is available, we want to keep two buffers - * on the dma queue. If the queue count reaches zero, - * we know the dma has stopped. - */ - while ((db->dma_qcount < 2) && (db->count >= db->fragsize)) { - if (au1xxx_dbdma_put_source(db->dmanr, - virt_to_phys(db->nextOut), db->fragsize, - DDMA_FLAGS_IE) == 0) { - err("qcount < 2 and no ring room!"); - } - db->nextOut += db->fragsize; - if (db->nextOut >= db->rawbuf + db->dmasize) - db->nextOut -= db->dmasize; - db->total_bytes += db->dma_fragsize; - if (db->dma_qcount == 0) - start_dac(s); - db->dma_qcount++; - } - spin_unlock_irqrestore(&s->lock, flags); - - count -= cnt; - usercnt = cnt / db->cnt_factor; - buffer += usercnt; - ret += usercnt; - } /* while (count > 0) */ - -out: - mutex_unlock(&s->sem); -out2: - remove_wait_queue(&db->wait, &wait); - set_current_state(TASK_RUNNING); - return ret; -} - - -/* No kernel lock - we have our own spinlock */ -static unsigned int -au1550_poll(struct file *file, struct poll_table_struct *wait) -{ - struct au1550_state *s = file->private_data; - unsigned long flags; - unsigned int mask = 0; - - if (file->f_mode & FMODE_WRITE) { - if (!s->dma_dac.ready) - return 0; - poll_wait(file, &s->dma_dac.wait, wait); - } - if (file->f_mode & FMODE_READ) { - if (!s->dma_adc.ready) - return 0; - poll_wait(file, &s->dma_adc.wait, wait); - } - - spin_lock_irqsave(&s->lock, flags); - - if (file->f_mode & FMODE_READ) { - if (s->dma_adc.count >= (signed)s->dma_adc.dma_fragsize) - mask |= POLLIN | POLLRDNORM; - } - if (file->f_mode & FMODE_WRITE) { - if (s->dma_dac.mapped) { - if (s->dma_dac.count >= - (signed)s->dma_dac.dma_fragsize) - mask |= POLLOUT | POLLWRNORM; - } else { - if ((signed) s->dma_dac.dmasize >= - s->dma_dac.count + (signed)s->dma_dac.dma_fragsize) - mask |= POLLOUT | POLLWRNORM; - } - } - spin_unlock_irqrestore(&s->lock, flags); - return mask; -} - -static int -au1550_mmap(struct file *file, struct vm_area_struct *vma) -{ - struct au1550_state *s = file->private_data; - struct dmabuf *db; - unsigned long size; - int ret = 0; - - mutex_lock(&au1550_ac97_mutex); - mutex_lock(&s->sem); - if (vma->vm_flags & VM_WRITE) - db = &s->dma_dac; - else if (vma->vm_flags & VM_READ) - db = &s->dma_adc; - else { - ret = -EINVAL; - goto out; - } - if (vma->vm_pgoff != 0) { - ret = -EINVAL; - goto out; - } - size = vma->vm_end - vma->vm_start; - if (size > (PAGE_SIZE << db->buforder)) { - ret = -EINVAL; - goto out; - } - if (remap_pfn_range(vma, vma->vm_start, page_to_pfn(virt_to_page(db->rawbuf)), - size, vma->vm_page_prot)) { - ret = -EAGAIN; - goto out; - } - vma->vm_flags &= ~VM_IO; - db->mapped = 1; -out: - mutex_unlock(&s->sem); - mutex_unlock(&au1550_ac97_mutex); - return ret; -} - -#ifdef DEBUG -static struct ioctl_str_t { - unsigned int cmd; - const char *str; -} ioctl_str[] = { - {SNDCTL_DSP_RESET, "SNDCTL_DSP_RESET"}, - {SNDCTL_DSP_SYNC, "SNDCTL_DSP_SYNC"}, - {SNDCTL_DSP_SPEED, "SNDCTL_DSP_SPEED"}, - {SNDCTL_DSP_STEREO, "SNDCTL_DSP_STEREO"}, - {SNDCTL_DSP_GETBLKSIZE, "SNDCTL_DSP_GETBLKSIZE"}, - {SNDCTL_DSP_SAMPLESIZE, "SNDCTL_DSP_SAMPLESIZE"}, - {SNDCTL_DSP_CHANNELS, "SNDCTL_DSP_CHANNELS"}, - {SOUND_PCM_WRITE_CHANNELS, "SOUND_PCM_WRITE_CHANNELS"}, - {SOUND_PCM_WRITE_FILTER, "SOUND_PCM_WRITE_FILTER"}, - {SNDCTL_DSP_POST, "SNDCTL_DSP_POST"}, - {SNDCTL_DSP_SUBDIVIDE, "SNDCTL_DSP_SUBDIVIDE"}, - {SNDCTL_DSP_SETFRAGMENT, "SNDCTL_DSP_SETFRAGMENT"}, - {SNDCTL_DSP_GETFMTS, "SNDCTL_DSP_GETFMTS"}, - {SNDCTL_DSP_SETFMT, "SNDCTL_DSP_SETFMT"}, - {SNDCTL_DSP_GETOSPACE, "SNDCTL_DSP_GETOSPACE"}, - {SNDCTL_DSP_GETISPACE, "SNDCTL_DSP_GETISPACE"}, - {SNDCTL_DSP_NONBLOCK, "SNDCTL_DSP_NONBLOCK"}, - {SNDCTL_DSP_GETCAPS, "SNDCTL_DSP_GETCAPS"}, - {SNDCTL_DSP_GETTRIGGER, "SNDCTL_DSP_GETTRIGGER"}, - {SNDCTL_DSP_SETTRIGGER, "SNDCTL_DSP_SETTRIGGER"}, - {SNDCTL_DSP_GETIPTR, "SNDCTL_DSP_GETIPTR"}, - {SNDCTL_DSP_GETOPTR, "SNDCTL_DSP_GETOPTR"}, - {SNDCTL_DSP_MAPINBUF, "SNDCTL_DSP_MAPINBUF"}, - {SNDCTL_DSP_MAPOUTBUF, "SNDCTL_DSP_MAPOUTBUF"}, - {SNDCTL_DSP_SETSYNCRO, "SNDCTL_DSP_SETSYNCRO"}, - {SNDCTL_DSP_SETDUPLEX, "SNDCTL_DSP_SETDUPLEX"}, - {SNDCTL_DSP_GETODELAY, "SNDCTL_DSP_GETODELAY"}, - {SNDCTL_DSP_GETCHANNELMASK, "SNDCTL_DSP_GETCHANNELMASK"}, - {SNDCTL_DSP_BIND_CHANNEL, "SNDCTL_DSP_BIND_CHANNEL"}, - {OSS_GETVERSION, "OSS_GETVERSION"}, - {SOUND_PCM_READ_RATE, "SOUND_PCM_READ_RATE"}, - {SOUND_PCM_READ_CHANNELS, "SOUND_PCM_READ_CHANNELS"}, - {SOUND_PCM_READ_BITS, "SOUND_PCM_READ_BITS"}, - {SOUND_PCM_READ_FILTER, "SOUND_PCM_READ_FILTER"} -}; -#endif - -static int -dma_count_done(struct dmabuf *db) -{ - if (db->stopped) - return 0; - - return db->dma_fragsize - au1xxx_get_dma_residue(db->dmanr); -} - - -static int -au1550_ioctl(struct file *file, unsigned int cmd, unsigned long arg) -{ - struct au1550_state *s = file->private_data; - unsigned long flags; - audio_buf_info abinfo; - count_info cinfo; - int count; - int val, mapped, ret, diff; - - mapped = ((file->f_mode & FMODE_WRITE) && s->dma_dac.mapped) || - ((file->f_mode & FMODE_READ) && s->dma_adc.mapped); - -#ifdef DEBUG - for (count = 0; count < ARRAY_SIZE(ioctl_str); count++) { - if (ioctl_str[count].cmd == cmd) - break; - } - if (count < ARRAY_SIZE(ioctl_str)) - pr_debug("ioctl %s, arg=0x%lxn", ioctl_str[count].str, arg); - else - pr_debug("ioctl 0x%x unknown, arg=0x%lx\n", cmd, arg); -#endif - - switch (cmd) { - case OSS_GETVERSION: - return put_user(SOUND_VERSION, (int *) arg); - - case SNDCTL_DSP_SYNC: - if (file->f_mode & FMODE_WRITE) - return drain_dac(s, file->f_flags & O_NONBLOCK); - return 0; - - case SNDCTL_DSP_SETDUPLEX: - return 0; - - case SNDCTL_DSP_GETCAPS: - return put_user(DSP_CAP_DUPLEX | DSP_CAP_REALTIME | - DSP_CAP_TRIGGER | DSP_CAP_MMAP, (int *)arg); - - case SNDCTL_DSP_RESET: - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - synchronize_irq(); - s->dma_dac.count = s->dma_dac.total_bytes = 0; - s->dma_dac.nextIn = s->dma_dac.nextOut = - s->dma_dac.rawbuf; - } - if (file->f_mode & FMODE_READ) { - stop_adc(s); - synchronize_irq(); - s->dma_adc.count = s->dma_adc.total_bytes = 0; - s->dma_adc.nextIn = s->dma_adc.nextOut = - s->dma_adc.rawbuf; - } - return 0; - - case SNDCTL_DSP_SPEED: - if (get_user(val, (int *) arg)) - return -EFAULT; - if (val >= 0) { - if (file->f_mode & FMODE_READ) { - stop_adc(s); - set_adc_rate(s, val); - } - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - set_dac_rate(s, val); - } - if (s->open_mode & FMODE_READ) - if ((ret = prog_dmabuf_adc(s))) - return ret; - if (s->open_mode & FMODE_WRITE) - if ((ret = prog_dmabuf_dac(s))) - return ret; - } - return put_user((file->f_mode & FMODE_READ) ? - s->dma_adc.sample_rate : - s->dma_dac.sample_rate, - (int *)arg); - - case SNDCTL_DSP_STEREO: - if (get_user(val, (int *) arg)) - return -EFAULT; - if (file->f_mode & FMODE_READ) { - stop_adc(s); - s->dma_adc.num_channels = val ? 2 : 1; - if ((ret = prog_dmabuf_adc(s))) - return ret; - } - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - s->dma_dac.num_channels = val ? 2 : 1; - if (s->codec_ext_caps & AC97_EXT_DACS) { - /* disable surround and center/lfe in AC'97 - */ - u16 ext_stat = rdcodec(s->codec, - AC97_EXTENDED_STATUS); - wrcodec(s->codec, AC97_EXTENDED_STATUS, - ext_stat | (AC97_EXTSTAT_PRI | - AC97_EXTSTAT_PRJ | - AC97_EXTSTAT_PRK)); - } - if ((ret = prog_dmabuf_dac(s))) - return ret; - } - return 0; - - case SNDCTL_DSP_CHANNELS: - if (get_user(val, (int *) arg)) - return -EFAULT; - if (val != 0) { - if (file->f_mode & FMODE_READ) { - if (val < 0 || val > 2) - return -EINVAL; - stop_adc(s); - s->dma_adc.num_channels = val; - if ((ret = prog_dmabuf_adc(s))) - return ret; - } - if (file->f_mode & FMODE_WRITE) { - switch (val) { - case 1: - case 2: - break; - case 3: - case 5: - return -EINVAL; - case 4: - if (!(s->codec_ext_caps & - AC97_EXTID_SDAC)) - return -EINVAL; - break; - case 6: - if ((s->codec_ext_caps & - AC97_EXT_DACS) != AC97_EXT_DACS) - return -EINVAL; - break; - default: - return -EINVAL; - } - - stop_dac(s); - if (val <= 2 && - (s->codec_ext_caps & AC97_EXT_DACS)) { - /* disable surround and center/lfe - * channels in AC'97 - */ - u16 ext_stat = - rdcodec(s->codec, - AC97_EXTENDED_STATUS); - wrcodec(s->codec, - AC97_EXTENDED_STATUS, - ext_stat | (AC97_EXTSTAT_PRI | - AC97_EXTSTAT_PRJ | - AC97_EXTSTAT_PRK)); - } else if (val >= 4) { - /* enable surround, center/lfe - * channels in AC'97 - */ - u16 ext_stat = - rdcodec(s->codec, - AC97_EXTENDED_STATUS); - ext_stat &= ~AC97_EXTSTAT_PRJ; - if (val == 6) - ext_stat &= - ~(AC97_EXTSTAT_PRI | - AC97_EXTSTAT_PRK); - wrcodec(s->codec, - AC97_EXTENDED_STATUS, - ext_stat); - } - - s->dma_dac.num_channels = val; - if ((ret = prog_dmabuf_dac(s))) - return ret; - } - } - return put_user(val, (int *) arg); - - case SNDCTL_DSP_GETFMTS: /* Returns a mask */ - return put_user(AFMT_S16_LE | AFMT_U8, (int *) arg); - - case SNDCTL_DSP_SETFMT: /* Selects ONE fmt */ - if (get_user(val, (int *) arg)) - return -EFAULT; - if (val != AFMT_QUERY) { - if (file->f_mode & FMODE_READ) { - stop_adc(s); - if (val == AFMT_S16_LE) - s->dma_adc.sample_size = 16; - else { - val = AFMT_U8; - s->dma_adc.sample_size = 8; - } - if ((ret = prog_dmabuf_adc(s))) - return ret; - } - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - if (val == AFMT_S16_LE) - s->dma_dac.sample_size = 16; - else { - val = AFMT_U8; - s->dma_dac.sample_size = 8; - } - if ((ret = prog_dmabuf_dac(s))) - return ret; - } - } else { - if (file->f_mode & FMODE_READ) - val = (s->dma_adc.sample_size == 16) ? - AFMT_S16_LE : AFMT_U8; - else - val = (s->dma_dac.sample_size == 16) ? - AFMT_S16_LE : AFMT_U8; - } - return put_user(val, (int *) arg); - - case SNDCTL_DSP_POST: - return 0; - - case SNDCTL_DSP_GETTRIGGER: - val = 0; - spin_lock_irqsave(&s->lock, flags); - if (file->f_mode & FMODE_READ && !s->dma_adc.stopped) - val |= PCM_ENABLE_INPUT; - if (file->f_mode & FMODE_WRITE && !s->dma_dac.stopped) - val |= PCM_ENABLE_OUTPUT; - spin_unlock_irqrestore(&s->lock, flags); - return put_user(val, (int *) arg); - - case SNDCTL_DSP_SETTRIGGER: - if (get_user(val, (int *) arg)) - return -EFAULT; - if (file->f_mode & FMODE_READ) { - if (val & PCM_ENABLE_INPUT) { - spin_lock_irqsave(&s->lock, flags); - start_adc(s); - spin_unlock_irqrestore(&s->lock, flags); - } else - stop_adc(s); - } - if (file->f_mode & FMODE_WRITE) { - if (val & PCM_ENABLE_OUTPUT) { - spin_lock_irqsave(&s->lock, flags); - start_dac(s); - spin_unlock_irqrestore(&s->lock, flags); - } else - stop_dac(s); - } - return 0; - - case SNDCTL_DSP_GETOSPACE: - if (!(file->f_mode & FMODE_WRITE)) - return -EINVAL; - abinfo.fragsize = s->dma_dac.fragsize; - spin_lock_irqsave(&s->lock, flags); - count = s->dma_dac.count; - count -= dma_count_done(&s->dma_dac); - spin_unlock_irqrestore(&s->lock, flags); - if (count < 0) - count = 0; - abinfo.bytes = (s->dma_dac.dmasize - count) / - s->dma_dac.cnt_factor; - abinfo.fragstotal = s->dma_dac.numfrag; - abinfo.fragments = abinfo.bytes >> s->dma_dac.fragshift; - pr_debug("ioctl SNDCTL_DSP_GETOSPACE: bytes=%d, fragments=%d\n", abinfo.bytes, abinfo.fragments); - return copy_to_user((void *) arg, &abinfo, - sizeof(abinfo)) ? -EFAULT : 0; - - case SNDCTL_DSP_GETISPACE: - if (!(file->f_mode & FMODE_READ)) - return -EINVAL; - abinfo.fragsize = s->dma_adc.fragsize; - spin_lock_irqsave(&s->lock, flags); - count = s->dma_adc.count; - count += dma_count_done(&s->dma_adc); - spin_unlock_irqrestore(&s->lock, flags); - if (count < 0) - count = 0; - abinfo.bytes = count / s->dma_adc.cnt_factor; - abinfo.fragstotal = s->dma_adc.numfrag; - abinfo.fragments = abinfo.bytes >> s->dma_adc.fragshift; - return copy_to_user((void *) arg, &abinfo, - sizeof(abinfo)) ? -EFAULT : 0; - - case SNDCTL_DSP_NONBLOCK: - spin_lock(&file->f_lock); - file->f_flags |= O_NONBLOCK; - spin_unlock(&file->f_lock); - return 0; - - case SNDCTL_DSP_GETODELAY: - if (!(file->f_mode & FMODE_WRITE)) - return -EINVAL; - spin_lock_irqsave(&s->lock, flags); - count = s->dma_dac.count; - count -= dma_count_done(&s->dma_dac); - spin_unlock_irqrestore(&s->lock, flags); - if (count < 0) - count = 0; - count /= s->dma_dac.cnt_factor; - return put_user(count, (int *) arg); - - case SNDCTL_DSP_GETIPTR: - if (!(file->f_mode & FMODE_READ)) - return -EINVAL; - spin_lock_irqsave(&s->lock, flags); - cinfo.bytes = s->dma_adc.total_bytes; - count = s->dma_adc.count; - if (!s->dma_adc.stopped) { - diff = dma_count_done(&s->dma_adc); - count += diff; - cinfo.bytes += diff; - cinfo.ptr = virt_to_phys(s->dma_adc.nextIn) + diff - - virt_to_phys(s->dma_adc.rawbuf); - } else - cinfo.ptr = virt_to_phys(s->dma_adc.nextIn) - - virt_to_phys(s->dma_adc.rawbuf); - if (s->dma_adc.mapped) - s->dma_adc.count &= (s->dma_adc.dma_fragsize-1); - spin_unlock_irqrestore(&s->lock, flags); - if (count < 0) - count = 0; - cinfo.blocks = count >> s->dma_adc.fragshift; - return copy_to_user((void *) arg, &cinfo, sizeof(cinfo)); - - case SNDCTL_DSP_GETOPTR: - if (!(file->f_mode & FMODE_READ)) - return -EINVAL; - spin_lock_irqsave(&s->lock, flags); - cinfo.bytes = s->dma_dac.total_bytes; - count = s->dma_dac.count; - if (!s->dma_dac.stopped) { - diff = dma_count_done(&s->dma_dac); - count -= diff; - cinfo.bytes += diff; - cinfo.ptr = virt_to_phys(s->dma_dac.nextOut) + diff - - virt_to_phys(s->dma_dac.rawbuf); - } else - cinfo.ptr = virt_to_phys(s->dma_dac.nextOut) - - virt_to_phys(s->dma_dac.rawbuf); - if (s->dma_dac.mapped) - s->dma_dac.count &= (s->dma_dac.dma_fragsize-1); - spin_unlock_irqrestore(&s->lock, flags); - if (count < 0) - count = 0; - cinfo.blocks = count >> s->dma_dac.fragshift; - return copy_to_user((void *) arg, &cinfo, sizeof(cinfo)); - - case SNDCTL_DSP_GETBLKSIZE: - if (file->f_mode & FMODE_WRITE) - return put_user(s->dma_dac.fragsize, (int *) arg); - else - return put_user(s->dma_adc.fragsize, (int *) arg); - - case SNDCTL_DSP_SETFRAGMENT: - if (get_user(val, (int *) arg)) - return -EFAULT; - if (file->f_mode & FMODE_READ) { - stop_adc(s); - s->dma_adc.ossfragshift = val & 0xffff; - s->dma_adc.ossmaxfrags = (val >> 16) & 0xffff; - if (s->dma_adc.ossfragshift < 4) - s->dma_adc.ossfragshift = 4; - if (s->dma_adc.ossfragshift > 15) - s->dma_adc.ossfragshift = 15; - if (s->dma_adc.ossmaxfrags < 4) - s->dma_adc.ossmaxfrags = 4; - if ((ret = prog_dmabuf_adc(s))) - return ret; - } - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - s->dma_dac.ossfragshift = val & 0xffff; - s->dma_dac.ossmaxfrags = (val >> 16) & 0xffff; - if (s->dma_dac.ossfragshift < 4) - s->dma_dac.ossfragshift = 4; - if (s->dma_dac.ossfragshift > 15) - s->dma_dac.ossfragshift = 15; - if (s->dma_dac.ossmaxfrags < 4) - s->dma_dac.ossmaxfrags = 4; - if ((ret = prog_dmabuf_dac(s))) - return ret; - } - return 0; - - case SNDCTL_DSP_SUBDIVIDE: - if ((file->f_mode & FMODE_READ && s->dma_adc.subdivision) || - (file->f_mode & FMODE_WRITE && s->dma_dac.subdivision)) - return -EINVAL; - if (get_user(val, (int *) arg)) - return -EFAULT; - if (val != 1 && val != 2 && val != 4) - return -EINVAL; - if (file->f_mode & FMODE_READ) { - stop_adc(s); - s->dma_adc.subdivision = val; - if ((ret = prog_dmabuf_adc(s))) - return ret; - } - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - s->dma_dac.subdivision = val; - if ((ret = prog_dmabuf_dac(s))) - return ret; - } - return 0; - - case SOUND_PCM_READ_RATE: - return put_user((file->f_mode & FMODE_READ) ? - s->dma_adc.sample_rate : - s->dma_dac.sample_rate, - (int *)arg); - - case SOUND_PCM_READ_CHANNELS: - if (file->f_mode & FMODE_READ) - return put_user(s->dma_adc.num_channels, (int *)arg); - else - return put_user(s->dma_dac.num_channels, (int *)arg); - - case SOUND_PCM_READ_BITS: - if (file->f_mode & FMODE_READ) - return put_user(s->dma_adc.sample_size, (int *)arg); - else - return put_user(s->dma_dac.sample_size, (int *)arg); - - case SOUND_PCM_WRITE_FILTER: - case SNDCTL_DSP_SETSYNCRO: - case SOUND_PCM_READ_FILTER: - return -EINVAL; - } - - return mixdev_ioctl(s->codec, cmd, arg); -} - -static long -au1550_unlocked_ioctl(struct file *file, unsigned int cmd, unsigned long arg) -{ - int ret; - - mutex_lock(&au1550_ac97_mutex); - ret = au1550_ioctl(file, cmd, arg); - mutex_unlock(&au1550_ac97_mutex); - - return ret; -} - -static int -au1550_open(struct inode *inode, struct file *file) -{ - int minor = MINOR(inode->i_rdev); - DECLARE_WAITQUEUE(wait, current); - struct au1550_state *s = &au1550_state; - int ret; - -#ifdef DEBUG - if (file->f_flags & O_NONBLOCK) - pr_debug("open: non-blocking\n"); - else - pr_debug("open: blocking\n"); -#endif - - file->private_data = s; - mutex_lock(&au1550_ac97_mutex); - /* wait for device to become free */ - mutex_lock(&s->open_mutex); - while (s->open_mode & file->f_mode) { - ret = -EBUSY; - if (file->f_flags & O_NONBLOCK) - goto out; - add_wait_queue(&s->open_wait, &wait); - __set_current_state(TASK_INTERRUPTIBLE); - mutex_unlock(&s->open_mutex); - schedule(); - remove_wait_queue(&s->open_wait, &wait); - set_current_state(TASK_RUNNING); - ret = -ERESTARTSYS; - if (signal_pending(current)) - goto out2; - mutex_lock(&s->open_mutex); - } - - stop_dac(s); - stop_adc(s); - - if (file->f_mode & FMODE_READ) { - s->dma_adc.ossfragshift = s->dma_adc.ossmaxfrags = - s->dma_adc.subdivision = s->dma_adc.total_bytes = 0; - s->dma_adc.num_channels = 1; - s->dma_adc.sample_size = 8; - set_adc_rate(s, 8000); - if ((minor & 0xf) == SND_DEV_DSP16) - s->dma_adc.sample_size = 16; - } - - if (file->f_mode & FMODE_WRITE) { - s->dma_dac.ossfragshift = s->dma_dac.ossmaxfrags = - s->dma_dac.subdivision = s->dma_dac.total_bytes = 0; - s->dma_dac.num_channels = 1; - s->dma_dac.sample_size = 8; - set_dac_rate(s, 8000); - if ((minor & 0xf) == SND_DEV_DSP16) - s->dma_dac.sample_size = 16; - } - - if (file->f_mode & FMODE_READ) { - if ((ret = prog_dmabuf_adc(s))) - goto out; - } - if (file->f_mode & FMODE_WRITE) { - if ((ret = prog_dmabuf_dac(s))) - goto out; - } - - s->open_mode |= file->f_mode & (FMODE_READ | FMODE_WRITE); - mutex_init(&s->sem); - ret = 0; -out: - mutex_unlock(&s->open_mutex); -out2: - mutex_unlock(&au1550_ac97_mutex); - return ret; -} - -static int -au1550_release(struct inode *inode, struct file *file) -{ - struct au1550_state *s = file->private_data; - - mutex_lock(&au1550_ac97_mutex); - - if (file->f_mode & FMODE_WRITE) { - mutex_unlock(&au1550_ac97_mutex); - drain_dac(s, file->f_flags & O_NONBLOCK); - mutex_lock(&au1550_ac97_mutex); - } - - mutex_lock(&s->open_mutex); - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - kfree(s->dma_dac.rawbuf); - s->dma_dac.rawbuf = NULL; - } - if (file->f_mode & FMODE_READ) { - stop_adc(s); - kfree(s->dma_adc.rawbuf); - s->dma_adc.rawbuf = NULL; - } - s->open_mode &= ((~file->f_mode) & (FMODE_READ|FMODE_WRITE)); - mutex_unlock(&s->open_mutex); - wake_up(&s->open_wait); - mutex_unlock(&au1550_ac97_mutex); - return 0; -} - -static /*const */ struct file_operations au1550_audio_fops = { - .owner = THIS_MODULE, - .llseek = au1550_llseek, - .read = au1550_read, - .write = au1550_write, - .poll = au1550_poll, - .unlocked_ioctl = au1550_unlocked_ioctl, - .mmap = au1550_mmap, - .open = au1550_open, - .release = au1550_release, -}; - -MODULE_AUTHOR("Advanced Micro Devices (AMD), dan@embeddededge.com"); -MODULE_DESCRIPTION("Au1550 AC97 Audio Driver"); -MODULE_LICENSE("GPL"); - - -static int __devinit -au1550_probe(void) -{ - struct au1550_state *s = &au1550_state; - int val; - - memset(s, 0, sizeof(struct au1550_state)); - - init_waitqueue_head(&s->dma_adc.wait); - init_waitqueue_head(&s->dma_dac.wait); - init_waitqueue_head(&s->open_wait); - mutex_init(&s->open_mutex); - spin_lock_init(&s->lock); - - s->codec = ac97_alloc_codec(); - if(s->codec == NULL) { - err("Out of memory"); - return -1; - } - s->codec->private_data = s; - s->codec->id = 0; - s->codec->codec_read = rdcodec; - s->codec->codec_write = wrcodec; - s->codec->codec_wait = waitcodec; - - if (!request_mem_region(CPHYSADDR(AC97_PSC_SEL), - 0x30, "Au1550 AC97")) { - err("AC'97 ports in use"); - } - - /* Allocate the DMA Channels - */ - if ((s->dma_dac.dmanr = au1xxx_dbdma_chan_alloc(DBDMA_MEM_CHAN, - DBDMA_AC97_TX_CHAN, dac_dma_interrupt, (void *)s)) == 0) { - err("Can't get DAC DMA"); - goto err_dma1; - } - au1xxx_dbdma_set_devwidth(s->dma_dac.dmanr, 16); - if (au1xxx_dbdma_ring_alloc(s->dma_dac.dmanr, - NUM_DBDMA_DESCRIPTORS) == 0) { - err("Can't get DAC DMA descriptors"); - goto err_dma1; - } - - if ((s->dma_adc.dmanr = au1xxx_dbdma_chan_alloc(DBDMA_AC97_RX_CHAN, - DBDMA_MEM_CHAN, adc_dma_interrupt, (void *)s)) == 0) { - err("Can't get ADC DMA"); - goto err_dma2; - } - au1xxx_dbdma_set_devwidth(s->dma_adc.dmanr, 16); - if (au1xxx_dbdma_ring_alloc(s->dma_adc.dmanr, - NUM_DBDMA_DESCRIPTORS) == 0) { - err("Can't get ADC DMA descriptors"); - goto err_dma2; - } - - pr_info("DAC: DMA%d, ADC: DMA%d", DBDMA_AC97_TX_CHAN, DBDMA_AC97_RX_CHAN); - - /* register devices */ - - if ((s->dev_audio = register_sound_dsp(&au1550_audio_fops, -1)) < 0) - goto err_dev1; - if ((s->codec->dev_mixer = - register_sound_mixer(&au1550_mixer_fops, -1)) < 0) - goto err_dev2; - - /* The GPIO for the appropriate PSC was configured by the - * board specific start up. - * - * configure PSC for AC'97 - */ - au_writel(0, AC97_PSC_CTRL); /* Disable PSC */ - au_sync(); - au_writel((PSC_SEL_CLK_SERCLK | PSC_SEL_PS_AC97MODE), AC97_PSC_SEL); - au_sync(); - - /* cold reset the AC'97 - */ - au_writel(PSC_AC97RST_RST, PSC_AC97RST); - au_sync(); - au1550_delay(10); - au_writel(0, PSC_AC97RST); - au_sync(); - - /* need to delay around 500msec(bleech) to give - some CODECs enough time to wakeup */ - au1550_delay(500); - - /* warm reset the AC'97 to start the bitclk - */ - au_writel(PSC_AC97RST_SNC, PSC_AC97RST); - au_sync(); - udelay(100); - au_writel(0, PSC_AC97RST); - au_sync(); - - /* Enable PSC - */ - au_writel(PSC_CTRL_ENABLE, AC97_PSC_CTRL); - au_sync(); - - /* Wait for PSC ready. - */ - do { - val = au_readl(PSC_AC97STAT); - au_sync(); - } while ((val & PSC_AC97STAT_SR) == 0); - - /* Configure AC97 controller. - * Deep FIFO, 16-bit sample, DMA, make sure DMA matches fifo size. - */ - val = PSC_AC97CFG_SET_LEN(16); - val |= PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8; - - /* Enable device so we can at least - * talk over the AC-link. - */ - au_writel(val, PSC_AC97CFG); - au_writel(PSC_AC97MSK_ALLMASK, PSC_AC97MSK); - au_sync(); - val |= PSC_AC97CFG_DE_ENABLE; - au_writel(val, PSC_AC97CFG); - au_sync(); - - /* Wait for Device ready. - */ - do { - val = au_readl(PSC_AC97STAT); - au_sync(); - } while ((val & PSC_AC97STAT_DR) == 0); - - /* codec init */ - if (!ac97_probe_codec(s->codec)) - goto err_dev3; - - s->codec_base_caps = rdcodec(s->codec, AC97_RESET); - s->codec_ext_caps = rdcodec(s->codec, AC97_EXTENDED_ID); - pr_info("AC'97 Base/Extended ID = %04x/%04x", - s->codec_base_caps, s->codec_ext_caps); - - if (!(s->codec_ext_caps & AC97_EXTID_VRA)) { - /* codec does not support VRA - */ - s->no_vra = 1; - } else if (!vra) { - /* Boot option says disable VRA - */ - u16 ac97_extstat = rdcodec(s->codec, AC97_EXTENDED_STATUS); - wrcodec(s->codec, AC97_EXTENDED_STATUS, - ac97_extstat & ~AC97_EXTSTAT_VRA); - s->no_vra = 1; - } - if (s->no_vra) - pr_info("no VRA, interpolating and decimating"); - - /* set mic to be the recording source */ - val = SOUND_MASK_MIC; - mixdev_ioctl(s->codec, SOUND_MIXER_WRITE_RECSRC, - (unsigned long) &val); - - return 0; - - err_dev3: - unregister_sound_mixer(s->codec->dev_mixer); - err_dev2: - unregister_sound_dsp(s->dev_audio); - err_dev1: - au1xxx_dbdma_chan_free(s->dma_adc.dmanr); - err_dma2: - au1xxx_dbdma_chan_free(s->dma_dac.dmanr); - err_dma1: - release_mem_region(CPHYSADDR(AC97_PSC_SEL), 0x30); - - ac97_release_codec(s->codec); - return -1; -} - -static void __devinit -au1550_remove(void) -{ - struct au1550_state *s = &au1550_state; - - if (!s) - return; - synchronize_irq(); - au1xxx_dbdma_chan_free(s->dma_adc.dmanr); - au1xxx_dbdma_chan_free(s->dma_dac.dmanr); - release_mem_region(CPHYSADDR(AC97_PSC_SEL), 0x30); - unregister_sound_dsp(s->dev_audio); - unregister_sound_mixer(s->codec->dev_mixer); - ac97_release_codec(s->codec); -} - -static int __init -init_au1550(void) -{ - return au1550_probe(); -} - -static void __exit -cleanup_au1550(void) -{ - au1550_remove(); -} - -module_init(init_au1550); -module_exit(cleanup_au1550); - -#ifndef MODULE - -static int __init -au1550_setup(char *options) -{ - char *this_opt; - - if (!options || !*options) - return 0; - - while ((this_opt = strsep(&options, ","))) { - if (!