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authorThomas Gleixner <tglx@linutronix.de>2010-06-28 22:33:13 +0200
committerThomas Gleixner <tglx@linutronix.de>2010-06-28 22:33:24 +0200
commitf384c954c9fe3d3c6fce5ae66b67f2ddd947d098 (patch)
treea38541b8083a2304435e9a153d408bd7cd44116e /sound
parent9a15a07fe2175dc25cd928a354b3839f562ac8cc (diff)
parent5904b3b81d25166e5e39b9727645bb47937618e3 (diff)
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Merge branch 'linus' into perf/core
Reason: Further changes conflict with upstream fixes Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Diffstat (limited to 'sound')
-rw-r--r--sound/pci/asihpi/hpi6205.c22
-rw-r--r--sound/pci/hda/patch_realtek.c35
-rw-r--r--sound/soc/davinci/davinci-mcasp.c2
-rw-r--r--sound/soc/fsl/mpc5200_dma.h2
-rw-r--r--sound/usb/clock.c12
-rw-r--r--sound/usb/endpoint.c1
-rw-r--r--sound/usb/format.c104
-rw-r--r--sound/usb/helper.h4
-rw-r--r--sound/usb/mixer.c19
9 files changed, 141 insertions, 60 deletions
diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c
index e89991e..3b44134 100644
--- a/sound/pci/asihpi/hpi6205.c
+++ b/sound/pci/asihpi/hpi6205.c
@@ -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;
}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index fc767b6..f1ce7d7 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -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)},
{ }
};
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 79f0f4a..d395509 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -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);
}
}
diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h
index 22208b3..e1ec6d9 100644
--- a/sound/soc/fsl/mpc5200_dma.h
+++ b/sound/soc/fsl/mpc5200_dma.h
@@ -73,7 +73,7 @@ struct psc_dma {
};
/* Utility for retrieving psc_dma_stream structure from a substream */
-inline struct psc_dma_stream *
+static inline struct psc_dma_stream *
to_psc_dma_stream(struct snd_pcm_substream *substream, struct psc_dma *psc_dma)
{
if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index b7aadd6..b585511 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -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);
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 9593b91..6f6596c 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -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;
}
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 5367cd1..30364ab 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -206,6 +206,60 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
}
/*
+ * 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);
diff --git a/sound/usb/helper.h b/sound/usb/helper.h
index a6b0e51..09bd943 100644
--- a/sound/usb/helper.h
+++ b/sound/usb/helper.h
@@ -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 */
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index a060d00..6939d0f 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -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;