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: mixart: range checking proc file
ALSA: hda - Fix a wrong array range check in patch_realtek.c
ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream
ALSA: hda - Enable amplifiers on Acer Inspire 6530G
ASoC: Only do WM8994 bias off transition from standby
ASoC: Don't use DCS_DATAPATH_BUSY for WM hubs devices
ASoC: Don't do runtime wm_hubs DC servo updates if using offset correction
ASoC: Support second DC servo readback method for wm_hubs
ASoC: Avoid wraparound in wm_hubs DC servo correction
ALSA: echoaudio - Eliminate use after free
ALSA: i2c: cleanup: change parameter to pointer
ALSA: hda - Add MSI blacklist for Aopen MZ915-M
ASoC: OMAP: Fix capture pointer handling for OMAP1510 to work correctly with recent ALSA PCM code
ALSA: hda - Update document about MSI and interrupts
ALSA: hda: Fix 0 dB offset for Lenovo Thinkpad models using AD1981
ALSA: hda - Add missing printk argument in previous patch
ASoC: Fix passing platform_data to ac97 bus users and fix a leak
ALSA: hda - Fix ADC/MUX assignment of ALC269 codec
ALSA: hda - Fix invalid bit values passed to snd_hda_codec_amp_stereo()
ASoC: wm8994: playback => capture
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index f4dd3bf..98d14cb 100644
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -119,10 +119,18 @@
Interrupt Handling
~~~~~~~~~~~~~~~~~~
-In rare but some cases, the interrupt isn't properly handled as
-default. You would notice this by the DMA transfer error reported by
-ALSA PCM core, for example. Using MSI might help in such a case.
-Pass `enable_msi=1` option for enabling MSI.
+HD-audio driver uses MSI as default (if available) since 2.6.33
+kernel as MSI works better on some machines, and in general, it's
+better for performance. However, Nvidia controllers showed bad
+regressions with MSI (especially in a combination with AMD chipset),
+thus we disabled MSI for them.
+
+There seem also still other devices that don't work with MSI. If you
+see a regression wrt the sound quality (stuttering, etc) or a lock-up
+in the recent kernel, try to pass `enable_msi=0` option to disable
+MSI. If it works, you can add the known bad device to the blacklist
+defined in hda_intel.c. In such a case, please report and give the
+patch back to the upstream developer.
HD-AUDIO CODEC
diff --git a/include/sound/ak4113.h b/include/sound/ak4113.h
index 8988eda..2609048 100644
--- a/include/sound/ak4113.h
+++ b/include/sound/ak4113.h
@@ -307,7 +307,7 @@
int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read,
ak4113_write_t *write,
- const unsigned char pgm[AK4113_WRITABLE_REGS],
+ const unsigned char *pgm,
void *private_data, struct ak4113 **r_ak4113);
void snd_ak4113_reg_write(struct ak4113 *ak4113, unsigned char reg,
unsigned char mask, unsigned char val);
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 061f16d..0a0b019 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -219,7 +219,6 @@
struct snd_soc_codec *codec;
unsigned int active;
unsigned char pop_wait:1;
- void *dma_data;
/* DAI private data */
void *private_data;
@@ -230,4 +229,21 @@
struct list_head list;
};
+static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
+ const struct snd_pcm_substream *ss)
+{
+ return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ dai->playback.dma_data : dai->capture.dma_data;
+}
+
+static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
+ const struct snd_pcm_substream *ss,
+ void *data)
+{
+ if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dai->playback.dma_data = data;
+ else
+ dai->capture.dma_data = data;
+}
+
#endif
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 5d234a8..a57fbfc 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -375,6 +375,7 @@
unsigned int channels_min; /* min channels */
unsigned int channels_max; /* max channels */
unsigned int active:1; /* stream is in use */
+ void *dma_data; /* used by platform code */
};
/* SoC audio ops */
diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c
index fff62cc..971a84a 100644
--- a/sound/i2c/other/ak4113.c
+++ b/sound/i2c/other/ak4113.c
@@ -70,7 +70,7 @@
}
int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read,
- ak4113_write_t *write, const unsigned char pgm[5],
+ ak4113_write_t *write, const unsigned char *pgm,
void *private_data, struct ak4113 **r_ak4113)
{
struct ak4113 *chip;
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 8dab82d..668a5ec 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -2184,10 +2184,9 @@
goto ctl_error;
#endif
- if ((err = snd_card_register(card)) < 0) {
- snd_card_free(card);
+ err = snd_card_register(card);
+ if (err < 0)
goto ctl_error;
- }
snd_printk(KERN_INFO "Card registered: %s\n", card->longname);
pci_set_drvdata(pci, chip);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 4bb9067..f8fd586 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2362,6 +2362,7 @@
SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */
SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */
SND_PCI_QUIRK(0x1849, 0x0888, "ASRock", 0), /* Athlon64 X2 + nvidia */
+ SND_PCI_QUIRK(0xa0a0, 0x0575, "Aopen MZ915-M", 0), /* ICH6 */
{}
};
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index e6d1bdf..af34606 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -1896,6 +1896,14 @@
case AD1981_THINKPAD:
spec->mixers[0] = ad1981_thinkpad_mixers;
spec->input_mux = &ad1981_thinkpad_capture_source;
+ /* set the upper-limit for mixer amp to 0dB for avoiding the
+ * possible damage by overloading
+ */
+ snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT,
+ (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