*this_opt) - continue; - if (!strncmp(this_opt, "vra", 3)) { - vra = 1; - } - } - - return 1; -} - -__setup("au1550_audio=", au1550_setup); - -#endif /* MODULE */ diff --git a/sound/oss/audio.c b/sound/oss/audio.c index 7df48a25c4ee..4b958b1c497c 100644 --- a/sound/oss/audio.c +++ b/sound/oss/audio.c @@ -514,7 +514,7 @@ int audio_ioctl(int dev, struct file *file, unsigned int cmd, void __user *arg) count += dmap->bytes_in_use; /* Pointer wrap not handled yet */ count += dmap->byte_counter; - /* Substract current count from the number of bytes written by app */ + /* Subtract current count from the number of bytes written by app */ count = dmap->user_counter - count; if (count < 0) count = 0; @@ -931,7 +931,7 @@ static int dma_ioctl(int dev, unsigned int cmd, void __user *arg) if (count < dmap_out->fragment_size && dmap_out->qhead != 0) count += dmap_out->bytes_in_use; /* Pointer wrap not handled yet */ count += dmap_out->byte_counter; - /* Substract current count from the number of bytes written by app */ + /* Subtract current count from the number of bytes written by app */ count = dmap_out->user_counter - count; if (count < 0) count = 0; diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index 87e2c72651f5..c918313c2206 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -1021,7 +1021,7 @@ static int sq_ioctl(struct file *file, u_int cmd, u_long arg) case SNDCTL_DSP_SYNC: /* This call, effectively, has the same behaviour as SNDCTL_DSP_RESET except that it waits for output to finish before resetting - everything - read, however, is killed imediately. + everything - read, however, is killed immediately. */ result = 0 ; if (file->f_mode & FMODE_WRITE) { diff --git a/sound/oss/midibuf.c b/sound/oss/midibuf.c index ceedb1eff203..8cdb2cfe65c8 100644 --- a/sound/oss/midibuf.c +++ b/sound/oss/midibuf.c @@ -295,7 +295,7 @@ int MIDIbuf_write(int dev, struct file *file, const char __user *buf, int count) for (i = 0; i < n; i++) { - /* BROKE BROKE BROKE - CANT DO THIS WITH CLI !! */ + /* BROKE BROKE BROKE - CAN'T DO THIS WITH CLI !! */ /* yes, think the same, so I removed the cli() brackets QUEUE_BYTE is protected against interrupts */ if (copy_from_user((char *) &tmp_data, &(buf)[c], 1)) { diff --git a/sound/oss/sb_card.c b/sound/oss/sb_card.c index 84ef4d06c1c2..fb5d7250de38 100644 --- a/sound/oss/sb_card.c +++ b/sound/oss/sb_card.c @@ -1,7 +1,7 @@ /* * sound/oss/sb_card.c * - * Detection routine for the ISA Sound Blaster and compatable sound + * Detection routine for the ISA Sound Blaster and compatible sound * cards. * * This file is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index 9890cf2066ff..5c773dff5ac5 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -168,7 +168,7 @@ * corresponding playback levels, unless recmask says they aren't recorded. In * the latter case the recording volumes are 0. * Now recording levels of inputs can be controlled, by changing the playback - * levels. Futhermore several devices can be recorded together (which is not + * levels. Furthermore several devices can be recorded together (which is not * possible with the ES1688). * Besides the separate recording level control for each input, the common * recording level can also be controlled by RECLEV as described above. diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c index 44357d877a27..09d46484bc1a 100644 --- a/sound/oss/swarm_cs4297a.c +++ b/sound/oss/swarm_cs4297a.c @@ -875,7 +875,7 @@ static void start_adc(struct cs4297a_state *s) if (s->prop_adc.fmt & AFMT_S8 || s->prop_adc.fmt & AFMT_U8) { // // now only use 16 bit capture, due to truncation issue - // in the chip, noticable distortion occurs. + // in the chip, noticeable distortion occurs. // allocate buffer and then convert from 16 bit to // 8 bit for the user buffer. // diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c index f0e0caa53200..12ba28e7b933 100644 --- a/sound/oss/vidc.c +++ b/sound/oss/vidc.c @@ -227,7 +227,7 @@ static int vidc_audio_set_speed(int dev, int rate) } else { /*printk("VIDC: internal %d %d %d\n", rate, rate_int, hwrate);*/ hwctrl=0x00000003; - /* Allow rougly 0.4% tolerance */ + /* Allow roughly 0.4% tolerance */ if (diff_int > (rate/256)) rate=rate_int; } diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index ef41825a7b10..e90d103e177e 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -666,6 +666,15 @@ config SND_KORG1212 To compile this driver as a module, choose M here: the module will be called snd-korg1212. +config SND_LOLA + tristate "Digigram Lola" + select SND_PCM + help + Say Y to include support for Digigram Lola boards. + + To compile this driver as a module, choose M here: the module + will be called snd-lola. + config SND_LX6464ES tristate "Digigram LX6464ES" select SND_PCM diff --git a/sound/pci/Makefile b/sound/pci/Makefile index 9cf4348ec137..54fe325e3aa5 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -64,6 +64,7 @@ obj-$(CONFIG_SND) += \ ca0106/ \ cs46xx/ \ cs5535audio/ \ + lola/ \ lx6464es/ \ echoaudio/ \ emu10k1/ \ diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 4382d0fa6b9a..d8f6fd65ebbb 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -29,7 +29,7 @@ * PM support * MIDI support * Game Port support - * SG DMA support (this will need *alot* of work) + * SG DMA support (this will need *a lot* of work) */ #include <linux/init.h> diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index eac7b08a161d..2ca6f4f85b41 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1017,7 +1017,7 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) /*? also check ASI5000 samplerate source If external, only support external rate. - If internal and other stream playing, cant switch + If internal and other stream playing, can't switch */ init_timer(&dpcm->timer); diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h index 6fc025c448de..255429c32c1c 100644 --- a/sound/pci/asihpi/hpi.h +++ b/sound/pci/asihpi/hpi.h @@ -725,7 +725,7 @@ enum HPI_AESEBU_ERRORS { #define HPI_PAD_TITLE_LEN 64 /** The text string containing the comment. */ #define HPI_PAD_COMMENT_LEN 256 -/** The PTY when the tuner has not recieved any PTY. */ +/** The PTY when the tuner has not received any PTY. */ #define HPI_PAD_PROGRAM_TYPE_INVALID 0xffff /** \} */ diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c index 09befdafe09f..df4aed5295dd 100644 --- a/sound/pci/asihpi/hpi6000.c +++ b/sound/pci/asihpi/hpi6000.c @@ -420,7 +420,7 @@ static void subsys_create_adapter(struct hpi_message *phm, ao.priv = kzalloc(sizeof(struct hpi_hw_obj), GFP_KERNEL); if (!ao.priv) { - HPI_DEBUG_LOG(ERROR, "cant get mem for adapter object\n"); + HPI_DEBUG_LOG(ERROR, "can't get mem for adapter object\n"); phr->error = HPI_ERROR_MEMORY_ALLOC; return; } diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 3b5562acb55e..9d5df54a6b46 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -469,7 +469,7 @@ static void subsys_create_adapter(struct hpi_message *phm, ao.priv = kzalloc(sizeof(struct hpi_hw_obj), GFP_KERNEL); if (!ao.priv) { - HPI_DEBUG_LOG(ERROR, "cant get mem for adapter object\n"); + HPI_DEBUG_LOG(ERROR, "can't get mem for adapter object\n"); phr->error = HPI_ERROR_MEMORY_ALLOC; return; } diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index 4fab1595a6a2..bf5eced76bac 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -608,7 +608,7 @@ struct hpi_data_compat32 { #endif struct hpi_buffer { - /** placehoder for backward compatability (see dwBufferSize) */ + /** placehoder for backward compatibility (see dwBufferSize) */ struct hpi_msg_format reserved; u32 command; /**< HPI_BUFFER_CMD_xxx*/ u32 pci_address; /**< PCI physical address of buffer for DSP DMA */ diff --git a/sound/pci/asihpi/hpimsgx.c b/sound/pci/asihpi/hpimsgx.c index 888e37966c3f..7352a5f7b4f7 100644 --- a/sound/pci/asihpi/hpimsgx.c +++ b/sound/pci/asihpi/hpimsgx.c @@ -717,7 +717,7 @@ static u16 HPIMSGX__init(struct hpi_message *phm, return phr->error; } if (hr.error == 0) { - /* the adapter was created succesfully + /* the adapter was created successfully save the mapping for future use */ hpi_entry_points[hr.u.s.adapter_index] = entry_point_func; /* prepare adapter (pre-open streams etc.) */ diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index ecb8f4daf408..02f6e08f7592 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -104,7 +104,7 @@ #define MIX_PLAYB(x) (vortex->mixplayb[x]) #define MIX_SPDIF(x) (vortex->mixspdif[x]) -#define NR_WTPB 0x20 /* WT channels per eahc bank. */ +#define NR_WTPB 0x20 /* WT channels per each bank. */ /* Structs */ typedef struct { diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c index f4aa8ff6f5f9..9ae8b3b17651 100644 --- a/sound/pci/au88x0/au88x0_a3d.c +++ b/sound/pci/au88x0/au88x0_a3d.c @@ -53,7 +53,7 @@ a3dsrc_GetTimeConsts(a3dsrc_t * a, short *HrtfTrack, short *ItdTrack, } #endif -/* Atmospheric absorbtion. */ +/* Atmospheric absorption. */ static void a3dsrc_SetAtmosTarget(a3dsrc_t * a, short aa, short b, short c, short d, @@ -835,7 +835,7 @@ snd_vortex_a3d_filter_put(struct snd_kcontrol *kcontrol, params[i] = ucontrol->value.integer.value[i]; /* Translate generic filter params to a3d filter params. */ vortex_a3d_translate_filter(a->filter, params); - /* Atmospheric absorbtion and filtering. */ + /* Atmospheric absorption and filtering. */ a3dsrc_SetAtmosTarget(a, a->filter[0], a->filter[1], a->filter[2], a->filter[3], a->filter[4]); diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 04b10fdb0d71..c5f7ae46afef 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -44,10 +44,10 @@ static struct snd_pcm_hardware snd_vortex_playback_hw_adb = { .channels_min = 1, .channels_max = 2, .buffer_bytes_max = 0x10000, - .period_bytes_min = 0x1, + .period_bytes_min = 0x20, .period_bytes_max = 0x1000, .periods_min = 2, - .periods_max = 32, + .periods_max = 1024, }; #ifndef CHIP_AU8820 @@ -140,6 +140,9 @@ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_PERIOD_BYTES)) < 0) return err; + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 64); + if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) { #ifndef CHIP_AU8820 if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_A3D) { @@ -515,7 +518,7 @@ static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) return -ENODEV; /* idx indicates which kind of PCM device. ADB, SPDIF, I2S and A3D share the - * same dma engine. WT uses it own separate dma engine whcih cant capture. */ + * same dma engine. WT uses it own separate dma engine which can't capture. */ if (idx == VORTEX_PCM_ADB) nr_capt = nr; else diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 5715c4d05573..9b7a6346037a 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -140,7 +140,7 @@ * Possible remedies: * - use speaker (amplifier) output instead of headphone output * (in case crackling is due to overloaded output clipping) - * - plug card into a different PCI slot, preferrably one that isn't shared + * - plug card into a different PCI slot, preferably one that isn't shared * too much (this helps a lot, but not completely!) * - get rid of PCI VGA card, use AGP instead * - upgrade or downgrade BIOS diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h index fc53b9bca26d..e8e8ccc96403 100644 --- a/sound/pci/ca0106/ca0106.h +++ b/sound/pci/ca0106/ca0106.h @@ -51,7 +51,7 @@ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) * * - * This code was initally based on code from ALSA's emu10k1x.c which is: + * This code was initially based on code from ALSA's emu10k1x.c which is: * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> * * This program is free software; you can redistribute it and/or modify @@ -175,7 +175,7 @@ /* CA0106 pointer-offset register set, accessed through the PTR and DATA registers */ /********************************************************************************************************/ -/* Initally all registers from 0x00 to 0x3f have zero contents. */ +/* Initially all registers from 0x00 to 0x3f have zero contents. */ #define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */ /* One list entry: 4 bytes for DMA address, * 4 bytes for period_size << 16. @@ -223,7 +223,7 @@ * The jack has 4 poles. I will call 1 - Tip, 2 - Next to 1, 3 - Next to 2, 4 - Next to 3 * For Analogue: 1 -> Center Speaker, 2 -> Sub Woofer, 3 -> Ground, 4 -> Ground * For Digital: 1 -> Front SPDIF, 2 -> Rear SPDIF, 3 -> Center/Subwoofer SPDIF, 4 -> Ground. - * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Sheild on all three, 4 -> Red. + * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Shield on all three, 4 -> Red. * So, from this you can see that you cannot use a Standard 4 pole Video A/V cable with the SB Audigy LS card. */ /* The Front SPDIF PCM gets mixed with samples from the AC97 codec, so can only work for Stereo PCM and not AC3/DTS diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 01b49388fafd..437759239694 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -117,7 +117,7 @@ * DAC: Unknown * Trying to handle it like the SB0410. * - * This code was initally based on code from ALSA's emu10k1x.c which is: + * This code was initially based on code from ALSA's emu10k1x.c which is: * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> * * This program is free software; you can redistribute it and/or modify diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 630aa4998189..84f3f92436b5 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -42,7 +42,7 @@ * 0.0.18 * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) * - * This code was initally based on code from ALSA's emu10k1x.c which is: + * This code was initially based on code from ALSA's emu10k1x.c which is: * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> * * This program is free software; you can redistribute it and/or modify diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index ba96428c9f4c..c694464b1168 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -42,7 +42,7 @@ * 0.0.18 * Implement support for Line-in capture on SB Live 24bit. * - * This code was initally based on code from ALSA's emu10k1x.c which is: + * This code was initially based on code from ALSA's emu10k1x.c which is: * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> * * This program is free software; you can redistribute it and/or modify diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index b5bb036ef73c..f4e573555da3 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -73,7 +73,7 @@ MODULE_PARM_DESC(mpu_port, "MPU-401 port."); module_param_array(fm_port, long, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM port."); module_param_array(soft_ac3, bool, NULL, 0444); -MODULE_PARM_DESC(soft_ac3, "Sofware-conversion of raw SPDIF packets (model 033 only)."); +MODULE_PARM_DESC(soft_ac3, "Software-conversion of raw SPDIF packets (model 033 only)."); #ifdef SUPPORT_JOYSTICK module_param_array(joystick_port, int, NULL, 0444); MODULE_PARM_DESC(joystick_port, "Joystick port address."); @@ -656,8 +656,8 @@ out: } /* - * Program pll register bits, I assume that the 8 registers 0xf8 upto 0xff - * are mapped onto the 8 ADC/DAC sampling frequency which can be choosen + * Program pll register bits, I assume that the 8 registers 0xf8 up to 0xff + * are mapped onto the 8 ADC/DAC sampling frequency which can be chosen * at the register CM_REG_FUNCTRL1 (0x04). * Problem: other ways are also possible (any information about that?) */ @@ -666,7 +666,7 @@ static void snd_cmipci_set_pll(struct cmipci *cm, unsigned int rate, unsigned in unsigned int reg = CM_REG_PLL + slot; /* * Guess that this programs at reg. 0x04 the pos 15:13/12:10 - * for DSFC/ASFC (000 upto 111). + * for DSFC/ASFC (000 up to 111). */ /* FIXME: Init (Do we've to set an other register first before programming?) */ diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index b9321544c31c..13f33c0719d3 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1627,7 +1627,7 @@ static struct ct_atc atc_preset __devinitdata = { * Creates and initializes a hardware manager. * * Creates kmallocated ct_atc structure. Initializes hardware. - * Returns 0 if suceeds, or negative error code if fails. + * Returns 0 if succeeds, or negative error code if fails. */ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 0cf400f879f9..a5c957db5cea 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1285,7 +1285,7 @@ static int hw_trn_init(struct hw *hw, const struct trn_conf *info) hw_write_20kx(hw, PTPALX, ptp_phys_low); hw_write_20kx(hw, PTPAHX, ptp_phys_high); hw_write_20kx(hw, TRNCTL, trnctl); - hw_write_20kx(hw, TRNIS, 0x200c01); /* realy needed? */ + hw_write_20kx(hw, TRNIS, 0x200c01); /* really needed? */ return 0; } diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 957a311514c8..c250614dadd0 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -248,7 +248,7 @@ static int is_valid_page(struct snd_emu10k1 *emu, dma_addr_t addr) /* * map the given memory block on PTB. * if the block is already mapped, update the link order. - * if no empty pages are found, tries to release unsed memory blocks + * if no empty pages are found, tries to release unused memory blocks * and retry the mapping. */ int snd_emu10k1_memblk_map(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk) diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c index 61b8ab39800f..a81dc44228ea 100644 --- a/sound/pci/emu10k1/p16v.c +++ b/sound/pci/emu10k1/p16v.c @@ -69,7 +69,7 @@ * ADC: Philips 1361T (Stereo 24bit) * DAC: CS4382-K (8-channel, 24bit, 192Khz) * - * This code was initally based on code from ALSA's emu10k1x.c which is: + * This code was initially based on code from ALSA's emu10k1x.c which is: * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> * * This program is free software; you can redistribute it and/or modify diff --git a/sound/pci/emu10k1/p16v.h b/sound/pci/emu10k1/p16v.h index 00f4817533b1..4e0ee1a9747a 100644 --- a/sound/pci/emu10k1/p16v.h +++ b/sound/pci/emu10k1/p16v.h @@ -59,7 +59,7 @@ * ADC: Philips 1361T (Stereo 24bit) * DAC: CS4382-K (8-channel, 24bit, 192Khz) * - * This code was initally based on code from ALSA's emu10k1x.c which is: + * This code was initially based on code from ALSA's emu10k1x.c which is: * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> * * This program is free software; you can redistribute it and/or modify @@ -86,7 +86,7 @@ * The sample rate is also controlled by the same registers that control the rate of the EMU10K2 sample rate converters. */ -/* Initally all registers from 0x00 to 0x3f have zero contents. */ +/* Initially all registers from 0x00 to 0x3f have zero contents. */ #define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */ /* One list entry: 4 bytes for DMA address, * 4 bytes for period_size << 16. diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2c79e96d0324..759ade12e758 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -937,6 +937,7 @@ void snd_hda_shutup_pins(struct hda_codec *codec) } EXPORT_SYMBOL_HDA(snd_hda_shutup_pins); +#ifdef SND_HDA_NEEDS_RESUME /* Restore the pin controls cleared previously via snd_hda_shutup_pins() */ static void restore_shutup_pins(struct hda_codec *codec) { @@ -953,6 +954,7 @@ static void restore_shutup_pins(struct hda_codec *codec) } codec->pins_shutup = 0; } +#endif static void init_hda_cache(struct hda_cache_rec *cache, unsigned int record_size); @@ -1329,6 +1331,7 @@ static void purify_inactive_streams(struct hda_codec *codec) } } +#ifdef SND_HDA_NEEDS_RESUME /* clean up all streams; called from suspend */ static void hda_cleanup_all_streams(struct hda_codec *codec) { @@ -1340,6 +1343,7 @@ static void hda_cleanup_all_streams(struct hda_codec *codec) really_cleanup_stream(codec, p); } } +#endif /* * amp access functions @@ -3661,7 +3665,7 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) * with the proper parameters for set up. * ops.cleanup should be called in hw_free for clean up of streams. * - * This function returns 0 if successfull, or a negative error code. + * This function returns 0 if successful, or a negative error code. */ int __devinit snd_hda_build_pcms(struct hda_bus *bus) { @@ -4851,7 +4855,7 @@ EXPORT_SYMBOL_HDA(snd_hda_suspend); * * Returns 0 if successful. * - * This fucntion is defined only when POWER_SAVE isn't set. + * This function is defined only when POWER_SAVE isn't set. * In the power-save mode, the codec is resumed dynamically. */ int snd_hda_resume(struct hda_bus *bus) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 69e33869a53e..ad97d937d3a8 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3035,6 +3035,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), 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, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ {} diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 251773e45f61..715615a88a8d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1280,6 +1280,39 @@ static int simple_playback_pcm_prepare(struct hda_pcm_stream *hinfo, stream_tag, format, substream); } +static void nvhdmi_8ch_7x_set_info_frame_parameters(struct hda_codec *codec, + int channels) +{ + unsigned int chanmask; + int chan = channels ? (channels - 1) : 1; + + switch (channels) { + default: + case 0: + case 2: + chanmask = 0x00; + break; + case 4: + chanmask = 0x08; + break; + case 6: + chanmask = 0x0b; + break; + case 8: + chanmask = 0x13; + break; + } + + /* Set the audio infoframe channel allocation and checksum fields. The + * channel count is computed implicitly by the hardware. */ + snd_hda_codec_write(codec, 0x1, 0, + Nv_VERB_SET_Channel_Allocation, chanmask); + + snd_hda_codec_write(codec, 0x1, 0, + Nv_VERB_SET_Info_Frame_Checksum, + (0x71 - chan - chanmask)); +} + static int nvhdmi_8ch_7x_pcm_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) @@ -1298,6 +1331,10 @@ static int nvhdmi_8ch_7x_pcm_close(struct hda_pcm_stream *hinfo, AC_VERB_SET_STREAM_FORMAT, 0); } + /* The audio hardware sends a channel count of 0x7 (8ch) when all the + * streams are disabled. */ + nvhdmi_8ch_7x_set_info_frame_parameters(codec, 8); + return snd_hda_multi_out_dig_close(codec, &spec->multiout); } @@ -1308,37 +1345,16 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { int chs; - unsigned int dataDCC1, dataDCC2, chan, chanmask, channel_id; + unsigned int dataDCC1, dataDCC2, channel_id; int i; mutex_lock(&codec->spdif_mutex); chs = substream->runtime->channels; - chan = chs ? (chs - 1) : 1; - switch (chs) { - default: - case 0: - case 2: - chanmask = 0x00; - break; - case 4: - chanmask = 0x08; - break; - case 6: - chanmask = 0x0b; - break; - case 8: - chanmask = 0x13; - break; - } dataDCC1 = AC_DIG1_ENABLE | AC_DIG1_COPYRIGHT; dataDCC2 = 0x2; - /* set the Audio InforFrame Channel Allocation */ - snd_hda_codec_write(codec, 0x1, 0, - Nv_VERB_SET_Channel_Allocation, chanmask); - /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) snd_hda_codec_write(codec, @@ -1413,10 +1429,7 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo, } } - /* set the Audio Info Frame Checksum */ - snd_hda_codec_write(codec, 0x1, 0, - Nv_VERB_SET_Info_Frame_Checksum, - (0x71 - chan - chanmask)); + nvhdmi_8ch_7x_set_info_frame_parameters(codec, chs); mutex_unlock(&codec->spdif_mutex); return 0; @@ -1512,6 +1525,11 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) spec->multiout.max_channels = 8; spec->pcm_playback = &nvhdmi_pcm_playback_8ch_7x; codec->patch_ops = nvhdmi_patch_ops_8ch_7x; + + /* Initialize the audio infoframe channel mask and checksum to something + * valid */ + nvhdmi_8ch_7x_set_info_frame_parameters(codec, 8); + return 0; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 12c6f4508c54..c82979a8cd09 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -549,7 +549,7 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, /* * Control the mode of pin widget settings via the mixer. "pc" is used - * instead of "%" to avoid consequences of accidently treating the % as + * instead of "%" to avoid consequences of accidentally treating the % as * being part of a format specifier. Maximum allowed length of a value is * 63 characters plus NULL terminator. * @@ -1704,11 +1704,11 @@ static void alc_apply_fixup(struct hda_codec *codec, int action) codec->chip_name, fix->type); break; } - if (!fix[id].chained) + if (!fix->chained) break; if (++depth > 10) break; - id = fix[id].chain_id; + id = fix->chain_id; } } @@ -5645,6 +5645,7 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, static struct snd_pci_quirk beep_white_list[] = { SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1), SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1), + SND_PCI_QUIRK(0x1043, 