From 38357ab2c83631728afa37a783c9b1bd474a0739 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 6 Jun 2009 19:03:23 +0100 Subject: ASoC: Sort DAPM power sequences while building lists In the past the DAPM power sequencing was done by iterating over the list of widgets once for each widget type and powering widgets of that type. Instead of doing that do the sorting at the time we insert the widgets into the lists of widgets to apply power changes to. This reduces the amount of computation required for seqencing still further, though the costs are generally dwarfed by the costs of the register writes implementing them. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 109 +++++++++++++++++++++++++++++++++------------------ 1 file changed, 71 insertions(+), 38 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 21c69074aa17..1b38e2195596 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -52,19 +52,37 @@ /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { - snd_soc_dapm_pre, snd_soc_dapm_supply, snd_soc_dapm_micbias, - snd_soc_dapm_mic, snd_soc_dapm_mux, snd_soc_dapm_value_mux, - snd_soc_dapm_dac, snd_soc_dapm_mixer, snd_soc_dapm_mixer_named_ctl, - snd_soc_dapm_pga, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, - snd_soc_dapm_post + [snd_soc_dapm_pre] = 0, + [snd_soc_dapm_supply] = 1, + [snd_soc_dapm_micbias] = 2, + [snd_soc_dapm_mic] = 3, + [snd_soc_dapm_mux] = 4, + [snd_soc_dapm_value_mux] = 5, + [snd_soc_dapm_dac] = 6, + [snd_soc_dapm_mixer] = 7, + [snd_soc_dapm_mixer_named_ctl] = 8, + [snd_soc_dapm_pga] = 9, + [snd_soc_dapm_adc] = 10, + [snd_soc_dapm_hp] = 11, + [snd_soc_dapm_spk] = 12, + [snd_soc_dapm_post] = 13, }; static int dapm_down_seq[] = { - snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, - snd_soc_dapm_pga, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_mixer, - snd_soc_dapm_dac, snd_soc_dapm_mic, snd_soc_dapm_micbias, - snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_supply, - snd_soc_dapm_post + [snd_soc_dapm_pre] = 0, + [snd_soc_dapm_adc] = 1, + [snd_soc_dapm_hp] = 2, + [snd_soc_dapm_spk] = 3, + [snd_soc_dapm_pga] = 4, + [snd_soc_dapm_mixer_named_ctl] = 5, + [snd_soc_dapm_mixer] = 6, + [snd_soc_dapm_dac] = 7, + [snd_soc_dapm_mic] = 8, + [snd_soc_dapm_micbias] = 9, + [snd_soc_dapm_mux] = 10, + [snd_soc_dapm_value_mux] = 11, + [snd_soc_dapm_supply] = 12, + [snd_soc_dapm_post] = 13, }; static void pop_wait(u32 pop_time) @@ -738,6 +756,32 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event, } } +static int dapm_seq_compare(struct snd_soc_dapm_widget *a, + struct snd_soc_dapm_widget *b, + int sort[]) +{ + if (sort[a->id] != sort[b->id]) + return sort[a->id] - sort[b->id]; + + return 0; +} + +/* Insert a widget in order into a DAPM power sequence. */ +static void dapm_seq_insert(struct snd_soc_dapm_widget *new_widget, + struct list_head *list, + int sort[]) +{ + struct snd_soc_dapm_widget *w; + + list_for_each_entry(w, list, power_list) + if (dapm_seq_compare(new_widget, w, sort) < 0) { + list_add_tail(&new_widget->power_list, &w->power_list); + return; + } + + list_add_tail(&new_widget->power_list, list); +} + /* * Scan each dapm widget for complete audio path. * A complete path is a route that has valid endpoints i.e.:- @@ -752,7 +796,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) struct snd_soc_device *socdev = codec->socdev; struct snd_soc_dapm_widget *w; int ret = 0; - int i, power; + int power; int sys_power = 0; INIT_LIST_HEAD(&codec->up_list); @@ -764,10 +808,10 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) list_for_each_entry(w, &codec->dapm_widgets, list) { switch (w->id) { case snd_soc_dapm_pre: - list_add_tail(&codec->down_list, &w->power_list); + dapm_seq_insert(w, &codec->down_list, dapm_down_seq); break; case snd_soc_dapm_post: - list_add_tail(&codec->up_list, &w->power_list); + dapm_seq_insert(w, &codec->up_list, dapm_up_seq); break; default: @@ -782,10 +826,11 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) continue; if (power) - list_add_tail(&w->power_list, &codec->up_list); + dapm_seq_insert(w, &codec->up_list, + dapm_up_seq); else - list_add_tail(&w->power_list, - &codec->down_list); + dapm_seq_insert(w, &codec->down_list, + dapm_down_seq); w->power = power; break; @@ -802,31 +847,19 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) } /* Power down widgets first; try to avoid amplifying pops. */ - for (i = 0; i < ARRAY_SIZE(dapm_down_seq); i++) { - list_for_each_entry(w, &codec->down_list, power_list) { - /* is widget in stream order */ - if (w->id != dapm_down_seq[i]) - continue; - - ret = dapm_power_widget(codec, event, w); - if (ret != 0) - pr_err("Failed to power down %s: %d\n", - w->name, ret); - } + list_for_each_entry(w, &codec->down_list, power_list) { + ret = dapm_power_widget(codec, event, w); + if (ret != 0) + pr_err("Failed to power down %s: %d\n", + w->name, ret); } /* Now power up. */ - for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) { - list_for_each_entry(w, &codec->up_list, power_list) { - /* is widget in stream order */ - if (w->id != dapm_up_seq[i]) - continue; - - ret = dapm_power_widget(codec, event, w); - if (ret != 0) - pr_err("Failed to power up %s: %d\n", - w->name, ret); - } + list_for_each_entry(w, &codec->up_list, power_list) { + ret = dapm_power_widget(codec, event, w); + if (ret != 0) + pr_err("Failed to power up %s: %d\n", + w->name, ret); } /* If we just powered the last thing off drop to standby bias */ -- cgit v1.2.3 From 163cac061c97394d4ef9c89efe5921dac937ddb8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 7 Jun 2009 10:12:52 +0100 Subject: ASoC: Factor out DAPM sequence execution Lump the list walk into a single function, and pull in the power application too so we can do some further refactoring. Pure code motion. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 122 +++++++++++++++++++++++++-------------------------- 1 file changed, 61 insertions(+), 61 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1b38e2195596..257d4f15e00e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -707,55 +707,6 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) return power; } -/* - * Scan a single DAPM widget for a complete audio path and update the - * power status appropriately. - */ -static int dapm_power_widget(struct snd_soc_codec *codec, int event, - struct snd_soc_dapm_widget *w) -{ - int ret; - - switch (w->id) { - case snd_soc_dapm_pre: - if (!w->event) - return 0; - - if (event == SND_SOC_DAPM_STREAM_START) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMU); - if (ret < 0) - return ret; - } else if (event == SND_SOC_DAPM_STREAM_STOP) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMD); - if (ret < 0) - return ret; - } - return 0; - - case snd_soc_dapm_post: - if (!w->event) - return 0; - - if (event == SND_SOC_DAPM_STREAM_START) { - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMU); - if (ret < 0) - return ret; - } else if (event == SND_SOC_DAPM_STREAM_STOP) { - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMD); - if (ret < 0) - return ret; - } - return 0; - - default: - return dapm_generic_apply_power(w); - } -} - static int dapm_seq_compare(struct snd_soc_dapm_widget *a, struct snd_soc_dapm_widget *b, int sort[]) @@ -782,6 +733,65 @@ static void dapm_seq_insert(struct snd_soc_dapm_widget *new_widget, list_add_tail(&new_widget->power_list, list); } +/* Apply a DAPM power sequence */ +static void dapm_seq_run(struct snd_soc_codec *codec, struct list_head *list, + int event) +{ + struct snd_soc_dapm_widget *w; + int ret; + + list_for_each_entry(w, list, power_list) { + switch (w->id) { + case snd_soc_dapm_pre: + if (!w->event) + list_for_each_entry_continue(w, list, + power_list); + + if (event == SND_SOC_DAPM_STREAM_START) { + ret = w->event(w, + NULL, SND_SOC_DAPM_PRE_PMU); + if (ret < 0) + pr_err("PRE widget failed: %d\n", + ret); + } else if (event == SND_SOC_DAPM_STREAM_STOP) { + ret = w->event(w, + NULL, SND_SOC_DAPM_PRE_PMD); + if (ret < 0) + pr_err("PRE widget failed: %d\n", + ret); + } + break; + + case snd_soc_dapm_post: + if (!w->event) + list_for_each_entry_continue(w, list, + power_list); + + if (event == SND_SOC_DAPM_STREAM_START) { + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMU); + if (ret < 0) + pr_err("POST widget failed: %d\n", + ret); + } else if (event == SND_SOC_DAPM_STREAM_STOP) { + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMD); + if (ret < 0) + pr_err("POST widget failed: %d\n", + ret); + } + break; + + default: + ret = dapm_generic_apply_power(w); + if (ret < 0) + pr_err("Failed to apply widget power: %d\n", + ret); + break; + } + } +} + /* * Scan each dapm widget for complete audio path. * A complete path is a route that has valid endpoints i.e.:- @@ -847,20 +857,10 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) } /* Power down widgets first; try to avoid amplifying pops. */ - list_for_each_entry(w, &codec->down_list, power_list) { - ret = dapm_power_widget(codec, event, w); - if (ret != 0) - pr_err("Failed to power down %s: %d\n", - w->name, ret); - } + dapm_seq_run(codec, &codec->down_list, event); /* Now power up. */ - list_for_each_entry(w, &codec->up_list, power_list) { - ret = dapm_power_widget(codec, event, w); - if (ret != 0) - pr_err("Failed to power up %s: %d\n", - w->name, ret); - } + dapm_seq_run(codec, &codec->up_list, event); /* If we just powered the last thing off drop to standby bias */ if (codec->bias_level == SND_SOC_BIAS_PREPARE && !sys_power) { -- cgit v1.2.3 From 46f5822f7841697d4aedaf4672661d7a765172cd Mon Sep 17 00:00:00 2001 From: Daniel Ribeiro Date: Sun, 7 Jun 2009 02:49:11 -0300 Subject: ASoC: Allow 32 bit registers for DAPM Replace the remaining unsigned shorts with unsigned ints. Tested with pcap2 codec (25 bits registers). Signed-off-by: Daniel Ribeiro Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ++-- sound/soc/soc-core.c | 28 ++++++++++++++-------------- sound/soc/soc-dapm.c | 16 ++++++++-------- 3 files changed, 24 insertions(+), 24 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index cf6111d72b17..a167b4930447 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -216,9 +216,9 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, /* codec register bit access */ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value); + unsigned int mask, unsigned int value); int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value); + unsigned int mask, unsigned int value); int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3f44150d8e30..e1a920cd8953 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1264,10 +1264,10 @@ EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); * Returns 1 for change else 0. */ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value) + unsigned int mask, unsigned int value) { int change; - unsigned short old, new; + unsigned int old, new; mutex_lock(&io_mutex); old = snd_soc_read(codec, reg); @@ -1294,10 +1294,10 @@ EXPORT_SYMBOL_GPL(snd_soc_update_bits); * Returns 1 for change else 0. */ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value) + unsigned int mask, unsigned int value) { int change; - unsigned short old, new; + unsigned int old, new; mutex_lock(&io_mutex); old = snd_soc_read(codec, reg); @@ -1583,7 +1583,7 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, bitmask; + unsigned int val, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -1612,8 +1612,8 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val; - unsigned short mask, bitmask; + unsigned int val; + unsigned int mask, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -1649,7 +1649,7 @@ int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short reg_val, val, mux; + unsigned int reg_val, val, mux; reg_val = snd_soc_read(codec, e->reg); val = (reg_val >> e->shift_l) & e->mask; @@ -1688,8 +1688,8 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val; - unsigned short mask; + unsigned int val; + unsigned int mask; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; @@ -1849,7 +1849,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned short val, val2, val_mask; + unsigned int val, val2, val_mask; val = (ucontrol->value.integer.value[0] & mask); if (invert) @@ -1915,7 +1915,7 @@ int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, unsigned int reg2 = mc->rreg; unsigned int shift = mc->shift; int max = mc->max; - unsigned int mask = (1<invert; ucontrol->value.integer.value[0] = @@ -1955,7 +1955,7 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; int err; - unsigned short val, val2, val_mask; + unsigned int val, val2, val_mask; val_mask = mask << shift; val = (ucontrol->value.integer.value[0] & mask); @@ -2047,7 +2047,7 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; int min = mc->min; - unsigned short val; + unsigned int val; val = (ucontrol->value.integer.value[0]+min) & 0xff; val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 21c69074aa17..7ad8afa8553d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -268,7 +268,7 @@ static int dapm_connect_mixer(struct snd_soc_codec *codec, static int dapm_update_bits(struct snd_soc_dapm_widget *widget) { int change, power; - unsigned short old, new; + unsigned int old, new; struct snd_soc_codec *codec = widget->codec; /* check for valid widgets */ @@ -1372,7 +1372,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned short val, val2, val_mask; + unsigned int val, val2, val_mask; int ret; val = (ucontrol->value.integer.value[0] & mask); @@ -1436,7 +1436,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, bitmask; + unsigned int val, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -1464,8 +1464,8 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, mux; - unsigned short mask, bitmask; + unsigned int val, mux; + unsigned int mask, bitmask; int ret = 0; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) @@ -1523,7 +1523,7 @@ int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short reg_val, val, mux; + unsigned int reg_val, val, mux; reg_val = snd_soc_read(widget->codec, e->reg); val = (reg_val >> e->shift_l) & e->mask; @@ -1563,8 +1563,8 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, mux; - unsigned short mask; + unsigned int val, mux; + unsigned int mask; int ret = 0; if (ucontrol->value.enumerated.item[0] > e->max - 1) -- cgit v1.2.3 From b22ead2a510fdb30440753f90237e86fdac70fae Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 7 Jun 2009 12:51:26 +0100 Subject: ASoC: Coalesce register writes for DAPM sequences Reduce the number of register writes we need to set the power state for a CODEC by coalescing updates to widgets with the same sequence order and same register into a single write. This can be a noticable performance improvement with slow or heavily contended control buses, such as I2C controllers with a low clock frequency, and is particularly noticable when resuming. It can also reduce the noticability of and pops and clicks by ensuring that left and right channels are powered simultaneously if they are in the same register. Currently widgets that have events are not coalesced, including PGAs which may use the volume ramping control. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 136 +++++++++++++++++++++++++++++++++++++++------------ 1 file changed, 104 insertions(+), 32 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 257d4f15e00e..66f07cdfb2f7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -713,6 +713,8 @@ static int dapm_seq_compare(struct snd_soc_dapm_widget *a, { if (sort[a->id] != sort[b->id]) return sort[a->id] - sort[b->id]; + if (a->reg != b->reg) + return a->reg - b->reg; return 0; } @@ -733,63 +735,133 @@ static void dapm_seq_insert(struct snd_soc_dapm_widget *new_widget, list_add_tail(&new_widget->power_list, list); } -/* Apply a DAPM power sequence */ -static void dapm_seq_run(struct snd_soc_codec *codec, struct list_head *list, - int event) +/* Apply the coalesced changes from a DAPM sequence */ +static void dapm_seq_run_coalesced(struct snd_soc_codec *codec, + struct list_head *pending) { struct snd_soc_dapm_widget *w; + int reg, power; + unsigned int value = 0; + unsigned int mask = 0; + unsigned int cur_mask; + + reg = list_first_entry(pending, struct snd_soc_dapm_widget, + power_list)->reg; + + list_for_each_entry(w, pending, power_list) { + cur_mask = 1 << w->shift; + BUG_ON(reg != w->reg); + + if (w->invert) + power = !w->power; + else + power = w->power; + + mask |= cur_mask; + if (power) + value |= cur_mask; + + pop_dbg(codec->pop_time, + "pop test : Queue %s: reg=0x%x, 0x%x/0x%x\n", + w->name, reg, value, mask); + } + + pop_dbg(codec->pop_time, + "pop test : Applying 0x%x/0x%x to %x in %dms\n", + value, mask, reg, codec->pop_time); + pop_wait(codec->pop_time); + snd_soc_update_bits(codec, reg, mask, value); +} + +/* Apply a DAPM power sequence. + * + * We walk over a pre-sorted list of widgets to apply power to. In + * order to minimise the number of writes to the device required + * multiple widgets will be updated in a single write where possible. + * Currently anything that requires more than a single write is not + * handled. + */ +static void dapm_seq_run(struct snd_soc_codec *codec, struct list_head *list, + int event, int sort[]) +{ + struct snd_soc_dapm_widget *w, *n; + LIST_HEAD(pending); + int cur_sort = -1; + int cur_reg = SND_SOC_NOPM; int ret; - list_for_each_entry(w, list, power_list) { + list_for_each_entry_safe(w, n, list, power_list) { + ret = 0; + + /* Do we need to apply any queued changes? */ + if (sort[w->id] != cur_sort || w->reg != cur_reg) { + if (!list_empty(&pending)) + dapm_seq_run_coalesced(codec, &pending); + + INIT_LIST_HEAD(&pending); + cur_sort = -1; + cur_reg = SND_SOC_NOPM; + } + switch (w->id) { case snd_soc_dapm_pre: if (!w->event) - list_for_each_entry_continue(w, list, - power_list); + list_for_each_entry_safe_continue(w, n, list, + power_list); - if (event == SND_SOC_DAPM_STREAM_START) { + if (event == SND_SOC_DAPM_STREAM_START) ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); - if (ret < 0) - pr_err("PRE widget failed: %d\n", - ret); - } else if (event == SND_SOC_DAPM_STREAM_STOP) { + else if (event == SND_SOC_DAPM_STREAM_STOP) ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); - if (ret < 0) - pr_err("PRE widget failed: %d\n", - ret); - } break; case snd_soc_dapm_post: if (!w->event) - list_for_each_entry_continue(w, list, - power_list); + list_for_each_entry_safe_continue(w, n, list, + power_list); - if (event == SND_SOC_DAPM_STREAM_START) { + if (event == SND_SOC_DAPM_STREAM_START) ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMU); - if (ret < 0) - pr_err("POST widget failed: %d\n", - ret); - } else if (event == SND_SOC_DAPM_STREAM_STOP) { + else if (event == SND_SOC_DAPM_STREAM_STOP) ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); - if (ret < 0) - pr_err("POST widget failed: %d\n", - ret); - } break; - default: + case snd_soc_dapm_input: + case snd_soc_dapm_output: + case snd_soc_dapm_hp: + case snd_soc_dapm_mic: + case snd_soc_dapm_line: + case snd_soc_dapm_spk: + /* No register support currently */ + case snd_soc_dapm_pga: + /* Don't coalsece these yet due to gain ramping */ ret = dapm_generic_apply_power(w); - if (ret < 0) - pr_err("Failed to apply widget power: %d\n", - ret); break; + + default: + /* If there's an event or an invalid register + * then run immediately, otherwise store the + * updates so that we can coalesce. */ + if (w->reg >= 0 && !w->event) { + cur_sort = sort[w->id]; + cur_reg = w->reg; + list_move(&w->power_list, &pending); + } else { + ret = dapm_generic_apply_power(w); + } } + + if (ret < 0) + pr_err("Failed to apply widget power: %d\n", + ret); } + + if (!list_empty(&pending)) + dapm_seq_run_coalesced(codec, &pending); } /* @@ -857,10 +929,10 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) } /* Power down widgets first; try to avoid amplifying pops. */ - dapm_seq_run(codec, &codec->down_list, event); + dapm_seq_run(codec, &codec->down_list, event, dapm_down_seq); /* Now power up. */ - dapm_seq_run(codec, &codec->up_list, event); + dapm_seq_run(codec, &codec->up_list, event, dapm_up_seq); /* If we just powered the last thing off drop to standby bias */ if (codec->bias_level == SND_SOC_BIAS_PREPARE && !sys_power) { -- cgit v1.2.3 From e3d4dabd2d9b74778f6f15a830eb3a0027bb3799 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 7 Jun 2009 13:08:45 +0100 Subject: ASoC: Sort specialised mixers and muxes together The more flexible value muxes and named mixers don't need to be sorted differently from a power management point of view, they are different only in terms of the control interface and not in terms of seqencing behaviour. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 66f07cdfb2f7..9187db18a042 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -57,15 +57,15 @@ static int dapm_up_seq[] = { [snd_soc_dapm_micbias] = 2, [snd_soc_dapm_mic] = 3, [snd_soc_dapm_mux] = 4, - [snd_soc_dapm_value_mux] = 5, - [snd_soc_dapm_dac] = 6, - [snd_soc_dapm_mixer] = 7, - [snd_soc_dapm_mixer_named_ctl] = 8, - [snd_soc_dapm_pga] = 9, - [snd_soc_dapm_adc] = 10, - [snd_soc_dapm_hp] = 11, - [snd_soc_dapm_spk] = 12, - [snd_soc_dapm_post] = 13, + [snd_soc_dapm_value_mux] = 4, + [snd_soc_dapm_dac] = 5, + [snd_soc_dapm_mixer] = 6, + [snd_soc_dapm_mixer_named_ctl] = 6, + [snd_soc_dapm_pga] = 7, + [snd_soc_dapm_adc] = 8, + [snd_soc_dapm_hp] = 9, + [snd_soc_dapm_spk] = 10, + [snd_soc_dapm_post] = 11, }; static int dapm_down_seq[] = { @@ -75,14 +75,14 @@ static int dapm_down_seq[] = { [snd_soc_dapm_spk] = 3, [snd_soc_dapm_pga] = 4, [snd_soc_dapm_mixer_named_ctl] = 5, - [snd_soc_dapm_mixer] = 6, - [snd_soc_dapm_dac] = 7, - [snd_soc_dapm_mic] = 8, - [snd_soc_dapm_micbias] = 9, - [snd_soc_dapm_mux] = 10, - [snd_soc_dapm_value_mux] = 11, - [snd_soc_dapm_supply] = 12, - [snd_soc_dapm_post] = 13, + [snd_soc_dapm_mixer] = 5, + [snd_soc_dapm_dac] = 6, + [snd_soc_dapm_mic] = 7, + [snd_soc_dapm_micbias] = 8, + [snd_soc_dapm_mux] = 9, + [snd_soc_dapm_value_mux] = 9, + [snd_soc_dapm_supply] = 10, + [snd_soc_dapm_post] = 11, }; static void pop_wait(u32 pop_time) -- cgit v1.2.3 From 81628103dd8527d99ea39b054a3f002d5859d7c3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 7 Jun 2009 13:21:24 +0100 Subject: ASoC: Coalesce power updates for DAPM widgets with events Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 76 +++++++++++++++++++++++++++++++++++++++++----------- 1 file changed, 60 insertions(+), 16 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9187db18a042..3fc791c28aa8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -740,7 +740,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_codec *codec, struct list_head *pending) { struct snd_soc_dapm_widget *w; - int reg, power; + int reg, power, ret; unsigned int value = 0; unsigned int mask = 0; unsigned int cur_mask; @@ -764,13 +764,62 @@ static void dapm_seq_run_coalesced(struct snd_soc_codec *codec, pop_dbg(codec->pop_time, "pop test : Queue %s: reg=0x%x, 0x%x/0x%x\n", w->name, reg, value, mask); + + /* power up pre event */ + if (w->power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMU)) { + pop_dbg(codec->pop_time, "pop test : %s PRE_PMU\n", + w->name); + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); + if (ret < 0) + pr_err("%s: pre event failed: %d\n", + w->name, ret); + } + + /* power down pre event */ + if (!w->power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMD)) { + pop_dbg(codec->pop_time, "pop test : %s PRE_PMD\n", + w->name); + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); + if (ret < 0) + pr_err("%s: pre event failed: %d\n", + w->name, ret); + } + } + + if (reg >= 0) { + pop_dbg(codec->pop_time, + "pop test : Applying 0x%x/0x%x to %x in %dms\n", + value, mask, reg, codec->pop_time); + pop_wait(codec->pop_time); + snd_soc_update_bits(codec, reg, mask, value); } - pop_dbg(codec->pop_time, - "pop test : Applying 0x%x/0x%x to %x in %dms\n", - value, mask, reg, codec->pop_time); - pop_wait(codec->pop_time); - snd_soc_update_bits(codec, reg, mask, value); + list_for_each_entry(w, pending, power_list) { + /* power up post event */ + if (w->power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMU)) { + pop_dbg(codec->pop_time, "pop test : %s POST_PMU\n", + w->name); + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMU); + if (ret < 0) + pr_err("%s: post event failed: %d\n", + w->name, ret); + } + + /* power down post event */ + if (!w->power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMD)) { + pop_dbg(codec->pop_time, "pop test : %s POST_PMD\n", + w->name); + ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); + if (ret < 0) + pr_err("%s: post event failed: %d\n", + w->name, ret); + } + } } /* Apply a DAPM power sequence. @@ -843,16 +892,11 @@ static void dapm_seq_run(struct snd_soc_codec *codec, struct list_head *list, break; default: - /* If there's an event or an invalid register - * then run immediately, otherwise store the - * updates so that we can coalesce. */ - if (w->reg >= 0 && !w->event) { - cur_sort = sort[w->id]; - cur_reg = w->reg; - list_move(&w->power_list, &pending); - } else { - ret = dapm_generic_apply_power(w); - } + /* Queue it up for application */ + cur_sort = sort[w->id]; + cur_reg = w->reg; + list_move(&w->power_list, &pending); + break; } if (ret < 0) -- cgit v1.2.3 From 4f1c1923851f9734c972812121e80a3b04ab3af4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 7 Jun 2009 13:37:44 +0100 Subject: ASoC: Coalesce power updates for PGAs Handle gain ramping for PGAs so we can coalesce their power updates too. This is not ideal since we can't cope properly with gain ramping for stereo paths but that was the case without coalescing and gain ramping is relatively infrequently used so the effects are limited. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3fc791c28aa8..7299ce405b2d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -786,6 +786,10 @@ static void dapm_seq_run_coalesced(struct snd_soc_codec *codec, pr_err("%s: pre event failed: %d\n", w->name, ret); } + + /* Lower PGA volume to reduce pops */ + if (w->id == snd_soc_dapm_pga && !w->power) + dapm_set_pga(w, w->power); } if (reg >= 0) { @@ -797,6 +801,10 @@ static void dapm_seq_run_coalesced(struct snd_soc_codec *codec, } list_for_each_entry(w, pending, power_list) { + /* Raise PGA volume to reduce pops */ + if (w->id == snd_soc_dapm_pga && w->power) + dapm_set_pga(w, w->power); + /* power up post event */ if (w->power && w->event && (w->event_flags & SND_SOC_DAPM_POST_PMU)) { @@ -886,8 +894,6 @@ static void dapm_seq_run(struct snd_soc_codec *codec, struct list_head *list, case snd_soc_dapm_line: case snd_soc_dapm_spk: /* No register support currently */ - case snd_soc_dapm_pga: - /* Don't coalsece these yet due to gain ramping */ ret = dapm_generic_apply_power(w); break; -- cgit v1.2.3 From 291f3bbcacf278726911c713e14cedb71c486b16 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 7 Jun 2009 13:57:17 +0100 Subject: ASoC: Make DAPM power sequence lists local variables They are now only accessed within dapm_power_widgets() so can be local to that function. Signed-off-by: Mark Brown --- include/sound/soc.h | 2 -- sound/soc/soc-dapm.c | 19 ++++++++----------- 2 files changed, 8 insertions(+), 13 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index cf6111d72b17..5964dd65bbd3 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -369,8 +369,6 @@ struct snd_soc_codec { enum snd_soc_bias_level bias_level; enum snd_soc_bias_level suspend_bias_level; struct delayed_work delayed_work; - struct list_head up_list; - struct list_head down_list; /* codec DAI's */ struct snd_soc_dai *dai; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7299ce405b2d..1c30da1535b5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -927,23 +927,22 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) { struct snd_soc_device *socdev = codec->socdev; struct snd_soc_dapm_widget *w; + LIST_HEAD(up_list); + LIST_HEAD(down_list); int ret = 0; int power; int sys_power = 0; - INIT_LIST_HEAD(&codec->up_list); - INIT_LIST_HEAD(&codec->down_list); - /* Check which widgets we need to power and store them in * lists indicating if they should be powered up or down. */ list_for_each_entry(w, &codec->dapm_widgets, list) { switch (w->id) { case snd_soc_dapm_pre: - dapm_seq_insert(w, &codec->down_list, dapm_down_seq); + dapm_seq_insert(w, &down_list, dapm_down_seq); break; case snd_soc_dapm_post: - dapm_seq_insert(w, &codec->up_list, dapm_up_seq); + dapm_seq_insert(w, &up_list, dapm_up_seq); break; default: @@ -958,11 +957,9 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) continue; if (power) - dapm_seq_insert(w, &codec->up_list, - dapm_up_seq); + dapm_seq_insert(w, &up_list, dapm_up_seq); else - dapm_seq_insert(w, &codec->down_list, - dapm_down_seq); + dapm_seq_insert(w, &down_list, dapm_down_seq); w->power = power; break; @@ -979,10 +976,10 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) } /* Power down widgets first; try to avoid amplifying pops. */ - dapm_seq_run(codec, &codec->down_list, event, dapm_down_seq); + dapm_seq_run(codec, &down_list, event, dapm_down_seq); /* Now power up. */ - dapm_seq_run(codec, &codec->up_list, event, dapm_up_seq); + dapm_seq_run(codec, &up_list, event, dapm_up_seq); /* If we just powered the last thing off drop to standby bias */ if (codec->bias_level == SND_SOC_BIAS_PREPARE && !sys_power) { -- cgit v1.2.3 From fa44c077eb2a17aa6913feceb493d13f45f3fa25 Mon Sep 17 00:00:00 2001 From: Daniel Ribeiro Date: Wed, 10 Jun 2009 15:23:24 -0300 Subject: ASoC: remove duplicated code on pxa-ssp.c * We don't need to write the registers twice, remove the first write. * DAIFMT_INV switch is duplicated inside DAIFMT_FORMAT switch, move it out. (This patch is for Mark's for-2.6.32 branch, I have not checked if the code is duplicated on current 2.6.30) Signed-off-by: Daniel Ribeiro Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 50 +++++++++++++++---------------------------------- 1 file changed, 15 insertions(+), 35 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 19c45409d94c..e22c5cef8fec 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -457,31 +457,27 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } - ssp_write_reg(ssp, SSCR0, sscr0); - ssp_write_reg(ssp, SSCR1, sscr1); - ssp_write_reg(ssp, SSPSP, sspsp); + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + sspsp |= SSPSP_SFRMP; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + case SND_SOC_DAIFMT_IB_IF: + sspsp |= SSPSP_SCMODE(2); + break; + case SND_SOC_DAIFMT_IB_NF: + sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; + break; + default: + return -EINVAL; + } switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; - /* See hw_params() */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - sspsp |= SSPSP_SFRMP; - break; - case SND_SOC_DAIFMT_NB_IF: - break; - case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SCMODE(2); - break; - case SND_SOC_DAIFMT_IB_NF: - sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; - break; - default: - return -EINVAL; - } break; case SND_SOC_DAIFMT_DSP_A: @@ -489,22 +485,6 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_DSP_B: sscr0 |= SSCR0_MOD | SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - sspsp |= SSPSP_SFRMP; - break; - case SND_SOC_DAIFMT_NB_IF: - break; - case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SCMODE(2); - break; - case SND_SOC_DAIFMT_IB_NF: - sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; - break; - default: - return -EINVAL; - } break; default: -- cgit v1.2.3 From 74dc55ed5b709e4a2a538252f98ea62358df82ce Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 9 Jun 2009 09:55:51 +0100 Subject: ASoC: Add WM8961 driver The WM8961 is a low power, high quality stereo CODEC designed for portable digital applications with headphone and stereo class D speaker drivers. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8961.c | 1309 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8961.h | 866 ++++++++++++++++++++++++++++++ 4 files changed, 2181 insertions(+) create mode 100644 sound/soc/codecs/wm8961.c create mode 100644 sound/soc/codecs/wm8961.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bbc97fd76648..021dbdfa5b92 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -39,6 +39,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8903 if I2C select SND_SOC_WM8940 if I2C select SND_SOC_WM8960 if I2C + select SND_SOC_WM8961 if I2C select SND_SOC_WM8971 if I2C select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C @@ -156,6 +157,9 @@ config SND_SOC_WM8940 config SND_SOC_WM8960 tristate +config SND_SOC_WM8961 + tristate + config SND_SOC_WM8971 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8b7530546f4d..e520c2b7f0e0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -27,6 +27,7 @@ snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8940-objs := wm8940.o snd-soc-wm8960-objs := wm8960.o +snd-soc-wm8961-objs := wm8961.o snd-soc-wm8971-objs := wm8971.o snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o @@ -65,6 +66,7 @@ obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o obj-$(CONFIG_SND_SOC_WM8940) += snd-soc-wm8940.o obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o +obj-$(CONFIG_SND_SOC_WM8961) += snd-soc-wm8961.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c new file mode 100644 index 000000000000..8e78959ca409 --- /dev/null +++ b/sound/soc/codecs/wm8961.c @@ -0,0 +1,1309 @@ +/* + * wm8961.c -- WM8961 ALSA SoC Audio driver + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Currently unimplemented features: + * - ALC + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8961.h" + +#define WM8961_MAX_REGISTER 0xFC + +static u16 wm8961_reg_defaults[] = { + 0x009F, /* R0 - Left Input volume */ + 0x009F, /* R1 - Right Input volume */ + 0x0000, /* R2 - LOUT1 volume */ + 0x0000, /* R3 - ROUT1 volume */ + 0x0020, /* R4 - Clocking1 */ + 0x0008, /* R5 - ADC & DAC Control 1 */ + 0x0000, /* R6 - ADC & DAC Control 2 */ + 0x000A, /* R7 - Audio Interface 0 */ + 0x01F4, /* R8 - Clocking2 */ + 0x0000, /* R9 - Audio Interface 1 */ + 0x00FF, /* R10 - Left DAC volume */ + 0x00FF, /* R11 - Right DAC volume */ + 0x0000, /* R12 */ + 0x0000, /* R13 */ + 0x0040, /* R14 - Audio Interface 2 */ + 0x0000, /* R15 - Software Reset */ + 0x0000, /* R16 */ + 0x007B, /* R17 - ALC1 */ + 0x0000, /* R18 - ALC2 */ + 0x0032, /* R19 - ALC3 */ + 0x0000, /* R20 - Noise Gate */ + 0x00C0, /* R21 - Left ADC volume */ + 0x00C0, /* R22 - Right ADC volume */ + 0x0120, /* R23 - Additional control(1) */ + 0x0000, /* R24 - Additional control(2) */ + 0x0000, /* R25 - Pwr Mgmt (1) */ + 0x0000, /* R26 - Pwr Mgmt (2) */ + 0x0000, /* R27 - Additional Control (3) */ + 0x0000, /* R28 - Anti-pop */ + 0x0000, /* R29 */ + 0x005F, /* R30 - Clocking 3 */ + 0x0000, /* R31 */ + 0x0000, /* R32 - ADCL signal path */ + 0x0000, /* R33 - ADCR signal path */ + 0x0000, /* R34 */ + 0x0000, /* R35 */ + 0x0000, /* R36 */ + 0x0000, /* R37 */ + 0x0000, /* R38 */ + 0x0000, /* R39 */ + 0x0000, /* R40 - LOUT2 volume */ + 0x0000, /* R41 - ROUT2 volume */ + 0x0000, /* R42 */ + 0x0000, /* R43 */ + 0x0000, /* R44 */ + 0x0000, /* R45 */ + 0x0000, /* R46 */ + 0x0000, /* R47 - Pwr Mgmt (3) */ + 0x0023, /* R48 - Additional Control (4) */ + 0x0000, /* R49 - Class D Control 1 */ + 0x0000, /* R50 */ + 0x0003, /* R51 - Class D Control 2 */ + 0x0000, /* R52 */ + 0x0000, /* R53 */ + 0x0000, /* R54 */ + 0x0000, /* R55 */ + 0x0106, /* R56 - Clocking 4 */ + 0x0000, /* R57 - DSP Sidetone 0 */ + 0x0000, /* R58 - DSP Sidetone 1 */ + 0x0000, /* R59 */ + 0x0000, /* R60 - DC Servo 0 */ + 0x0000, /* R61 - DC Servo 1 */ + 0x0000, /* R62 */ + 0x015E, /* R63 - DC Servo 3 */ + 0x0010, /* R64 */ + 0x0010, /* R65 - DC Servo 5 */ + 0x0000, /* R66 */ + 0x0001, /* R67 */ + 0x0003, /* R68 - Analogue PGA Bias */ + 0x0000, /* R69 - Analogue HP 0 */ + 0x0060, /* R70 */ + 0x01FB, /* R71 - Analogue HP 2 */ + 0x0000, /* R72 - Charge Pump 1 */ + 0x0065, /* R73 */ + 0x005F, /* R74 */ + 0x0059, /* R75 */ + 0x006B, /* R76 */ + 0x0038, /* R77 */ + 0x000C, /* R78 */ + 0x000A, /* R79 */ + 0x006B, /* R80 */ + 0x0000, /* R81 */ + 0x0000, /* R82 - Charge Pump B */ + 0x0087, /* R83 */ + 0x0000, /* R84 */ + 0x005C, /* R85 */ + 0x0000, /* R86 */ + 0x0000, /* R87 - Write Sequencer 1 */ + 0x0000, /* R88 - Write Sequencer 2 */ + 0x0000, /* R89 - Write Sequencer 3 */ + 0x0000, /* R90 - Write Sequencer 4 */ + 0x0000, /* R91 - Write Sequencer 5 */ + 0x0000, /* R92 - Write Sequencer 6 */ + 0x0000, /* R93 - Write Sequencer 7 */ + 0x0000, /* R94 */ + 0x0000, /* R95 */ + 0x0000, /* R96 */ + 0x0000, /* R97 */ + 0x0000, /* R98 */ + 0x0000, /* R99 */ + 0x0000, /* R100 */ + 0x0000, /* R101 */ + 0x0000, /* R102 */ + 0x0000, /* R103 */ + 0x0000, /* R104 */ + 0x0000, /* R105 */ + 0x0000, /* R106 */ + 0x0000, /* R107 */ + 0x0000, /* R108 */ + 0x0000, /* R109 */ + 0x0000, /* R110 */ + 0x0000, /* R111 */ + 0x0000, /* R112 */ + 0x0000, /* R113 */ + 0x0000, /* R114 */ + 0x0000, /* R115 */ + 0x0000, /* R116 */ + 0x0000, /* R117 */ + 0x0000, /* R118 */ + 0x0000, /* R119 */ + 0x0000, /* R120 */ + 0x0000, /* R121 */ + 0x0000, /* R122 */ + 0x0000, /* R123 */ + 0x0000, /* R124 */ + 0x0000, /* R125 */ + 0x0000, /* R126 */ + 0x0000, /* R127 */ + 0x0000, /* R128 */ + 0x0000, /* R129 */ + 0x0000, /* R130 */ + 0x0000, /* R131 */ + 0x0000, /* R132 */ + 0x0000, /* R133 */ + 0x0000, /* R134 */ + 0x0000, /* R135 */ + 0x0000, /* R136 */ + 0x0000, /* R137 */ + 0x0000, /* R138 */ + 0x0000, /* R139 */ + 0x0000, /* R140 */ + 0x0000, /* R141 */ + 0x0000, /* R142 */ + 0x0000, /* R143 */ + 0x0000, /* R144 */ + 0x0000, /* R145 */ + 0x0000, /* R146 */ + 0x0000, /* R147 */ + 0x0000, /* R148 */ + 0x0000, /* R149 */ + 0x0000, /* R150 */ + 0x0000, /* R151 */ + 0x0000, /* R152 */ + 0x0000, /* R153 */ + 0x0000, /* R154 */ + 0x0000, /* R155 */ + 0x0000, /* R156 */ + 0x0000, /* R157 */ + 0x0000, /* R158 */ + 0x0000, /* R159 */ + 0x0000, /* R160 */ + 0x0000, /* R161 */ + 0x0000, /* R162 */ + 0x0000, /* R163 */ + 0x0000, /* R164 */ + 0x0000, /* R165 */ + 0x0000, /* R166 */ + 0x0000, /* R167 */ + 0x0000, /* R168 */ + 0x0000, /* R169 */ + 0x0000, /* R170 */ + 0x0000, /* R171 */ + 0x0000, /* R172 */ + 0x0000, /* R173 */ + 0x0000, /* R174 */ + 0x0000, /* R175 */ + 0x0000, /* R176 */ + 0x0000, /* R177 */ + 0x0000, /* R178 */ + 0x0000, /* R179 */ + 0x0000, /* R180 */ + 0x0000, /* R181 */ + 0x0000, /* R182 */ + 0x0000, /* R183 */ + 0x0000, /* R184 */ + 0x0000, /* R185 */ + 0x0000, /* R186 */ + 0x0000, /* R187 */ + 0x0000, /* R188 */ + 0x0000, /* R189 */ + 0x0000, /* R190 */ + 0x0000, /* R191 */ + 0x0000, /* R192 */ + 0x0000, /* R193 */ + 0x0000, /* R194 */ + 0x0000, /* R195 */ + 0x0030, /* R196 */ + 0x0006, /* R197 */ + 0x0000, /* R198 */ + 0x0060, /* R199 */ + 0x0000, /* R200 */ + 0x003F, /* R201 */ + 0x0000, /* R202 */ + 0x0000, /* R203 */ + 0x0000, /* R204 */ + 0x0001, /* R205 */ + 0x0000, /* R206 */ + 0x0181, /* R207 */ + 0x0005, /* R208 */ + 0x0008, /* R209 */ + 0x0008, /* R210 */ + 0x0000, /* R211 */ + 0x013B, /* R212 */ + 0x0000, /* R213 */ + 0x0000, /* R214 */ + 0x0000, /* R215 */ + 0x0000, /* R216 */ + 0x0070, /* R217 */ + 0x0000, /* R218 */ + 0x0000, /* R219 */ + 0x0000, /* R220 */ + 0x0000, /* R221 */ + 0x0000, /* R222 */ + 0x0003, /* R223 */ + 0x0000, /* R224 */ + 0x0000, /* R225 */ + 0x0001, /* R226 */ + 0x0008, /* R227 */ + 0x0000, /* R228 */ + 0x0000, /* R229 */ + 0x0000, /* R230 */ + 0x0000, /* R231 */ + 0x0004, /* R232 */ + 0x0000, /* R233 */ + 0x0000, /* R234 */ + 0x0000, /* R235 */ + 0x0000, /* R236 */ + 0x0000, /* R237 */ + 0x0080, /* R238 */ + 0x0000, /* R239 */ + 0x0000, /* R240 */ + 0x0000, /* R241 */ + 0x0000, /* R242 */ + 0x0000, /* R243 */ + 0x0000, /* R244 */ + 0x0052, /* R245 */ + 0x0110, /* R246 */ + 0x0040, /* R247 */ + 0x0000, /* R248 */ + 0x0030, /* R249 */ + 0x0000, /* R250 */ + 0x0000, /* R251 */ + 0x0001, /* R252 - General test 1 */ +}; + +struct wm8961_priv { + struct snd_soc_codec codec; + int sysclk; + u16 reg_cache[WM8961_MAX_REGISTER]; +}; + +static int wm8961_reg_is_volatile(int reg) +{ + switch (reg) { + case WM8961_WRITE_SEQUENCER_7: + case WM8961_DC_SERVO_1: + return 1; + + default: + return 0; + } +} + +static unsigned int wm8961_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + BUG_ON(reg > WM8961_MAX_REGISTER); + return cache[reg]; +} + +static unsigned int wm8961_read_hw(struct snd_soc_codec *codec, u8 reg) +{ + struct i2c_msg xfer[2]; + u16 data; + int ret; + struct i2c_client *client = codec->control_data; + + BUG_ON(reg > WM8961_MAX_REGISTER); + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 1; + xfer[0].buf = ® + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 2; + xfer[1].buf = (u8 *)&data; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret != 2) { + dev_err(&client->dev, "i2c_transfer() returned %d\n", ret); + return 0; + } + + return (data >> 8) | ((data & 0xff) << 8); +} + +static unsigned int wm8961_read(struct snd_soc_codec *codec, unsigned int reg) +{ + if (wm8961_reg_is_volatile(reg)) + return wm8961_read_hw(codec, reg); + else + return wm8961_read_reg_cache(codec, reg); +} + +static int wm8961_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 *cache = codec->reg_cache; + u8 data[3]; + + BUG_ON(reg > WM8961_MAX_REGISTER); + + if (!wm8961_reg_is_volatile(reg)) + cache[reg] = value; + + data[0] = reg; + data[1] = value >> 8; + data[2] = value & 0x00ff; + + if (codec->hw_write(codec->control_data, data, 3) == 3) + return 0; + else + return -EIO; +} + +static int wm8961_reset(struct snd_soc_codec *codec) +{ + return wm8961_write(codec, WM8961_SOFTWARE_RESET, 0); +} + +/* + * The headphone output supports special anti-pop sequences giving + * silent power up and power down. + */ +static int wm8961_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u16 hp_reg = wm8961_read(codec, WM8961_ANALOGUE_HP_0); + u16 cp_reg = wm8961_read(codec, WM8961_CHARGE_PUMP_1); + u16 pwr_reg = wm8961_read(codec, WM8961_PWR_MGMT_2); + u16 dcs_reg = wm8961_read(codec, WM8961_DC_SERVO_1); + int timeout = 500; + + if (event & SND_SOC_DAPM_POST_PMU) { + /* Make sure the output is shorted */ + hp_reg &= ~(WM8961_HPR_RMV_SHORT | WM8961_HPL_RMV_SHORT); + wm8961_write(codec, WM8961_ANALOGUE_HP_0, hp_reg); + + /* Enable the charge pump */ + cp_reg |= WM8961_CP_ENA; + wm8961_write(codec, WM8961_CHARGE_PUMP_1, cp_reg); + mdelay(5); + + /* Enable the PGA */ + pwr_reg |= WM8961_LOUT1_PGA | WM8961_ROUT1_PGA; + wm8961_write(codec, WM8961_PWR_MGMT_2, pwr_reg); + + /* Enable the amplifier */ + hp_reg |= WM8961_HPR_ENA | WM8961_HPL_ENA; + wm8961_write(codec, WM8961_ANALOGUE_HP_0, hp_reg); + + /* Second stage enable */ + hp_reg |= WM8961_HPR_ENA_DLY | WM8961_HPL_ENA_DLY; + wm8961_write(codec, WM8961_ANALOGUE_HP_0, hp_reg); + + /* Enable the DC servo & trigger startup */ + dcs_reg |= + WM8961_DCS_ENA_CHAN_HPR | WM8961_DCS_TRIG_STARTUP_HPR | + WM8961_DCS_ENA_CHAN_HPL | WM8961_DCS_TRIG_STARTUP_HPL; + dev_dbg(codec->dev, "Enabling DC servo\n"); + + wm8961_write(codec, WM8961_DC_SERVO_1, dcs_reg); + do { + msleep(1); + dcs_reg = wm8961_read(codec, WM8961_DC_SERVO_1); + } while (--timeout && + dcs_reg & (WM8961_DCS_TRIG_STARTUP_HPR | + WM8961_DCS_TRIG_STARTUP_HPL)); + if (dcs_reg & (WM8961_DCS_TRIG_STARTUP_HPR | + WM8961_DCS_TRIG_STARTUP_HPL)) + dev_err(codec->dev, "DC servo timed out\n"); + else + dev_dbg(codec->dev, "DC servo startup complete\n"); + + /* Enable the output stage */ + hp_reg |= WM8961_HPR_ENA_OUTP | WM8961_HPL_ENA_OUTP; + wm8961_write(codec, WM8961_ANALOGUE_HP_0, hp_reg); + + /* Remove the short on the output stage */ + hp_reg |= WM8961_HPR_RMV_SHORT | WM8961_HPL_RMV_SHORT; + wm8961_write(codec, WM8961_ANALOGUE_HP_0, hp_reg); + } + + if (event & SND_SOC_DAPM_PRE_PMD) { + /* Short the output */ + hp_reg &= ~(WM8961_HPR_RMV_SHORT | WM8961_HPL_RMV_SHORT); + wm8961_write(codec, WM8961_ANALOGUE_HP_0, hp_reg); + + /* Disable the output stage */ + hp_reg &= ~(WM8961_HPR_ENA_OUTP | WM8961_HPL_ENA_OUTP); + wm8961_write(codec, WM8961_ANALOGUE_HP_0, hp_reg); + + /* Disable DC offset cancellation */ + dcs_reg &= ~(WM8961_DCS_ENA_CHAN_HPR | + WM8961_DCS_ENA_CHAN_HPL); + wm8961_write(codec, WM8961_DC_SERVO_1, dcs_reg); + + /* Finish up */ + hp_reg &= ~(WM8961_HPR_ENA_DLY | WM8961_HPR_ENA | + WM8961_HPL_ENA_DLY | WM8961_HPL_ENA); + wm8961_write(codec, WM8961_ANALOGUE_HP_0, hp_reg); + + /* Disable the PGA */ + pwr_reg &= ~(WM8961_LOUT1_PGA | WM8961_ROUT1_PGA); + wm8961_write(codec, WM8961_PWR_MGMT_2, pwr_reg); + + /* Disable the charge pump */ + dev_dbg(codec->dev, "Disabling charge pump\n"); + wm8961_write(codec, WM8961_CHARGE_PUMP_1, + cp_reg & ~WM8961_CP_ENA); + } + + return 0; +} + +static int wm8961_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u16 pwr_reg = wm8961_read(codec, WM8961_PWR_MGMT_2); + u16 spk_reg = wm8961_read(codec, WM8961_CLASS_D_CONTROL_1); + + if (event & SND_SOC_DAPM_POST_PMU) { + /* Enable the PGA */ + pwr_reg |= WM8961_SPKL_PGA | WM8961_SPKR_PGA; + wm8961_write(codec, WM8961_PWR_MGMT_2, pwr_reg); + + /* Enable the amplifier */ + spk_reg |= WM8961_SPKL_ENA | WM8961_SPKR_ENA; + wm8961_write(codec, WM8961_CLASS_D_CONTROL_1, spk_reg); + } + + if (event & SND_SOC_DAPM_PRE_PMD) { + /* Enable the amplifier */ + spk_reg &= ~(WM8961_SPKL_ENA | WM8961_SPKR_ENA); + wm8961_write(codec, WM8961_CLASS_D_CONTROL_1, spk_reg); + + /* Enable the PGA */ + pwr_reg &= ~(WM8961_SPKL_PGA | WM8961_SPKR_PGA); + wm8961_write(codec, WM8961_PWR_MGMT_2, pwr_reg); + } + + return 0; +} + +static const char *adc_hpf_text[] = { + "Hi-fi", "Voice 1", "Voice 2", "Voice 3", +}; + +static const struct soc_enum adc_hpf = + SOC_ENUM_SINGLE(WM8961_ADC_DAC_CONTROL_2, 7, 4, adc_hpf_text); + +static const char *dac_deemph_text[] = { + "None", "32kHz", "44.1kHz", "48kHz", +}; + +static const struct soc_enum dac_deemph = + SOC_ENUM_SINGLE(WM8961_ADC_DAC_CONTROL_1, 1, 4, dac_deemph_text); + +static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); +static const DECLARE_TLV_DB_SCALE(hp_sec_tlv, -700, 100, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -7200, 75, 1); +static const DECLARE_TLV_DB_SCALE(sidetone_tlv, -3600, 300, 0); +static unsigned int boost_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(13, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(20, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(29, 0, 0), +}; +static const DECLARE_TLV_DB_SCALE(pga_tlv, -2325, 75, 0); + +static const struct snd_kcontrol_new wm8961_snd_controls[] = { +SOC_DOUBLE_R_TLV("Headphone Volume", WM8961_LOUT1_VOLUME, WM8961_ROUT1_VOLUME, + 0, 127, 0, out_tlv), +SOC_DOUBLE_TLV("Headphone Secondary Volume", WM8961_ANALOGUE_HP_2, + 6, 3, 7, 0, hp_sec_tlv), +SOC_DOUBLE_R("Headphone ZC Switch", WM8961_LOUT1_VOLUME, WM8961_ROUT1_VOLUME, + 7, 1, 0), + +SOC_DOUBLE_R_TLV("Speaker Volume", WM8961_LOUT2_VOLUME, WM8961_ROUT2_VOLUME, + 0, 127, 0, out_tlv), +SOC_DOUBLE_R("Speaker ZC Switch", WM8961_LOUT2_VOLUME, WM8961_ROUT2_VOLUME, + 7, 1, 0), +SOC_SINGLE("Speaker AC Gain", WM8961_CLASS_D_CONTROL_2, 0, 7, 0), + +SOC_SINGLE("DAC x128 OSR Switch", WM8961_ADC_DAC_CONTROL_2, 0, 1, 0), +SOC_ENUM("DAC Deemphasis", dac_deemph), +SOC_SINGLE("DAC Soft Mute Switch", WM8961_ADC_DAC_CONTROL_2, 3, 1, 0), + +SOC_DOUBLE_R_TLV("Sidetone Volume", WM8961_DSP_SIDETONE_0, + WM8961_DSP_SIDETONE_1, 4, 12, 0, sidetone_tlv), + +SOC_SINGLE("ADC High Pass Filter Switch", WM8961_ADC_DAC_CONTROL_1, 0, 1, 0), +SOC_ENUM("ADC High Pass Filter Mode", adc_hpf), + +SOC_DOUBLE_R_TLV("Capture Volume", + WM8961_LEFT_ADC_VOLUME, WM8961_RIGHT_ADC_VOLUME, + 1, 119, 0, adc_tlv), +SOC_DOUBLE_R_TLV("Capture Boost Volume", + WM8961_ADCL_SIGNAL_PATH, WM8961_ADCR_SIGNAL_PATH, + 4, 3, 0, boost_tlv), +SOC_DOUBLE_R_TLV("Capture PGA Volume", + WM8961_LEFT_INPUT_VOLUME, WM8961_RIGHT_INPUT_VOLUME, + 0, 62, 0, pga_tlv), +SOC_DOUBLE_R("Capture PGA ZC Switch", + WM8961_LEFT_INPUT_VOLUME, WM8961_RIGHT_INPUT_VOLUME, + 6, 1, 1), +SOC_DOUBLE_R("Capture PGA Switch", + WM8961_LEFT_INPUT_VOLUME, WM8961_RIGHT_INPUT_VOLUME, + 7, 1, 1), +}; + +static const char *sidetone_text[] = { + "None", "Left", "Right" +}; + +static const struct soc_enum dacl_sidetone = + SOC_ENUM_SINGLE(WM8961_DSP_SIDETONE_0, 2, 3, sidetone_text); + +static const struct soc_enum dacr_sidetone = + SOC_ENUM_SINGLE(WM8961_DSP_SIDETONE_1, 2, 3, sidetone_text); + +static const struct snd_kcontrol_new dacl_mux = + SOC_DAPM_ENUM("DACL Sidetone", dacl_sidetone); + +static const struct snd_kcontrol_new dacr_mux = + SOC_DAPM_ENUM("DACR Sidetone", dacr_sidetone); + +static const struct snd_soc_dapm_widget wm8961_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("LINPUT"), +SND_SOC_DAPM_INPUT("RINPUT"), + +SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8961_CLOCKING2, 4, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Left Input", WM8961_PWR_MGMT_1, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Input", WM8961_PWR_MGMT_1, 4, 0, NULL, 0), + +SND_SOC_DAPM_ADC("ADCL", "HiFi Capture", WM8961_PWR_MGMT_1, 3, 0), +SND_SOC_DAPM_ADC("ADCR", "HiFi Capture", WM8961_PWR_MGMT_1, 2, 0), + +SND_SOC_DAPM_MICBIAS("MICBIAS", WM8961_PWR_MGMT_1, 1, 0), + +SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &dacl_mux), +SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &dacr_mux), + +SND_SOC_DAPM_DAC("DACL", "HiFi Playback", WM8961_PWR_MGMT_2, 8, 0), +SND_SOC_DAPM_DAC("DACR", "HiFi Playback", WM8961_PWR_MGMT_2, 7, 0), + +/* Handle as a mono path for DCS */ +SND_SOC_DAPM_PGA_E("Headphone Output", SND_SOC_NOPM, + 4, 0, NULL, 0, wm8961_hp_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_E("Speaker Output", SND_SOC_NOPM, + 4, 0, NULL, 0, wm8961_spk_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + +SND_SOC_DAPM_OUTPUT("HP_L"), +SND_SOC_DAPM_OUTPUT("HP_R"), +SND_SOC_DAPM_OUTPUT("SPK_LN"), +SND_SOC_DAPM_OUTPUT("SPK_LP"), +SND_SOC_DAPM_OUTPUT("SPK_RN"), +SND_SOC_DAPM_OUTPUT("SPK_RP"), +}; + + +static const struct snd_soc_dapm_route audio_paths[] = { + { "DACL", NULL, "CLK_DSP" }, + { "DACL", NULL, "DACL Sidetone" }, + { "DACR", NULL, "CLK_DSP" }, + { "DACR", NULL, "DACR Sidetone" }, + + { "DACL Sidetone", "Left", "ADCL" }, + { "DACL Sidetone", "Right", "ADCR" }, + + { "DACR Sidetone", "Left", "ADCL" }, + { "DACR Sidetone", "Right", "ADCR" }, + + { "HP_L", NULL, "Headphone Output" }, + { "HP_R", NULL, "Headphone Output" }, + { "Headphone Output", NULL, "DACL" }, + { "Headphone Output", NULL, "DACR" }, + + { "SPK_LN", NULL, "Speaker Output" }, + { "SPK_LP", NULL, "Speaker Output" }, + { "SPK_RN", NULL, "Speaker Output" }, + { "SPK_RP", NULL, "Speaker Output" }, + + { "Speaker Output", NULL, "DACL" }, + { "Speaker Output", NULL, "DACR" }, + + { "ADCL", NULL, "Left Input" }, + { "ADCL", NULL, "CLK_DSP" }, + { "ADCR", NULL, "Right Input" }, + { "ADCR", NULL, "CLK_DSP" }, + + { "Left Input", NULL, "LINPUT" }, + { "Right Input", NULL, "RINPUT" }, + +}; + +/* Values for CLK_SYS_RATE */ +static struct { + int ratio; + u16 val; +} wm8961_clk_sys_ratio[] = { + { 64, 0 }, + { 128, 1 }, + { 192, 2 }, + { 256, 3 }, + { 384, 4 }, + { 512, 5 }, + { 768, 6 }, + { 1024, 7 }, + { 1408, 8 }, + { 1536, 9 }, +}; + +/* Values for SAMPLE_RATE */ +static struct { + int rate; + u16 val; +} wm8961_srate[] = { + { 48000, 0 }, + { 44100, 0 }, + { 32000, 1 }, + { 22050, 2 }, + { 24000, 2 }, + { 16000, 3 }, + { 11250, 4 }, + { 12000, 4 }, + { 8000, 5 }, +}; + +static int wm8961_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8961_priv *wm8961 = codec->private_data; + int i, best, target, fs; + u16 reg; + + fs = params_rate(params); + + if (!wm8961->sysclk) { + dev_err(codec->dev, "MCLK has not been specified\n"); + return -EINVAL; + } + + /* Find the closest sample rate for the filters */ + best = 0; + for (i = 0; i < ARRAY_SIZE(wm8961_srate); i++) { + if (abs(wm8961_srate[i].rate - fs) < + abs(wm8961_srate[best].rate - fs)) + best = i; + } + reg = wm8961_read(codec, WM8961_ADDITIONAL_CONTROL_3); + reg &= ~WM8961_SAMPLE_RATE_MASK; + reg |= wm8961_srate[best].val; + wm8961_write(codec, WM8961_ADDITIONAL_CONTROL_3, reg); + dev_dbg(codec->dev, "Selected SRATE %dHz for %dHz\n", + wm8961_srate[best].rate, fs); + + /* Select a CLK_SYS/fs ratio equal to or higher than required */ + target = wm8961->sysclk / fs; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && target < 64) { + dev_err(codec->dev, + "SYSCLK must be at least 64*fs for DAC\n"); + return -EINVAL; + } + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && target < 256) { + dev_err(codec->dev, + "SYSCLK must be at least 256*fs for ADC\n"); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(wm8961_clk_sys_ratio); i++) { + if (wm8961_clk_sys_ratio[i].ratio >= target) + break; + } + if (i == ARRAY_SIZE(wm8961_clk_sys_ratio)) { + dev_err(codec->dev, "Unable to generate CLK_SYS_RATE\n"); + return -EINVAL; + } + dev_dbg(codec->dev, "Selected CLK_SYS_RATE of %d for %d/%d=%d\n", + wm8961_clk_sys_ratio[i].ratio, wm8961->sysclk, fs, + wm8961->sysclk / fs); + + reg = wm8961_read(codec, WM8961_CLOCKING_4); + reg &= ~WM8961_CLK_SYS_RATE_MASK; + reg |= wm8961_clk_sys_ratio[i].val << WM8961_CLK_SYS_RATE_SHIFT; + wm8961_write(codec, WM8961_CLOCKING_4, reg); + + reg = wm8961_read(codec, WM8961_AUDIO_INTERFACE_0); + reg &= ~WM8961_WL_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + reg |= 1 << WM8961_WL_SHIFT; + break; + case SNDRV_PCM_FORMAT_S24_LE: + reg |= 2 << WM8961_WL_SHIFT; + break; + case SNDRV_PCM_FORMAT_S32_LE: + reg |= 3 << WM8961_WL_SHIFT; + break; + default: + return -EINVAL; + } + wm8961_write(codec, WM8961_AUDIO_INTERFACE_0, reg); + + /* Sloping stop-band filter is recommended for <= 24kHz */ + reg = wm8961_read(codec, WM8961_ADC_DAC_CONTROL_2); + if (fs <= 24000) + reg |= WM8961_DACSLOPE; + else + reg &= WM8961_DACSLOPE; + wm8961_write(codec, WM8961_ADC_DAC_CONTROL_2, reg); + + return 0; +} + +static int wm8961_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, + int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8961_priv *wm8961 = codec->private_data; + u16 reg = wm8961_read(codec, WM8961_CLOCKING1); + + if (freq > 33000000) { + dev_err(codec->dev, "MCLK must be <33MHz\n"); + return -EINVAL; + } + + if (freq > 16500000) { + dev_dbg(codec->dev, "Using MCLK/2 for %dHz MCLK\n", freq); + reg |= WM8961_MCLKDIV; + freq /= 2; + } else { + dev_dbg(codec->dev, "Using MCLK/1 for %dHz MCLK\n", freq); + reg &= WM8961_MCLKDIV; + } + + wm8961_write(codec, WM8961_CLOCKING1, reg); + + wm8961->sysclk = freq; + + return 0; +} + +static int wm8961_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u16 aif = wm8961_read(codec, WM8961_AUDIO_INTERFACE_0); + + aif &= ~(WM8961_BCLKINV | WM8961_LRP | + WM8961_MS | WM8961_FORMAT_MASK); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + aif |= WM8961_MS; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + break; + + case SND_SOC_DAIFMT_LEFT_J: + aif |= 1; + break; + + case SND_SOC_DAIFMT_I2S: + aif |= 2; + break; + + case SND_SOC_DAIFMT_DSP_B: + aif |= WM8961_LRP; + case SND_SOC_DAIFMT_DSP_A: + aif |= 3; + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + case SND_SOC_DAIFMT_IB_NF: + break; + default: + return -EINVAL; + } + break; + + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + aif |= WM8961_LRP; + break; + case SND_SOC_DAIFMT_IB_NF: + aif |= WM8961_BCLKINV; + break; + case SND_SOC_DAIFMT_IB_IF: + aif |= WM8961_BCLKINV | WM8961_LRP; + break; + default: + return -EINVAL; + } + + return wm8961_write(codec, WM8961_AUDIO_INTERFACE_0, aif); +} + +static int wm8961_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + u16 reg = wm8961_read(codec, WM8961_ADDITIONAL_CONTROL_2); + + if (tristate) + reg |= WM8961_TRIS; + else + reg &= ~WM8961_TRIS; + + return wm8961_write(codec, WM8961_ADDITIONAL_CONTROL_2, reg); +} + +static int wm8961_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 reg = wm8961_read(codec, WM8961_ADC_DAC_CONTROL_1); + + if (mute) + reg |= WM8961_DACMU; + else + reg &= ~WM8961_DACMU; + + msleep(17); + + return wm8961_write(codec, WM8961_ADC_DAC_CONTROL_1, reg); +} + +static int wm8961_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct snd_soc_codec *codec = dai->codec; + u16 reg; + + switch (div_id) { + case WM8961_BCLK: + reg = wm8961_read(codec, WM8961_CLOCKING2); + reg &= ~WM8961_BCLKDIV_MASK; + reg |= div; + wm8961_write(codec, WM8961_CLOCKING2, reg); + break; + + case WM8961_LRCLK: + reg = wm8961_read(codec, WM8961_AUDIO_INTERFACE_2); + reg &= ~WM8961_LRCLK_RATE_MASK; + reg |= div; + wm8961_write(codec, WM8961_AUDIO_INTERFACE_2, reg); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int wm8961_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg; + + /* This is all slightly unusual since we have no bypass paths + * and the output amplifier structure means we can just slam + * the biases straight up rather than having to ramp them + * slowly. + */ + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + if (codec->bias_level == SND_SOC_BIAS_STANDBY) { + /* Enable bias generation */ + reg = wm8961_read(codec, WM8961_ANTI_POP); + reg |= WM8961_BUFIOEN | WM8961_BUFDCOPEN; + wm8961_write(codec, WM8961_ANTI_POP, reg); + + /* VMID=2*50k, VREF */ + reg = wm8961_read(codec, WM8961_PWR_MGMT_1); + reg &= ~WM8961_VMIDSEL_MASK; + reg |= (1 << WM8961_VMIDSEL_SHIFT) | WM8961_VREF; + wm8961_write(codec, WM8961_PWR_MGMT_1, reg); + } + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_PREPARE) { + /* VREF off */ + reg = wm8961_read(codec, WM8961_PWR_MGMT_1); + reg &= ~WM8961_VREF; + wm8961_write(codec, WM8961_PWR_MGMT_1, reg); + + /* Bias generation off */ + reg = wm8961_read(codec, WM8961_ANTI_POP); + reg &= ~(WM8961_BUFIOEN | WM8961_BUFDCOPEN); + wm8961_write(codec, WM8961_ANTI_POP, reg); + + /* VMID off */ + reg = wm8961_read(codec, WM8961_PWR_MGMT_1); + reg &= ~WM8961_VMIDSEL_MASK; + wm8961_write(codec, WM8961_PWR_MGMT_1, reg); + } + break; + + case SND_SOC_BIAS_OFF: + break; + } + + codec->bias_level = level; + + return 0; +} + + +#define WM8961_RATES SNDRV_PCM_RATE_8000_48000 + +#define WM8961_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops wm8961_dai_ops = { + .hw_params = wm8961_hw_params, + .set_sysclk = wm8961_set_sysclk, + .set_fmt = wm8961_set_fmt, + .digital_mute = wm8961_digital_mute, + .set_tristate = wm8961_set_tristate, + .set_clkdiv = wm8961_set_clkdiv, +}; + +struct snd_soc_dai wm8961_dai = { + .name = "WM8961", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8961_RATES, + .formats = WM8961_FORMATS,}, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8961_RATES, + .formats = WM8961_FORMATS,}, + .ops = &wm8961_dai_ops, +}; +EXPORT_SYMBOL_GPL(wm8961_dai); + + +static struct snd_soc_codec *wm8961_codec; + +static int wm8961_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8961_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8961_codec; + codec = wm8961_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8961_snd_controls, + ARRAY_SIZE(wm8961_snd_controls)); + snd_soc_dapm_new_controls(codec, wm8961_dapm_widgets, + ARRAY_SIZE(wm8961_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); + snd_soc_dapm_new_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +static int wm8961_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +#ifdef CONFIG_PM +static int wm8961_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8961_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8961_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + u16 *reg_cache = codec->reg_cache; + int i; + + for (i = 0; i < codec->reg_cache_size; i++) { + if (i == WM8961_SOFTWARE_RESET) + continue; + + wm8961_write(codec, i, reg_cache[i]); + } + + wm8961_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define wm8961_suspend NULL +#define wm8961_resume NULL +#endif + +struct snd_soc_codec_device soc_codec_dev_wm8961 = { + .probe = wm8961_probe, + .remove = wm8961_remove, + .suspend = wm8961_suspend, + .resume = wm8961_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8961); + +static int wm8961_register(struct wm8961_priv *wm8961) +{ + struct snd_soc_codec *codec = &wm8961->codec; + int ret; + u16 reg; + + if (wm8961_codec) { + dev_err(codec->dev, "Another WM8961 is registered\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8961; + codec->name = "WM8961"; + codec->owner = THIS_MODULE; + codec->read = wm8961_read; + codec->write = wm8961_write; + codec->dai = &wm8961_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(wm8961->reg_cache); + codec->reg_cache = &wm8961->reg_cache; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8961_set_bias_level; + + memcpy(codec->reg_cache, wm8961_reg_defaults, + sizeof(wm8961_reg_defaults)); + + reg = wm8961_read_hw(codec, WM8961_SOFTWARE_RESET); + if (reg != 0x1801) { + dev_err(codec->dev, "Device is not a WM8961: ID=0x%x\n", reg); + ret = -EINVAL; + goto err; + } + + reg = wm8961_read_hw(codec, WM8961_RIGHT_INPUT_VOLUME); + dev_info(codec->dev, "WM8961 family %d revision %c\n", + (reg & WM8961_DEVICE_ID_MASK) >> WM8961_DEVICE_ID_SHIFT, + ((reg & WM8961_CHIP_REV_MASK) >> WM8961_CHIP_REV_SHIFT) + + 'A'); + + ret = wm8961_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + return ret; + } + + /* Enable class W */ + reg = wm8961_read(codec, WM8961_CHARGE_PUMP_B); + reg |= WM8961_CP_DYN_PWR_MASK; + wm8961_write(codec, WM8961_CHARGE_PUMP_B, reg); + + /* Latch volume update bits (right channel only, we always + * write both out) and default ZC on. */ + reg = wm8961_read(codec, WM8961_ROUT1_VOLUME); + wm8961_write(codec, WM8961_ROUT1_VOLUME, + reg | WM8961_LO1ZC | WM8961_OUT1VU); + wm8961_write(codec, WM8961_LOUT1_VOLUME, reg | WM8961_LO1ZC); + reg = wm8961_read(codec, WM8961_ROUT2_VOLUME); + wm8961_write(codec, WM8961_ROUT2_VOLUME, + reg | WM8961_SPKRZC | WM8961_SPKVU); + wm8961_write(codec, WM8961_LOUT2_VOLUME, reg | WM8961_SPKLZC); + + reg = wm8961_read(codec, WM8961_RIGHT_ADC_VOLUME); + wm8961_write(codec, WM8961_RIGHT_ADC_VOLUME, reg | WM8961_ADCVU); + reg = wm8961_read(codec, WM8961_RIGHT_INPUT_VOLUME); + wm8961_write(codec, WM8961_RIGHT_INPUT_VOLUME, reg | WM8961_IPVU); + + /* Use soft mute by default */ + reg = wm8961_read(codec, WM8961_ADC_DAC_CONTROL_2); + reg |= WM8961_DACSMM; + wm8961_write(codec, WM8961_ADC_DAC_CONTROL_2, reg); + + /* Use automatic clocking mode by default; for now this is all + * we support. + */ + reg = wm8961_read(codec, WM8961_CLOCKING_3); + reg &= ~WM8961_MANUAL_MODE; + wm8961_write(codec, WM8961_CLOCKING_3, reg); + + wm8961_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + wm8961_dai.dev = codec->dev; + + wm8961_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8961_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; + +err: + kfree(wm8961); + return ret; +} + +static void wm8961_unregister(struct wm8961_priv *wm8961) +{ + wm8961_set_bias_level(&wm8961->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8961_dai); + snd_soc_unregister_codec(&wm8961->codec); + kfree(wm8961); + wm8961_codec = NULL; +} + +static __devinit int wm8961_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8961_priv *wm8961; + struct snd_soc_codec *codec; + + wm8961 = kzalloc(sizeof(struct wm8961_priv), GFP_KERNEL); + if (wm8961 == NULL) + return -ENOMEM; + + codec = &wm8961->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8961); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8961_register(wm8961); +} + +static __devexit int wm8961_i2c_remove(struct i2c_client *client) +{ + struct wm8961_priv *wm8961 = i2c_get_clientdata(client); + wm8961_unregister(wm8961); + return 0; +} + +static const struct i2c_device_id wm8961_i2c_id[] = { + { "wm8961", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8961_i2c_id); + +static struct i2c_driver wm8961_i2c_driver = { + .driver = { + .name = "wm8961", + .owner = THIS_MODULE, + }, + .probe = wm8961_i2c_probe, + .remove = __devexit_p(wm8961_i2c_remove), + .id_table = wm8961_i2c_id, +}; + +static int __init wm8961_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&wm8961_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8961 I2C driver: %d\n", + ret); + } + + return ret; +} +module_init(wm8961_modinit); + +static void __exit wm8961_exit(void) +{ + i2c_del_driver(&wm8961_i2c_driver); +} +module_exit(wm8961_exit); + + +MODULE_DESCRIPTION("ASoC WM8961 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8961.h b/sound/soc/codecs/wm8961.h new file mode 100644 index 000000000000..5513bfd720d6 --- /dev/null +++ b/sound/soc/codecs/wm8961.h @@ -0,0 +1,866 @@ +/* + * wm8961.h -- WM8961 Soc Audio driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8961_H +#define _WM8961_H + +#include + +extern struct snd_soc_codec_device soc_codec_dev_wm8961; +extern struct snd_soc_dai wm8961_dai; + +#define WM8961_BCLK 1 +#define WM8961_LRCLK 2 + +#define WM8961_BCLK_DIV_1 0 +#define WM8961_BCLK_DIV_1_5 1 +#define WM8961_BCLK_DIV_2 2 +#define WM8961_BCLK_DIV_3 3 +#define WM8961_BCLK_DIV_4 4 +#define WM8961_BCLK_DIV_5_5 5 +#define WM8961_BCLK_DIV_6 6 +#define WM8961_BCLK_DIV_8 7 +#define WM8961_BCLK_DIV_11 8 +#define WM8961_BCLK_DIV_12 9 +#define WM8961_BCLK_DIV_16 10 +#define WM8961_BCLK_DIV_24 11 +#define WM8961_BCLK_DIV_32 13 + + +/* + * Register values. + */ +#define WM8961_LEFT_INPUT_VOLUME 0x00 +#define WM8961_RIGHT_INPUT_VOLUME 0x01 +#define WM8961_LOUT1_VOLUME 0x02 +#define WM8961_ROUT1_VOLUME 0x03 +#define WM8961_CLOCKING1 0x04 +#define WM8961_ADC_DAC_CONTROL_1 0x05 +#define WM8961_ADC_DAC_CONTROL_2 0x06 +#define WM8961_AUDIO_INTERFACE_0 0x07 +#define WM8961_CLOCKING2 0x08 +#define WM8961_AUDIO_INTERFACE_1 0x09 +#define WM8961_LEFT_DAC_VOLUME 0x0A +#define WM8961_RIGHT_DAC_VOLUME 0x0B +#define WM8961_AUDIO_INTERFACE_2 0x0E +#define WM8961_SOFTWARE_RESET 0x0F +#define WM8961_ALC1 0x11 +#define WM8961_ALC2 0x12 +#define WM8961_ALC3 0x13 +#define WM8961_NOISE_GATE 0x14 +#define WM8961_LEFT_ADC_VOLUME 0x15 +#define WM8961_RIGHT_ADC_VOLUME 0x16 +#define WM8961_ADDITIONAL_CONTROL_1 0x17 +#define WM8961_ADDITIONAL_CONTROL_2 0x18 +#define WM8961_PWR_MGMT_1 0x19 +#define WM8961_PWR_MGMT_2 0x1A +#define WM8961_ADDITIONAL_CONTROL_3 0x1B +#define WM8961_ANTI_POP 0x1C +#define WM8961_CLOCKING_3 0x1E +#define WM8961_ADCL_SIGNAL_PATH 0x20 +#define WM8961_ADCR_SIGNAL_PATH 0x21 +#define WM8961_LOUT2_VOLUME 0x28 +#define WM8961_ROUT2_VOLUME 0x29 +#define WM8961_PWR_MGMT_3 0x2F +#define WM8961_ADDITIONAL_CONTROL_4 0x30 +#define WM8961_CLASS_D_CONTROL_1 0x31 +#define WM8961_CLASS_D_CONTROL_2 0x33 +#define WM8961_CLOCKING_4 0x38 +#define WM8961_DSP_SIDETONE_0 0x39 +#define WM8961_DSP_SIDETONE_1 0x3A +#define WM8961_DC_SERVO_0 0x3C +#define WM8961_DC_SERVO_1 0x3D +#define WM8961_DC_SERVO_3 0x3F +#define WM8961_DC_SERVO_5 0x41 +#define WM8961_ANALOGUE_PGA_BIAS 0x44 +#define WM8961_ANALOGUE_HP_0 0x45 +#define WM8961_ANALOGUE_HP_2 0x47 +#define WM8961_CHARGE_PUMP_1 0x48 +#define WM8961_CHARGE_PUMP_B 0x52 +#define WM8961_WRITE_SEQUENCER_1 0x57 +#define WM8961_WRITE_SEQUENCER_2 0x58 +#define WM8961_WRITE_SEQUENCER_3 0x59 +#define WM8961_WRITE_SEQUENCER_4 0x5A +#define WM8961_WRITE_SEQUENCER_5 0x5B +#define WM8961_WRITE_SEQUENCER_6 0x5C +#define WM8961_WRITE_SEQUENCER_7 0x5D +#define WM8961_GENERAL_TEST_1 0xFC + + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - Left Input volume + */ +#define WM8961_IPVU 0x0100 /* IPVU */ +#define WM8961_IPVU_MASK 0x0100 /* IPVU */ +#define WM8961_IPVU_SHIFT 8 /* IPVU */ +#define WM8961_IPVU_WIDTH 1 /* IPVU */ +#define WM8961_LINMUTE 0x0080 /* LINMUTE */ +#define WM8961_LINMUTE_MASK 0x0080 /* LINMUTE */ +#define WM8961_LINMUTE_SHIFT 7 /* LINMUTE */ +#define WM8961_LINMUTE_WIDTH 1 /* LINMUTE */ +#define WM8961_LIZC 0x0040 /* LIZC */ +#define WM8961_LIZC_MASK 0x0040 /* LIZC */ +#define WM8961_LIZC_SHIFT 6 /* LIZC */ +#define WM8961_LIZC_WIDTH 1 /* LIZC */ +#define WM8961_LINVOL_MASK 0x003F /* LINVOL - [5:0] */ +#define WM8961_LINVOL_SHIFT 0 /* LINVOL - [5:0] */ +#define WM8961_LINVOL_WIDTH 6 /* LINVOL - [5:0] */ + +/* + * R1 (0x01) - Right Input volume + */ +#define WM8961_DEVICE_ID_MASK 0xF000 /* DEVICE_ID - [15:12] */ +#define WM8961_DEVICE_ID_SHIFT 12 /* DEVICE_ID - [15:12] */ +#define WM8961_DEVICE_ID_WIDTH 4 /* DEVICE_ID - [15:12] */ +#define WM8961_CHIP_REV_MASK 0x0E00 /* CHIP_REV - [11:9] */ +#define WM8961_CHIP_REV_SHIFT 9 /* CHIP_REV - [11:9] */ +#define WM8961_CHIP_REV_WIDTH 3 /* CHIP_REV - [11:9] */ +#define WM8961_IPVU 0x0100 /* IPVU */ +#define WM8961_IPVU_MASK 0x0100 /* IPVU */ +#define WM8961_IPVU_SHIFT 8 /* IPVU */ +#define WM8961_IPVU_WIDTH 1 /* IPVU */ +#define WM8961_RINMUTE 0x0080 /* RINMUTE */ +#define WM8961_RINMUTE_MASK 0x0080 /* RINMUTE */ +#define WM8961_RINMUTE_SHIFT 7 /* RINMUTE */ +#define WM8961_RINMUTE_WIDTH 1 /* RINMUTE */ +#define WM8961_RIZC 0x0040 /* RIZC */ +#define WM8961_RIZC_MASK 0x0040 /* RIZC */ +#define WM8961_RIZC_SHIFT 6 /* RIZC */ +#define WM8961_RIZC_WIDTH 1 /* RIZC */ +#define WM8961_RINVOL_MASK 0x003F /* RINVOL - [5:0] */ +#define WM8961_RINVOL_SHIFT 0 /* RINVOL - [5:0] */ +#define WM8961_RINVOL_WIDTH 6 /* RINVOL - [5:0] */ + +/* + * R2 (0x02) - LOUT1 volume + */ +#define WM8961_OUT1VU 0x0100 /* OUT1VU */ +#define WM8961_OUT1VU_MASK 0x0100 /* OUT1VU */ +#define WM8961_OUT1VU_SHIFT 8 /* OUT1VU */ +#define WM8961_OUT1VU_WIDTH 1 /* OUT1VU */ +#define WM8961_LO1ZC 0x0080 /* LO1ZC */ +#define WM8961_LO1ZC_MASK 0x0080 /* LO1ZC */ +#define WM8961_LO1ZC_SHIFT 7 /* LO1ZC */ +#define WM8961_LO1ZC_WIDTH 1 /* LO1ZC */ +#define WM8961_LOUT1VOL_MASK 0x007F /* LOUT1VOL - [6:0] */ +#define WM8961_LOUT1VOL_SHIFT 0 /* LOUT1VOL - [6:0] */ +#define WM8961_LOUT1VOL_WIDTH 7 /* LOUT1VOL - [6:0] */ + +/* + * R3 (0x03) - ROUT1 volume + */ +#define WM8961_OUT1VU 0x0100 /* OUT1VU */ +#define WM8961_OUT1VU_MASK 0x0100 /* OUT1VU */ +#define WM8961_OUT1VU_SHIFT 8 /* OUT1VU */ +#define WM8961_OUT1VU_WIDTH 1 /* OUT1VU */ +#define WM8961_RO1ZC 0x0080 /* RO1ZC */ +#define WM8961_RO1ZC_MASK 0x0080 /* RO1ZC */ +#define WM8961_RO1ZC_SHIFT 7 /* RO1ZC */ +#define WM8961_RO1ZC_WIDTH 1 /* RO1ZC */ +#define WM8961_ROUT1VOL_MASK 0x007F /* ROUT1VOL - [6:0] */ +#define WM8961_ROUT1VOL_SHIFT 0 /* ROUT1VOL - [6:0] */ +#define WM8961_ROUT1VOL_WIDTH 7 /* ROUT1VOL - [6:0] */ + +/* + * R4 (0x04) - Clocking1 + */ +#define WM8961_ADCDIV_MASK 0x01C0 /* ADCDIV - [8:6] */ +#define WM8961_ADCDIV_SHIFT 6 /* ADCDIV - [8:6] */ +#define WM8961_ADCDIV_WIDTH 3 /* ADCDIV - [8:6] */ +#define WM8961_DACDIV_MASK 0x0038 /* DACDIV - [5:3] */ +#define WM8961_DACDIV_SHIFT 3 /* DACDIV - [5:3] */ +#define WM8961_DACDIV_WIDTH 3 /* DACDIV - [5:3] */ +#define WM8961_MCLKDIV 0x0004 /* MCLKDIV */ +#define WM8961_MCLKDIV_MASK 0x0004 /* MCLKDIV */ +#define WM8961_MCLKDIV_SHIFT 2 /* MCLKDIV */ +#define WM8961_MCLKDIV_WIDTH 1 /* MCLKDIV */ + +/* + * R5 (0x05) - ADC & DAC Control 1 + */ +#define WM8961_ADCPOL_MASK 0x0060 /* ADCPOL - [6:5] */ +#define WM8961_ADCPOL_SHIFT 5 /* ADCPOL - [6:5] */ +#define WM8961_ADCPOL_WIDTH 2 /* ADCPOL - [6:5] */ +#define WM8961_DACMU 0x0008 /* DACMU */ +#define WM8961_DACMU_MASK 0x0008 /* DACMU */ +#define WM8961_DACMU_SHIFT 3 /* DACMU */ +#define WM8961_DACMU_WIDTH 1 /* DACMU */ +#define WM8961_DEEMPH_MASK 0x0006 /* DEEMPH - [2:1] */ +#define WM8961_DEEMPH_SHIFT 1 /* DEEMPH - [2:1] */ +#define WM8961_DEEMPH_WIDTH 2 /* DEEMPH - [2:1] */ +#define WM8961_ADCHPD 0x0001 /* ADCHPD */ +#define WM8961_ADCHPD_MASK 0x0001 /* ADCHPD */ +#define WM8961_ADCHPD_SHIFT 0 /* ADCHPD */ +#define WM8961_ADCHPD_WIDTH 1 /* ADCHPD */ + +/* + * R6 (0x06) - ADC & DAC Control 2 + */ +#define WM8961_ADC_HPF_CUT_MASK 0x0180 /* ADC_HPF_CUT - [8:7] */ +#define WM8961_ADC_HPF_CUT_SHIFT 7 /* ADC_HPF_CUT - [8:7] */ +#define WM8961_ADC_HPF_CUT_WIDTH 2 /* ADC_HPF_CUT - [8:7] */ +#define WM8961_DACPOL_MASK 0x0060 /* DACPOL - [6:5] */ +#define WM8961_DACPOL_SHIFT 5 /* DACPOL - [6:5] */ +#define WM8961_DACPOL_WIDTH 2 /* DACPOL - [6:5] */ +#define WM8961_DACSMM 0x0008 /* DACSMM */ +#define WM8961_DACSMM_MASK 0x0008 /* DACSMM */ +#define WM8961_DACSMM_SHIFT 3 /* DACSMM */ +#define WM8961_DACSMM_WIDTH 1 /* DACSMM */ +#define WM8961_DACMR 0x0004 /* DACMR */ +#define WM8961_DACMR_MASK 0x0004 /* DACMR */ +#define WM8961_DACMR_SHIFT 2 /* DACMR */ +#define WM8961_DACMR_WIDTH 1 /* DACMR */ +#define WM8961_DACSLOPE 0x0002 /* DACSLOPE */ +#define WM8961_DACSLOPE_MASK 0x0002 /* DACSLOPE */ +#define WM8961_DACSLOPE_SHIFT 1 /* DACSLOPE */ +#define WM8961_DACSLOPE_WIDTH 1 /* DACSLOPE */ +#define WM8961_DAC_OSR128 0x0001 /* DAC_OSR128 */ +#define WM8961_DAC_OSR128_MASK 0x0001 /* DAC_OSR128 */ +#define WM8961_DAC_OSR128_SHIFT 0 /* DAC_OSR128 */ +#define WM8961_DAC_OSR128_WIDTH 1 /* DAC_OSR128 */ + +/* + * R7 (0x07) - Audio Interface 0 + */ +#define WM8961_ALRSWAP 0x0100 /* ALRSWAP */ +#define WM8961_ALRSWAP_MASK 0x0100 /* ALRSWAP */ +#define WM8961_ALRSWAP_SHIFT 8 /* ALRSWAP */ +#define WM8961_ALRSWAP_WIDTH 1 /* ALRSWAP */ +#define WM8961_BCLKINV 0x0080 /* BCLKINV */ +#define WM8961_BCLKINV_MASK 0x0080 /* BCLKINV */ +#define WM8961_BCLKINV_SHIFT 7 /* BCLKINV */ +#define WM8961_BCLKINV_WIDTH 1 /* BCLKINV */ +#define WM8961_MS 0x0040 /* MS */ +#define WM8961_MS_MASK 0x0040 /* MS */ +#define WM8961_MS_SHIFT 6 /* MS */ +#define WM8961_MS_WIDTH 1 /* MS */ +#define WM8961_DLRSWAP 0x0020 /* DLRSWAP */ +#define WM8961_DLRSWAP_MASK 0x0020 /* DLRSWAP */ +#define WM8961_DLRSWAP_SHIFT 5 /* DLRSWAP */ +#define WM8961_DLRSWAP_WIDTH 1 /* DLRSWAP */ +#define WM8961_LRP 0x0010 /* LRP */ +#define WM8961_LRP_MASK 0x0010 /* LRP */ +#define WM8961_LRP_SHIFT 4 /* LRP */ +#define WM8961_LRP_WIDTH 1 /* LRP */ +#define WM8961_WL_MASK 0x000C /* WL - [3:2] */ +#define WM8961_WL_SHIFT 2 /* WL - [3:2] */ +#define WM8961_WL_WIDTH 2 /* WL - [3:2] */ +#define WM8961_FORMAT_MASK 0x0003 /* FORMAT - [1:0] */ +#define WM8961_FORMAT_SHIFT 0 /* FORMAT - [1:0] */ +#define WM8961_FORMAT_WIDTH 2 /* FORMAT - [1:0] */ + +/* + * R8 (0x08) - Clocking2 + */ +#define WM8961_DCLKDIV_MASK 0x01C0 /* DCLKDIV - [8:6] */ +#define WM8961_DCLKDIV_SHIFT 6 /* DCLKDIV - [8:6] */ +#define WM8961_DCLKDIV_WIDTH 3 /* DCLKDIV - [8:6] */ +#define WM8961_CLK_SYS_ENA 0x0020 /* CLK_SYS_ENA */ +#define WM8961_CLK_SYS_ENA_MASK 0x0020 /* CLK_SYS_ENA */ +#define WM8961_CLK_SYS_ENA_SHIFT 5 /* CLK_SYS_ENA */ +#define WM8961_CLK_SYS_ENA_WIDTH 1 /* CLK_SYS_ENA */ +#define WM8961_CLK_DSP_ENA 0x0010 /* CLK_DSP_ENA */ +#define WM8961_CLK_DSP_ENA_MASK 0x0010 /* CLK_DSP_ENA */ +#define WM8961_CLK_DSP_ENA_SHIFT 4 /* CLK_DSP_ENA */ +#define WM8961_CLK_DSP_ENA_WIDTH 1 /* CLK_DSP_ENA */ +#define WM8961_BCLKDIV_MASK 0x000F /* BCLKDIV - [3:0] */ +#define WM8961_BCLKDIV_SHIFT 0 /* BCLKDIV - [3:0] */ +#define WM8961_BCLKDIV_WIDTH 4 /* BCLKDIV - [3:0] */ + +/* + * R9 (0x09) - Audio Interface 1 + */ +#define WM8961_DACCOMP_MASK 0x0018 /* DACCOMP - [4:3] */ +#define WM8961_DACCOMP_SHIFT 3 /* DACCOMP - [4:3] */ +#define WM8961_DACCOMP_WIDTH 2 /* DACCOMP - [4:3] */ +#define WM8961_ADCCOMP_MASK 0x0006 /* ADCCOMP - [2:1] */ +#define WM8961_ADCCOMP_SHIFT 1 /* ADCCOMP - [2:1] */ +#define WM8961_ADCCOMP_WIDTH 2 /* ADCCOMP - [2:1] */ +#define WM8961_LOOPBACK 0x0001 /* LOOPBACK */ +#define WM8961_LOOPBACK_MASK 0x0001 /* LOOPBACK */ +#define WM8961_LOOPBACK_SHIFT 0 /* LOOPBACK */ +#define WM8961_LOOPBACK_WIDTH 1 /* LOOPBACK */ + +/* + * R10 (0x0A) - Left DAC volume + */ +#define WM8961_DACVU 0x0100 /* DACVU */ +#define WM8961_DACVU_MASK 0x0100 /* DACVU */ +#define WM8961_DACVU_SHIFT 8 /* DACVU */ +#define WM8961_DACVU_WIDTH 1 /* DACVU */ +#define WM8961_LDACVOL_MASK 0x00FF /* LDACVOL - [7:0] */ +#define WM8961_LDACVOL_SHIFT 0 /* LDACVOL - [7:0] */ +#define WM8961_LDACVOL_WIDTH 8 /* LDACVOL - [7:0] */ + +/* + * R11 (0x0B) - Right DAC volume + */ +#define WM8961_DACVU 0x0100 /* DACVU */ +#define WM8961_DACVU_MASK 0x0100 /* DACVU */ +#define WM8961_DACVU_SHIFT 8 /* DACVU */ +#define WM8961_DACVU_WIDTH 1 /* DACVU */ +#define WM8961_RDACVOL_MASK 0x00FF /* RDACVOL - [7:0] */ +#define WM8961_RDACVOL_SHIFT 0 /* RDACVOL - [7:0] */ +#define WM8961_RDACVOL_WIDTH 8 /* RDACVOL - [7:0] */ + +/* + * R14 (0x0E) - Audio Interface 2 + */ +#define WM8961_LRCLK_RATE_MASK 0x01FF /* LRCLK_RATE - [8:0] */ +#define WM8961_LRCLK_RATE_SHIFT 0 /* LRCLK_RATE - [8:0] */ +#define WM8961_LRCLK_RATE_WIDTH 9 /* LRCLK_RATE - [8:0] */ + +/* + * R15 (0x0F) - Software Reset + */ +#define WM8961_SW_RST_DEV_ID1_MASK 0xFFFF /* SW_RST_DEV_ID1 - [15:0] */ +#define WM8961_SW_RST_DEV_ID1_SHIFT 0 /* SW_RST_DEV_ID1 - [15:0] */ +#define WM8961_SW_RST_DEV_ID1_WIDTH 16 /* SW_RST_DEV_ID1 - [15:0] */ + +/* + * R17 (0x11) - ALC1 + */ +#define WM8961_ALCSEL_MASK 0x0180 /* ALCSEL - [8:7] */ +#define WM8961_ALCSEL_SHIFT 7 /* ALCSEL - [8:7] */ +#define WM8961_ALCSEL_WIDTH 2 /* ALCSEL - [8:7] */ +#define WM8961_MAXGAIN_MASK 0x0070 /* MAXGAIN - [6:4] */ +#define WM8961_MAXGAIN_SHIFT 4 /* MAXGAIN - [6:4] */ +#define WM8961_MAXGAIN_WIDTH 3 /* MAXGAIN - [6:4] */ +#define WM8961_ALCL_MASK 0x000F /* ALCL - [3:0] */ +#define WM8961_ALCL_SHIFT 0 /* ALCL - [3:0] */ +#define WM8961_ALCL_WIDTH 4 /* ALCL - [3:0] */ + +/* + * R18 (0x12) - ALC2 + */ +#define WM8961_ALCZC 0x0080 /* ALCZC */ +#define WM8961_ALCZC_MASK 0x0080 /* ALCZC */ +#define WM8961_ALCZC_SHIFT 7 /* ALCZC */ +#define WM8961_ALCZC_WIDTH 1 /* ALCZC */ +#define WM8961_MINGAIN_MASK 0x0070 /* MINGAIN - [6:4] */ +#define WM8961_MINGAIN_SHIFT 4 /* MINGAIN - [6:4] */ +#define WM8961_MINGAIN_WIDTH 3 /* MINGAIN - [6:4] */ +#define WM8961_HLD_MASK 0x000F /* HLD - [3:0] */ +#define WM8961_HLD_SHIFT 0 /* HLD - [3:0] */ +#define WM8961_HLD_WIDTH 4 /* HLD - [3:0] */ + +/* + * R19 (0x13) - ALC3 + */ +#define WM8961_ALCMODE 0x0100 /* ALCMODE */ +#define WM8961_ALCMODE_MASK 0x0100 /* ALCMODE */ +#define WM8961_ALCMODE_SHIFT 8 /* ALCMODE */ +#define WM8961_ALCMODE_WIDTH 1 /* ALCMODE */ +#define WM8961_DCY_MASK 0x00F0 /* DCY - [7:4] */ +#define WM8961_DCY_SHIFT 4 /* DCY - [7:4] */ +#define WM8961_DCY_WIDTH 4 /* DCY - [7:4] */ +#define WM8961_ATK_MASK 0x000F /* ATK - [3:0] */ +#define WM8961_ATK_SHIFT 0 /* ATK - [3:0] */ +#define WM8961_ATK_WIDTH 4 /* ATK - [3:0] */ + +/* + * R20 (0x14) - Noise Gate + */ +#define WM8961_NGTH_MASK 0x00F8 /* NGTH - [7:3] */ +#define WM8961_NGTH_SHIFT 3 /* NGTH - [7:3] */ +#define WM8961_NGTH_WIDTH 5 /* NGTH - [7:3] */ +#define WM8961_NGG 0x0002 /* NGG */ +#define WM8961_NGG_MASK 0x0002 /* NGG */ +#define WM8961_NGG_SHIFT 1 /* NGG */ +#define WM8961_NGG_WIDTH 1 /* NGG */ +#define WM8961_NGAT 0x0001 /* NGAT */ +#define WM8961_NGAT_MASK 0x0001 /* NGAT */ +#define WM8961_NGAT_SHIFT 0 /* NGAT */ +#define WM8961_NGAT_WIDTH 1 /* NGAT */ + +/* + * R21 (0x15) - Left ADC volume + */ +#define WM8961_ADCVU 0x0100 /* ADCVU */ +#define WM8961_ADCVU_MASK 0x0100 /* ADCVU */ +#define WM8961_ADCVU_SHIFT 8 /* ADCVU */ +#define WM8961_ADCVU_WIDTH 1 /* ADCVU */ +#define WM8961_LADCVOL_MASK 0x00FF /* LADCVOL - [7:0] */ +#define WM8961_LADCVOL_SHIFT 0 /* LADCVOL - [7:0] */ +#define WM8961_LADCVOL_WIDTH 8 /* LADCVOL - [7:0] */ + +/* + * R22 (0x16) - Right ADC volume + */ +#define WM8961_ADCVU 0x0100 /* ADCVU */ +#define WM8961_ADCVU_MASK 0x0100 /* ADCVU */ +#define WM8961_ADCVU_SHIFT 8 /* ADCVU */ +#define WM8961_ADCVU_WIDTH 1 /* ADCVU */ +#define WM8961_RADCVOL_MASK 0x00FF /* RADCVOL - [7:0] */ +#define WM8961_RADCVOL_SHIFT 0 /* RADCVOL - [7:0] */ +#define WM8961_RADCVOL_WIDTH 8 /* RADCVOL - [7:0] */ + +/* + * R23 (0x17) - Additional control(1) + */ +#define WM8961_TSDEN 0x0100 /* TSDEN */ +#define WM8961_TSDEN_MASK 0x0100 /* TSDEN */ +#define WM8961_TSDEN_SHIFT 8 /* TSDEN */ +#define WM8961_TSDEN_WIDTH 1 /* TSDEN */ +#define WM8961_DMONOMIX 0x0010 /* DMONOMIX */ +#define WM8961_DMONOMIX_MASK 0x0010 /* DMONOMIX */ +#define WM8961_DMONOMIX_SHIFT 4 /* DMONOMIX */ +#define WM8961_DMONOMIX_WIDTH 1 /* DMONOMIX */ +#define WM8961_TOEN 0x0001 /* TOEN */ +#define WM8961_TOEN_MASK 0x0001 /* TOEN */ +#define WM8961_TOEN_SHIFT 0 /* TOEN */ +#define WM8961_TOEN_WIDTH 1 /* TOEN */ + +/* + * R24 (0x18) - Additional control(2) + */ +#define WM8961_TRIS 0x0008 /* TRIS */ +#define WM8961_TRIS_MASK 0x0008 /* TRIS */ +#define WM8961_TRIS_SHIFT 3 /* TRIS */ +#define WM8961_TRIS_WIDTH 1 /* TRIS */ + +/* + * R25 (0x19) - Pwr Mgmt (1) + */ +#define WM8961_VMIDSEL_MASK 0x0180 /* VMIDSEL - [8:7] */ +#define WM8961_VMIDSEL_SHIFT 7 /* VMIDSEL - [8:7] */ +#define WM8961_VMIDSEL_WIDTH 2 /* VMIDSEL - [8:7] */ +#define WM8961_VREF 0x0040 /* VREF */ +#define WM8961_VREF_MASK 0x0040 /* VREF */ +#define WM8961_VREF_SHIFT 6 /* VREF */ +#define WM8961_VREF_WIDTH 1 /* VREF */ +#define WM8961_AINL 0x0020 /* AINL */ +#define WM8961_AINL_MASK 0x0020 /* AINL */ +#define WM8961_AINL_SHIFT 5 /* AINL */ +#define WM8961_AINL_WIDTH 1 /* AINL */ +#define WM8961_AINR 0x0010 /* AINR */ +#define WM8961_AINR_MASK 0x0010 /* AINR */ +#define WM8961_AINR_SHIFT 4 /* AINR */ +#define WM8961_AINR_WIDTH 1 /* AINR */ +#define WM8961_ADCL 0x0008 /* ADCL */ +#define WM8961_ADCL_MASK 0x0008 /* ADCL */ +#define WM8961_ADCL_SHIFT 3 /* ADCL */ +#define WM8961_ADCL_WIDTH 1 /* ADCL */ +#define WM8961_ADCR 0x0004 /* ADCR */ +#define WM8961_ADCR_MASK 0x0004 /* ADCR */ +#define WM8961_ADCR_SHIFT 2 /* ADCR */ +#define WM8961_ADCR_WIDTH 1 /* ADCR */ +#define WM8961_MICB 0x0002 /* MICB */ +#define WM8961_MICB_MASK 0x0002 /* MICB */ +#define WM8961_MICB_SHIFT 1 /* MICB */ +#define WM8961_MICB_WIDTH 1 /* MICB */ + +/* + * R26 (0x1A) - Pwr Mgmt (2) + */ +#define WM8961_DACL 0x0100 /* DACL */ +#define WM8961_DACL_MASK 0x0100 /* DACL */ +#define WM8961_DACL_SHIFT 8 /* DACL */ +#define WM8961_DACL_WIDTH 1 /* DACL */ +#define WM8961_DACR 0x0080 /* DACR */ +#define WM8961_DACR_MASK 0x0080 /* DACR */ +#define WM8961_DACR_SHIFT 7 /* DACR */ +#define WM8961_DACR_WIDTH 1 /* DACR */ +#define WM8961_LOUT1_PGA 0x0040 /* LOUT1_PGA */ +#define WM8961_LOUT1_PGA_MASK 0x0040 /* LOUT1_PGA */ +#define WM8961_LOUT1_PGA_SHIFT 6 /* LOUT1_PGA */ +#define WM8961_LOUT1_PGA_WIDTH 1 /* LOUT1_PGA */ +#define WM8961_ROUT1_PGA 0x0020 /* ROUT1_PGA */ +#define WM8961_ROUT1_PGA_MASK 0x0020 /* ROUT1_PGA */ +#define WM8961_ROUT1_PGA_SHIFT 5 /* ROUT1_PGA */ +#define WM8961_ROUT1_PGA_WIDTH 1 /* ROUT1_PGA */ +#define WM8961_SPKL_PGA 0x0010 /* SPKL_PGA */ +#define WM8961_SPKL_PGA_MASK 0x0010 /* SPKL_PGA */ +#define WM8961_SPKL_PGA_SHIFT 4 /* SPKL_PGA */ +#define WM8961_SPKL_PGA_WIDTH 1 /* SPKL_PGA */ +#define WM8961_SPKR_PGA 0x0008 /* SPKR_PGA */ +#define WM8961_SPKR_PGA_MASK 0x0008 /* SPKR_PGA */ +#define WM8961_SPKR_PGA_SHIFT 3 /* SPKR_PGA */ +#define WM8961_SPKR_PGA_WIDTH 1 /* SPKR_PGA */ + +/* + * R27 (0x1B) - Additional Control (3) + */ +#define WM8961_SAMPLE_RATE_MASK 0x0007 /* SAMPLE_RATE - [2:0] */ +#define WM8961_SAMPLE_RATE_SHIFT 0 /* SAMPLE_RATE - [2:0] */ +#define WM8961_SAMPLE_RATE_WIDTH 3 /* SAMPLE_RATE - [2:0] */ + +/* + * R28 (0x1C) - Anti-pop + */ +#define WM8961_BUFDCOPEN 0x0010 /* BUFDCOPEN */ +#define WM8961_BUFDCOPEN_MASK 0x0010 /* BUFDCOPEN */ +#define WM8961_BUFDCOPEN_SHIFT 4 /* BUFDCOPEN */ +#define WM8961_BUFDCOPEN_WIDTH 1 /* BUFDCOPEN */ +#define WM8961_BUFIOEN 0x0008 /* BUFIOEN */ +#define WM8961_BUFIOEN_MASK 0x0008 /* BUFIOEN */ +#define WM8961_BUFIOEN_SHIFT 3 /* BUFIOEN */ +#define WM8961_BUFIOEN_WIDTH 1 /* BUFIOEN */ +#define WM8961_SOFT_ST 0x0004 /* SOFT_ST */ +#define WM8961_SOFT_ST_MASK 0x0004 /* SOFT_ST */ +#define WM8961_SOFT_ST_SHIFT 2 /* SOFT_ST */ +#define WM8961_SOFT_ST_WIDTH 1 /* SOFT_ST */ + +/* + * R30 (0x1E) - Clocking 3 + */ +#define WM8961_CLK_TO_DIV_MASK 0x0180 /* CLK_TO_DIV - [8:7] */ +#define WM8961_CLK_TO_DIV_SHIFT 7 /* CLK_TO_DIV - [8:7] */ +#define WM8961_CLK_TO_DIV_WIDTH 2 /* CLK_TO_DIV - [8:7] */ +#define WM8961_CLK_256K_DIV_MASK 0x007E /* CLK_256K_DIV - [6:1] */ +#define WM8961_CLK_256K_DIV_SHIFT 1 /* CLK_256K_DIV - [6:1] */ +#define WM8961_CLK_256K_DIV_WIDTH 6 /* CLK_256K_DIV - [6:1] */ +#define WM8961_MANUAL_MODE 0x0001 /* MANUAL_MODE */ +#define WM8961_MANUAL_MODE_MASK 0x0001 /* MANUAL_MODE */ +#define WM8961_MANUAL_MODE_SHIFT 0 /* MANUAL_MODE */ +#define WM8961_MANUAL_MODE_WIDTH 1 /* MANUAL_MODE */ + +/* + * R32 (0x20) - ADCL signal path + */ +#define WM8961_LMICBOOST_MASK 0x0030 /* LMICBOOST - [5:4] */ +#define WM8961_LMICBOOST_SHIFT 4 /* LMICBOOST - [5:4] */ +#define WM8961_LMICBOOST_WIDTH 2 /* LMICBOOST - [5:4] */ + +/* + * R33 (0x21) - ADCR signal path + */ +#define WM8961_RMICBOOST_MASK 0x0030 /* RMICBOOST - [5:4] */ +#define WM8961_RMICBOOST_SHIFT 4 /* RMICBOOST - [5:4] */ +#define WM8961_RMICBOOST_WIDTH 2 /* RMICBOOST - [5:4] */ + +/* + * R40 (0x28) - LOUT2 volume + */ +#define WM8961_SPKVU 0x0100 /* SPKVU */ +#define WM8961_SPKVU_MASK 0x0100 /* SPKVU */ +#define WM8961_SPKVU_SHIFT 8 /* SPKVU */ +#define WM8961_SPKVU_WIDTH 1 /* SPKVU */ +#define WM8961_SPKLZC 0x0080 /* SPKLZC */ +#define WM8961_SPKLZC_MASK 0x0080 /* SPKLZC */ +#define WM8961_SPKLZC_SHIFT 7 /* SPKLZC */ +#define WM8961_SPKLZC_WIDTH 1 /* SPKLZC */ +#define WM8961_SPKLVOL_MASK 0x007F /* SPKLVOL - [6:0] */ +#define WM8961_SPKLVOL_SHIFT 0 /* SPKLVOL - [6:0] */ +#define WM8961_SPKLVOL_WIDTH 7 /* SPKLVOL - [6:0] */ + +/* + * R41 (0x29) - ROUT2 volume + */ +#define WM8961_SPKVU 0x0100 /* SPKVU */ +#define WM8961_SPKVU_MASK 0x0100 /* SPKVU */ +#define WM8961_SPKVU_SHIFT 8 /* SPKVU */ +#define WM8961_SPKVU_WIDTH 1 /* SPKVU */ +#define WM8961_SPKRZC 0x0080 /* SPKRZC */ +#define WM8961_SPKRZC_MASK 0x0080 /* SPKRZC */ +#define WM8961_SPKRZC_SHIFT 7 /* SPKRZC */ +#define WM8961_SPKRZC_WIDTH 1 /* SPKRZC */ +#define WM8961_SPKRVOL_MASK 0x007F /* SPKRVOL - [6:0] */ +#define WM8961_SPKRVOL_SHIFT 0 /* SPKRVOL - [6:0] */ +#define WM8961_SPKRVOL_WIDTH 7 /* SPKRVOL - [6:0] */ + +/* + * R47 (0x2F) - Pwr Mgmt (3) + */ +#define WM8961_TEMP_SHUT 0x0002 /* TEMP_SHUT */ +#define WM8961_TEMP_SHUT_MASK 0x0002 /* TEMP_SHUT */ +#define WM8961_TEMP_SHUT_SHIFT 1 /* TEMP_SHUT */ +#define WM8961_TEMP_SHUT_WIDTH 1 /* TEMP_SHUT */ +#define WM8961_TEMP_WARN 0x0001 /* TEMP_WARN */ +#define WM8961_TEMP_WARN_MASK 0x0001 /* TEMP_WARN */ +#define WM8961_TEMP_WARN_SHIFT 0 /* TEMP_WARN */ +#define WM8961_TEMP_WARN_WIDTH 1 /* TEMP_WARN */ + +/* + * R48 (0x30) - Additional Control (4) + */ +#define WM8961_TSENSEN 0x0002 /* TSENSEN */ +#define WM8961_TSENSEN_MASK 0x0002 /* TSENSEN */ +#define WM8961_TSENSEN_SHIFT 1 /* TSENSEN */ +#define WM8961_TSENSEN_WIDTH 1 /* TSENSEN */ +#define WM8961_MBSEL 0x0001 /* MBSEL */ +#define WM8961_MBSEL_MASK 0x0001 /* MBSEL */ +#define WM8961_MBSEL_SHIFT 0 /* MBSEL */ +#define WM8961_MBSEL_WIDTH 1 /* MBSEL */ + +/* + * R49 (0x31) - Class D Control 1 + */ +#define WM8961_SPKR_ENA 0x0080 /* SPKR_ENA */ +#define WM8961_SPKR_ENA_MASK 0x0080 /* SPKR_ENA */ +#define WM8961_SPKR_ENA_SHIFT 7 /* SPKR_ENA */ +#define WM8961_SPKR_ENA_WIDTH 1 /* SPKR_ENA */ +#define WM8961_SPKL_ENA 0x0040 /* SPKL_ENA */ +#define WM8961_SPKL_ENA_MASK 0x0040 /* SPKL_ENA */ +#define WM8961_SPKL_ENA_SHIFT 6 /* SPKL_ENA */ +#define WM8961_SPKL_ENA_WIDTH 1 /* SPKL_ENA */ + +/* + * R51 (0x33) - Class D Control 2 + */ +#define WM8961_CLASSD_ACGAIN_MASK 0x0007 /* CLASSD_ACGAIN - [2:0] */ +#define WM8961_CLASSD_ACGAIN_SHIFT 0 /* CLASSD_ACGAIN - [2:0] */ +#define WM8961_CLASSD_ACGAIN_WIDTH 3 /* CLASSD_ACGAIN - [2:0] */ + +/* + * R56 (0x38) - Clocking 4 + */ +#define WM8961_CLK_DCS_DIV_MASK 0x01E0 /* CLK_DCS_DIV - [8:5] */ +#define WM8961_CLK_DCS_DIV_SHIFT 5 /* CLK_DCS_DIV - [8:5] */ +#define WM8961_CLK_DCS_DIV_WIDTH 4 /* CLK_DCS_DIV - [8:5] */ +#define WM8961_CLK_SYS_RATE_MASK 0x001E /* CLK_SYS_RATE - [4:1] */ +#define WM8961_CLK_SYS_RATE_SHIFT 1 /* CLK_SYS_RATE - [4:1] */ +#define WM8961_CLK_SYS_RATE_WIDTH 4 /* CLK_SYS_RATE - [4:1] */ + +/* + * R57 (0x39) - DSP Sidetone 0 + */ +#define WM8961_ADCR_DAC_SVOL_MASK 0x00F0 /* ADCR_DAC_SVOL - [7:4] */ +#define WM8961_ADCR_DAC_SVOL_SHIFT 4 /* ADCR_DAC_SVOL - [7:4] */ +#define WM8961_ADCR_DAC_SVOL_WIDTH 4 /* ADCR_DAC_SVOL - [7:4] */ +#define WM8961_ADC_TO_DACR_MASK 0x000C /* ADC_TO_DACR - [3:2] */ +#define WM8961_ADC_TO_DACR_SHIFT 2 /* ADC_TO_DACR - [3:2] */ +#define WM8961_ADC_TO_DACR_WIDTH 2 /* ADC_TO_DACR - [3:2] */ + +/* + * R58 (0x3A) - DSP Sidetone 1 + */ +#define WM8961_ADCL_DAC_SVOL_MASK 0x00F0 /* ADCL_DAC_SVOL - [7:4] */ +#define WM8961_ADCL_DAC_SVOL_SHIFT 4 /* ADCL_DAC_SVOL - [7:4] */ +#define WM8961_ADCL_DAC_SVOL_WIDTH 4 /* ADCL_DAC_SVOL - [7:4] */ +#define WM8961_ADC_TO_DACL_MASK 0x000C /* ADC_TO_DACL - [3:2] */ +#define WM8961_ADC_TO_DACL_SHIFT 2 /* ADC_TO_DACL - [3:2] */ +#define WM8961_ADC_TO_DACL_WIDTH 2 /* ADC_TO_DACL - [3:2] */ + +/* + * R60 (0x3C) - DC Servo 0 + */ +#define WM8961_DCS_ENA_CHAN_INL 0x0080 /* DCS_ENA_CHAN_INL */ +#define WM8961_DCS_ENA_CHAN_INL_MASK 0x0080 /* DCS_ENA_CHAN_INL */ +#define WM8961_DCS_ENA_CHAN_INL_SHIFT 7 /* DCS_ENA_CHAN_INL */ +#define WM8961_DCS_ENA_CHAN_INL_WIDTH 1 /* DCS_ENA_CHAN_INL */ +#define WM8961_DCS_TRIG_STARTUP_INL 0x0040 /* DCS_TRIG_STARTUP_INL */ +#define WM8961_DCS_TRIG_STARTUP_INL_MASK 0x0040 /* DCS_TRIG_STARTUP_INL */ +#define WM8961_DCS_TRIG_STARTUP_INL_SHIFT 6 /* DCS_TRIG_STARTUP_INL */ +#define WM8961_DCS_TRIG_STARTUP_INL_WIDTH 1 /* DCS_TRIG_STARTUP_INL */ +#define WM8961_DCS_TRIG_SERIES_INL 0x0010 /* DCS_TRIG_SERIES_INL */ +#define WM8961_DCS_TRIG_SERIES_INL_MASK 0x0010 /* DCS_TRIG_SERIES_INL */ +#define WM8961_DCS_TRIG_SERIES_INL_SHIFT 4 /* DCS_TRIG_SERIES_INL */ +#define WM8961_DCS_TRIG_SERIES_INL_WIDTH 1 /* DCS_TRIG_SERIES_INL */ +#define WM8961_DCS_ENA_CHAN_INR 0x0008 /* DCS_ENA_CHAN_INR */ +#define WM8961_DCS_ENA_CHAN_INR_MASK 0x0008 /* DCS_ENA_CHAN_INR */ +#define WM8961_DCS_ENA_CHAN_INR_SHIFT 3 /* DCS_ENA_CHAN_INR */ +#define WM8961_DCS_ENA_CHAN_INR_WIDTH 1 /* DCS_ENA_CHAN_INR */ +#define WM8961_DCS_TRIG_STARTUP_INR 0x0004 /* DCS_TRIG_STARTUP_INR */ +#define WM8961_DCS_TRIG_STARTUP_INR_MASK 0x0004 /* DCS_TRIG_STARTUP_INR */ +#define WM8961_DCS_TRIG_STARTUP_INR_SHIFT 2 /* DCS_TRIG_STARTUP_INR */ +#define WM8961_DCS_TRIG_STARTUP_INR_WIDTH 1 /* DCS_TRIG_STARTUP_INR */ +#define WM8961_DCS_TRIG_SERIES_INR 0x0001 /* DCS_TRIG_SERIES_INR */ +#define WM8961_DCS_TRIG_SERIES_INR_MASK 0x0001 /* DCS_TRIG_SERIES_INR */ +#define WM8961_DCS_TRIG_SERIES_INR_SHIFT 0 /* DCS_TRIG_SERIES_INR */ +#define WM8961_DCS_TRIG_SERIES_INR_WIDTH 1 /* DCS_TRIG_SERIES_INR */ + +/* + * R61 (0x3D) - DC Servo 1 + */ +#define WM8961_DCS_ENA_CHAN_HPL 0x0080 /* DCS_ENA_CHAN_HPL */ +#define WM8961_DCS_ENA_CHAN_HPL_MASK 0x0080 /* DCS_ENA_CHAN_HPL */ +#define WM8961_DCS_ENA_CHAN_HPL_SHIFT 7 /* DCS_ENA_CHAN_HPL */ +#define WM8961_DCS_ENA_CHAN_HPL_WIDTH 1 /* DCS_ENA_CHAN_HPL */ +#define WM8961_DCS_TRIG_STARTUP_HPL 0x0040 /* DCS_TRIG_STARTUP_HPL */ +#define WM8961_DCS_TRIG_STARTUP_HPL_MASK 0x0040 /* DCS_TRIG_STARTUP_HPL */ +#define WM8961_DCS_TRIG_STARTUP_HPL_SHIFT 6 /* DCS_TRIG_STARTUP_HPL */ +#define WM8961_DCS_TRIG_STARTUP_HPL_WIDTH 1 /* DCS_TRIG_STARTUP_HPL */ +#define WM8961_DCS_TRIG_SERIES_HPL 0x0010 /* DCS_TRIG_SERIES_HPL */ +#define WM8961_DCS_TRIG_SERIES_HPL_MASK 0x0010 /* DCS_TRIG_SERIES_HPL */ +#define WM8961_DCS_TRIG_SERIES_HPL_SHIFT 4 /* DCS_TRIG_SERIES_HPL */ +#define WM8961_DCS_TRIG_SERIES_HPL_WIDTH 1 /* DCS_TRIG_SERIES_HPL */ +#define WM8961_DCS_ENA_CHAN_HPR 0x0008 /* DCS_ENA_CHAN_HPR */ +#define WM8961_DCS_ENA_CHAN_HPR_MASK 0x0008 /* DCS_ENA_CHAN_HPR */ +#define WM8961_DCS_ENA_CHAN_HPR_SHIFT 3 /* DCS_ENA_CHAN_HPR */ +#define WM8961_DCS_ENA_CHAN_HPR_WIDTH 1 /* DCS_ENA_CHAN_HPR */ +#define WM8961_DCS_TRIG_STARTUP_HPR 0x0004 /* DCS_TRIG_STARTUP_HPR */ +#define WM8961_DCS_TRIG_STARTUP_HPR_MASK 0x0004 /* DCS_TRIG_STARTUP_HPR */ +#define WM8961_DCS_TRIG_STARTUP_HPR_SHIFT 2 /* DCS_TRIG_STARTUP_HPR */ +#define WM8961_DCS_TRIG_STARTUP_HPR_WIDTH 1 /* DCS_TRIG_STARTUP_HPR */ +#define WM8961_DCS_TRIG_SERIES_HPR 0x0001 /* DCS_TRIG_SERIES_HPR */ +#define WM8961_DCS_TRIG_SERIES_HPR_MASK 0x0001 /* DCS_TRIG_SERIES_HPR */ +#define WM8961_DCS_TRIG_SERIES_HPR_SHIFT 0 /* DCS_TRIG_SERIES_HPR */ +#define WM8961_DCS_TRIG_SERIES_HPR_WIDTH 1 /* DCS_TRIG_SERIES_HPR */ + +/* + * R63 (0x3F) - DC Servo 3 + */ +#define WM8961_DCS_FILT_BW_SERIES_MASK 0x0030 /* DCS_FILT_BW_SERIES - [5:4] */ +#define WM8961_DCS_FILT_BW_SERIES_SHIFT 4 /* DCS_FILT_BW_SERIES - [5:4] */ +#define WM8961_DCS_FILT_BW_SERIES_WIDTH 2 /* DCS_FILT_BW_SERIES - [5:4] */ + +/* + * R65 (0x41) - DC Servo 5 + */ +#define WM8961_DCS_SERIES_NO_HP_MASK 0x007F /* DCS_SERIES_NO_HP - [6:0] */ +#define WM8961_DCS_SERIES_NO_HP_SHIFT 0 /* DCS_SERIES_NO_HP - [6:0] */ +#define WM8961_DCS_SERIES_NO_HP_WIDTH 7 /* DCS_SERIES_NO_HP - [6:0] */ + +/* + * R68 (0x44) - Analogue PGA Bias + */ +#define WM8961_HP_PGAS_BIAS_MASK 0x0007 /* HP_PGAS_BIAS - [2:0] */ +#define WM8961_HP_PGAS_BIAS_SHIFT 0 /* HP_PGAS_BIAS - [2:0] */ +#define WM8961_HP_PGAS_BIAS_WIDTH 3 /* HP_PGAS_BIAS - [2:0] */ + +/* + * R69 (0x45) - Analogue HP 0 + */ +#define WM8961_HPL_RMV_SHORT 0x0080 /* HPL_RMV_SHORT */ +#define WM8961_HPL_RMV_SHORT_MASK 0x0080 /* HPL_RMV_SHORT */ +#define WM8961_HPL_RMV_SHORT_SHIFT 7 /* HPL_RMV_SHORT */ +#define WM8961_HPL_RMV_SHORT_WIDTH 1 /* HPL_RMV_SHORT */ +#define WM8961_HPL_ENA_OUTP 0x0040 /* HPL_ENA_OUTP */ +#define WM8961_HPL_ENA_OUTP_MASK 0x0040 /* HPL_ENA_OUTP */ +#define WM8961_HPL_ENA_OUTP_SHIFT 6 /* HPL_ENA_OUTP */ +#define WM8961_HPL_ENA_OUTP_WIDTH 1 /* HPL_ENA_OUTP */ +#define WM8961_HPL_ENA_DLY 0x0020 /* HPL_ENA_DLY */ +#define WM8961_HPL_ENA_DLY_MASK 0x0020 /* HPL_ENA_DLY */ +#define WM8961_HPL_ENA_DLY_SHIFT 5 /* HPL_ENA_DLY */ +#define WM8961_HPL_ENA_DLY_WIDTH 1 /* HPL_ENA_DLY */ +#define WM8961_HPL_ENA 0x0010 /* HPL_ENA */ +#define WM8961_HPL_ENA_MASK 0x0010 /* HPL_ENA */ +#define WM8961_HPL_ENA_SHIFT 4 /* HPL_ENA */ +#define WM8961_HPL_ENA_WIDTH 1 /* HPL_ENA */ +#define WM8961_HPR_RMV_SHORT 0x0008 /* HPR_RMV_SHORT */ +#define WM8961_HPR_RMV_SHORT_MASK 0x0008 /* HPR_RMV_SHORT */ +#define WM8961_HPR_RMV_SHORT_SHIFT 3 /* HPR_RMV_SHORT */ +#define WM8961_HPR_RMV_SHORT_WIDTH 1 /* HPR_RMV_SHORT */ +#define WM8961_HPR_ENA_OUTP 0x0004 /* HPR_ENA_OUTP */ +#define WM8961_HPR_ENA_OUTP_MASK 0x0004 /* HPR_ENA_OUTP */ +#define WM8961_HPR_ENA_OUTP_SHIFT 2 /* HPR_ENA_OUTP */ +#define WM8961_HPR_ENA_OUTP_WIDTH 1 /* HPR_ENA_OUTP */ +#define WM8961_HPR_ENA_DLY 0x0002 /* HPR_ENA_DLY */ +#define WM8961_HPR_ENA_DLY_MASK 0x0002 /* HPR_ENA_DLY */ +#define WM8961_HPR_ENA_DLY_SHIFT 1 /* HPR_ENA_DLY */ +#define WM8961_HPR_ENA_DLY_WIDTH 1 /* HPR_ENA_DLY */ +#define WM8961_HPR_ENA 0x0001 /* HPR_ENA */ +#define WM8961_HPR_ENA_MASK 0x0001 /* HPR_ENA */ +#define WM8961_HPR_ENA_SHIFT 0 /* HPR_ENA */ +#define WM8961_HPR_ENA_WIDTH 1 /* HPR_ENA */ + +/* + * R71 (0x47) - Analogue HP 2 + */ +#define WM8961_HPL_VOL_MASK 0x01C0 /* HPL_VOL - [8:6] */ +#define WM8961_HPL_VOL_SHIFT 6 /* HPL_VOL - [8:6] */ +#define WM8961_HPL_VOL_WIDTH 3 /* HPL_VOL - [8:6] */ +#define WM8961_HPR_VOL_MASK 0x0038 /* HPR_VOL - [5:3] */ +#define WM8961_HPR_VOL_SHIFT 3 /* HPR_VOL - [5:3] */ +#define WM8961_HPR_VOL_WIDTH 3 /* HPR_VOL - [5:3] */ +#define WM8961_HP_BIAS_BOOST_MASK 0x0007 /* HP_BIAS_BOOST - [2:0] */ +#define WM8961_HP_BIAS_BOOST_SHIFT 0 /* HP_BIAS_BOOST - [2:0] */ +#define WM8961_HP_BIAS_BOOST_WIDTH 3 /* HP_BIAS_BOOST - [2:0] */ + +/* + * R72 (0x48) - Charge Pump 1 + */ +#define WM8961_CP_ENA 0x0001 /* CP_ENA */ +#define WM8961_CP_ENA_MASK 0x0001 /* CP_ENA */ +#define WM8961_CP_ENA_SHIFT 0 /* CP_ENA */ +#define WM8961_CP_ENA_WIDTH 1 /* CP_ENA */ + +/* + * R82 (0x52) - Charge Pump B + */ +#define WM8961_CP_DYN_PWR_MASK 0x0003 /* CP_DYN_PWR - [1:0] */ +#define WM8961_CP_DYN_PWR_SHIFT 0 /* CP_DYN_PWR - [1:0] */ +#define WM8961_CP_DYN_PWR_WIDTH 2 /* CP_DYN_PWR - [1:0] */ + +/* + * R87 (0x57) - Write Sequencer 1 + */ +#define WM8961_WSEQ_ENA 0x0020 /* WSEQ_ENA */ +#define WM8961_WSEQ_ENA_MASK 0x0020 /* WSEQ_ENA */ +#define WM8961_WSEQ_ENA_SHIFT 5 /* WSEQ_ENA */ +#define WM8961_WSEQ_ENA_WIDTH 1 /* WSEQ_ENA */ +#define WM8961_WSEQ_WRITE_INDEX_MASK 0x001F /* WSEQ_WRITE_INDEX - [4:0] */ +#define WM8961_WSEQ_WRITE_INDEX_SHIFT 0 /* WSEQ_WRITE_INDEX - [4:0] */ +#define WM8961_WSEQ_WRITE_INDEX_WIDTH 5 /* WSEQ_WRITE_INDEX - [4:0] */ + +/* + * R88 (0x58) - Write Sequencer 2 + */ +#define WM8961_WSEQ_EOS 0x0100 /* WSEQ_EOS */ +#define WM8961_WSEQ_EOS_MASK 0x0100 /* WSEQ_EOS */ +#define WM8961_WSEQ_EOS_SHIFT 8 /* WSEQ_EOS */ +#define WM8961_WSEQ_EOS_WIDTH 1 /* WSEQ_EOS */ +#define WM8961_WSEQ_ADDR_MASK 0x00FF /* WSEQ_ADDR - [7:0] */ +#define WM8961_WSEQ_ADDR_SHIFT 0 /* WSEQ_ADDR - [7:0] */ +#define WM8961_WSEQ_ADDR_WIDTH 8 /* WSEQ_ADDR - [7:0] */ + +/* + * R89 (0x59) - Write Sequencer 3 + */ +#define WM8961_WSEQ_DATA_MASK 0x00FF /* WSEQ_DATA - [7:0] */ +#define WM8961_WSEQ_DATA_SHIFT 0 /* WSEQ_DATA - [7:0] */ +#define WM8961_WSEQ_DATA_WIDTH 8 /* WSEQ_DATA - [7:0] */ + +/* + * R90 (0x5A) - Write Sequencer 4 + */ +#define WM8961_WSEQ_ABORT 0x0100 /* WSEQ_ABORT */ +#define WM8961_WSEQ_ABORT_MASK 0x0100 /* WSEQ_ABORT */ +#define WM8961_WSEQ_ABORT_SHIFT 8 /* WSEQ_ABORT */ +#define WM8961_WSEQ_ABORT_WIDTH 1 /* WSEQ_ABORT */ +#define WM8961_WSEQ_START 0x0080 /* WSEQ_START */ +#define WM8961_WSEQ_START_MASK 0x0080 /* WSEQ_START */ +#define WM8961_WSEQ_START_SHIFT 7 /* WSEQ_START */ +#define WM8961_WSEQ_START_WIDTH 1 /* WSEQ_START */ +#define WM8961_WSEQ_START_INDEX_MASK 0x003F /* WSEQ_START_INDEX - [5:0] */ +#define WM8961_WSEQ_START_INDEX_SHIFT 0 /* WSEQ_START_INDEX - [5:0] */ +#define WM8961_WSEQ_START_INDEX_WIDTH 6 /* WSEQ_START_INDEX - [5:0] */ + +/* + * R91 (0x5B) - Write Sequencer 5 + */ +#define WM8961_WSEQ_DATA_WIDTH_MASK 0x0070 /* WSEQ_DATA_WIDTH - [6:4] */ +#define WM8961_WSEQ_DATA_WIDTH_SHIFT 4 /* WSEQ_DATA_WIDTH - [6:4] */ +#define WM8961_WSEQ_DATA_WIDTH_WIDTH 3 /* WSEQ_DATA_WIDTH - [6:4] */ +#define WM8961_WSEQ_DATA_START_MASK 0x000F /* WSEQ_DATA_START - [3:0] */ +#define WM8961_WSEQ_DATA_START_SHIFT 0 /* WSEQ_DATA_START - [3:0] */ +#define WM8961_WSEQ_DATA_START_WIDTH 4 /* WSEQ_DATA_START - [3:0] */ + +/* + * R92 (0x5C) - Write Sequencer 6 + */ +#define WM8961_WSEQ_DELAY_MASK 0x000F /* WSEQ_DELAY - [3:0] */ +#define WM8961_WSEQ_DELAY_SHIFT 0 /* WSEQ_DELAY - [3:0] */ +#define WM8961_WSEQ_DELAY_WIDTH 4 /* WSEQ_DELAY - [3:0] */ + +/* + * R93 (0x5D) - Write Sequencer 7 + */ +#define WM8961_WSEQ_BUSY 0x0001 /* WSEQ_BUSY */ +#define WM8961_WSEQ_BUSY_MASK 0x0001 /* WSEQ_BUSY */ +#define WM8961_WSEQ_BUSY_SHIFT 0 /* WSEQ_BUSY */ +#define WM8961_WSEQ_BUSY_WIDTH 1 /* WSEQ_BUSY */ + +/* + * R252 (0xFC) - General test 1 + */ +#define WM8961_ARA_ENA 0x0002 /* ARA_ENA */ +#define WM8961_ARA_ENA_MASK 0x0002 /* ARA_ENA */ +#define WM8961_ARA_ENA_SHIFT 1 /* ARA_ENA */ +#define WM8961_ARA_ENA_WIDTH 1 /* ARA_ENA */ +#define WM8961_AUTO_INC 0x0001 /* AUTO_INC */ +#define WM8961_AUTO_INC_MASK 0x0001 /* AUTO_INC */ +#define WM8961_AUTO_INC_SHIFT 0 /* AUTO_INC */ +#define WM8961_AUTO_INC_WIDTH 1 /* AUTO_INC */ + +#endif -- cgit v1.2.3 From df205936d5d1dfec9a52c90af77bb54a2c9c9728 Mon Sep 17 00:00:00 2001 From: "Lopez Cruz, Misael" Date: Thu, 11 Jun 2009 17:35:27 -0500 Subject: ASoC: Zoom2: Add machine driver for Zoom2 board Add support for Zoom2 board. Zoom2 machine driver connects both codec DAIs (audio and voice) to omap3 McBSP ports in the following way: HiFi <-> McBSP2 Voice <-> McBSP3 The zoom2 driver has the following DAPM widgets: * Ext Mic: MAINMIC, SUBMIC (with bias) * Ext Spk: HFL, HFR * Headset Stereophone: HSOL, HSOR * Headset Mic: HSMIC (with bias) * Aux In: AUXL, AUXR Signed-off-by: Misael Lopez Cruz Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 7 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/zoom2.c | 301 ++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 310 insertions(+) create mode 100644 sound/soc/omap/zoom2.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index b771238662b6..a5a90e594535 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -72,4 +72,11 @@ config SND_OMAP_SOC_OMAP3_BEAGLE help Say Y if you want to add support for SoC audio on the Beagleboard. +config SND_OMAP_SOC_ZOOM2 + tristate "SoC Audio support for Zoom2" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_ZOOM2 + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for Soc audio on Zoom2 board. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index a37f49862389..fefc48f02bd3 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -14,6 +14,7 @@ snd-soc-omap3evm-objs := omap3evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o +snd-soc-zoom2-objs := zoom2.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o @@ -23,3 +24,4 @@ obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o +obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c new file mode 100644 index 000000000000..be2e307fdf72 --- /dev/null +++ b/sound/soc/omap/zoom2.c @@ -0,0 +1,301 @@ +/* + * zoom2.c -- SoC audio for Zoom2 + * + * Author: Misael Lopez Cruz + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +#define ZOOM2_HEADSET_MUX_GPIO (OMAP_MAX_GPIO_LINES + 15) + +static int zoom2_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops zoom2_ops = { + .hw_params = zoom2_hw_params, +}; + +static int zoom2_hw_voice_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBS_CFM); + if (ret) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops zoom2_voice_ops = { + .hw_params = zoom2_hw_voice_params, +}; + +/* Zoom2 machine DAPM */ +static const struct snd_soc_dapm_widget zoom2_twl4030_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Ext Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_LINE("Aux In", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* External Mics: MAINMIC, SUBMIC with bias*/ + {"MAINMIC", NULL, "Mic Bias 1"}, + {"SUBMIC", NULL, "Mic Bias 2"}, + {"Mic Bias 1", NULL, "Ext Mic"}, + {"Mic Bias 2", NULL, "Ext Mic"}, + + /* External Speakers: HFL, HFR */ + {"Ext Spk", NULL, "HFL"}, + {"Ext Spk", NULL, "HFR"}, + + /* Headset Stereophone: HSOL, HSOR */ + {"Headset Stereophone", NULL, "HSOL"}, + {"Headset Stereophone", NULL, "HSOR"}, + + /* Headset Mic: HSMIC with bias */ + {"HSMIC", NULL, "Headset Mic Bias"}, + {"Headset Mic Bias", NULL, "Headset Mic"}, + + /* Aux In: AUXL, AUXR */ + {"Aux In", NULL, "AUXL"}, + {"Aux In", NULL, "AUXR"}, +}; + +static int zoom2_twl4030_init(struct snd_soc_codec *codec) +{ + int ret; + + /* Add Zoom2 specific widgets */ + ret = snd_soc_dapm_new_controls(codec, zoom2_twl4030_dapm_widgets, + ARRAY_SIZE(zoom2_twl4030_dapm_widgets)); + if (ret) + return ret; + + /* Set up Zoom2 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* Zoom2 connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Headset Stereophone"); + snd_soc_dapm_enable_pin(codec, "Aux In"); + + /* TWL4030 not connected pins */ + snd_soc_dapm_nc_pin(codec, "CARKITMIC"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC0"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC1"); + + snd_soc_dapm_nc_pin(codec, "OUTL"); + snd_soc_dapm_nc_pin(codec, "OUTR"); + snd_soc_dapm_nc_pin(codec, "EARPIECE"); + snd_soc_dapm_nc_pin(codec, "PREDRIVEL"); + snd_soc_dapm_nc_pin(codec, "PREDRIVER"); + snd_soc_dapm_nc_pin(codec, "CARKITL"); + snd_soc_dapm_nc_pin(codec, "CARKITR"); + + ret = snd_soc_dapm_sync(codec); + + return ret; +} + +static int zoom2_twl4030_voice_init(struct snd_soc_codec *codec) +{ + unsigned short reg; + + /* Enable voice interface */ + reg = codec->read(codec, TWL4030_REG_VOICE_IF); + reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN; + codec->write(codec, TWL4030_REG_VOICE_IF, reg); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link zoom2_dai[] = { + { + .name = "TWL4030 I2S", + .stream_name = "TWL4030 Audio", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], + .init = zoom2_twl4030_init, + .ops = &zoom2_ops, + }, + { + .name = "TWL4030 PCM", + .stream_name = "TWL4030 Voice", + .cpu_dai = &omap_mcbsp_dai[1], + .codec_dai = &twl4030_dai[TWL4030_DAI_VOICE], + .init = zoom2_twl4030_voice_init, + .ops = &zoom2_voice_ops, + }, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_zoom2 = { + .name = "Zoom2", + .platform = &omap_soc_platform, + .dai_link = zoom2_dai, + .num_links = ARRAY_SIZE(zoom2_dai), +}; + +/* twl4030 setup */ +static struct twl4030_setup_data twl4030_setup = { + .ramp_delay_value = 2, /* 81 ms */ + .sysclk = 26000, +}; + +/* Audio subsystem */ +static struct snd_soc_device zoom2_snd_devdata = { + .card = &snd_soc_zoom2, + .codec_dev = &soc_codec_dev_twl4030, + .codec_data = &twl4030_setup, +}; + +static struct platform_device *zoom2_snd_device; + +static int __init zoom2_soc_init(void) +{ + int ret; + + if (!machine_is_omap_zoom2()) { + pr_debug("Not Zoom2!\n"); + return -ENODEV; + } + printk(KERN_INFO "Zoom2 SoC init\n"); + + zoom2_snd_device = platform_device_alloc("soc-audio", -1); + if (!zoom2_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(zoom2_snd_device, &zoom2_snd_devdata); + zoom2_snd_devdata.dev = &zoom2_snd_device->dev; + *(unsigned int *)zoom2_dai[0].cpu_dai->private_data = 1; /* McBSP2 */ + *(unsigned int *)zoom2_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ + + ret = platform_device_add(zoom2_snd_device); + if (ret) + goto err1; + + BUG_ON(gpio_request(ZOOM2_HEADSET_MUX_GPIO, "hs_mux") < 0); + gpio_direction_output(ZOOM2_HEADSET_MUX_GPIO, 0); + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(zoom2_snd_device); + + return ret; +} +module_init(zoom2_soc_init); + +static void __exit zoom2_soc_exit(void) +{ + gpio_free(ZOOM2_HEADSET_MUX_GPIO); + + platform_device_unregister(zoom2_snd_device); +} +module_exit(zoom2_soc_exit); + +MODULE_AUTHOR("Misael Lopez Cruz "); +MODULE_DESCRIPTION("ALSA SoC Zoom2"); +MODULE_LICENSE("GPL"); + -- cgit v1.2.3 From 9e79261f302083cbc6aa95e0f778e3583b1ab36e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 12 Jun 2009 17:27:07 +0100 Subject: ASoC: Automatically control WM8903 sloping stopband filter For best performance the DAC sloping stopband filter should be enabled below 24kHz and not enabled above that so remove the user visible control for this and do it autonomously in the driver. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index d8a9222fbf74..3ebd770463e3 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -715,8 +715,6 @@ SOC_ENUM("DAC Soft Mute Rate", soft_mute), SOC_ENUM("DAC Mute Mode", mute_mode), SOC_SINGLE("DAC Mono Switch", WM8903_DAC_DIGITAL_1, 12, 1, 0), SOC_ENUM("DAC De-emphasis", dac_deemphasis), -SOC_SINGLE("DAC Sloping Stopband Filter Switch", - WM8903_DAC_DIGITAL_1, 11, 1, 0), SOC_ENUM("DAC Companding Mode", dac_companding), SOC_SINGLE("DAC Companding Switch", WM8903_AUDIO_INTERFACE_0, 1, 1, 0), @@ -1377,12 +1375,19 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, u16 aif3 = wm8903_read(codec, WM8903_AUDIO_INTERFACE_3); u16 clock0 = wm8903_read(codec, WM8903_CLOCK_RATES_0); u16 clock1 = wm8903_read(codec, WM8903_CLOCK_RATES_1); + u16 dac_digital1 = wm8903_read(codec, WM8903_DAC_DIGITAL_1); if (substream == wm8903->slave_substream) { dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n"); return 0; } + /* Enable sloping stopband filter for low sample rates */ + if (fs <= 24000) + dac_digital1 |= WM8903_DAC_SB_FILT; + else + dac_digital1 &= ~WM8903_DAC_SB_FILT; + /* Configure sample rate logic for DSP - choose nearest rate */ dsp_config = 0; best_val = abs(sample_rates[dsp_config].rate - fs); @@ -1507,6 +1512,7 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, wm8903_write(codec, WM8903_AUDIO_INTERFACE_1, aif1); wm8903_write(codec, WM8903_AUDIO_INTERFACE_2, aif2); wm8903_write(codec, WM8903_AUDIO_INTERFACE_3, aif3); + wm8903_write(codec, WM8903_DAC_DIGITAL_1, dac_digital1); return 0; } -- cgit v1.2.3 From 21002e20767292d85701154cdf12a591b45f0979 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 12 Jun 2009 17:27:52 +0100 Subject: ASoC: Automatically manage WM8900 sloping stopband filter For best performance the DAC sloping stopband filter should be enabled below 24kHz and not enabled above that so remove the user visible control for this and do it autonomously in the driver. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8900.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 3c78945244b8..de99206a0103 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -116,6 +116,7 @@ #define WM8900_REG_CLOCKING2_DAC_CLKDIV 0x1c #define WM8900_REG_DACCTRL_MUTE 0x004 +#define WM8900_REG_DACCTRL_DAC_SB_FILT 0x100 #define WM8900_REG_DACCTRL_AIF_LRCLKRATE 0x400 #define WM8900_REG_AUDIO3_ADCLRC_DIR 0x0800 @@ -439,7 +440,6 @@ SOC_SINGLE("DAC Soft Mute Switch", WM8900_REG_DACCTRL, 6, 1, 1), SOC_ENUM("DAC Mute Rate", dac_mute_rate), SOC_SINGLE("DAC Mono Switch", WM8900_REG_DACCTRL, 9, 1, 0), SOC_ENUM("DAC Deemphasis", dac_deemphasis), -SOC_SINGLE("DAC Sloping Stopband Filter Switch", WM8900_REG_DACCTRL, 8, 1, 0), SOC_SINGLE("DAC Sigma-Delta Modulator Clock Switch", WM8900_REG_DACCTRL, 12, 1, 0), @@ -743,6 +743,17 @@ static int wm8900_hw_params(struct snd_pcm_substream *substream, wm8900_write(codec, WM8900_REG_AUDIO1, reg); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + reg = wm8900_read(codec, WM8900_REG_DACCTRL); + + if (params_rate(params) <= 24000) + reg |= WM8900_REG_DACCTRL_DAC_SB_FILT; + else + reg &= ~WM8900_REG_DACCTRL_DAC_SB_FILT; + + wm8900_write(codec, WM8900_REG_DACCTRL, reg); + } + return 0; } -- cgit v1.2.3 From 619439998ac32953d737fbe2dc82eb67024547d0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 12 Jun 2009 22:56:59 +0100 Subject: ASoC: Automatically manage WM8350 sloping stopband filter For best performance the DAC sloping stopband filter should be enabled below 24kHz and not enabled above that so remove the user visible control for this and do it autonomously in the driver. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 23 +++++++++++++++++------ 1 file changed, 17 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index e7348d341b76..f6bb59951f96 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -406,7 +406,6 @@ static const char *wm8350_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; static const char *wm8350_pol[] = { "Normal", "Inv R", "Inv L", "Inv L & R" }; static const char *wm8350_dacmutem[] = { "Normal", "Soft" }; static const char *wm8350_dacmutes[] = { "Fast", "Slow" }; -static const char *wm8350_dacfilter[] = { "Normal", "Sloping" }; static const char *wm8350_adcfilter[] = { "None", "High Pass" }; static const char *wm8350_adchp[] = { "44.1kHz", "8kHz", "16kHz", "32kHz" }; static const char *wm8350_lr[] = { "Left", "Right" }; @@ -416,7 +415,6 @@ static const struct soc_enum wm8350_enum[] = { SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 0, 4, wm8350_pol), SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 14, 2, wm8350_dacmutem), SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 13, 2, wm8350_dacmutes), - SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 12, 2, wm8350_dacfilter), SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 15, 2, wm8350_adcfilter), SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 8, 4, wm8350_adchp), SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 0, 4, wm8350_pol), @@ -444,10 +442,9 @@ static const struct snd_kcontrol_new wm8350_snd_controls[] = { 0, 255, 0, dac_pcm_tlv), SOC_ENUM("Playback PCM Mute Function", wm8350_enum[2]), SOC_ENUM("Playback PCM Mute Speed", wm8350_enum[3]), - SOC_ENUM("Playback PCM Filter", wm8350_enum[4]), - SOC_ENUM("Capture PCM Filter", wm8350_enum[5]), - SOC_ENUM("Capture PCM HP Filter", wm8350_enum[6]), - SOC_ENUM("Capture ADC Inversion", wm8350_enum[7]), + SOC_ENUM("Capture PCM Filter", wm8350_enum[4]), + SOC_ENUM("Capture PCM HP Filter", wm8350_enum[5]), + SOC_ENUM("Capture ADC Inversion", wm8350_enum[6]), SOC_WM8350_DOUBLE_R_TLV("Capture PCM Volume", WM8350_ADC_DIGITAL_VOLUME_L, WM8350_ADC_DIGITAL_VOLUME_R, @@ -993,6 +990,7 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai) { struct snd_soc_codec *codec = codec_dai->codec; + struct wm8350 *wm8350 = codec->control_data; u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) & ~WM8350_AIF_WL_MASK; @@ -1012,6 +1010,19 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream, } wm8350_codec_write(codec, WM8350_AI_FORMATING, iface); + + /* The sloping stopband filter is recommended for use with + * lower sample rates to improve performance. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (params_rate(params) < 24000) + wm8350_set_bits(wm8350, WM8350_DAC_MUTE_VOLUME, + WM8350_DAC_SB_FILT); + else + wm8350_clear_bits(wm8350, WM8350_DAC_MUTE_VOLUME, + WM8350_DAC_SB_FILT); + } + return 0; } -- cgit v1.2.3 From 831dc0f10f7b2a4856094ff160c018bf19f77527 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 13 Jun 2009 19:55:02 +0100 Subject: ASoC: Add stub suspend and resume calls for ASoC subdevices Now that ASoC subdevices can be regular devices they can have normal suspend and resume calls from their buses. However, suspending them individually is not desirable since this can lead to problems such as pops and clicks from devices being suspended with their signals being amplified or clocks being stopped suddenly. This will be resolved by having the normal device model suspend and resume calls call into ASoC which will suspend the entire card while any of its components are suspended. At present this is not yet implemented but in order to aid the transition of drivers to the standard device model this patch adds API calls for the notifications. Signed-off-by: Mark Brown --- include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 39 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 44 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5297ba7e2c41..e6704c0a4404 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -192,6 +192,11 @@ void snd_soc_unregister_platform(struct snd_soc_platform *platform); int snd_soc_register_codec(struct snd_soc_codec *codec); void snd_soc_unregister_codec(struct snd_soc_codec *codec); +#ifdef CONFIG_PM +int snd_soc_suspend_device(struct device *dev); +int snd_soc_resume_device(struct device *dev); +#endif + /* pcm <-> DAI connect */ void snd_soc_free_pcms(struct snd_soc_device *socdev); int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e1a920cd8953..44141178ff4a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -788,6 +788,45 @@ static int soc_resume(struct platform_device *pdev) return 0; } +/** + * snd_soc_suspend_device: Notify core of device suspend + * + * @dev: Device being suspended. + * + * In order to ensure that the entire audio subsystem is suspended in a + * coordinated fashion ASoC devices should suspend themselves when + * called by ASoC. When the standard kernel suspend process asks the + * device to suspend it should call this function to initiate a suspend + * of the entire ASoC card. + * + * \note Currently this function is stubbed out. + */ +int snd_soc_suspend_device(struct device *dev) +{ + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_suspend_device); + +/** + * snd_soc_resume_device: Notify core of device resume + * + * @dev: Device being resumed. + * + * In order to ensure that the entire audio subsystem is resumed in a + * coordinated fashion ASoC devices should resume themselves when called + * by ASoC. When the standard kernel resume process asks the device + * to resume it should call this function. Once all the components of + * the card have notified that they are ready to be resumed the card + * will be resumed. + * + * \note Currently this function is stubbed out. + */ +int snd_soc_resume_device(struct device *dev) +{ + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_resume_device); + #else #define soc_suspend NULL #define soc_resume NULL -- cgit v1.2.3 From b3b50b3f31775be5d2e441618bbc1c5cbee4d9f1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 13 Jun 2009 22:30:18 +0100 Subject: ASoC: Add suspend and resume callbacks to Wolfson CODEC drivers Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 17 +++++++++++++++++ sound/soc/codecs/wm8400.c | 17 +++++++++++++++++ sound/soc/codecs/wm8580.c | 17 +++++++++++++++++ sound/soc/codecs/wm8731.c | 39 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8753.c | 35 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8900.c | 17 +++++++++++++++++ sound/soc/codecs/wm8903.c | 17 +++++++++++++++++ sound/soc/codecs/wm8940.c | 17 +++++++++++++++++ sound/soc/codecs/wm8960.c | 17 +++++++++++++++++ sound/soc/codecs/wm8988.c | 34 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm9081.c | 17 +++++++++++++++++ 11 files changed, 244 insertions(+) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index f6bb59951f96..4ded0e3a35e0 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1671,6 +1671,21 @@ static int __devexit wm8350_codec_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM +static int wm8350_codec_suspend(struct platform_device *pdev, pm_message_t m) +{ + return snd_soc_suspend_device(&pdev->dev); +} + +static int wm8350_codec_resume(struct platform_device *pdev) +{ + return snd_soc_resume_device(&pdev->dev); +} +#else +#define wm8350_codec_suspend NULL +#define wm8350_codec_resume NULL +#endif + static struct platform_driver wm8350_codec_driver = { .driver = { .name = "wm8350-codec", @@ -1678,6 +1693,8 @@ static struct platform_driver wm8350_codec_driver = { }, .probe = wm8350_codec_probe, .remove = __devexit_p(wm8350_codec_remove), + .suspend = wm8350_codec_suspend, + .resume = wm8350_codec_resume, }; static __init int wm8350_init(void) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 502eefac1ecd..0bf903f27564 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1553,6 +1553,21 @@ static int __exit wm8400_codec_remove(struct platform_device *dev) return 0; } +#ifdef CONFIG_PM +static int wm8400_pdev_suspend(struct platform_device *pdev, pm_message_t msg) +{ + return snd_soc_suspend_device(&pdev->dev); +} + +static int wm8400_pdev_resume(struct platform_device *pdev) +{ + return snd_soc_resume_device(&pdev->dev); +} +#else +#define wm8400_pdev_suspend NULL +#define wm8400_pdev_resume NULL +#endif + static struct platform_driver wm8400_codec_driver = { .driver = { .name = "wm8400-codec", @@ -1560,6 +1575,8 @@ static struct platform_driver wm8400_codec_driver = { }, .probe = wm8400_codec_probe, .remove = __exit_p(wm8400_codec_remove), + .suspend = wm8400_pdev_suspend, + .resume = wm8400_pdev_resume, }; static int __init wm8400_codec_init(void) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 86c4b24db817..261ef101d4fc 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -995,6 +995,21 @@ static int wm8580_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM +static int wm8580_i2c_suspend(struct i2c_client *client, pm_message_t msg) +{ + return snd_soc_suspend_device(&client->dev); +} + +static int wm8580_i2c_resume(struct i2c_client *client) +{ + return snd_soc_resume_device(&client->dev); +} +#else +#define wm8580_i2c_suspend NULL +#define wm8580_i2c_resume NULL +#endif + static const struct i2c_device_id wm8580_i2c_id[] = { { "wm8580", 0 }, { } @@ -1008,6 +1023,8 @@ static struct i2c_driver wm8580_i2c_driver = { }, .probe = wm8580_i2c_probe, .remove = wm8580_i2c_remove, + .suspend = wm8580_i2c_suspend, + .resume = wm8580_i2c_resume, .id_table = wm8580_i2c_id, }; #endif diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 7a205876ef4f..d7f4788f7ace 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -460,6 +460,7 @@ struct snd_soc_dai wm8731_dai = { }; EXPORT_SYMBOL_GPL(wm8731_dai); +#ifdef CONFIG_PM static int wm8731_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -488,6 +489,10 @@ static int wm8731_resume(struct platform_device *pdev) wm8731_set_bias_level(codec, codec->suspend_bias_level); return 0; } +#else +#define wm8731_suspend NULL +#define wm8731_resume NULL +#endif static int wm8731_probe(struct platform_device *pdev) { @@ -680,6 +685,21 @@ static int __devexit wm8731_spi_remove(struct spi_device *spi) return 0; } +#ifdef CONFIG_PM +static int wm8731_spi_suspend(struct spi_device *spi, pm_message_t msg) +{ + return snd_soc_suspend_device(&spi->dev); +} + +static int wm8731_spi_resume(struct spi_device *spi) +{ + return snd_soc_resume_device(&spi->dev); +} +#else +#define wm8731_spi_suspend NULL +#define wm8731_spi_resume NULL +#endif + static struct spi_driver wm8731_spi_driver = { .driver = { .name = "wm8731", @@ -687,6 +707,8 @@ static struct spi_driver wm8731_spi_driver = { .owner = THIS_MODULE, }, .probe = wm8731_spi_probe, + .suspend = wm8731_spi_suspend, + .resume = wm8731_spi_resume, .remove = __devexit_p(wm8731_spi_remove), }; #endif /* CONFIG_SPI_MASTER */ @@ -720,6 +742,21 @@ static __devexit int wm8731_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM +static int wm8731_i2c_suspend(struct i2c_client *i2c, pm_message_t msg) +{ + return snd_soc_suspend_device(&i2c->dev); +} + +static int wm8731_i2c_resume(struct i2c_client *i2c) +{ + return snd_soc_resume_device(&i2c->dev); +} +#else +#define wm8731_i2c_suspend NULL +#define wm8731_i2c_resume NULL +#endif + static const struct i2c_device_id wm8731_i2c_id[] = { { "wm8731", 0 }, { } @@ -733,6 +770,8 @@ static struct i2c_driver wm8731_i2c_driver = { }, .probe = wm8731_i2c_probe, .remove = __devexit_p(wm8731_i2c_remove), + .suspend = wm8731_i2c_suspend, + .resume = wm8731_i2c_resume, .id_table = wm8731_i2c_id, }; #endif diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index d28eeaceb857..370f7df03628 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1766,6 +1766,21 @@ static int wm8753_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM +static int wm8753_i2c_suspend(struct i2c_client *client, pm_message_t msg) +{ + return snd_soc_suspend_device(&client->dev); +} + +static int wm8753_i2c_resume(struct i2c_client *client) +{ + return snd_soc_resume_device(&client->dev); +} +#else +#define wm8753_i2c_suspend NULL +#define wm8753_i2c_resume NULL +#endif + static const struct i2c_device_id wm8753_i2c_id[] = { { "wm8753", 0 }, { } @@ -1779,6 +1794,8 @@ static struct i2c_driver wm8753_i2c_driver = { }, .probe = wm8753_i2c_probe, .remove = wm8753_i2c_remove, + .suspend = wm8753_i2c_suspend, + .resume = wm8753_i2c_resume, .id_table = wm8753_i2c_id, }; #endif @@ -1834,6 +1851,22 @@ static int __devexit wm8753_spi_remove(struct spi_device *spi) return 0; } +#ifdef CONFIG_PM +static int wm8753_spi_suspend(struct spi_device *spi, pm_message_t msg) +{ + return snd_soc_suspend_device(&spi->dev); +} + +static int wm8753_spi_resume(struct spi_device *spi) +{ + return snd_soc_resume_device(&spi->dev); +} + +#else +#define wm8753_spi_suspend NULL +#define wm8753_spi_resume NULL +#endif + static struct spi_driver wm8753_spi_driver = { .driver = { .name = "wm8753", @@ -1842,6 +1875,8 @@ static struct spi_driver wm8753_spi_driver = { }, .probe = wm8753_spi_probe, .remove = __devexit_p(wm8753_spi_remove), + .suspend = wm8753_spi_suspend, + .resume = wm8753_spi_resume, }; #endif diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index de99206a0103..ac308993ac5a 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1399,6 +1399,21 @@ static __devexit int wm8900_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM +static int wm8900_i2c_suspend(struct i2c_client *client, pm_message_t msg) +{ + return snd_soc_suspend_device(&client->dev); +} + +static int wm8900_i2c_resume(struct i2c_client *client) +{ + return snd_soc_resume_device(&client->dev); +} +#else +#define wm8900_i2c_suspend NULL +#define wm8900_i2c_resume NULL +#endif + static const struct i2c_device_id wm8900_i2c_id[] = { { "wm8900", 0 }, { } @@ -1412,6 +1427,8 @@ static struct i2c_driver wm8900_i2c_driver = { }, .probe = wm8900_i2c_probe, .remove = __devexit_p(wm8900_i2c_remove), + .suspend = wm8900_i2c_suspend, + .resume = wm8900_i2c_resume, .id_table = wm8900_i2c_id, }; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 3ebd770463e3..6239af8fdf71 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1731,6 +1731,21 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM +static int wm8903_i2c_suspend(struct i2c_client *client, pm_message_t msg) +{ + return snd_soc_suspend_device(&client->dev); +} + +static int wm8903_i2c_resume(struct i2c_client *client) +{ + return snd_soc_resume_device(&client->dev); +} +#else +#define wm8903_i2c_suspend NULL +#define wm8903_i2c_resume NULL +#endif + /* i2c codec control layer */ static const struct i2c_device_id wm8903_i2c_id[] = { { "wm8903", 0 }, @@ -1745,6 +1760,8 @@ static struct i2c_driver wm8903_i2c_driver = { }, .probe = wm8903_i2c_probe, .remove = __devexit_p(wm8903_i2c_remove), + .suspend = wm8903_i2c_suspend, + .resume = wm8903_i2c_resume, .id_table = wm8903_i2c_id, }; diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index b8e17d6bc1f7..b69210a77423 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -916,6 +916,21 @@ static int __devexit wm8940_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM +static int wm8940_i2c_suspend(struct i2c_client *client, pm_message_t msg) +{ + return snd_soc_suspend_device(&client->dev); +} + +static int wm8940_i2c_resume(struct i2c_client *client) +{ + return snd_soc_resume_device(&client->dev); +} +#else +#define wm8940_i2c_suspend NULL +#define wm8940_i2c_resume NULL +#endif + static const struct i2c_device_id wm8940_i2c_id[] = { { "wm8940", 0 }, { } @@ -929,6 +944,8 @@ static struct i2c_driver wm8940_i2c_driver = { }, .probe = wm8940_i2c_probe, .remove = __devexit_p(wm8940_i2c_remove), + .suspend = wm8940_i2c_suspend, + .resume = wm8940_i2c_resume, .id_table = wm8940_i2c_id, }; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index e224d8add170..b7894d6dffc0 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -927,6 +927,21 @@ static __devexit int wm8960_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM +static int wm8960_i2c_suspend(struct i2c_client *client, pm_message_t msg) +{ + return snd_soc_suspend_device(&client->dev); +} + +static int wm8960_i2c_resume(struct i2c_client *client) +{ + return snd_soc_resume_device(&client->dev); +} +#else +#define wm8960_i2c_suspend NULL +#define wm8960_i2c_resume NULL +#endif + static const struct i2c_device_id wm8960_i2c_id[] = { { "wm8960", 0 }, { } @@ -940,6 +955,8 @@ static struct i2c_driver wm8960_i2c_driver = { }, .probe = wm8960_i2c_probe, .remove = __devexit_p(wm8960_i2c_remove), + .suspend = wm8960_i2c_suspend, + .resume = wm8960_i2c_resume, .id_table = wm8960_i2c_id, }; diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index c05f71803aa8..03fac6a0f805 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -981,6 +981,21 @@ static int wm8988_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM +static int wm8988_i2c_suspend(struct i2c_client *client, pm_message_t msg) +{ + return snd_soc_suspend_device(&client->dev); +} + +static int wm8988_i2c_resume(struct i2c_client *client) +{ + return snd_soc_resume_device(&client->dev); +} +#else +#define wm8988_i2c_suspend NULL +#define wm8988_i2c_resume NULL +#endif + static const struct i2c_device_id wm8988_i2c_id[] = { { "wm8988", 0 }, { } @@ -994,6 +1009,8 @@ static struct i2c_driver wm8988_i2c_driver = { }, .probe = wm8988_i2c_probe, .remove = wm8988_i2c_remove, + .suspend = wm8988_i2c_suspend, + .resume = wm8988_i2c_resume, .id_table = wm8988_i2c_id, }; #endif @@ -1051,6 +1068,21 @@ static int __devexit wm8988_spi_remove(struct spi_device *spi) return 0; } +#ifdef CONFIG_PM +static int wm8988_spi_suspend(struct spi_device *spi, pm_message_t msg) +{ + return snd_soc_suspend_device(&spi->dev); +} + +static int wm8988_spi_resume(struct spi_device *spi) +{ + return snd_soc_resume_device(&spi->dev); +} +#else +#define wm8988_spi_suspend NULL +#define wm8988_spi_resume NULL +#endif + static struct spi_driver wm8988_spi_driver = { .driver = { .name = "wm8988", @@ -1059,6 +1091,8 @@ static struct spi_driver wm8988_spi_driver = { }, .probe = wm8988_spi_probe, .remove = __devexit_p(wm8988_spi_remove), + .suspend = wm8988_spi_suspend, + .resume = wm8988_spi_resume, }; #endif diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 86fc57e25f97..dbe20597d872 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1492,6 +1492,21 @@ static __devexit int wm9081_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM +static int wm9081_i2c_suspend(struct i2c_client *client, pm_message_t msg) +{ + return snd_soc_suspend_device(&client->dev); +} + +static int wm9081_i2c_resume(struct i2c_client *client) +{ + return snd_soc_resume_device(&client->dev); +} +#else +#define wm9081_i2c_suspend NULL +#define wm9081_i2c_resume NULL +#endif + static const struct i2c_device_id wm9081_i2c_id[] = { { "wm9081", 0 }, { } @@ -1505,6 +1520,8 @@ static struct i2c_driver wm9081_i2c_driver = { }, .probe = wm9081_i2c_probe, .remove = __devexit_p(wm9081_i2c_remove), + .suspend = wm9081_i2c_suspend, + .resume = wm9081_i2c_resume, .id_table = wm9081_i2c_id, }; -- cgit v1.2.3 From 1abd91849990ed61d6468ffa8b7fc1ae61db4b1a Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Mon, 15 Jun 2009 22:18:23 +0200 Subject: ASoC: UDA1380: refactor device registration This patch mostly follows commit 5998102b9095fdb7c67755812038612afea315c5 "ASoC: Refactor WM8731 device registration" to make UDA1380 use standard device instantiation. Similarly, the I2C device registration temporarily moves into the magician machine driver before it will find its final resting place in the board file. At the same time, platform specific configuration is moved to platform data and common power/reset GPIO handling moves into the codec driver. Signed-off-by: Philipp Zabel Signed-off-by: Mark Brown --- include/sound/uda1380.h | 22 ++++ sound/soc/codecs/uda1380.c | 313 +++++++++++++++++++++++++-------------------- sound/soc/codecs/uda1380.h | 8 -- sound/soc/pxa/magician.c | 54 ++++---- 4 files changed, 221 insertions(+), 176 deletions(-) create mode 100644 include/sound/uda1380.h diff --git a/include/sound/uda1380.h b/include/sound/uda1380.h new file mode 100644 index 000000000000..381319c7000c --- /dev/null +++ b/include/sound/uda1380.h @@ -0,0 +1,22 @@ +/* + * UDA1380 ALSA SoC Codec driver + * + * Copyright 2009 Philipp Zabel + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __UDA1380_H +#define __UDA1380_H + +struct uda1380_platform_data { + int gpio_power; + int gpio_reset; + int dac_clk; +#define UDA1380_DAC_CLK_SYSCLK 0 +#define UDA1380_DAC_CLK_WSPLL 1 +}; + +#endif /* __UDA1380_H */ diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 5b21594e0e58..92ec03442154 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -5,9 +5,7 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. * - * Copyright (c) 2007 Philipp Zabel - * Improved support for DAPM and audio routing/mixing capabilities, - * added TLV support. + * Copyright (c) 2007-2009 Philipp Zabel * * Modified by Richard Purdie to fit into SoC * codec model. @@ -19,26 +17,32 @@ #include #include #include -#include #include #include -#include +#include #include #include #include #include #include #include -#include #include #include #include +#include #include "uda1380.h" -static struct work_struct uda1380_work; static struct snd_soc_codec *uda1380_codec; +/* codec private data */ +struct uda1380_priv { + struct snd_soc_codec codec; + u16 reg_cache[UDA1380_CACHEREGNUM]; + unsigned int dac_clk; + struct work_struct work; +}; + /* * uda1380 register cache */ @@ -473,6 +477,7 @@ static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; + struct uda1380_priv *uda1380 = codec->private_data; int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER); switch (cmd) { @@ -480,13 +485,13 @@ static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: uda1380_write_reg_cache(codec, UDA1380_MIXER, mixer & ~R14_SILENCE); - schedule_work(&uda1380_work); + schedule_work(&uda1380->work); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: uda1380_write_reg_cache(codec, UDA1380_MIXER, mixer | R14_SILENCE); - schedule_work(&uda1380_work); + schedule_work(&uda1380->work); break; } return 0; @@ -670,44 +675,33 @@ static int uda1380_resume(struct platform_device *pdev) return 0; } -/* - * initialise the UDA1380 driver - * register mixer and dsp interfaces with the kernel - */ -static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) +static int uda1380_probe(struct platform_device *pdev) { - struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct uda1380_platform_data *pdata; int ret = 0; - codec->name = "UDA1380"; - codec->owner = THIS_MODULE; - codec->read = uda1380_read_reg_cache; - codec->write = uda1380_write; - codec->set_bias_level = uda1380_set_bias_level; - codec->dai = uda1380_dai; - codec->num_dai = ARRAY_SIZE(uda1380_dai); - codec->reg_cache = kmemdup(uda1380_reg, sizeof(uda1380_reg), - GFP_KERNEL); - if (codec->reg_cache == NULL) - return -ENOMEM; - codec->reg_cache_size = ARRAY_SIZE(uda1380_reg); - codec->reg_cache_step = 1; - uda1380_reset(codec); + if (uda1380_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } - uda1380_codec = codec; - INIT_WORK(&uda1380_work, uda1380_flush_work); + socdev->card->codec = uda1380_codec; + codec = uda1380_codec; + pdata = codec->dev->platform_data; /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { - pr_err("uda1380: failed to create pcms\n"); + dev_err(codec->dev, "failed to create pcms: %d\n", ret); goto pcm_err; } /* power on device */ uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* set clock input */ - switch (dac_clk) { + switch (pdata->dac_clk) { case UDA1380_DAC_CLK_SYSCLK: uda1380_write(codec, UDA1380_CLK, 0); break; @@ -716,13 +710,12 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) break; } - /* uda1380 init */ snd_soc_add_controls(codec, uda1380_snd_controls, ARRAY_SIZE(uda1380_snd_controls)); uda1380_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { - pr_err("uda1380: failed to register card\n"); + dev_err(codec->dev, "failed to register card: %d\n", ret); goto card_err; } @@ -732,165 +725,201 @@ card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); pcm_err: - kfree(codec->reg_cache); return ret; } -static struct snd_soc_device *uda1380_socdev; - -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - -static int uda1380_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +/* power down chip */ +static int uda1380_remove(struct platform_device *pdev) { - struct snd_soc_device *socdev = uda1380_socdev; - struct uda1380_setup_data *setup = socdev->codec_data; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - int ret; - - i2c_set_clientdata(i2c, codec); - codec->control_data = i2c; - ret = uda1380_init(socdev, setup->dac_clk); - if (ret < 0) - pr_err("uda1380: failed to initialise UDA1380\n"); + if (codec->control_data) + uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); - return ret; -} + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); -static int uda1380_i2c_remove(struct i2c_client *client) -{ - struct snd_soc_codec *codec = i2c_get_clientdata(client); - kfree(codec->reg_cache); return 0; } -static const struct i2c_device_id uda1380_i2c_id[] = { - { "uda1380", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, uda1380_i2c_id); - -static struct i2c_driver uda1380_i2c_driver = { - .driver = { - .name = "UDA1380 I2C Codec", - .owner = THIS_MODULE, - }, - .probe = uda1380_i2c_probe, - .remove = uda1380_i2c_remove, - .id_table = uda1380_i2c_id, +struct snd_soc_codec_device soc_codec_dev_uda1380 = { + .probe = uda1380_probe, + .remove = uda1380_remove, + .suspend = uda1380_suspend, + .resume = uda1380_resume, }; +EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380); -static int uda1380_add_i2c_device(struct platform_device *pdev, - const struct uda1380_setup_data *setup) +static int uda1380_register(struct uda1380_priv *uda1380) { - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - int ret; + int ret, i; + struct snd_soc_codec *codec = &uda1380->codec; + struct uda1380_platform_data *pdata = codec->dev->platform_data; - ret = i2c_add_driver(&uda1380_i2c_driver); - if (ret != 0) { - dev_err(&pdev->dev, "can't add i2c driver\n"); - return ret; + if (uda1380_codec) { + dev_err(codec->dev, "Another UDA1380 is registered\n"); + return -EINVAL; + } + + if (!pdata || !pdata->gpio_power || !pdata->gpio_reset) + return -EINVAL; + + ret = gpio_request(pdata->gpio_power, "uda1380 power"); + if (ret) + goto err_out; + ret = gpio_request(pdata->gpio_reset, "uda1380 reset"); + if (ret) + goto err_gpio; + + gpio_direction_output(pdata->gpio_power, 1); + + /* we may need to have the clock running here - pH5 */ + gpio_direction_output(pdata->gpio_reset, 1); + udelay(5); + gpio_set_value(pdata->gpio_reset, 0); + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = uda1380; + codec->name = "UDA1380"; + codec->owner = THIS_MODULE; + codec->read = uda1380_read_reg_cache; + codec->write = uda1380_write; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = uda1380_set_bias_level; + codec->dai = uda1380_dai; + codec->num_dai = ARRAY_SIZE(uda1380_dai); + codec->reg_cache_size = ARRAY_SIZE(uda1380_reg); + codec->reg_cache = &uda1380->reg_cache; + codec->reg_cache_step = 1; + + memcpy(codec->reg_cache, uda1380_reg, sizeof(uda1380_reg)); + + ret = uda1380_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err_reset; } - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = setup->i2c_address; - strlcpy(info.type, "uda1380", I2C_NAME_SIZE); + INIT_WORK(&uda1380->work, uda1380_flush_work); + + for (i = 0; i < ARRAY_SIZE(uda1380_dai); i++) + uda1380_dai[i].dev = codec->dev; - adapter = i2c_get_adapter(setup->i2c_bus); - if (!adapter) { - dev_err(&pdev->dev, "can't get i2c adapter %d\n", - setup->i2c_bus); - goto err_driver; + uda1380_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err_reset; } - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", - (unsigned int)info.addr); - goto err_driver; + ret = snd_soc_register_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai)); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", ret); + goto err_dai; } return 0; -err_driver: - i2c_del_driver(&uda1380_i2c_driver); - return -ENODEV; +err_dai: + snd_soc_unregister_codec(codec); +err_reset: + gpio_set_value(pdata->gpio_power, 0); + gpio_free(pdata->gpio_reset); +err_gpio: + gpio_free(pdata->gpio_power); +err_out: + return ret; } -#endif -static int uda1380_probe(struct platform_device *pdev) +static void uda1380_unregister(struct uda1380_priv *uda1380) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct uda1380_setup_data *setup; + struct snd_soc_codec *codec = &uda1380->codec; + struct uda1380_platform_data *pdata = codec->dev->platform_data; + + snd_soc_unregister_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai)); + snd_soc_unregister_codec(&uda1380->codec); + + gpio_set_value(pdata->gpio_power, 0); + gpio_free(pdata->gpio_reset); + gpio_free(pdata->gpio_power); + + kfree(uda1380); + uda1380_codec = NULL; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static __devinit int uda1380_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct uda1380_priv *uda1380; struct snd_soc_codec *codec; int ret; - setup = socdev->codec_data; - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) + uda1380 = kzalloc(sizeof(struct uda1380_priv), GFP_KERNEL); + if (uda1380 == NULL) return -ENOMEM; - socdev->card->codec = codec; - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); + codec = &uda1380->codec; + codec->hw_write = (hw_write_t)i2c_master_send; - uda1380_socdev = socdev; - ret = -ENODEV; + i2c_set_clientdata(i2c, uda1380); + codec->control_data = i2c; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - codec->hw_write = (hw_write_t)i2c_master_send; - ret = uda1380_add_i2c_device(pdev, setup); - } -#endif + codec->dev = &i2c->dev; + ret = uda1380_register(uda1380); if (ret != 0) - kfree(codec); + kfree(uda1380); + return ret; } -/* power down chip */ -static int uda1380_remove(struct platform_device *pdev) +static int __devexit uda1380_i2c_remove(struct i2c_client *i2c) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - - if (codec->control_data) - uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); - - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_unregister_device(codec->control_data); - i2c_del_driver(&uda1380_i2c_driver); -#endif - kfree(codec); - + struct uda1380_priv *uda1380 = i2c_get_clientdata(i2c); + uda1380_unregister(uda1380); return 0; } -struct snd_soc_codec_device soc_codec_dev_uda1380 = { - .probe = uda1380_probe, - .remove = uda1380_remove, - .suspend = uda1380_suspend, - .resume = uda1380_resume, +static const struct i2c_device_id uda1380_i2c_id[] = { + { "uda1380", 0 }, + { } }; -EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380); +MODULE_DEVICE_TABLE(i2c, uda1380_i2c_id); + +static struct i2c_driver uda1380_i2c_driver = { + .driver = { + .name = "UDA1380 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = uda1380_i2c_probe, + .remove = __devexit_p(uda1380_i2c_remove), + .id_table = uda1380_i2c_id, +}; +#endif static int __init uda1380_modinit(void) { - return snd_soc_register_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai)); + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&uda1380_i2c_driver); + if (ret != 0) + pr_err("Failed to register UDA1380 I2C driver: %d\n", ret); +#endif + return 0; } module_init(uda1380_modinit); static void __exit uda1380_exit(void) { - snd_soc_unregister_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai)); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&uda1380_i2c_driver); +#endif } module_exit(uda1380_exit); diff --git a/sound/soc/codecs/uda1380.h b/sound/soc/codecs/uda1380.h index c55c17a52a12..9cefa8a54770 100644 --- a/sound/soc/codecs/uda1380.h +++ b/sound/soc/codecs/uda1380.h @@ -72,14 +72,6 @@ #define R22_SKIP_DCFIL 0x0002 #define R23_AGC_EN 0x0001 -struct uda1380_setup_data { - int i2c_bus; - unsigned short i2c_address; - int dac_clk; -#define UDA1380_DAC_CLK_SYSCLK 0 -#define UDA1380_DAC_CLK_WSPLL 1 -}; - #define UDA1380_DAI_DUPLEX 0 /* playback and capture on single DAI */ #define UDA1380_DAI_PLAYBACK 1 /* playback DAI */ #define UDA1380_DAI_CAPTURE 2 /* capture DAI */ diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index c89a3cdf31e4..9fe4ad20615c 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -20,12 +20,14 @@ #include #include #include +#include #include #include #include #include #include +#include #include #include @@ -447,34 +449,47 @@ static struct snd_soc_card snd_soc_card_magician = { .platform = &pxa2xx_soc_platform, }; -/* magician audio private data */ -static struct uda1380_setup_data magician_uda1380_setup = { - .i2c_address = 0x18, - .dac_clk = UDA1380_DAC_CLK_WSPLL, -}; - /* magician audio subsystem */ static struct snd_soc_device magician_snd_devdata = { .card = &snd_soc_card_magician, .codec_dev = &soc_codec_dev_uda1380, - .codec_data = &magician_uda1380_setup, }; static struct platform_device *magician_snd_device; +/* + * FIXME: move into magician board file once merged into the pxa tree + */ +static struct uda1380_platform_data uda1380_info = { + .gpio_power = EGPIO_MAGICIAN_CODEC_POWER, + .gpio_reset = EGPIO_MAGICIAN_CODEC_RESET, + .dac_clk = UDA1380_DAC_CLK_WSPLL, +}; + +static struct i2c_board_info i2c_board_info[] = { + { + I2C_BOARD_INFO("uda1380", 0x18), + .platform_data = &uda1380_info, + }, +}; + static int __init magician_init(void) { int ret; + struct i2c_adapter *adapter; + struct i2c_client *client; if (!machine_is_magician()) return -ENODEV; - ret = gpio_request(EGPIO_MAGICIAN_CODEC_POWER, "CODEC_POWER"); - if (ret) - goto err_request_power; - ret = gpio_request(EGPIO_MAGICIAN_CODEC_RESET, "CODEC_RESET"); - if (ret) - goto err_request_reset; + adapter = i2c_get_adapter(0); + if (!adapter) + return -ENODEV; + client = i2c_new_device(adapter, i2c_board_info); + i2c_put_adapter(adapter); + if (!client) + return -ENODEV; + ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER"); if (ret) goto err_request_spk; @@ -491,14 +506,8 @@ static int __init magician_init(void) if (ret) goto err_request_in_sel1; - gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 1); gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0); - /* we may need to have the clock running here - pH5 */ - gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 1); - udelay(5); - gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 0); - magician_snd_device = platform_device_alloc("soc-audio", -1); if (!magician_snd_device) { ret = -ENOMEM; @@ -526,10 +535,6 @@ err_request_mic: err_request_ep: gpio_free(EGPIO_MAGICIAN_SPK_POWER); err_request_spk: - gpio_free(EGPIO_MAGICIAN_CODEC_RESET); -err_request_reset: - gpio_free(EGPIO_MAGICIAN_CODEC_POWER); -err_request_power: return ret; } @@ -540,15 +545,12 @@ static void __exit magician_exit(void) gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0); gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0); gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0); - gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 0); gpio_free(EGPIO_MAGICIAN_IN_SEL1); gpio_free(EGPIO_MAGICIAN_IN_SEL0); gpio_free(EGPIO_MAGICIAN_MIC_POWER); gpio_free(EGPIO_MAGICIAN_EP_POWER); gpio_free(EGPIO_MAGICIAN_SPK_POWER); - gpio_free(EGPIO_MAGICIAN_CODEC_RESET); - gpio_free(EGPIO_MAGICIAN_CODEC_POWER); } module_init(magician_init); -- cgit v1.2.3 From 085f30654175a91c28d2b66b9ea6cceab627fed0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Jun 2009 13:57:07 +0200 Subject: ALSA: Add new TLV types for dBwith min/max Add new types for TLV dB scale specified with min/max values instead of min/step since the resolution can't match always with the one a device provides. For example, usb audio devices give 1/256 dB resolution while ALSA TLV is based on 1/100 dB resolution. The new min/max types have less problems because the possible rounding error happens only at min/max. Signed-off-by: Takashi Iwai --- include/sound/tlv.h | 14 ++++++++++++++ sound/core/vmaster.c | 8 ++++++-- 2 files changed, 20 insertions(+), 2 deletions(-) diff --git a/include/sound/tlv.h b/include/sound/tlv.h index d136ea2181ed..9fd5b19ccf5c 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -35,6 +35,8 @@ #define SNDRV_CTL_TLVT_DB_SCALE 1 /* dB scale */ #define SNDRV_CTL_TLVT_DB_LINEAR 2 /* linear volume */ #define SNDRV_CTL_TLVT_DB_RANGE 3 /* dB range container */ +#define SNDRV_CTL_TLVT_DB_MINMAX 4 /* dB scale with min/max */ +#define SNDRV_CTL_TLVT_DB_MINMAX_MUTE 5 /* dB scale with min/max with mute */ #define TLV_DB_SCALE_ITEM(min, step, mute) \ SNDRV_CTL_TLVT_DB_SCALE, 2 * sizeof(unsigned int), \ @@ -42,6 +44,18 @@ #define DECLARE_TLV_DB_SCALE(name, min, step, mute) \ unsigned int name[] = { TLV_DB_SCALE_ITEM(min, step, mute) } +/* dB scale specified with min/max values instead of step */ +#define TLV_DB_MINMAX_ITEM(min_dB, max_dB) \ + SNDRV_CTL_TLVT_DB_MINMAX, 2 * sizeof(unsigned int), \ + (min_dB), (max_dB) +#define TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \ + SNDRV_CTL_TLVT_DB_MINMAX_MUTE, 2 * sizeof(unsigned int), \ + (min_dB), (max_dB) +#define DECLARE_TLV_DB_MINMAX(name, min_dB, max_dB) \ + unsigned int name[] = { TLV_DB_MINMAX_ITEM(min_dB, max_dB) } +#define DECLARE_TLV_DB_MINMAX_MUTE(name, min_dB, max_dB) \ + unsigned int name[] = { TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) } + /* linear volume between min_dB and max_dB (.01dB unit) */ #define TLV_DB_LINEAR_ITEM(min_dB, max_dB) \ SNDRV_CTL_TLVT_DB_LINEAR, 2 * sizeof(unsigned int), \ diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 257624bd1997..3b9b550109cb 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -353,7 +353,8 @@ static void master_free(struct snd_kcontrol *kcontrol) * * The optional argument @tlv can be used to specify the TLV information * for dB scale of the master control. It should be a single element - * with #SNDRV_CTL_TLVT_DB_SCALE type, and should be the max 0dB. + * with #SNDRV_CTL_TLVT_DB_SCALE, #SNDRV_CTL_TLV_DB_MINMAX or + * #SNDRV_CTL_TLVT_DB_MINMAX_MUTE type, and should be the max 0dB. */ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, const unsigned int *tlv) @@ -384,7 +385,10 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, kctl->private_free = master_free; /* additional (constant) TLV read */ - if (tlv && tlv[0] == SNDRV_CTL_TLVT_DB_SCALE) { + if (tlv && + (tlv[0] == SNDRV_CTL_TLVT_DB_SCALE || + tlv[0] == SNDRV_CTL_TLVT_DB_MINMAX || + tlv[0] == SNDRV_CTL_TLVT_DB_MINMAX_MUTE)) { kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; memcpy(master->tlv, tlv, sizeof(master->tlv)); kctl->tlv.p = master->tlv; -- cgit v1.2.3 From b8e1c73f4608b8b9ca1e8f1a09f9fd8684e78071 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Jun 2009 14:04:37 +0200 Subject: ALSA: usb-audio - Use the new TLV_DB_MINMAX type Use the new TLV_DB_MINMAX type instead of TLV_DB_SCALE. Signed-off-by: Takashi Iwai --- sound/usb/usbmixer.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index ecb58e7a6245..f127bfd97c07 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -461,7 +461,7 @@ static int mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *_tlv) { struct usb_mixer_elem_info *cval = kcontrol->private_data; - DECLARE_TLV_DB_SCALE(scale, 0, 0, 0); + DECLARE_TLV_DB_MINMAX(scale, 0, 0); if (size < sizeof(scale)) return -ENOMEM; @@ -469,7 +469,7 @@ static int mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, * while ALSA TLV contains in 1/100 dB unit */ scale[2] = (convert_signed_value(cval, cval->min) * 100) / 256; - scale[3] = (convert_signed_value(cval, cval->res) * 100) / 256; + scale[3] = (convert_signed_value(cval, cval->max) * 100) / 256; if (copy_to_user(_tlv, scale, sizeof(scale))) return -EFAULT; return 0; -- cgit v1.2.3 From eedbdf03a25ab3b2c332ad7fa205aa8ffbe477ba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Jun 2009 14:27:35 +0200 Subject: ALSA: usb-audio - Correct bogus volume dB information Some USB devices give bogus dB information and it screws up PA. It's better to detect a broken value and correct it in the driver before exposing the value to the outside. Signed-off-by: Takashi Iwai --- sound/usb/usbmixer.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index f127bfd97c07..539b427d08c6 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -470,6 +470,15 @@ static int mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, */ scale[2] = (convert_signed_value(cval, cval->min) * 100) / 256; scale[3] = (convert_signed_value(cval, cval->max) * 100) / 256; + if (scale[3] <= scale[2]) { + /* something is wrong; assume it's either from/to 0dB */ + if (scale[2] < 0) + scale[3] = 0; + else if (scale[2] > 0) + scale[2] = 0; + else /* totally crap, return an error */ + return -EINVAL; + } if (copy_to_user(_tlv, scale, sizeof(scale))) return -EFAULT; return 0; -- cgit v1.2.3 From a583cd53478f0c55b92f084bdbe3b66d2b4496df Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 17 Jun 2009 17:30:14 +0100 Subject: ASoC: Regulator support for WM8580 Add basic support for integration with the regulator API to WM8580. Since the core cannot yet disable biases when the CODEC is idle we simply request and enable the regulators for the entire time the driver is active. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8580.c | 38 +++++++++++++++++++++++++++++++++++--- 1 file changed, 35 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 261ef101d4fc..97b9ed95d289 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -24,6 +24,8 @@ #include #include #include +#include + #include #include #include @@ -187,15 +189,22 @@ struct pll_state { unsigned int out; }; +#define WM8580_NUM_SUPPLIES 3 +static const char *wm8580_supply_names[WM8580_NUM_SUPPLIES] = { + "AVDD", + "DVDD", + "PVDD", +}; + /* codec private data */ struct wm8580_priv { struct snd_soc_codec codec; + struct regulator_bulk_data supplies[WM8580_NUM_SUPPLIES]; u16 reg_cache[WM8580_MAX_REGISTER + 1]; struct pll_state a; struct pll_state b; }; - /* * read wm8580 register cache */ @@ -922,11 +931,28 @@ static int wm8580_register(struct wm8580_priv *wm8580) memcpy(codec->reg_cache, wm8580_reg, sizeof(wm8580_reg)); + for (i = 0; i < ARRAY_SIZE(wm8580->supplies); i++) + wm8580->supplies[i].supply = wm8580_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8580->supplies), + wm8580->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8580->supplies), + wm8580->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_regulator_get; + } + /* Get the codec into a known state */ ret = wm8580_write(codec, WM8580_RESET, 0); if (ret != 0) { dev_err(codec->dev, "Failed to reset codec: %d\n", ret); - goto err; + goto err_regulator_enable; } for (i = 0; i < ARRAY_SIZE(wm8580_dai); i++) @@ -939,7 +965,7 @@ static int wm8580_register(struct wm8580_priv *wm8580) ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - goto err; + goto err_regulator_enable; } ret = snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai)); @@ -952,6 +978,10 @@ static int wm8580_register(struct wm8580_priv *wm8580) err_codec: snd_soc_unregister_codec(codec); +err_regulator_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8580->supplies), wm8580->supplies); +err_regulator_get: + regulator_bulk_free(ARRAY_SIZE(wm8580->supplies), wm8580->supplies); err: kfree(wm8580); return ret; @@ -962,6 +992,8 @@ static void wm8580_unregister(struct wm8580_priv *wm8580) wm8580_set_bias_level(&wm8580->codec, SND_SOC_BIAS_OFF); snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai)); snd_soc_unregister_codec(&wm8580->codec); + regulator_bulk_disable(ARRAY_SIZE(wm8580->supplies), wm8580->supplies); + regulator_bulk_free(ARRAY_SIZE(wm8580->supplies), wm8580->supplies); kfree(wm8580); wm8580_codec = NULL; } -- cgit v1.2.3 From b53109db5e016425b767e8e33669a5f41257e6e5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 20 Jun 2009 13:53:20 +0100 Subject: ASoC: Fix shadowed variables in twl4030 No need to define second copies of mode and format, they're declared with exactly the same type at the head of the function and there's no conflict in their use. Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 4dbb853eef5a..a5062ccd28ee 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1609,8 +1609,6 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, /* If the substream has 4 channel, do the necessary setup */ if (params_channels(params) == 4) { - u8 format, mode; - format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); -- cgit v1.2.3 From 423c238d7185a0a8f1b28ddb0be44e0286927909 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 20 Jun 2009 13:54:02 +0100 Subject: ASoC: Staticise put_twl4030_opmode_enum_double() It's an operation for a control and doesn't need to be exported. Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index a5062ccd28ee..4275eea31450 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -924,7 +924,7 @@ static const struct soc_enum twl4030_op_modes_enum = ARRAY_SIZE(twl4030_op_modes_texts), twl4030_op_modes_texts); -int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, +static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -- cgit v1.2.3 From f274143f127dd9420e6def7dc47340b55f29c885 Mon Sep 17 00:00:00 2001 From: Barry Song Date: Sat, 20 Jun 2009 11:29:56 -0400 Subject: ASoC: Blackfin: convert internal names from bf52x to bf5xx These drivers aren't BF52x specific, so don't use bf52x in the names. Signed-off-by: Barry Song Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ad73311.c | 16 ++++++++-------- sound/soc/blackfin/bf5xx-ssm2602.c | 16 ++++++++-------- 2 files changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index edfbdc024e66..9825b71d0e28 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -203,23 +203,23 @@ static struct snd_soc_device bf5xx_ad73311_snd_devdata = { .codec_dev = &soc_codec_dev_ad73311, }; -static struct platform_device *bf52x_ad73311_snd_device; +static struct platform_device *bf5xx_ad73311_snd_device; static int __init bf5xx_ad73311_init(void) { int ret; pr_debug("%s enter\n", __func__); - bf52x_ad73311_snd_device = platform_device_alloc("soc-audio", -1); - if (!bf52x_ad73311_snd_device) + bf5xx_ad73311_snd_device = platform_device_alloc("soc-audio", -1); + if (!bf5xx_ad73311_snd_device) return -ENOMEM; - platform_set_drvdata(bf52x_ad73311_snd_device, &bf5xx_ad73311_snd_devdata); - bf5xx_ad73311_snd_devdata.dev = &bf52x_ad73311_snd_device->dev; - ret = platform_device_add(bf52x_ad73311_snd_device); + platform_set_drvdata(bf5xx_ad73311_snd_device, &bf5xx_ad73311_snd_devdata); + bf5xx_ad73311_snd_devdata.dev = &bf5xx_ad73311_snd_device->dev; + ret = platform_device_add(bf5xx_ad73311_snd_device); if (ret) - platform_device_put(bf52x_ad73311_snd_device); + platform_device_put(bf5xx_ad73311_snd_device); return ret; } @@ -227,7 +227,7 @@ static int __init bf5xx_ad73311_init(void) static void __exit bf5xx_ad73311_exit(void) { pr_debug("%s enter\n", __func__); - platform_device_unregister(bf52x_ad73311_snd_device); + platform_device_unregister(bf5xx_ad73311_snd_device); } module_init(bf5xx_ad73311_init); diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index bc0cdded7116..3a00fa4dbe6d 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -148,24 +148,24 @@ static struct snd_soc_device bf5xx_ssm2602_snd_devdata = { .codec_data = &bf5xx_ssm2602_setup, }; -static struct platform_device *bf52x_ssm2602_snd_device; +static struct platform_device *bf5xx_ssm2602_snd_device; static int __init bf5xx_ssm2602_init(void) { int ret; pr_debug("%s enter\n", __func__); - bf52x_ssm2602_snd_device = platform_device_alloc("soc-audio", -1); - if (!bf52x_ssm2602_snd_device) + bf5xx_ssm2602_snd_device = platform_device_alloc("soc-audio", -1); + if (!bf5xx_ssm2602_snd_device) return -ENOMEM; - platform_set_drvdata(bf52x_ssm2602_snd_device, + platform_set_drvdata(bf5xx_ssm2602_snd_device, &bf5xx_ssm2602_snd_devdata); - bf5xx_ssm2602_snd_devdata.dev = &bf52x_ssm2602_snd_device->dev; - ret = platform_device_add(bf52x_ssm2602_snd_device); + bf5xx_ssm2602_snd_devdata.dev = &bf5xx_ssm2602_snd_device->dev; + ret = platform_device_add(bf5xx_ssm2602_snd_device); if (ret) - platform_device_put(bf52x_ssm2602_snd_device); + platform_device_put(bf5xx_ssm2602_snd_device); return ret; } @@ -173,7 +173,7 @@ static int __init bf5xx_ssm2602_init(void) static void __exit bf5xx_ssm2602_exit(void) { pr_debug("%s enter\n", __func__); - platform_device_unregister(bf52x_ssm2602_snd_device); + platform_device_unregister(bf5xx_ssm2602_snd_device); } module_init(bf5xx_ssm2602_init); -- cgit v1.2.3 From c264301c777840b2df130e042b7f5a0c1041646f Mon Sep 17 00:00:00 2001 From: "Lopez Cruz, Misael" Date: Fri, 19 Jun 2009 03:23:42 -0500 Subject: ASoC: TWL4030: Fix voice interface clock masters Voice interface of twl4030 codec supports: CBM_CFM and CBS_CFS. It doesn't support CBS_CFM. Signed-off-by: Misael Lopez Cruz Acked-By: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 2 +- sound/soc/omap/sdp3430.c | 2 +- sound/soc/omap/zoom2.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 4275eea31450..091125c888ed 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1946,7 +1946,7 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFM: format &= ~(TWL4030_VIF_SLAVE_EN); break; case SND_SOC_DAIFMT_CBS_CFS: diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index b719e5db4f57..c51594d8fd13 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -96,7 +96,7 @@ static int sdp3430_hw_voice_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBS_CFM); + SND_SOC_DAIFMT_CBM_CFM); if (ret) { printk(KERN_ERR "can't set codec DAI configuration\n"); return ret; diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index be2e307fdf72..3de6d2bd3903 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -92,7 +92,7 @@ static int zoom2_hw_voice_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBS_CFM); + SND_SOC_DAIFMT_CBM_CFM); if (ret) { printk(KERN_ERR "can't set codec DAI configuration\n"); return ret; -- cgit v1.2.3 From 328d0a138e3d7761f4db53fabf82279b90ea66dd Mon Sep 17 00:00:00 2001 From: "Lopez Cruz, Misael" Date: Mon, 22 Jun 2009 10:51:52 -0500 Subject: ASoC: TWL4030: Add AVADC Clock Priority AVDAC clk priority allows to determine the path ADC must be connected when the codec is in option2 and both HiFi and Voice paths are enabled. Signed-off-by: Misael Lopez Cruz Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 091125c888ed..f916a9a46712 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1005,6 +1005,16 @@ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); */ static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0); +/* AVADC clock priority */ +static const char *twl4030_avadc_clk_priority_texts[] = { + "Voice high priority", "HiFi high priority" +}; + +static const struct soc_enum twl4030_avadc_clk_priority_enum = + SOC_ENUM_SINGLE(TWL4030_REG_AVADC_CTL, 2, + ARRAY_SIZE(twl4030_avadc_clk_priority_texts), + twl4030_avadc_clk_priority_texts); + static const char *twl4030_rampdelay_texts[] = { "27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms", "437/323/218 ms", "874/645/437 ms", "1748/1291/874 ms", @@ -1106,6 +1116,8 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN, 0, 3, 5, 0, input_gain_tlv), + SOC_ENUM("AVADC Clock Priority", twl4030_avadc_clk_priority_enum), + SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum), SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum), -- cgit v1.2.3 From 30808ca751c3b8d81e948efb8fed7451a8321010 Mon Sep 17 00:00:00 2001 From: "Lopez Cruz, Misael" Date: Mon, 22 Jun 2009 19:34:07 -0500 Subject: ASoC: TWL4030: Correct bypass event for voice sidetone Event for voice sidetone was being interpreted as an analog HiFi bypass event because VSTPGA register offset is less than ARXR2_APGA_CTL offset. Reordering the register checks allows to handle voice digital bypass event properly. Signed-off-by: Misael Lopez Cruz Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 20 +++++++++++++------- 1 file changed, 13 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index f916a9a46712..df42fa2abd91 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -712,7 +712,19 @@ static int bypass_event(struct snd_soc_dapm_widget *w, reg = twl4030_read_reg_cache(w->codec, m->reg); - if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { + /* + * bypass_state[0:3] - analog HiFi bypass + * bypass_state[4] - analog voice bypass + * bypass_state[5] - digital voice bypass + * bypass_state[6:7] - digital HiFi bypass + */ + if (m->reg == TWL4030_REG_VSTPGA) { + /* Voice digital bypass */ + if (reg) + twl4030->bypass_state |= (1 << 5); + else + twl4030->bypass_state &= ~(1 << 5); + } else if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { /* Analog bypass */ if (reg & (1 << m->shift)) twl4030->bypass_state |= @@ -726,12 +738,6 @@ static int bypass_event(struct snd_soc_dapm_widget *w, twl4030->bypass_state |= (1 << 4); else twl4030->bypass_state &= ~(1 << 4); - } else if (m->reg == TWL4030_REG_VSTPGA) { - /* Voice digital bypass */ - if (reg) - twl4030->bypass_state |= (1 << 5); - else - twl4030->bypass_state &= ~(1 << 5); } else { /* Digital bypass */ if (reg & (0x7 << m->shift)) -- cgit v1.2.3 From 517374704da44c1ba77c1600714fe214524af286 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 22 Jun 2009 13:16:51 +0100 Subject: ASoC: Add a shutdown callback Ensure that the audio subsystem is powered down cleanly when the system shuts down by providing a shutdown operation. This ensures that all the components have been returned to an off state cleanly which should avoid audio issues from partially charged capacitors or noise on digital inputs if the system is restarted quickly. Signed-off-by: Mark Brown Tested-by: Ben Dooks --- include/sound/soc-dapm.h | 1 + sound/soc/soc-core.c | 16 ++++++++++++++++ sound/soc/soc-dapm.c | 29 +++++++++++++++++++++++++++++ 3 files changed, 46 insertions(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ec8a45f9a069..35814ced2d22 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -279,6 +279,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, /* dapm events */ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, int event); +void snd_soc_dapm_shutdown(struct snd_soc_device *socdev); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 44141178ff4a..55d45c43ba16 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1020,6 +1020,21 @@ static int soc_remove(struct platform_device *pdev) return 0; } +static void soc_shutdown(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; + + if (!card->instantiated) + return; + + /* Flush out pmdown_time work - we actually do want to run it + * now, we're shutting down so no imminent restart. */ + run_delayed_work(&card->delayed_work); + + snd_soc_dapm_shutdown(socdev); +} + /* ASoC platform driver */ static struct platform_driver soc_driver = { .driver = { @@ -1030,6 +1045,7 @@ static struct platform_driver soc_driver = { .remove = soc_remove, .suspend = soc_suspend, .resume = soc_resume, + .shutdown = soc_shutdown, }; /* create a new pcm */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 653435930ad8..b9129efeedf3 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2032,6 +2032,35 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev) } EXPORT_SYMBOL_GPL(snd_soc_dapm_free); +/* + * snd_soc_dapm_shutdown - callback for system shutdown + */ +void snd_soc_dapm_shutdown(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_dapm_widget *w; + LIST_HEAD(down_list); + int powerdown = 0; + + list_for_each_entry(w, &codec->dapm_widgets, list) { + if (w->power) { + dapm_seq_insert(w, &down_list, dapm_down_seq); + powerdown = 1; + } + } + + /* If there were no widgets to power down we're already in + * standby. + */ + if (powerdown) { + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_PREPARE); + dapm_seq_run(codec, &down_list, 0, dapm_down_seq); + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_STANDBY); + } + + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_OFF); +} + /* Module information */ MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk"); MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC"); -- cgit v1.2.3 From a1e21c9078fb8005e5accb921696ec9e2f38176e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 17 Jun 2009 09:33:52 +0200 Subject: ALSA: hda - Don't call