diff options
author | Linus Torvalds <torvalds@linux-foundation.org> | 2010-05-26 08:41:25 -0700 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2010-05-26 08:41:25 -0700 |
commit | 2214482cb00e6da1397c2ecde5b392490eb9637f (patch) | |
tree | 7375817fa8b76741a0e362716b59860255e526ba /sound | |
parent | 13da9e200fe4740b02cd51e07ab454627e228920 (diff) | |
parent | d21921215af2fe33190a3b5b166b145e607e537d (diff) | |
download | linux-2214482cb00e6da1397c2ecde5b392490eb9637f.tar.bz2 |
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: emu10k1: allow high-resolution mixer controls
ALSA: pcm: fix delta calculation at boundary wraparound
ALSA: hda_intel: fix handling of non-completion stream interrupts
ALSA: usb/caiaq: fix Traktor Kontrol X1 ABS_HAT2X axis
ALSA: hda: Fix model quirk for Dell M1730
ALSA: hda - iMac9,1 sound fixes
ALSA: hda: Use LPIB for Toshiba A100-259
ALSA: hda: Use LPIB for Acer Aspire 5110
ALSA: aw2-alsa.c: use pci_ids.h defines and fix checkpatch.pl noise
ALSA: usb-audio: add support for Akai MPD16
ALSA: pcm: fix the fix of the runtime->boundary calculation
Diffstat (limited to 'sound')
-rw-r--r-- | sound/core/pcm_lib.c | 13 | ||||
-rw-r--r-- | sound/core/pcm_native.c | 39 | ||||
-rw-r--r-- | sound/pci/aw2/aw2-alsa.c | 11 | ||||
-rw-r--r-- | sound/pci/emu10k1/emufx.c | 36 | ||||
-rw-r--r-- | sound/pci/hda/hda_intel.c | 9 | ||||
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 84 | ||||
-rw-r--r-- | sound/pci/hda/patch_sigmatel.c | 2 | ||||
-rw-r--r-- | sound/usb/caiaq/input.c | 2 | ||||
-rw-r--r-- | sound/usb/midi.c | 110 | ||||
-rw-r--r-- | sound/usb/midi.h | 2 | ||||
-rw-r--r-- | sound/usb/quirks-table.h | 11 | ||||
-rw-r--r-- | sound/usb/quirks.c | 1 | ||||
-rw-r--r-- | sound/usb/usbaudio.h | 1 |
13 files changed, 219 insertions, 102 deletions
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a2ff86189d2a..e9d98be190c5 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -345,7 +345,9 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, new_hw_ptr = hw_base + pos; } __delta: - delta = (new_hw_ptr - old_hw_ptr) % runtime->boundary; + delta = new_hw_ptr - old_hw_ptr; + if (delta < 0) + delta += runtime->boundary; if (xrun_debug(substream, in_interrupt ? XRUN_DEBUG_PERIODUPDATE : XRUN_DEBUG_HWPTRUPDATE)) { char name[16]; @@ -439,8 +441,13 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, snd_pcm_playback_silence(substream, new_hw_ptr); if (in_interrupt) { - runtime->hw_ptr_interrupt = new_hw_ptr - - (new_hw_ptr % runtime->period_size); + delta = new_hw_ptr - runtime->hw_ptr_interrupt; + if (delta < 0) + delta += runtime->boundary; + delta -= (snd_pcm_uframes_t)delta % runtime->period_size; + runtime->hw_ptr_interrupt += delta; + if (runtime->hw_ptr_interrupt >= runtime->boundary) + runtime->hw_ptr_interrupt -= runtime->boundary; } runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 644c2bb17b86..303ac04ff6e4 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -27,7 +27,6 @@ #include <linux/pm_qos_params.h> #include <linux/uio.h> #include <linux/dma-mapping.h> -#include <linux/math64.h> #include <sound/core.h> #include <sound/control.h> #include <sound/info.h> @@ -370,38 +369,6 @@ static int period_to_usecs(struct snd_pcm_runtime *runtime) return usecs; } -static int calc_boundary(struct snd_pcm_runtime *runtime) -{ - u_int64_t boundary; - - boundary = (u_int64_t)runtime->buffer_size * - (u_int64_t)runtime->period_size; -#if BITS_PER_LONG < 64 - /* try to find lowest common multiple for buffer and period */ - if (boundary > LONG_MAX - runtime->buffer_size) { - u_int32_t remainder = -1; - u_int32_t divident = runtime->buffer_size; - u_int32_t divisor = runtime->period_size; - while (remainder) { - remainder = divident % divisor; - if (remainder) { - divident = divisor; - divisor = remainder; - } - } - boundary = div_u64(boundary, divisor); - if (boundary > LONG_MAX - runtime->buffer_size) - return -ERANGE; - } -#endif - if (boundary == 0) - return -ERANGE; - runtime->boundary = boundary; - while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size) - runtime->boundary *= 2; - return 0; -} - static int snd_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -477,9 +444,9 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, runtime->stop_threshold = runtime->buffer_size; runtime->silence_threshold = 0; runtime->silence_size = 0; - err = calc_boundary(runtime); - if (err < 0) - goto _error; + runtime->boundary = runtime->buffer_size; + while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size) + runtime->boundary *= 2; snd_pcm_timer_resolution_change(substream); runtime->status->state = SNDRV_PCM_STATE_SETUP; diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 67921f93a41e..c15002242d98 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -26,7 +26,7 @@ #include <linux/slab.h> #include <linux/interrupt.h> #include <linux/delay.h> -#include <asm/io.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/initval.h> #include <sound/pcm.h> @@ -44,9 +44,6 @@ MODULE_LICENSE("GPL"); /********************************* * DEFINES ********************************/ -#define PCI_VENDOR_ID_SAA7146 0x1131 -#define PCI_DEVICE_ID_SAA7146 0x7146 - #define CTL_ROUTE_ANALOG 0 #define CTL_ROUTE_DIGITAL 1 @@ -165,7 +162,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard."); static DEFINE_PCI_DEVICE_TABLE(snd_aw2_ids) = { - {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0, + {PCI_VENDOR_ID_PHILIPS, PCI_DEVICE_ID_PHILIPS_SAA7146, 0, 0, 0, 0, 0}, {0} }; @@ -419,7 +416,7 @@ static int snd_aw2_pcm_playback_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - snd_printdd(KERN_DEBUG "aw2: Playback_open \n"); + snd_printdd(KERN_DEBUG "aw2: Playback_open\n"); runtime->hw = snd_aw2_playback_hw; return 0; } @@ -435,7 +432,7 @@ static int snd_aw2_pcm_capture_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - snd_printdd(KERN_DEBUG "aw2: Capture_open \n"); + snd_printdd(KERN_DEBUG "aw2: Capture_open\n"); runtime->hw = snd_aw2_capture_hw; return 0; } diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 4b302d86f5f2..7a9401462c1c 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -35,6 +35,7 @@ #include <linux/vmalloc.h> #include <linux/init.h> #include <linux/mutex.h> +#include <linux/moduleparam.h> #include <sound/core.h> #include <sound/tlv.h> @@ -50,6 +51,10 @@ #define EMU10K1_CENTER_LFE_FROM_FRONT #endif +static bool high_res_gpr_volume; +module_param(high_res_gpr_volume, bool, 0444); +MODULE_PARM_DESC(high_res_gpr_volume, "GPR mixer controls use 31-bit range."); + /* * Tables */ @@ -296,6 +301,7 @@ static const u32 db_table[101] = { /* EMU10k1/EMU10k2 DSP control db gain */ static const DECLARE_TLV_DB_SCALE(snd_emu10k1_db_scale1, -4000, 40, 1); +static const DECLARE_TLV_DB_LINEAR(snd_emu10k1_db_linear, TLV_DB_GAIN_MUTE, 0); static const u32 onoff_table[2] = { 0x00000000, 0x00000001 @@ -1072,10 +1078,17 @@ snd_emu10k1_init_mono_control(struct snd_emu10k1_fx8010_control_gpr *ctl, strcpy(ctl->id.name, name); ctl->vcount = ctl->count = 1; ctl->gpr[0] = gpr + 0; ctl->value[0] = defval; - ctl->min = 0; - ctl->max = 100; - ctl->tlv = snd_emu10k1_db_scale1; - ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100; + if (high_res_gpr_volume) { + ctl->min = 0; + ctl->max = 0x7fffffff; + ctl->tlv = snd_emu10k1_db_linear; + ctl->translation = EMU10K1_GPR_TRANSLATION_NONE; + } else { + ctl->min = 0; + ctl->max = 100; + ctl->tlv = snd_emu10k1_db_scale1; + ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100; + } } static void __devinit @@ -1087,10 +1100,17 @@ snd_emu10k1_init_stereo_control(struct snd_emu10k1_fx8010_control_gpr *ctl, ctl->vcount = ctl->count = 2; ctl->gpr[0] = gpr + 0; ctl->value[0] = defval; ctl->gpr[1] = gpr + 1; ctl->value[1] = defval; - ctl->min = 0; - ctl->max = 100; - ctl->tlv = snd_emu10k1_db_scale1; - ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100; + if (high_res_gpr_volume) { + ctl->min = 0; + ctl->max = 0x7fffffff; + ctl->tlv = snd_emu10k1_db_linear; + ctl->translation = EMU10K1_GPR_TRANSLATION_NONE; + } else { + ctl->min = 0; + ctl->max = 100; + ctl->tlv = snd_emu10k1_db_scale1; + ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100; + } } static