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Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

Browse source 
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: usb/endpoint, fix dangling pointer use
  ALSA: asihpi - Get rid of incorrect "long" types and casts.
  ASoC: DaVinci: Fix McASP hardware FIFO configuration
  ALSA: hda - Fix line-in for mb5 model MacBook (Pro) 5,1 / 5,2
  ALSA: usb-audio: fix UAC2 control value queries
  ALSA: usb-audio: parse UAC2 sample rate ranges correctly
  ALSA: usb-audio: fix control messages for USB_RECIP_INTERFACE
  ALSA: usb-audio: add check for faulty clock in parse_audio_format_rates_v2()
  ALSA: hda - Don't check capture source mixer if no ADC is available
This commit is contained in:
Linus Torvalds 2010-06-27 07:39:57 -07:00
commit 29ccb201a2
8 changed files with 140 additions and 59 deletions
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22
sound/pci/asihpi/hpi6205.c
View file

@ -941,11 +941,11 @@ static void outstream_host_buffer_free(struct hpi_adapter_obj *pao,
}
static long outstream_get_space_available(struct hpi_hostbuffer_status
static u32 outstream_get_space_available(struct hpi_hostbuffer_status
*status)
{
return status->size_in_bytes - ((long)(status->host_index) -
(long)(status->dSP_index));
return status->size_in_bytes - (status->host_index -
status->dSP_index);
}
static void outstream_write(struct hpi_adapter_obj *pao,
@ -954,7 +954,7 @@ static void outstream_write(struct hpi_adapter_obj *pao,
struct hpi_hw_obj *phw = pao->priv;
struct bus_master_interface *interface = phw->p_interface_buffer;
struct hpi_hostbuffer_status *status;
long space_available;
u32 space_available;
if (!phw->outstream_host_buffer_size[phm->obj_index]) {
/* there is no BBM buffer, write via message */
@ -1007,7 +1007,7 @@ static void outstream_write(struct hpi_adapter_obj *pao,
}
space_available = outstream_get_space_available(status);
if (space_available < (long)phm->u.d.u.data.data_size) {
if (space_available < phm->u.d.u.data.data_size) {
phr->error = HPI_ERROR_INVALID_DATASIZE;
return;
}
@ -1018,7 +1018,7 @@ static void outstream_write(struct hpi_adapter_obj *pao,
&& hpios_locked_mem_valid(&phw->outstream_host_buffers[phm->
obj_index])) {
u8 *p_bbm_data;
long l_first_write;
u32 l_first_write;
u8 *p_app_data = (u8 *)phm->u.d.u.data.pb_data;
if (hpios_locked_mem_get_virt_addr(&phw->
@ -1248,9 +1248,9 @@ static void instream_start(struct hpi_adapter_obj *pao,
hw_message(pao, phm, phr);
}
static long instream_get_bytes_available(struct hpi_hostbuffer_status *status)
static u32 instream_get_bytes_available(struct hpi_hostbuffer_status *status)
{
return (long)(status->dSP_index) - (long)(status->host_index);
return status->dSP_index - status->host_index;
}
static void instream_read(struct hpi_adapter_obj *pao,
@ -1259,9 +1259,9 @@ static void instream_read(struct hpi_adapter_obj *pao,
struct hpi_hw_obj *phw = pao->priv;
struct bus_master_interface *interface = phw->p_interface_buffer;
struct hpi_hostbuffer_status *status;
long data_available;
u32 data_available;
u8 *p_bbm_data;
long l_first_read;
u32 l_first_read;
u8 *p_app_data = (u8 *)phm->u.d.u.data.pb_data;
if (!phw->instream_host_buffer_size[phm->obj_index]) {
@ -1272,7 +1272,7 @@ static void instream_read(struct hpi_adapter_obj *pao,
status = &interface->instream_host_buffer_status[phm->obj_index];
data_available = instream_get_bytes_available(status);
if (data_available < (long)phm->u.d.u.data.data_size) {
if (data_available < phm->u.d.u.data.data_size) {
phr->error = HPI_ERROR_INVALID_DATASIZE;
return;
}