+ (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+ (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+ (1 << AC_AMPCAP_MUTE_SHIFT));
break;
case AD1981_TOSHIBA:
spec->mixers[0] = ad1981_hp_mixers;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9a23444..c7730db 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1621,6 +1621,11 @@
*/
static struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
+/* Route to built-in subwoofer as well as speakers */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* Bias voltage on for external mic port */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80},
/* Front Mic: set to PIN_IN (empty by default) */
@@ -1632,10 +1637,12 @@
/* Enable speaker output */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
/* Enable headphone output */
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
{ }
};
@@ -4984,6 +4991,70 @@
}
}
+/* fill adc_nids (and capsrc_nids) containing all active input pins */
+static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids,
+ int num_nids)
+{
+ struct alc_spec *spec = codec->spec;
+ int n;
+ hda_nid_t fallback_adc = 0, fallback_cap = 0;
+
+ for (n = 0; n < num_nids; n++) {
+ hda_nid_t adc, cap;
+ hda_nid_t conn[HDA_MAX_NUM_INPUTS];
+ int nconns, i, j;
+
+ adc = nids[n];
+ if (get_wcaps_type(get_wcaps(codec, adc)) != AC_WID_AUD_IN)
+ continue;
+ cap = adc;
+ nconns = snd_hda_get_connections(codec, cap, conn,
+ ARRAY_SIZE(conn));
+ if (nconns == 1) {
+ cap = conn[0];
+ nconns = snd_hda_get_connections(codec, cap, conn,
+ ARRAY_SIZE(conn));
+ }
+ if (nconns <= 0)
+ continue;
+ if (!fallback_adc) {
+ fallback_adc = adc;
+ fallback_cap = cap;
+ }
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ hda_nid_t nid = spec->autocfg.input_pins[i];
+ if (!nid)
+ continue;
+ for (j = 0; j < nconns; j++) {
+ if (conn[j] == nid)
+ break;
+ }
+ if (j >= nconns)
+ break;
+ }
+ if (i >= AUTO_PIN_LAST) {
+ int num_adcs = spec->num_adc_nids;
+ spec->private_adc_nids[num_adcs] = adc;
+ spec->private_capsrc_nids[num_adcs] = cap;
+ spec->num_adc_nids++;
+ spec->adc_nids = spec->private_adc_nids;
+ if (adc != cap)
+ spec->capsrc_nids = spec->private_capsrc_nids;
+ }
+ }
+ if (!spec->num_adc_nids) {
+ printk(KERN_WARNING "hda_codec: %s: no valid ADC found;"
+ " using fallback 0x%x\n",
+ codec->chip_name, fallback_adc);
+ spec->private_adc_nids[0] = fallback_adc;
+ spec->adc_nids = spec->private_adc_nids;
+ if (fallback_adc != fallback_cap) {
+ spec->private_capsrc_nids[0] = fallback_cap;
+ spec->capsrc_nids = spec->private_adc_nids;
+ }
+ }
+}
+
#ifdef CONFIG_SND_HDA_INPUT_BEEP
#define set_beep_amp(spec, nid, idx, dir) \
((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir))
@@ -8398,9 +8469,7 @@
static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 2, 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("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
@@ -10041,13 +10110,12 @@
int idx;
alc_set_pin_output(codec, nid, pin_type);
+ if (dac_idx >= spec->multiout.num_dacs)
+ return;
if (spec->multiout.dac_nids[dac_idx] == 0x25)
idx = 4;
- else {
- if (spec->multiout.num_dacs >= dac_idx)
- return;
+ else
idx = spec->multiout.dac_nids[dac_idx] - 2;
- }
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
}
@@ -12459,11 +12527,11 @@
unsigned char bits;
present = snd_hda_jack_detect(codec, 0x15);
- bits = present ? AMP_IN_MUTE(0) : 0;
+ bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 0,
- AMP_IN_MUTE(0), bits);
+ HDA_AMP_MUTE, bits);
snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 1,
- AMP_IN_MUTE(0), bits);
+ HDA_AMP_MUTE, bits);
}
static void alc268_acer_lc_unsol_event(struct hda_codec *codec,
@@ -13333,9 +13401,9 @@
0x22,
};
-/* NOTE: ADC2 (0x07) is connected from a recording *MIXER* (0x24),
- * not a mux!
- */
+static hda_nid_t alc269_adc_candidates[] = {
+ 0x08, 0x09, 0x07,
+};
#define alc269_modes alc260_modes
#define alc269_capture_source alc880_lg_lw_capture_source
@@ -13482,11 +13550,11 @@
unsigned char bits;
present = snd_hda_jack_detect(codec, 0x15);
- bits = present ? AMP_IN_MUTE(0) : 0;
+ bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
- AMP_IN_MUTE(0), bits);
+ HDA_AMP_MUTE, bits);
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
- AMP_IN_MUTE(0), bits);
+ HDA_AMP_MUTE, bits);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_COEF_INDEX, 0x0c);
@@ -13511,11 +13579,11 @@
/* Check port replicator headphone socket */
present |= snd_hda_jack_detect(codec, 0x1a);
- bits = present ? AMP_IN_MUTE(0) : 0;
+ bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
- AMP_IN_MUTE(0), bits);
+ HDA_AMP_MUTE, bits);
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
- AMP_IN_MUTE(0), bits);
+ HDA_AMP_MUTE, bits);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_COEF_INDEX, 0x0c);
@@ -13646,11 +13714,11 @@
unsigned char bits;
present = snd_hda_jack_detect(codec, nid);
- bits = present ? AMP_IN_MUTE(0) : 0;
+ bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
- AMP_IN_MUTE(0), bits);
+ HDA_AMP_MUTE, bits);
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
- AMP_IN_MUTE(0), bits);
+ HDA_AMP_MUTE, bits);
}
/* unsolicited event for HP jack sensing */
@@ -13842,7 +13910,6 @@
struct alc_spec *spec = codec->spec;
int err;
static hda_nid_t alc269_ignore[] = { 0x1d, 0 };
- hda_nid_t real_capsrc_nids;
err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
alc269_ignore);
@@ -13866,18 +13933,19 @@
if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010) {
add_verb(spec, alc269vb_init_verbs);
- real_capsrc_nids = alc269vb_capsrc_nids[0];
alc_ssid_check(codec, 0, 0x1b, 0x14, 0x21);
} else {
add_verb(spec, alc269_init_verbs);
- real_capsrc_nids = alc269_capsrc_nids[0];
alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0);
}
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux[0];
+ fillup_priv_adc_nids(codec, alc269_adc_candidates,
+ sizeof(alc269_adc_candidates));
+
/* set default input source */
- snd_hda_codec_write_cache(codec, real_capsrc_nids,
+ snd_hda_codec_write_cache(codec, spec->capsrc_nids[0],
0, AC_VERB_SET_CONNECT_SEL,
spec->input_mux->items[0].index);
@@ -14156,14 +14224,16 @@
spec->stream_digital_playback = &alc269_pcm_digital_playback;
spec->stream_digital_capture = &alc269_pcm_digital_capture;
- if (!is_alc269vb) {
- spec->adc_nids = alc269_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
- spec->capsrc_nids = alc269_capsrc_nids;
- } else {
- spec->adc_nids = alc269vb_adc_nids;
- spec->num_adc_nids = ARRAY_SIZE(alc269vb_adc_nids);
- spec->capsrc_nids = alc269vb_capsrc_nids;
+ if (!spec->adc_nids) { /* wasn't filled automatically? use default */
+ if (!is_alc269vb) {
+ spec->adc_nids = alc269_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
+ spec->capsrc_nids = alc269_capsrc_nids;
+ } else {
+ spec->adc_nids = alc269vb_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(alc269vb_adc_nids);
+ spec->capsrc_nids = alc269vb_capsrc_nids;
+ }
}
if (!spec->cap_mixer)
@@ -17115,9 +17185,9 @@
present = snd_hda_jack_detect(codec, 0x21);
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
- AMP_IN_MUTE(0), bits);
+ HDA_AMP_MUTE, bits);
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
- AMP_IN_MUTE(0), bits);
+ HDA_AMP_MUTE, bits);
}
static void alc663_21jd_two_speaker_automute(struct hda_codec *codec)
@@ -17128,13 +17198,13 @@
present = snd_hda_jack_detect(codec, 0x21);
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
- AMP_IN_MUTE(0), bits);
+ HDA_AMP_MUTE, bits);
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
- AMP_IN_MUTE(0), bits);
+ HDA_AMP_MUTE, bits);
snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0,
- AMP_IN_MUTE(0), bits);
+ HDA_AMP_MUTE, bits);
snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1,
- AMP_IN_MUTE(0), bits);
+ HDA_AMP_MUTE, bits);
}
static void alc663_15jd_two_speaker_automute(struct hda_codec *codec)
@@ -17145,13 +17215,13 @@
present = snd_hda_jack_detect(codec, 0x15);
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
- AMP_IN_MUTE(0), bits);
+ HDA_AMP_MUTE, bits);
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
- AMP_IN_MUTE(0), bits);
+ HDA_AMP_MUTE, bits);
snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0,
- AMP_IN_MUTE(0), bits);
+ HDA_AMP_MUTE, bits);
snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1,
- AMP_IN_MUTE(0), bits);
+ HDA_AMP_MUTE, bits);
}
static void alc662_f5z_speaker_automute(struct hda_codec *codec)
@@ -17190,14 +17260,14 @@
if (present1 || present2) {
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
- AMP_IN_MUTE(0), AMP_IN_MUTE(0));
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
- AMP_IN_MUTE(0), AMP_IN_MUTE(0));
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
} else {
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
- AMP_IN_MUTE(0), 0);
+ HDA_AMP_MUTE, 0);
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
- AMP_IN_MUTE(0), 0);
+ HDA_AMP_MUTE, 0);
}
}
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index 55e9315..3be8f97 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -1162,13 +1162,15 @@
unsigned long count, unsigned long pos)
{
struct mixart_mgr *mgr = entry->private_data;
+ unsigned long maxsize;
- count = count & ~3; /* make sure the read size is a multiple of 4 bytes */
- if(count <= 0)
+ if (pos >= MIXART_BA0_SIZE)
return 0;
- if(pos + count > MIXART_BA0_SIZE)
- count = (long)(MIXART_BA0_SIZE - pos);
- if(copy_to_user_fromio(buf, MIXART_MEM( mgr, pos ), count))
+ maxsize = MIXART_BA0_SIZE - pos;
+ if (count > maxsize)
+ count = maxsize;
+ count = count & ~3; /* make sure the read size is a multiple of 4 bytes */
+ if (copy_to_user_fromio(buf, MIXART_MEM(mgr, pos), count))
return -EFAULT;
return count;
}
@@ -1181,13 +1183,15 @@
unsigned long count, unsigned long pos)
{
struct mixart_mgr *mgr = entry->private_data;
+ unsigned long maxsize;
- count = count & ~3; /* make sure the read size is a multiple of 4 bytes */
- if(count <= 0)
+ if (pos > MIXART_BA1_SIZE)
return 0;
- if(pos + count > MIXART_BA1_SIZE)
- count = (long)(MIXART_BA1_SIZE - pos);
- if(copy_to_user_fromio(buf, MIXART_REG( mgr, pos ), count))
+ maxsize = MIXART_BA1_SIZE - pos;
+ if (count > maxsize)
+ count = maxsize;
+ count = count & ~3; /* make sure the read size is a multiple of 4 bytes */
+ if (copy_to_user_fromio(buf, MIXART_REG(mgr, pos), count))
return -EFAULT;
return count;
}
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index 9ef6b96..3e6628c8 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -180,7 +180,7 @@
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
runtime->dma_bytes = params_buffer_bytes(params);
- prtd->params = rtd->dai->cpu_dai->dma_data;
+ prtd->params = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
prtd->params->dma_intr_handler = atmel_pcm_dma_irq;
prtd->dma_buffer = runtime->dma_addr;
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index e588e63..0b59806 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -363,12 +363,12 @@
ssc_p->dma_params[dir] = dma_params;
/*
- * The cpu_dai->dma_data field is only used to communicate the
- * appropriate DMA parameters to the pcm driver hw_params()
+ * The snd_soc_pcm_stream->dma_data field is only used to communicate
+ * the appropriate DMA parameters to the pcm driver hw_params()
* function. It should not be used for other purposes
* as it is common to all substreams.