void __devinit diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 170610e1d7da..77e22c2a8caa 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1097,6 +1097,7 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) struct azx *chip = dev_id; struct azx_dev *azx_dev; u32 status; + u8 sd_status; int i, ok; spin_lock(&chip->reg_lock); @@ -1110,8 +1111,10 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) for (i = 0; i < chip->num_streams; i++) { azx_dev = &chip->azx_dev[i]; if (status & azx_dev->sd_int_sta_mask) { + sd_status = azx_sd_readb(azx_dev, SD_STS); azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); - if (!azx_dev->substream || !azx_dev->running) + if (!azx_dev->substream || !azx_dev->running || + !(sd_status & SD_INT_COMPLETE)) continue; /* check whether this IRQ is really acceptable */ ok = azx_position_ok(chip, azx_dev); @@ -2279,12 +2282,14 @@ static int azx_dev_free(struct snd_device *device) * white/black-listing for position_fix */ static struct snd_pci_quirk position_fix_list[] __devinitdata = { + SND_PCI_QUIRK(0x1025, 0x009f, "Acer Aspire 5110", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB), SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB), SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 53538b0f9991..17d4548cc353 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7025,6 +7025,14 @@ static struct hda_input_mux alc889A_mb31_capture_source = { }, }; +static struct hda_input_mux alc889A_imac91_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x01 }, + { "Line", 0x2 }, /* Not sure! */ + }, +}; + /* * 2ch mode */ @@ -7486,15 +7494,8 @@ static struct snd_kcontrol_new alc885_macmini3_mixer[] = { }; static struct snd_kcontrol_new alc885_imac91_mixer[] = { - HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Line-Out Playback Switch", 0x0c, 0x02, HDA_INPUT), - HDA_CODEC_MUTE ("Speaker Playback Switch", 0x14, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT), - HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT), { } /* end */ }; @@ -7995,61 +7996,56 @@ static struct hda_verb alc885_mbp3_init_verbs[] = { /* iMac 9,1 */ static struct hda_verb alc885_imac91_init_verbs[] = { - /* Line-Out mixer: unmute input/output amp left and right (volume = 0) */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Rear mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* HP Pin: output 0 (0x0c) */ + /* Internal Speaker Pin (0x0c) */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) }, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) }, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP Pin: Rear */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, - /* Internal Speakers: output 0 (0x0d) */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC880_HP_EVENT | AC_USRSP_EN)}, + /* Line in Rear */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* Mic (rear) pin: input vref at 80% */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Front Mic pin: input vref at 80% */ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line In pin: use output 1 when in LineOut mode */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, - - /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ + /* Rear mixer */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Line-Out mixer: unmute input/output amp left and right (volume = 0) */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Input mixer2 */ + /* 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Input mixer3 */ + /* 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* ADC1: mute amp left and right */ + /* 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In */ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* ADC2: mute amp left and right */ + /* 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In */ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* ADC3: mute amp left and right */ + /* 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In */ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - { } }; @@ -8118,7 +8114,7 @@ static void