0x831a, "EeePC", 1), SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1), {} }; @@ -9836,7 +9837,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL), - SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavilion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), @@ -9863,6 +9864,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL), SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG), @@ -10699,7 +10701,6 @@ enum { PINFIX_LENOVO_Y530, PINFIX_PB_M5210, PINFIX_ACER_ASPIRE_7736, - PINFIX_GIGABYTE_880GM, }; static const struct alc_fixup alc882_fixups[] = { @@ -10731,13 +10732,6 @@ static const struct alc_fixup alc882_fixups[] = { .type = ALC_FIXUP_SKU, .v.sku = ALC_FIXUP_SKU_IGNORE, }, - [PINFIX_GIGABYTE_880GM] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x14, 0x1114410 }, /* set as speaker */ - { } - } - }, }; static struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -10745,7 +10739,6 @@ static struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736), - SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", PINFIX_GIGABYTE_880GM), {} }; @@ -14124,7 +14117,7 @@ static hda_nid_t alc269vb_capsrc_nids[1] = { }; static hda_nid_t alc269_adc_candidates[] = { - 0x08, 0x09, 0x07, + 0x08, 0x09, 0x07, 0x11, }; #define alc269_modes alc260_modes @@ -14868,6 +14861,23 @@ static void alc269_fixup_hweq(struct hda_codec *codec, alc_write_coef_idx(codec, 0x1e, coef | 0x80); } +static void alc271_fixup_dmic(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + static struct hda_verb verbs[] = { + {0x20, AC_VERB_SET_COEF_INDEX, 0x0d}, + {0x20, AC_VERB_SET_PROC_COEF, 0x4000}, + {} + }; + unsigned int cfg; + + if (strcmp(codec->chip_name, "ALC271X")) + return; + cfg = snd_hda_codec_get_pincfg(codec, 0x12); + if (get_defcfg_connect(cfg) == AC_JACK_PORT_FIXED) + snd_hda_sequence_write(codec, verbs); +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -14876,6 +14886,7 @@ enum { ALC269_FIXUP_ASUS_G73JW, ALC269_FIXUP_LENOVO_EAPD, ALC275_FIXUP_SONY_HWEQ, + ALC271_FIXUP_DMIC, }; static const struct alc_fixup alc269_fixups[] = { @@ -14929,7 +14940,11 @@ static const struct alc_fixup alc269_fixups[] = { .v.func = alc269_fixup_hweq, .chained = true, .chain_id = ALC275_FIXUP_SONY_VAIO_GPIO2 - } + }, + [ALC271_FIXUP_DMIC] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc271_fixup_dmic, + }, }; static struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -14938,6 +14953,7 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), + SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), @@ -18782,6 +18798,8 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO), SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", + ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x152d, 0x2304, "Quanta WH1", ALC663_ASUS_H13), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1631, 0xc10c, "PB RS65", ALC663_ASUS_M51VA), @@ -19455,7 +19473,7 @@ enum { ALC662_FIXUP_IDEAPAD, ALC272_FIXUP_MARIO, ALC662_FIXUP_CZC_P10T, - ALC662_FIXUP_GIGABYTE, + ALC662_FIXUP_SKU_IGNORE, }; static const struct alc_fixup alc662_fixups[] = { @@ -19484,20 +19502,17 @@ static const struct alc_fixup alc662_fixups[] = { {} } }, - [ALC662_FIXUP_GIGABYTE] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x14, 0x1114410 }, /* set as speaker */ - { } - } + [ALC662_FIXUP_SKU_IGNORE] = { + .type = ALC_FIXUP_SKU, + .v.sku = ALC_FIXUP_SKU_IGNORE, }, }; static struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), + SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), - SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", ALC662_FIXUP_GIGABYTE), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 05fcd60cc46f..94d19c03a7f4 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2475,7 +2475,7 @@ static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol, spec->hp_switch = ucontrol->value.integer.value[0] ? nid : 0; - /* check to be sure that the ports are upto date with + /* check to be sure that the ports are up to date with * switch changes */ stac_issue_unsol_event(codec, nid); @@ -3408,6 +3408,9 @@ static int get_connection_index(struct hda_codec *codec, hda_nid_t mux, hda_nid_t conn[HDA_MAX_NUM_INPUTS]; int i, nums; + if (!(get_wcaps(codec, mux) & AC_WCAP_CONN_LIST)) + return -1; + nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn)); for (i = 0; i < nums; i++) if (conn[i] == nid) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 1371b57c11e8..0997031c48d2 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1292,14 +1292,18 @@ static void notify_aa_path_ctls(struct hda_codec *codec) { int i; struct snd_ctl_elem_id id; - const char *labels[] = {"Mic", "Front Mic", "Line"}; + const char *labels[] = {"Mic", "Front Mic", "Line", "Rear Mic"}; + struct snd_kcontrol *ctl; memset(&id, 0, sizeof(id)); id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; for (i = 0; i < ARRAY_SIZE(labels); i++) { sprintf(id.name, "%s Playback Volume", labels[i]); - snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE, - &id); + ctl = snd_hda_find_mixer_ctl(codec, id.name); + if (ctl) + snd_ctl_notify(codec->bus->card, + SNDRV_CTL_EVENT_MASK_VALUE, + &ctl->id); } } diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 2f6252266a02..3e4f8c12ffce 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -148,7 +148,7 @@ static void aureon_pca9554_write(struct snd_ice1712 *ice, unsigned char reg, udelay(100); /* * send device address, command and value, - * skipping ack cycles inbetween + * skipping ack cycles in between */ for (j = 0; j < 3; j++) { switch (j) { @@ -2143,7 +2143,7 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) ice->num_total_adcs = 2; } - /* to remeber the register values of CS8415 */ + /* to remember the register values of CS8415 */ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); if (!ice->akm) return -ENOMEM; diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 4fc6d8bc637e..f4594d76b6ea 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2755,7 +2755,7 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, return err; } if (c->mpu401_1_name) - /* Prefered name available in card_info */ + /* Preferred name available in card_info */ snprintf(ice->rmidi[0]->name, sizeof(ice->rmidi[0]->name), "%s %d", c->mpu401_1_name, card->number); @@ -2772,7 +2772,7 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, return err; } if (c->mpu401_2_name) - /* Prefered name available in card_info */ + /* Preferred name available in card_info */ snprintf(ice->rmidi[1]->name, sizeof(ice->rmidi[1]->name), "%s %d", c->mpu401_2_name, diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c index cdb873f5da50..92c1160d7ab5 100644 --- a/sound/pci/ice1712/pontis.c +++ b/sound/pci/ice1712/pontis.c @@ -768,7 +768,7 @@ static int __devinit pontis_init(struct snd_ice1712 *ice) ice->num_total_dacs = 2; ice->num_total_adcs = 2; - /* to remeber the register values */ + /* to remember the register values */ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); if (! ice->akm) return -ENOMEM; diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 6a9fee3ee78f..764cc93dbca4 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -1046,7 +1046,7 @@ static int __devinit prodigy_hifi_init(struct snd_ice1712 *ice) * don't call snd_ice1712_gpio_get/put(), otherwise it's overwritten */ ice->gpio.saved[0] = 0; - /* to remeber the register values */ + /* to remember the register values */ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); if (! ice->akm) @@ -1128,7 +1128,7 @@ static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice) * don't call snd_ice1712_gpio_get/put(), otherwise it's overwritten */ ice->gpio.saved[0] = 0; - /* to remeber the register values */ + /* to remember the register values */ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); if (! ice->akm) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 629a5494347a..6c896dbfd796 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -534,7 +534,7 @@ static int snd_intel8x0_codec_semaphore(struct intel8x0 *chip, unsigned int code udelay(10); } while (time--); - /* access to some forbidden (non existant) ac97 registers will not + /* access to some forbidden (non existent) ac97 registers will not * reset the semaphore. So even if you don't get the semaphore, still * continue the access. We don't need the semaphore anyway. */ snd_printk(KERN_ERR "codec_semaphore: semaphore is not ready [0x%x][0x%x]\n", diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index bae3a279f75f..f3353b49c785 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -331,7 +331,7 @@ static int snd_intel8x0m_codec_semaphore(struct intel8x0m *chip, unsigned int co udelay(10); } while (time--); - /* access to some forbidden (non existant) ac97 registers will not + /* access to some forbidden (non existent) ac97 registers will not * reset the semaphore. So even if you don't get the semaphore, still * continue the access. We don't need the semaphore anyway. */ snd_printk(KERN_ERR "codec_semaphore: semaphore is not ready [0x%x][0x%x]\n", diff --git a/sound/pci/lola/Makefile b/sound/pci/lola/Makefile new file mode 100644 index 000000000000..8178a2a59d00 --- /dev/null +++ b/sound/pci/lola/Makefile @@ -0,0 +1,4 @@ +snd-lola-y := lola.o lola_pcm.o lola_clock.o lola_mixer.o +snd-lola-$(CONFIG_SND_DEBUG) += lola_proc.o + +obj-$(CONFIG_SND_LOLA) += snd-lola.o diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c new file mode 100644 index 000000000000..34b24286d279 --- /dev/null +++ b/sound/pci/lola/lola.c @@ -0,0 +1,791 @@ +/* + * Support for Digigram Lola PCI-e boards + * + * Copyright (c) 2011 Takashi Iwai <tiwai@suse.de> + * + * 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/kernel.h> +#include <linux/init.h> +#include <linux/moduleparam.h> +#include <linux/dma-mapping.h> +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <linux/slab.h> +#include <linux/pci.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include "lola.h" + +/* Standard options */ +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Digigram Lola driver."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for Digigram Lola driver."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Digigram Lola driver."); + +/* Lola-specific options */ + +/* for instance use always max granularity which is compatible + * with all sample rates + */ +static int granularity[SNDRV_CARDS] = { + [0 ... (SNDRV_CARDS - 1)] = LOLA_GRANULARITY_MAX +}; + +/* below a sample_rate of 16kHz the analogue audio quality is NOT excellent */ +static int sample_rate_min[SNDRV_CARDS] = { + [0 ... (SNDRV_CARDS - 1) ] = 16000 +}; + +module_param_array(granularity, int, NULL, 0444); +MODULE_PARM_DESC(granularity, "Granularity value"); +module_param_array(sample_rate_min, int, NULL, 0444); +MODULE_PARM_DESC(sample_rate_min, "Minimal sample rate"); + +/* + */ + +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Digigram, Lola}}"); +MODULE_DESCRIPTION("Digigram Lola driver"); +MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>"); + +#ifdef CONFIG_SND_DEBUG_VERBOSE +static int debug; +module_param(debug, int, 0644); +#define verbose_debug(fmt, args...) \ + do { if (debug > 1) printk(KERN_DEBUG SFX fmt, ##args); } while (0) +#else +#define verbose_debug(fmt, args...) +#endif + +/* + * pseudo-codec read/write via CORB/RIRB + */ + +static int corb_send_verb(struct lola *chip, unsigned int nid, + unsigned int verb, unsigned int data, + unsigned int extdata) +{ + unsigned long flags; + int ret = -EIO; + + chip->last_cmd_nid = nid; + chip->last_verb = verb; + chip->last_data = data; + chip->last_extdata = extdata; + data |= (nid << 20) | (verb << 8); + + spin_lock_irqsave(&chip->reg_lock, flags); + if (chip->rirb.cmds < LOLA_CORB_ENTRIES - 1) { + unsigned int wp = chip->corb.wp + 1; + wp %= LOLA_CORB_ENTRIES; + chip->corb.wp = wp; + chip->corb.buf[wp * 2] = cpu_to_le32(data); + chip->corb.buf[wp * 2 + 1] = cpu_to_le32(extdata); + lola_writew(chip, BAR0, CORBWP, wp); + chip->rirb.cmds++; + smp_wmb(); + ret = 0; + } + spin_unlock_irqrestore(&chip->reg_lock, flags); + return ret; +} + +static void lola_queue_unsol_event(struct lola *chip, unsigned int res, + unsigned int res_ex) +{ + lola_update_ext_clock_freq(chip, res); +} + +/* retrieve RIRB entry - called from interrupt handler */ +static void lola_update_rirb(struct lola *chip) +{ + unsigned int rp, wp; + u32 res, res_ex; + + wp = lola_readw(chip, BAR0, RIRBWP); + if (wp == chip->rirb.wp) + return; + chip->rirb.wp = wp; + + while (chip->rirb.rp != wp) { + chip->rirb.rp++; + chip->rirb.rp %= LOLA_CORB_ENTRIES; + + rp = chip->rirb.rp << 1; /* an RIRB entry is 8-bytes */ + res_ex = le32_to_cpu(chip->rirb.buf[rp + 1]); + res = le32_to_cpu(chip->rirb.buf[rp]); + if (res_ex & LOLA_RIRB_EX_UNSOL_EV) + lola_queue_unsol_event(chip, res, res_ex); + else if (chip->rirb.cmds) { + chip->res = res; + chip->res_ex = res_ex; + smp_wmb(); + chip->rirb.cmds--; + } + } +} + +static int rirb_get_response(struct lola *chip, unsigned int *val, + unsigned int *extval) +{ + unsigned long timeout; + + again: + timeout = jiffies + msecs_to_jiffies(1000); + for (;;) { + if (chip->polling_mode) { + spin_lock_irq(&chip->reg_lock); + lola_update_rirb(chip); + spin_unlock_irq(&chip->reg_lock); + } + if (!chip->rirb.cmds) { + *val = chip->res; + if (extval) + *extval = chip->res_ex; + verbose_debug("get_response: %x, %x\n", + chip->res, chip->res_ex); + if (chip->res_ex & LOLA_RIRB_EX_ERROR) { + printk(KERN_WARNING SFX "RIRB ERROR: " + "NID=%x, verb=%x, data=%x, ext=%x\n", + chip->last_cmd_nid, + chip->last_verb, chip->last_data, + chip->last_extdata); + return -EIO; + } + return 0; + } + if (time_after(jiffies, timeout)) + break; + udelay(20); + cond_resched(); + } + printk(KERN_WARNING SFX "RIRB response error\n"); + if (!chip->polling_mode) { + printk(KERN_WARNING SFX "switching to polling mode\n"); + chip->polling_mode = 1; + goto again; + } + return -EIO; +} + +/* aynchronous write of a codec verb with data */ +int lola_codec_write(struct lola *chip, unsigned int nid, unsigned int verb, + unsigned int data, unsigned int extdata) +{ + verbose_debug("codec_write NID=%x, verb=%x, data=%x, ext=%x\n", + nid, verb, data, extdata); + return corb_send_verb(chip, nid, verb, data, extdata); +} + +/* write a codec verb with data and read the returned status */ +int lola_codec_read(struct lola *chip, unsigned int nid, unsigned int verb, + unsigned int data, unsigned int extdata, + unsigned int *val, unsigned int *extval) +{ + int err; + + verbose_debug("codec_read NID=%x, verb=%x, data=%x, ext=%x\n", + nid, verb, data, extdata); + err = corb_send_verb(chip, nid, verb, data, extdata); + if (err < 0) + return err; + err = rirb_get_response(chip, val, extval); + return err; +} + +/* flush all pending codec writes */ +int lola_codec_flush(struct lola *chip) +{ + unsigned int tmp; + return rirb_get_response(chip, &tmp, NULL); +} + +/* + * interrupt handler + */ +static irqreturn_t lola_interrupt(int irq, void *dev_id) +{ + struct lola *chip = dev_id; + unsigned int notify_ins, notify_outs, error_ins, error_outs; + int handled = 0; + int i; + + notify_ins = notify_outs = error_ins = error_outs = 0; + spin_lock(&chip->reg_lock); + for (;;) { + unsigned int status, in_sts, out_sts; + unsigned int reg; + + status = lola_readl(chip, BAR1, DINTSTS); + if (!status || status == -1) + break; + + in_sts = lola_readl(chip, BAR1, DIINTSTS); + out_sts = lola_readl(chip, BAR1, DOINTSTS); + + /* clear Input Interrupts */ + for (i = 0; in_sts && i < chip->pcm[CAPT].num_streams; i++) { + if (!(in_sts & (1 << i))) + continue; + in_sts &= ~(1 << i); + reg = lola_dsd_read(chip, i, STS); + if (reg & LOLA_DSD_STS_DESE) /* error */ + error_ins |= (1 << i); + if (reg & LOLA_DSD_STS_BCIS) /* notify */ + notify_ins |= (1 << i); + /* clear */ + lola_dsd_write(chip, i, STS, reg); + } + + /* clear Output Interrupts */ + for (i = 0; out_sts && i < chip->pcm[PLAY].num_streams; i++) { + if (!(out_sts & (1 << i))) + continue; + out_sts &= ~(1 << i); + reg = lola_dsd_read(chip, i + MAX_STREAM_IN_COUNT, STS); + if (reg & LOLA_DSD_STS_DESE) /* error */ + error_outs |= (1 << i); + if (reg & LOLA_DSD_STS_BCIS) /* notify */ + notify_outs |= (1 << i); + lola_dsd_write(chip, i + MAX_STREAM_IN_COUNT, STS, reg); + } + + if (status & LOLA_DINT_CTRL) { + unsigned char rbsts; /* ring status is byte access */ + rbsts = lola_readb(chip, BAR0, RIRBSTS); + rbsts &= LOLA_RIRB_INT_MASK; + if (rbsts) + lola_writeb(chip, BAR0, RIRBSTS, rbsts); + rbsts = lola_readb(chip, BAR0, CORBSTS); + rbsts &= LOLA_CORB_INT_MASK; + if (rbsts) + lola_writeb(chip, BAR0, CORBSTS, rbsts); + + lola_update_rirb(chip); + } + + if (status & (LOLA_DINT_FIFOERR | LOLA_DINT_MUERR)) { + /* clear global fifo error interrupt */ + lola_writel(chip, BAR1, DINTSTS, + (status & (LOLA_DINT_FIFOERR | LOLA_DINT_MUERR))); + } + handled = 1; + } + spin_unlock(&chip->reg_lock); + + lola_pcm_update(chip, &chip->pcm[CAPT], notify_ins); + lola_pcm_update(chip, &chip->pcm[PLAY], notify_outs); + + return IRQ_RETVAL(handled); +} + + +/* + * controller + */ +static int reset_controller(struct lola *chip) +{ + unsigned int gctl = lola_readl(chip, BAR0, GCTL); + unsigned long end_time; + + if (gctl) { + /* to be sure */ + lola_writel(chip, BAR1, BOARD_MODE, 0); + return 0; + } + + chip->cold_reset = 1; + lola_writel(chip, BAR0, GCTL, LOLA_GCTL_RESET); + end_time = jiffies + msecs_to_jiffies(200); + do { + msleep(1); + gctl = lola_readl(chip, BAR0, GCTL); + if (gctl) + break; + } while (time_before(jiffies, end_time)); + if (!gctl) { + printk(KERN_ERR SFX "cannot reset controller\n"); + return -EIO; + } + return 0; +} + +static void lola_irq_enable(struct lola *chip) +{ + unsigned int val; + + /* enalbe all I/O streams */ + val = (1 << chip->pcm[PLAY].num_streams) - 1; + lola_writel(chip, BAR1, DOINTCTL, val); + val = (1 << chip->pcm[CAPT].num_streams) - 1; + lola_writel(chip, BAR1, DIINTCTL, val); + + /* enable global irqs */ + val = LOLA_DINT_GLOBAL | LOLA_DINT_CTRL | LOLA_DINT_FIFOERR | + LOLA_DINT_MUERR; + lola_writel(chip, BAR1, DINTCTL, val); +} + +static void lola_irq_disable(struct lola *chip) +{ + lola_writel(chip, BAR1, DINTCTL, 0); + lola_writel(chip, BAR1, DIINTCTL, 0); + lola_writel(chip, BAR1, DOINTCTL, 0); +} + +static int setup_corb_rirb(struct lola *chip) +{ + int err; + unsigned char tmp; + unsigned long end_time; + + err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + PAGE_SIZE, &chip->rb); + if (err < 0) + return err; + + chip->corb.addr = chip->rb.addr; + chip->corb.buf = (u32 *)chip->rb.area; + chip->rirb.addr = chip->rb.addr + 2048; + chip->rirb.buf = (u32 *)(chip->rb.area + 2048); + + /* disable ringbuffer DMAs */ + lola_writeb(chip, BAR0, RIRBCTL, 0); + lola_writeb(chip, BAR0, CORBCTL, 0); + + end_time = jiffies + msecs_to_jiffies(200); + do { + if (!lola_readb(chip, BAR0, RIRBCTL) && + !lola_readb(chip, BAR0, CORBCTL)) + break; + msleep(1); + } while (time_before(jiffies, end_time)); + + /* CORB set up */ + lola_writel(chip, BAR0, CORBLBASE, (u32)chip->corb.addr); + lola_writel(chip, BAR0, CORBUBASE, upper_32_bits(chip->corb.addr)); + /* set the corb size to 256 entries */ + lola_writeb(chip, BAR0, CORBSIZE, 0x02); + /* set the corb write pointer to 0 */ + lola_writew(chip, BAR0, CORBWP, 0); + /* reset the corb hw read pointer */ + lola_writew(chip, BAR0, CORBRP, LOLA_RBRWP_CLR); + /* enable corb dma */ + lola_writeb(chip, BAR0, CORBCTL, LOLA_RBCTL_DMA_EN); + /* clear flags if set */ + tmp = lola_readb(chip, BAR0, CORBSTS) & LOLA_CORB_INT_MASK; + if (tmp) + lola_writeb(chip, BAR0, CORBSTS, tmp); + chip->corb.wp = 0; + + /* RIRB set up */ + lola_writel(chip, BAR0, RIRBLBASE, (u32)chip->rirb.addr); + lola_writel(chip, BAR0, RIRBUBASE, upper_32_bits(chip->rirb.addr)); + /* set the rirb size to 256 entries */ + lola_writeb(chip, BAR0, RIRBSIZE, 0x02); + /* reset the rirb hw write pointer */ + lola_writew(chip, BAR0, RIRBWP, LOLA_RBRWP_CLR); + /* set N=1, get RIRB response interrupt for new entry */ + lola_writew(chip, BAR0, RINTCNT, 1); + /* enable rirb dma and response irq */ + lola_writeb(chip, BAR0, RIRBCTL, LOLA_RBCTL_DMA_EN | LOLA_RBCTL_IRQ_EN); + /* clear flags if set */ + tmp = lola_readb(chip, BAR0, RIRBSTS) & LOLA_RIRB_INT_MASK; + if (tmp) + lola_writeb(chip, BAR0, RIRBSTS, tmp); + chip->rirb.rp = chip->rirb.cmds = 0; + + return 0; +} + +static void stop_corb_rirb(struct lola *chip) +{ + /* disable ringbuffer DMAs */ + lola_writeb(chip, BAR0, RIRBCTL, 0); + lola_writeb(chip, BAR0, CORBCTL, 0); +} + +static void lola_reset_setups(struct lola *chip) +{ + /* update the granularity */ + lola_set_granularity(chip, chip->granularity, true); + /* update the sample clock */ + lola_set_clock_index(chip, chip->clock.cur_index); + /* enable unsolicited events of the clock widget */ + lola_enable_clock_events(chip); + /* update the analog gains */ + lola_setup_all_analog_gains(chip, CAPT, false); /* input, update */ + /* update SRC configuration if applicable */ + lola_set_src_config(chip, chip->input_src_mask, false); + /* update the analog outputs */ + lola_setup_all_analog_gains(chip, PLAY, false); /* output, update */ +} + +static int lola_parse_tree(struct lola *chip) +{ + unsigned int val; + int nid, err; + + err = lola_read_param(chip, 0, LOLA_PAR_VENDOR_ID, &val); + if (err < 0) { + printk(KERN_ERR SFX "Can't read VENDOR_ID\n"); + return err; + } + val >>= 16; + if (val != 0x1369) { + printk(KERN_ERR SFX "Unknown codec vendor 0x%x\n", val); + return -EINVAL; + } + + err = lola_read_param(chip, 1, LOLA_PAR_FUNCTION_TYPE, &val); + if (err < 0) { + printk(KERN_ERR SFX "Can't read FUNCTION_TYPE for 0x%x\n", nid); + return err; + } + if (val != 1) { + printk(KERN_ERR SFX "Unknown function type %d\n", val); + return -EINVAL; + } + + err = lola_read_param(chip, 1, LOLA_PAR_SPECIFIC_CAPS, &val); + if (err < 0) { + printk(KERN_ERR SFX "Can't read SPECCAPS\n"); + return err; + } + chip->lola_caps = val; + chip->pin[CAPT].num_pins = LOLA_AFG_INPUT_PIN_COUNT(chip->lola_caps); + chip->pin[PLAY].num_pins = LOLA_AFG_OUTPUT_PIN_COUNT(chip->lola_caps); + snd_printdd(SFX "speccaps=0x%x, pins in=%d, out=%d\n", + chip->lola_caps, + chip->pin[CAPT].num_pins, chip->pin[PLAY].num_pins); + + if (chip->pin[CAPT].num_pins > MAX_AUDIO_INOUT_COUNT || + chip->pin[PLAY].num_pins > MAX_AUDIO_INOUT_COUNT) { + printk(KERN_ERR SFX "Invalid Lola-spec caps 0x%x\n", val); + return -EINVAL; + } + + nid = 0x02; + err = lola_init_pcm(chip, CAPT, &nid); + if (err < 0) + return err; + err = lola_init_pcm(chip, PLAY, &nid); + if (err < 0) + return err; + + err = lola_init_pins(chip, CAPT, &nid); + if (err < 0) + return err; + err = lola_init_pins(chip, PLAY, &nid); + if (err < 0) + return err; + + if (LOLA_AFG_CLOCK_WIDGET_PRESENT(chip->lola_caps)) { + err = lola_init_clock_widget(chip, nid); + if (err < 0) + return err; + nid++; + } + if (LOLA_AFG_MIXER_WIDGET_PRESENT(chip->lola_caps)) { + err = lola_init_mixer_widget(chip, nid); + if (err < 0) + return err; + nid++; + } + + /* enable unsolicited events of the clock widget */ + err = lola_enable_clock_events(chip); + if (err < 0) + return err; + + /* if last ResetController was not a ColdReset, we don't know + * the state of the card; initialize here again + */ + if (!chip->cold_reset) { + lola_reset_setups(chip); + chip->cold_reset = 1; + } else { + /* set the granularity if it is not the default */ + if (chip->granularity != LOLA_GRANULARITY_MIN) + lola_set_granularity(chip, chip->granularity, true); + } + + return 0; +} + +static void lola_stop_hw(struct lola *chip) +{ + stop_corb_rirb(chip); + lola_irq_disable(chip); +} + +static void lola_free(struct lola *chip) +{ + if (chip->initialized) + lola_stop_hw(chip); + lola_free_pcm(chip); + lola_free_mixer(chip); + if (chip->irq >= 0) + free_irq(chip->irq, (void *)chip); + if (chip->bar[0].remap_addr) + iounmap(chip->bar[0].remap_addr); + if (chip->bar[1].remap_addr) + iounmap(chip->bar[1].remap_addr); + if (chip->rb.area) + snd_dma_free_pages(&chip->rb); + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + kfree(chip); +} + +static int lola_dev_free(struct snd_device *device) +{ + lola_free(device->device_data); + return 0; +} + +static int __devinit lola_create(struct snd_card *card, struct pci_dev *pci, + int dev, struct lola **rchip) +{ + struct lola *chip; + int err; + unsigned int dever; + static struct snd_device_ops ops = { + .dev_free = lola_dev_free, + }; + + *rchip = NULL; + + err = pci_enable_device(pci); + if (err < 0) + return err; + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (!chip) { + snd_printk(KERN_ERR SFX "cannot allocate chip\n"); + pci_disable_device(pci); + return -ENOMEM; + } + + spin_lock_init(&chip->reg_lock); + mutex_init(&chip->open_mutex); + chip->card = card; + chip->pci = pci; + chip->irq = -1; + + chip->granularity = granularity[dev]; + switch (chip->granularity) { + case 8: + chip->sample_rate_max = 48000; + break; + case 16: + chip->sample_rate_max = 96000; + break; + case 32: + chip->sample_rate_max = 192000; + break; + default: + snd_printk(KERN_WARNING SFX + "Invalid granularity %d, reset to %d\n", + chip->granularity, LOLA_GRANULARITY_MAX); + chip->granularity = LOLA_GRANULARITY_MAX; + chip->sample_rate_max = 192000; + break; + } + chip->sample_rate_min = sample_rate_min[dev]; + if (chip->sample_rate_min > chip->sample_rate_max) { + snd_printk(KERN_WARNING SFX + "Invalid sample_rate_min %d, reset to 16000\n", + chip->sample_rate_min); + chip->sample_rate_min = 16000; + } + + err = pci_request_regions(pci, DRVNAME); + if (err < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + + chip->bar[0].addr = pci_resource_start(pci, 0); + chip->bar[0].remap_addr = pci_ioremap_bar(pci, 0); + chip->bar[1].addr = pci_resource_start(pci, 2); + chip->bar[1].remap_addr = pci_ioremap_bar(pci, 2); + if (!chip->bar[0].remap_addr || !chip->bar[1].remap_addr) { + snd_printk(KERN_ERR SFX "ioremap error\n"); + err = -ENXIO; + goto errout; + } + + pci_set_master(pci); + + err = reset_controller(chip); + if (err < 0) + goto errout; + + if (request_irq(pci->irq, lola_interrupt, IRQF_SHARED, + DRVNAME, chip)) { + printk(KERN_ERR SFX "unable to grab IRQ %d\n", pci->irq); + err = -EBUSY; + goto errout; + } + chip->irq = pci->irq; + synchronize_irq(chip->irq); + + dever = lola_readl(chip, BAR1, DEVER); + chip->pcm[CAPT].num_streams = (dever >> 0) & 0x3ff; + chip->pcm[PLAY].num_streams = (dever >> 10) & 0x3ff; + chip->version = (dever >> 24) & 0xff; + snd_printdd(SFX "streams in=%d, out=%d, version=0x%x\n", + chip->pcm[CAPT].num_streams, chip->pcm[PLAY].num_streams, + chip->version); + + /* Test LOLA_BAR1_DEVER */ + if (chip->pcm[CAPT].num_streams > MAX_STREAM_IN_COUNT || + chip->pcm[PLAY].num_streams > MAX_STREAM_OUT_COUNT || + (!chip->pcm[CAPT].num_streams && + !chip->pcm[PLAY].num_streams)) { + printk(KERN_ERR SFX "invalid DEVER = %x\n", dever); + err = -EINVAL; + goto errout; + } + + err = setup_corb_rirb(chip); + if (err < 0) + goto errout; + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_printk(KERN_ERR SFX "Error creating device [card]!