*/
- rtd->dai->cpu_dai->dma_data = dma_params;
+ snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_params);
channels = params_channels(params);
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index fd101d4..1f5e57a 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -81,9 +81,11 @@
static int ac97_soc_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_card *card = socdev->card;
struct snd_soc_codec *codec;
struct snd_ac97_bus *ac97_bus;
struct snd_ac97_template ac97_template;
+ int i;
int ret = 0;
printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION);
@@ -103,12 +105,6 @@
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
- ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
- if (ret < 0) {
- printk(KERN_ERR "ASoC: failed to init gen ac97 glue\n");
- goto err;
- }
-
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0)
@@ -124,6 +120,13 @@
if (ret < 0)
goto bus_err;
+ for (i = 0; i < card->num_links; i++) {
+ if (card->dai_link[i].codec_dai->ac97_control) {
+ snd_ac97_dev_add_pdata(codec->ac97,
+ card->dai_link[i].cpu_dai->ac97_pdata);
+ }
+ }
+
return 0;
bus_err:
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 8d1c637..9da0724 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3008,34 +3008,39 @@
break;
case SND_SOC_BIAS_OFF:
- /* Switch over to startup biases */
- snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
- WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA |
- WM8994_VMID_BUF_ENA |
- WM8994_VMID_RAMP_MASK,
- WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA |
- WM8994_VMID_BUF_ENA |
- (1 << WM8994_VMID_RAMP_SHIFT));
+ if (codec->bias_level == SND_SOC_BIAS_STANDBY) {
+ /* Switch over to startup biases */
+ snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
+ WM8994_BIAS_SRC |
+ WM8994_STARTUP_BIAS_ENA |
+ WM8994_VMID_BUF_ENA |
+ WM8994_VMID_RAMP_MASK,
+ WM8994_BIAS_SRC |
+ WM8994_STARTUP_BIAS_ENA |
+ WM8994_VMID_BUF_ENA |
+ (1 << WM8994_VMID_RAMP_SHIFT));
- /* Disable main biases */
- snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
- WM8994_BIAS_ENA | WM8994_VMID_SEL_MASK, 0);
+ /* Disable main biases */
+ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
+ WM8994_BIAS_ENA |
+ WM8994_VMID_SEL_MASK, 0);
- /* Discharge line */
- snd_soc_update_bits(codec, WM8994_ANTIPOP_1,
- WM8994_LINEOUT1_DISCH |
- WM8994_LINEOUT2_DISCH,
- WM8994_LINEOUT1_DISCH |
- WM8994_LINEOUT2_DISCH);
+ /* Discharge line */
+ snd_soc_update_bits(codec, WM8994_ANTIPOP_1,
+ WM8994_LINEOUT1_DISCH |
+ WM8994_LINEOUT2_DISCH,
+ WM8994_LINEOUT1_DISCH |
+ WM8994_LINEOUT2_DISCH);
- msleep(5);
+ msleep(5);
- /* Switch off startup biases */
- snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
- WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA |
- WM8994_VMID_BUF_ENA |
- WM8994_VMID_RAMP_MASK, 0);
-
+ /* Switch off startup biases */
+ snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
+ WM8994_BIAS_SRC |
+ WM8994_STARTUP_BIAS_ENA |
+ WM8994_VMID_BUF_ENA |
+ WM8994_VMID_RAMP_MASK, 0);
+ }
break;
}
codec->bias_level = level;
@@ -3402,7 +3407,7 @@
.rates = WM8994_RATES,
.formats = WM8994_FORMATS,
},
- .playback = {
+ .capture = {
.stream_name = "AIF3 Capture",
.channels_min = 2,
.channels_max = 2,
@@ -3731,11 +3736,12 @@
case 3:
wm8994->hubs.dcs_codes = -5;
wm8994->hubs.hp_startup_mode = 1;
+ wm8994->hubs.dcs_readback_mode = 1;
break;
default:
+ wm8994->hubs.dcs_readback_mode = 1;
break;
}
-
/* Remember if AIFnLRCLK is configured as a GPIO. This should be
* configured on init - if a system wants to do this dynamically
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 486bdd2..e1f225a 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -62,21 +62,27 @@
static const struct soc_enum speaker_mode =
SOC_ENUM_SINGLE(WM8993_SPKMIXR_ATTENUATION, 8, 2, speaker_mode_text);
-static void wait_for_dc_servo(struct snd_soc_codec *codec)
+static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op)
{
unsigned int reg;
int count = 0;
+ unsigned int val;
+
+ val = op | WM8993_DCS_ENA_CHAN_0 | WM8993_DCS_ENA_CHAN_1;
+
+ /* Trigger the command */
+ snd_soc_write(codec, WM8993_DC_SERVO_0, val);
dev_dbg(codec->dev, "Waiting for DC servo...\n");
do {
count++;
msleep(1);
- reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_0);
+ reg = snd_soc_read(codec, WM8993_DC_SERVO_0);
dev_dbg(codec->dev, "DC servo: %x\n", reg);
- } while (reg & WM8993_DCS_DATAPATH_BUSY && count < 400);
+ } while (reg & op && count < 400);
- if (reg & WM8993_DCS_DATAPATH_BUSY)
+ if (reg & op)
dev_err(codec->dev, "Timed out waiting for DC Servo\n");
}
@@ -86,51 +92,58 @@
static void calibrate_dc_servo(struct snd_soc_codec *codec)
{
struct wm_hubs_data *hubs = codec->private_data;
- u16 reg, dcs_cfg;
+ u16 reg, reg_l, reg_r, dcs_cfg;
/* Set for 32 series updates */
snd_soc_update_bits(codec, WM8993_DC_SERVO_1,
WM8993_DCS_SERIES_NO_01_MASK,
32 << WM8993_DCS_SERIES_NO_01_SHIFT);
-
- /* Enable the DC servo. Write all bits to avoid triggering startup
- * or write calibration.