35
sound/pci/hda/patch_realtek.c
View file

@ -2619,16 +2619,18 @@ static int alc_build_controls(struct hda_codec *codec)
}
/* assign Capture Source enums to NID */
kctl = snd_hda_find_mixer_ctl(codec, "Capture Source");
if (!kctl)
kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
for (i = 0; kctl && i < kctl->count; i++) {
hda_nid_t *nids = spec->capsrc_nids;
if (!nids)
nids = spec->adc_nids;
err = snd_hda_add_nid(codec, kctl, i, nids[i]);
if (err < 0)
return err;
if (spec->capsrc_nids || spec->adc_nids) {
kctl = snd_hda_find_mixer_ctl(codec, "Capture Source");
if (!kctl)
kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
for (i = 0; kctl && i < kctl->count; i++) {
hda_nid_t *nids = spec->capsrc_nids;
if (!nids)
nids = spec->adc_nids;
err = snd_hda_add_nid(codec, kctl, i, nids[i]);
if (err < 0)
return err;
}
}
if (spec->cap_mixer) {
const char *kname = kctl ? kctl->id.name : NULL;
@ -6948,7 +6950,7 @@ static struct hda_input_mux mb5_capture_source = {
.num_items = 3,
.items = {
{ "Mic", 0x1 },
{ "Line", 0x2 },
{ "Line", 0x7 },
{ "CD", 0x4 },
},
};
@ -7469,8 +7471,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = {
HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT),
HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Line Boost", 0x15, 0x00, HDA_INPUT),
@ -7853,10 +7855,9 @@ static struct hda_verb alc885_mb5_init_verbs[] = {
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{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)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0x1)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x7)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x4)},
{ }
};

2
sound/soc/davinci/davinci-mcasp.c
View file

@ -612,7 +612,6 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream)
NUMDMA_MASK);
mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK);
mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
}
if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) {
@ -623,7 +622,6 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream)
NUMDMA_MASK);
mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK);
mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
}
}

12
sound/usb/clock.c
View file

@ -103,7 +103,8 @@ static int uac_clock_selector_get_val(struct snd_usb_audio *chip, int selector_i
ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0),
UAC2_CS_CUR,
USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
UAC2_CX_CLOCK_SELECTOR << 8, selector_id << 8,
UAC2_CX_CLOCK_SELECTOR << 8,
snd_usb_ctrl_intf(chip) | (selector_id << 8),
&buf, sizeof(buf), 1000);
if (ret < 0)
@ -120,7 +121,8 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id)
err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
UAC2_CS_CONTROL_CLOCK_VALID << 8, source_id << 8,
UAC2_CS_CONTROL_CLOCK_VALID << 8,
snd_usb_ctrl_intf(chip) | (source_id << 8),
&data, sizeof(data), 1000);
if (err < 0) {
@ -269,7 +271,8 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
data[3] = rate >> 24;
if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR,
USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT,
UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8,
UAC2_CS_CONTROL_SAM_FREQ << 8,
snd_usb_ctrl_intf(chip) | (clock << 8),
data, sizeof(data), 1000)) < 0) {
snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2)\n",
dev->devnum, iface, fmt->altsetting, rate);
@ -278,7 +281,8 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8,
UAC2_CS_CONTROL_SAM_FREQ << 8,
snd_usb_ctrl_intf(chip) | (clock << 8),
data, sizeof(data), 1000)) < 0) {
snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
dev->devnum, iface, fmt->altsetting);

1
sound/usb/endpoint.c
View file

@ -427,6 +427,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) {
kfree(fp->rate_table);
kfree(fp);
fp = NULL;
continue;
}