\n"); + goto errout; + } + + strcpy(card->driver, "Lola"); + strlcpy(card->shortname, "Digigram Lola", sizeof(card->shortname)); + snprintf(card->longname, sizeof(card->longname), + "%s at 0x%lx irq %i", + card->shortname, chip->bar[0].addr, chip->irq); + strcpy(card->mixername, card->shortname); + + lola_irq_enable(chip); + + chip->initialized = 1; + *rchip = chip; + return 0; + + errout: + lola_free(chip); + return err; +} + +static int __devinit lola_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct lola *chip; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) { + snd_printk(KERN_ERR SFX "Error creating card!\n"); + return err; + } + + snd_card_set_dev(card, &pci->dev); + + err = lola_create(card, pci, dev, &chip); + if (err < 0) + goto out_free; + card->private_data = chip; + + err = lola_parse_tree(chip); + if (err < 0) + goto out_free; + + err = lola_create_pcm(chip); + if (err < 0) + goto out_free; + + err = lola_create_mixer(chip); + if (err < 0) + goto out_free; + + lola_proc_debug_new(chip); + + err = snd_card_register(card); + if (err < 0) + goto out_free; + + pci_set_drvdata(pci, card); + dev++; + return err; +out_free: + snd_card_free(card); + return err; +} + +static void __devexit lola_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +/* PCI IDs */ +static DEFINE_PCI_DEVICE_TABLE(lola_ids) = { + { PCI_VDEVICE(DIGIGRAM, 0x0001) }, + { 0, } +}; +MODULE_DEVICE_TABLE(pci, lola_ids); + +/* pci_driver definition */ +static struct pci_driver driver = { + .name = DRVNAME, + .id_table = lola_ids, + .probe = lola_probe, + .remove = __devexit_p(lola_remove), +}; + +static int __init alsa_card_lola_init(void) +{ + return pci_register_driver(&driver); +} + +static void __exit alsa_card_lola_exit(void) +{ + pci_unregister_driver(&driver); +} + +module_init(alsa_card_lola_init) +module_exit(alsa_card_lola_exit) diff --git a/sound/pci/lola/lola.h b/sound/pci/lola/lola.h new file mode 100644 index 000000000000..d5708e29b16d --- /dev/null +++ b/sound/pci/lola/lola.h @@ -0,0 +1,527 @@ +/* + * Support for Digigram Lola PCI-e boards + * + * Copyright (c) 2011 Takashi Iwai <tiwai@suse.de> + * + * 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. + */ + +#ifndef _LOLA_H +#define _LOLA_H + +#define DRVNAME "snd-lola" +#define SFX DRVNAME ": " + +/* + * Lola HD Audio Registers BAR0 + */ +#define LOLA_BAR0_GCAP 0x00 +#define LOLA_BAR0_VMIN 0x02 +#define LOLA_BAR0_VMAJ 0x03 +#define LOLA_BAR0_OUTPAY 0x04 +#define LOLA_BAR0_INPAY 0x06 +#define LOLA_BAR0_GCTL 0x08 +#define LOLA_BAR0_WAKEEN 0x0c +#define LOLA_BAR0_STATESTS 0x0e +#define LOLA_BAR0_GSTS 0x10 +#define LOLA_BAR0_OUTSTRMPAY 0x18 +#define LOLA_BAR0_INSTRMPAY 0x1a +#define LOLA_BAR0_INTCTL 0x20 +#define LOLA_BAR0_INTSTS 0x24 +#define LOLA_BAR0_WALCLK 0x30 +#define LOLA_BAR0_SSYNC 0x38 + +#define LOLA_BAR0_CORBLBASE 0x40 +#define LOLA_BAR0_CORBUBASE 0x44 +#define LOLA_BAR0_CORBWP 0x48 /* no ULONG access */ +#define LOLA_BAR0_CORBRP 0x4a /* no ULONG access */ +#define LOLA_BAR0_CORBCTL 0x4c /* no ULONG access */ +#define LOLA_BAR0_CORBSTS 0x4d /* UCHAR access only */ +#define LOLA_BAR0_CORBSIZE 0x4e /* no ULONG access */ + +#define LOLA_BAR0_RIRBLBASE 0x50 +#define LOLA_BAR0_RIRBUBASE 0x54 +#define LOLA_BAR0_RIRBWP 0x58 +#define LOLA_BAR0_RINTCNT 0x5a /* no ULONG access */ +#define LOLA_BAR0_RIRBCTL 0x5c +#define LOLA_BAR0_RIRBSTS 0x5d /* UCHAR access only */ +#define LOLA_BAR0_RIRBSIZE 0x5e /* no ULONG access */ + +#define LOLA_BAR0_ICW 0x60 +#define LOLA_BAR0_IRR 0x64 +#define LOLA_BAR0_ICS 0x68 +#define LOLA_BAR0_DPLBASE 0x70 +#define LOLA_BAR0_DPUBASE 0x74 + +/* stream register offsets from stream base 0x80 */ +#define LOLA_BAR0_SD0_OFFSET 0x80 +#define LOLA_REG0_SD_CTL 0x00 +#define LOLA_REG0_SD_STS 0x03 +#define LOLA_REG0_SD_LPIB 0x04 +#define LOLA_REG0_SD_CBL 0x08 +#define LOLA_REG0_SD_LVI 0x0c +#define LOLA_REG0_SD_FIFOW 0x0e +#define LOLA_REG0_SD_FIFOSIZE 0x10 +#define LOLA_REG0_SD_FORMAT 0x12 +#define LOLA_REG0_SD_BDLPL 0x18 +#define LOLA_REG0_SD_BDLPU 0x1c + +/* + * Lola Digigram Registers BAR1 + */ +#define LOLA_BAR1_FPGAVER 0x00 +#define LOLA_BAR1_DEVER 0x04 +#define LOLA_BAR1_UCBMV 0x08 +#define LOLA_BAR1_JTAG 0x0c +#define LOLA_BAR1_UARTRX 0x10 +#define LOLA_BAR1_UARTTX 0x14 +#define LOLA_BAR1_UARTCR 0x18 +#define LOLA_BAR1_NVRAMVER 0x1c +#define LOLA_BAR1_CTRLSPI 0x20 +#define LOLA_BAR1_DSPI 0x24 +#define LOLA_BAR1_AISPI 0x28 +#define LOLA_BAR1_GRAN 0x2c + +#define LOLA_BAR1_DINTCTL 0x80 +#define LOLA_BAR1_DIINTCTL 0x84 +#define LOLA_BAR1_DOINTCTL 0x88 +#define LOLA_BAR1_LRC 0x90 +#define LOLA_BAR1_DINTSTS 0x94 +#define LOLA_BAR1_DIINTSTS 0x98 +#define LOLA_BAR1_DOINTSTS 0x9c + +#define LOLA_BAR1_DSD0_OFFSET 0xa0 +#define LOLA_BAR1_DSD_SIZE 0x18 + +#define LOLA_BAR1_DSDnSTS 0x00 +#define LOLA_BAR1_DSDnLPIB 0x04 +#define LOLA_BAR1_DSDnCTL 0x08 +#define LOLA_BAR1_DSDnLVI 0x0c +#define LOLA_BAR1_DSDnBDPL 0x10 +#define LOLA_BAR1_DSDnBDPU 0x14 + +#define LOLA_BAR1_SSYNC 0x03e8 + +#define LOLA_BAR1_BOARD_CTRL 0x0f00 +#define LOLA_BAR1_BOARD_MODE 0x0f02 + +#define LOLA_BAR1_SOURCE_GAIN_ENABLE 0x1000 +#define LOLA_BAR1_DEST00_MIX_GAIN_ENABLE 0x1004 +#define LOLA_BAR1_DEST31_MIX_GAIN_ENABLE 0x1080 +#define LOLA_BAR1_SOURCE00_01_GAIN 0x1084 +#define LOLA_BAR1_SOURCE30_31_GAIN 0x10c0 +#define LOLA_BAR1_SOURCE_GAIN(src) \ + (LOLA_BAR1_SOURCE00_01_GAIN + (src) * 2) +#define LOLA_BAR1_DEST00_MIX00_01_GAIN 0x10c4 +#define LOLA_BAR1_DEST00_MIX30_31_GAIN 0x1100 +#define LOLA_BAR1_DEST01_MIX00_01_GAIN 0x1104 +#define LOLA_BAR1_DEST01_MIX30_31_GAIN 0x1140 +#define LOLA_BAR1_DEST31_MIX00_01_GAIN 0x1884 +#define LOLA_BAR1_DEST31_MIX30_31_GAIN 0x18c0 +#define LOLA_BAR1_MIX_GAIN(dest, mix) \ + (LOLA_BAR1_DEST00_MIX00_01_GAIN + (dest) * 0x40 + (mix) * 2) +#define LOLA_BAR1_ANALOG_CLIP_IN 0x18c4 +#define LOLA_BAR1_PEAKMETERS_SOURCE00_01 0x18c8 +#define LOLA_BAR1_PEAKMETERS_SOURCE30_31 0x1904 +#define LOLA_BAR1_PEAKMETERS_SOURCE(src) \ + (LOLA_BAR1_PEAKMETERS_SOURCE00_01 + (src) * 2) +#define LOLA_BAR1_PEAKMETERS_DEST00_01 0x1908 +#define LOLA_BAR1_PEAKMETERS_DEST30_31 0x1944 +#define LOLA_BAR1_PEAKMETERS_DEST(dest) \ + (LOLA_BAR1_PEAKMETERS_DEST00_01 + (dest) * 2) +#define LOLA_BAR1_PEAKMETERS_AGC00_01 0x1948 +#define LOLA_BAR1_PEAKMETERS_AGC14_15 0x1964 +#define LOLA_BAR1_PEAKMETERS_AGC(x) \ + (LOLA_BAR1_PEAKMETERS_AGC00_01 + (x) * 2) + +/* GCTL reset bit */ +#define LOLA_GCTL_RESET (1 << 0) +/* GCTL unsolicited response enable bit */ +#define LOLA_GCTL_UREN (1 << 8) + +/* CORB/RIRB control, read/write pointer */ +#define LOLA_RBCTL_DMA_EN 0x02 /* enable DMA */ +#define LOLA_RBCTL_IRQ_EN 0x01 /* enable IRQ */ +#define LOLA_RBRWP_CLR 0x8000 /* read/write pointer clear */ + +#define LOLA_RIRB_EX_UNSOL_EV 0x40000000 +#define LOLA_RIRB_EX_ERROR 0x80000000 + +/* CORB int mask: CMEI[0] */ +#define LOLA_CORB_INT_CMEI 0x01 +#define LOLA_CORB_INT_MASK LOLA_CORB_INT_CMEI + +/* RIRB int mask: overrun[2], response[0] */ +#define LOLA_RIRB_INT_RESPONSE 0x01 +#define LOLA_RIRB_INT_OVERRUN 0x04 +#define LOLA_RIRB_INT_MASK (LOLA_RIRB_INT_RESPONSE | LOLA_RIRB_INT_OVERRUN) + +/* DINTCTL and DINTSTS */ +#define LOLA_DINT_GLOBAL 0x80000000 /* global interrupt enable bit */ +#define LOLA_DINT_CTRL 0x40000000 /* controller interrupt enable bit */ +#define LOLA_DINT_FIFOERR 0x20000000 /* global fifo error enable bit */ +#define LOLA_DINT_MUERR 0x10000000 /* global microcontroller underrun error */ + +/* DSDnCTL bits */ +#define LOLA_DSD_CTL_SRST 0x01 /* stream reset bit */ +#define LOLA_DSD_CTL_SRUN 0x02 /* stream DMA start bit */ +#define LOLA_DSD_CTL_IOCE 0x04 /* interrupt on completion enable */ +#define LOLA_DSD_CTL_DEIE 0x10 /* descriptor error interrupt enable */ +#define LOLA_DSD_CTL_VLRCV 0x20 /* valid LRCountValue information in bits 8..31 */ +#define LOLA_LRC_MASK 0xffffff00 + +/* DSDnSTS */ +#define LOLA_DSD_STS_BCIS 0x04 /* buffer completion interrupt status */ +#define LOLA_DSD_STS_DESE 0x10 /* descriptor error interrupt */ +#define LOLA_DSD_STS_FIFORDY 0x20 /* fifo ready */ + +#define LOLA_CORB_ENTRIES 256 + +#define MAX_STREAM_IN_COUNT 16 +#define MAX_STREAM_OUT_COUNT 16 +#define MAX_STREAM_COUNT 16 +#define MAX_PINS MAX_STREAM_COUNT +#define MAX_STREAM_BUFFER_COUNT 16 +#define MAX_AUDIO_INOUT_COUNT 16 + +#define LOLA_CLOCK_TYPE_INTERNAL 0 +#define LOLA_CLOCK_TYPE_AES 1 +#define LOLA_CLOCK_TYPE_AES_SYNC 2 +#define LOLA_CLOCK_TYPE_WORDCLOCK 3 +#define LOLA_CLOCK_TYPE_ETHERSOUND 4 +#define LOLA_CLOCK_TYPE_VIDEO 5 + +#define LOLA_CLOCK_FORMAT_NONE 0 +#define LOLA_CLOCK_FORMAT_NTSC 1 +#define LOLA_CLOCK_FORMAT_PAL 2 + +#define MAX_SAMPLE_CLOCK_COUNT 48 + +/* parameters used with mixer widget's mixer capabilities */ +#define LOLA_PEAK_METER_CAN_AGC_MASK 1 +#define LOLA_PEAK_METER_CAN_ANALOG_CLIP_MASK 2 + +struct lola_bar { + unsigned long addr; + void __iomem *remap_addr; +}; + +/* CORB/RIRB */ +struct lola_rb { + u32 *buf; /* CORB/RIRB buffer, 8 byte per each entry */ + dma_addr_t addr; /* physical address of CORB/RIRB buffer */ + unsigned short rp, wp; /* read/write pointers */ + int cmds; /* number of pending requests */ +}; + +/* Pin widget setup */ +struct lola_pin { + unsigned int nid; + bool is_analog; + unsigned int amp_mute; + unsigned int amp_step_size; + unsigned int amp_num_steps; + unsigned int amp_offset; + unsigned int max_level; + unsigned int config_default_reg; + unsigned int fixed_gain_list_len; + unsigned int cur_gain_step; +}; + +struct lola_pin_array { + unsigned int num_pins; + unsigned int num_analog_pins; + struct lola_pin pins[MAX_PINS]; +}; + +/* Clock widget setup */ +struct lola_sample_clock { + unsigned int type; + unsigned int format; + unsigned int freq; +}; + +struct lola_clock_widget { + unsigned int nid; + unsigned int items; + unsigned int cur_index; + unsigned int cur_freq; + bool cur_valid; + struct lola_sample_clock sample_clock[MAX_SAMPLE_CLOCK_COUNT]; + unsigned int idx_lookup[MAX_SAMPLE_CLOCK_COUNT]; +}; + +#define LOLA_MIXER_DIM 32 +struct lola_mixer_array { + u32 src_gain_enable; + u32 dest_mix_gain_enable[LOLA_MIXER_DIM]; + u16 src_gain[LOLA_MIXER_DIM]; + u16 dest_mix_gain[LOLA_MIXER_DIM][LOLA_MIXER_DIM]; +}; + +/* Mixer widget setup */ +struct lola_mixer_widget { + unsigned int nid; + unsigned int caps; + struct lola_mixer_array __user *array; + struct lola_mixer_array *array_saved; + unsigned int src_stream_outs; + unsigned int src_phys_ins; + unsigned int dest_stream_ins; + unsigned int dest_phys_outs; + unsigned int src_stream_out_ofs; + unsigned int dest_phys_out_ofs; + unsigned int src_mask; + unsigned int dest_mask; +}; + +/* Audio stream */ +struct lola_stream { + unsigned int nid; /* audio widget NID */ + unsigned int index; /* array index */ + unsigned int dsd; /* DSD index */ + bool can_float; + struct snd_pcm_substream *substream; /* assigned PCM substream */ + struct lola_stream *master; /* master stream (for multi-channel) */ + + /* buffer setup */ + unsigned int bufsize; + unsigned int period_bytes; + unsigned int frags; + + /* format + channel setup */ + unsigned int format_verb; + + /* flags */ + unsigned int opened:1; + unsigned int prepared:1; + unsigned int paused:1; + unsigned int running:1; +}; + +#define PLAY SNDRV_PCM_STREAM_PLAYBACK +#define CAPT SNDRV_PCM_STREAM_CAPTURE + +struct lola_pcm { + unsigned int num_streams; + struct snd_dma_buffer bdl; /* BDL buffer */ + struct lola_stream streams[MAX_STREAM_COUNT]; +}; + +/* card instance */ +struct lola { + struct snd_card *card; + struct pci_dev *pci; + + /* pci resources */ + struct lola_bar bar[2]; + int irq; + + /* locks */ + spinlock_t reg_lock; + struct mutex open_mutex; + + /* CORB/RIRB */ + struct lola_rb corb; + struct lola_rb rirb; + unsigned int res, res_ex; /* last read values */ + /* last command (for debugging) */ + unsigned int last_cmd_nid, last_verb, last_data, last_extdata; + + /* CORB/RIRB buffers */ + struct snd_dma_buffer rb; + + /* unsolicited events */ + unsigned int last_unsol_res; + + /* streams */ + struct lola_pcm pcm[2]; + + /* input src */ + unsigned int input_src_caps_mask; + unsigned int input_src_mask; + + /* pins */ + struct lola_pin_array pin[2]; + + /* clock */ + struct lola_clock_widget clock; + int ref_count_rate; + unsigned int sample_rate; + + /* mixer */ + struct lola_mixer_widget mixer; + + /* hw info */ + unsigned int version; + unsigned int lola_caps; + + /* parameters */ + unsigned int granularity; + unsigned int sample_rate_min; + unsigned int sample_rate_max; + + /* flags */ + unsigned int initialized:1; + unsigned int cold_reset:1; + unsigned int polling_mode:1; + + /* for debugging */ + unsigned int debug_res; + unsigned int debug_res_ex; +}; + +#define BAR0 0 +#define BAR1 1 + +/* Helper macros */ +#define lola_readl(chip, idx, name) \ + readl((chip)->bar[idx].remap_addr + LOLA_##idx##_##name) +#define lola_readw(chip, idx, name) \ + readw((chip)->bar[idx].remap_addr + LOLA_##idx##_##name) +#define lola_readb(chip, idx, name) \ + readb((chip)->bar[idx].remap_addr + LOLA_##idx##_##name) +#define lola_writel(chip, idx, name, val) \ + writel((val), (chip)->bar[idx].remap_addr + LOLA_##idx##_##name) +#define lola_writew(chip, idx, name, val) \ + writew((val), (chip)->bar[idx].remap_addr + LOLA_##idx##_##name) +#define lola_writeb(chip, idx, name, val) \ + writeb((val), (chip)->bar[idx].remap_addr + LOLA_##idx##_##name) + +#define lola_dsd_read(chip, dsd, name) \ + readl((chip)->bar[BAR1].remap_addr + LOLA_BAR1_DSD0_OFFSET + \ + (LOLA_BAR1_DSD_SIZE * (dsd)) + LOLA_BAR1_DSDn##name) +#define lola_dsd_write(chip, dsd, name, val) \ + writel((val), (chip)->bar[BAR1].remap_addr + LOLA_BAR1_DSD0_OFFSET + \ + (LOLA_BAR1_DSD_SIZE * (dsd)) + LOLA_BAR1_DSDn##name) + +/* GET verbs HDAudio */ +#define LOLA_VERB_GET_STREAM_FORMAT 0xa00 +#define LOLA_VERB_GET_AMP_GAIN_MUTE 0xb00 +#define LOLA_VERB_PARAMETERS 0xf00 +#define LOLA_VERB_GET_POWER_STATE 0xf05 +#define LOLA_VERB_GET_CONV 0xf06 +#define LOLA_VERB_GET_UNSOLICITED_RESPONSE 0xf08 +#define LOLA_VERB_GET_DIGI_CONVERT_1 0xf0d +#define LOLA_VERB_GET_CONFIG_DEFAULT 0xf1c +#define LOLA_VERB_GET_SUBSYSTEM_ID 0xf20 +/* GET verbs Digigram */ +#define LOLA_VERB_GET_FIXED_GAIN 0xfc0 +#define LOLA_VERB_GET_GAIN_SELECT 0xfc1 +#define LOLA_VERB_GET_MAX_LEVEL 0xfc2 +#define LOLA_VERB_GET_CLOCK_LIST 0xfc3 +#define LOLA_VERB_GET_CLOCK_SELECT 0xfc4 +#define LOLA_VERB_GET_CLOCK_STATUS 0xfc5 + +/* SET verbs HDAudio */ +#define LOLA_VERB_SET_STREAM_FORMAT 0x200 +#define LOLA_VERB_SET_AMP_GAIN_MUTE 0x300 +#define LOLA_VERB_SET_POWER_STATE 0x705 +#define LOLA_VERB_SET_CHANNEL_STREAMID 0x706 +#define LOLA_VERB_SET_UNSOLICITED_ENABLE 0x708 +#define LOLA_VERB_SET_DIGI_CONVERT_1 0x70d +/* SET verbs Digigram */ +#define LOLA_VERB_SET_GAIN_SELECT 0xf81 +#define LOLA_VERB_SET_CLOCK_SELECT 0xf84 +#define LOLA_VERB_SET_GRANULARITY_STEPS 0xf86 +#define LOLA_VERB_SET_SOURCE_GAIN 0xf87 +#define LOLA_VERB_SET_MIX_GAIN 0xf88 +#define LOLA_VERB_SET_DESTINATION_GAIN 0xf89 +#define LOLA_VERB_SET_SRC 0xf8a + +/* Parameter IDs used with LOLA_VERB_PARAMETERS */ +#define LOLA_PAR_VENDOR_ID 0x00 +#define LOLA_PAR_FUNCTION_TYPE 0x05 +#define LOLA_PAR_AUDIO_WIDGET_CAP 0x09 +#define LOLA_PAR_PCM 0x0a +#define LOLA_PAR_STREAM_FORMATS 0x0b +#define LOLA_PAR_PIN_CAP 0x0c +#define LOLA_PAR_AMP_IN_CAP 0x0d +#define LOLA_PAR_CONNLIST_LEN 0x0e +#define LOLA_PAR_POWER_STATE 0x0f +#define LOLA_PAR_GPIO_CAP 0x11 +#define LOLA_PAR_AMP_OUT_CAP 0x12 +#define LOLA_PAR_SPECIFIC_CAPS 0x80 +#define LOLA_PAR_FIXED_GAIN_LIST 0x81 + +/* extract results of LOLA_PAR_SPECIFIC_CAPS */ +#define LOLA_AFG_MIXER_WIDGET_PRESENT(res) ((res & (1 << 21)) != 0) +#define LOLA_AFG_CLOCK_WIDGET_PRESENT(res) ((res & (1 << 20)) != 0) +#define LOLA_AFG_INPUT_PIN_COUNT(res) ((res >> 10) & 0x2ff) +#define LOLA_AFG_OUTPUT_PIN_COUNT(res) ((res) & 0x2ff) + +/* extract results of LOLA_PAR_AMP_IN_CAP / LOLA_PAR_AMP_OUT_CAP */ +#define LOLA_AMP_MUTE_CAPABLE(res) ((res & (1 << 31)) != 0) +#define LOLA_AMP_STEP_SIZE(res) ((res >> 24) & 0x7f) +#define LOLA_AMP_NUM_STEPS(res) ((res >> 12) & 0x3ff) +#define LOLA_AMP_OFFSET(res) ((res) & 0x3ff) + +#define LOLA_GRANULARITY_MIN 8 +#define LOLA_GRANULARITY_MAX 32 +#define LOLA_GRANULARITY_STEP 8 + +/* parameters used with unsolicited command/response */ +#define LOLA_UNSOLICITED_TAG_MASK 0x3f +#define LOLA_UNSOLICITED_TAG 0x1a +#define LOLA_UNSOLICITED_ENABLE 0x80 +#define LOLA_UNSOL_RESP_TAG_OFFSET 26 + +/* count values in the Vendor Specific Mixer Widget's Audio Widget Capabilities */ +#define LOLA_MIXER_SRC_INPUT_PLAY_SEPARATION(res) ((res >> 2) & 0x1f) +#define LOLA_MIXER_DEST_REC_OUTPUT_SEPATATION(res) ((res >> 7) & 0x1f) + +int lola_codec_write(struct lola *chip, unsigned int nid, unsigned int verb, + unsigned int data, unsigned int extdata); +int lola_codec_read(struct lola *chip, unsigned int nid, unsigned int verb, + unsigned int data, unsigned int extdata, + unsigned int *val, unsigned int *extval); +int lola_codec_flush(struct lola *chip); +#define lola_read_param(chip, nid, param, val) \ + lola_codec_read(chip, nid, LOLA_VERB_PARAMETERS, param, 0, val, NULL) + +/* PCM */ +int lola_create_pcm(struct lola *chip); +void lola_free_pcm(struct lola *chip); +int lola_init_pcm(struct lola *chip, int dir, int *nidp); +void lola_pcm_update(struct lola *chip, struct lola_pcm *pcm, unsigned int bits); + +/* clock */ +int lola_init_clock_widget(struct lola *chip, int nid); +int lola_set_granularity(struct lola *chip, unsigned int val, bool force); +int lola_enable_clock_events(struct lola *chip); +int lola_set_clock_index(struct lola *chip, unsigned int idx); +int lola_set_clock(struct lola *chip, int idx); +int lola_set_sample_rate(struct lola *chip, int rate); +bool lola_update_ext_clock_freq(struct lola *chip, unsigned int val); +unsigned int lola_sample_rate_convert(unsigned int coded); + +/* mixer */ +int lola_init_pins(struct lola *chip, int dir, int *nidp); +int lola_init_mixer_widget(struct lola *chip, int nid); +void lola_free_mixer(struct lola *chip); +int lola_create_mixer(struct lola *chip); +int lola_setup_all_analog_gains(struct lola *chip, int dir, bool mute); +void lola_save_mixer(struct lola *chip); +void lola_restore_mixer(struct lola *chip); +int lola_set_src_config(struct lola *chip, unsigned int src_mask, bool update); + +/* proc */ +#ifdef CONFIG_SND_DEBUG +void lola_proc_debug_new(struct lola *chip); +#else +#define lola_proc_debug_new(chip) +#endif + +#endif /* _LOLA_H */ diff --git a/sound/pci/lola/lola_clock.c b/sound/pci/lola/lola_clock.c new file mode 100644 index 000000000000..72f8ef0ac865 --- /dev/null +++ b/sound/pci/lola/lola_clock.c @@ -0,0 +1,323 @@ +/* + * Support for Digigram Lola PCI-e boards + * + * Copyright (c) 2011 Takashi Iwai <tiwai@suse.de> + * + * 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/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include "lola.h" + +unsigned int lola_sample_rate_convert(unsigned int coded) +{ + unsigned int freq; + + /* base frequency */ + switch (coded & 0x3) { + case 0: freq = 48000; break; + case 1: freq = 44100; break; + case 2: freq = 32000; break; + default: return 0; /* error */ + } + + /* multiplier / devisor */ + switch (coded & 0x1c) { + case (0 << 2): break; + case (4 << 2): break; + case (1 << 2): freq *= 2; break; + case (2 << 2): freq *= 4; break; + case (5 << 2): freq /= 2; break; + case (6 << 2): freq /= 4; break; + default: return 0; /* error */ + } + + /* ajustement */ + switch (coded & 0x60) { + case (0 << 5): break; + case (1 << 5): freq = (freq * 999) / 1000; break; + case (2 << 5): freq = (freq * 1001) / 1000; break; + default: return 0; /* error */ + } + return freq; +} + +/* + * Granualrity + */ + +#define LOLA_MAXFREQ_AT_GRANULARITY_MIN 48000 +#define LOLA_MAXFREQ_AT_GRANULARITY_BELOW_MAX 96000 + +static bool check_gran_clock_compatibility(struct lola *chip, + unsigned int val, + unsigned int freq) +{ + if (!chip->granularity) + return true; + + if (val < LOLA_GRANULARITY_MIN || val > LOLA_GRANULARITY_MAX || + (val % LOLA_GRANULARITY_STEP) != 0) + return false; + + if (val == LOLA_GRANULARITY_MIN) { + if (freq > LOLA_MAXFREQ_AT_GRANULARITY_MIN) + return false; + } else if (val < LOLA_GRANULARITY_MAX) { + if (freq > LOLA_MAXFREQ_AT_GRANULARITY_BELOW_MAX) + return false; + } + return true; +} + +int lola_set_granularity(struct lola *chip, unsigned int val, bool force) +{ + int err; + + if (!force) { + if (val == chip->granularity) + return 0; +#if 0 + /* change Gran only if there are no streams allocated ! */ + if (chip->audio_in_alloc_mask || chip->audio_out_alloc_mask) + return -EBUSY; +#endif + if (!check_gran_clock_compatibility(chip, val, + chip->clock.cur_freq)) + return -EINVAL; + } + + chip->granularity = val; + val /= LOLA_GRANULARITY_STEP; + + /* audio function group */ + err = lola_codec_write(chip, 1, LOLA_VERB_SET_GRANULARITY_STEPS, + val, 0); + if (err < 0) + return err; + /* this can be a very slow function !!! */ + usleep_range(400 * val, 20000); + return lola_codec_flush(chip); +} + +/* + * Clock widget handling + */ + +int __devinit lola_init_clock_widget(struct lola *chip, int nid) +{ + unsigned int val; + int i, j, nitems, nb_verbs, idx, idx_list; + int err; + + err = lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val); + if (err < 0) { + printk(KERN_ERR SFX "Can't read wcaps for 0x%x\n", nid); + return err; + } + + if ((val & 0xfff00000) != 0x01f00000) { /* test SubType and Type */ + snd_printdd("No valid clock widget\n"); + return 0; + } + + chip->clock.nid = nid; + chip->clock.items = val & 0xff; + snd_printdd("clock_list nid=%x, entries=%d\n", nid, + chip->clock.items); + if (chip->clock.items > MAX_SAMPLE_CLOCK_COUNT) { + printk(KERN_ERR SFX "CLOCK_LIST too big: %d\n", + chip->clock.items); + return -EINVAL; + } + + nitems = chip->clock.items; + nb_verbs = (nitems + 3) / 4; + idx = 0; + idx_list = 0; + for (i = 0; i < nb_verbs; i++) { + unsigned int res_ex; + unsigned short items[4]; + + err = lola_codec_read(chip, nid, LOLA_VERB_GET_CLOCK_LIST, + idx, 0, &val, &res_ex); + if (err < 0) { + printk(KERN_ERR SFX "Can't read CLOCK_LIST\n"); + return -EINVAL; + } + + items[0] = val & 0xfff; + items[1] = (val >> 16) & 0xfff; + items[2] = res_ex & 0xfff; + items[3] = (res_ex >> 16) & 0xfff; + + for (j = 0; j < 4; j++) { + unsigned char type = items[j] >> 8; + unsigned int freq = items[j] & 0xff; + int format = LOLA_CLOCK_FORMAT_NONE; + bool add_clock = true; + if (type == LOLA_CLOCK_TYPE_INTERNAL) { + freq = lola_sample_rate_convert(freq); + if (freq < chip->sample_rate_min) + add_clock = false; + else if (freq == 48000) { + chip->clock.cur_index = idx_list; + chip->clock.cur_freq = 48000; + chip->clock.cur_valid = true; + } + } else if (type == LOLA_CLOCK_TYPE_VIDEO) { + freq = lola_sample_rate_convert(freq); + if (freq < chip->sample_rate_min) + add_clock = false; + /* video clock has a format (0:NTSC, 1:PAL)*/ + if (items[j] & 0x80) + format = LOLA_CLOCK_FORMAT_NTSC; + else + format = LOLA_CLOCK_FORMAT_PAL; + } + if (add_clock) { + struct lola_sample_clock *sc; + sc = &chip->clock.sample_clock[idx_list]; + sc->type = type; + sc->format = format; + sc->freq = freq; + /* keep the index used with the board */ + chip->clock.idx_lookup[idx_list] = idx; + idx_list++; + } else { + chip->clock.items--; + } + if (++idx >= nitems) + break; + } + } + return 0; +} + +/* enable unsolicited events of the clock widget */ +int lola_enable_clock_events(struct lola *chip) +{ + unsigned int res; + int err; + + err = lola_codec_read(chip, chip->clock.nid, + LOLA_VERB_SET_UNSOLICITED_ENABLE, + LOLA_UNSOLICITED_ENABLE | LOLA_UNSOLICITED_TAG, + 0, &res, NULL); + if (err < 0) + return err; + if (res) { + printk(KERN_WARNING SFX "error in enable_clock_events %d\n", + res); + return -EINVAL; + } + return 0; +} + +int lola_set_clock_index(struct lola *chip, unsigned int idx) +{ + unsigned int res; + int err; + + err = lola_codec_read(chip, chip->clock.nid, + LOLA_VERB_SET_CLOCK_SELECT, + chip->clock.idx_lookup[idx], + 0, &res, NULL); + if (err < 0) + return err; + if (res) { + printk(KERN_WARNING SFX "error in set_clock %d\n", res); + return -EINVAL; + } + return 0; +} + +bool lola_update_ext_clock_freq(struct lola *chip, unsigned int val) +{ + unsigned int tag; + + /* the current EXTERNAL clock information gets updated by interrupt + * with an unsolicited response + */ + if (!val) + return false; + tag = (val >> LOLA_UNSOL_RESP_TAG_OFFSET) & LOLA_UNSOLICITED_TAG_MASK; + if (tag != LOLA_UNSOLICITED_TAG) + return false; + + /* only for current = external clocks */ + if (chip->clock.sample_clock[chip->clock.cur_index].type != + LOLA_CLOCK_TYPE_INTERNAL) { + chip->clock.cur_freq = lola_sample_rate_convert(val & 0x7f); + chip->clock.cur_valid = (val & 0x100) != 0; + } + return true; +} + +int lola_set_clock(struct lola *chip, int idx) +{ + int freq = 0; + bool valid = false; + + if (idx == chip->clock.cur_index) { + /* current clock is allowed */ + freq = chip->clock.cur_freq; + valid = chip->clock.cur_valid; + } else if (chip->clock.sample_clock[idx].type == + LOLA_CLOCK_TYPE_INTERNAL) { + /* internal clocks allowed */ + freq = chip->clock.sample_clock[idx].freq; + valid = true; + } + + if (!freq || !valid) + return -EINVAL; + + if (!check_gran_clock_compatibility(chip, chip->granularity, freq)) + return -EINVAL; + + if (idx != chip->clock.cur_index) { + int err = lola_set_clock_index(chip, idx); + if (err < 0) + return err; + /* update new settings */ + chip->clock.cur_index = idx; + chip->clock.cur_freq = freq; + chip->clock.cur_valid = true; + } + return 0; +} + +int lola_set_sample_rate(struct lola *chip, int rate) +{ + int i; + + if (chip->clock.cur_freq == rate && chip->clock.cur_valid) + return 0; + /* search for new dwClockIndex */ + for (i = 0; i < chip->clock.items; i++) { + if (chip->clock.sample_clock[i].type == LOLA_CLOCK_TYPE_INTERNAL && + chip->clock.sample_clock[i].freq == rate) + break; + } + if (i >= chip->clock.items) + return -EINVAL; + return lola_set_clock(chip, i); +} + diff --git a/sound/pci/lola/lola_mixer.c b/sound/pci/lola/lola_mixer.c new file mode 100644 index 000000000000..5d518f1a712c --- /dev/null +++ b/sound/pci/lola/lola_mixer.c @@ -0,0 +1,839 @@ +/* + * Support for Digigram Lola PCI-e boards + * + * Copyright (c) 2011 Takashi Iwai <tiwai@suse.de> + * + * 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/kernel.h> +#include <linux/init.h> +#include <linux/vmalloc.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/tlv.h> +#include "lola.h" + +static int __devinit lola_init_pin(struct lola *chip, struct lola_pin *pin, + int dir, int nid) +{ + unsigned int val; + int err; + + pin->nid = nid; + err = lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val); + if (err < 0) { + printk(KERN_ERR SFX "Can't read wcaps for 0x%x\n", nid); + return err; + } + val &= 0x00f00fff; /* test TYPE and bits 0..11 */ + if (val == 0x00400200) /* Type = 4, Digital = 1 */ + pin->is_analog = false; + else if (val == 0x0040000a && dir == CAPT) /* Dig=0, InAmp/ovrd */ + pin->is_analog = true; + else if (val == 0x0040000c && dir == PLAY) /* Dig=0, OutAmp/ovrd */ + pin->is_analog = true; + else { + printk(KERN_ERR SFX "Invalid wcaps 0x%x for 0x%x\n", val, nid); + return -EINVAL; + } + + /* analog parameters only following, so continue in case of Digital pin + */ + if (!pin->is_analog) + return 0; + + if (dir == PLAY) + err = lola_read_param(chip, nid, LOLA_PAR_AMP_OUT_CAP, &val); + else + err = lola_read_param(chip, nid, LOLA_PAR_AMP_IN_CAP, &val); + if (err < 0) { + printk(KERN_ERR SFX "Can't read AMP-caps for 0x%x\n", nid); + return err; + } + + pin->amp_mute = LOLA_AMP_MUTE_CAPABLE(val); + pin->amp_step_size = LOLA_AMP_STEP_SIZE(val); + pin->amp_num_steps = LOLA_AMP_NUM_STEPS(val); + if (pin->amp_num_steps) { + /* zero as mute state */ + pin->amp_num_steps++; + pin->amp_step_size++; + } + pin->amp_offset = LOLA_AMP_OFFSET(val); + + err = lola_codec_read(chip, nid, LOLA_VERB_GET_MAX_LEVEL, 0, 0, &val, + NULL); + if (err < 0) { + printk(KERN_ERR SFX "Can't get MAX_LEVEL 0x%x\n", nid); + return err; + } + pin->max_level = val & 0x3ff; /* 10 bits */ + + pin->config_default_reg = 0; + pin->fixed_gain_list_len = 0; + pin->cur_gain_step = 0; + + return 0; +} + +int __devinit lola_init_pins(struct lola *chip, int dir, int *nidp) +{ + int i, err, nid; + nid = *nidp; + for (i = 0; i < chip->pin[dir].num_pins; i++, nid++) { + err = lola_init_pin(chip, &chip->pin[dir].pins[i], dir, nid); + if (err < 0) + return err; + if (chip->pin[dir].pins[i].is_analog) + chip->pin[dir].num_analog_pins++; + } + *nidp = nid; + return 0; +} + +void lola_free_mixer(struct lola *chip) +{ + if (chip->mixer.array_saved) + vfree(chip->mixer.array_saved); +} + +int __devinit lola_init_mixer_widget(struct lola *chip, int nid) +{ + unsigned int val; + int err; + + err = lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val); + if (err < 0) { + printk(KERN_ERR SFX "Can't read wcaps for 0x%x\n", nid); + return err; + } + + if ((val & 0xfff00000) != 0x02f00000) { /* test SubType and Type */ + snd_printdd("No valid mixer widget\n"); + return 0; + } + + chip->mixer.nid = nid; + chip->mixer.caps = val; + chip->mixer.array = (struct lola_mixer_array __iomem *) + (chip->bar[BAR1].remap_addr + LOLA_BAR1_SOURCE_GAIN_ENABLE); + + /* reserve memory to copy mixer data for sleep mode transitions */ + chip->mixer.array_saved = vmalloc(sizeof(struct lola_mixer_array)); + + /* mixer matrix sources are physical input data and play streams */ + chip->mixer.src_stream_outs = chip->pcm[PLAY].num_streams; + chip->mixer.src_phys_ins = chip->pin[CAPT].num_pins; + + /* mixer matrix destinations are record streams and physical output */ + chip->mixer.dest_stream_ins = chip->pcm[CAPT].num_streams; + chip->mixer.dest_phys_outs = chip->pin[PLAY].num_pins; + + /* mixer matrix can have unused areas between PhysIn and + * Play or Record and PhysOut zones + */ + chip->mixer.src_stream_out_ofs = chip->mixer.src_phys_ins + + LOLA_MIXER_SRC_INPUT_PLAY_SEPARATION(val); + chip->mixer.dest_phys_out_ofs = chip->mixer.dest_stream_ins + + LOLA_MIXER_DEST_REC_OUTPUT_SEPATATION(val); + + /* example : MixerMatrix of LoLa881 + * 0-------8------16-------8------16 + * | | | | | + * | INPUT | | INPUT | | + * | -> |unused | -> |unused | + * | RECORD| | OUTPUT| | + * | | | | | + * 8-------------------------------- + * | | | | | + * | | | | | + * |unused |unused |unused |unused | + * | | | | | + * | | | | | + * 16------------------------------- + * | | | | | + * | PLAY | | PLAY | | + * | -> |unused | -> |unused | + * | RECORD| | OUTPUT| | + * | | | | | + * 8-------------------------------- + * | | | | | + * | | | | | + * |unused |unused |unused |unused | + * | | | | | + * | | | | | + * 16------------------------------- + */ + if (chip->mixer.src_stream_out_ofs > MAX_AUDIO_INOUT_COUNT || + chip->mixer.dest_phys_out_ofs > MAX_STREAM_IN_COUNT) { + printk(KERN_ERR SFX "Invalid mixer widget size\n"); + return -EINVAL; + } + + chip->mixer.src_mask = ((1U << chip->mixer.src_phys_ins) - 1) | + (((1U << chip->mixer.src_stream_outs) - 1) + << chip->mixer.src_stream_out_ofs); + chip->mixer.dest_mask = ((1U << chip->mixer.dest_stream_ins) - 1) | + (((1U << chip->mixer.dest_phys_outs) - 1) + << chip->mixer.dest_phys_out_ofs); + + return 0; +} + +static int lola_mixer_set_src_gain(struct lola *chip, unsigned int id, + unsigned short gain, bool on) +{ + unsigned int oldval, val; + + if (!(chip->mixer.src_mask & (1 << id))) + return -EINVAL; + writew(gain, &chip->mixer.array->src_gain[id]); + oldval = val = readl(&chip->mixer.array->src_gain_enable); + if (on) + val |= (1 << id); + else + val &= ~(1 << id); + writel(val, &chip->mixer.array->src_gain_enable); + lola_codec_flush(chip); + /* inform micro-controller about the new source gain */ + return lola_codec_write(chip, chip->mixer.nid, + LOLA_VERB_SET_SOURCE_GAIN, id, 0); +} + +#if 0 /* not used */ +static int lola_mixer_set_src_gains(struct lola *chip, unsigned int mask, + unsigned short *gains) +{ + int i; + + if ((chip->mixer.src_mask & mask) != mask) + return -EINVAL; + for (i = 0; i < LOLA_MIXER_DIM; i++) { + if (mask & (1 << i)) { + writew(*gains, &chip->mixer.array->src_gain[i]); + gains++; + } + } + writel(mask, &chip->mixer.array->src_gain_enable); + lola_codec_flush(chip); + if (chip->mixer.caps & LOLA_PEAK_METER_CAN_AGC_MASK) { + /* update for all srcs at once */ + return lola_codec_write(chip, chip->mixer.nid, + LOLA_VERB_SET_SOURCE_GAIN, 0x80, 0); + } + /* update manually */ + for (i = 0; i < LOLA_MIXER_DIM; i++) { + if (mask & (1 << i)) { + lola_codec_write(chip, chip->mixer.nid, + LOLA_VERB_SET_SOURCE_GAIN, i, 0); + } + } + return 0; +} +#endif /* not used */ + +static int lola_mixer_set_mapping_gain(struct lola *chip, + unsigned int src, unsigned int dest, + unsigned short gain, bool on) +{ + unsigned int val; + + if (!(chip->mixer.src_mask & (1 << src)) || + !(chip->mixer.dest_mask & (1 << dest))) + return -EINVAL; + if (on) + writew(gain, &chip->mixer.array->dest_mix_gain[dest][src]); + val = readl(&chip->mixer.array->dest_mix_gain_enable[dest]); + if (on) + val |= (1 << src); + else + val &= ~(1 << src); + writel(val, &chip->mixer.array->dest_mix_gain_enable[dest]); + lola_codec_flush(chip); + return lola_codec_write(chip, chip->mixer.nid, LOLA_VERB_SET_MIX_GAIN, + src, dest); +} + +static int lola_mixer_set_dest_gains(struct lola *chip, unsigned int id, + unsigned int mask, unsigned short *gains) +{ + int i; + + if (!(chip->mixer.dest_mask & (1 << id)) || + (chip->mixer.src_mask & mask) != mask) + return -EINVAL; + for (i = 0; i < LOLA_MIXER_DIM; i++) { + if (mask & (1 << i)) { + writew(*gains, &chip->mixer.array->dest_mix_gain[id][i]); + gains++; + } + } + writel(mask, &chip->mixer.array->dest_mix_gain_enable[id]); + lola_codec_flush(chip); + /* update for all dests at once */ + return lola_codec_write(chip, chip->mixer.nid, + LOLA_VERB_SET_DESTINATION_GAIN, id, 0); +} + +/* + */ + +static int set_analog_volume(struct lola *chip, int dir, + unsigned int idx, unsigned int val, + bool external_call); + +int lola_setup_all_analog_gains(struct lola *chip, int dir, bool mute) +{ + struct lola_pin *pin; + int idx, max_idx; + + pin = chip->pin[dir].pins; + max_idx = chip->pin[dir].num_pins; + for (idx = 0; idx < max_idx; idx++) { + if (pin[idx].is_analog) { + unsigned int val = mute ? 0 : pin[idx].cur_gain_step; + /* set volume and do not save the value */ + set_analog_volume(chip, dir, idx, val, false); + } + } + return lola_codec_flush(chip); +} + +void lola_save_mixer(struct lola *chip) +{ + /* mute analog output */ + if (chip->mixer.array_saved) { + /* store contents of mixer array */ + memcpy_fromio(chip->mixer.array_saved, chip->mixer.array, + sizeof(*chip->mixer.array)); + } + lola_setup_all_analog_gains(chip, PLAY, true); /* output mute */ +} + +void lola_restore_mixer(struct lola *chip) +{ + int i; + + /*lola_reset_setups(chip);*/ + if (chip->mixer.array_saved) { + /* restore contents of mixer array */ + memcpy_toio(chip->mixer.array, chip->mixer.array_saved, + sizeof(*chip->mixer.array)); + /* inform micro-controller about all restored values + * and ignore return values + */ + for (i = 0; i < chip->mixer.src_phys_ins; i++) + lola_codec_write(chip, chip->mixer.nid, + LOLA_VERB_SET_SOURCE_GAIN, + i, 0); + for (i = 0; i < chip->mixer.src_stream_outs; i++) + lola_codec_write(chip, chip->mixer.nid, + LOLA_VERB_SET_SOURCE_GAIN, + chip->mixer.src_stream_out_ofs + i, 0); + for (i = 0; i < chip->mixer.dest_stream_ins; i++) + lola_codec_write(chip, chip->mixer.nid, + LOLA_VERB_SET_DESTINATION_GAIN, + i, 0); + for (i = 0; i < chip->mixer.dest_phys_outs; i++) + lola_codec_write(chip, chip->mixer.nid, + LOLA_VERB_SET_DESTINATION_GAIN, + chip->mixer.dest_phys_out_ofs + i, 0); + lola_codec_flush(chip); + } +} + +/* + */ + +static int set_analog_volume(struct lola *chip, int dir, + unsigned int idx, unsigned int val, + bool external_call) +{ + struct lola_pin *pin; + int err; + + if (idx >= chip->pin[dir].num_pins) + return -EINVAL; + pin = &chip->pin[dir].pins[idx]; + if (!pin->is_analog || pin->amp_num_steps <= val) + return -EINVAL; + if (external_call && pin->cur_gain_step == val) + return 0; + if (external_call) + lola_codec_flush(chip); + err = lola_codec_write(chip, pin->nid, + LOLA_VERB_SET_AMP_GAIN_MUTE, val, 0); + if (err < 0) + return err; + if (external_call) + pin->cur_gain_step = val; + return 0; +} + +int lola_set_src_config(struct lola *chip, unsigned int src_mask, bool update) +{ + int ret = 0; + int success = 0; + int n, err; + + /* SRC can be activated and the dwInputSRCMask is valid? */ + if ((chip->input_src_caps_mask & src_mask) != src_mask) + return -EINVAL; + /* handle all even Inputs - SRC is a stereo setting !!! */ + for (n = 0; n < chip->pin[CAPT].num_pins; n += 2) { + unsigned int mask = 3U << n; /* handle the stereo case */ + unsigned int new_src, src_state; + if (!(chip->input_src_caps_mask & mask)) + continue; + /* if one IO needs SRC, both stereo IO will get SRC */ + new_src = (src_mask & mask) != 0; + if (update) { + src_state = (chip->input_src_mask & mask) != 0; + if (src_state == new_src) + continue; /* nothing to change for this IO */ + } + err = lola_codec_write(chip, chip->pcm[CAPT].streams[n].nid, + LOLA_VERB_SET_SRC, new_src, 0); + if (!err) + success++; + else + ret = err; + } + if (success) + ret = lola_codec_flush(chip); + if (!ret) + chip->input_src_mask = src_mask; + return ret; +} + +/* + */ +static int init_mixer_values(struct lola *chip) +{ + int i; + + /* all src on */ + lola_set_src_config(chip, (1 << chip->pin[CAPT].num_pins) - 1, false); + + /* clear all matrix */ + memset_io(chip->mixer.array, 0, sizeof(*chip->mixer.array)); + /* set src gain to 0dB */ + for (i = 0; i < chip->mixer.src_phys_ins; i++) + lola_mixer_set_src_gain(chip, i, 336, true); /* 0dB */ + for (i = 0; i < chip->mixer.src_stream_outs; i++) + lola_mixer_set_src_gain(chip, + i + chip->mixer.src_stream_out_ofs, + 336, true); /* 0dB */ + /* set 1:1 dest gain */ + for (i = 0; i < chip->mixer.dest_stream_ins; i++) { + int src = i % chip->mixer.src_phys_ins; + lola_mixer_set_mapping_gain(chip, src, i, 336, true); + } + for (i = 0; i < chip->mixer.src_stream_outs; i++) { + int src = chip->mixer.src_stream_out_ofs + i; + int dst = chip->mixer.dest_phys_out_ofs + + i % chip->mixer.dest_phys_outs; + lola_mixer_set_mapping_gain(chip, src, dst, 336, true); + } + return 0; +} + +/* + * analog mixer control element + */ +static int lola_analog_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + int dir = kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = chip->pin[dir].num_pins; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = chip->pin[dir].pins[0].amp_num_steps; + return 0; +} + +static int lola_analog_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + int dir = kcontrol->private_value; + int i; + + for (i = 0; i < chip->pin[dir].num_pins; i++) + ucontrol->value.integer.value[i] = + chip->pin[dir].pins[i].cur_gain_step; + return 0; +} + +static int lola_analog_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + int dir = kcontrol->private_value; + int i, err; + + for (i = 0; i < chip->pin[dir].num_pins; i++) { + err = set_analog_volume(chip, dir, i, + ucontrol->value.integer.value[i], + true); + if (err < 0) + return err; + } + return 0; +} + +static int lola_analog_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + int dir = kcontrol->private_value; + unsigned int val1, val2; + struct lola_pin *pin; + + if (size < 4 * sizeof(unsigned int)) + return -ENOMEM; + pin = &chip->pin[dir].pins[0]; + + val2 = pin->amp_step_size * 25; + val1 = -1 * (int)pin->amp_offset * (int)val2; +#ifdef TLV_DB_SCALE_MUTE + val2 |= TLV_DB_SCALE_MUTE; +#endif + if (put_user(SNDRV_CTL_TLVT_DB_SCALE, tlv)) + return -EFAULT; + if (put_user(2 * sizeof(unsigned int), tlv + 1)) + return -EFAULT; + if (put_user(val1, tlv + 2)) + return -EFAULT; + if (put_user(val2, tlv + 3)) + return -EFAULT; + return 0; +} + +static struct snd_kcontrol_new lola_analog_mixer __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK), + .info = lola_analog_vol_info, + .get = lola_analog_vol_get, + .put = lola_analog_vol_put, + .tlv.c = lola_analog_vol_tlv, +}; + +static int __devinit create_analog_mixer(struct lola *chip, int dir, char *name) +{ + if (!chip->pin[dir].num_pins) + return 0; + /* no analog volumes on digital only adapters */ + if (chip->pin[dir].num_pins != chip->pin[dir].num_analog_pins) + return 0; + lola_analog_mixer.name = name; + lola_analog_mixer.private_value = dir; + return snd_ctl_add(chip->card, + snd_ctl_new1(&lola_analog_mixer, chip)); +} + +/* + * Hardware sample rate converter on digital input + */ +static int lola_input_src_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = chip->pin[CAPT].num_pins; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int lola_input_src_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + int i; + + for (i = 0; i < chip->pin[CAPT].num_pins; i++) + ucontrol->value.integer.value[i] = + !!(chip->input_src_mask & (1 << i)); + return 0; +} + +static int lola_input_src_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + int i; + unsigned int mask; + + mask = 0; + for (i = 0; i < chip->pin[CAPT].num_pins; i++) + if (ucontrol->value.integer.value[i]) + mask |= 1 << i; + return lola_set_src_config(chip, mask, true); +} + +static struct snd_kcontrol_new lola_input_src_mixer __devinitdata = { + .name = "Digital SRC Capture Switch", + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = lola_input_src_info, + .get = lola_input_src_get, + .put = lola_input_src_put, +}; + +/* + * Lola16161 or Lola881 can have Hardware sample rate converters + * on its digital input pins + */ +static int __devinit create_input_src_mixer(struct lola *chip) +{ + if (!chip->input_src_caps_mask) + return 0; + + return snd_ctl_add(chip->card, + snd_ctl_new1(&lola_input_src_mixer, chip)); +} + +/* + * src gain mixer + */ +static int lola_src_gain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + unsigned int count = (kcontrol->private_value >> 8) & 0xff; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = count; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 409; + return 0; +} + +static int lola_src_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + unsigned int ofs = kcontrol->private_value & 0xff; + unsigned int count = (kcontrol->private_value >> 8) & 0xff; + unsigned int mask, i; + + mask = readl(&chip->mixer.array->src_gain_enable); + for (i = 0; i < count; i++) { + unsigned int idx = ofs + i; + unsigned short val; + if (!(chip->mixer.src_mask & (1 << idx))) + return -EINVAL; + if (mask & (1 << idx)) + val = readw(&chip->mixer.array->src_gain[idx]) + 1; + else + val = 0; + ucontrol->value.integer.value[i] = val; + } + return 0; +} + +static int lola_src_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + unsigned int ofs = kcontrol->private_value & 0xff; + unsigned int count = (kcontrol->private_value >> 8) & 0xff; + int i, err; + + for (i = 0; i < count; i++) { + unsigned int idx = ofs + i; + unsigned short val = ucontrol->value.integer.value[i]; + if (val) + val--; + err = lola_mixer_set_src_gain(chip, idx, val, !!val); + if (err < 0) + return err; + } + return 0; +} + +/* raw value: 0 = -84dB, 336 = 0dB, 408=18dB, incremented 1 for mute */ +static const DECLARE_TLV_DB_SCALE(lola_src_gain_tlv, -8425, 25, 1); + +static struct snd_kcontrol_new lola_src_gain_mixer __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .info = lola_src_gain_info, + .get = lola_src_gain_get, + .put = lola_src_gain_put, + .tlv.p = lola_src_gain_tlv, +}; + +static int __devinit create_src_gain_mixer(struct lola *chip, + int num, int ofs, char *name) +{ + lola_src_gain_mixer.name = name; + lola_src_gain_mixer.private_value = ofs + (num << 8); + return snd_ctl_add(chip->card, + snd_ctl_new1(&lola_src_gain_mixer, chip)); +} + +/* + * destination gain (matrix-like) mixer + */ +static int lola_dest_gain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + unsigned int src_num = (kcontrol->private_value >> 8) & 0xff; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = src_num; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 433; + return 0; +} + +static int lola_dest_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + unsigned int src_ofs = kcontrol->private_value & 0xff; + unsigned int src_num = (kcontrol->private_value >> 8) & 0xff; + unsigned int dst_ofs = (kcontrol->private_value >> 16) & 0xff; + unsigned int dst, mask, i; + + dst = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + dst_ofs; + mask = readl(&chip->mixer.array->dest_mix_gain_enable[dst]); + for (i = 0; i < src_num; i++) { + unsigned int src = src_ofs + i; + unsigned short val; + if (!(chip->mixer.src_mask & (1 << src))) + return -EINVAL; + if (mask & (1 << dst)) + val = readw(&chip->mixer.array->dest_mix_gain[dst][src]) + 1; + else + val = 0; + ucontrol->value.integer.value[i] = val; + } + return 0; +} + +static int lola_dest_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + unsigned int src_ofs = kcontrol->private_value & 0xff; + unsigned int src_num = (kcontrol->private_value >> 8) & 0xff; + unsigned int dst_ofs = (kcontrol->private_value >> 16) & 0xff; + unsigned int dst, mask; + unsigned short gains[MAX_STREAM_COUNT]; + int i, num; + + mask = 0; + num = 0; + for (i = 0; i < src_num; i++) { + unsigned short val = ucontrol->value.integer.value[i]; + if (val) { + gains[num++] = val - 1; + mask |= 1 << i; + } + } + mask <<= src_ofs; + dst = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + dst_ofs; + return lola_mixer_set_dest_gains(chip, dst, mask, gains); +} + +static const DECLARE_TLV_DB_SCALE(lola_dest_gain_tlv, -8425, 25, 1); + +static struct snd_kcontrol_new lola_dest_gain_mixer __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .info = lola_dest_gain_info, + .get = lola_dest_gain_get, + .put = lola_dest_gain_put, + .tlv.p = lola_dest_gain_tlv, +}; + +static int __devinit create_dest_gain_mixer(struct lola *chip, + int src_num, int src_ofs, + int num, int ofs, char *name) +{ + lola_dest_gain_mixer.count = num; + lola_dest_gain_mixer.name = name; + lola_dest_gain_mixer.private_value = + src_ofs + (src_num << 8) + (ofs << 16) + (num << 24); + return snd_ctl_add(chip->card, + snd_ctl_new1(&lola_dest_gain_mixer, chip)); +} + +/* + */ +int __devinit lola_create_mixer(struct lola *chip) +{ + int err; + + err = create_analog_mixer(chip, PLAY, "Analog Playback Volume"); + if (err < 0) + return err; + err = create_analog_mixer(chip, CAPT, "Analog Capture Volume"); + if (err < 0) + return err; + err = create_input_src_mixer(chip); + if (err < 0) + return err; + err = create_src_gain_mixer(chip, chip->mixer.src_phys_ins, 0, + "Line Source Gain Volume"); + if (err < 0) + return err; + err = create_src_gain_mixer(chip, chip->mixer.src_stream_outs, + chip->mixer.src_stream_out_ofs, + "Stream Source Gain Volume"); + if (err < 0) + return err; + err = create_dest_gain_mixer(chip, + chip->mixer.src_phys_ins, 0, + chip->mixer.dest_stream_ins, 0, + "Line Capture Volume"); + if (err < 0) + return err; + err = create_dest_gain_mixer(chip, + chip->mixer.src_stream_outs, + chip->mixer.src_stream_out_ofs, + chip->mixer.dest_stream_ins, 0, + "Stream-Loopback Capture Volume"); + if (err < 0) + return err; + err = create_dest_gain_mixer(chip, + chip->mixer.src_phys_ins, 0, + chip->mixer.dest_phys_outs, + chip->mixer.dest_phys_out_ofs, + "Line-Loopback Playback Volume"); + if (err < 0) + return err; + err = create_dest_gain_mixer(chip, + chip->mixer.src_stream_outs, + chip->mixer.src_stream_out_ofs, + chip->mixer.dest_phys_outs, + chip->mixer.dest_phys_out_ofs, + "Stream Playback Volume"); + if (err < 0) + return err; + + return init_mixer_values(chip); +} diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c new file mode 100644 index 000000000000..c44db68eecb5 --- /dev/null +++ b/sound/pci/lola/lola_pcm.c @@ -0,0 +1,706 @@ +/* + * Support for Digigram Lola PCI-e boards + * + * Copyright (c) 2011 Takashi Iwai <tiwai@suse.de> + * + * 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/kernel.h> +#include <linux/init.h> +#include <linux/dma-mapping.h> +#include <linux/pci.