- */
- snd_soc_update_bits(codec, WM8993_DC_SERVO_0,
- 0xFFFF,
- WM8993_DCS_ENA_CHAN_0 |
- WM8993_DCS_ENA_CHAN_1 |
- WM8993_DCS_TRIG_SERIES_1 |
- WM8993_DCS_TRIG_SERIES_0);
-
- wait_for_dc_servo(codec);
+ wait_for_dc_servo(codec,
+ WM8993_DCS_TRIG_SERIES_0 | WM8993_DCS_TRIG_SERIES_1);
/* Apply correction to DC servo result */
if (hubs->dcs_codes) {
dev_dbg(codec->dev, "Applying %d code DC servo correction\n",
hubs->dcs_codes);
+ /* Different chips in the family support different
+ * readback methods.
+ */
+ switch (hubs->dcs_readback_mode) {
+ case 0:
+ reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1)
+ & WM8993_DCS_INTEG_CHAN_0_MASK;;
+ reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2)
+ & WM8993_DCS_INTEG_CHAN_1_MASK;
+ break;
+ case 1:
+ reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
+ reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
+ >> WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+ reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
+ break;
+ default:
+ WARN(1, "Unknown DCS readback method");
+ break;
+ }
+
/* HPOUT1L */
- reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) &
- WM8993_DCS_INTEG_CHAN_0_MASK;;
- reg += hubs->dcs_codes;
- dcs_cfg = reg << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+ if (reg_l + hubs->dcs_codes > 0 &&
+ reg_l + hubs->dcs_codes < 0xff)
+ reg_l += hubs->dcs_codes;
+ dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
/* HPOUT1R */
- reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) &
- WM8993_DCS_INTEG_CHAN_1_MASK;
- reg += hubs->dcs_codes;
- dcs_cfg |= reg;
+ if (reg_r + hubs->dcs_codes > 0 &&
+ reg_r + hubs->dcs_codes < 0xff)
+ reg_r += hubs->dcs_codes;
+ dcs_cfg |= reg_r;
/* Do it */
snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg);
- snd_soc_update_bits(codec, WM8993_DC_SERVO_0,
- WM8993_DCS_TRIG_DAC_WR_0 |
- WM8993_DCS_TRIG_DAC_WR_1,
- WM8993_DCS_TRIG_DAC_WR_0 |
- WM8993_DCS_TRIG_DAC_WR_1);
-
- wait_for_dc_servo(codec);
+ wait_for_dc_servo(codec,
+ WM8993_DCS_TRIG_DAC_WR_0 |
+ WM8993_DCS_TRIG_DAC_WR_1);
}
}
@@ -141,10 +154,16 @@
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wm_hubs_data *hubs = codec->private_data;
int ret;
ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
+ /* If we're applying an offset correction then updating the
+ * callibration would be likely to introduce further offsets. */
+ if (hubs->dcs_codes)
+ return ret;
+
/* Only need to do this if the outputs are active */
if (snd_soc_read(codec, WM8993_POWER_MANAGEMENT_1)
& (WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA))
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index 420104f..e51c166 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -21,6 +21,7 @@
/* This *must* be the first element of the codec->private_data struct */
struct wm_hubs_data {
int dcs_codes;
+ int dcs_readback_mode;
int hp_startup_mode;
};
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 62af7e0..adadcd3 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -586,7 +586,8 @@
dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start;
davinci_i2s_dai.private_data = dev;
- davinci_i2s_dai.dma_data = dev->dma_params;
+ davinci_i2s_dai.capture.dma_data = dev->dma_params;
+ davinci_i2s_dai.playback.dma_data = dev->dma_params;
ret = snd_soc_register_dai(&davinci_i2s_dai);
if (ret != 0)
goto err_free_mem;
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 6c80cc3..79f0f4a 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -918,7 +918,8 @@
dma_data->channel = res->start;
davinci_mcasp_dai[pdata->op_mode].private_data = dev;
- davinci_mcasp_dai[pdata->op_mode].dma_data = dev->dma_params;
+ davinci_mcasp_dai[pdata->op_mode].capture.dma_data = dev->dma_params;
+ davinci_mcasp_dai[pdata->op_mode].playback.dma_data = dev->dma_params;
davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev;
ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]);
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 80c7fdf..2dc406f 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -649,8 +649,10 @@
struct snd_pcm_hardware *ppcm;
int ret = 0;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->dma_data;
+ struct davinci_pcm_dma_params *pa;
struct davinci_pcm_dma_params *params;
+
+ pa = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
if (!pa)
return -ENODEV;
params = &pa[substream->stream];
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index 86668ab..2e79d71 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -84,11 +84,13 @@
static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data;
+ struct imx_pcm_dma_params *dma_params;
struct snd_pcm_runtime *runtime = substream->runtime;
struct imx_pcm_runtime_data *iprtd = runtime->private_data;
int ret;
+ dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream);
+
iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH);
if (iprtd->dma < 0) {
pr_err("Failed to claim the audio DMA\n");
@@ -193,10 +195,12 @@
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data;
+ struct imx_pcm_dma_params *dma_params;
struct imx_pcm_runtime_data *iprtd = runtime->private_data;
int err;
+ dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream);
+
iprtd->substream = substream;
iprtd->buf = (unsigned int *)substream->dma_buffer.area;
iprtd->period_cnt = 0;
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 6546b06..0bcc6d7 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -235,17 +235,20 @@
struct snd_soc_dai *cpu_dai)
{
struct imx_ssi *ssi = cpu_dai->private_data;
+ struct imx_pcm_dma_params *dma_data;
u32 reg, sccr;
/* Tx/Rx config */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
reg = SSI_STCCR;
- cpu_dai->dma_data = &ssi->dma_params_tx;
+ dma_data = &ssi->dma_params_tx;
} else {
reg = SSI_SRCCR;
- cpu_dai->dma_data = &ssi->dma_params_rx;
+ dma_data = &ssi->dma_params_rx;
}
+ snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
+
sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK;
/* DAI data (word) size */
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index e814a95..8ad9dc9 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -297,7 +297,9 @@
omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode;
omap_mcbsp_dai_dma_params[id][substream->stream].data_type =
OMAP_DMA_DATA_TYPE_S16;
- cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream];
+
+ snd_soc_dai_set_dma_data(cpu_dai, substream,
+ &omap_mcbsp_dai_dma_params[id][substream->stream]);
if (mcbsp_data->configured) {
/* McBSP already configured by another stream */
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 25f19e4..b7f4f7e 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -150,7 +150,8 @@
int stream = substream->stream;
int channels, err, link_mask = 0;
- cpu_dai->dma_data = &omap_mcpdm_dai_dma_params[stream];
+ snd_soc_dai_set_dma_data(cpu_dai, substream,
+ &omap_mcpdm_dai_dma_params[stream]);
channels = params_channels(params);
switch (channels) {
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index ba8acbb..1e52190 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -61,12 +61,11 @@
struct omap_runtime_data *prtd = runtime->private_data;
unsigned long flags;
- if ((cpu_is_omap1510()) &&
- (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) {
+ if ((cpu_is_omap1510())) {
/*
* OMAP1510 doesn't fully support DMA progress counter
* and there is no software emulation implemented yet,
- * so have to maintain our own playback progress counter
+ * so have to maintain our own progress counters
* that can be used by omap_pcm_pointer() instead.
