diff options
author | Linus Torvalds <torvalds@linux-foundation.org> | 2009-07-27 12:17:29 -0700 |
---|---|---|
committer | Linus Torvalds <torvalds@linux-foundation.org> | 2009-07-27 12:17:29 -0700 |
commit | b54c3835469c9548d470e7788cb22a2fd7e21133 (patch) | |
tree | 62e89f7b5ec4acc66bdca7bc7d51bcc44b62357f | |
parent | 04fc0a4097014db7c22da33a56494e3e8a1895d5 (diff) | |
parent | 57e4a5c4f8cfb4b198830c5400f9fc9eb7b75091 (diff) | |
download | op-kernel-dev-b54c3835469c9548d470e7788cb22a2fd7e21133.zip op-kernel-dev-b54c3835469c9548d470e7788cb22a2fd7e21133.tar.gz |
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: hda - Fix mute control with some ALC262 models
ALSA: snd_usb_caiaq: add support for Audio2DJ
ALSA: pcm - Fix hwptr buffer-size overlap bug
ALSA: pcm - Fix warnings in debug loggings
ALSA: pcm - Add logging of hwptr updates and interrupt updates
ASoC: tlv320aic3x: Enable PLL when not bypassed
ALSA: hda - Restore GPIO1 properly at resume with AD1984A
ALSA: ctxfi - Fix uninitialized error checks
ALSA: hda - Use snprintf() to be safer
ALSA: usb-audio - Volume control quirk for QuickCam E 3500
ALSA: pcm - Fix regressions with VMware
-rw-r--r-- | Documentation/sound/alsa/Procfile.txt | 5 | ||||
-rw-r--r-- | sound/core/pcm_lib.c | 36 | ||||
-rw-r--r-- | sound/pci/ctxfi/ctamixer.c | 14 | ||||
-rw-r--r-- | sound/pci/ctxfi/ctsrc.c | 7 | ||||
-rw-r--r-- | sound/pci/hda/patch_analog.c | 2 | ||||
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 33 | ||||
-rw-r--r-- | sound/pci/hda/patch_sigmatel.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/tlv320aic3x.c | 11 | ||||
-rw-r--r-- | sound/usb/Kconfig | 1 | ||||
-rw-r--r-- | sound/usb/caiaq/audio.c | 1 | ||||
-rw-r--r-- | sound/usb/caiaq/device.c | 8 | ||||
-rw-r--r-- | sound/usb/caiaq/device.h | 1 | ||||
-rw-r--r-- | sound/usb/usbmixer.c | 25 |
13 files changed, 107 insertions, 39 deletions
diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt index 381908d..719a819 100644 --- a/Documentation/sound/alsa/Procfile.txt +++ b/Documentation/sound/alsa/Procfile.txt @@ -101,6 +101,8 @@ card*/pcm*/xrun_debug bit 0 = Enable XRUN/jiffies debug messages bit 1 = Show stack trace at XRUN / jiffies check bit 2 = Enable additional jiffies check + bit 3 = Log hwptr update at each period interrupt + bit 4 = Log hwptr update at each snd_pcm_update_hw_ptr() When the bit 0 is set, the driver will show the messages to kernel log when an xrun is detected. The debug message is @@ -117,6 +119,9 @@ card*/pcm*/xrun_debug buggy) hardware that doesn't give smooth pointer updates. This feature is enabled via the bit 2. + Bits 3 and 4 are for logging the hwptr records. Note that + these will give flood of kernel messages. + card*/pcm*/sub*/info The general information of this PCM sub-stream. diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 333e4dd..72cfd47 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -233,6 +233,18 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) xrun(substream); return -EPIPE; } + if (xrun_debug(substream, 8)) { + char name[16]; + pcm_debug_name(substream, name, sizeof(name)); + snd_printd("period_update: %s: pos=0x%x/0x%x/0x%x, " + "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", + name, (unsigned int)pos, + (unsigned int)runtime->period_size, + (unsigned int)runtime->buffer_size, + (unsigned long)old_hw_ptr, + (unsigned long)runtime->hw_ptr_base, + (unsigned long)runtime->hw_ptr_interrupt); + } hw_base = runtime->hw_ptr_base; new_hw_ptr = hw_base + pos; hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size; @@ -244,18 +256,27 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) delta = new_hw_ptr - hw_ptr_interrupt; } if (delta < 0) { - delta += runtime->buffer_size; + if (runtime->periods == 1 || new_hw_ptr < old_hw_ptr) + delta += runtime->buffer_size; if (delta < 0) { hw_ptr_error(substream, "Unexpected hw_pointer value " "(stream=%i, pos=%ld, intr_ptr=%ld)\n", substream->stream, (long)pos, (long)hw_ptr_interrupt); +#if 1 + /* simply skipping the hwptr update seems more + * robust in some cases, e.g. on VMware with + * inaccurate timer source + */ + return 0; /* skip this update */ +#else /* rebase to interrupt position */ hw_base = new_hw_ptr = hw_ptr_interrupt; /* align hw_base to buffer_size */ hw_base -= hw_base % runtime->buffer_size; delta = 0; +#endif } else { hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) @@ -344,6 +365,19 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) xrun(substream); return -EPIPE; } + if (xrun_debug(substream, 16)) { + char name[16]; + pcm_debug_name(substream, name, sizeof(name)); + snd_printd("hw_update: %s: pos=0x%x/0x%x/0x%x, " + "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", + name, (unsigned int)pos, + (unsigned int)runtime->period_size, + (unsigned int)runtime->buffer_size, + (unsigned long)old_hw_ptr, + (unsigned long)runtime->hw_ptr_base, + (unsigned long)runtime->hw_ptr_interrupt); + } + hw_base = runtime->hw_ptr_base; new_hw_ptr = hw_base + pos; diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c index a1db51b3..a7f4a67 100644 --- a/sound/pci/ctxfi/ctamixer.c +++ b/sound/pci/ctxfi/ctamixer.c @@ -242,13 +242,12 @@ static int get_amixer_rsc(struct amixer_mgr *mgr, /* Allocate mem for amixer resource */ amixer = kzalloc(sizeof(*amixer), GFP_KERNEL); - if (NULL == amixer) { - err = -ENOMEM; - return err; - } + if (!amixer) + return -ENOMEM; /* Check whether there are sufficient * amixer resources to meet request. */ + err = 0; spin_lock_irqsave(&mgr->mgr_lock, flags); for (i = 0; i < desc->msr; i++) { err = mgr_get_resource(&mgr->mgr, 1, &idx); @@ -397,12 +396,11 @@ static int get_sum_rsc(struct sum_mgr *mgr, /* Allocate mem for sum resource */ sum = kzalloc(sizeof(*sum), GFP_KERNEL); - if (NULL == sum) { - err = -ENOMEM; - return err; - } + if (!sum) + return -ENOMEM; /* Check whether there are sufficient sum resources to meet request. */ + err = 0; spin_lock_irqsave(&mgr->mgr_lock, flags); for (i = 0; i < desc->msr; i++) { err = mgr_get_resource(&mgr->mgr, 1, &idx); diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c index e1c145d..df43a5c 100644 --- a/sound/pci/ctxfi/ctsrc.c +++ b/sound/pci/ctxfi/ctsrc.c @@ -724,12 +724,11 @@ static int get_srcimp_rsc(struct srcimp_mgr *mgr, /* Allocate mem for SRCIMP resource */ srcimp = kzalloc(sizeof(*srcimp), GFP_KERNEL); - if (NULL == srcimp) { - err = -ENOMEM; - return err; - } + if (!srcimp) + return -ENOMEM; /* Check whether there are sufficient SRCIMP resources. */ + err = 0; spin_lock_irqsave(&mgr->mgr_lock, flags); for (i = 0; i < desc->msr; i++) { err = mgr_get_resource(&mgr->mgr, 1, &idx); diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index be7d25f..3da85ca 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3754,7 +3754,7 @@ static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol, int mute = (!ucontrol->value.integer.value[0] && !ucontrol->value.integer.value[1]); /* toggle GPIO1 according to the mute state */ - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, mute ? 0x02 : 0x0); return ret; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7e99763..8c8b273 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10631,6 +10631,18 @@ static void alc262_lenovo_3000_unsol_event(struct hda_codec *codec, alc262_lenovo_3000_automute(codec, 1); } +static int amp_stereo_mute_update(struct hda_codec *codec, hda_nid_t nid, + int dir, int idx, long *valp) +{ + int i, change = 0; + + for (i = 0; i < 2; i++, valp++) + change |= snd_hda_codec_amp_update(codec, nid, i, dir, idx, + HDA_AMP_MUTE, + *valp ? 0 : HDA_AMP_MUTE); + return change; +} + /* bind hp and internal speaker mute (with plug check) */ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -10639,13 +10651,8 @@ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change; - change = snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp ? 0 : HDA_AMP_MUTE); - change |= snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp ? 0 : HDA_AMP_MUTE); - + change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp); + change |= amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp); if (change) alc262_fujitsu_automute(codec, 0); return change; @@ -10680,10 +10687,7 @@ static int alc262_lenovo_3000_master_sw_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change; - change = snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp ? 0 : HDA_AMP_MUTE); - + change = amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp); if (change) alc262_lenovo_3000_automute(codec, 0); return change; @@ -11854,12 +11858,7 @@ static int alc268_acer_master_sw_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change; - change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp[0] ? 0 : HDA_AMP_MUTE); - change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp[1] ? 