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include "lola.h" + +#define LOLA_MAX_BDL_ENTRIES 8 +#define LOLA_MAX_BUF_SIZE (1024*1024*1024) +#define LOLA_BDL_ENTRY_SIZE (16 * 16) + +static struct lola_pcm *lola_get_pcm(struct snd_pcm_substream *substream) +{ + struct lola *chip = snd_pcm_substream_chip(substream); + return &chip->pcm[substream->stream]; +} + +static struct lola_stream *lola_get_stream(struct snd_pcm_substream *substream) +{ + struct lola_pcm *pcm = lola_get_pcm(substream); + unsigned int idx = substream->number; + return &pcm->streams[idx]; +} + +static unsigned int lola_get_lrc(struct lola *chip) +{ + return lola_readl(chip, BAR1, LRC); +} + +static unsigned int lola_get_tstamp(struct lola *chip, bool quick_no_sync) +{ + unsigned int tstamp = lola_get_lrc(chip) >> 8; + if (chip->granularity) { + unsigned int wait_banks = quick_no_sync ? 0 : 8; + tstamp += (wait_banks + 1) * chip->granularity - 1; + tstamp -= tstamp % chip->granularity; + } + return tstamp << 8; +} + +/* clear any pending interrupt status */ +static void lola_stream_clear_pending_irq(struct lola *chip, + struct lola_stream *str) +{ + unsigned int val = lola_dsd_read(chip, str->dsd, STS); + val &= LOLA_DSD_STS_DESE | LOLA_DSD_STS_BCIS; + if (val) + lola_dsd_write(chip, str->dsd, STS, val); +} + +static void lola_stream_start(struct lola *chip, struct lola_stream *str, + unsigned int tstamp) +{ + lola_stream_clear_pending_irq(chip, str); + lola_dsd_write(chip, str->dsd, CTL, + LOLA_DSD_CTL_SRUN | + LOLA_DSD_CTL_IOCE | + LOLA_DSD_CTL_DEIE | + LOLA_DSD_CTL_VLRCV | + tstamp); +} + +static void lola_stream_stop(struct lola *chip, struct lola_stream *str, + unsigned int tstamp) +{ + lola_dsd_write(chip, str->dsd, CTL, + LOLA_DSD_CTL_IOCE | + LOLA_DSD_CTL_DEIE | + LOLA_DSD_CTL_VLRCV | + tstamp); + lola_stream_clear_pending_irq(chip, str); +} + +static void wait_for_srst_clear(struct lola *chip, struct lola_stream *str) +{ + unsigned long end_time = jiffies + msecs_to_jiffies(200); + while (time_before(jiffies, end_time)) { + unsigned int val; + val = lola_dsd_read(chip, str->dsd, CTL); + if (!(val & LOLA_DSD_CTL_SRST)) + return; + msleep(1); + } + printk(KERN_WARNING SFX "SRST not clear (stream %d)\n", str->dsd); +} + +static int lola_stream_wait_for_fifo(struct lola *chip, + struct lola_stream *str, + bool ready) +{ + unsigned int val = ready ? LOLA_DSD_STS_FIFORDY : 0; + unsigned long end_time = jiffies + msecs_to_jiffies(200); + while (time_before(jiffies, end_time)) { + unsigned int reg = lola_dsd_read(chip, str->dsd, STS); + if ((reg & LOLA_DSD_STS_FIFORDY) == val) + return 0; + msleep(1); + } + printk(KERN_WARNING SFX "FIFO not ready (stream %d)\n", str->dsd); + return -EIO; +} + +/* sync for FIFO ready/empty for all linked streams; + * clear paused flag when FIFO gets ready again + */ +static int lola_sync_wait_for_fifo(struct lola *chip, + struct snd_pcm_substream *substream, + bool ready) +{ + unsigned int val = ready ? LOLA_DSD_STS_FIFORDY : 0; + unsigned long end_time = jiffies + msecs_to_jiffies(200); + struct snd_pcm_substream *s; + int pending = 0; + + while (time_before(jiffies, end_time)) { + pending = 0; + snd_pcm_group_for_each_entry(s, substream) { + struct lola_stream *str; + if (s->pcm->card != substream->pcm->card) + continue; + str = lola_get_stream(s); + if (str->prepared && str->paused) { + unsigned int reg; + reg = lola_dsd_read(chip, str->dsd, STS); + if ((reg & LOLA_DSD_STS_FIFORDY) != val) { + pending = str->dsd + 1; + break; + } + if (ready) + str->paused = 0; + } + } + if (!pending) + return 0; + msleep(1); + } + printk(KERN_WARNING SFX "FIFO not ready (pending %d)\n", pending - 1); + return -EIO; +} + +/* finish pause - prepare for a new resume */ +static void lola_sync_pause(struct lola *chip, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_substream *s; + + lola_sync_wait_for_fifo(chip, substream, false); + snd_pcm_group_for_each_entry(s, substream) { + struct lola_stream *str; + if (s->pcm->card != substream->pcm->card) + continue; + str = lola_get_stream(s); + if (str->paused && str->prepared) + lola_dsd_write(chip, str->dsd, CTL, LOLA_DSD_CTL_SRUN | + LOLA_DSD_CTL_IOCE | LOLA_DSD_CTL_DEIE); + } + lola_sync_wait_for_fifo(chip, substream, true); +} + +static void lola_stream_reset(struct lola *chip, struct lola_stream *str) +{ + if (str->prepared) { + if (str->paused) + lola_sync_pause(chip, str->substream); + str->prepared = 0; + lola_dsd_write(chip, str->dsd, CTL, + LOLA_DSD_CTL_IOCE | LOLA_DSD_CTL_DEIE); + lola_stream_wait_for_fifo(chip, str, false); + lola_stream_clear_pending_irq(chip, str); + lola_dsd_write(chip, str->dsd, CTL, LOLA_DSD_CTL_SRST); + lola_dsd_write(chip, str->dsd, LVI, 0); + lola_dsd_write(chip, str->dsd, BDPU, 0); + lola_dsd_write(chip, str->dsd, BDPL, 0); + wait_for_srst_clear(chip, str); + } +} + +static struct snd_pcm_hardware lola_pcm_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_FLOAT_LE), + .rates = SNDRV_PCM_RATE_8000_192000, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = LOLA_MAX_BUF_SIZE, + .period_bytes_min = 128, + .period_bytes_max = LOLA_MAX_BUF_SIZE / 2, + .periods_min = 2, + .periods_max = LOLA_MAX_BDL_ENTRIES, + .fifo_size = 0, +}; + +static int lola_pcm_open(struct snd_pcm_substream *substream) +{ + struct lola *chip = snd_pcm_substream_chip(substream); + struct lola_pcm *pcm = lola_get_pcm(substream); + struct lola_stream *str = lola_get_stream(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + mutex_lock(&chip->open_mutex); + if (str->opened) { + mutex_unlock(&chip->open_mutex); + return -EBUSY; + } + str->substream = substream; + str->master = NULL; + str->opened = 1; + runtime->hw = lola_pcm_hw; + runtime->hw.channels_max = pcm->num_streams - str->index; + if (chip->sample_rate) { + /* sample rate is locked */ + runtime->hw.rate_min = chip->sample_rate; + runtime->hw.rate_max = chip->sample_rate; + } else { + runtime->hw.rate_min = chip->sample_rate_min; + runtime->hw.rate_max = chip->sample_rate_max; + } + chip->ref_count_rate++; + snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + /* period size = multiple of chip->granularity (8, 16 or 32 frames)*/ + snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + chip->granularity); + snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + chip->granularity); + mutex_unlock(&chip->open_mutex); + return 0; +} + +static void lola_cleanup_slave_streams(struct lola_pcm *pcm, + struct lola_stream *str) +{ + int i; + for (i = str->index + 1; i < pcm->num_streams; i++) { + struct lola_stream *s = &pcm->streams[i]; + if (s->master != str) + break; + s->master = NULL; + s->opened = 0; + } +} + +static int lola_pcm_close(struct snd_pcm_substream *substream) +{ + struct lola *chip = snd_pcm_substream_chip(substream); + struct lola_stream *str = lola_get_stream(substream); + + mutex_lock(&chip->open_mutex); + if (str->substream == substream) { + str->substream = NULL; + str->opened = 0; + } + if (--chip->ref_count_rate == 0) { + /* release sample rate */ + chip->sample_rate = 0; + } + mutex_unlock(&chip->open_mutex); + return 0; +} + +static int lola_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct lola_stream *str = lola_get_stream(substream); + + str->bufsize = 0; + str->period_bytes = 0; + str->format_verb = 0; + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int lola_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct lola *chip = snd_pcm_substream_chip(substream); + struct lola_pcm *pcm = lola_get_pcm(substream); + struct lola_stream *str = lola_get_stream(substream); + + mutex_lock(&chip->open_mutex); + lola_stream_reset(chip, str); + lola_cleanup_slave_streams(pcm, str); + mutex_unlock(&chip->open_mutex); + return snd_pcm_lib_free_pages(substream); +} + +/* + * set up a BDL entry + */ +static int setup_bdle(struct snd_pcm_substream *substream, + struct lola_stream *str, u32 **bdlp, + int ofs, int size) +{ + u32 *bdl = *bdlp; + + while (size > 0) { + dma_addr_t addr; + int chunk; + + if (str->frags >= LOLA_MAX_BDL_ENTRIES) + return -EINVAL; + + addr = snd_pcm_sgbuf_get_addr(substream, ofs); + /* program the address field of the BDL entry */ + bdl[0] = cpu_to_le32((u32)addr); + bdl[1] = cpu_to_le32(upper_32_bits(addr)); + /* program the size field of the BDL entry */ + chunk = snd_pcm_sgbuf_get_chunk_size(substream, ofs, size); + bdl[2] = cpu_to_le32(chunk); + /* program the IOC to enable interrupt + * only when the whole fragment is processed + */ + size -= chunk; + bdl[3] = size ? 0 : cpu_to_le32(0x01); + bdl += 4; + str->frags++; + ofs += chunk; + } + *bdlp = bdl; + return ofs; +} + +/* + * set up BDL entries + */ +static int lola_setup_periods(struct lola *chip, struct lola_pcm *pcm, + struct snd_pcm_substream *substream, + struct lola_stream *str) +{ + u32 *bdl; + int i, ofs, periods, period_bytes; + + period_bytes = str->period_bytes; + periods = str->bufsize / period_bytes; + + /* program the initial BDL entries */ + bdl = (u32 *)(pcm->bdl.area + LOLA_BDL_ENTRY_SIZE * str->index); + ofs = 0; + str->frags = 0; + for (i = 0; i < periods; i++) { + ofs = setup_bdle(substream, str, &bdl, ofs, period_bytes); + if (ofs < 0) + goto error; + } + return 0; + + error: + snd_printk(KERN_ERR SFX "Too many BDL entries: buffer=%d, period=%d\n", + str->bufsize, period_bytes); + return -EINVAL; +} + +static unsigned int lola_get_format_verb(struct snd_pcm_substream *substream) +{ + unsigned int verb; + + switch (substream->runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + verb = 0x00000000; + break; + case SNDRV_PCM_FORMAT_S24_LE: + verb = 0x00000200; + break; + case SNDRV_PCM_FORMAT_S32_LE: + verb = 0x00000300; + break; + case SNDRV_PCM_FORMAT_FLOAT_LE: + verb = 0x00001300; + break; + default: + return 0; + } + verb |= substream->runtime->channels; + return verb; +} + +static int lola_set_stream_config(struct lola *chip, + struct lola_stream *str, + int channels) +{ + int i, err; + unsigned int verb, val; + + /* set format info for all channels + * (with only one command for the first channel) + */ + err = lola_codec_read(chip, str->nid, LOLA_VERB_SET_STREAM_FORMAT, + str->format_verb, 0, &val, NULL); + if (err < 0) { + printk(KERN_ERR SFX "Cannot set stream format 0x%x\n", + str->format_verb); + return err; + } + + /* update stream - channel config */ + for (i = 0; i < channels; i++) { + verb = (str->index << 6) | i; + err = lola_codec_read(chip, str[i].nid, + LOLA_VERB_SET_CHANNEL_STREAMID, 0, verb, + &val, NULL); + if (err < 0) { + printk(KERN_ERR SFX "Cannot set stream channel %d\n", i); + return err; + } + } + return 0; +} + +/* + * set up the SD for streaming + */ +static int lola_setup_controller(struct lola *chip, struct lola_pcm *pcm, + struct lola_stream *str) +{ + dma_addr_t bdl; + + if (str->prepared) + return -EINVAL; + + /* set up BDL */ + bdl = pcm->bdl.addr + LOLA_BDL_ENTRY_SIZE * str->index; + lola_dsd_write(chip, str->dsd, BDPL, (u32)bdl); + lola_dsd_write(chip, str->dsd, BDPU, upper_32_bits(bdl)); + /* program the stream LVI (last valid index) of the BDL */ + lola_dsd_write(chip, str->dsd, LVI, str->frags - 1); + lola_stream_clear_pending_irq(chip, str); + + lola_dsd_write(chip, str->dsd, CTL, + LOLA_DSD_CTL_IOCE | LOLA_DSD_CTL_DEIE | LOLA_DSD_CTL_SRUN); + + str->prepared = 1; + + return lola_stream_wait_for_fifo(chip, str, true); +} + +static int lola_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct lola *chip = snd_pcm_substream_chip(substream); + struct lola_pcm *pcm = lola_get_pcm(substream); + struct lola_stream *str = lola_get_stream(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int bufsize, period_bytes, format_verb; + int i, err; + + mutex_lock(&chip->open_mutex); + lola_stream_reset(chip, str); + lola_cleanup_slave_streams(pcm, str); + if (str->index + runtime->channels > pcm->num_streams) { + mutex_unlock(&chip->open_mutex); + return -EINVAL; + } + for (i = 1; i < runtime->channels; i++) { + str[i].master = str; + str[i].opened = 1; + } + mutex_unlock(&chip->open_mutex); + + bufsize = snd_pcm_lib_buffer_bytes(substream); + period_bytes = snd_pcm_lib_period_bytes(substream); + format_verb = lola_get_format_verb(substream); + + str->bufsize = bufsize; + str->period_bytes = period_bytes; + str->format_verb = format_verb; + + err = lola_setup_periods(chip, pcm, substream, str); + if (err < 0) + return err; + + err = lola_set_sample_rate(chip, runtime->rate); + if (err < 0) + return err; + chip->sample_rate = runtime->rate; /* sample rate gets locked */ + + err = lola_set_stream_config(chip, str, runtime->channels); + if (err < 0) + return err; + + err = lola_setup_controller(chip, pcm, str); + if (err < 0) { + lola_stream_reset(chip, str); + return err; + } + + return 0; +} + +static int lola_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct lola *chip = snd_pcm_substream_chip(substream); + struct lola_stream *str; + struct snd_pcm_substream *s; + unsigned int start; + unsigned int tstamp; + bool sync_streams; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + start = 1; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + start = 0; + break; + default: + return -EINVAL; + } + + /* + * sample correct synchronization is only needed starting several + * streams. On stop or if only one stream do as quick as possible + */ + sync_streams = (start && snd_pcm_stream_linked(substream)); + tstamp = lola_get_tstamp(chip, !sync_streams); + spin_lock(&chip->reg_lock); + snd_pcm_group_for_each_entry(s, substream) { + if (s->pcm->card != substream->pcm->card) + continue; + str = lola_get_stream(s); + if (start) + lola_stream_start(chip, str, tstamp); + else + lola_stream_stop(chip, str, tstamp); + str->running = start; + str->paused = !start; + snd_pcm_trigger_done(s, substream); + } + spin_unlock(&chip->reg_lock); + return 0; +} + +static snd_pcm_uframes_t lola_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct lola *chip = snd_pcm_substream_chip(substream); + struct lola_stream *str = lola_get_stream(substream); + unsigned int pos = lola_dsd_read(chip, str->dsd, LPIB); + + if (pos >= str->bufsize) + pos = 0; + return bytes_to_frames(substream->runtime, pos); +} + +void lola_pcm_update(struct lola *chip, struct lola_pcm *pcm, unsigned int bits) +{ + int i; + + for (i = 0; bits && i < pcm->num_streams; i++) { + if (bits & (1 << i)) { + struct lola_stream *str = &pcm->streams[i]; + if (str->substream && str->running) + snd_pcm_period_elapsed(str->substream); + bits &= ~(1 << i); + } + } +} + +static struct snd_pcm_ops lola_pcm_ops = { + .open = lola_pcm_open, + .close = lola_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = lola_pcm_hw_params, + .hw_free = lola_pcm_hw_free, + .prepare = lola_pcm_prepare, + .trigger = lola_pcm_trigger, + .pointer = lola_pcm_pointer, + .page = snd_pcm_sgbuf_ops_page, +}; + +int __devinit lola_create_pcm(struct lola *chip) +{ + struct snd_pcm *pcm; + int i, err; + + for (i = 0; i < 2; i++) { + err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + PAGE_SIZE, &chip->pcm[i].bdl); + if (err < 0) + return err; + } + + err = snd_pcm_new(chip->card, "Digigram Lola", 0, + chip->pcm[SNDRV_PCM_STREAM_PLAYBACK].num_streams, + chip->pcm[SNDRV_PCM_STREAM_CAPTURE].num_streams, + &pcm); + if (err < 0) + return err; + strlcpy(pcm->name, "Digigram Lola", sizeof(pcm->name)); + pcm->private_data = chip; + for (i = 0; i < 2; i++) { + if (chip->pcm[i].num_streams) + snd_pcm_set_ops(pcm, i, &lola_pcm_ops); + } + /* buffer pre-allocation */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, + snd_dma_pci_data(chip->pci), + 1024 * 64, 32 * 1024 * 1024); + return 0; +} + +void lola_free_pcm(struct lola *chip) +{ + snd_dma_free_pages(&chip->pcm[0].bdl); + snd_dma_free_pages(&chip->pcm[1].bdl); +} + +/* + */ + +static int lola_init_stream(struct lola *chip, struct lola_stream *str, + int idx, int nid, int dir) +{ + unsigned int val; + int err; + + str->nid = nid; + str->index = idx; + str->dsd = idx; + if (dir == PLAY) + str->dsd += MAX_STREAM_IN_COUNT; + err = lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val); + if (err < 0) { + printk(KERN_ERR SFX "Can't read wcaps for 0x%x\n", nid); + return err; + } + if (dir == PLAY) { + /* test TYPE and bits 0..11 (no test bit9 : Digital = 0/1) */ + if ((val & 0x00f00dff) != 0x00000010) { + printk(KERN_ERR SFX "Invalid wcaps 0x%x for 0x%x\n", + val, nid); + return -EINVAL; + } + } else { + /* test TYPE and bits 0..11 (no test bit9 : Digital = 0/1) + * (bug : ignore bit8: Conn list = 0/1) + */ + if ((val & 0x00f00cff) != 0x00100010) { + printk(KERN_ERR SFX "Invalid wcaps 0x%x for 0x%x\n", + val, nid); + return -EINVAL; + } + /* test bit9:DIGITAL and bit12:SRC_PRESENT*/ + if ((val & 0x00001200) == 0x00001200) + chip->input_src_caps_mask |= (1 << idx); + } + + err = lola_read_param(chip, nid, LOLA_PAR_STREAM_FORMATS, &val); + if (err < 0) { + printk(KERN_ERR SFX "Can't read FORMATS 0x%x\n", nid); + return err; + } + val &= 3; + if (val == 3) + str->can_float = true; + if (!(val & 1)) { + printk(KERN_ERR SFX "Invalid formats 0x%x for 0x%x", val, nid); + return -EINVAL; + } + return 0; +} + +int __devinit lola_init_pcm(struct lola *chip, int dir, int *nidp) +{ + struct lola_pcm *pcm = &chip->pcm[dir]; + int i, nid, err; + + nid = *nidp; + for (i = 0; i < pcm->num_streams; i++, nid++) { + err = lola_init_stream(chip, &pcm->streams[i], i, nid, dir); + if (err < 0) + return err; + } + *nidp = nid; + return 0; +} diff --git a/sound/pci/lola/lola_proc.c b/sound/pci/lola/lola_proc.c new file mode 100644 index 000000000000..9d7daf897c9d --- /dev/null +++ b/sound/pci/lola/lola_proc.c @@ -0,0 +1,222 @@ +/* + * Support for Digigram Lola PCI-e boards + * + * Copyright (c) 2011 Takashi Iwai <tiwai@suse.de> + * + * 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/kernel.h> +#include <linux/init.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/info.h> +#include <sound/pcm.h> +#include "lola.h" + +static void print_audio_widget(struct snd_info_buffer *buffer, + struct lola *chip, int nid, const char *name) +{ + unsigned int val; + + lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val); + snd_iprintf(buffer, "Node 0x%02x %s wcaps 0x%x\n", nid, name, val); + lola_read_param(chip, nid, LOLA_PAR_STREAM_FORMATS, &val); + snd_iprintf(buffer, " Formats: 0x%x\n", val); +} + +static void print_pin_widget(struct snd_info_buffer *buffer, + struct lola *chip, int nid, unsigned int ampcap, + const char *name) +{ + unsigned int val; + + lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val); + snd_iprintf(buffer, "Node 0x%02x %s wcaps 0x%x\n", nid, name, val); + if (val == 0x00400200) + return; + lola_read_param(chip, nid, ampcap, &val); + snd_iprintf(buffer, " Amp-Caps: 0x%x\n", val); + snd_iprintf(buffer, " mute=%d, step-size=%d, steps=%d, ofs=%d\n", + LOLA_AMP_MUTE_CAPABLE(val), + LOLA_AMP_STEP_SIZE(val), + LOLA_AMP_NUM_STEPS(val), + LOLA_AMP_OFFSET(val)); + lola_codec_read(chip, nid, LOLA_VERB_GET_MAX_LEVEL, 0, 0, &val, NULL); + snd_iprintf(buffer, " Max-level: 0x%x\n", val); +} + +static void print_clock_widget(struct snd_info_buffer *buffer, + struct lola *chip, int nid) +{ + int i, j, num_clocks; + unsigned int val; + + lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val); + snd_iprintf(buffer, "Node 0x%02x [Clock] wcaps 0x%x\n", nid, val); + num_clocks = val & 0xff; + for (i = 0; i < num_clocks; i += 4) { + unsigned int res_ex; + unsigned short items[4]; + const char *name; + + lola_codec_read(chip, nid, LOLA_VERB_GET_CLOCK_LIST, + i, 0, &val, &res_ex); + items[0] = val & 0xfff; + items[1] = (val >> 16) & 0xfff; + items[2] = res_ex & 0xfff; + items[3] = (res_ex >> 16) & 0xfff; + for (j = 0; j < 4; j++) { + unsigned char type = items[j] >> 8; + unsigned int freq = items[j] & 0xff; + if (i + j >= num_clocks) + break; + if (type == LOLA_CLOCK_TYPE_INTERNAL) { + name = "Internal"; + freq = lola_sample_rate_convert(freq); + } else if (type == LOLA_CLOCK_TYPE_VIDEO) { + name = "Video"; + freq = lola_sample_rate_convert(freq); + } else { + name = "Other"; + } + snd_iprintf(buffer, " Clock %d: Type %d:%s, freq=%d\n", + i + j, type, name, freq); + } + } +} + +static void print_mixer_widget(struct snd_info_buffer *buffer, + struct lola *chip, int nid) +{ + unsigned int val; + + lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val); + snd_iprintf(buffer, "Node 0x%02x [Mixer] wcaps 0x%x\n", nid, val); +} + +static void lola_proc_codec_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct lola *chip = entry->private_data; + unsigned int val; + int i, nid; + + lola_read_param(chip, 0, LOLA_PAR_VENDOR_ID, &val); + snd_iprintf(buffer, "Vendor: 0x%08x\n", val); + lola_read_param(chip, 1, LOLA_PAR_FUNCTION_TYPE, &val); + snd_iprintf(buffer, "Function Type: %d\n", val); + lola_read_param(chip, 1, LOLA_PAR_SPECIFIC_CAPS, &val); + snd_iprintf(buffer, "Specific-Caps: 0x%08x\n", val); + snd_iprintf(buffer, " Pins-In %d, Pins-Out %d\n", + chip->pin[CAPT].num_pins, chip->pin[PLAY].num_pins); + nid = 2; + for (i = 0; i < chip->pcm[CAPT].num_streams; i++, nid++) + print_audio_widget(buffer, chip, nid, "[Audio-In]"); + for (i = 0; i < chip->pcm[PLAY].num_streams; i++, nid++) + print_audio_widget(buffer, chip, nid, "[Audio-Out]"); + for (i = 0; i < chip->pin[CAPT].num_pins; i++, nid++) + print_pin_widget(buffer, chip, nid, LOLA_PAR_AMP_IN_CAP, + "[Pin-In]"); + for (i = 0; i < chip->pin[PLAY].num_pins; i++, nid++) + print_pin_widget(buffer, chip, nid, LOLA_PAR_AMP_OUT_CAP, + "[Pin-Out]"); + if (LOLA_AFG_CLOCK_WIDGET_PRESENT(chip->lola_caps)) { + print_clock_widget(buffer, chip, nid); + nid++; + } + if (LOLA_AFG_MIXER_WIDGET_PRESENT(chip->lola_caps)) { + print_mixer_widget(buffer, chip, nid); + nid++; + } +} + +/* direct codec access for debugging */ +static void lola_proc_codec_rw_write(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct lola *chip = entry->private_data; + char line[64]; + unsigned int id, verb, data, extdata; + while (!snd_info_get_line(buffer, line, sizeof(line))) { + if (sscanf(line, "%i %i %i %i", &id, &verb, &data, &extdata) != 4) + continue; + lola_codec_read(chip, id, verb, data, extdata, + &chip->debug_res, + &chip->debug_res_ex); + } +} + +static void lola_proc_codec_rw_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct lola *chip = entry->private_data; + snd_iprintf(buffer, "0x%x 0x%x\n", chip->debug_res, chip->debug_res_ex); +} + +/* + * dump some registers + */ +static void lola_proc_regs_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct lola *chip = entry->private_data; + int i; + + for (i = 0; i < 0x40; i += 4) { + snd_iprintf(buffer, "BAR0 %02x: %08x\n", i, + readl(chip->bar[BAR0].remap_addr + i)); + } + snd_iprintf(buffer, "\n"); + for (i = 0; i < 0x30; i += 4) { + snd_iprintf(buffer, "BAR1 %02x: %08x\n", i, + readl(chip->bar[BAR1].remap_addr + i)); + } + snd_iprintf(buffer, "\n"); + for (i = 0x80; i < 0xa0; i += 4) { + snd_iprintf(buffer, "BAR1 %02x: %08x\n", i, + readl(chip->bar[BAR1].remap_addr + i)); + } + snd_iprintf(buffer, "\n"); + for (i = 0; i < 32; i++) { + snd_iprintf(buffer, "DSD %02x STS %08x\n", i, + lola_dsd_read(chip, i, STS)); + snd_iprintf(buffer, "DSD %02x LPIB %08x\n", i, + lola_dsd_read(chip, i, LPIB)); + snd_iprintf(buffer, "DSD %02x CTL %08x\n", i, + lola_dsd_read(chip, i, CTL)); + snd_iprintf(buffer, "DSD %02x LVIL %08x\n", i, + lola_dsd_read(chip, i, LVI)); + snd_iprintf(buffer, "DSD %02x BDPL %08x\n", i, + lola_dsd_read(chip, i, BDPL)); + snd_iprintf(buffer, "DSD %02x BDPU %08x\n", i, + lola_dsd_read(chip, i, BDPU)); + } +} + +void __devinit lola_proc_debug_new(struct lola *chip) +{ + struct snd_info_entry *entry; + + if (!snd_card_proc_new(chip->card, "codec", &entry)) + snd_info_set_text_ops(entry, chip, lola_proc_codec_read); + if (!snd_card_proc_new(chip->card, "codec_rw", &entry)) { + snd_info_set_text_ops(entry, chip, lola_proc_codec_rw_read); + entry->mode |= S_IWUSR; + entry->c.text.write = lola_proc_codec_rw_write; + } + if (!snd_card_proc_new(chip->card, "regs", &entry)) + snd_info_set_text_ops(entry, chip, lola_proc_regs_read); +} diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index d3350f383966..3df0f530f67c 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -265,7 +265,7 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int if (! timeout) { /* error - no ack */ mutex_unlock(&mgr->msg_mutex); - snd_printk(KERN_ERR "error: no reponse on msg %x\n", msg_frame); + snd_printk(KERN_ERR "error: no response on msg %x\n", msg_frame); return -EIO; } @@ -278,7 +278,7 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int err = get_msg(mgr, &resp, msg_frame); if( request->message_id != resp.message_id ) - snd_printk(KERN_ERR "REPONSE ERROR!