alc885_imac91_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x18; spec->autocfg.speaker_pins[1] = 0x1a; } @@ -9627,14 +9623,14 @@ static struct alc_config_preset alc882_presets[] = { .init_hook = alc885_imac24_init_hook, }, [ALC885_IMAC91] = { - .mixers = { alc885_imac91_mixer, alc882_chmode_mixer }, + .mixers = {alc885_imac91_mixer}, .init_verbs = { alc885_imac91_init_verbs, alc880_gpio1_init_verbs }, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, - .channel_mode = alc885_mbp_4ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes), - .input_mux = &alc882_capture_source, + .channel_mode = alc885_mba21_ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes), + .input_mux = &alc889A_imac91_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, .unsol_event = alc_automute_amp_unsol_event, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a0e06d82da1f..f1e7babd6920 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2078,12 +2078,12 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2000, "Intel D965", STAC_D965_3ST), /* Dell 3 stack systems */ - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f7, "Dell XPS M1730", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01dd, "Dell Dimension E520", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ed, "Dell ", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f4, "Dell ", STAC_DELL_3ST), /* Dell 3 stack systems with verb table in BIOS */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f7, "Dell XPS M1730", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0227, "Dell Vostro 1400 ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022e, "Dell ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022f, "Dell Inspiron 1525", STAC_DELL_BIOS), diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index 8bbfbfd4c658..dcb620796d9e 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -171,7 +171,7 @@ static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *dev, input_report_abs(input_dev, ABS_HAT0Y, (buf[4] << 8) | buf[5]); input_report_abs(input_dev, ABS_HAT1X, (buf[12] << 8) | buf[13]); input_report_abs(input_dev, ABS_HAT1Y, (buf[2] << 8) | buf[3]); - input_report_abs(input_dev, ABS_HAT2X, (buf[15] << 8) | buf[15]); + input_report_abs(input_dev, ABS_HAT2X, (buf[14] << 8) | buf[15]); input_report_abs(input_dev, ABS_HAT2Y, (buf[0] << 8) | buf[1]); input_report_abs(input_dev, ABS_HAT3X, (buf[10] << 8) | buf[11]); input_report_abs(input_dev, ABS_HAT3Y, (buf[6] << 8) | buf[7]); diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 8b1e4b124a9f..46785643c66d 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -645,6 +645,105 @@ static struct usb_protocol_ops snd_usbmidi_cme_ops = { }; /* + * AKAI MPD16 protocol: + * + * For control port (endpoint 1): + * ============================== + * One or more chunks consisting of first byte of (0x10 | msg_len) and then a + * SysEx message (msg_len=9 bytes long). + * + * For data port (endpoint 2): + * =========================== + * One or more chunks consisting of first byte of (0x20 | msg_len) and then a + * MIDI message (msg_len bytes long) + * + * Messages sent: Active Sense, Note On, Poly Pressure, Control Change. + */ +static void snd_usbmidi_akai_input(struct snd_usb_midi_in_endpoint *ep, + uint8_t *buffer, int buffer_length) +{ + unsigned int pos = 0; + unsigned int len = (unsigned int)buffer_length; + while (pos < len) { + unsigned int port = (buffer[pos] >> 4) - 1; + unsigned int msg_len = buffer[pos] & 0x0f; + pos++; + if (pos + msg_len <= len && port < 2) + snd_usbmidi_input_data(ep, 0, &buffer[pos], msg_len); + pos += msg_len; + } +} + +#define MAX_AKAI_SYSEX_LEN 9 + +static void snd_usbmidi_akai_output(struct snd_usb_midi_out_endpoint *ep, + struct urb *urb) +{ + uint8_t *msg; + int pos, end, count, buf_end; + uint8_t tmp[MAX_AKAI_SYSEX_LEN]; + struct snd_rawmidi_substream *substream = ep->ports[0].substream; + + if (!ep->ports[0].active) + return; + + msg = urb->transfer_buffer + urb->transfer_buffer_length; + buf_end = ep->max_transfer - MAX_AKAI_SYSEX_LEN - 1; + + /* only try adding more data when there's space for at