*/
spin_lock_irqsave(&prtd->lock, flags);
@@ -101,9 +100,11 @@
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct omap_runtime_data *prtd = runtime->private_data;
- struct omap_pcm_dma_data *dma_data = rtd->dai->cpu_dai->dma_data;
+ struct omap_pcm_dma_data *dma_data;
int err = 0;
+ dma_data = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
+
/* return if this is a bufferless transfer e.g.
* codec <--> BT codec or GSM modem -- lg FIXME */
if (!dma_data)
@@ -190,8 +191,7 @@
dma_params.frame_count = runtime->periods;
omap_set_dma_params(prtd->dma_ch, &dma_params);
- if ((cpu_is_omap1510()) &&
- (substream->stream == SNDRV_PCM_STREAM_PLAYBACK))
+ if ((cpu_is_omap1510()))
omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ |
OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ);
else
@@ -249,14 +249,15 @@
dma_addr_t ptr;
snd_pcm_uframes_t offset;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ if (cpu_is_omap1510()) {
+ offset = prtd->period_index * runtime->period_size;
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
ptr = omap_get_dma_dst_pos(prtd->dma_ch);
offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
- } else if (!(cpu_is_omap1510())) {
+ } else {
ptr = omap_get_dma_src_pos(prtd->dma_ch);
offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
- } else
- offset = prtd->period_index * runtime->period_size;
+ }
if (offset >= runtime->buffer_size)
offset = 0;
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index d5fc52d..544fd95 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -122,10 +122,9 @@
ssp_disable(ssp);
}
- if (cpu_dai->dma_data) {
- kfree(cpu_dai->dma_data);
- cpu_dai->dma_data = NULL;
- }
+ kfree(snd_soc_dai_get_dma_data(cpu_dai, substream));
+ snd_soc_dai_set_dma_data(cpu_dai, substream, NULL);
+
return ret;
}
@@ -142,10 +141,8 @@
clk_disable(ssp->clk);
}
- if (cpu_dai->dma_data) {
- kfree(cpu_dai->dma_data);
- cpu_dai->dma_data = NULL;
- }
+ kfree(snd_soc_dai_get_dma_data(cpu_dai, substream));
+ snd_soc_dai_set_dma_data(cpu_dai, substream, NULL);
}
#ifdef CONFIG_PM
@@ -570,19 +567,23 @@
u32 sspsp;
int width = snd_pcm_format_physical_width(params_format(params));
int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf;
+ struct pxa2xx_pcm_dma_params *dma_data;
+
+ dma_data = snd_soc_dai_get_dma_data(dai, substream);
/* generate correct DMA params */
- if (cpu_dai->dma_data)
- kfree(cpu_dai->dma_data);
+ kfree(dma_data);
/* Network mode with one active slot (ttsa == 1) can be used
* to force 16-bit frame width on the wire (for S16_LE), even
* with two channels. Use 16-bit DMA transfers for this case.
*/
- cpu_dai->dma_data = ssp_get_dma_params(ssp,
+ dma_data = ssp_get_dma_params(ssp,
((chn == 2) && (ttsa != 1)) || (width == 32),
substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+ snd_soc_dai_set_dma_data(dai, substream, dma_data);
+
/* we can only change the settings if the port is not in use */
if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE)
return 0;
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index e9ae7b3..d314115 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -122,11 +122,14 @@
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct pxa2xx_pcm_dma_params *dma_data;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out;
+ dma_data = &pxa2xx_ac97_pcm_stereo_out;
else
- cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_in;
+ dma_data = &pxa2xx_ac97_pcm_stereo_in;
+
+ snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
return 0;
}
@@ -137,11 +140,14 @@
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct pxa2xx_pcm_dma_params *dma_data;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
+ dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
else
- cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_in;
+ dma_data = &pxa2xx_ac97_pcm_aux_mono_in;
+
+ snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
return 0;
}
@@ -156,7 +162,8 @@
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
return -ENODEV;
else
- cpu_dai->dma_data = &pxa2xx_ac97_pcm_mic_mono_in;
+ snd_soc_dai_set_dma_data(cpu_dai, substream,
+ &pxa2xx_ac97_pcm_mic_mono_in);
return 0;
}
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 6b8f655..c1a52757 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -164,6 +164,7 @@
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct pxa2xx_pcm_dma_params *dma_data;
BUG_ON(IS_ERR(clk_i2s));
clk_enable(clk_i2s);
@@ -171,9 +172,11 @@
pxa_i2s_wait();
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_out;
+ dma_data = &pxa2xx_i2s_pcm_stereo_out;
else
- cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_in;
+ dma_data = &pxa2xx_i2s_pcm_stereo_in;
+
+ snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
/* is port used by another stream */
if (!(SACR0 & SACR0_ENB)) {
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index d38e395..adc7e6f 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -25,9 +25,11 @@
struct snd_pcm_runtime *runtime = substream->runtime;
struct pxa2xx_runtime_data *prtd = runtime->private_data;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct pxa2xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data;
+ struct pxa2xx_pcm_dma_params *dma;
int ret;
+ dma = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
+
/* return if this is a bufferless transfer e.g.