snd_hda_codec_configure in snd_hda_codec_new() The codec setup call via snd_hda_codec_configure() isn't necessarily called in snd_hda_codec_new(). For the later added feature, it's better to change the code flow like: - create all codec instances - configure each codec Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 8 ++------ sound/pci/hda/hda_codec.h | 3 ++- sound/pci/hda/hda_intel.c | 21 +++++++++++++++++---- sound/pci/hda/hda_local.h | 1 - 4 files changed, 21 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 462e2cedaa6a..506f46ef0304 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -885,7 +885,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, * Returns 0 if successful, or a negative error code. */ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, - int do_init, struct hda_codec **codecp) + struct hda_codec **codecp) { struct hda_codec *codec; char component[31]; @@ -978,11 +978,6 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr codec->afg ? codec->afg : codec->mfg, AC_PWRST_D0); - if (do_init) { - err = snd_hda_codec_configure(codec); - if (err < 0) - goto error; - } snd_hda_codec_proc_new(codec); snd_hda_create_hwdep(codec); @@ -1036,6 +1031,7 @@ int snd_hda_codec_configure(struct hda_codec *codec) err = init_unsol_queue(codec->bus); return err; } +EXPORT_SYMBOL_HDA(snd_hda_codec_configure); /** * snd_hda_codec_setup_stream - set up the codec for streaming diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index cad79efaabc9..b7ca7d5bbe8c 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -830,7 +830,8 @@ enum { int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, struct hda_bus **busp); int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, - int do_init, struct hda_codec **codecp); + struct hda_codec **codecp); +int snd_hda_codec_configure(struct hda_codec *codec); /* * low level functions diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4e9ea7080270..da58f2ca9151 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1286,8 +1286,7 @@ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] __devinitdata = { [AZX_DRIVER_TERA] = 1, }; -static int __devinit azx_codec_create(struct azx *chip, const char *model, - int no_init) +static int __devinit azx_codec_create(struct azx *chip, const char *model) { struct hda_bus_template bus_temp; int c, codecs, err; @@ -1346,7 +1345,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, for (c = 0; c < max_slots; c++) { if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { struct hda_codec *codec; - err = snd_hda_codec_new(chip->bus, c, !no_init, &codec); + err = snd_hda_codec_new(chip->bus, c, &codec); if (err < 0) continue; codecs++; @@ -1356,7 +1355,16 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, snd_printk(KERN_ERR SFX "no codecs initialized\n"); return -ENXIO; } + return 0; +} +/* configure each codec instance */ +static int __devinit azx_codec_configure(struct azx *chip) +{ + struct hda_codec *codec; + list_for_each_entry(codec, &chip->bus->codec_list, list) { + snd_hda_codec_configure(codec); + } return 0; } @@ -2466,9 +2474,14 @@ static int __devinit azx_probe(struct pci_dev *pci, card->private_data = chip; /* create codec instances */ - err = azx_codec_create(chip, model[dev], probe_only[dev]); + err = azx_codec_create(chip, model[dev]); if (err < 0) goto out_free; + if (!probe_only[dev]) { + err = azx_codec_configure(chip); + if (err < 0) + goto out_free; + } /* create PCM streams */ err = snd_hda_build_pcms(chip->bus); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 83349013b4df..75aa3785212f 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -99,7 +99,6 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, int snd_hda_add_vmaster(struct hda_codec *codec, char *name, unsigned int *tlv, const char **slaves); int snd_hda_codec_reset(struct hda_codec *codec); -int snd_hda_codec_configure(struct hda_codec *codec); /* amp value bits */ #define HDA_AMP_MUTE 0x80 -- cgit v1.2.3 From 4ea6fbc8eb23c3ae5fd2fb55a340ab85c8649bce Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 17 Jun 2009 09:52:54 +0200 Subject: ALSA: hda - Add patch module option Added the patch module option to apply a "patch" as a firmware to modify pin configurations or give additional hints to the driver before actually initializing and configuring the codec. This can be used as a workaround when the BIOS doesn't give sufficient information or give wrong information that doesn't match with the real hardware setup, until it's fixed statically in the driver via a quirk. Signed-off-by: Takashi Iwai --- sound/pci/hda/Kconfig | 14 +++ sound/pci/hda/hda_codec.h | 7 ++ sound/pci/hda/hda_hwdep.c | 236 ++++++++++++++++++++++++++++++++++++++++++---- sound/pci/hda/hda_intel.c | 21 ++++- 4 files changed, 257 insertions(+), 21 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 04438f1d682d..b8a77f9b0827 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -46,6 +46,20 @@ config SND_HDA_INPUT_JACK Say Y here to enable the jack plugging notification via input layer. +config SND_HDA_PATCH_LOADER + bool "Support initialization patch loading for HD-audio" + depends on EXPERIMENTAL + select FW_LOADER + select SND_HDA_HWDEP + select SND_HDA_RECONFIG + help + Say Y here to allow the HD-audio driver to load a pseudo + firmware file ("patch") for overriding the BIOS setup at + start up. The "patch" file can be specified via patch module + option, such as patch=hda-init. + + This option turns on hwdep and reconfig features automatically. + config SND_HDA_CODEC_REALTEK bool "Build Realtek HD-audio codec support" default y diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index b7ca7d5bbe8c..72c997592eed 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -939,6 +939,13 @@ static inline void snd_hda_power_down(struct hda_codec *codec) {} #define snd_hda_codec_needs_resume(codec) 1 #endif +#ifdef CONFIG_SND_HDA_PATCH_LOADER +/* + * patch firmware + */ +int snd_hda_load_patch(struct hda_bus *bus, const char *patch); +#endif + /* * Codec modularization */ diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 6812fbe80fa4..cc24e6721d74 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include "hda_codec.h" #include "hda_local.h" @@ -312,12 +313,8 @@ static ssize_t init_verbs_show(struct device *dev, return len; } -static ssize_t init_verbs_store(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t count) +static int parse_init_verbs(struct hda_codec *codec, const char *buf) { - struct snd_hwdep *hwdep = dev_get_drvdata(dev); - struct hda_codec *codec = hwdep->private_data; struct hda_verb *v; int nid, verb, param; @@ -331,6 +328,18 @@ static ssize_t init_verbs_store(struct device *dev, v->nid = nid; v->verb = verb; v->param = param; + return 0; +} + +static ssize_t init_verbs_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + int err = parse_init_verbs(codec, buf); + if (err < 0) + return err; return count; } @@ -376,19 +385,15 @@ static void remove_trail_spaces(char *str) #define MAX_HINTS 1024 -static ssize_t hints_store(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t count) +static int parse_hints(struct hda_codec *codec, const char *buf) { - struct snd_hwdep *hwdep = dev_get_drvdata(dev); - struct hda_codec *codec = hwdep->private_data; char *key, *val; struct hda_hint *hint; while (isspace(*buf)) buf++; if (!*buf || *buf == '#' || *buf == '\n') - return count; + return 0; if (*buf == '=') return -EINVAL; key = kstrndup_noeol(buf, 1024); @@ -411,7 +416,7 @@ static ssize_t hints_store(struct device *dev, kfree(hint->key); hint->key = key; hint->val = val; - return count; + return 0; } /* allocate a new hint entry */ if (codec->hints.used >= MAX_HINTS) @@ -424,6 +429,18 @@ static ssize_t hints_store(struct device *dev, } hint->key = key; hint->val = val; + return 0; +} + +static ssize_t hints_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + int err = parse_hints(codec, buf); + if (err < 0) + return err; return count; } @@ -469,20 +486,24 @@ static ssize_t driver_pin_configs_show(struct device *dev, #define MAX_PIN_CONFIGS 32 -static ssize_t user_pin_configs_store(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t count) +static int parse_user_pin_configs(struct hda_codec *codec, const char *buf) { - struct snd_hwdep *hwdep = dev_get_drvdata(dev); - struct hda_codec *codec = hwdep->private_data; int nid, cfg; - int err; if (sscanf(buf, "%i %i", &nid, &cfg) != 2) return -EINVAL; if (!nid) return -EINVAL; - err = snd_hda_add_pincfg(codec, &codec->user_pins, nid, cfg); + return snd_hda_add_pincfg(codec, &codec->user_pins, nid, cfg); +} + +static ssize_t user_pin_configs_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + int err = parse_user_pin_configs(codec, buf); if (err < 0) return err; return count; @@ -553,3 +574,180 @@ int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key) EXPORT_SYMBOL_HDA(snd_hda_get_bool_hint); #endif /* CONFIG_SND_HDA_RECONFIG */ + +#ifdef CONFIG_SND_HDA_PATCH_LOADER + +/* parser mode */ +enum { + LINE_MODE_NONE, + LINE_MODE_CODEC, + LINE_MODE_MODEL, + LINE_MODE_PINCFG, + LINE_MODE_VERB, + LINE_MODE_HINT, + NUM_LINE_MODES, +}; + +static inline int strmatch(const char *a, const char *b) +{ + return strnicmp(a, b, strlen(b)) == 0; +} + +/* parse the contents after the line "[codec]" + * accept only the line with three numbers, and assign the current codec + */ +static void parse_codec_mode(char *buf, struct hda_bus *bus, + struct hda_codec **codecp) +{ + unsigned int vendorid, subid, caddr; + struct hda_codec *codec; + + *codecp = NULL; + if (sscanf(buf, "%i %i %i", &vendorid, &subid, &caddr) == 3) { + list_for_each_entry(codec, &bus->codec_list, list) { + if (codec->addr == caddr) { + *codecp = codec; + break; + } + } + } +} + +/* parse the contents after the other command tags, [pincfg], [verb], + * [hint] and [model] + * just pass to the sysfs helper (only when any codec was specified) + */ +static void parse_pincfg_mode(char *buf, struct hda_bus *bus, + struct hda_codec **codecp) +{ + if (!*codecp) + return; + parse_user_pin_configs(*codecp, buf); +} + +static void parse_verb_mode(char *buf, struct hda_bus *bus, + struct hda_codec **codecp) +{ + if (!*codecp) + return; + parse_init_verbs(*codecp, buf); +} + +static void parse_hint_mode(char *buf, struct hda_bus *bus, + struct hda_codec **codecp) +{ + if (!*codecp) + return; + parse_hints(*codecp, buf); +} + +static void parse_model_mode(char *buf, struct hda_bus *bus, + struct hda_codec **codecp) +{ + if (!*codecp) + return; + kfree((*codecp)->modelname); + (*codecp)->modelname = kstrdup(buf, GFP_KERNEL); +} + +struct hda_patch_item { + const char *tag; + void (*parser)(char *buf, struct hda_bus *bus, struct hda_codec **retc); +}; + +static struct hda_patch_item patch_items[NUM_LINE_MODES] = { + [LINE_MODE_CODEC] = { "[codec]", parse_codec_mode }, + [LINE_MODE_MODEL] = { "[model]", parse_model_mode }, + [LINE_MODE_VERB] = { "[verb]", parse_verb_mode }, + [LINE_MODE_PINCFG] = { "[pincfg]", parse_pincfg_mode }, + [LINE_MODE_HINT] = { "[hint]", parse_hint_mode }, +}; + +/* check the line starting with '[' -- change the parser mode accodingly */ +static int parse_line_mode(char *buf, struct hda_bus *bus) +{ + int i; + for (i = 0; i < ARRAY_SIZE(patch_items); i++) { + if (!patch_items[i].tag) + continue; + if (strmatch(buf, patch_items[i].tag)) + return i; + } + return LINE_MODE_NONE; +} + +/* copy one line from the buffer in fw, and update the fields in fw + * return zero if it reaches to the end of the buffer, or non-zero + * if successfully copied a line + * + * the spaces at the beginning and the end of the line are stripped + */ +static int get_line_from_fw(char *buf, int size, struct firmware *fw) +{ + int len; + const char *p = fw->data; + while (isspace(*p) && fw->size) { + p++; + fw->size--; + } + if (!fw->size) + return 0; + if (size < fw->size) + size = fw->size; + + for (len = 0; len < fw->size; len++) { + if (!*p) + break; + if (*p == '\n') { + p++; + len++; + break; + } + if (len < size) + *buf++ = *p++; + } + *buf = 0; + fw->size -= len; + fw->data = p; + remove_trail_spaces(buf); + return 1; +} + +/* + * load a "patch" firmware file and parse it + */ +int snd_hda_load_patch(struct hda_bus *bus, const char *patch) +{ + int err; + const struct firmware *fw; + struct firmware tmp; + char buf[128]; + struct hda_codec *codec; + int line_mode; + struct device *dev = bus->card->dev; + + if (snd_BUG_ON(!dev)) + return -ENODEV; + err = request_firmware(&fw, patch, dev); + if (err < 0) { + printk(KERN_ERR "hda-codec: Cannot load the patch '%s'\n", + patch); + return err; + } + + tmp = *fw; + line_mode = LINE_MODE_NONE; + codec = NULL; + while (get_line_from_fw(buf, sizeof(buf) - 1, &tmp)) { + if (!*buf || *buf == '#' || *buf == '\n') + continue; + if (*buf == '[') + line_mode = parse_line_mode(buf, bus); + else if (patch_items[line_mode].parser) + patch_items[line_mode].parser(buf, bus, &codec); + } + release_firmware(fw); + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_load_patch); +#endif /* CONFIG_SND_HDA_PATCH_LOADER */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index da58f2ca9151..a2f4a116f872 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -61,6 +61,9 @@ static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int probe_only[SNDRV_CARDS]; static int single_cmd; static int enable_msi; +#ifdef CONFIG_SND_HDA_PATCH_LOADER +static char *patch[SNDRV_CARDS]; +#endif module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -84,6 +87,10 @@ MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs " "(for debugging only)."); module_param(enable_msi, int, 0444); MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)"); +#ifdef CONFIG_SND_HDA_PATCH_LOADER +module_param_array(patch, charp, NULL, 0444); +MODULE_PARM_DESC(patch, "Patch file for Intel HD audio interface."); +#endif #ifdef CONFIG_SND_HDA_POWER_SAVE static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; @@ -2468,6 +2475,9 @@ static int __devinit azx_probe(struct pci_dev *pci, return err; } + /* set this here since it's referred in snd_hda_load_patch() */ + snd_card_set_dev(card, &pci->dev); + err = azx_create(card, pci, dev, pci_id->driver_data, &chip); if (err < 0) goto out_free; @@ -2477,6 +2487,15 @@ static int __devinit azx_probe(struct pci_dev *pci, err = azx_codec_create(chip, model[dev]); if (err < 0) goto out_free; +#ifdef CONFIG_SND_HDA_PATCH_LOADER + if (patch[dev]) { + snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n", + patch[dev]); + err = snd_hda_load_patch(chip->bus, patch[dev]); + if (err < 0) + goto out_free; + } +#endif if (!probe_only[dev]) { err = azx_codec_configure(chip); if (err < 0) @@ -2493,8 +2512,6 @@ static int __devinit azx_probe(struct pci_dev *pci, if (err < 0) goto out_free; - snd_card_set_dev(card, &pci->dev); - err = snd_card_register(card); if (err < 0) goto out_free; -- cgit v1.2.3 From 768248256339da88d65088c8beffe1a3dcd9f1ca Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 17 Jun 2009 17:17:18 +0200 Subject: ALSA: hda - Add description about patch loading Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio.txt | 46 +++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 88b7433d2f11..8bc9867c0a3a 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -403,6 +403,52 @@ re-configure based on that state, run like below: ------------------------------------------------------------------------ +Early Patching +~~~~~~~~~~~~~~ +When CONFIG_SND_HDA_PATCH_LOADER=y is set, you can pass a "patch" as a +firmware file for modifying the HD-audio setup before initializing the +codec. This can work basically like the reconfiguration via sysfs in +the above, but it does it before the first codec configuration. + +The patch file looks like below: + +------------------------------------------------------------------------ + [codec] + 0x12345678 0xabcd1234 2 + + [model] + auto + + [pincfg] + 0x12 0x411111f0 + + [verb] + 0x20 0x500 0x03 + 0x20 0x400 0xff + + [hint] + hp_detect = yes +------------------------------------------------------------------------ + +The file needs to have a line `[codec]`. The next line should contain +three numbers indicating the codec vendor-id (0x12345678 in the +example), the codec subsystem-id (0xabcd1234) and the address (2) of +the codec. The rest patch entries are applied to this specified codec +until another codec entry is given. + +The `[model]` line allows to change the model name of the each codec. +In the example above, it will be changed to model=auto. +Note that this overrides the module option. + +After the `[pincfg]` line, the contents are parsed as the initial +default pin-configurations just like `user_pin_configs` sysfs above. +The values can be shown in user_pin_configs sysfs file, too. + +Similarly, the lines after `[verb]` are parsed as `init_verbs` +sysfs entries, and the lines after `[hint]` are parsed as `hints` +sysfs entries, respectively. + + Power-Saving ~~~~~~~~~~~~ The power-saving is a kind of auto-suspend of the device. When the -- cgit v1.2.3 From 1e7b8c87cb53d9a14f1a9ef35eed739f68851f5c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 17 Jun 2009 17:30:54 +0200 Subject: ALSA: hda - More description about patch module option Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 4 ++++ Documentation/sound/alsa/HD-Audio.txt | 16 +++++++++++++++- 2 files changed, 19 insertions(+), 1 deletion(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 012858d2b119..414700b996ae 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -749,6 +749,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. bdl_pos_adj - Specifies the DMA IRQ timing delay in samples. Passing -1 will make the driver to choose the appropriate value based on the controller chip. + patch - Specifies the early "patch" files to modify the HD-audio + setup before initializing the codecs. This option is + available only when CONFIG_SND_HDA_PATCH_LOADER=y is set. + See HD-Audio.txt for details. [Single (global) options] single_cmd - Use single immediate commands to communicate with diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 8bc9867c0a3a..55aab1168236 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -410,7 +410,7 @@ firmware file for modifying the HD-audio setup before initializing the codec. This can work basically like the reconfiguration via sysfs in the above, but it does it before the first codec configuration. -The patch file looks like below: +A patch file is a plain text file which looks like below: ------------------------------------------------------------------------ [codec] @@ -448,6 +448,20 @@ Similarly, the lines after `[verb]` are parsed as `init_verbs` sysfs entries, and the lines after `[hint]` are parsed as `hints` sysfs entries, respectively. +The hd-audio driver reads the file via request_firmware(). Thus, +a patch file has to be located on the appropriate firmware path, +typically, /lib/firmware. For example, when you pass the option +`patch=hda-init.fw`, the file /lib/firmware/hda-init-fw must be +present. + +The patch module option is specific to each card instance, and you +need to give one file name for each instance, separated by commas. +For example, if you have two cards, one for an on-board analog and one +for an HDMI video board, you may pass patch option like below: +------------------------------------------------------------------------ + options snd-hda-intel patch=on-board-patch,hdmi-patch +------------------------------------------------------------------------ + Power-Saving ~~~~~~~~~~~~ -- cgit v1.2.3 From b40e9538124fc9b9333e3eea0fc514da4a185dae Mon Sep 17 00:00:00 2001 From: Igor Chernyshev Date: Thu, 25 Jun 2009 09:31:07 +0200 Subject: ALSA: ice1724 - Patch for suspend/resume for Audiotrak Prodigy HD2 I've built a small HTPC and had to add suspend/resume support in ice1724 driver. There seem to be 3 existing bugs related to that: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4413 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3748 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2314 Due to hardware (un)availability, I only enabled the fix for Audiotrak Prodigy HD2 card, which is installed in my HTPC. However, most of my code should be reusable in the future on other ice1724-based cards as well (as long as people add card-specific peices of code). The fix is currently based on ALSA 1.0.20 and works on my MythBuntu 9.04 HTPC (using 2.6.28-11 kernel). Signed-off-by: Igor Chernyshev Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1712.h | 9 ++++ sound/pci/ice1712/ice1724.c | 110 +++++++++++++++++++++++++++++++++++---- sound/pci/ice1712/prodigy_hifi.c | 46 ++++++++++++---- 3 files changed, 146 insertions(+), 19 deletions(-) diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index adc909ec125c..9da2dae64c5b 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -379,6 +379,15 @@ struct snd_ice1712 { unsigned char (*set_mclk)(struct snd_ice1712 *ice, unsigned int rate); void (*set_spdif_clock)(struct snd_ice1712 *ice); +#ifdef CONFIG_PM + int (*pm_suspend)(struct snd_ice1712 *); + int (*pm_resume)(struct snd_ice1712 *); + int pm_suspend_enabled:1; + int pm_saved_is_spdif_master:1; + unsigned int pm_saved_spdif_ctrl; + unsigned char pm_saved_spdif_cfg; + unsigned int pm_saved_route; +#endif }; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 36ade77cf371..6a560021e11f 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -560,6 +560,7 @@ static int snd_vt1724_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: spin_lock(&ice->reg_lock); old = inb(ICEMT1724(ice, DMA_CONTROL)); if (cmd == SNDRV_PCM_TRIGGER_START) @@ -570,6 +571,10 @@ static int snd_vt1724_pcm_trigger(struct snd_pcm_substream *substream, int cmd) spin_unlock(&ice->reg_lock); break; + case SNDRV_PCM_TRIGGER_RESUME: + /* apps will have to restart stream */ + break; + default: return -EINVAL; } @@ -2272,7 +2277,7 @@ static void __devinit snd_vt1724_chip_reset(struct snd_ice1712 *ice) msleep(10); } -static int __devinit snd_vt1724_chip_init(struct snd_ice1712 *ice) +static int snd_vt1724_chip_init(struct snd_ice1712 *ice) { outb(ice->eeprom.data[ICE_EEP2_SYSCONF], ICEREG1724(ice, SYS_CFG)); outb(ice->eeprom.data[ICE_EEP2_ACLINK], ICEREG1724(ice, AC97_CFG)); @@ -2287,6 +2292,14 @@ static int __devinit snd_vt1724_chip_init(struct snd_ice1712 *ice) outb(0, ICEREG1724(ice, POWERDOWN)); + /* MPU_RX and TX irq masks are cleared later dynamically */ + outb(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX , ICEREG1724(ice, IRQMASK)); + + /* don't handle FIFO overrun/underruns (just yet), + * since they cause machine lockups + */ + outb(VT1724_MULTI_FIFO_ERR, ICEMT1724(ice, DMA_INT_MASK)); + return 0; } @@ -2431,6 +2444,8 @@ static int __devinit snd_vt1724_create(struct snd_card *card, snd_vt1724_proc_init(ice); synchronize_irq(pci->irq); + card->private_data = ice; + err = pci_request_regions(pci, "ICE1724"); if (err < 0) { kfree(ice); @@ -2459,14 +2474,6 @@ static int __devinit snd_vt1724_create(struct snd_card *card, return -EIO; } - /* MPU_RX and TX irq masks are cleared later dynamically */ - outb(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX , ICEREG1724(ice, IRQMASK)); - - /* don't handle FIFO overrun/underruns (just yet), - * since they cause machine lockups - */ - outb(VT1724_MULTI_FIFO_ERR, ICEMT1724(ice, DMA_INT_MASK)); - err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, ice, &ops); if (err < 0) { snd_vt1724_free(ice); @@ -2650,11 +2657,96 @@ static void __devexit snd_vt1724_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } +#ifdef CONFIG_PM +static int snd_vt1724_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct snd_ice1712 *ice = card->private_data; + + if (!ice->pm_suspend_enabled) + return 0; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + + snd_pcm_suspend_all(ice->pcm); + snd_pcm_suspend_all(ice->pcm_pro); + snd_pcm_suspend_all(ice->pcm_ds); + snd_ac97_suspend(ice->ac97); + + spin_lock_irq(&ice->reg_lock); + ice->pm_saved_is_spdif_master = ice->is_spdif_master(ice); + ice->pm_saved_spdif_ctrl = inw(ICEMT1724(ice, SPDIF_CTRL)); + ice->pm_saved_spdif_cfg = inb(ICEREG1724(ice, SPDIF_CFG)); + ice->pm_saved_route = inl(ICEMT1724(ice, ROUTE_PLAYBACK)); + spin_unlock_irq(&ice->reg_lock); + + if (ice->pm_suspend) + ice->pm_suspend(ice); + + pci_disable_device(pci); + pci_save_state(pci); + pci_set_power_state(pci, pci_choose_state(pci, state)); + return 0; +} + +static int snd_vt1724_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct snd_ice1712 *ice = card->private_data; + + if (!ice->pm_suspend_enabled) + return 0; + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + + if (pci_enable_device(pci) < 0) { + snd_card_disconnect(card); + return -EIO; + } + + pci_set_master(pci); + + snd_vt1724_chip_reset(ice); + + if (snd_vt1724_chip_init(ice) < 0) { + snd_card_disconnect(card); + return -EIO; + } + + if (ice->pm_resume) + ice->pm_resume(ice); + + if (ice->pm_saved_is_spdif_master) { + /* switching to external clock via SPDIF */ + ice->set_spdif_clock(ice); + } else { + /* internal on-card clock */ + snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 1); + } + + update_spdif_bits(ice, ice->pm_saved_spdif_ctrl); + + outb(ice->pm_saved_spdif_cfg, ICEREG1724(ice, SPDIF_CFG)); + outl(ice->pm_saved_route, ICEMT1724(ice, ROUTE_PLAYBACK)); + + if (ice->ac97) + snd_ac97_resume(ice->ac97); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +#endif + static struct pci_driver driver = { .name = "ICE1724", .id_table = snd_vt1724_ids, .probe = snd_vt1724_probe, .remove = __devexit_p(snd_vt1724_remove), +#ifdef CONFIG_PM + .suspend = snd_vt1724_suspend, + .resume = snd_vt1724_resume, +#endif }; static int __init alsa_card_ice1724_init(void) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 043a93879bd5..c75515f5be6f 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -1077,7 +1077,7 @@ static int __devinit prodigy_hifi_init(struct snd_ice1712 *ice) /* * initialize the chip */ -static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice) +static void ak4396_init(struct snd_ice1712 *ice) { static unsigned short ak4396_inits[] = { AK4396_CTRL1, 0x87, /* I2S Normal Mode, 24 bit */ @@ -1087,9 +1087,37 @@ static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice) AK4396_RCH_ATT, 0x00, }; - struct prodigy_hifi_spec *spec; unsigned int i; + /* initialize ak4396 codec */ + /* reset codec */ + ak4396_write(ice, AK4396_CTRL1, 0x86); + msleep(100); + ak4396_write(ice, AK4396_CTRL1, 0x87); + + for (i = 0; i < ARRAY_SIZE(ak4396_inits); i += 2) + ak4396_write(ice, ak4396_inits[i], ak4396_inits[i+1]); +} + +#ifdef CONFIG_PM +static int __devinit prodigy_hd2_resume(struct snd_ice1712 *ice) +{ + /* initialize ak4396 codec and restore previous mixer volumes */ + struct prodigy_hifi_spec *spec = ice->spec; + int i; + mutex_lock(&ice->gpio_mutex); + ak4396_init(ice); + for (i = 0; i < 2; i++) + ak4396_write(ice, AK4396_LCH_ATT + i, spec->vol[i] & 0xff); + mutex_unlock(&ice->gpio_mutex); + return 0; +} +#endif + +static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice) +{ + struct prodigy_hifi_spec *spec; + ice->vt1720 = 0; ice->vt1724 = 1; @@ -1112,14 +1140,12 @@ static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice) return -ENOMEM; ice->spec = spec; - /* initialize ak4396 codec */ - /* reset codec */ - ak4396_write(ice, AK4396_CTRL1, 0x86); - msleep(100); - ak4396_write(ice, AK4396_CTRL1, 0x87); - - for (i = 0; i < ARRAY_SIZE(ak4396_inits); i += 2) - ak4396_write(ice, ak4396_inits[i], ak4396_inits[i+1]); +#ifdef CONFIG_PM + ice->pm_resume = &prodigy_hd2_resume; + ice->pm_suspend_enabled = 1; +#endif + + ak4396_init(ice); return 0; } -- cgit v1.2.3 From e2f551dacbdff8e40365a989ab66104b03316f4d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 25 Jun 2009 13:57:59 +0100 Subject: ASoC: Add core suspend and resume callbacks to WM8961 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8961.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 8e78959ca409..1af2d10702f4 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1267,6 +1267,21 @@ static __devexit int wm8961_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM +static int wm8961_i2c_suspend(struct i2c_client *client) +{ + return snd_soc_suspend_device(&client->dev); +} + +static int wm8961_i2c_resume(struct i2c_client *client) +{ + return snd_soc_resume_device(&client->dev); +} +#else +#define wm8961_i2c_suspend NULL +#define wm8961_i2c_resume NULL +#endif + static const struct i2c_device_id wm8961_i2c_id[] = { { "wm8961", 0 }, { } @@ -1280,6 +1295,8 @@ static struct i2c_driver wm8961_i2c_driver = { }, .probe = wm8961_i2c_probe, .remove = __devexit_p(wm8961_i2c_remove), + .suspend = wm8961_i2c_suspend, + .resume = wm8961_i2c_resume, .id_table = wm8961_i2c_id, }; -- cgit v1.2.3 From 647613e97fa46f6c25cf38b0f2fa81eba5f278d4 Mon Sep 17 00:00:00 2001 From: Atsushi Nemoto Date: Thu, 25 Jun 2009 22:36:58 +0900 Subject: ASoC: txx9aclc: dynamically allocate dmaengine devname Use kasprintf to allocate temporary devname string instead of a fixed size string. This fixes "FIXME" introduced on removal of BUS_ID_SIZE. Signed-off-by: Atsushi Nemoto Signed-off-by: Mark Brown --- sound/soc/txx9/txx9aclc.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 938a58a5a244..efed64b8b026 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -297,15 +297,17 @@ static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, static bool filter(struct dma_chan *chan, void *param) { struct txx9aclc_dmadata *dmadata = param; - char devname[20 + 2]; /* FIXME: old BUS_ID_SIZE + 2 */ + char *devname; + bool found = false; - snprintf(devname, sizeof(devname), "%s.%d", dmadata->dma_res->name, + devname = kasprintf(GFP_KERNEL, "%s.%d", dmadata->dma_res->name, (int)dmadata->dma_res->start); if (strcmp(dev_name(chan->device->dev), devname) == 0) { chan->private = &dmadata->dma_slave; - return true; + found = true; } - return false; + kfree(devname); + return found; } static int txx9aclc_dma_init(struct txx9aclc_soc_device *dev, -- cgit v1.2.3 From e3c7dbb07cde886ba97f183c4cc98a2a99b46eaa Mon Sep 17 00:00:00 2001 From: "Lopez Cruz, Misael" Date: Thu, 25 Jun 2009 12:36:14 -0500 Subject: ASoC: Remove word "Switch" from Handsfree switch name SoC dapm adds the suffix "Switch" to SND_SOC_DAPM_SWITCH controls, removing word "Switch" from HandsfreeL/HandsfreeR widget name for avoiding to duplicate it. Signed-off-by: Misael Lopez Cruz Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index df42fa2abd91..49ceb620678c 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1258,14 +1258,14 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* HandsfreeL/R */ SND_SOC_DAPM_MUX("HandsfreeL Mux", SND_SOC_NOPM, 0, 0, &twl4030_dapm_handsfreel_control), - SND_SOC_DAPM_SWITCH("HandsfreeL Switch", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SWITCH("HandsfreeL", SND_SOC_NOPM, 0, 0, &twl4030_dapm_handsfreelmute_control), SND_SOC_DAPM_PGA_E("HandsfreeL PGA", SND_SOC_NOPM, 0, 0, NULL, 0, handsfreelpga_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MUX("HandsfreeR Mux", SND_SOC_NOPM, 5, 0, &twl4030_dapm_handsfreer_control), - SND_SOC_DAPM_SWITCH("HandsfreeR Switch", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SWITCH("HandsfreeR", SND_SOC_NOPM, 0, 0, &twl4030_dapm_handsfreermute_control), SND_SOC_DAPM_PGA_E("HandsfreeR PGA", SND_SOC_NOPM, 0, 0, NULL, 0, handsfreerpga_event, @@ -1377,15 +1377,15 @@ static const struct snd_soc_dapm_route intercon[] = { {"HandsfreeL Mux", "AudioL1", "Analog L1 Playback Mixer"}, {"HandsfreeL Mux", "AudioL2", "Analog L2 Playback Mixer"}, {"HandsfreeL Mux", "AudioR2", "Analog R2 Playback Mixer"}, - {"HandsfreeL Switch", "Switch", "HandsfreeL Mux"}, - {"HandsfreeL PGA", NULL, "HandsfreeL Switch"}, + {"HandsfreeL", "Switch", "HandsfreeL Mux"}, + {"HandsfreeL PGA", NULL, "HandsfreeL"}, /* HandsfreeR */ {"HandsfreeR Mux", "Voice", "Analog Voice Playback Mixer"}, {"HandsfreeR Mux", "AudioR1", "Analog R1 Playback Mixer"}, {"HandsfreeR Mux", "AudioR2", "Analog R2 Playback Mixer"}, {"HandsfreeR Mux", "AudioL2", "Analog L2 Playback Mixer"}, - {"HandsfreeR Switch", "Switch", "HandsfreeR Mux"}, - {"HandsfreeR PGA", NULL, "HandsfreeR Switch"}, + {"HandsfreeR", "Switch", "HandsfreeR Mux"}, + {"HandsfreeR PGA", NULL, "HandsfreeR"}, /* Vibra */ {"Vibra Mux", "AudioL1", "DAC Left1"}, {"Vibra Mux", "AudioR1", "DAC Right1"}, -- cgit v1.2.3 From c2caa4da46a41899b29f90ec30ef3ba86665b334 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 26 Jun 2009 15:36:56 +0100 Subject: ASoC: Fix widget powerdown on shutdown We need to set the widget power state we want to implement. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b9129efeedf3..5157ec110cfa 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2045,6 +2045,7 @@ void snd_soc_dapm_shutdown(struct snd_soc_device *socdev) list_for_each_entry(w, &codec->dapm_widgets, list) { if (w->power) { dapm_seq_insert(w, &down_list, dapm_down_seq); + w->power = 0; powerdown = 1; } } -- cgit v1.2.3 From 74a0094cd9d030d7a684e6ce1cbd1658eb63bd7d Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Sat, 27 Jun 2009 16:26:35 +0200 Subject: ALSA: cmi8330: revert comments about AD1848 back In ALSA 1.0.20, the comments were changed to say CMI8330 instead of AD1848. The CMI8330 chip includes two codecs - AD1848 and SB16, so the comments were correct and are misleading now. Revert them back. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/isa/cmi8330.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index 3ee0269e5bd0..d510c76c537f 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -35,7 +35,7 @@ * * This card has two mixers and two PCM devices. I've cheesed it such * that recording and playback can be done through the same device. - * The driver "magically" routes the capturing to the CMI8330 codec, + * The driver "magically" routes the capturing to the AD1848 codec, * and playback to the SB16 codec. This allows for full-duplex mode * to some extent. * The utilities in alsa-utils are aware of both devices, so passing @@ -345,7 +345,7 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "CMI8330/C3D PnP configure failure\n"); + snd_printk(KERN_ERR "CMI8330/C3D (AD1848) PnP configure failure\n"); return -EBUSY; } wssport[dev] = pnp_port_start(pdev, 0); @@ -527,11 +527,11 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) wssdma[dev], -1, WSS_HW_DETECT, 0, &acard->wss); if (err < 0) { - snd_printk(KERN_ERR PFX "(CMI8330) device busy??\n"); + snd_printk(KERN_ERR PFX "(AD1848) device busy??\n"); return err; } if (acard->wss->hardware != WSS_HW_CMI8330) { - snd_printk(KERN_ERR PFX "(CMI8330) not found during probe\n"); + snd_printk(KERN_ERR PFX "(AD1848) not found during probe\n"); return -ENODEV; } -- cgit v1.2.3 From b7b51141b4fb6f9059a20c03dd2a5bf77c466c7e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Jun 2009 08:34:06 +0200 Subject: ALSA: hda - Check "beep" hint Check the hint "beep" in snd_hda_attach_beep_device() to avoid the beep device creation if user doesn't want. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 29272f2e95a0..08fe6592ad44 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -24,6 +24,7 @@ #include #include #include "hda_beep.h" +#include "hda_local.h" enum { DIGBEEP_HZ_STEP = 46875, /* 46.875 Hz */ @@ -115,6 +116,9 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) struct hda_beep *beep; int err; + if (!snd_hda_get_bool_hint(codec, "beep")) + return 0; /* disabled explicitly */ + beep = kzalloc(sizeof(*beep), GFP_KERNEL); if (beep == NULL) return -ENOMEM; -- cgit v1.2.3 From 9ea21ebca1a1c4caf3bdaecb0879034107cac1e9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 29 Jun 2009 16:29:06 +0100 Subject: ASoC: Fix WM8961 suspend function type Signed-off-by: Mark Brown --- sound/soc/codecs/wm8961.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 1af2d10702f4..bd1af92a122f 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1268,7 +1268,7 @@ static __devexit int wm8961_i2c_remove(struct i2c_client *client) } #ifdef CONFIG_PM -static int wm8961_i2c_suspend(struct i2c_client *client) +static int wm8961_i2c_suspend(struct i2c_client *client, pm_message_t state) { return snd_soc_suspend_device(&client->dev); } -- cgit v1.2.3 From 4953550a6ca399b644ef057626617465d8be9a7b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 30 Jun 2009 15:28:30 +0200 Subject: ALSA: hda - Merge patch_alc882() and patch_alc883() Merge patch_alc882() and patch_alc883() to the former one since both codecs have fairly similar connections but just a slight difference. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 12 +- sound/pci/hda/patch_realtek.c | 1979 ++++++++++---------------- 2 files changed, 766 insertions(+), 1225 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 939a3dd58148..a1895d7f3cf7 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -114,8 +114,8 @@ ALC662/663/272 samsung-nc10 Samsung NC10 mini notebook auto auto-config reading BIOS (default) -ALC882/885 -========== +ALC882/883/885/888/889 +====================== 3stack-dig 3-jack with SPDIF I/O 6stack-dig 6-jack digital with SPDIF I/O arima Arima W820Di1 @@ -127,12 +127,8 @@ ALC882/885 mbp3 Macbook Pro rev3 imac24 iMac 24'' with jack detection w2jc ASUS W2JC - auto auto-config reading BIOS (default) - -ALC883/888 -========== - 3stack-dig 3-jack with SPDIF I/O - 6stack-dig 6-jack digital with SPDIF I/O + 3stack-2ch-dig 3-jack with SPDIF I/O (ALC883) + alc883-6stack-dig 6-jack digital with SPDIF I/O (ALC883) 3stack-6ch 3-jack 6-channel 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O 6stack-dig-demo 6-jack digital for Intel demo board diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3a8e58c483df..6a899e8fdd0c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -208,12 +208,6 @@ enum { ALC885_MBP3, ALC885_MB5, ALC885_IMAC24, - ALC882_AUTO, - ALC882_MODEL_LAST, -}; - -/* ALC883 models */ -enum { ALC883_3ST_2ch_DIG, ALC883_3ST_6ch_DIG, ALC883_3ST_6ch, @@ -246,8 +240,8 @@ enum { ALC889A_MB31, ALC1200_ASUS_P5Q, ALC883_SONY_VAIO_TT, - ALC883_AUTO, - ALC883_MODEL_LAST, + ALC882_AUTO, + ALC882_MODEL_LAST, }; /* for GPIO Poll */ @@ -320,6 +314,8 @@ struct alc_spec { struct snd_array kctls; struct hda_input_mux private_imux[3]; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; + hda_nid_t private_adc_nids[AUTO_CFG_MAX_OUTS]; + hda_nid_t private_capsrc_nids[AUTO_CFG_MAX_OUTS]; /* hooks */ void (*init_hook)(struct hda_codec *codec); @@ -6295,7 +6291,7 @@ static int patch_alc260(struct hda_codec *codec) /* - * ALC882 support + * ALC882/883/885/888/889 support * * ALC882 is almost identical with ALC880 but has cleaner and more flexible * configuration. Each pin widget can choose any input DACs and a mixer. @@ -6307,22 +6303,35 @@ static int patch_alc260(struct hda_codec *codec) */ #define ALC882_DIGOUT_NID 0x06 #define ALC882_DIGIN_NID 0x0a +#define ALC883_DIGOUT_NID ALC882_DIGOUT_NID +#define ALC883_DIGIN_NID ALC882_DIGIN_NID +#define ALC1200_DIGOUT_NID 0x10 + static struct hda_channel_mode alc882_ch_modes[1] = { { 8, NULL } }; +/* DACs */ static hda_nid_t alc882_dac_nids[4] = { /* front, rear, clfe, rear_surr */ 0x02, 0x03, 0x04, 0x05 }; +#define alc883_dac_nids alc882_dac_nids -/* identical with ALC880 */ +/* ADCs */ #define alc882_adc_nids alc880_adc_nids #define alc882_adc_nids_alt alc880_adc_nids_alt +#define alc883_adc_nids alc882_adc_nids_alt +static hda_nid_t alc883_adc_nids_alt[1] = { 0x08 }; +static hda_nid_t alc883_adc_nids_rev[2] = { 0x09, 0x08 }; +#define alc889_adc_nids alc880_adc_nids static hda_nid_t alc882_capsrc_nids[3] = { 0x24, 0x23, 0x22 }; static hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 }; +#define alc883_capsrc_nids alc882_capsrc_nids_alt +static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 }; +#define alc889_capsrc_nids alc882_capsrc_nids /* input MUX */ /* FIXME: should be a matrix-type input source selection */ @@ -6337,6 +6346,8 @@ static struct hda_input_mux alc882_capture_source = { }, }; +#define alc883_capture_source alc882_capture_source + static struct hda_input_mux mb5_capture_source = { .num_items = 3, .items = { @@ -6346,6 +6357,77 @@ static struct hda_input_mux mb5_capture_source = { }, }; +static struct hda_input_mux alc883_3stack_6ch_intel = { + .num_items = 4, + .items = { + { "Mic", 0x1 }, + { "Front Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + +static struct hda_input_mux alc883_lenovo_101e_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x1 }, + { "Line", 0x2 }, + }, +}; + +static struct hda_input_mux alc883_lenovo_nb0763_capture_source = { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "iMic", 0x1 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + +static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + { "Int Mic", 0x1 }, + }, +}; + +static struct hda_input_mux alc883_lenovo_sky_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Front Mic", 0x1 }, + { "Line", 0x4 }, + }, +}; + +static struct hda_input_mux alc883_asus_eee1601_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + }, +}; + +static struct hda_input_mux alc889A_mb31_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + /* Front Mic (0x01) unused */ + { "Line", 0x2 }, + /* Line 2 (0x03) unused */ + /* CD (0x04) unsused? */ + }, +}; + +/* + * 2ch mode + */ +static struct hda_channel_mode alc883_3ST_2ch_modes[1] = { + { 2, NULL } +}; + /* * 2ch mode */ @@ -6357,6 +6439,18 @@ static struct hda_verb alc882_3ST_ch2_init[] = { { } /* end */ }; +/* + * 4ch mode + */ +static struct hda_verb alc882_3ST_ch4_init[] = { + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + /* * 6ch mode */ @@ -6370,11 +6464,14 @@ static struct hda_verb alc882_3ST_ch6_init[] = { { } /* end */ }; -static struct hda_channel_mode alc882_3ST_6ch_modes[2] = { +static struct hda_channel_mode alc882_3ST_6ch_modes[3] = { { 2, alc882_3ST_ch2_init }, + { 4, alc882_3ST_ch4_init }, { 6, alc882_3ST_ch6_init }, }; +#define alc883_3ST_6ch_modes alc882_3ST_6ch_modes + /* * 6ch mode */ @@ -6462,6 +6559,143 @@ static struct hda_channel_mode alc885_mb5_6ch_modes[2] = { { 6, alc885_mb5_ch6_init }, }; + +/* + * 2ch mode + */ +static struct hda_verb alc883_4ST_ch2_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } /* end */ +}; + +/* + * 4ch mode + */ +static struct hda_verb alc883_4ST_ch4_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* + * 6ch mode + */ +static struct hda_verb alc883_4ST_ch6_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* + * 8ch mode + */ +static struct hda_verb alc883_4ST_ch8_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +static struct hda_channel_mode alc883_4ST_8ch_modes[4] = { + { 2, alc883_4ST_ch2_init }, + { 4, alc883_4ST_ch4_init }, + { 6, alc883_4ST_ch6_init }, + { 8, alc883_4ST_ch8_init }, +}; + + +/* + * 2ch mode + */ +static struct hda_verb alc883_3ST_ch2_intel_init[] = { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } /* end */ +}; + +/* + * 4ch mode + */ +static struct hda_verb alc883_3ST_ch4_intel_init[] = { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* + * 6ch mode + */ +static struct hda_verb alc883_3ST_ch6_intel_init[] = { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x19, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +static struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = { + { 2, alc883_3ST_ch2_intel_init }, + { 4, alc883_3ST_ch4_intel_init }, + { 6, alc883_3ST_ch6_intel_init }, +}; + +/* + * 6ch mode + */ +static struct hda_verb alc883_sixstack_ch6_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, + { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { } /* end */ +}; + +/* + * 8ch mode + */ +static struct hda_verb alc883_sixstack_ch8_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { } /* end */ +}; + +static struct hda_channel_mode alc883_sixstack_modes[2] = { + { 6, alc883_sixstack_ch6_init }, + { 8, alc883_sixstack_ch8_init }, +}; + + /* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17 * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b */ @@ -6597,7 +6831,7 @@ static struct snd_kcontrol_new alc882_chmode_mixer[] = { { } /* end */ }; -static struct hda_verb alc882_init_verbs[] = { +static struct hda_verb alc882_base_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -6649,11 +6883,6 @@ static struct hda_verb alc882_init_verbs[] = { /* FIXME: use matrix-type input source selection */ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* Input mixer2 */ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, @@ -6664,9 +6893,6 @@ static struct hda_verb alc882_init_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* ADC1: mute amp left and right */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, /* ADC2: mute amp left and right */ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -6677,6 +6903,21 @@ static struct hda_verb alc882_init_verbs[] = { { } }; +static struct hda_verb alc882_adc1_init_verbs[] = { + /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* ADC1: mute amp left and right */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, + { } +}; + +/* HACK - expand to two elements */ +#define alc882_init_verbs alc882_base_init_verbs, alc882_adc1_init_verbs + static struct hda_verb alc882_eapd_verbs[] = { /* change to EAPD mode */ {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, @@ -6684,6 +6925,8 @@ static struct hda_verb alc882_eapd_verbs[] = { { } }; +#define alc883_init_verbs alc882_base_init_verbs + /* Mac Pro test */ static struct snd_kcontrol_new alc882_macpro_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -7034,882 +7277,62 @@ static void alc885_imac24_init_hook(struct hda_codec *codec) /* * generic initialization of ADC, input mixers and output mixers */ -static struct hda_verb alc882_auto_init_verbs[] = { +static struct hda_verb alc883_auto_init_verbs[] = { /* * Unmute ADC0-2 and set the default input to mic-in */ - {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback - * mixer widget - * Note: PASD motherboards uses the Line In 2 as the input for - * front panel mic (mic 2) - */ - /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - - /* - * Set up output mixers (0x0c - 0x0f) - */ - /* set vol=0 to output mixers */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* set up input amps for analog loopback */ - /* Amp Indices: DAC = 0, mixer = 1 */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, 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)}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, - /* Input mixer2 */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, - /* Input mixer3 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, - - { } -}; - -#ifdef CONFIG_SND_HDA_POWER_SAVE -#define alc882_loopbacks alc880_loopbacks -#endif - -/* pcm configuration: identical with ALC880 */ -#define alc882_pcm_analog_playback alc880_pcm_analog_playback -#define alc882_pcm_analog_capture alc880_pcm_analog_capture -#define alc882_pcm_digital_playback alc880_pcm_digital_playback -#define alc882_pcm_digital_capture alc880_pcm_digital_capture - -/* - * configuration and preset - */ -static const char *alc882_models[ALC882_MODEL_LAST] = { - [ALC882_3ST_DIG] = "3stack-dig", - [ALC882_6ST_DIG] = "6stack-dig", - [ALC882_ARIMA] = "arima", - [ALC882_W2JC] = "w2jc", - [ALC882_TARGA] = "targa", - [ALC882_ASUS_A7J] = "asus-a7j", - [ALC882_ASUS_A7M] = "asus-a7m", - [ALC885_MACPRO] = "macpro", - [ALC885_MB5] = "mb5", - [ALC885_MBP3] = "mbp3", - [ALC885_IMAC24] = "imac24", - [ALC882_AUTO] = "auto", -}; - -static struct snd_pci_quirk alc882_cfg_tbl[] = { - SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J), - SND_PCI_QUIRK(0x1043, 0x1243, "Asus A7J", ALC882_ASUS_A7J), - SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_ASUS_A7M), - SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC), - SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */ - SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), - {} -}; - -static struct alc_config_preset alc882_presets[] = { - [ALC882_3ST_DIG] = { - .mixers = { alc882_base_mixer }, - .init_verbs = { alc882_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), - .channel_mode = alc882_ch_modes, - .need_dac_fix = 1, - .input_mux = &alc882_capture_source, - }, - [ALC882_6ST_DIG] = { - .mixers = { alc882_base_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes), - .channel_mode = alc882_sixstack_modes, - .input_mux = &alc882_capture_source, - }, - [ALC882_ARIMA] = { - .mixers = { alc882_base_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc882_eapd_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes), - .channel_mode = alc882_sixstack_modes, - .input_mux = &alc882_capture_source, - }, - [ALC882_W2JC] = { - .mixers = { alc882_w2jc_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc882_eapd_verbs, - alc880_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), - .channel_mode = alc880_threestack_modes, - .need_dac_fix = 1, - .input_mux = &alc882_capture_source, - .dig_out_nid = ALC882_DIGOUT_NID, - }, - [ALC885_MBP3] = { - .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer }, - .init_verbs = { alc885_mbp3_init_verbs, - alc880_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .channel_mode = alc885_mbp_6ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_mbp_6ch_modes), - .input_mux = &alc882_capture_source, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc885_mbp3_init_hook, - }, - [ALC885_MB5] = { - .mixers = { alc885_mb5_mixer, alc882_chmode_mixer }, - .init_verbs = { alc885_mb5_init_verbs, - alc880_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .channel_mode = alc885_mb5_6ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes), - .input_mux = &mb5_capture_source, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - }, - [ALC885_MACPRO] = { - .mixers = { alc882_macpro_mixer }, - .init_verbs = { alc882_macpro_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), - .channel_mode = alc882_ch_modes, - .input_mux = &alc882_capture_source, - .init_hook = alc885_macpro_init_hook, - }, - [ALC885_IMAC24] = { - .mixers = { alc885_imac24_mixer }, - .init_verbs = { alc885_imac24_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), - .channel_mode = alc882_ch_modes, - .input_mux = &alc882_capture_source, - .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc885_imac24_init_hook, - }, - [ALC882_TARGA] = { - .mixers = { alc882_targa_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc882_targa_verbs}, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), - .adc_nids = alc882_adc_nids, - .capsrc_nids = alc882_capsrc_nids, - .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes), - .channel_mode = alc882_3ST_6ch_modes, - .need_dac_fix = 1, - .input_mux = &alc882_capture_source, - .unsol_event = alc882_targa_unsol_event, - .init_hook = alc882_targa_init_hook, - }, - [ALC882_ASUS_A7J] = { - .mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc882_asus_a7j_verbs}, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), - .adc_nids = alc882_adc_nids, - .capsrc_nids = alc882_capsrc_nids, - .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes), - .channel_mode = alc882_3ST_6ch_modes, - .need_dac_fix = 1, - .input_mux = &alc882_capture_source, - }, - [ALC882_ASUS_A7M] = { - .mixers = { alc882_asus_a7m_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc882_eapd_verbs, - alc880_gpio1_init_verbs, - alc882_asus_a7m_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), - .channel_mode = alc880_threestack_modes, - .need_dac_fix = 1, - .input_mux = &alc882_capture_source, - }, -}; - - -/* - * Pin config fixes - */ -enum { - PINFIX_ABIT_AW9D_MAX -}; - -static struct alc_pincfg alc882_abit_aw9d_pinfix[] = { - { 0x15, 0x01080104 }, /* side */ - { 0x16, 0x01011012 }, /* rear */ - { 0x17, 0x01016011 }, /* clfe */ - { } -}; - -static const struct alc_pincfg *alc882_pin_fixes[] = { - [PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix, -}; - -static struct snd_pci_quirk alc882_pinfix_tbl[] = { - SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), - {} -}; - -/* - * BIOS auto configuration - */ -static void alc882_auto_set_output_and_unmute(struct hda_codec *codec, - hda_nid_t nid, int pin_type, - int dac_idx) -{ - /* set as output */ - struct alc_spec *spec = codec->spec; - int idx; - - alc_set_pin_output(codec, nid, pin_type); - if (spec->multiout.dac_nids[dac_idx] == 0x25) - idx = 4; - else - idx = spec->multiout.dac_nids[dac_idx] - 2; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx); - -} - -static void alc882_auto_init_multi_out(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - int i; - - for (i = 0; i <= HDA_SIDE; i++) { - hda_nid_t nid = spec->autocfg.line_out_pins[i]; - int pin_type = get_pin_type(spec->autocfg.line_out_type); - if (nid) - alc882_auto_set_output_and_unmute(codec, nid, pin_type, - i); - } -} - -static void alc882_auto_init_hp_out(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - hda_nid_t pin; - - pin = spec->autocfg.hp_pins[0]; - if (pin) /* connect to front */ - /* use dac 0 */ - alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); - pin = spec->autocfg.speaker_pins[0]; - if (pin) - alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); -} - -#define alc882_is_input_pin(nid) alc880_is_input_pin(nid) -#define ALC882_PIN_CD_NID ALC880_PIN_CD_NID - -static void alc882_auto_init_analog_input(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - int i; - - for (i = 0; i < AUTO_PIN_LAST; i++) { - hda_nid_t nid = spec->autocfg.input_pins[i]; - if (!nid) - continue; - alc_set_input_pin(codec, nid, AUTO_PIN_FRONT_MIC /*i*/); - if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_MUTE); - } -} - -static void alc882_auto_init_input_src(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - int c; - - for (c = 0; c < spec->num_adc_nids; c++) { - hda_nid_t conn_list[HDA_MAX_NUM_INPUTS]; - hda_nid_t nid = spec->capsrc_nids[c]; - unsigned int mux_idx; - const struct hda_input_mux *imux; - int conns, mute, idx, item; - - conns = snd_hda_get_connections(codec, nid, conn_list, - ARRAY_SIZE(conn_list)); - if (conns < 0) - continue; - mux_idx = c >= spec->num_mux_defs ? 0 : c; - imux = &spec->input_mux[mux_idx]; - for (idx = 0; idx < conns; idx++) { - /* if the current connection is the selected one, - * unmute it as default - otherwise mute it - */ - mute = AMP_IN_MUTE(idx); - for (item = 0; item < imux->num_items; item++) { - if (imux->items[item].index == idx) { - if (spec->cur_mux[c] == item) - mute = AMP_IN_UNMUTE(idx); - break; - } - } - /* check if we have a selector or mixer - * we could check for the widget type instead, but - * just check for Amp-In presence (in case of mixer - * without amp-in there is something wrong, this - * function shouldn't be used or capsrc nid is wrong) - */ - if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - mute); - else if (mute != AMP_IN_MUTE(idx)) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CONNECT_SEL, - idx); - } - } -} - -/* add mic boosts if needed */ -static int alc_auto_add_mic_boost(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - int err; - hda_nid_t nid; - - nid = spec->autocfg.input_pins[AUTO_PIN_MIC]; - if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) { - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "Mic Boost", - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); - if (err < 0) - return err; - } - nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC]; - if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) { - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "Front Mic Boost", - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); - if (err < 0) - return err; - } - return 0; -} - -/* almost identical with ALC880 parser... */ -static int alc882_parse_auto_config(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - int err = alc880_parse_auto_config(codec); - - if (err < 0) - return err; - else if (!err) - return 0; /* no config found */ - - err = alc_auto_add_mic_boost(codec); - if (err < 0) - return err; - - /* hack - override the init verbs */ - spec->init_verbs[0] = alc882_auto_init_verbs; - - return 1; /* config found */ -} - -/* additional initialization for auto-configuration model */ -static void alc882_auto_init(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - alc882_auto_init_multi_out(codec); - alc882_auto_init_hp_out(codec); - alc882_auto_init_analog_input(codec); - alc882_auto_init_input_src(codec); - if (spec->unsol_event) - alc_inithook(codec); -} - -static int patch_alc883(struct hda_codec *codec); /* called in patch_alc882() */ - -static int patch_alc882(struct hda_codec *codec) -{ - struct alc_spec *spec; - int err, board_config; - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; - - board_config = snd_hda_check_board_config(codec, ALC882_MODEL_LAST, - alc882_models, - alc882_cfg_tbl); - - if (board_config < 0 || board_config >= ALC882_MODEL_LAST) { - /* Pick up systems that don't supply PCI SSID */ - switch (codec->subsystem_id) { - case 0x106b0c00: /* Mac Pro */ - board_config = ALC885_MACPRO; - break; - case 0x106b1000: /* iMac 24 */ - case 0x106b2800: /* AppleTV */ - case 0x106b3e00: /* iMac 24 Aluminium */ - board_config = ALC885_IMAC24; - break; - case 0x106b00a0: /* MacBookPro3,1 - Another revision */ - case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ - case 0x106b00a4: /* MacbookPro4,1 */ - case 0x106b2c00: /* Macbook Pro rev3 */ - /* Macbook 3.1 (0x106b3600) is handled by patch_alc883() */ - case 0x106b3800: /* MacbookPro4,1 - latter revision */ - board_config = ALC885_MBP3; - break; - case 0x106b3f00: /* Macbook 5,1 */ - case 0x106b4000: /* Macbook Pro 5,1 - FIXME: HP jack sense - * seems not working, so apparently - * no perfect solution yet - */ - board_config = ALC885_MB5; - break; - default: - /* ALC889A is handled better as ALC888-compatible */ - if (codec->revision_id == 0x100101 || - codec->revision_id == 0x100103) { - alc_free(codec); - return patch_alc883(codec); - } - printk(KERN_INFO "hda_codec: Unknown model for %s, " - "trying auto-probe from BIOS...\n", - codec->chip_name); - board_config = ALC882_AUTO; - } - } - - alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes); - - if (board_config == ALC882_AUTO) { - /* automatic parse from the BIOS config */ - err = alc882_parse_auto_config(codec); - if (err < 0) { - alc_free(codec); - return err; - } else if (!err) { - printk(KERN_INFO - "hda_codec: Cannot set up configuration " - "from BIOS. Using base mode...\n"); - board_config = ALC882_3ST_DIG; - } - } - - err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) { - alc_free(codec); - return err; - } - - if (board_config != ALC882_AUTO) - setup_preset(spec, &alc882_presets[board_config]); - - spec->stream_analog_playback = &alc882_pcm_analog_playback; - spec->stream_analog_capture = &alc882_pcm_analog_capture; - /* FIXME: setup DAC5 */ - /*spec->stream_analog_alt_playback = &alc880_pcm_analog_alt_playback;*/ - spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture; - - spec->stream_digital_playback = &alc882_pcm_digital_playback; - spec->stream_digital_capture = &alc882_pcm_digital_capture; - - if (!spec->adc_nids && spec->input_mux) { - /* check whether NID 0x07 is valid */ - unsigned int wcap = get_wcaps(codec, 0x07); - /* get type */ - wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - if (wcap != AC_WID_AUD_IN) { - spec->adc_nids = alc882_adc_nids_alt; - spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids_alt); - spec->capsrc_nids = alc882_capsrc_nids_alt; - } else { - spec->adc_nids = alc882_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids); - spec->capsrc_nids = alc882_capsrc_nids; - } - } - set_capture_mixer(spec); - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - - spec->vmaster_nid = 0x0c; - - codec->patch_ops = alc_patch_ops; - if (board_config == ALC882_AUTO) - spec->init_hook = alc882_auto_init; -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (!spec->loopback.amplist) - spec->loopback.amplist = alc882_loopbacks; -#endif - codec->proc_widget_hook = print_realtek_coef; - - return 0; -} - -/* - * ALC883 support - * - * ALC883 is almost identical with ALC880 but has cleaner and more flexible - * configuration. Each pin widget can choose any input DACs and a mixer. - * Each ADC is connected from a mixer of all inputs. This makes possible - * 6-channel independent captures. - * - * In addition, an independent DAC for the multi-playback (not used in this - * driver yet). - */ -#define ALC883_DIGOUT_NID 0x06 -#define ALC883_DIGIN_NID 0x0a - -#define ALC1200_DIGOUT_NID 0x10 - -static hda_nid_t alc883_dac_nids[4] = { - /* front, rear, clfe, rear_surr */ - 0x02, 0x03, 0x04, 0x05 -}; - -static hda_nid_t alc883_adc_nids[2] = { - /* ADC1-2 */ - 0x08, 0x09, -}; - -static hda_nid_t alc883_adc_nids_alt[1] = { - /* ADC1 */ - 0x08, -}; - -static hda_nid_t alc883_adc_nids_rev[2] = { - /* ADC2-1 */ - 0x09, 0x08 -}; - -#define alc889_adc_nids alc880_adc_nids - -static hda_nid_t alc883_capsrc_nids[2] = { 0x23, 0x22 }; - -static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 }; - -#define alc889_capsrc_nids alc882_capsrc_nids - -/* input MUX */ -/* FIXME: should be a matrix-type input source selection */ - -static struct hda_input_mux alc883_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x1 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -static struct hda_input_mux alc883_3stack_6ch_intel = { - .num_items = 4, - .items = { - { "Mic", 0x1 }, - { "Front Mic", 0x0 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -static struct hda_input_mux alc883_lenovo_101e_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x1 }, - { "Line", 0x2 }, - }, -}; - -static struct hda_input_mux alc883_lenovo_nb0763_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "iMic", 0x1 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x0 }, - { "Int Mic", 0x1 }, - }, -}; - -static struct hda_input_mux alc883_lenovo_sky_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x1 }, - { "Line", 0x4 }, - }, -}; - -static struct hda_input_mux alc883_asus_eee1601_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x0 }, - { "Line", 0x2 }, - }, -}; - -static struct hda_input_mux alc889A_mb31_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x0 }, - /* Front Mic (0x01) unused */ - { "Line", 0x2 }, - /* Line 2 (0x03) unused */ - /* CD (0x04) unsused? */ - }, -}; - -/* - * 2ch mode - */ -static struct hda_channel_mode alc883_3ST_2ch_modes[1] = { - { 2, NULL } -}; - -/* - * 2ch mode - */ -static struct hda_verb alc883_3ST_ch2_init[] = { - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } /* end */ -}; - -/* - * 4ch mode - */ -static struct hda_verb alc883_3ST_ch4_init[] = { - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { } /* end */ -}; - -/* - * 6ch mode - */ -static struct hda_verb alc883_3ST_ch6_init[] = { - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { } /* end */ -}; - -static struct hda_channel_mode alc883_3ST_6ch_modes[3] = { - { 2, alc883_3ST_ch2_init }, - { 4, alc883_3ST_ch4_init }, - { 6, alc883_3ST_ch6_init }, -}; - - -/* - * 2ch mode - */ -static struct hda_verb alc883_4ST_ch2_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } /* end */ -}; - -/* - * 4ch mode - */ -static struct hda_verb alc883_4ST_ch4_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { } /* end */ -}; - -/* - * 6ch mode - */ -static struct hda_verb alc883_4ST_ch6_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { } /* end */ -}; - -/* - * 8ch mode - */ -static struct hda_verb alc883_4ST_ch8_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 }, - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { } /* end */ -}; - -static struct hda_channel_mode alc883_4ST_8ch_modes[4] = { - { 2, alc883_4ST_ch2_init }, - { 4, alc883_4ST_ch4_init }, - { 6, alc883_4ST_ch6_init }, - { 8, alc883_4ST_ch8_init }, -}; - - -/* - * 2ch mode - */ -static struct hda_verb alc883_3ST_ch2_intel_init[] = { - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } /* end */ -}; - -/* - * 4ch mode - */ -static struct hda_verb alc883_3ST_ch4_intel_init[] = { - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { } /* end */ -}; - -/* - * 6ch mode - */ -static struct hda_verb alc883_3ST_ch6_intel_init[] = { - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x19, AC_VERB_SET_CONNECT_SEL, 0x02 }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { } /* end */ -}; + {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, -static struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = { - { 2, alc883_3ST_ch2_intel_init }, - { 4, alc883_3ST_ch4_intel_init }, - { 6, alc883_3ST_ch6_intel_init }, -}; + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + * Note: PASD motherboards uses the Line In 2 as the input for + * front panel mic (mic 2) + */ + /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, -/* - * 6ch mode - */ -static struct hda_verb alc883_sixstack_ch6_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, - { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { } /* end */ -}; + /* + * Set up output mixers (0x0c - 0x0f) + */ + /* set vol=0 to output mixers */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* set up input amps for analog loopback */ + /* Amp Indices: DAC = 0, mixer = 1 */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0e, 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)}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, -/* - * 8ch mode - */ -static struct hda_verb alc883_sixstack_ch8_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { } /* end */ -}; + /* FIXME: use matrix-type input source selection */ + /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ + /* Input mixer2 */ + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + /* Input mixer3 */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, -static struct hda_channel_mode alc883_sixstack_modes[2] = { - { 6, alc883_sixstack_ch6_init }, - { 8, alc883_sixstack_ch8_init }, + { } }; /* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */ @@ -7962,34 +7385,7 @@ static struct hda_verb alc883_medion_eapd_verbs[] = { { } }; -/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17 - * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b - */ - -static struct snd_kcontrol_new alc883_base_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, 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("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; +#define alc883_base_mixer alc882_base_mixer static struct snd_kcontrol_new alc883_mitac_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -8309,113 +7705,35 @@ static struct snd_kcontrol_new alc883_asus_eee1601_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { } /* end */ -}; - -static struct snd_kcontrol_new alc883_asus_eee1601_cap_mixer[] = { - HDA_BIND_VOL("Capture Volume", &alc883_bind_cap_vol), - HDA_BIND_SW("Capture Switch", &alc883_bind_cap_switch), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, - { } /* end */ -}; - -static struct snd_kcontrol_new alc883_chmode_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - }, - { } /* end */ -}; - -static struct hda_verb alc883_init_verbs[] = { - /* ADC1: mute amp left and right */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* ADC2: mute amp left and right */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* Front mixer: unmute input/output amp left and right (volume = 0) */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Rear mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* CLFE mixer */ - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Side mixer */ - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - - /* mute analog input loopbacks */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; - /* Front Pin: output 0 (0x0c) */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* Rear Pin: output 1 (0x0d) */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* CLFE Pin: output 2 (0x0e) */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, - /* Side Pin: output 3 (0x0f) */ - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, - /* Mic (rear) pin: input vref at 80% */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Front Mic pin: input vref at 80% */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line In pin: input */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line-2 In: Headphone output (output 0 - 0x0c) */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* CD pin widget for input */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, +static struct snd_kcontrol_new alc883_asus_eee1601_cap_mixer[] = { + HDA_BIND_VOL("Capture Volume", &alc883_bind_cap_vol), + HDA_BIND_SW("Capture Switch", &alc883_bind_cap_switch), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 1, + .info = alc_mux_enum_info, + .get = alc_mux_enum_get, + .put = alc_mux_enum_put, + }, + { } /* end */ +}; - /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer2 */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Input mixer3 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - { } +static struct snd_kcontrol_new alc883_chmode_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, + }, + { } /* end */ }; /* toggle speaker-output according to the hp-jack state */ @@ -8850,69 +8168,6 @@ static void alc883_vaiott_init_hook(struct hda_codec *codec) alc_automute_amp(codec); } -/* - * generic initialization of ADC, input mixers and output mixers - */ -static struct hda_verb alc883_auto_init_verbs[] = { - /* - * Unmute ADC0-2 and set the default input to mic-in - */ - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback - * mixer widget - * Note: PASD motherboards uses the Line In 2 as the input for - * front panel mic (mic 2) - */ - /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - - /* - * Set up output mixers (0x0c - 0x0f) - */ - /* set vol=0 to output mixers */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* set up input amps for analog loopback */ - /* Amp Indices: DAC = 0, mixer = 1 */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, 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)}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer1 */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - /* {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, - /* Input mixer2 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - /* {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, - - { } -}; - static struct hda_verb alc888_asus_m90v_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -9023,25 +8278,44 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res) alc889A_mb31_automute(codec); } + #ifdef CONFIG_SND_HDA_POWER_SAVE -#define alc883_loopbacks alc880_loopbacks +#define alc882_loopbacks alc880_loopbacks #endif /* pcm configuration: identical with ALC880 */ -#define alc883_pcm_analog_playback alc880_pcm_analog_playback -#define alc883_pcm_analog_capture alc880_pcm_analog_capture -#define alc883_pcm_analog_alt_capture alc880_pcm_analog_alt_capture -#define alc883_pcm_digital_playback alc880_pcm_digital_playback -#define alc883_pcm_digital_capture alc880_pcm_digital_capture +#define alc882_pcm_analog_playback alc880_pcm_analog_playback +#define alc882_pcm_analog_capture alc880_pcm_analog_capture +#define alc882_pcm_digital_playback alc880_pcm_digital_playback +#define alc882_pcm_digital_capture alc880_pcm_digital_capture + +static hda_nid_t alc883_slave_dig_outs[] = { + ALC1200_DIGOUT_NID, 0, +}; + +static hda_nid_t alc1200_slave_dig_outs[] = { + ALC883_DIGOUT_NID, 0, +}; /* * configuration and preset */ -static const char *alc883_models[ALC883_MODEL_LAST] = { - [ALC883_3ST_2ch_DIG] = "3stack-dig", +static const char *alc882_models[ALC882_MODEL_LAST] = { + [ALC882_3ST_DIG] = "3stack-dig", + [ALC882_6ST_DIG] = "6stack-dig", + [ALC882_ARIMA] = "arima", + [ALC882_W2JC] = "w2jc", + [ALC882_TARGA] = "targa", + [ALC882_ASUS_A7J] = "asus-a7j", + [ALC882_ASUS_A7M] = "asus-a7m", + [ALC885_MACPRO] = "macpro", + [ALC885_MB5] = "mb5", + [ALC885_MBP3] = "mbp3", + [ALC885_IMAC24] = "imac24", + [ALC883_3ST_2ch_DIG] = "3stack-2ch-dig", [ALC883_3ST_6ch_DIG] = "3stack-6ch-dig", [ALC883_3ST_6ch] = "3stack-6ch", - [ALC883_6ST_DIG] = "6stack-dig", + [ALC883_6ST_DIG] = "alc883-6stack-dig", [ALC883_TARGA_DIG] = "targa-dig", [ALC883_TARGA_2ch_DIG] = "targa-2ch-dig", [ALC883_TARGA_8ch_DIG] = "targa-8ch-dig", @@ -9068,11 +8342,12 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC1200_ASUS_P5Q] = "asus-p5q", [ALC889A_MB31] = "mb31", [ALC883_SONY_VAIO_TT] = "sony-vaio-tt", - [ALC883_AUTO] = "auto", + [ALC882_AUTO] = "auto", }; -static struct snd_pci_quirk alc883_cfg_tbl[] = { - SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_DIG), +static struct snd_pci_quirk alc882_cfg_tbl[] = { + SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC882_6ST_DIG), + SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x010a, "Acer Ferrari 5000", ALC883_ACER_ASPIRE), @@ -9087,8 +8362,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { ALC888_ACER_ASPIRE_8930G), SND_PCI_QUIRK(0x1025, 0x0146, "Acer Aspire 6935G", ALC888_ACER_ASPIRE_8930G), - SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC883_AUTO), - SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC883_AUTO), + SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC882_AUTO), + SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC882_AUTO), SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC888_ACER_ASPIRE_6530G), SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", @@ -9097,30 +8372,44 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { * model=auto should work fine now */ /* SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER), */ + SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL), + SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a72, "HP Educ.ar", ALC888_3ST_HP), + + SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J), + SND_PCI_QUIRK(0x1043, 0x1243, "Asus A7J", ALC882_ASUS_A7J), + SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_ASUS_A7M), SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V), + SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC), + SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG), + SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601), + + SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC883_SONY_VAIO_TT), SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1071, 0x8227, "Mitac 82801H", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL), SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch), - SND_PCI_QUIRK(0x1458, 0xa002, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */ SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG), @@ -9142,11 +8431,13 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), + /* SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), */ SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), SND_PCI_QUIRK_MASK(0x1734, 0xfff0, 0x1100, "FSC AMILO Xi/Pi25xx", ALC883_FUJITSU_PI2515), @@ -9161,24 +8452,175 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x17c0, 0x4085, "MEDION MD96630", ALC888_LENOVO_MS7195_DIG), SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66), + SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC), SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), - SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC883_SONY_VAIO_TT), - {} -}; -static hda_nid_t alc883_slave_dig_outs[] = { - ALC1200_DIGOUT_NID, 0, + {} }; -static hda_nid_t alc1200_slave_dig_outs[] = { - ALC883_DIGOUT_NID, 0, +/* codec SSID table for Intel Mac */ +static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { + SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC885_MBP3), + SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC885_MBP3), + SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC885_MBP3), + SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC885_MACPRO), + SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_IMAC24), + SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_IMAC24), + SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3), + SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31), + SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3), + SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24), + SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5), + /* FIXME: HP jack sense seems not working for MBP 5,1, so apparently + * no perfect solution yet + */ + SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5), + {} /* terminator */ }; -static struct alc_config_preset alc883_presets[] = { +static struct alc_config_preset alc882_presets[] = { + [ALC882_3ST_DIG] = { + .mixers = { alc882_base_mixer }, + .init_verbs = { alc882_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), + .channel_mode = alc882_ch_modes, + .need_dac_fix = 1, + .input_mux = &alc882_capture_source, + }, + [ALC882_6ST_DIG] = { + .mixers = { alc882_base_mixer, alc882_chmode_mixer }, + .init_verbs = { alc882_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes), + .channel_mode = alc882_sixstack_modes, + .input_mux = &alc882_capture_source, + }, + [ALC882_ARIMA] = { + .mixers = { alc882_base_mixer, alc882_chmode_mixer }, + .init_verbs = { alc882_init_verbs, alc882_eapd_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes), + .channel_mode = alc882_sixstack_modes, + .input_mux = &alc882_capture_source, + }, + [ALC882_W2JC] = { + .mixers = { alc882_w2jc_mixer, alc882_chmode_mixer }, + .init_verbs = { alc882_init_verbs, alc882_eapd_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), + .channel_mode = alc880_threestack_modes, + .need_dac_fix = 1, + .input_mux = &alc882_capture_source, + .dig_out_nid = ALC882_DIGOUT_NID, + }, + [ALC885_MBP3] = { + .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer }, + .init_verbs = { alc885_mbp3_init_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .channel_mode = alc885_mbp_6ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mbp_6ch_modes), + .input_mux = &alc882_capture_source, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc885_mbp3_init_hook, + }, + [ALC885_MB5] = { + .mixers = { alc885_mb5_mixer, alc882_chmode_mixer }, + .init_verbs = { alc885_mb5_init_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .channel_mode = alc885_mb5_6ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes), + .input_mux = &mb5_capture_source, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + }, + [ALC885_MACPRO] = { + .mixers = { alc882_macpro_mixer }, + .init_verbs = { alc882_macpro_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), + .channel_mode = alc882_ch_modes, + .input_mux = &alc882_capture_source, + .init_hook = alc885_macpro_init_hook, + }, + [ALC885_IMAC24] = { + .mixers = { alc885_imac24_mixer }, + .init_verbs = { alc885_imac24_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), + .channel_mode = alc882_ch_modes, + .input_mux = &alc882_capture_source, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc885_imac24_init_hook, + }, + [ALC882_TARGA] = { + .mixers = { alc882_targa_mixer, alc882_chmode_mixer }, + .init_verbs = { alc882_init_verbs, alc882_targa_verbs}, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), + .adc_nids = alc882_adc_nids, + .capsrc_nids = alc882_capsrc_nids, + .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes), + .channel_mode = alc882_3ST_6ch_modes, + .need_dac_fix = 1, + .input_mux = &alc882_capture_source, + .unsol_event = alc882_targa_unsol_event, + .init_hook = alc882_targa_init_hook, + }, + [ALC882_ASUS_A7J] = { + .mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer }, + .init_verbs = { alc882_init_verbs, alc882_asus_a7j_verbs}, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), + .adc_nids = alc882_adc_nids, + .capsrc_nids = alc882_capsrc_nids, + .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes), + .channel_mode = alc882_3ST_6ch_modes, + .need_dac_fix = 1, + .input_mux = &alc882_capture_source, + }, + [ALC882_ASUS_A7M] = { + .mixers = { alc882_asus_a7m_mixer, alc882_chmode_mixer }, + .init_verbs = { alc882_init_verbs, alc882_eapd_verbs, + alc880_gpio1_init_verbs, + alc882_asus_a7m_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), + .channel_mode = alc880_threestack_modes, + .need_dac_fix = 1, + .input_mux = &alc882_capture_source, + }, [ALC883_3ST_2ch_DIG] = { .mixers = { alc883_3ST_2ch_mixer }, .init_verbs = { alc883_init_verbs }, @@ -9612,10 +9054,33 @@ static struct alc_config_preset alc883_presets[] = { }; +/* + * Pin config fixes + */ +enum { + PINFIX_ABIT_AW9D_MAX +}; + +static struct alc_pincfg alc882_abit_aw9d_pinfix[] = { + { 0x15, 0x01080104 }, /* side */ + { 0x16, 0x01011012 }, /* rear */ + { 0x17, 0x01016011 }, /* clfe */ + { } +}; + +static const struct alc_pincfg *alc882_pin_fixes[] = { + [PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix, +}; + +static struct snd_pci_quirk alc882_pinfix_tbl[] = { + SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), + {} +}; + /* * BIOS auto configuration */ -static void alc883_auto_set_output_and_unmute(struct hda_codec *codec, +static void alc882_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, int dac_idx) { @@ -9632,7 +9097,7 @@ static void alc883_auto_set_output_and_unmute(struct hda_codec *codec, } -static void alc883_auto_init_multi_out(struct hda_codec *codec) +static void alc882_auto_init_multi_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int i; @@ -9641,12 +9106,12 @@ static void alc883_auto_init_multi_out(struct hda_codec *codec) hda_nid_t nid = spec->autocfg.line_out_pins[i]; int pin_type = get_pin_type(spec->autocfg.line_out_type); if (nid) - alc883_auto_set_output_and_unmute(codec, nid, pin_type, + alc882_auto_set_output_and_unmute(codec, nid, pin_type, i); } } -static void alc883_auto_init_hp_out(struct hda_codec *codec) +static void alc882_auto_init_hp_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; hda_nid_t pin; @@ -9654,42 +9119,114 @@ static void alc883_auto_init_hp_out(struct hda_codec *codec) pin = spec->autocfg.hp_pins[0]; if (pin) /* connect to front */ /* use dac 0 */ - alc883_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); pin = spec->autocfg.speaker_pins[0]; if (pin) - alc883_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); + alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); } -#define alc883_is_input_pin(nid) alc880_is_input_pin(nid) -#define ALC883_PIN_CD_NID ALC880_PIN_CD_NID +#define alc882_is_input_pin(nid) alc880_is_input_pin(nid) +#define ALC882_PIN_CD_NID ALC880_PIN_CD_NID -static void alc883_auto_init_analog_input(struct hda_codec *codec) +static void alc882_auto_init_analog_input(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int i; for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; - if (alc883_is_input_pin(nid)) { - alc_set_input_pin(codec, nid, i); - if (nid != ALC883_PIN_CD_NID && - (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) + if (!nid) + continue; + alc_set_input_pin(codec, nid, AUTO_PIN_FRONT_MIC /*i*/); + if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_MUTE); + } +} + +static void alc882_auto_init_input_src(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int c; + + for (c = 0; c < spec->num_adc_nids; c++) { + hda_nid_t conn_list[HDA_MAX_NUM_INPUTS]; + hda_nid_t nid = spec->capsrc_nids[c]; + unsigned int mux_idx; + const struct hda_input_mux *imux; + int conns, mute, idx, item; + + conns = snd_hda_get_connections(codec, nid, conn_list, + ARRAY_SIZE(conn_list)); + if (conns < 0) + continue; + mux_idx = c >= spec->num_mux_defs ? 