104
sound/usb/format.c
View file

@ -205,6 +205,60 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
return 0;
}
/*
* Helper function to walk the array of sample rate triplets reported by
* the device. The problem is that we need to parse whole array first to
* get to know how many sample rates we have to expect.
* Then fp->rate_table can be allocated and filled.
*/
static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets,
const unsigned char *data)
{
int i, nr_rates = 0;
fp->rates = fp->rate_min = fp->rate_max = 0;
for (i = 0; i < nr_triplets; i++) {
int min = combine_quad(&data[2 + 12 * i]);
int max = combine_quad(&data[6 + 12 * i]);
int res = combine_quad(&data[10 + 12 * i]);
int rate;
if ((max < 0) || (min < 0) || (res < 0) || (max < min))
continue;
/*
* for ranges with res == 1, we announce a continuous sample
* rate range, and this function should return 0 for no further
* parsing.
*/
if (res == 1) {
fp->rate_min = min;
fp->rate_max = max;
fp->rates = SNDRV_PCM_RATE_CONTINUOUS;
return 0;
}
for (rate = min; rate <= max; rate += res) {
if (fp->rate_table)
fp->rate_table[nr_rates] = rate;
if (!fp->rate_min || rate < fp->rate_min)
fp->rate_min = rate;
if (!fp->rate_max || rate > fp->rate_max)
fp->rate_max = rate;
fp->rates |= snd_pcm_rate_to_rate_bit(rate);
nr_rates++;
/* avoid endless loop */
if (res == 0)
break;
}
}
return nr_rates;
}
/*
* parse the format descriptor and stores the possible sample rates
* on the audioformat table (audio class v2).
@ -215,13 +269,20 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
{
struct usb_device *dev = chip->dev;
unsigned char tmp[2], *data;
int i, nr_rates, data_size, ret = 0;
int nr_triplets, data_size, ret = 0;
int clock = snd_usb_clock_find_source(chip, chip->ctrl_intf, fp->clock);
if (clock < 0) {
snd_printk(KERN_ERR "%s(): unable to find clock source (clock %d)\n",
__func__, clock);
goto err;
}
/* get the number of sample rates first by only fetching 2 bytes */
ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE,
USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8,
UAC2_CS_CONTROL_SAM_FREQ << 8,
snd_usb_ctrl_intf(chip) | (clock << 8),
tmp, sizeof(tmp), 1000);
if (ret < 0) {
@ -230,8 +291,8 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
goto err;
}
nr_rates = (tmp[1] << 8) | tmp[0];
data_size = 2 + 12 * nr_rates;
nr_triplets = (tmp[1] << 8) | tmp[0];
data_size = 2 + 12 * nr_triplets;
data = kzalloc(data_size, GFP_KERNEL);
if (!data) {
ret = -ENOMEM;
@ -241,7 +302,8 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
/* now get the full information */
ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE,
USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8,
UAC2_CS_CONTROL_SAM_FREQ << 8,
snd_usb_ctrl_intf(chip) | (clock << 8),
data, data_size, 1000);
if (ret < 0) {
@ -251,26 +313,28 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
goto err_free;
}
fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL);
/* Call the triplet parser, and make sure fp->rate_table is NULL.
* We just use the return value to know how many sample rates we
* will have to deal with. */
kfree(fp->rate_table);
fp->rate_table = NULL;
fp->nr_rates = parse_uac2_sample_rate_range(fp, nr_triplets, data);
if (fp->nr_rates == 0) {
/* SNDRV_PCM_RATE_CONTINUOUS */
ret = 0;
goto err_free;
}
fp->rate_table = kmalloc(sizeof(int) * fp->nr_rates, GFP_KERNEL);
if (!fp->rate_table) {
ret = -ENOMEM;
goto err_free;
}
fp->nr_rates = 0;
fp->rate_min = fp->rate_max = 0;
for (i = 0; i < nr_rates; i++) {
int rate = combine_quad(&data[2 + 12 * i]);
fp->rate_table[fp->nr_rates] = rate;
if (!fp->rate_min || rate < fp->rate_min)
fp->rate_min = rate;
if (!fp->rate_max || rate > fp->rate_max)
fp->rate_max = rate;
fp->rates |= snd_pcm_rate_to_rate_bit(rate);
fp->nr_rates++;
}
/* Call the triplet parser again, but this time, fp->rate_table is
* allocated, so the rates will be stored */
parse_uac2_sample_rate_range(fp, nr_triplets, data);
err_free:
kfree(data);

4
sound/usb/helper.h
View file

@ -28,5 +28,9 @@ unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip,
#define snd_usb_get_speed(dev) ((dev)->speed)
#endif
static inline int snd_usb_ctrl_intf(struct snd_usb_audio *chip)
{
return get_iface_desc(chip->ctrl_intf)->bInterfaceNumber;
}
#endif /* __USBAUDIO_HELPER_H */

19
sound/usb/mixer.c
View file

@ -297,20 +297,27 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v
static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret)
{
unsigned char buf[14]; /* enough space for one range of 4 bytes */
unsigned char buf[2 + 3*sizeof(__u16)]; /* enough space for one range */
unsigned char *val;
int ret;
int ret, size;
__u8 bRequest;
bRequest = (request == UAC_GET_CUR) ?
UAC2_CS_CUR : UAC2_CS_RANGE;
if (request == UAC_GET_CUR) {
bRequest = UAC2_CS_CUR;
size = sizeof(__u16);
} else {
bRequest = UAC2_CS_RANGE;
size = sizeof(buf);
}
memset(buf, 0, sizeof(buf));
ret = snd_usb_ctl_msg(cval->mixer->chip->dev,
usb_rcvctrlpipe(cval->mixer->chip->dev, 0),
bRequest,
USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
validx, cval->mixer->ctrlif | (cval->id << 8),
buf, sizeof(buf), 1000);
buf, size, 1000);
if (ret < 0) {
snd_printk(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n",
@ -318,6 +325,8 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v
return ret;
}
/* FIXME: how should we handle multiple triplets here? */
switch (request) {
case UAC_GET_CUR:
val = buf;