* codec <--> BT codec or GSM modem -- lg FIXME */
if (!dma)
diff --git a/sound/soc/s3c24xx/s3c-ac97.c b/sound/soc/s3c24xx/s3c-ac97.c
index ee8ed9d7..ecf4fd0 100644
--- a/sound/soc/s3c24xx/s3c-ac97.c
+++ b/sound/soc/s3c24xx/s3c-ac97.c
@@ -224,11 +224,14 @@
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct s3c_dma_params *dma_data;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- cpu_dai->dma_data = &s3c_ac97_pcm_out;
+ dma_data = &s3c_ac97_pcm_out;
else
- cpu_dai->dma_data = &s3c_ac97_pcm_in;
+ dma_data = &s3c_ac97_pcm_in;
+
+ snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
return 0;
}
@@ -238,8 +241,8 @@
{
u32 ac_glbctrl;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- int channel = ((struct s3c_dma_params *)
- rtd->dai->cpu_dai->dma_data)->channel;
+ struct s3c_dma_params *dma_data =
+ snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL);
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
@@ -265,7 +268,7 @@
writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL);
- s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+ s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED);
return 0;
}
@@ -280,7 +283,7 @@
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
return -ENODEV;
else
- cpu_dai->dma_data = &s3c_ac97_mic_in;
+ snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in);
return 0;
}
@@ -290,8 +293,8 @@
{
u32 ac_glbctrl;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- int channel = ((struct s3c_dma_params *)
- rtd->dai->cpu_dai->dma_data)->channel;
+ struct s3c_dma_params *dma_data =
+ snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL);
ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK;
@@ -311,7 +314,7 @@
writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL);
- s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+ s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED);
return 0;
}
diff --git a/sound/soc/s3c24xx/s3c-dma.c b/sound/soc/s3c24xx/s3c-dma.c
index 7725e26..1b61c23 100644
--- a/sound/soc/s3c24xx/s3c-dma.c
+++ b/sound/soc/s3c24xx/s3c-dma.c
@@ -145,10 +145,12 @@
struct snd_pcm_runtime *runtime = substream->runtime;
struct s3c24xx_runtime_data *prtd = runtime->private_data;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct s3c_dma_params *dma = rtd->dai->cpu_dai->dma_data;
unsigned long totbytes = params_buffer_bytes(params);
+ struct s3c_dma_params *dma =
+ snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
int ret = 0;
+
pr_debug("Entered %s\n", __func__);
/* return if this is a bufferless transfer e.g.
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index e994d83..8851594 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -339,14 +339,17 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai_link *dai = rtd->dai;
struct s3c_i2sv2_info *i2s = to_info(dai->cpu_dai);
+ struct s3c_dma_params *dma_data;
u32 iismod;
pr_debug("Entered %s\n", __func__);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- dai->cpu_dai->dma_data = i2s->dma_playback;
+ dma_data = i2s->dma_playback;
else
- dai->cpu_dai->dma_data = i2s->dma_capture;
+ dma_data = i2s->dma_capture;
+
+ snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data);
/* Working copies of register */
iismod = readl(i2s->regs + S3C2412_IISMOD);
@@ -394,8 +397,8 @@
int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
unsigned long irqs;
int ret = 0;
- int channel = ((struct s3c_dma_params *)
- rtd->dai->cpu_dai->dma_data)->channel;
+ struct s3c_dma_params *dma_data =
+ snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
pr_debug("Entered %s\n", __func__);
@@ -431,7 +434,7 @@
* of the auto reload mechanism of S3C24XX.
* This call won't bother S3C64XX.