0 : c; + imux = &spec->input_mux[mux_idx]; + for (idx = 0; idx < conns; idx++) { + /* if the current connection is the selected one, + * unmute it as default - otherwise mute it + */ + mute = AMP_IN_MUTE(idx); + for (item = 0; item < imux->num_items; item++) { + if (imux->items[item].index == idx) { + if (spec->cur_mux[c] == item) + mute = AMP_IN_UNMUTE(idx); + break; + } + } + /* check if we have a selector or mixer + * we could check for the widget type instead, but + * just check for Amp-In presence (in case of mixer + * without amp-in there is something wrong, this + * function shouldn't be used or capsrc nid is wrong) + */ + if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_MUTE); + mute); + else if (mute != AMP_IN_MUTE(idx)) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, + idx); } } } -#define alc883_auto_init_input_src alc882_auto_init_input_src +/* add mic boosts if needed */ +static int alc_auto_add_mic_boost(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int err; + hda_nid_t nid; + + nid = spec->autocfg.input_pins[AUTO_PIN_MIC]; + if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) { + err = add_control(spec, ALC_CTL_WIDGET_VOL, + "Mic Boost", + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + } + nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC]; + if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) { + err = add_control(spec, ALC_CTL_WIDGET_VOL, + "Front Mic Boost", + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + } + return 0; +} /* almost identical with ALC880 parser... */ -static int alc883_parse_auto_config(struct hda_codec *codec) +static int alc882_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int err = alc880_parse_auto_config(codec); - struct auto_pin_cfg *cfg = &spec->autocfg; + struct auto_pin_cfg *autocfg = &spec->autocfg; + unsigned int wcap; int i; + int err = alc880_parse_auto_config(codec); if (err < 0) return err; @@ -9702,43 +9239,45 @@ static int alc883_parse_auto_config(struct hda_codec *codec) /* hack - override the init verbs */ spec->init_verbs[0] = alc883_auto_init_verbs; + /* if ADC 0x07 is available, initialize it, too */ + wcap = get_wcaps(codec, 0x07); + wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + if (wcap == AC_WID_AUD_IN) + add_verb(spec, alc882_adc1_init_verbs); - /* setup input_mux for ALC889 */ - if (codec->vendor_id == 0x10ec0889) { - /* digital-mic input pin is excluded in alc880_auto_create..() - * because it's under 0x18 - */ - if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 || - cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) { - struct hda_input_mux *imux = &spec->private_imux[0]; - for (i = 1; i < 3; i++) - memcpy(&spec->private_imux[i], - &spec->private_imux[0], - sizeof(spec->private_imux[0])); - imux->items[imux->num_items].label = "Int DMic"; - imux->items[imux->num_items].index = 0x0b; - imux->num_items++; - spec->num_mux_defs = 3; - spec->input_mux = spec->private_imux; - } + /* digital-mic input pin is excluded in alc880_auto_create..() + * because it's under 0x18 + */ + if (autocfg->input_pins[AUTO_PIN_MIC] == 0x12 || + autocfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) { + struct hda_input_mux *imux = &spec->private_imux[0]; + for (i = 1; i < 3; i++) + memcpy(&spec->private_imux[i], + &spec->private_imux[0], + sizeof(spec->private_imux[0])); + imux->items[imux->num_items].label = "Int DMic"; + imux->items[imux->num_items].index = 0x0b; + imux->num_items++; + spec->num_mux_defs = 3; + spec->input_mux = spec->private_imux; } return 1; /* config found */ } /* additional initialization for auto-configuration model */ -static void alc883_auto_init(struct hda_codec *codec) +static void alc882_auto_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - alc883_auto_init_multi_out(codec); - alc883_auto_init_hp_out(codec); - alc883_auto_init_analog_input(codec); - alc883_auto_init_input_src(codec); + alc882_auto_init_multi_out(codec); + alc882_auto_init_hp_out(codec); + alc882_auto_init_analog_input(codec); + alc882_auto_init_input_src(codec); if (spec->unsol_event) alc_inithook(codec); } -static int patch_alc883(struct hda_codec *codec) +static int patch_alc882(struct hda_codec *codec) { struct alc_spec *spec; int err, board_config; @@ -9749,28 +9288,36 @@ static int patch_alc883(struct hda_codec *codec) codec->spec = spec; - alc_fix_pll_init(codec, 0x20, 0x0a, 10); + switch (codec->vendor_id) { + case 0x10ec0882: + case 0x10ec0885: + break; + default: + /* ALC883 and variants */ + alc_fix_pll_init(codec, 0x20, 0x0a, 10); + break; + } - board_config = snd_hda_check_board_config(codec, ALC883_MODEL_LAST, - alc883_models, - alc883_cfg_tbl); - if (board_config < 0 || board_config >= ALC883_MODEL_LAST) { - /* Pick up systems that don't supply PCI SSID */ - switch (codec->subsystem_id) { - case 0x106b3600: /* Macbook 3.1 */ - board_config = ALC889A_MB31; - break; - default: - printk(KERN_INFO - "hda_codec: Unknown model for %s, trying " - "auto-probe from BIOS...\n", codec->chip_name); - board_config = ALC883_AUTO; - } + board_config = snd_hda_check_board_config(codec, ALC882_MODEL_LAST, + alc882_models, + alc882_cfg_tbl); + + if (board_config < 0 || board_config >= ALC882_MODEL_LAST) + board_config = snd_hda_check_board_codec_sid_config(codec, + ALC882_MODEL_LAST, alc882_models, alc882_ssid_cfg_tbl); + + if (board_config < 0 || board_config >= ALC882_MODEL_LAST) { + printk(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", + codec->chip_name); + board_config = ALC882_AUTO; } - if (board_config == ALC883_AUTO) { + alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes); + + if (board_config == ALC882_AUTO) { /* automatic parse from the BIOS config */ - err = alc883_parse_auto_config(codec); + err = alc882_parse_auto_config(codec); if (err < 0) { alc_free(codec); return err; @@ -9778,7 +9325,7 @@ static int patch_alc883(struct hda_codec *codec) printk(KERN_INFO "hda_codec: Cannot set up configuration " "from BIOS. Using base mode...\n"); - board_config = ALC883_3ST_2ch_DIG; + board_config = ALC882_3ST_DIG; } } @@ -9788,63 +9335,61 @@ static int patch_alc883(struct hda_codec *codec) return err; } - if (board_config != ALC883_AUTO) - setup_preset(spec, &alc883_presets[board_config]); + if (board_config != ALC882_AUTO) + setup_preset(spec, &alc882_presets[board_config]); - switch (codec->vendor_id) { - case 0x10ec0888: - if (!spec->num_adc_nids) { - spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); - spec->adc_nids = alc883_adc_nids; - } - if (!spec->capsrc_nids) - spec->capsrc_nids = alc883_capsrc_nids; + spec->stream_analog_playback = &alc882_pcm_analog_playback; + spec->stream_analog_capture = &alc882_pcm_analog_capture; + /* FIXME: setup DAC5 */ + /*spec->stream_analog_alt_playback = &alc880_pcm_analog_alt_playback;*/ + spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture; + + spec->stream_digital_playback = &alc882_pcm_digital_playback; + spec->stream_digital_capture = &alc882_pcm_digital_capture; + + if (codec->vendor_id == 0x10ec0888) spec->init_amp = ALC_INIT_DEFAULT; /* always initialize */ - break; - case 0x10ec0889: - if (!spec->num_adc_nids) { - spec->num_adc_nids = ARRAY_SIZE(alc889_adc_nids); - spec->adc_nids = alc889_adc_nids; - } - if (!spec->capsrc_nids) - spec->capsrc_nids = alc889_capsrc_nids; - break; - default: - if (!spec->num_adc_nids) { - spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); - spec->adc_nids = alc883_adc_nids; + + if (!spec->adc_nids && spec->input_mux) { + int i; + spec->num_adc_nids = 0; + for (i = 0; i < ARRAY_SIZE(alc882_adc_nids); i++) { + hda_nid_t cap; + hda_nid_t nid = alc882_adc_nids[i]; + unsigned int wcap = get_wcaps(codec, nid); + /* get type */ + wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + if (wcap != AC_WID_AUD_IN) + continue; + spec->private_adc_nids[spec->num_adc_nids] = nid; + err = snd_hda_get_connections(codec, nid, &cap, 1); + if (err < 0) + continue; + spec->private_capsrc_nids[spec->num_adc_nids] = cap; + spec->num_adc_nids++; } - if (!spec->capsrc_nids) - spec->capsrc_nids = alc883_capsrc_nids; - break; + spec->adc_nids = spec->private_adc_nids; + spec->capsrc_nids = spec->private_capsrc_nids; } - spec->stream_analog_playback = &alc883_pcm_analog_playback; - spec->stream_analog_capture = &alc883_pcm_analog_capture; - spec->stream_analog_alt_capture = &alc883_pcm_analog_alt_capture; - - spec->stream_digital_playback = &alc883_pcm_digital_playback; - spec->stream_digital_capture = &alc883_pcm_digital_capture; - - if (!spec->cap_mixer) - set_capture_mixer(spec); + set_capture_mixer(spec); set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; codec->patch_ops = alc_patch_ops; - if (board_config == ALC883_AUTO) - spec->init_hook = alc883_auto_init; - + if (board_config == ALC882_AUTO) + spec->init_hook = alc882_auto_init; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) - spec->loopback.amplist = alc883_loopbacks; + spec->loopback.amplist = alc882_loopbacks; #endif codec->proc_widget_hook = print_realtek_coef; return 0; } + /* * ALC262 support */ @@ -17546,23 +17091,23 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0861, .name = "ALC861", .patch = patch_alc861 }, { .id = 0x10ec0862, .name = "ALC861-VD", .patch = patch_alc861vd }, { .id = 0x10ec0662, .rev = 0x100002, .name = "ALC662 rev2", - .patch = patch_alc883 }, + .patch = patch_alc882 }, { .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1", .patch = patch_alc662 }, { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, - { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc883 }, + { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 }, { .id = 0x10ec0885, .rev = 0x100101, .name = "ALC889A", - .patch = patch_alc882 }, /* should be patch_alc883() in future */ + .patch = patch_alc882 }, { .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A", - .patch = patch_alc882 }, /* should be patch_alc883() in future */ + .patch = patch_alc882 }, { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 }, - { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc883 }, + { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 }, { .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200", - .patch = patch_alc883 }, - { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 }, - { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 }, + .patch = patch_alc882 }, + { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc882 }, + { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 }, {} /* terminator */ }; -- cgit v1.2.3 From 0d971c9fcf06d22663040570c3cfe08b137c4b2f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 30 Jun 2009 16:11:11 +0200 Subject: ALSA: hda - Fix input pinctl for ALC882 auto mode alc882_auto_init_analog_input() sets the input pins to VREF-80 regardless of the input pin types although it shouldn't be for line-in pins. This patch fixes the behavior to follow other codecs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6a899e8fdd0c..0f6b6a6f72e3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9125,9 +9125,6 @@ static void alc882_auto_init_hp_out(struct hda_codec *codec) alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); } -#define alc882_is_input_pin(nid) alc880_is_input_pin(nid) -#define ALC882_PIN_CD_NID ALC880_PIN_CD_NID - static void alc882_auto_init_analog_input(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -9137,7 +9134,7 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) hda_nid_t nid = spec->autocfg.input_pins[i]; if (!nid) continue; - alc_set_input_pin(codec, nid, AUTO_PIN_FRONT_MIC /*i*/); + alc_set_input_pin(codec, nid, i); if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, -- cgit v1.2.3 From 416356fcfad46bdebcf8e2afdb94919401ff99d3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 30 Jun 2009 19:05:15 +0100 Subject: ASoC: Convert to dev_pm_ops Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 30 +++++++++++++++++++----------- 1 file changed, 19 insertions(+), 11 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 235503230fe7..dfc03c0bacb6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -619,8 +619,9 @@ static struct snd_pcm_ops soc_pcm_ops = { #ifdef CONFIG_PM /* powers down audio subsystem for suspend */ -static int soc_suspend(struct platform_device *pdev, pm_message_t state) +static int soc_suspend(struct device *dev) { + struct platform_device *pdev = to_platform_device(dev); struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_card *card = socdev->card; struct snd_soc_platform *platform = card->platform; @@ -656,7 +657,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) snd_pcm_suspend_all(card->dai_link[i].pcm); if (card->suspend_pre) - card->suspend_pre(pdev, state); + card->suspend_pre(pdev, PMSG_SUSPEND); for (i = 0; i < card->num_links; i++) { struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; @@ -682,7 +683,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) } if (codec_dev->suspend) - codec_dev->suspend(pdev, state); + codec_dev->suspend(pdev, PMSG_SUSPEND); for (i = 0; i < card->num_links; i++) { struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; @@ -691,7 +692,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) } if (card->suspend_post) - card->suspend_post(pdev, state); + card->suspend_post(pdev, PMSG_SUSPEND); return 0; } @@ -765,8 +766,9 @@ static void soc_resume_deferred(struct work_struct *work) } /* powers up audio subsystem after a suspend */ -static int soc_resume(struct platform_device *pdev) +static int soc_resume(struct device *dev) { + struct platform_device *pdev = to_platform_device(dev); struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_card *card = socdev->card; struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai; @@ -826,7 +828,6 @@ int snd_soc_resume_device(struct device *dev) return 0; } EXPORT_SYMBOL_GPL(snd_soc_resume_device); - #else #define soc_suspend NULL #define soc_resume NULL @@ -1020,32 +1021,39 @@ static int soc_remove(struct platform_device *pdev) return 0; } -static void soc_shutdown(struct platform_device *pdev) +static int soc_poweroff(struct device *dev) { + struct platform_device *pdev = to_platform_device(dev); struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_card *card = socdev->card; if (!card->instantiated) - return; + return 0; /* Flush out pmdown_time work - we actually do want to run it * now, we're shutting down so no imminent restart. */ run_delayed_work(&card->delayed_work); snd_soc_dapm_shutdown(socdev); + + return 0; } +static struct dev_pm_ops soc_pm_ops = { + .suspend = soc_suspend, + .resume = soc_resume, + .poweroff = soc_poweroff, +}; + /* ASoC platform driver */ static struct platform_driver soc_driver = { .driver = { .name = "soc-audio", .owner = THIS_MODULE, + .pm = &soc_pm_ops, }, .probe = soc_probe, .remove = soc_remove, - .suspend = soc_suspend, - .resume = soc_resume, - .shutdown = soc_shutdown, }; /* create a new pcm */ -- cgit v1.2.3 From b5025c50b5e817b3e509ad7e569f131b80d7c223 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Jul 2009 18:05:27 +0200 Subject: ALSA: hda - Allow FLOAT PCM format So far, the FLOAT PCM format is used only exclusivley set. But this can be a combination with other formats. This patch changes the parser to allow the FLOAT format in addition to other PCM formats. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 506f46ef0304..263d124de611 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2690,11 +2690,11 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, bps = 20; } } - else if (streams == AC_SUPFMT_FLOAT32) { - /* should be exclusive */ + if (streams & AC_SUPFMT_FLOAT32) { formats |= SNDRV_PCM_FMTBIT_FLOAT_LE; bps = 32; - } else if (streams == AC_SUPFMT_AC3) { + } + if (streams == AC_SUPFMT_AC3) { /* should be exclusive */ /* temporary hack: we have still no proper support * for the direct AC3 stream... -- cgit v1.2.3 From 1dcf98ff8e2a4571a2accb852686023b47ca629a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 1 Jul 2009 18:28:54 +0100 Subject: ASoC: Add WM8523 CODEC driver The WM8523 is a high performance stereo DAC with integral charge pump providing 2Vrms line driver outputs using a single 3.3V power supply rail. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8523.c | 755 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8523.h | 160 ++++++++++ 4 files changed, 921 insertions(+) create mode 100644 sound/soc/codecs/wm8523.c create mode 100644 sound/soc/codecs/wm8523.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 021dbdfa5b92..68ea5b6648df 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -30,6 +30,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8350 if MFD_WM8350 select SND_SOC_WM8400 if MFD_WM8400 select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8523 if I2C select SND_SOC_WM8580 if I2C select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI @@ -130,6 +131,9 @@ config SND_SOC_WM8400 config SND_SOC_WM8510 tristate +config SND_SOC_WM8523 + tristate + config SND_SOC_WM8580 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index e520c2b7f0e0..8ce28a34e8e9 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -18,6 +18,7 @@ snd-soc-uda1380-objs := uda1380.o snd-soc-wm8350-objs := wm8350.o snd-soc-wm8400-objs := wm8400.o snd-soc-wm8510-objs := wm8510.o +snd-soc-wm8523-objs := wm8523.o snd-soc-wm8580-objs := wm8580.o snd-soc-wm8728-objs := wm8728.o snd-soc-wm8731-objs := wm8731.o @@ -56,6 +57,7 @@ obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o +obj-$(CONFIG_SND_SOC_WM8523) += snd-soc-wm8523.o obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c new file mode 100644 index 000000000000..3b499ae7ce6c --- /dev/null +++ b/sound/soc/codecs/wm8523.c @@ -0,0 +1,755 @@ +/* + * wm8523.c -- WM8523 ALSA SoC Audio driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8523.h" + +static struct snd_soc_codec *wm8523_codec; +struct snd_soc_codec_device soc_codec_dev_wm8523; + +#define WM8523_NUM_SUPPLIES 2 +static const char *wm8523_supply_names[WM8523_NUM_SUPPLIES] = { + "AVDD", + "LINEVDD", +}; + +#define WM8523_NUM_RATES 7 + +/* codec private data */ +struct wm8523_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8523_REGISTER_COUNT]; + struct regulator_bulk_data supplies[WM8523_NUM_SUPPLIES]; + unsigned int sysclk; + unsigned int rate_constraint_list[WM8523_NUM_RATES]; + struct snd_pcm_hw_constraint_list rate_constraint; +}; + +static const u16 wm8523_reg[WM8523_REGISTER_COUNT] = { + 0x8523, /* R0 - DEVICE_ID */ + 0x0001, /* R1 - REVISION */ + 0x0000, /* R2 - PSCTRL1 */ + 0x1812, /* R3 - AIF_CTRL1 */ + 0x0000, /* R4 - AIF_CTRL2 */ + 0x0001, /* R5 - DAC_CTRL3 */ + 0x0190, /* R6 - DAC_GAINL */ + 0x0190, /* R7 - DAC_GAINR */ + 0x0000, /* R8 - ZERO_DETECT */ +}; + +static int wm8523_volatile(unsigned int reg) +{ + switch (reg) { + case WM8523_DEVICE_ID: + case WM8523_REVISION: + return 1; + default: + return 0; + } +} + +static int wm8523_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct wm8523_priv *wm8523 = codec->private_data; + u8 data[3]; + + BUG_ON(reg > WM8523_MAX_REGISTER); + + data[0] = reg; + data[1] = (value >> 8) & 0x00ff; + data[2] = value & 0x00ff; + + if (!wm8523_volatile(reg)) + wm8523->reg_cache[reg] = value; + if (codec->hw_write(codec->control_data, data, 3) == 3) + return 0; + else + return -EIO; +} + +static int wm8523_reset(struct snd_soc_codec *codec) +{ + return wm8523_write(codec, WM8523_DEVICE_ID, 0); +} + +static unsigned int wm8523_read_hw(struct snd_soc_codec *codec, u8 reg) +{ + struct i2c_msg xfer[2]; + u16 data; + int ret; + struct i2c_client *i2c = codec->control_data; + + /* Write register */ + xfer[0].addr = i2c->addr; + xfer[0].flags = 0; + xfer[0].len = 1; + xfer[0].buf = ® + + /* Read data */ + xfer[1].addr = i2c->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 2; + xfer[1].buf = (u8 *)&data; + + ret = i2c_transfer(i2c->adapter, xfer, 2); + if (ret != 2) { + dev_err(codec->dev, "Failed to read 0x%x: %d\n", reg, ret); + return 0; + } + + return (data >> 8) | ((data & 0xff) << 8); +} + + +static unsigned int wm8523_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *reg_cache = codec->reg_cache; + + BUG_ON(reg > WM8523_MAX_REGISTER); + + if (wm8523_volatile(reg)) + return wm8523_read_hw(codec, reg); + else + return reg_cache[reg]; +} + +static const DECLARE_TLV_DB_SCALE(dac_tlv, -10000, 25, 0); + +static const char *wm8523_zd_count_text[] = { + "1024", + "2048", +}; + +static const struct soc_enum wm8523_zc_count = + SOC_ENUM_SINGLE(WM8523_ZERO_DETECT, 0, 2, wm8523_zd_count_text); + +static const struct snd_kcontrol_new wm8523_snd_controls[] = { +SOC_DOUBLE_R_TLV("Playback Volume", WM8523_DAC_GAINL, WM8523_DAC_GAINR, + 0, 448, 0, dac_tlv), +SOC_SINGLE("ZC Switch", WM8523_DAC_CTRL3, 4, 1, 0), +SOC_SINGLE("Playback Deemphasis Switch", WM8523_AIF_CTRL1, 8, 1, 0), +SOC_DOUBLE("Playback Switch", WM8523_DAC_CTRL3, 2, 3, 1, 1), +SOC_SINGLE("Volume Ramp Up Switch", WM8523_DAC_CTRL3, 1, 1, 0), +SOC_SINGLE("Volume Ramp Down Switch", WM8523_DAC_CTRL3, 0, 1, 0), +SOC_ENUM("Zero Detect Count", wm8523_zc_count), +}; + +static const struct snd_soc_dapm_widget wm8523_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("LINEVOUTL"), +SND_SOC_DAPM_OUTPUT("LINEVOUTR"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + { "LINEVOUTL", NULL, "DAC" }, + { "LINEVOUTR", NULL, "DAC" }, +}; + +static int wm8523_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8523_dapm_widgets, + ARRAY_SIZE(wm8523_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static struct { + int value; + int ratio; +} lrclk_ratios[WM8523_NUM_RATES] = { + { 1, 128 }, + { 2, 192 }, + { 3, 256 }, + { 4, 384 }, + { 5, 512 }, + { 6, 768 }, + { 7, 1152 }, +}; + +static int wm8523_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8523_priv *wm8523 = codec->private_data; + + /* The set of sample rates that can be supported depends on the + * MCLK supplied to the CODEC - enforce this. + */ + if (!wm8523->sysclk) { + dev_err(codec->dev, + "No MCLK configured, call set_sysclk() on init\n"); + return -EINVAL; + } + + return 0; + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &wm8523->rate_constraint); + + return 0; +} + +static int wm8523_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8523_priv *wm8523 = codec->private_data; + int i; + u16 aifctrl1 = wm8523_read(codec, WM8523_AIF_CTRL1); + u16 aifctrl2 = wm8523_read(codec, WM8523_AIF_CTRL2); + + /* Find a supported LRCLK ratio */ + for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) { + if (wm8523->sysclk / params_rate(params) == + lrclk_ratios[i].ratio) + break; + } + + /* Should never happen, should be handled by constraints */ + if (i == ARRAY_SIZE(lrclk_ratios)) { + dev_err(codec->dev, "MCLK/fs ratio %d unsupported\n", + wm8523->sysclk / params_rate(params)); + return -EINVAL; + } + + aifctrl2 &= ~WM8523_SR_MASK; + aifctrl2 |= lrclk_ratios[i].value; + + aifctrl1 &= ~WM8523_WL_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + aifctrl1 |= 0x8; + break; + case SNDRV_PCM_FORMAT_S24_LE: + aifctrl1 |= 0x10; + break; + case SNDRV_PCM_FORMAT_S32_LE: + aifctrl1 |= 0x18; + break; + } + + wm8523_write(codec, WM8523_AIF_CTRL1, aifctrl1); + wm8523_write(codec, WM8523_AIF_CTRL2, aifctrl2); + + return 0; +} + +static int wm8523_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8523_priv *wm8523 = codec->private_data; + unsigned int val; + int i; + + wm8523->sysclk = freq; + + wm8523->rate_constraint.count = 0; + for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) { + val = freq / lrclk_ratios[i].ratio; + /* Check that it's a standard rate since core can't + * cope with others and having the odd rates confuses + * constraint matching. + */ + switch (val) { + case 8000: + case 11025: + case 16000: + case 22050: + case 32000: + case 44100: + case 48000: + case 64000: + case 88200: + case 96000: + case 176400: + case 192000: + dev_dbg(codec->dev, "Supported sample rate: %dHz\n", + val); + wm8523->rate_constraint_list[i] = val; + wm8523->rate_constraint.count++; + break; + default: + dev_dbg(codec->dev, "Skipping sample rate: %dHz\n", + val); + } + } + + /* Need at least one supported rate... */ + if (wm8523->rate_constraint.count == 0) + return -EINVAL; + + return 0; +} + + +static int wm8523_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 aifctrl1 = wm8523_read(codec, WM8523_AIF_CTRL1); + + aifctrl1 &= ~(WM8523_BCLK_INV_MASK | WM8523_LRCLK_INV_MASK | + WM8523_FMT_MASK | WM8523_AIF_MSTR_MASK); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + aifctrl1 |= WM8523_AIF_MSTR; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aifctrl1 |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + aifctrl1 |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + aifctrl1 |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + aifctrl1 |= 0x0023; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + aifctrl1 |= WM8523_BCLK_INV | WM8523_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + aifctrl1 |= WM8523_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + aifctrl1 |= WM8523_LRCLK_INV; + break; + default: + return -EINVAL; + } + + wm8523_write(codec, WM8523_AIF_CTRL1, aifctrl1); + + return 0; +} + +static int wm8523_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm8523_priv *wm8523 = codec->private_data; + int ret, i; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* Full power on */ + snd_soc_update_bits(codec, WM8523_PSCTRL1, + WM8523_SYS_ENA_MASK, 3); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(wm8523->supplies), + wm8523->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", + ret); + return ret; + } + + /* Initial power up */ + snd_soc_update_bits(codec, WM8523_PSCTRL1, + WM8523_SYS_ENA_MASK, 1); + + /* Sync back default/cached values */ + for (i = WM8523_AIF_CTRL1; + i < WM8523_MAX_REGISTER; i++) + wm8523_write(codec, i, wm8523->reg_cache[i]); + + + msleep(100); + } + + /* Power up to mute */ + snd_soc_update_bits(codec, WM8523_PSCTRL1, + WM8523_SYS_ENA_MASK, 2); + + break; + + case SND_SOC_BIAS_OFF: + /* The chip runs through the power down sequence for us. */ + snd_soc_update_bits(codec, WM8523_PSCTRL1, + WM8523_SYS_ENA_MASK, 0); + msleep(100); + + regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), + wm8523->supplies); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8523_RATES SNDRV_PCM_RATE_8000_192000 + +#define WM8523_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops wm8523_dai_ops = { + .startup = wm8523_startup, + .hw_params = wm8523_hw_params, + .set_sysclk = wm8523_set_dai_sysclk, + .set_fmt = wm8523_set_dai_fmt, +}; + +struct snd_soc_dai wm8523_dai = { + .name = "WM8523", + .playback = { + .stream_name = "Playback", + .channels_min = 2, /* Mono modes not yet supported */ + .channels_max = 2, + .rates = WM8523_RATES, + .formats = WM8523_FORMATS, + }, + .ops = &wm8523_dai_ops, +}; +EXPORT_SYMBOL_GPL(wm8523_dai); + +#ifdef CONFIG_PM +static int wm8523_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8523_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8523_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8523_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define wm8523_suspend NULL +#define wm8523_resume NULL +#endif + +static int wm8523_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8523_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8523_codec; + codec = wm8523_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8523_snd_controls, + ARRAY_SIZE(wm8523_snd_controls)); + wm8523_add_widgets(codec); + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +static int wm8523_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8523 = { + .probe = wm8523_probe, + .remove = wm8523_remove, + .suspend = wm8523_suspend, + .resume = wm8523_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8523); + +static int wm8523_register(struct wm8523_priv *wm8523) +{ + int ret; + struct snd_soc_codec *codec = &wm8523->codec; + int i; + + if (wm8523_codec) { + dev_err(codec->dev, "Another WM8523 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8523; + codec->name = "WM8523"; + codec->owner = THIS_MODULE; + codec->read = wm8523_read; + codec->write = wm8523_write; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8523_set_bias_level; + codec->dai = &wm8523_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8523_REGISTER_COUNT; + codec->reg_cache = &wm8523->reg_cache; + + wm8523->rate_constraint.list = &wm8523->rate_constraint_list[0]; + wm8523->rate_constraint.count = + ARRAY_SIZE(wm8523->rate_constraint_list); + + memcpy(codec->reg_cache, wm8523_reg, sizeof(wm8523_reg)); + + for (i = 0; i < ARRAY_SIZE(wm8523->supplies); i++) + wm8523->supplies[i].supply = wm8523_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8523->supplies), + wm8523->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8523->supplies), + wm8523->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } + + ret = wm8523_read(codec, WM8523_DEVICE_ID); + if (ret < 0) { + dev_err(codec->dev, "Failed to read ID register\n"); + goto err_enable; + } + if (ret != wm8523_reg[WM8523_DEVICE_ID]) { + dev_err(codec->dev, "Device is not a WM8523, ID is %x\n", ret); + ret = -EINVAL; + goto err_enable; + } + + ret = wm8523_read(codec, WM8523_REVISION); + if (ret < 0) { + dev_err(codec->dev, "Failed to read revision register\n"); + goto err_enable; + } + dev_info(codec->dev, "revision %c\n", + (ret & WM8523_CHIP_REV_MASK) + 'A'); + + ret = wm8523_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err_enable; + } + + wm8523_dai.dev = codec->dev; + + /* Change some default settings - latch VU and enable ZC */ + wm8523->reg_cache[WM8523_DAC_GAINR] |= WM8523_DACR_VU; + wm8523->reg_cache[WM8523_DAC_CTRL3] |= WM8523_ZC; + + wm8523_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Bias level configuration will have done an extra enable */ + regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); + + wm8523_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8523_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); +err_get: + regulator_bulk_free(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); +err: + kfree(wm8523); + return ret; +} + +static void wm8523_unregister(struct wm8523_priv *wm8523) +{ + wm8523_set_bias_level(&wm8523->codec, SND_SOC_BIAS_OFF); + regulator_bulk_free(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); + snd_soc_unregister_dai(&wm8523_dai); + snd_soc_unregister_codec(&wm8523->codec); + kfree(wm8523); + wm8523_codec = NULL; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static __devinit int wm8523_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8523_priv *wm8523; + struct snd_soc_codec *codec; + + wm8523 = kzalloc(sizeof(struct wm8523_priv), GFP_KERNEL); + if (wm8523 == NULL) + return -ENOMEM; + + codec = &wm8523->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8523); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8523_register(wm8523); +} + +static __devexit int wm8523_i2c_remove(struct i2c_client *client) +{ + struct wm8523_priv *wm8523 = i2c_get_clientdata(client); + wm8523_unregister(wm8523); + return 0; +} + +#ifdef CONFIG_PM +static int wm8523_i2c_suspend(struct i2c_client *i2c, pm_message_t msg) +{ + return snd_soc_suspend_device(&i2c->dev); +} + +static int wm8523_i2c_resume(struct i2c_client *i2c) +{ + return snd_soc_resume_device(&i2c->dev); +} +#else +#define wm8523_i2c_suspend NULL +#define wm8523_i2c_resume NULL +#endif + +static const struct i2c_device_id wm8523_i2c_id[] = { + { "wm8523", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8523_i2c_id); + +static struct i2c_driver wm8523_i2c_driver = { + .driver = { + .name = "WM8523", + .owner = THIS_MODULE, + }, + .probe = wm8523_i2c_probe, + .remove = __devexit_p(wm8523_i2c_remove), + .suspend = wm8523_i2c_suspend, + .resume = wm8523_i2c_resume, + .id_table = wm8523_i2c_id, +}; +#endif + +static int __init wm8523_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8523_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8523 I2C driver: %d\n", + ret); + } +#endif + return 0; +} +module_init(wm8523_modinit); + +static void __exit wm8523_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8523_i2c_driver); +#endif +} +module_exit(wm8523_exit); + +MODULE_DESCRIPTION("ASoC WM8523 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8523.h b/sound/soc/codecs/wm8523.h new file mode 100644 index 000000000000..1aa9ce3e1357 --- /dev/null +++ b/sound/soc/codecs/wm8523.h @@ -0,0 +1,160 @@ +/* + * wm8523.h -- WM8423 ASoC driver + * + * Copyright 2009 Wolfson Microelectronics, plc + * + * Author: Mark Brown + * + * Based on wm8753.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8523_H +#define _WM8523_H + +/* + * Register values. + */ +#define WM8523_DEVICE_ID 0x00 +#define WM8523_REVISION 0x01 +#define WM8523_PSCTRL1 0x02 +#define WM8523_AIF_CTRL1 0x03 +#define WM8523_AIF_CTRL2 0x04 +#define WM8523_DAC_CTRL3 0x05 +#define WM8523_DAC_GAINL 0x06 +#define WM8523_DAC_GAINR 0x07 +#define WM8523_ZERO_DETECT 0x08 + +#define WM8523_REGISTER_COUNT 9 +#define WM8523_MAX_REGISTER 0x08 + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - DEVICE_ID + */ +#define WM8523_CHIP_ID_MASK 0xFFFF /* CHIP_ID - [15:0] */ +#define WM8523_CHIP_ID_SHIFT 0 /* CHIP_ID - [15:0] */ +#define WM8523_CHIP_ID_WIDTH 16 /* CHIP_ID - [15:0] */ + +/* + * R1 (0x01) - REVISION + */ +#define WM8523_CHIP_REV_MASK 0x0007 /* CHIP_REV - [2:0] */ +#define WM8523_CHIP_REV_SHIFT 0 /* CHIP_REV - [2:0] */ +#define WM8523_CHIP_REV_WIDTH 3 /* CHIP_REV - [2:0] */ + +/* + * R2 (0x02) - PSCTRL1 + */ +#define WM8523_SYS_ENA_MASK 0x0003 /* SYS_ENA - [1:0] */ +#define WM8523_SYS_ENA_SHIFT 0 /* SYS_ENA - [1:0] */ +#define WM8523_SYS_ENA_WIDTH 2 /* SYS_ENA - [1:0] */ + +/* + * R3 (0x03) - AIF_CTRL1 + */ +#define WM8523_TDM_MODE_MASK 0x1800 /* TDM_MODE - [12:11] */ +#define WM8523_TDM_MODE_SHIFT 11 /* TDM_MODE - [12:11] */ +#define WM8523_TDM_MODE_WIDTH 2 /* TDM_MODE - [12:11] */ +#define WM8523_TDM_SLOT_MASK 0x0600 /* TDM_SLOT - [10:9] */ +#define WM8523_TDM_SLOT_SHIFT 9 /* TDM_SLOT - [10:9] */ +#define WM8523_TDM_SLOT_WIDTH 2 /* TDM_SLOT - [10:9] */ +#define WM8523_DEEMPH 0x0100 /* DEEMPH */ +#define WM8523_DEEMPH_MASK 0x0100 /* DEEMPH */ +#define WM8523_DEEMPH_SHIFT 8 /* DEEMPH */ +#define WM8523_DEEMPH_WIDTH 1 /* DEEMPH */ +#define WM8523_AIF_MSTR 0x0080 /* AIF_MSTR */ +#define WM8523_AIF_MSTR_MASK 0x0080 /* AIF_MSTR */ +#define WM8523_AIF_MSTR_SHIFT 7 /* AIF_MSTR */ +#define WM8523_AIF_MSTR_WIDTH 1 /* AIF_MSTR */ +#define WM8523_LRCLK_INV 0x0040 /* LRCLK_INV */ +#define WM8523_LRCLK_INV_MASK 0x0040 /* LRCLK_INV */ +#define WM8523_LRCLK_INV_SHIFT 6 /* LRCLK_INV */ +#define WM8523_LRCLK_INV_WIDTH 1 /* LRCLK_INV */ +#define WM8523_BCLK_INV 0x0020 /* BCLK_INV */ +#define WM8523_BCLK_INV_MASK 0x0020 /* BCLK_INV */ +#define WM8523_BCLK_INV_SHIFT 5 /* BCLK_INV */ +#define WM8523_BCLK_INV_WIDTH 1 /* BCLK_INV */ +#define WM8523_WL_MASK 0x0018 /* WL - [4:3] */ +#define WM8523_WL_SHIFT 3 /* WL - [4:3] */ +#define WM8523_WL_WIDTH 2 /* WL - [4:3] */ +#define WM8523_FMT_MASK 0x0007 /* FMT - [2:0] */ +#define WM8523_FMT_SHIFT 0 /* FMT - [2:0] */ +#define WM8523_FMT_WIDTH 3 /* FMT - [2:0] */ + +/* + * R4 (0x04) - AIF_CTRL2 + */ +#define WM8523_DAC_OP_MUX_MASK 0x00C0 /* DAC_OP_MUX - [7:6] */ +#define WM8523_DAC_OP_MUX_SHIFT 6 /* DAC_OP_MUX - [7:6] */ +#define WM8523_DAC_OP_MUX_WIDTH 2 /* DAC_OP_MUX - [7:6] */ +#define WM8523_BCLKDIV_MASK 0x0038 /* BCLKDIV - [5:3] */ +#define WM8523_BCLKDIV_SHIFT 3 /* BCLKDIV - [5:3] */ +#define WM8523_BCLKDIV_WIDTH 3 /* BCLKDIV - [5:3] */ +#define WM8523_SR_MASK 0x0007 /* SR - [2:0] */ +#define WM8523_SR_SHIFT 0 /* SR - [2:0] */ +#define WM8523_SR_WIDTH 3 /* SR - [2:0] */ + +/* + * R5 (0x05) - DAC_CTRL3 + */ +#define WM8523_ZC 0x0010 /* ZC */ +#define WM8523_ZC_MASK 0x0010 /* ZC */ +#define WM8523_ZC_SHIFT 4 /* ZC */ +#define WM8523_ZC_WIDTH 1 /* ZC */ +#define WM8523_DACR 0x0008 /* DACR */ +#define WM8523_DACR_MASK 0x0008 /* DACR */ +#define WM8523_DACR_SHIFT 3 /* DACR */ +#define WM8523_DACR_WIDTH 1 /* DACR */ +#define WM8523_DACL 0x0004 /* DACL */ +#define WM8523_DACL_MASK 0x0004 /* DACL */ +#define WM8523_DACL_SHIFT 2 /* DACL */ +#define WM8523_DACL_WIDTH 1 /* DACL */ +#define WM8523_VOL_UP_RAMP 0x0002 /* VOL_UP_RAMP */ +#define WM8523_VOL_UP_RAMP_MASK 0x0002 /* VOL_UP_RAMP */ +#define WM8523_VOL_UP_RAMP_SHIFT 1 /* VOL_UP_RAMP */ +#define WM8523_VOL_UP_RAMP_WIDTH 1 /* VOL_UP_RAMP */ +#define WM8523_VOL_DOWN_RAMP 0x0001 /* VOL_DOWN_RAMP */ +#define WM8523_VOL_DOWN_RAMP_MASK 0x0001 /* VOL_DOWN_RAMP */ +#define WM8523_VOL_DOWN_RAMP_SHIFT 0 /* VOL_DOWN_RAMP */ +#define WM8523_VOL_DOWN_RAMP_WIDTH 1 /* VOL_DOWN_RAMP */ + +/* + * R6 (0x06) - DAC_GAINL + */ +#define WM8523_DACL_VU 0x0200 /* DACL_VU */ +#define WM8523_DACL_VU_MASK 0x0200 /* DACL_VU */ +#define WM8523_DACL_VU_SHIFT 9 /* DACL_VU */ +#define WM8523_DACL_VU_WIDTH 1 /* DACL_VU */ +#define WM8523_DACL_VOL_MASK 0x01FF /* DACL_VOL - [8:0] */ +#define WM8523_DACL_VOL_SHIFT 0 /* DACL_VOL - [8:0] */ +#define WM8523_DACL_VOL_WIDTH 9 /* DACL_VOL - [8:0] */ + +/* + * R7 (0x07) - DAC_GAINR + */ +#define WM8523_DACR_VU 0x0200 /* DACR_VU */ +#define WM8523_DACR_VU_MASK 0x0200 /* DACR_VU */ +#define WM8523_DACR_VU_SHIFT 9 /* DACR_VU */ +#define WM8523_DACR_VU_WIDTH 1 /* DACR_VU */ +#define WM8523_DACR_VOL_MASK 0x01FF /* DACR_VOL - [8:0] */ +#define WM8523_DACR_VOL_SHIFT 0 /* DACR_VOL - [8:0] */ +#define WM8523_DACR_VOL_WIDTH 9 /* DACR_VOL - [8:0] */ + +/* + * R8 (0x08) - ZERO_DETECT + */ +#define WM8523_ZD_COUNT_MASK 0x0003 /* ZD_COUNT - [1:0] */ +#define WM8523_ZD_COUNT_SHIFT 0 /* ZD_COUNT - [1:0] */ +#define WM8523_ZD_COUNT_WIDTH 2 /* ZD_COUNT - [1:0] */ + +extern struct snd_soc_dai wm8523_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8523; + +#endif -- cgit v1.2.3 From cd775387244e379ef9b284f9b6aff6ee069f4d12 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 2 Jul 2009 10:28:56 +0200 Subject: ALSA: ice1724 - Fix section mismatch Now snd_vt1724_chip_reset() is used in the resume callback, thus it cannot be __devinit. Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1724.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 6a560021e11f..fdd6c6d28bce 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2267,7 +2267,7 @@ static int __devinit snd_vt1724_read_eeprom(struct snd_ice1712 *ice, -static void __devinit snd_vt1724_chip_reset(struct snd_ice1712 *ice) +static void snd_vt1724_chip_reset(struct snd_ice1712 *ice) { outb(VT1724_RESET , ICEREG1724(ice, CONTROL)); inb(ICEREG1724(ice, CONTROL)); /* pci posting flush */ -- cgit v1.2.3 From cb6381225a8064b0911dced3eb10f00bd5520c85 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jul 2009 10:56:10 +0200 Subject: ALSA: hda - Add missing mixer amp initialization for ALC882 After merting patch_alc882() and patch_alc883(), the initialization of mixer amp 0x0b was missing in alc882_base_init_verbs[]. This is usually no critical problem, but it can disable the power-saving as the default state, so better to put to mute these channels. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0f6b6a6f72e3..d15d83ee4ad8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6849,6 +6849,13 @@ static struct hda_verb alc882_base_init_verbs[] = { {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* mute analog input loopbacks */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* Front Pin: output 0 (0x0c) */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, -- cgit v1.2.3 From 8ab9e0af6d7709a781b60a51711ddf1d43bd22bb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jul 2009 10:58:12 +0200 Subject: ALSA: hda - Manually expand alc882_init_verbs Instead of expanding alc882_init_verbs to two elements via a macro, manually expand to each entry. This makes clear that some have already the full slot for init_verbs array (currently 5). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 26 ++++++++++++++------------ 1 file changed, 14 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d15d83ee4ad8..07e260a5ddf0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6922,9 +6922,6 @@ static struct hda_verb alc882_adc1_init_verbs[] = { { } }; -/* HACK - expand to two elements */ -#define alc882_init_verbs alc882_base_init_verbs, alc882_adc1_init_verbs - static struct hda_verb alc882_eapd_verbs[] = { /* change to EAPD mode */ {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, @@ -8492,7 +8489,8 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { static struct alc_config_preset alc882_presets[] = { [ALC882_3ST_DIG] = { .mixers = { alc882_base_mixer }, - .init_verbs = { alc882_init_verbs }, + .init_verbs = { alc882_base_init_verbs, + alc882_adc1_init_verbs }, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .dig_out_nid = ALC882_DIGOUT_NID, @@ -8504,7 +8502,8 @@ static struct alc_config_preset alc882_presets[] = { }, [ALC882_6ST_DIG] = { .mixers = { alc882_base_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs }, + .init_verbs = { alc882_base_init_verbs, + alc882_adc1_init_verbs }, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .dig_out_nid = ALC882_DIGOUT_NID, @@ -8515,7 +8514,8 @@ static struct alc_config_preset alc882_presets[] = { }, [ALC882_ARIMA] = { .mixers = { alc882_base_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc882_eapd_verbs }, + .