0 : HDA_AMP_MUTE); + change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp); if (change) alc268_acer_automute(codec, 0); return change; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index da7f9f6..512f3b9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4066,7 +4066,7 @@ static int stac92xx_add_jack(struct hda_codec *codec, jack->nid = nid; jack->type = type; - sprintf(name, "%s at %s %s Jack", + snprintf(name, sizeof(name), "%s at %s %s Jack", snd_hda_get_jack_type(def_conf), snd_hda_get_jack_connectivity(def_conf), snd_hda_get_jack_location(def_conf)); diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index ab099f4..cb0d1bf 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -767,6 +767,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; u16 pll_d = 1; + u8 reg; /* select data word length */ data = @@ -801,8 +802,16 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, pll_q &= 0xf; aic3x_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT); aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV); - } else + /* disable PLL if it is bypassed */ + reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); + aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg & ~PLL_ENABLE); + + } else { aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV); + /* enable PLL when it is used */ + reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); + aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg | PLL_ENABLE); + } /* Route Left DAC to left channel input and * right DAC to right channel input */ diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 523aec1..73525c0 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -48,6 +48,7 @@ config SND_USB_CAIAQ * Native Instruments Kore Controller * Native Instruments Kore Controller 2 * Native Instruments Audio Kontrol 1 + * Native Instruments Audio 2 DJ * Native Instruments Audio 4 DJ * Native Instruments Audio 8 DJ * Native Instruments Guitar Rig Session I/O diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 8f9b60c..121af06 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -646,6 +646,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_GUITARRIGMOBILE): dev->samplerates |= SNDRV_PCM_RATE_192000; /* fall thru */ + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO2DJ): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): dev->samplerates |= SNDRV_PCM_RATE_88200; diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index de38108..83e6c13 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,13 +35,14 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.18"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.19"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," "{Native Instruments, Kore Controller}," "{Native Instruments, Kore Controller 2}," "{Native Instruments, Audio Kontrol 1}," + "{Native Instruments, Audio 2 DJ}," "{Native Instruments, Audio 4 DJ}," "{Native Instruments, Audio 8 DJ}," "{Native Instruments, Session I/O}," @@ -121,6 +122,11 @@ static struct usb_device_id snd_usb_id_table[] = { .idVendor = USB_VID_NATIVEINSTRUMENTS, .idProduct = USB_PID_AUDIO4DJ }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_AUDIO2DJ + }, { /* terminator */ } }; diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index ece7351..44e3edf 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -10,6 +10,7 @@ #define USB_PID_KORECONTROLLER 0x4711 #define USB_PID_KORECONTROLLER2 0x4712 #define USB_PID_AK1 0x0815 +#define USB_PID_AUDIO2DJ 0x041c #define USB_PID_AUDIO4DJ 0x0839 #define USB_PID_AUDIO8DJ 0x1978 #define USB_PID_SESSIONIO 0x1915 diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 4bd3a7a..ec9cdf9 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -990,20 +990,35 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, break; } - /* quirk for UDA1321/N101 */ - /* note that detection between firmware 2.1.1.7 (N101) and later 2.1.1.21 */ - /* is not very clear from datasheets */ - /* I hope that the min value is -15360 for newer firmware --jk */ + /* volume control quirks */ switch (state->chip->usb_id) { case USB_ID(0x0471, 0x0101): case USB_ID(0x0471, 0x0104): case USB_ID(0x0471, 0x0105): case USB_ID(0x0672, 0x1041): + /* quirk for UDA1321/N101. + * note that detection between firmware 2.1.1.7 (N101) + * and later 2.1.1.21 is not very clear from datasheets. + * I hope that the min value is -15360 for newer firmware --jk + */ if (!strcmp(kctl->id.name, "PCM Playback Volume") && cval->min == -15616) { - snd_printk(KERN_INFO "using volume control quirk for the UDA1321/N101 chip\n"); + snd_printk(KERN_INFO + "set volume quirk for UDA1321/N101 chip\n"); cval->max = -256; } + break; + + case USB_ID(0x046d, 0x09a4): + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + snd_printk(KERN_INFO + "set volume quirk for QuickCam E3500\n"); + cval->min = 6080; + cval->max = 8768; + cval->res = 192; + } + break; + } snd_printdd(KERN_INFO "[%d] FU [%s] ch = %d, val = %d/%d/%d\n", |