least 1 SysEx */ + while (urb->transfer_buffer_length < buf_end) { + count = snd_rawmidi_transmit_peek(substream, + tmp, MAX_AKAI_SYSEX_LEN); + if (!count) { + ep->ports[0].active = 0; + return; + } + /* try to skip non-SysEx data */ + for (pos = 0; pos < count && tmp[pos] != 0xF0; pos++) + ; + + if (pos > 0) { + snd_rawmidi_transmit_ack(substream, pos); + continue; + } + + /* look for the start or end marker */ + for (end = 1; end < count && tmp[end] < 0xF0; end++) + ; + + /* next SysEx started before the end of current one */ + if (end < count && tmp[end] == 0xF0) { + /* it's incomplete - drop it */ + snd_rawmidi_transmit_ack(substream, end); + continue; + } + /* SysEx complete */ + if (end < count && tmp[end] == 0xF7) { + /* queue it, ack it, and get the next one */ + count = end + 1; + msg[0] = 0x10 | count; + memcpy(&msg[1], tmp, count); + snd_rawmidi_transmit_ack(substream, count); + urb->transfer_buffer_length += count + 1; + msg += count + 1; + continue; + } + /* less than 9 bytes and no end byte - wait for more */ + if (count < MAX_AKAI_SYSEX_LEN) { + ep->ports[0].active = 0; + return; + } + /* 9 bytes and no end marker in sight - malformed, skip it */ + snd_rawmidi_transmit_ack(substream, count); + } +} + +static struct usb_protocol_ops snd_usbmidi_akai_ops = { + .input = snd_usbmidi_akai_input, + .output = snd_usbmidi_akai_output, +}; + +/* * Novation USB MIDI protocol: number of data bytes is in the first byte * (when receiving) (+1!) or in the second byte (when sending); data begins * at the third byte. @@ -1434,6 +1533,11 @@ static struct port_info { EXTERNAL_PORT(0x086a, 0x0001, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0002, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0003, 4, "%s Broadcast"), + /* Akai MPD16 */ + CONTROL_PORT(0x09e8, 0x0062, 0, "%s Control"), + PORT_INFO(0x09e8, 0x0062, 1, "%s MIDI", 0, + SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | + SNDRV_SEQ_PORT_TYPE_HARDWARE), /* Access Music Virus TI */ EXTERNAL_PORT(0x133e, 0x0815, 0, "%s MIDI"), PORT_INFO(0x133e, 0x0815, 1, "%s Synth", 0, @@ -2035,6 +2139,12 @@ int snd_usbmidi_create(struct snd_card *card, umidi->usb_protocol_ops = &snd_usbmidi_cme_ops; err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; + case QUIRK_MIDI_AKAI: + umidi->usb_protocol_ops = &snd_usbmidi_akai_ops; + err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); + /* endpoint 1 is input-only */ + endpoints[1].out_cables = 0; + break; default: snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type); err = -ENXIO; diff --git a/sound/usb/midi.h b/sound/usb/midi.h index 2089ec987c66..2fca80b744c0 100644 --- a/sound/usb/midi.h +++ b/sound/usb/midi.h @@ -37,6 +37,8 @@ struct snd_usb_midi_endpoint_info { /* for QUIRK_MIDI_CME, data is NULL */ +/* for QUIRK_MIDI_AKAI, data is NULL */ + int snd_usbmidi_create(struct snd_card *card, struct usb_interface *iface, struct list_head *midi_list, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 91ddef31bcbd..f8797f61a24b 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1973,6 +1973,17 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* AKAI devices */ +{ + USB_DEVICE(0x09e8, 0x0062), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "AKAI", + .product_name = "MPD16", + .ifnum = 0, + .type = QUIRK_MIDI_AKAI, + } +}, + /* TerraTec devices */ { USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0012), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 136e5b4cf6de..b45e54c09ba2 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -289,6 +289,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_FASTLANE] = create_any_midi_quirk, [QUIRK_MIDI_EMAGIC] = create_any_midi_quirk, [QUIRK_MIDI_CME] = create_any_midi_quirk, + [QUIRK_MIDI_AKAI] = create_any_midi_quirk, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index d679e72a3e5c..06ebf24d3a4d 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -74,6 +74,7 @@ enum quirk_type { QUIRK_MIDI_FASTLANE, QUIRK_MIDI_EMAGIC, QUIRK_MIDI_CME, + QUIRK_MIDI_AKAI, QUIRK_MIDI_US122L, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, |