\n"); + snd_printk(KERN_ERR "RESPONSE ERROR!\n"); mutex_unlock(&mgr->msg_mutex); return err; diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index 833e7180ad2d..304411c1fe4b 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -1042,11 +1042,11 @@ void pcxhr_msg_tasklet(unsigned long arg) int i, j; if (mgr->src_it_dsp & PCXHR_IRQ_FREQ_CHANGE) - snd_printdd("TASKLET : PCXHR_IRQ_FREQ_CHANGE event occured\n"); + snd_printdd("TASKLET : PCXHR_IRQ_FREQ_CHANGE event occurred\n"); if (mgr->src_it_dsp & PCXHR_IRQ_TIME_CODE) - snd_printdd("TASKLET : PCXHR_IRQ_TIME_CODE event occured\n"); + snd_printdd("TASKLET : PCXHR_IRQ_TIME_CODE event occurred\n"); if (mgr->src_it_dsp & PCXHR_IRQ_NOTIFY) - snd_printdd("TASKLET : PCXHR_IRQ_NOTIFY event occured\n"); + snd_printdd("TASKLET : PCXHR_IRQ_NOTIFY event occurred\n"); if (mgr->src_it_dsp & (PCXHR_IRQ_FREQ_CHANGE | PCXHR_IRQ_TIME_CODE)) { /* clear events FREQ_CHANGE and TIME_CODE */ pcxhr_init_rmh(prmh, CMD_TEST_IT); @@ -1055,7 +1055,7 @@ void pcxhr_msg_tasklet(unsigned long arg) err, prmh->stat[0]); } if (mgr->src_it_dsp & PCXHR_IRQ_ASYNC) { - snd_printdd("TASKLET : PCXHR_IRQ_ASYNC event occured\n"); + snd_printdd("TASKLET : PCXHR_IRQ_ASYNC event occurred\n"); pcxhr_init_rmh(prmh, CMD_ASYNC); prmh->cmd[0] |= 1; /* add SEL_ASYNC_EVENTS */ @@ -1233,7 +1233,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) reg = PCXHR_INPL(mgr, PCXHR_PLX_L2PCIDB); PCXHR_OUTPL(mgr, PCXHR_PLX_L2PCIDB, reg); - /* timer irq occured */ + /* timer irq occurred */ if (reg & PCXHR_IRQ_TIMER) { int timer_toggle = reg & PCXHR_IRQ_TIMER; /* is a 24 bit counter */ @@ -1288,7 +1288,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) if (reg & PCXHR_IRQ_MASK) { if (reg & PCXHR_IRQ_ASYNC) { /* as we didn't request any async notifications, - * some kind of xrun error will probably occured + * some kind of xrun error will probably occurred */ /* better resynchronize all streams next interrupt : */ mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID; diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index d5f5b440fc40..9ff247fc8871 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -150,7 +150,7 @@ MODULE_PARM_DESC(enable, "Enable RME Digi96 soundcard."); #define RME96_RCR_BITPOS_F1 28 #define RME96_RCR_BITPOS_F2 29 -/* Additonal register bits */ +/* Additional register bits */ #define RME96_AR_WSEL (1 << 0) #define RME96_AR_ANALOG (1 << 1) #define RME96_AR_FREQPAD_0 (1 << 2) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index a323eafb9e03..949691a876d3 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -391,7 +391,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); /* Status2 Register bits */ /* MADI ONLY */ -#define HDSPM_version0 (1<<0) /* not realy defined but I guess */ +#define HDSPM_version0 (1<<0) /* not really defined but I guess */ #define HDSPM_version1 (1<<1) /* in former cards it was ??? */ #define HDSPM_version2 (1<<2) @@ -936,7 +936,7 @@ struct hdspm { struct snd_kcontrol *playback_mixer_ctls[HDSPM_MAX_CHANNELS]; /* but input to much, so not used */ struct snd_kcontrol *input_mixer_ctls[HDSPM_MAX_CHANNELS]; - /* full mixer accessable over mixer ioctl or hwdep-device */ + /* full mixer accessible over mixer ioctl or hwdep-device */ struct hdspm_mixer *mixer; struct hdspm_tco *tco; /* NULL if no TCO detected */ diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 1b8f6742b5fa..2b5c7a95ae1f 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -308,7 +308,7 @@ static irqreturn_t sis_interrupt(int irq, void *dev) u32 intr, status; /* We only use the DMA interrupts, and we don't enable any other - * source of interrupts. But, it is possible to see an interupt + * source of interrupts. But, it is possible to see an interrupt * status that didn't actually interrupt us, so eliminate anything * we're not expecting to avoid falsely claiming an IRQ, and an * ensuing endless loop. @@ -773,7 +773,7 @@ static void sis_prepare_timing_voice(struct voice *voice, vperiod = 0; } - /* The interrupt handler implements the timing syncronization, so + /* The interrupt handler implements the timing synchronization, so * setup its state. */ timing->flags |= VOICE_SYNC_TIMING; @@ -1139,7 +1139,7 @@ static int sis_chip_init(struct sis7019 *sis) */ outl(SIS_DMA_CSR_PCI_SETTINGS, io + SIS_DMA_CSR); - /* Reset the syncronization groups for all of the channels + /* Reset the synchronization groups for all of the channels * to be asyncronous. If we start doing SPDIF or 5.1 sound, etc. * we'll need to change how we handle these. Until then, we just * assign sub-mixer 0 to all playback channels, and avoid any diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index edce8a27e3ee..bc823a547550 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -358,7 +358,7 @@ static irqreturn_t snd_ps3_interrupt(int irq, void *dev_id) * filling dummy data, serial automatically start to * consume them and then will generate normal buffer * empty interrupts. - * If both buffer underflow and buffer empty are occured, + * If both buffer underflow and buffer empty are occurred, * it is better to do nomal data transfer than empty one */ snd_ps3_program_dma(card, diff --git a/sound/ppc/snd_ps3_reg.h b/sound/ppc/snd_ps3_reg.h index 03fdee4aaaf2..2e6302079566 100644 --- a/sound/ppc/snd_ps3_reg.h +++ b/sound/ppc/snd_ps3_reg.h @@ -125,7 +125,7 @@ transfers. Any interrupts associated with the canceled transfers will occur as if the transfer had finished. Since this bit is designed to recover from DMA related issues - which are caused by unpredictable situations, it is prefered to wait + which are caused by unpredictable situations, it is preferred to wait for normal DMA transfer end without using this bit. */ #define PS3_AUDIO_CONFIG_CLEAR (1 << 8) /* RWIVF */ @@ -316,13 +316,13 @@ DISABLED=Interrupt generation disabled. /* Audio Port Interrupt Status Register -Indicates Interrupt status, which interrupt has occured, and can clear +Indicates Interrupt status, which interrupt has occurred, and can clear each interrupt in this register. Writing 1b to a field containing 1b clears field and de-asserts interrupt. Writing 0b to a field has no effect. Field vaules are the following: -0 - Interrupt hasn't occured. -1 - Interrupt has occured. +0 - Interrupt hasn't occurred. +1 - Interrupt has occurred. 31 24 23 16 15 8 7 0 @@ -473,7 +473,7 @@ Channel N is out of action by setting 0 to asoen. /* Sampling Rate Specifies the divide ratio of the bit clock (clock output -from bclko) used by the 3-wire Audio Output Clock, whcih +from bclko) used by the 3-wire Audio Output Clock, which is applied to the master clock selected by mcksel. Data output is synchronized with this clock. */ @@ -756,7 +756,7 @@ The STATUS field can be used to monitor the progress of a DMA request. DONE indicates the previous request has completed. EVENT indicates that the DMA engine is waiting for the EVENT to occur. PENDING indicates that the DMA engine has not started processing this -request, but the EVENT has occured. +request, but the EVENT has occurred. DMA indicates that the data transfer is in progress. NOTIFY indicates that the notifier signalling end of transfer is being written. CLEAR indicated that the previous transfer was cleared. @@ -824,7 +824,7 @@ AUDIOFIFO = Audio WriteData FIFO, /* PS3_AUDIO_DMASIZE specifies the number of 128-byte blocks + 1 to transfer. -So a value of 0 means 128-bytes will get transfered. +So a value of 0 means 128-bytes will get transferred. 31 24 23 16 15 8 7 0 diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 5d230cee3fa7..7fbfa051f6e1 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -672,7 +672,7 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) /* re-enable interrupts */ ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); - /* Re-enable recieve and transmit as appropriate */ + /* Re-enable receive and transmit as appropriate */ cr = 0; cr |= (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 4f377c9e868d..eecffb548947 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -481,7 +481,7 @@ struct _pll_div { }; /* Note : pll code from original alc5623 driver. Not sure of how good it is */ -/* usefull only for master mode */ +/* useful only for master mode */ static const struct _pll_div codec_master_pll_div[] = { { 2048000, 8192000, 0x0ea0}, diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index f7cd346fd727..f5ccdbf7ebc6 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -308,8 +308,6 @@ static int jz4740_codec_dev_probe(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, jz4740_codec_dapm_routes, ARRAY_SIZE(jz4740_codec_dapm_routes)); - snd_soc_dapm_new_widgets(codec); - jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 72de47e5d040..2c2a681da0d7 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -161,7 +161,7 @@ static const struct snd_kcontrol_new lm4857_controls[] = { lm4857_get_mode, lm4857_set_mode), }; -/* There is a demux inbetween the the input signal and the output signals. +/* There is a demux between the input signal and the output signals. * Currently there is no easy way to model it in ASoC and since it does not make * much of a difference in practice simply connect the input direclty to the * outputs. */ diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 2a30eae1881c..4d9fb279e146 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -26,7 +26,9 @@ #define pr_fmt(fmt) KBUILD_MODNAME ": " fmt #include <linux/platform_device.h> +#include <linux/delay.h> #include <linux/slab.h> + #include <asm/intel_scu_ipc.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -925,7 +927,7 @@ static struct platform_driver sn95031_codec_driver = { .owner = THIS_MODULE, }, .probe = sn95031_device_probe, - .remove = sn95031_device_remove, + .remove = __devexit_p(sn95031_device_remove), }; static int __init sn95031_init(void) diff --git a/sound/soc/codecs/tlv320aic26.h b/sound/soc/codecs/tlv320aic26.h index 62b1f2261429..67f19c3bebe6 100644 --- a/sound/soc/codecs/tlv320aic26.h +++ b/sound/soc/codecs/tlv320aic26.h @@ -14,14 +14,14 @@ #define AIC26_PAGE_ADDR(page, offset) ((page << 6) | offset) #define AIC26_NUM_REGS AIC26_PAGE_ADDR(3, 0) -/* Page 0: Auxillary data registers */ +/* Page 0: Auxiliary data registers */ #define AIC26_REG_BAT1 AIC26_PAGE_ADDR(0, 0x05) #define AIC26_REG_BAT2 AIC26_PAGE_ADDR(0, 0x06) #define AIC26_REG_AUX AIC26_PAGE_ADDR(0, 0x07) #define AIC26_REG_TEMP1 AIC26_PAGE_ADDR(0, 0x09) #define AIC26_REG_TEMP2 AIC26_PAGE_ADDR(0, 0x0A) -/* Page 1: Auxillary control registers */ +/* Page 1: Auxiliary control registers */ #define AIC26_REG_AUX_ADC AIC26_PAGE_ADDR(1, 0x00) #define AIC26_REG_STATUS AIC26_PAGE_ADDR(1, 0x01) #define AIC26_REG_REFERENCE AIC26_PAGE_ADDR(1, 0x03) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 3bedab26892f..6c43c13f0430 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -884,7 +884,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, if (bypass_pll) return 0; - /* Use PLL, compute apropriate setup for j, d, r and p, the closest + /* Use PLL, compute appropriate setup for j, d, r and p, the closest * one wins the game. Try with d==0 first, next with d!=0. * Constraints for j are according to the datasheet. * The sysclk is divided by 1000 to prevent integer overflows. diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 00b6d87e7bdb..082e9d51963f 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -324,6 +324,10 @@ static void dac33_init_chip(struct snd_soc_codec *codec) dac33_write(codec, DAC33_OUT_AMP_CTRL, dac33_read_reg_cache(codec, DAC33_OUT_AMP_CTRL)); + dac33_write(codec, DAC33_LDAC_PWR_CTRL, + dac33_read_reg_cache(codec, DAC33_LDAC_PWR_CTRL)); + dac33_write(codec, DAC33_RDAC_PWR_CTRL, + dac33_read_reg_cache(codec, DAC33_RDAC_PWR_CTRL)); } static inline int dac33_read_id(struct snd_soc_codec *codec) @@ -670,6 +674,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) { struct snd_soc_codec *codec = dac33->codec; unsigned int delay; + unsigned long flags; switch (dac33->fifo_mode) { case DAC33_FIFO_MODE1: @@ -677,10 +682,10 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) DAC33_THRREG(dac33->nsample)); /* Take the timestamps */ - spin_lock_irq(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp2 = ktime_to_us(ktime_get()); dac33->t_stamp1 = dac33->t_stamp2; - spin_unlock_irq(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); dac33_write16(codec, DAC33_PREFILL_MSB, DAC33_THRREG(dac33->alarm_threshold)); @@ -692,11 +697,11 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) break; case DAC33_FIFO_MODE7: /* Take the timestamp */ - spin_lock_irq(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp1 = ktime_to_us(ktime_get()); /* Move back the timestamp with drain time */ dac33->t_stamp1 -= dac33->mode7_us_to_lthr; - spin_unlock_irq(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); dac33_write16(codec, DAC33_PREFILL_MSB, DAC33_THRREG(DAC33_MODE7_MARGIN)); @@ -714,13 +719,14 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) static inline void dac33_playback_handler(struct tlv320dac33_priv *dac33) { struct snd_soc_codec *codec = dac33->codec; + unsigned long flags; switch (dac33->fifo_mode) { case DAC33_FIFO_MODE1: /* Take the timestamp */ - spin_lock_irq(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp2 = ktime_to_us(ktime_get()); - spin_unlock_irq(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); dac33_write16(codec, DAC33_NSAMPLE_MSB, DAC33_THRREG(dac33->nsample)); @@ -773,10 +779,11 @@ static irqreturn_t dac33_interrupt_handler(int irq, void *dev) { struct snd_soc_codec *codec = dev; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); + unsigned long flags; - spin_lock(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); dac33->t_stamp1 = ktime_to_us(ktime_get()); - spin_unlock(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); /* Do not schedule the workqueue in Mode7 */ if (dac33->fifo_mode != DAC33_FIFO_MODE7) @@ -1020,7 +1027,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) /* * For FIFO bypass mode: * Enable the FIFO bypass (Disable the FIFO use) - * Set the BCLK as continous + * Set the BCLK as continuous */ fifoctrl_a |= DAC33_FBYPAS; aictrl_b |= DAC33_BCLKON; @@ -1173,15 +1180,16 @@ static snd_pcm_sframes_t dac33_dai_delay( unsigned int time_delta, uthr; int samples_out, samples_in, samples; snd_pcm_sframes_t delay = 0; + unsigned long flags; switch (dac33->fifo_mode) { case DAC33_FIFO_BYPASS: break; case DAC33_FIFO_MODE1: - spin_lock(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); t0 = dac33->t_stamp1; t1 = dac33->t_stamp2; - spin_unlock(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); t_now = ktime_to_us(ktime_get()); /* We have not started to fill the FIFO yet, delay is 0 */ @@ -1246,10 +1254,10 @@ static snd_pcm_sframes_t dac33_dai_delay( } break; case DAC33_FIFO_MODE7: - spin_lock(&dac33->lock); + spin_lock_irqsave(&dac33->lock, flags); t0 = dac33->t_stamp1; uthr = dac33->uthr; - spin_unlock(&dac33->lock); + spin_unlock_irqrestore(&dac33->lock, flags); t_now = ktime_to_us(ktime_get()); /* We have not started to fill the FIFO yet, delay is 0 */ diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 8512800f6326..575238d68e5e 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -281,7 +281,7 @@ static inline void twl4030_check_defaults(struct snd_soc_codec *codec) i, val, twl4030_reg[i]); } } - dev_dbg(codec->dev, "Found %d non maching registers. %s\n", + dev_dbg(codec->dev, "Found %d non-matching registers. %s\n", difference, difference ? "Not OK" : "OK"); } @@ -2018,7 +2018,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, u8 mode; /* If the system master clock is not 26MHz, the voice PCM interface is - * not avilable. + * not available. */ if (twl4030->sysclk != 26000) { dev_err(codec->dev, "The board is configured for %u Hz, while" @@ -2028,7 +2028,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, } /* If the codec mode is not option2, the voice PCM interface is not - * avilable. + * available. */ mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & TWL4030_OPT_MODE; diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 482fcdb59bfa..255901c4460d 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1629,8 +1629,10 @@ static int twl6040_probe(struct snd_soc_codec *codec) priv->naudint = naudint; priv->workqueue = create_singlethread_workqueue("twl6040-codec"); - if (!priv->workqueue) + if (!priv->workqueue) { + ret = -ENOMEM; goto work_err; + } INIT_DELAYED_WORK(&priv->delayed_work, twl6040_accessory_work); diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 8f6b5ee6645b..4bbc0a79f01e 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -772,7 +772,7 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec, reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD); snd_soc_write(codec, WM8580_PWRDN1, reg); - /* Make VMID high impedence */ + /* Make VMID high impedance */ reg = snd_soc_read(codec, WM8580_ADC_CONTROL1); reg &= ~0x100; snd_soc_write(codec, WM8580_ADC_CONTROL1, reg); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 3f09deea8d9d..ffa2ffe5ec11 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1312,7 +1312,7 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S24_LE) /* - * The WM8753 supports upto 4 different and mutually exclusive DAI + * The WM8753 supports up to 4 different and mutually exclusive DAI * configurations. This gives 2 PCM's available for use, hifi and voice. * NOTE: The Voice PCM cannot play or capture audio to the CPU as it's DAI * is connected between the wm8753 and a BT codec or GSM modem. diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index ae1cadfae84c..f52b623bb692 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -247,8 +247,6 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re case WM8903_REVISION_NUMBER: case WM8903_INTERRUPT_STATUS_1: case WM8903_WRITE_SEQUENCER_4: - case WM8903_POWER_MANAGEMENT_3: - case WM8903_POWER_MANAGEMENT_2: case WM8903_DC_SERVO_READBACK_1: case WM8903_DC_SERVO_READBACK_2: case WM8903_DC_SERVO_READBACK_3: @@ -875,34 +873,40 @@ SND_SOC_DAPM_MIXER("Left Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 1, 0, SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0, right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), -SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0, - 4, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0, +SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2, + 1, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2, 0, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 4, 0, +SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 1, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 0, 0, +SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 0, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPL_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 7, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPL_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 6, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPL_ENA", 1, WM8903_ANALOGUE_HP_0, 4, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPR_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 3, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPR_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_ENA", 1, WM8903_ANALOGUE_HP_0, 0, 0, NULL, 0), SND_SOC_DAPM_PGA_S("LINEOUTL_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 7, 0, NULL, 0), SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 6, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 5, 0, +SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 5, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTL_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 4, 0, NULL, 0), SND_SOC_DAPM_PGA_S("LINEOUTR_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 3, 0, NULL, 0), SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0, +SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 1, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTR_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DCS Master", WM8903_DC_SERVO_0, 4, 0, NULL, 0), @@ -1037,10 +1041,14 @@ static const struct snd_soc_dapm_route intercon[] = { { "Left Speaker PGA", NULL, "Left Speaker Mixer" }, { "Right Speaker PGA", NULL, "Right Speaker Mixer" }, - { "HPL_ENA_DLY", NULL, "Left Headphone Output PGA" }, - { "HPR_ENA_DLY", NULL, "Right Headphone Output PGA" }, - { "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" }, - { "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" }, + { "HPL_ENA", NULL, "Left Headphone Output PGA" }, + { "HPR_ENA", NULL, "Right Headphone Output PGA" }, + { "HPL_ENA_DLY", NULL, "HPL_ENA" }, + { "HPR_ENA_DLY", NULL, "HPR_ENA" }, + { "LINEOUTL_ENA", NULL, "Left Line Output PGA" }, + { "LINEOUTR_ENA", NULL, "Right Line Output PGA" }, + { "LINEOUTL_ENA_DLY", NULL, "LINEOUTL_ENA" }, + { "LINEOUTR_ENA_DLY", NULL, "LINEOUTR_ENA" }, { "HPL_DCS", NULL, "DCS Master" }, { "HPR_DCS", NULL, "DCS Master" }, diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 443ae580445c..9b3bba4df5b3 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1895,7 +1895,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, pr_debug("Fvco=%dHz\n", target); - /* Find an appropraite FLL_FRATIO and factor it out of the target */ + /* Find an appropriate FLL_FRATIO and factor it out of the target */ for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { fll_div->fll_fratio = fll_fratios[i].fll_fratio; diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 5e0214d6293e..3c7198779c31 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -176,7 +176,7 @@ static int wm8995_pll_factors(struct device *dev, return 0; } -/* Lookup table specifiying SRATE (table 25 in datasheet); some of the +/* Lookup table specifying SRATE (table 25 in datasheet); some of the * output frequencies have been rounded to the standard frequencies * they are intended to match where the error is slight. */ static struct { diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 3b71dd65c966..500011eb8b2b 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3137,7 +3137,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, pr_debug("FLL Fvco=%dHz\n", target); - /* Find an appropraite FLL_FRATIO and factor it out of the target */ + /* Find an appropriate FLL_FRATIO and factor it out of the target */ for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { fll_div->fll_fratio = fll_fratios[i].fll_fratio; diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 28fdfd66661d..3c2ee1bb73cd 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -981,7 +981,7 @@ static int wm8991_set_dai_pll(struct snd_soc_dai *codec_dai, reg = snd_soc_read(codec, WM8991_CLOCKING_2); snd_soc_write(codec, WM8991_CLOCKING_2, reg | WM8991_SYSCLK_SRC); - /* set up N , fractional mode and pre-divisor if neccessary */ + /* set up N , fractional mode and pre-divisor if necessary */ snd_soc_write(codec, WM8991_PLL1, pll_div.n | WM8991_SDM | (pll_div.div2 ? WM8991_PRESCALE : 0)); snd_soc_write(codec, WM8991_PLL2, (u8)(pll_div.k>>8)); diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 379fa22c5b6c..056aef904347 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -324,7 +324,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, pr_debug("Fvco=%dHz\n", target); - /* Find an appropraite FLL_FRATIO and factor it out of the target */ + /* Find an appropriate FLL_FRATIO and factor it out of the target */ for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { fll_div->fll_fratio = fll_fratios[i].fll_fratio; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3dc64c8b6a5c..84e1bd1d2822 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -82,18 +82,18 @@ struct wm8994_priv { int mbc_ena[3]; - /* Platform dependant DRC configuration */ + /* Platform dependent DRC configuration */ const char **drc_texts; int drc_cfg[WM8994_NUM_DRC]; struct soc_enum drc_enum; - /* Platform dependant ReTune mobile configuration */ + /* Platform dependent ReTune mobile configuration */ int num_retune_mobile_texts; const char **retune_mobile_texts; int retune_mobile_cfg[WM8994_NUM_EQ]; struct soc_enum retune_mobile_enum; - /* Platform dependant MBC configuration */ + /* Platform dependent MBC configuration */ int mbc_cfg; const char **mbc_texts; struct soc_enum mbc_enum; @@ -3261,20 +3261,36 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch volume updates (right only; we always do left then right). */ + snd_soc_update_bits(codec, WM8994_AIF1_DAC1_LEFT_VOLUME, + WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU); snd_soc_update_bits(codec, WM8994_AIF1_DAC1_RIGHT_VOLUME, WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU); + snd_soc_update_bits(codec, WM8994_AIF1_DAC2_LEFT_VOLUME, + WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU); snd_soc_update_bits(codec, WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU); + snd_soc_update_bits(codec, WM8994_AIF2_DAC_LEFT_VOLUME, + WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU); snd_soc_update_bits(codec, WM8994_AIF2_DAC_RIGHT_VOLUME, WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU); + snd_soc_update_bits(codec, WM8994_AIF1_ADC1_LEFT_VOLUME, + WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU); snd_soc_update_bits(codec, WM8994_AIF1_ADC1_RIGHT_VOLUME, WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU); + snd_soc_update_bits(codec, WM8994_AIF1_ADC2_LEFT_VOLUME, + WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU); snd_soc_update_bits(codec, WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU); + snd_soc_update_bits(codec, WM8994_AIF2_ADC_LEFT_VOLUME, + WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU); snd_soc_update_bits(codec, WM8994_AIF2_ADC_RIGHT_VOLUME, WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU); + snd_soc_update_bits(codec, WM8994_DAC1_LEFT_VOLUME, + WM8994_DAC1_VU, WM8994_DAC1_VU); snd_soc_update_bits(codec, WM8994_DAC1_RIGHT_VOLUME, WM8994_DAC1_VU, WM8994_DAC1_VU); + snd_soc_update_bits(codec, WM8994_DAC2_LEFT_VOLUME, + WM8994_DAC2_VU, WM8994_DAC2_VU); snd_soc_update_bits(codec, WM8994_DAC2_RIGHT_VOLUME, WM8994_DAC2_VU, WM8994_DAC2_VU); diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 55cdf2982020..91c6b39de50c 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -305,7 +305,7 @@ static int speaker_mode_get(struct snd_kcontrol *kcontrol, /* * Stop any attempts to change speaker mode while the speaker is enabled. * - * We also have some special anti-pop controls dependant on speaker + * We also have some special anti-pop controls dependent on speaker * mode which must be changed along with the mode. */ static int speaker_mode_put(struct snd_kcontrol *kcontrol, @@ -456,7 +456,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, pr_debug("Fvco=%dHz\n", target); - /* Find an appropraite FLL_FRATIO and factor it out of the target */ + /* Find an appropriate FLL_FRATIO and factor it out of the target */ for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { fll_div->fll_fratio = fll_fratios[i].fll_fratio; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 7b6b3c18e299..4005e9af5d61 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -740,12 +740,12 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "SPKL", "Input Switch", "MIXINL" }, { "SPKL", "IN1LP Switch", "IN1LP" }, - { "SPKL", "Output Switch", "Left Output Mixer" }, + { "SPKL", "Output Switch", "Left Output PGA" }, { "SPKL", NULL, "TOCLK" }, { "SPKR", "Input Switch", "MIXINR" }, { "SPKR", "IN1RP Switch", "IN1RP" }, - { "SPKR", "Output Switch", "Right Output Mixer" }, + { "SPKR", "Output Switch", "Right Output PGA" }, { "SPKR", NULL, "TOCLK" }, { "SPKL Boost", "Direct Voice Switch", "Direct Voice" }, @@ -767,8 +767,8 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "SPKOUTRP", NULL, "SPKR Driver" }, { "SPKOUTRN", NULL, "SPKR Driver" }, - { "Left Headphone Mux", "Mixer", "Left Output Mixer" }, - { "Right Headphone Mux", "Mixer", "Right Output Mixer" }, + { "Left Headphone Mux", "Mixer", "Left Output PGA" }, + { "Right Headphone Mux", "Mixer", "Right Output PGA" }, { "Headphone PGA", NULL, "Left Headphone Mux" }, { "Headphone PGA", NULL, "Right Headphone Mux" }, diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index a5af834c8ef5..4ddc6d3b6678 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -434,17 +434,21 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); - mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26)); + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, + ACLKX | AHCLKX | AFSX); break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ - mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); - mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); - mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x2d << 26)); + mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, + ACLKX | ACLKR); + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, + AFSX | AFSR); break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is clock and frame master */ @@ -454,7 +458,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); - mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, (0x3f << 26)); + mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, + ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR); break; default: @@ -644,7 +649,7 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask); mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD); - if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32)) mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXMOD(dev->tdm_slots), FSXMOD(0x1FF)); else @@ -660,7 +665,7 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) AHCLKRE); mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask); - if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32)) mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRMOD(dev->tdm_slots), FSRMOD(0x1FF)); else diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 671ef8dd524c..aab7765f401a 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -110,12 +110,12 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream, slave_config.direction = DMA_TO_DEVICE; slave_config.dst_addr = dma_params->dma_addr; slave_config.dst_addr_width = buswidth; - slave_config.dst_maxburst = dma_params->burstsize; + slave_config.dst_maxburst = dma_params->burstsize * buswidth; } else { slave_config.direction = DMA_FROM_DEVICE; slave_config.src_addr = dma_params->dma_addr; slave_config.src_addr_width = buswidth; - slave_config.src_maxburst = dma_params->burstsize; + slave_config.src_maxburst = dma_params->burstsize * buswidth; } ret = dmaengine_slave_config(iprtd->dma_chan, &slave_config); @@ -303,6 +303,11 @@ static struct snd_soc_platform_driver imx_soc_platform_mx2 = { static int __devinit imx_soc_platform_probe(struct platform_device *pdev) { + struct imx_ssi *ssi = platform_get_drvdata(pdev); + + ssi->dma_params_tx.burstsize = 6; + ssi->dma_params_rx.burstsize = 4; + return snd_soc_register_platform(&pdev->dev, &imx_soc_platform_mx2); } diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index bc92ec620004..ac2ded969253 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -16,7 +16,7 @@ * sane processor vendors have a FIFO per AC97 slot, the i.MX has only * one FIFO which combines all valid receive slots. We cannot even select * which slots we want to receive. The WM9712 with which this driver - * was developped with always sends GPIO status data in slot 12 which + * was developed with always sends GPIO status data in slot 12 which * we receive in our (PCM-) data stream. The only chance we have is to * manually skip this data in the FIQ handler. With sampling rates different * from 48000Hz not every frame has valid receive data, so the ratio diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h index a4406a134892..dc8a87530e3e 100644 --- a/sound/soc/imx/imx-ssi.h +++ b/sound/soc/imx/imx-ssi.h @@ -234,7 +234,4 @@ void imx_pcm_free(struct snd_pcm *pcm); */ #define IMX_SSI_DMABUF_SIZE (64 * 1024) -#define DMA_RXFIFO_BURST 0x4 -#define DMA_TXFIFO_BURST 0x6 - #endif /* _IMX_SSI_H */ diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 0fd6a630db01..e13c6ce46328 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -132,7 +132,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) priv = snd_soc_dai_get_dma_data(cpu_dai, substream); snd_soc_set_runtime_hwparams(substream, &kirkwood_dma_snd_hw); - /* Ensure that all constraints linked to dma burst are fullfilled */ + /* Ensure that all constraints linked to dma burst are fulfilled */ err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, priv->burst * 2, @@ -170,7 +170,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) /* * Enable Error interrupts. We're only ack'ing them but - * it's usefull for diagnostics + * it's useful for diagnostics */ writel((unsigned long)-1, priv->io + KIRKWOOD_ERR_MASK); } diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index ee2c22475a76..d567c322a2fb 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -116,18 +116,20 @@ struct snd_soc_dai_driver sst_platform_dai[] = { static inline void sst_set_stream_status(struct sst_runtime_stream *stream, int state) { - spin_lock(&stream->status_lock); + unsigned long flags; + spin_lock_irqsave(&stream->status_lock, flags); stream->stream_status = state; - spin_unlock(&stream->status_lock); + spin_unlock_irqrestore(&stream->status_lock, flags); } static inline int sst_get_stream_status(struct sst_runtime_stream *stream) { int state; + unsigned long flags; - spin_lock(&stream->status_lock); + spin_lock_irqsave(&stream->status_lock, flags); state = stream->stream_status; - spin_unlock(&stream->status_lock); + spin_unlock_irqrestore(&stream->status_lock, flags); return state; } @@ -440,7 +442,7 @@ static int sst_platform_remove(struct platform_device *pdev) snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sst_platform_dai)); snd_soc_unregister_platform(&pdev->dev); - pr_debug("sst_platform_remove sucess\n"); + pr_debug("sst_platform_remove success\n"); return 0; } @@ -463,7 +465,7 @@ module_init(sst_soc_platform_init); static void __exit sst_soc_platform_exit(void) { platform_driver_unregister(&sst_platform_driver); - pr_debug("sst_soc_platform_exit sucess\n"); + pr_debug("sst_soc_platform_exit success\n"); } module_exit(sst_soc_platform_exit); diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 3167be689621..462cbcbea74a 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -248,7 +248,7 @@ static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = { */ /* To actually apply any modem controlled configuration changes to the codec, - * we must connect codec DAI pins to the modem for a moment. Be carefull not + * we must connect codec DAI pins to the modem for a moment. Be careful not * to interfere with our digital mute function that shares the same hardware. */ static struct timer_list cx81801_timer; static bool cx81801_cmd_pending; @@ -402,9 +402,9 @@ static struct tty_ldisc_ops cx81801_ops = { /* - * Even if not very usefull, the sound card can still work without any of the + * Even if not very useful, the sound card can still work without any of the * above functonality activated. You can still control its audio input/output - * constellation and speakerphone gain from userspace by issueing AT commands + * constellation and speakerphone gain from userspace by issuing AT commands * over the modem port. */ diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 784cff5f67e8..9027da466cae 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -310,7 +310,7 @@ static struct snd_soc_dai_link corgi_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8731-codec-0.001b", + .codec_name = "wm8731-codec.0-001b", .init = corgi_wm8731_init, .ops = &corgi_ops, }; diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 02fb66416ddc..2ce0b2d891d5 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -65,6 +65,7 @@ static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) if (prtd->dma_ch >= 0) { pxa_free_dma(prtd->dma_ch); prtd->dma_ch = -1; + prtd->params = NULL; } return 0; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index ac577263b3e3..b6445757fc54 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -167,7 +167,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { .codec_name = "wm9713-codec", .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa2xx-ac97", - .codec_name = "wm9713-hifi", + .codec_dai_name = "wm9713-hifi", .init = zylonite_wm9713_init, }, { @@ -176,7 +176,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { .codec_name = "wm9713-codec", .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa2xx-ac97-aux", - .codec_name = "wm9713-aux", + .codec_dai_name = "wm9713-aux", }, { .name = "WM9713 Voice", @@ -184,7 +184,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { .codec_name = "wm9713-codec", .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa-ssp-dai.2", - .codec_name = "wm9713-voice", + .codec_dai_name = "wm9713-voice", .ops = &zylonite_voice_ops, }, }; diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c index f6b3a3ce5919..0e80daee8b6f 100644 --- a/sound/soc/samsung/goni_wm8994.c +++ b/sound/soc/samsung/goni_wm8994.c @@ -236,18 +236,18 @@ static struct snd_soc_dai_link goni_dai[] = { .name = "WM8994", .stream_name = "WM8994 HiFi", .cpu_dai_name = "samsung-i2s.0", - .codec_dai_name = "wm8994-hifi", + .codec_dai_name = "wm8994-aif1", .platform_name = "samsung-audio", - .codec_name = "wm8994-codec.0-0x1a", + .codec_name = "wm8994-codec.0-001a", .init = goni_wm8994_init, .ops = &goni_hifi_ops, }, { .name = "WM8994 Voice", .stream_name = "Voice", .cpu_dai_name = "goni-voice-dai", - .codec_dai_name = "wm8994-voice", + .codec_dai_name = "wm8994-aif2", .platform_name = "samsung-audio", - .codec_name = "wm8994-codec.0-0x1a", + .codec_name = "wm8994-codec.0-001a", .ops = &goni_voice_ops, }, }; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 78bfdb3f5d7e..452230975632 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -228,7 +228,7 @@ static const struct snd_kcontrol_new neo1973_wm8753_controls[] = { SOC_DAPM_PIN_SWITCH("Handset Mic"), }; -/* GTA02 specific routes and controlls */ +/* GTA02 specific routes and controls */ #ifdef CONFIG_MACH_NEO1973_GTA02 @@ -372,7 +372,7 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) return 0; } -/* GTA01 specific controlls */ +/* GTA01 specific controls */ #ifdef CONFIG_MACH_NEO1973_GTA01 diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 38aac7d57a59..9c7e8b48aed6 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -350,8 +350,8 @@ static int s3c_pcm_set_fmt(struct snd_soc_dai *cpu_dai, ctl = readl(regs + S3C_PCM_CTL); switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - /* Nothing to do, NB_NF by default */ + case SND_SOC_DAIFMT_IB_NF: + /* Nothing to do, IB_NF by default */ break; default: dev_err(pcm->dev, "Unsupported clock inversion!\n"); diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 0c9997e2d8c0..23c0e83d4c19 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1200,10 +1200,11 @@ static int fsi_probe(struct platform_device *pdev) master->fsib.master = master; pm_runtime_enable(&pdev->dev); - pm_runtime_resume(&pdev->dev); dev_set_drvdata(&pdev->dev, master); + pm_runtime_get_sync(&pdev->dev); fsi_soft_all_reset(master); + pm_runtime_put_sync(&pdev->dev); ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, id_entry->name, master); @@ -1218,8 +1219,17 @@ static int fsi_probe(struct platform_device *pdev) goto exit_free_irq; } - return snd_soc_register_dais(&pdev->dev, fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); + ret = snd_soc_register_dais(&pdev->dev, fsi_soc_dai, + ARRAY_SIZE(fsi_soc_dai)); + if (ret < 0) { + dev_err(&pdev->dev, "cannot snd dai register\n"); + goto exit_snd_soc; + } + + return ret; +exit_snd_soc: + snd_soc_unregister_platform(&pdev->dev); exit_free_irq: free_irq(irq, master); exit_iounmap: @@ -1238,12 +1248,11 @@ static int fsi_remove(struct platform_device *pdev) master = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai)); - snd_soc_unregister_platform(&pdev->dev); - + free_irq(master->irq, master); pm_runtime_disable(&pdev->dev); - free_irq(master->irq, master); + snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai)); + snd_soc_unregister_platform(&pdev->dev); iounmap(master->base); kfree(master); @@ -1321,3 +1330,4 @@ module_exit(fsi_mobile_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("SuperH onchip FSI audio driver"); MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>"); +MODULE_ALIAS("platform:fsi-pcm-audio"); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4dda58926bc5..d8562ce4de7a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -92,8 +92,8 @@ static int min_bytes_needed(unsigned long val) static int format_register_str(struct snd_soc_codec *codec, unsigned int reg, char *buf, size_t len) { - int wordsize = codec->driver->reg_word_size * 2; - int regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; + int wordsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; + int regsize = codec->driver->reg_word_size * 2; int ret; char tmpbuf[len + 1]; char regbuf[regsize + 1]; @@ -132,8 +132,8 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf, size_t total = 0; loff_t p = 0; - wordsize = codec->driver->reg_word_size * 2; - regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; + wordsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; + regsize = codec->driver->reg_word_size * 2; len = wordsize + regsize + 2 + 1; @@ -629,6 +629,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) runtime->hw.rates |= codec_dai_drv->capture.rates; } + ret = -EINVAL; snd_pcm_limit_hw_rates(runtime); if (!runtime->hw.rates) { printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", @@ -640,7 +641,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->name, cpu_dai->name); goto config_err; } - if (!runtime->hw.channels_min || !runtime->hw.channels_max) { + if (!runtime->hw.channels_min || !runtime->hw.channels_max || + runtime->hw.channels_min > runtime->hw.channels_max) { printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", codec_dai->name, cpu_dai->name); goto config_err; @@ -2060,6 +2062,7 @@ const struct dev_pm_ops snd_soc_pm_ops = { .resume = snd_soc_resume, .poweroff = snd_soc_poweroff, }; +EXPORT_SYMBOL_GPL(snd_soc_pm_ops); /* ASoC platform driver */ static struct platform_driver soc_driver = { diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index fcab80b36a37..fc017c0a7b5d 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -331,7 +331,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, goto err; if (gpios[i].wake) { - ret = set_irq_wake(gpio_to_irq(gpios[i].gpio), 1); + ret = irq_set_irq_wake(gpio_to_irq(gpios[i].gpio), 1); if (ret != 0) printk(KERN_ERR "Failed to mark GPIO %d as wake source: %d\n", diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index 8585957477eb..556a57133925 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -370,6 +370,7 @@ static struct platform_driver tegra_snd_harmony_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = tegra_snd_harmony_probe, .remove = __devexit_p(tegra_snd_harmony_remove), diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index 45ffa12f0dab..d47beffedb0f 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -49,7 +49,7 @@ struct ihex_record { u16 address; u8 len; u8 data[256]; - char error; /* true if an error occured parsing this record */ + char error; /* true if an error occurred parsing this record */ u8 max_len; /* maximum record length in whole ihex */ @@ -80,7 +80,7 @@ static u8 usb6fire_fw_ihex_hex(const u8 *data, u8 *crc) /* * returns true if record is available, false otherwise. - * iff an error occured, false will be returned and record->error will be true. + * iff an error occurred, false will be returned and record->error will be true. */ static bool usb6fire_fw_ihex_next_record(struct ihex_record *record) { diff --git a/sound/usb/format.c b/sound/usb/format.c index 95d5d7c4f957..8d042dce0d16 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -177,9 +177,11 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof if (!rate) continue; /* C-Media CM6501 mislabels its 96 kHz altsetting */ + /* Terratec Aureon 7.1 USB C-Media 6206, too */ if (rate == 48000 && nr_rates == 1 && (chip->usb_id == USB_ID(0x0d8c, 0x0201) || - chip->usb_id == USB_ID(0x0d8c, 0x0102)) && + chip->usb_id == USB_ID(0x0d8c, 0x0102) || + chip->usb_id == USB_ID(0x0ccd, 0x00b1)) && fp->altsetting == 5 && fp->maxpacksize == 392) rate = 96000; /* Creative VF0470 Live Cam reports 16 kHz instead of 8kHz */ diff --git a/sound/usb/midi.c b/sound/usb/midi.c index b4b39c0b6c9e..f9289102886a 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1301,6 +1301,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, case USB_ID(0x15ca, 0x0101): /* Textech USB Midi Cable */ case USB_ID(0x15ca, 0x1806): /* Textech USB Midi Cable */ case USB_ID(0x1a86, 0x752d): /* QinHeng CH345 "USB2.0-MIDI" */ + case USB_ID(0xfc08, 0x0101): /* Unknown vendor Cable */ ep->max_transfer = 4; break; /* diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c8c28cd75534..eab06edcc9b7 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1184,7 +1184,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, /* * parse a feature unit * - * most of controlls are defined here. + * most of controls are defined here. */ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void *_ftr) { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index c1a5d7d4580b..bd13d7257240 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -266,7 +266,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, * audio-interface quirks * * returns zero if no standard audio/MIDI parsing is needed. - * returns a postive value if standard audio/midi interfaces are parsed + * returns a positive value if standard audio/midi interfaces are parsed * after this. * returns a negative value at error. */ @@ -533,6 +533,7 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, case USB_ID(0x0d8c, 0x0102): /* C-Media CM6206 / CM106-Like Sound Device */ + case USB_ID(0x0ccd, 0x00b1): /* Terratec Aureon 7.1 USB */ return snd_usb_cm6206_boot_quirk(dev); case USB_ID(0x133e, 0x0815): diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 287ef73b1237..a51340f6f2db 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -20,7 +20,7 @@ at standard samplerates, what led to this part of the usx2y module: It provides the alsa kernel half of the usx2y-alsa-jack driver pair. - The pair uses a hardware dependant alsa-device for mmaped pcm transport. + The pair uses a hardware dependent alsa-device for mmaped pcm transport. Advantage achieved: The usb_hc moves pcm data from/into memory via DMA. That memory is mmaped by jack's usx2y driver. @@ -38,7 +38,7 @@ 2periods works but is useless cause of crackling). This is a first "proof of concept" implementation. - Later, functionalities should migrate to more apropriate places: + Later, functionalities should migrate to more appropriate places: Userland: - The jackd could mmap its float-pcm buffers directly from alsa-lib. - alsa-lib could provide power of 2 period sized shaping combined with int/float |