*/
- s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+ s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED);
break;
diff --git a/sound/soc/s3c24xx/s3c-pcm.c b/sound/soc/s3c24xx/s3c-pcm.c
index a98f40c..326f0a9 100644
--- a/sound/soc/s3c24xx/s3c-pcm.c
+++ b/sound/soc/s3c24xx/s3c-pcm.c
@@ -178,6 +178,7 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai_link *dai = rtd->dai;
struct s3c_pcm_info *pcm = to_info(dai->cpu_dai);
+ struct s3c_dma_params *dma_data;
void __iomem *regs = pcm->regs;
struct clk *clk;
int sclk_div, sync_div;
@@ -187,9 +188,11 @@
dev_dbg(pcm->dev, "Entered %s\n", __func__);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- dai->cpu_dai->dma_data = pcm->dma_playback;
+ dma_data = pcm->dma_playback;
else
- dai->cpu_dai->dma_data = pcm->dma_capture;
+ dma_data = pcm->dma_capture;
+
+ snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data);
/* Strictly check for sample size */
switch (params_format(params)) {
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 0bc5950..c3ac890 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -242,14 +242,17 @@
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct s3c_dma_params *dma_data;
u32 iismod;
pr_debug("Entered %s\n", __func__);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out;
+ dma_data = &s3c24xx_i2s_pcm_stereo_out;
else
- rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_in;
+ dma_data = &s3c24xx_i2s_pcm_stereo_in;
+
+ snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_data);
/* Working copies of register */
iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
@@ -258,13 +261,11 @@
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8:
iismod &= ~S3C2410_IISMOD_16BIT;
- ((struct s3c_dma_params *)
- rtd->dai->cpu_dai->dma_data)->dma_size = 1;
+ dma_data->dma_size = 1;
break;
case SNDRV_PCM_FORMAT_S16_LE:
iismod |= S3C2410_IISMOD_16BIT;
- ((struct s3c_dma_params *)
- rtd->dai->cpu_dai->dma_data)->dma_size = 2;
+ dma_data->dma_size = 2;
break;
default:
return -EINVAL;
@@ -280,8 +281,8 @@
{
int ret = 0;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- int channel = ((struct s3c_dma_params *)
- rtd->dai->cpu_dai->dma_data)->channel;
+ struct s3c_dma_params *dma_data =
+ snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
pr_debug("Entered %s\n", __func__);
@@ -300,7 +301,7 @@
else
s3c24xx_snd_txctrl(1);
- s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+ s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c
index 0664fac..5b9ac17 100644
--- a/sound/soc/s6000/s6000-i2s.c
+++ b/sound/soc/s6000/s6000-i2s.c
@@ -519,7 +519,8 @@
s6000_i2s_dai.dev = &pdev->dev;
s6000_i2s_dai.private_data = dev;
- s6000_i2s_dai.dma_data = &dev->dma_params;
+ s6000_i2s_dai.capture.dma_data = &dev->dma_params;
+ s6000_i2s_dai.playback.dma_data = &dev->dma_params;
dev->sifbase = sifmem->start;
dev->scbbase = mmio;
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
index 1d61109..9c7f7f0 100644
--- a/sound/soc/s6000/s6000-pcm.c
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -58,13 +58,15 @@
struct snd_pcm_runtime *runtime = substream->runtime;
struct s6000_runtime_data *prtd = runtime->private_data;
struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
- struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+ struct s6000_pcm_dma_params *par;
int channel;
unsigned int period_size;
unsigned int dma_offset;
dma_addr_t dma_pos;
dma_addr_t src, dst;
+ par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
period_size = snd_pcm_lib_period_bytes(substream);
dma_offset = prtd->period * period_size;
dma_pos = runtime->dma_addr + dma_offset;
@@ -101,7 +103,8 @@
{
struct snd_pcm *pcm = data;
struct snd_soc_pcm_runtime *runtime = pcm->private_data;
- struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data;
+ struct s6000_pcm_dma_params *params =
+ snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
struct s6000_runtime_data *prtd;
unsigned int has_xrun;
int i, ret = IRQ_NONE;
@@ -172,11 +175,13 @@
{
struct s6000_runtime_data *prtd = substream->runtime->private_data;
struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
- struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+ struct s6000_pcm_dma_params *par;
unsigned long flags;
int srcinc;
u32 dma;
+ par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
spin_lock_irqsave(&prtd->lock, flags);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
@@ -212,10 +217,12 @@
{
struct s6000_runtime_data *prtd = substream->runtime->private_data;
struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
- struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+ struct s6000_pcm_dma_params *par;
unsigned long flags;
u32 channel;
+ par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
channel = par->dma_out;
else
@@ -236,9 +243,11 @@
static int s6000_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
- struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+ struct s6000_pcm_dma_params *par;
int ret;
+ par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
ret = par->trigger(substream, cmd, 0);
if (ret < 0)
return ret;
@@ -275,13 +284,15 @@
static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
- struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+ struct s6000_pcm_dma_params *par;
struct snd_pcm_runtime *runtime = substream->runtime;
struct s6000_runtime_data *prtd = runtime->private_data;
unsigned long flags;
unsigned int offset;
dma_addr_t count;
+ par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
spin_lock_irqsave(&prtd->lock, flags);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -305,11 +316,12 @@
static int s6000_pcm_open(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
- struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+ struct s6000_pcm_dma_params *par;
struct snd_pcm_runtime *runtime = substream->runtime;
struct s6000_runtime_data *prtd;
int ret;
+ par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
snd_soc_set_runtime_hwparams(substream, &s6000_pcm_hardware);
ret = snd_pcm_hw_constraint_step(runtime, 0,
@@ -364,7 +376,7 @@
struct snd_pcm_hw_params *hw_params)
{
struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
- struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+ struct s6000_pcm_dma_params *par;
int ret;
ret = snd_pcm_lib_malloc_pages(substream,
params_buffer_bytes(hw_params));
@@ -373,6 +385,8 @@
return ret;
}
+ par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
if (par->same_rate) {
spin_lock(&par->lock);
if (par->rate == -1 ||
@@ -392,7 +406,8 @@
static int s6000_pcm_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
- struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+ struct s6000_pcm_dma_params *par =
+ snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
spin_lock(&par->lock);
par->in_use &= ~(1 << substream->stream);
@@ -417,7 +432,8 @@
static void s6000_pcm_free(struct snd_pcm *pcm)
{
struct snd_soc_pcm_runtime *runtime = pcm->private_data;
- struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data;
+ struct s6000_pcm_dma_params *params =
+ snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
free_irq(params->irq, pcm);
snd_pcm_lib_preallocate_free_for_all(pcm);
@@ -429,9 +445,11 @@
struct snd_soc_dai *dai, struct snd_pcm *pcm)
{
struct snd_soc_pcm_runtime *runtime = pcm->private_data;
- struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data;
+ struct s6000_pcm_dma_params *params;
int res;
+ params = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
if (!card->dev->dma_mask)
card->dev->dma_mask = &s6000_pcm_dmamask;
if (!card->dev->coherent_dma_mask)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 2320153..ad7f952 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1549,7 +1549,8 @@
mutex_unlock(&codec->mutex);
return ret;
}
- if (card->dai_link[i].codec_dai->ac97_control) {
+ /* Check for codec->ac97 to handle the ac97.c fun */
+ if (card->dai_link[i].codec_dai->ac97_control && codec->ac97) {
snd_ac97_dev_add_pdata(codec->ac97,
card->dai_link[i].cpu_dai->ac97_pdata);
}