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs, + alc882_eapd_verbs }, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes), @@ -8524,8 +8524,8 @@ static struct alc_config_preset alc882_presets[] = { }, [ALC882_W2JC] = { .mixers = { alc882_w2jc_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc882_eapd_verbs, - alc880_gpio1_init_verbs }, + .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs, + alc882_eapd_verbs, alc880_gpio1_init_verbs }, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), @@ -8587,7 +8587,8 @@ static struct alc_config_preset alc882_presets[] = { }, [ALC882_TARGA] = { .mixers = { alc882_targa_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc882_targa_verbs}, + .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs, + alc882_targa_verbs}, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .dig_out_nid = ALC882_DIGOUT_NID, @@ -8603,7 +8604,8 @@ static struct alc_config_preset alc882_presets[] = { }, [ALC882_ASUS_A7J] = { .mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc882_asus_a7j_verbs}, + .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs, + alc882_asus_a7j_verbs}, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .dig_out_nid = ALC882_DIGOUT_NID, @@ -8617,8 +8619,8 @@ static struct alc_config_preset alc882_presets[] = { }, [ALC882_ASUS_A7M] = { .mixers = { alc882_asus_a7m_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc882_eapd_verbs, - alc880_gpio1_init_verbs, + .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs, + alc882_eapd_verbs, alc880_gpio1_init_verbs, alc882_asus_a7m_verbs }, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, -- cgit v1.2.3 From b0bb3aa6233dccfccd040793d0d9ce838e4890e1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jul 2009 23:25:37 +0200 Subject: ALSA: hda - Don't override maxbps for FLOAT sharing with linear formats When FLOAT PCM format is available but together with other linear PCM formats, don't override maxbps value. For FLOAT format, it's always 32, thus it can be better checked in snd_hda_calc_stream_format(). Otherwise the maxbps 32 might be used wrongly even if the linear PCM doesn't support it. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 263d124de611..fb5760c64e74 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2563,7 +2563,7 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, case 20: case 24: case 32: - if (maxbps >= 32) + if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE) val |= 0x40; else if (maxbps >= 24) val |= 0x30; @@ -2692,7 +2692,8 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, } if (streams & AC_SUPFMT_FLOAT32) { formats |= SNDRV_PCM_FMTBIT_FLOAT_LE; - bps = 32; + if (!bps) + bps = 32; } if (streams == AC_SUPFMT_AC3) { /* should be exclusive */ -- cgit v1.2.3 From 02358fcfa54ce018a0bb56ca9f5a898de574a9d3 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Sat, 4 Jul 2009 01:44:59 -0300 Subject: ALSA: hda - move 8086:fb30 quirk (stac9205) to the proper section Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 14f3c3e0f62d..41b5b3a18c1e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1590,8 +1590,6 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xfb30, - "SigmaTel",STAC_9205_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_REF), /* Dell laptops have BIOS problem */ @@ -2344,6 +2342,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_9205_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xfb30, + "SigmaTel", STAC_9205_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_9205_REF), /* Dell */ -- cgit v1.2.3 From aba6653617754e12763a0d3c9dda332b66190a50 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 5 Jul 2009 11:44:46 +0200 Subject: ALSA: hda - Fix error path in the sanity check in azx_pcm_open() Release resources cleanly after errors in the sanity check in azx_pcm_open(). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1877d95d4aa6..16e09d740572 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1455,6 +1455,17 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) return err; } snd_pcm_limit_hw_rates(runtime); + /* sanity check */ + if (snd_BUG_ON(!runtime->hw.channels_min) || + snd_BUG_ON(!runtime->hw.channels_max) || + snd_BUG_ON(!runtime->hw.formats) || + snd_BUG_ON(!runtime->hw.rates)) { + azx_release_device(azx_dev); + hinfo->ops.close(hinfo, apcm->codec, substream); + snd_hda_power_down(apcm->codec); + mutex_unlock(&chip->open_mutex); + return -EINVAL; + } spin_lock_irqsave(&chip->reg_lock, flags); azx_dev->substream = substream; azx_dev->running = 0; @@ -1463,13 +1474,6 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) runtime->private_data = azx_dev; snd_pcm_set_sync(substream); mutex_unlock(&chip->open_mutex); - - if (snd_BUG_ON(!runtime->hw.channels_min || !runtime->hw.channels_max)) - return -EINVAL; - if (snd_BUG_ON(!runtime->hw.formats)) - return -EINVAL; - if (snd_BUG_ON(!runtime->hw.rates)) - return -EINVAL; return 0; } -- cgit v1.2.3 From 69eb88825a7a562ee3564bdae20c35b0238307b0 Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Sat, 4 Jul 2009 22:25:44 +0200 Subject: cmi8330: Add basic CMI8329 support Add basic support for CMI8329 cards. Makes PCM and OPL3 work. Does not break CMI8330 (tested). Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/isa/cmi8330.c | 71 +++++++++++++++++++++++++++++++---------------------- 1 file changed, 42 insertions(+), 29 deletions(-) diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index d510c76c537f..33e63faf6aa1 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -1,5 +1,5 @@ /* - * Driver for C-Media's CMI8330 soundcards. + * Driver for C-Media's CMI8330 and CMI8329 soundcards. * Copyright (c) by George Talusan * http://www.undergrad.math.uwaterloo.ca/~gstalusa * @@ -64,7 +64,7 @@ /* */ MODULE_AUTHOR("George Talusan "); -MODULE_DESCRIPTION("C-Media CMI8330"); +MODULE_DESCRIPTION("C-Media CMI8330/CMI8329"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8330,isapnp:{CMI0001,@@@0001,@X@0001}}}"); @@ -86,38 +86,38 @@ static long mpuport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; static int mpuirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; module_param_array(index, int, NULL, 0444); -MODULE_PARM_DESC(index, "Index value for CMI8330 soundcard."); +MODULE_PARM_DESC(index, "Index value for CMI8330/CMI8329 soundcard."); module_param_array(id, charp, NULL, 0444); -MODULE_PARM_DESC(id, "ID string for CMI8330 soundcard."); +MODULE_PARM_DESC(id, "ID string for CMI8330/CMI8329 soundcard."); module_param_array(enable, bool, NULL, 0444); -MODULE_PARM_DESC(enable, "Enable CMI8330 soundcard."); +MODULE_PARM_DESC(enable, "Enable CMI8330/CMI8329 soundcard."); #ifdef CONFIG_PNP module_param_array(isapnp, bool, NULL, 0444); MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard."); #endif module_param_array(sbport, long, NULL, 0444); -MODULE_PARM_DESC(sbport, "Port # for CMI8330 SB driver."); +MODULE_PARM_DESC(sbport, "Port # for CMI8330/CMI8329 SB driver."); module_param_array(sbirq, int, NULL, 0444); -MODULE_PARM_DESC(sbirq, "IRQ # for CMI8330 SB driver."); +MODULE_PARM_DESC(sbirq, "IRQ # for CMI8330/CMI8329 SB driver."); module_param_array(sbdma8, int, NULL, 0444); -MODULE_PARM_DESC(sbdma8, "DMA8 for CMI8330 SB driver."); +MODULE_PARM_DESC(sbdma8, "DMA8 for CMI8330/CMI8329 SB driver."); module_param_array(sbdma16, int, NULL, 0444); -MODULE_PARM_DESC(sbdma16, "DMA16 for CMI8330 SB driver."); +MODULE_PARM_DESC(sbdma16, "DMA16 for CMI8330/CMI8329 SB driver."); module_param_array(wssport, long, NULL, 0444); -MODULE_PARM_DESC(wssport, "Port # for CMI8330 WSS driver."); +MODULE_PARM_DESC(wssport, "Port # for CMI8330/CMI8329 WSS driver."); module_param_array(wssirq, int, NULL, 0444); -MODULE_PARM_DESC(wssirq, "IRQ # for CMI8330 WSS driver."); +MODULE_PARM_DESC(wssirq, "IRQ # for CMI8330/CMI8329 WSS driver."); module_param_array(wssdma, int, NULL, 0444); -MODULE_PARM_DESC(wssdma, "DMA for CMI8330 WSS driver."); +MODULE_PARM_DESC(wssdma, "DMA for CMI8330/CMI8329 WSS driver."); module_param_array(fmport, long, NULL, 0444); -MODULE_PARM_DESC(fmport, "FM port # for CMI8330 driver."); +MODULE_PARM_DESC(fmport, "FM port # for CMI8330/CMI8329 driver."); module_param_array(mpuport, long, NULL, 0444); -MODULE_PARM_DESC(mpuport, "MPU-401 port # for CMI8330 driver."); +MODULE_PARM_DESC(mpuport, "MPU-401 port # for CMI8330/CMI8329 driver."); module_param_array(mpuirq, int, NULL, 0444); -MODULE_PARM_DESC(mpuirq, "IRQ # for CMI8330 MPU-401 port."); +MODULE_PARM_DESC(mpuirq, "IRQ # for CMI8330/CMI8329 MPU-401 port."); #ifdef CONFIG_PNP static int isa_registered; static int pnp_registered; @@ -156,6 +156,11 @@ static unsigned char snd_cmi8330_image[((CMI8330_CDINGAIN)-16) + 1] = typedef int (*snd_pcm_open_callback_t)(struct snd_pcm_substream *); +enum card_type { + CMI8330, + CMI8329 +}; + struct snd_cmi8330 { #ifdef CONFIG_PNP struct pnp_dev *cap; @@ -172,11 +177,14 @@ struct snd_cmi8330 { snd_pcm_open_callback_t open; void *private_data; /* sb or wss */ } streams[2]; + + enum card_type type; }; #ifdef CONFIG_PNP static struct pnp_card_device_id snd_cmi8330_pnpids[] = { + { .id = "CMI0001", .devs = { { "@X@0001" }, { "@@@0001" }, { "@H@0001" }, { "A@@0001" } } }, { .id = "CMI0001", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } } }, { .id = "" } }; @@ -304,7 +312,7 @@ static int __devinit snd_cmi8330_mixer(struct snd_card *card, struct snd_cmi8330 unsigned int idx; int err; - strcpy(card->mixername, "CMI8330/C3D"); + strcpy(card->mixername, (acard->type == CMI8329) ? "CMI8329" : "CMI8330/C3D"); for (idx = 0; idx < ARRAY_SIZE(snd_cmi8330_controls); idx++) { err = snd_ctl_add(card, @@ -329,6 +337,9 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, struct pnp_dev *pdev; int err; + /* CMI8329 has a device with ID A@@0001, CMI8330 does not */ + acard->type = (id->devs[3].id[0]) ? CMI8329 : CMI8330; + acard->cap = pnp_request_card_device(card, id->devs[0].id, NULL); if (acard->cap == NULL) return -EBUSY; @@ -345,34 +356,36 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "CMI8330/C3D (AD1848) PnP configure failure\n"); + snd_printk(KERN_ERR "AD1848 PnP configure failure\n"); return -EBUSY; } wssport[dev] = pnp_port_start(pdev, 0); wssdma[dev] = pnp_dma(pdev, 0); wssirq[dev] = pnp_irq(pdev, 0); - fmport[dev] = pnp_port_start(pdev, 1); + if (acard->type == CMI8330) + fmport[dev] = pnp_port_start(pdev, 1); /* allocate SB16 resources */ pdev = acard->play; err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "CMI8330/C3D (SB16) PnP configure failure\n"); + snd_printk(KERN_ERR "SB16 PnP configure failure\n"); return -EBUSY; } sbport[dev] = pnp_port_start(pdev, 0); sbdma8[dev] = pnp_dma(pdev, 0); sbdma16[dev] = pnp_dma(pdev, 1); sbirq[dev] = pnp_irq(pdev, 0); + if (acard->type == CMI8329) + fmport[dev] = pnp_port_start(pdev, 1); /* allocate MPU-401 resources */ pdev = acard->mpu; err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR - "CMI8330/C3D (MPU-401) PnP configure failure\n"); + snd_printk(KERN_ERR "MPU-401 PnP configure failure\n"); return -EBUSY; } mpuport[dev] = pnp_port_start(pdev, 0); @@ -430,9 +443,9 @@ static int __devinit snd_cmi8330_pcm(struct snd_card *card, struct snd_cmi8330 * snd_cmi8330_capture_open }; - if ((err = snd_pcm_new(card, "CMI8330", 0, 1, 1, &pcm)) < 0) + if ((err = snd_pcm_new(card, (chip->type == CMI8329) ? "CMI8329" : "CMI8330", 0, 1, 1, &pcm)) < 0) return err; - strcpy(pcm->name, "CMI8330"); + strcpy(pcm->name, (chip->type == CMI8329) ? "CMI8329" : "CMI8330"); pcm->private_data = chip; /* SB16 */ @@ -527,11 +540,11 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) wssdma[dev], -1, WSS_HW_DETECT, 0, &acard->wss); if (err < 0) { - snd_printk(KERN_ERR PFX "(AD1848) device busy??\n"); + snd_printk(KERN_ERR PFX "AD1848 device busy??\n"); return err; } if (acard->wss->hardware != WSS_HW_CMI8330) { - snd_printk(KERN_ERR PFX "(AD1848) not found during probe\n"); + snd_printk(KERN_ERR PFX "AD1848 not found during probe\n"); return -ENODEV; } @@ -541,11 +554,11 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) sbdma8[dev], sbdma16[dev], SB_HW_AUTO, &acard->sb)) < 0) { - snd_printk(KERN_ERR PFX "(SB16) device busy??\n"); + snd_printk(KERN_ERR PFX "SB16 device busy??\n"); return err; } if (acard->sb->hardware != SB_HW_16) { - snd_printk(KERN_ERR PFX "(SB16) not found during probe\n"); + snd_printk(KERN_ERR PFX "SB16 not found during probe\n"); return err; } @@ -585,8 +598,8 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) mpuport[dev]); } - strcpy(card->driver, "CMI8330/C3D"); - strcpy(card->shortname, "C-Media CMI8330/C3D"); + strcpy(card->driver, (acard->type == CMI8329) ? "CMI8329" : "CMI8330/C3D"); + strcpy(card->shortname, (acard->type == CMI8329) ? "C-Media CMI8329" : "C-Media CMI8330/C3D"); sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d", card->shortname, acard->wss->port, -- cgit v1.2.3 From 3eff8958308ed875a4e845d59a498288f8bbad77 Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Sun, 5 Jul 2009 13:28:48 +0200 Subject: ALSA: azt3328: fix Kconfig entry This driver is about as far from being experimental as it can ever get for an undocumented card, thus create this patch (interestingly it was the only EXPERIMENTAL remaining in the entire Kconfig file). Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 748f6b7d90b7..fb5ee3cc3968 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -135,11 +135,11 @@ config SND_AW2 config SND_AZT3328 - tristate "Aztech AZF3328 / PCI168 (EXPERIMENTAL)" - depends on EXPERIMENTAL + tristate "Aztech AZF3328 / PCI168" select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM + select SND_RAWMIDI help Say Y here to include support for Aztech AZF3328 (PCI168) soundcards. -- cgit v1.2.3 From dfbf9511155d3584b8747c935216077f46eb9a4f Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Sun, 5 Jul 2009 13:55:46 +0200 Subject: ALSA: azt3328: large codec cleanup, add I2S port etc. - fully separate codec I/O port handling, enabling the use of a single function each for all codecs (playback, capture, I2S out) - add a new separate pcm for I2S out port (UNTESTED, no I2S DAC available yet) - switch gameport to low frequency while idle, to try to reduce noise/power - improve snd_azf3328_codec_setdmaa() calculation - minor variable type cleanup (u16, bool etc.) - add some doc updates (help those lost Windows users, debug help, ...) Note that due to the large cleanup aspect of the codec I/O change, I was able to fit everything including all improvements into the same binary size!! (a measly 10 bytes more or so) This should now be the almost last patch to this driver (minus some possible kernel clocksource patch and x86_64 fixes or so). I just felt like taking a break from the usual stuff and wanted to get this driver's structure finished, and it's rather clean now... Tested, working and checkpatch.pl:ed on 2.6.30-rc5, applies cleanly to 2.6.30 proper. Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai --- sound/pci/azt3328.c | 978 ++++++++++++++++++++++++++++------------------------ sound/pci/azt3328.h | 87 +++-- 2 files changed, 564 insertions(+), 501 deletions(-) diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index f290bc56178f..39dfdaa6a56f 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1,6 +1,6 @@ /* * azt3328.c - driver for Aztech AZF3328 based soundcards (e.g. PCI168). - * Copyright (C) 2002, 2005 - 2008 by Andreas Mohr + * Copyright (C) 2002, 2005 - 2009 by Andreas Mohr * * Framework borrowed from Bart Hartgers's als4000.c. * Driver developed on PCI168 AP(W) version (PCI rev. 10, subsystem ID 1801), @@ -10,6 +10,13 @@ * PCI168 A/AP, sub ID 8000 * Please give me feedback in case you try my driver with one of these!! * + * Keywords: Windows XP Vista 168nt4-125.zip 168win95-125.zip PCI 168 download + * (XP/Vista do not support this card at all but every Linux distribution + * has very good support out of the box; + * just to make sure that the right people hit this and get to know that, + * despite the high level of Internet ignorance - as usual :-P - + * about Linux support for this card) + * * GPL LICENSE * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -71,10 +78,11 @@ * - built-in General DirectX timer having a 20 bits counter * with 1us resolution (see below!) * - I2S serial output port for external DAC + * [FIXME: 3.3V or 5V level? maximum rate is 66.2kHz right?] * - supports 33MHz PCI spec 2.1, PCI power management 1.0, compliant with ACPI * - supports hardware volume control * - single chip low cost solution (128 pin QFP) - * - supports programmable Sub-vendor and Sub-system ID + * - supports programmable Sub-vendor and Sub-system ID [24C02 SEEPROM chip] * required for Microsoft's logo compliance (FIXME: where?) * At least the Trident 4D Wave DX has one bit somewhere * to enable writes to PCI subsystem VID registers, that should be it. @@ -82,6 +90,7 @@ * some custom data starting at 0x80. What kind of config settings * are located in our extended PCI space anyway?? * - PCI168 AP(W) card: power amplifier with 4 Watts/channel at 4 Ohms + * [TDA1517P chip] * * Note that this driver now is actually *better* than the Windows driver, * since it additionally supports the card's 1MHz DirectX timer - just try @@ -146,10 +155,15 @@ * to read the Digital Enhanced Game Port. Not sure whether it is fixable. * * TODO + * - use PCI_VDEVICE + * - verify driver status on x86_64 + * - test multi-card driver operation + * - (ab)use 1MHz DirectX timer as kernel clocksource * - test MPU401 MIDI playback etc. * - add more power micro-management (disable various units of the card - * as long as they're unused). However this requires more I/O ports which I - * haven't figured out yet and which thus might not even exist... + * as long as they're unused, to improve audio quality and save power). + * However this requires more I/O ports which I haven't figured out yet + * and which thus might not even exist... * The standard suspend/resume functionality could probably make use of * some improvement, too... * - figure out what all unknown port bits are responsible for @@ -185,6 +199,26 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); #define SUPPORT_GAMEPORT 1 #endif +/* === Debug settings === + Further diagnostic functionality than the settings below + does not need to be provided, since one can easily write a bash script + to dump the card's I/O ports (those listed in lspci -v -v): + function dump() + { + local descr=$1; local addr=$2; local count=$3 + + echo "${descr}: ${count} @ ${addr}:" + dd if=/dev/port skip=$[${addr}] count=${count} bs=1 2>/dev/null| hexdump -C + } + and then use something like + "dump joy200 0x200 8", "dump mpu388 0x388 4", "dump joy 0xb400 8", + "dump codec00 0xa800 32", "dump mixer 0xb800 64", "dump synth 0xbc00 8", + possibly within a "while true; do ... sleep 1; done" loop. + Tweaking ports could be done using + VALSTRING="`printf "%02x" $value`" + printf "\x""$VALSTRING"|dd of=/dev/port seek=$[${addr}] bs=1 2>/dev/null +*/ + #define DEBUG_MISC 0 #define DEBUG_CALLS 0 #define DEBUG_MIXER 0 @@ -250,22 +284,23 @@ static int seqtimer_scaling = 128; module_param(seqtimer_scaling, int, 0444); MODULE_PARM_DESC(seqtimer_scaling, "Set 1024000Hz sequencer timer scale factor (lockup danger!). Default 128."); -struct snd_azf3328_audio_stream { +struct snd_azf3328_codec_data { + unsigned long io_base; struct snd_pcm_substream *substream; - int enabled; - int running; - unsigned long portbase; + bool running; + const char *name; }; -enum snd_azf3328_stream_index { - AZF_PLAYBACK = 0, - AZF_CAPTURE = 1, +enum snd_azf3328_codec_type { + AZF_CODEC_PLAYBACK = 0, + AZF_CODEC_CAPTURE = 1, + AZF_CODEC_I2S_OUT = 2, }; struct snd_azf3328 { /* often-used fields towards beginning, then grouped */ - unsigned long codec_io; /* usually 0xb000, size 128 */ + unsigned long ctrl_io; /* usually 0xb000, size 128 */ unsigned long game_io; /* usually 0xb400, size 8 */ unsigned long mpu_io; /* usually 0xb800, size 4 */ unsigned long opl3_io; /* usually 0xbc00, size 8 */ @@ -275,15 +310,17 @@ struct snd_azf3328 { struct snd_timer *timer; - struct snd_pcm *pcm; - struct snd_azf3328_audio_stream audio_stream[2]; + struct snd_pcm *pcm[3]; + + /* playback, recording and I2S out codecs */ + struct snd_azf3328_codec_data codecs[3]; struct snd_card *card; struct snd_rawmidi *rmidi; #ifdef SUPPORT_GAMEPORT struct gameport *gameport; - int axes[4]; + u16 axes[4]; #endif struct pci_dev *pci; @@ -293,12 +330,12 @@ struct snd_azf3328 { * If we need to add more registers here, then we might try to fold this * into some transparent combined shadow register handling with * CONFIG_PM register storage below, but that's slightly difficult. */ - u16 shadow_reg_codec_6AH; + u16 shadow_reg_ctrl_6AH; #ifdef CONFIG_PM /* register value containers for power management * Note: not always full I/O range preserved (just like Win driver!) */ - u16 saved_regs_codec[AZF_IO_SIZE_CODEC_PM / 2]; + u16 saved_regs_ctrl[AZF_IO_SIZE_CTRL_PM / 2]; u16 saved_regs_game [AZF_IO_SIZE_GAME_PM / 2]; u16 saved_regs_mpu [AZF_IO_SIZE_MPU_PM / 2]; u16 saved_regs_opl3 [AZF_IO_SIZE_OPL3_PM / 2]; @@ -316,7 +353,7 @@ MODULE_DEVICE_TABLE(pci, snd_azf3328_ids); static int -snd_azf3328_io_reg_setb(unsigned reg, u8 mask, int do_set) +snd_azf3328_io_reg_setb(unsigned reg, u8 mask, bool do_set) { u8 prev = inb(reg), new; @@ -331,39 +368,72 @@ snd_azf3328_io_reg_setb(unsigned reg, u8 mask, int do_set) } static inline void -snd_azf3328_codec_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value) +snd_azf3328_codec_outb(const struct snd_azf3328_codec_data *codec, + unsigned reg, + u8 value +) { - outb(value, chip->codec_io + reg); + outb(value, codec->io_base + reg); } static inline u8 -snd_azf3328_codec_inb(const struct snd_azf3328 *chip, unsigned reg) +snd_azf3328_codec_inb(const struct snd_azf3328_codec_data *codec, unsigned reg) { - return inb(chip->codec_io + reg); + return inb(codec->io_base + reg); } static inline void -snd_azf3328_codec_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value) +snd_azf3328_codec_outw(const struct snd_azf3328_codec_data *codec, + unsigned reg, + u16 value +) { - outw(value, chip->codec_io + reg); + outw(value, codec->io_base + reg); } static inline u16 -snd_azf3328_codec_inw(const struct snd_azf3328 *chip, unsigned reg) +snd_azf3328_codec_inw(const struct snd_azf3328_codec_data *codec, unsigned reg) { - return inw(chip->codec_io + reg); + return inw(codec->io_base + reg); } static inline void -snd_azf3328_codec_outl(const struct snd_azf3328 *chip, unsigned reg, u32 value) +snd_azf3328_codec_outl(const struct snd_azf3328_codec_data *codec, + unsigned reg, + u32 value +) { - outl(value, chip->codec_io + reg); + outl(value, codec->io_base + reg); } static inline u32 -snd_azf3328_codec_inl(const struct snd_azf3328 *chip, unsigned reg) +snd_azf3328_codec_inl(const struct snd_azf3328_codec_data *codec, unsigned reg) +{ + return inl(codec->io_base + reg); +} + +static inline void +snd_azf3328_ctrl_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value) { - return inl(chip->codec_io + reg); + outb(value, chip->ctrl_io + reg); +} + +static inline u8 +snd_azf3328_ctrl_inb(const struct snd_azf3328 *chip, unsigned reg) +{ + return inb(chip->ctrl_io + reg); +} + +static inline void +snd_azf3328_ctrl_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value) +{ + outw(value, chip->ctrl_io + reg); +} + +static inline void +snd_azf3328_ctrl_outl(const struct snd_azf3328 *chip, unsigned reg, u32 value) +{ + outl(value, chip->ctrl_io + reg); } static inline void @@ -404,13 +474,13 @@ snd_azf3328_mixer_inw(const struct snd_azf3328 *chip, unsigned reg) #define AZF_MUTE_BIT 0x80 -static int +static bool snd_azf3328_mixer_set_mute(const struct snd_azf3328 *chip, - unsigned reg, int do_mute + unsigned reg, bool do_mute ) { unsigned long portbase = chip->mixer_io + reg + 1; - int updated; + bool updated; /* the mute bit is on the *second* (i.e. right) register of a * left/right channel setting */ @@ -569,7 +639,7 @@ snd_azf3328_get_mixer(struct snd_kcontrol *kcontrol, { struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol); struct azf3328_mixer_reg reg; - unsigned int oreg, val; + u16 oreg, val; snd_azf3328_dbgcallenter(); snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); @@ -600,7 +670,7 @@ snd_azf3328_put_mixer(struct snd_kcontrol *kcontrol, { struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol); struct azf3328_mixer_reg reg; - unsigned int oreg, nreg, val; + u16 oreg, nreg, val; snd_azf3328_dbgcallenter(); snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); @@ -709,7 +779,7 @@ snd_azf3328_put_mixer_enum(struct snd_kcontrol *kcontrol, { struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol); struct azf3328_mixer_reg reg; - unsigned int oreg, nreg, val; + u16 oreg, nreg, val; snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); oreg = snd_azf3328_mixer_inw(chip, reg.reg); @@ -867,14 +937,15 @@ snd_azf3328_hw_free(struct snd_pcm_substream *substream) static void snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, - unsigned reg, + enum snd_azf3328_codec_type codec_type, enum azf_freq_t bitrate, unsigned int format_width, unsigned int channels ) { - u16 val = 0xff00; unsigned long flags; + const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; + u16 val = 0xff00; snd_azf3328_dbgcallenter(); switch (bitrate) { @@ -917,7 +988,7 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, spin_lock_irqsave(&chip->reg_lock, flags); /* set bitrate/format */ - snd_azf3328_codec_outw(chip, reg, val); + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_SOUNDFORMAT, val); /* changing the bitrate/format settings switches off the * audio output with an annoying click in case of 8/16bit format change @@ -926,11 +997,11 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, * (FIXME: yes, it works, but what exactly am I doing here?? :) * FIXME: does this have some side effects for full-duplex * or other dramatic side effects? */ - if (reg == IDX_IO_PLAY_SOUNDFORMAT) /* only do it for playback */ - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) | - DMA_PLAY_SOMETHING1 | - DMA_PLAY_SOMETHING2 | + if (codec_type == AZF_CODEC_PLAYBACK) /* only do it for playback */ + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, + snd_azf3328_codec_inw(codec, IDX_IO_CODEC_DMA_FLAGS) | + DMA_RUN_SOMETHING1 | + DMA_RUN_SOMETHING2 | SOMETHING_ALMOST_ALWAYS_SET | DMA_EPILOGUE_SOMETHING | DMA_SOMETHING_ELSE @@ -942,134 +1013,132 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, static inline void snd_azf3328_codec_setfmt_lowpower(struct snd_azf3328 *chip, - unsigned reg + enum snd_azf3328_codec_type codec_type ) { /* choose lowest frequency for low power consumption. * While this will cause louder noise due to rather coarse frequency, * it should never matter since output should always * get disabled properly when idle anyway. */ - snd_azf3328_codec_setfmt(chip, reg, AZF_FREQ_4000, 8, 1); + snd_azf3328_codec_setfmt(chip, codec_type, AZF_FREQ_4000, 8, 1); } static void -snd_azf3328_codec_reg_6AH_update(struct snd_azf3328 *chip, +snd_azf3328_ctrl_reg_6AH_update(struct snd_azf3328 *chip, unsigned bitmask, - int enable + bool enable ) { if (enable) - chip->shadow_reg_codec_6AH &= ~bitmask; + chip->shadow_reg_ctrl_6AH &= ~bitmask; else - chip->shadow_reg_codec_6AH |= bitmask; + chip->shadow_reg_ctrl_6AH |= bitmask; snd_azf3328_dbgplay("6AH_update mask 0x%04x enable %d: val 0x%04x\n", - bitmask, enable, chip->shadow_reg_codec_6AH); - snd_azf3328_codec_outw(chip, IDX_IO_6AH, chip->shadow_reg_codec_6AH); + bitmask, enable, chip->shadow_reg_ctrl_6AH); + snd_azf3328_ctrl_outw(chip, IDX_IO_6AH, chip->shadow_reg_ctrl_6AH); } static inline void -snd_azf3328_codec_enable(struct snd_azf3328 *chip, int enable) +snd_azf3328_ctrl_enable_codecs(struct snd_azf3328 *chip, bool enable) { snd_azf3328_dbgplay("codec_enable %d\n", enable); /* no idea what exactly is being done here, but I strongly assume it's * PM related */ - snd_azf3328_codec_reg_6AH_update( + snd_azf3328_ctrl_reg_6AH_update( chip, IO_6A_PAUSE_PLAYBACK_BIT8, enable ); } static void -snd_azf3328_codec_activity(struct snd_azf3328 *chip, - enum snd_azf3328_stream_index stream_type, - int enable +snd_azf3328_ctrl_codec_activity(struct snd_azf3328 *chip, + enum snd_azf3328_codec_type codec_type, + bool enable ) { - int need_change = (chip->audio_stream[stream_type].running != enable); + struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; + bool need_change = (codec->running != enable); snd_azf3328_dbgplay( - "codec_activity: type %d, enable %d, need_change %d\n", - stream_type, enable, need_change + "codec_activity: %s codec, enable %d, need_change %d\n", + codec->name, enable, need_change ); if (need_change) { - enum snd_azf3328_stream_index other = - (stream_type == AZF_PLAYBACK) ? - AZF_CAPTURE : AZF_PLAYBACK; - /* small check to prevent shutting down the other party - * in case it's active */ - if ((enable) || !(chip->audio_stream[other].running)) - snd_azf3328_codec_enable(chip, enable); + static const struct { + enum snd_azf3328_codec_type other1; + enum snd_azf3328_codec_type other2; + } peer_codecs[3] = + { { AZF_CODEC_CAPTURE, AZF_CODEC_I2S_OUT }, + { AZF_CODEC_PLAYBACK, AZF_CODEC_I2S_OUT }, + { AZF_CODEC_PLAYBACK, AZF_CODEC_CAPTURE } }; + bool call_function; + + if (enable) + /* if enable codec, call enable_codecs func + to enable codec supply... */ + call_function = 1; + else { + /* ...otherwise call enable_codecs func + (which globally shuts down operation of codecs) + only in case the other codecs are currently + not active either! */ + if ((!chip->codecs[peer_codecs[codec_type].other1] + .running) + && (!chip->codecs[peer_codecs[codec_type].other2] + .running)) + call_function = 1; + } + if (call_function) + snd_azf3328_ctrl_enable_codecs(chip, enable); /* ...and adjust clock, too * (reduce noise and power consumption) */ if (!enable) snd_azf3328_codec_setfmt_lowpower( chip, - chip->audio_stream[stream_type].portbase - + IDX_IO_PLAY_SOUNDFORMAT + codec_type ); } - chip->audio_stream[stream_type].running = enable; + codec->running = enable; } static void -snd_azf3328_setdmaa(struct snd_azf3328 *chip, - long unsigned int addr, - unsigned int count, - unsigned int size, - enum snd_azf3328_stream_index stream_type +snd_azf3328_codec_setdmaa(struct snd_azf3328 *chip, + enum snd_azf3328_codec_type codec_type, + unsigned long addr, + unsigned int count, + unsigned int size ) { + const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; snd_azf3328_dbgcallenter(); - if (!chip->audio_stream[stream_type].running) { - /* AZF3328 uses a two buffer pointer DMA playback approach */ + if (!codec->running) { + /* AZF3328 uses a two buffer pointer DMA transfer approach */ - unsigned long flags, portbase, addr_area2; + unsigned long flags; /* width 32bit (prevent overflow): */ - unsigned long count_areas, count_tmp; + u32 addr_area2, count_areas, lengths; - portbase = chip->audio_stream[stream_type].portbase; count_areas = size/2; addr_area2 = addr+count_areas; count_areas--; /* max. index */ snd_azf3328_dbgplay("set DMA: buf1 %08lx[%lu], buf2 %08lx[%lu]\n", addr, count_areas, addr_area2, count_areas); /* build combined I/O buffer length word */ - count_tmp = count_areas; - count_areas |= (count_tmp << 16); + lengths = (count_areas << 16) | (count_areas); spin_lock_irqsave(&chip->reg_lock, flags); - outl(addr, portbase + IDX_IO_PLAY_DMA_START_1); - outl(addr_area2, portbase + IDX_IO_PLAY_DMA_START_2); - outl(count_areas, portbase + IDX_IO_PLAY_DMA_LEN_1); + snd_azf3328_codec_outl(codec, IDX_IO_CODEC_DMA_START_1, addr); + snd_azf3328_codec_outl(codec, IDX_IO_CODEC_DMA_START_2, + addr_area2); + snd_azf3328_codec_outl(codec, IDX_IO_CODEC_DMA_LENGTHS, + lengths); spin_unlock_irqrestore(&chip->reg_lock, flags); } snd_azf3328_dbgcallleave(); } static int -snd_azf3328_playback_prepare(struct snd_pcm_substream *substream) -{ -#if 0 - struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int size = snd_pcm_lib_buffer_bytes(substream); - unsigned int count = snd_pcm_lib_period_bytes(substream); -#endif - - snd_azf3328_dbgcallenter(); -#if 0 - snd_azf3328_codec_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, - runtime->rate, - snd_pcm_format_width(runtime->format), - runtime->channels); - snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, AZF_PLAYBACK); -#endif - snd_azf3328_dbgcallleave(); - return 0; -} - -static int -snd_azf3328_capture_prepare(struct snd_pcm_substream *substream) +snd_azf3328_codec_prepare(struct snd_pcm_substream *substream) { #if 0 struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); @@ -1080,135 +1149,161 @@ snd_azf3328_capture_prepare(struct snd_pcm_substream *substream) snd_azf3328_dbgcallenter(); #if 0 - snd_azf3328_codec_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, AZF_CODEC_..., runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); - snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, AZF_CAPTURE); + snd_azf3328_codec_setdmaa(chip, AZF_CODEC_..., + runtime->dma_addr, count, size); #endif snd_azf3328_dbgcallleave(); return 0; } static int -snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) +snd_azf3328_codec_trigger(enum snd_azf3328_codec_type codec_type, + struct snd_pcm_substream *substream, int cmd) { struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); + const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; struct snd_pcm_runtime *runtime = substream->runtime; int result = 0; - unsigned int status1; - int previously_muted; + u16 flags1; + bool previously_muted = 0; + bool is_playback_codec = (AZF_CODEC_PLAYBACK == codec_type); - snd_azf3328_dbgcalls("snd_azf3328_playback_trigger cmd %d\n", cmd); + snd_azf3328_dbgcalls("snd_azf3328_codec_trigger cmd %d\n", cmd); switch (cmd) { case SNDRV_PCM_TRIGGER_START: - snd_azf3328_dbgplay("START PLAYBACK\n"); - - /* mute WaveOut (avoid clicking during setup) */ - previously_muted = - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); + snd_azf3328_dbgplay("START %s\n", codec->name); + + if (is_playback_codec) { + /* mute WaveOut (avoid clicking during setup) */ + previously_muted = + snd_azf3328_mixer_set_mute( + chip, IDX_MIXER_WAVEOUT, 1 + ); + } - snd_azf3328_codec_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, codec_type, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); spin_lock(&chip->reg_lock); /* first, remember current value: */ - status1 = snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS); + flags1 = snd_azf3328_codec_inw(codec, IDX_IO_CODEC_DMA_FLAGS); - /* stop playback */ - status1 &= ~DMA_RESUME; - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); + /* stop transfer */ + flags1 &= ~DMA_RESUME; + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1); /* FIXME: clear interrupts or what??? */ - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_IRQTYPE, 0xffff); + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_IRQTYPE, 0xffff); spin_unlock(&chip->reg_lock); - snd_azf3328_setdmaa(chip, runtime->dma_addr, + snd_azf3328_codec_setdmaa(chip, codec_type, runtime->dma_addr, snd_pcm_lib_period_bytes(substream), - snd_pcm_lib_buffer_bytes(substream), - AZF_PLAYBACK); + snd_pcm_lib_buffer_bytes(substream) + ); spin_lock(&chip->reg_lock); #ifdef WIN9X /* FIXME: enable playback/recording??? */ - status1 |= DMA_PLAY_SOMETHING1 | DMA_PLAY_SOMETHING2; - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); + flags1 |= DMA_RUN_SOMETHING1 | DMA_RUN_SOMETHING2; + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1); - /* start playback again */ + /* start transfer again */ /* FIXME: what is this value (0x0010)??? */ - status1 |= DMA_RESUME | DMA_EPILOGUE_SOMETHING; - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); + flags1 |= DMA_RESUME | DMA_EPILOGUE_SOMETHING; + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1); #else /* NT4 */ - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, 0x0000); - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, - DMA_PLAY_SOMETHING1); - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, - DMA_PLAY_SOMETHING1 | - DMA_PLAY_SOMETHING2); - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, + DMA_RUN_SOMETHING1); + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, + DMA_RUN_SOMETHING1 | + DMA_RUN_SOMETHING2); + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, DMA_RESUME | SOMETHING_ALMOST_ALWAYS_SET | DMA_EPILOGUE_SOMETHING | DMA_SOMETHING_ELSE); #endif spin_unlock(&chip->reg_lock); - snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 1); - - /* now unmute WaveOut */ - if (!previously_muted) - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); + snd_azf3328_ctrl_codec_activity(chip, codec_type, 1); + + if (is_playback_codec) { + /* now unmute WaveOut */ + if (!previously_muted) + snd_azf3328_mixer_set_mute( + chip, IDX_MIXER_WAVEOUT, 0 + ); + } - snd_azf3328_dbgplay("STARTED PLAYBACK\n"); + snd_azf3328_dbgplay("STARTED %s\n", codec->name); break; case SNDRV_PCM_TRIGGER_RESUME: - snd_azf3328_dbgplay("RESUME PLAYBACK\n"); - /* resume playback if we were active */ + snd_azf3328_dbgplay("RESUME %s\n", codec->name); + /* resume codec if we were active */ spin_lock(&chip->reg_lock); - if (chip->audio_stream[AZF_PLAYBACK].running) - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) | DMA_RESUME); + if (codec->running) + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, + snd_azf3328_codec_inw( + codec, IDX_IO_CODEC_DMA_FLAGS + ) | DMA_RESUME + ); spin_unlock(&chip->reg_lock); break; case SNDRV_PCM_TRIGGER_STOP: - snd_azf3328_dbgplay("STOP PLAYBACK\n"); - - /* mute WaveOut (avoid clicking during setup) */ - previously_muted = - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); + snd_azf3328_dbgplay("STOP %s\n", codec->name); + + if (is_playback_codec) { + /* mute WaveOut (avoid clicking during setup) */ + previously_muted = + snd_azf3328_mixer_set_mute( + chip, IDX_MIXER_WAVEOUT, 1 + ); + } spin_lock(&chip->reg_lock); /* first, remember current value: */ - status1 = snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS); + flags1 = snd_azf3328_codec_inw(codec, IDX_IO_CODEC_DMA_FLAGS); - /* stop playback */ - status1 &= ~DMA_RESUME; - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); + /* stop transfer */ + flags1 &= ~DMA_RESUME; + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1); /* hmm, is this really required? we're resetting the same bit * immediately thereafter... */ - status1 |= DMA_PLAY_SOMETHING1; - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); + flags1 |= DMA_RUN_SOMETHING1; + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1); - status1 &= ~DMA_PLAY_SOMETHING1; - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); + flags1 &= ~DMA_RUN_SOMETHING1; + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1); spin_unlock(&chip->reg_lock); - snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 0); - - /* now unmute WaveOut */ - if (!previously_muted) - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); + snd_azf3328_ctrl_codec_activity(chip, codec_type, 0); + + if (is_playback_codec) { + /* now unmute WaveOut */ + if (!previously_muted) + snd_azf3328_mixer_set_mute( + chip, IDX_MIXER_WAVEOUT, 0 + ); + } - snd_azf3328_dbgplay("STOPPED PLAYBACK\n"); + snd_azf3328_dbgplay("STOPPED %s\n", codec->name); break; case SNDRV_PCM_TRIGGER_SUSPEND: - snd_azf3328_dbgplay("SUSPEND PLAYBACK\n"); - /* make sure playback is stopped */ - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) & ~DMA_RESUME); + snd_azf3328_dbgplay("SUSPEND %s\n", codec->name); + /* make sure codec is stopped */ + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, + snd_azf3328_codec_inw( + codec, IDX_IO_CODEC_DMA_FLAGS + ) & ~DMA_RESUME + ); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_PUSH NIY!\n"); @@ -1225,172 +1320,74 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) return result; } -/* this is just analogous to playback; I'm not quite sure whether recording - * should actually be triggered like that */ static int -snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) +snd_azf3328_codec_playback_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int result = 0; - unsigned int status1; - - snd_azf3328_dbgcalls("snd_azf3328_capture_trigger cmd %d\n", cmd); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - - snd_azf3328_dbgplay("START CAPTURE\n"); - - snd_azf3328_codec_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, - runtime->rate, - snd_pcm_format_width(runtime->format), - runtime->channels); - - spin_lock(&chip->reg_lock); - /* first, remember current value: */ - status1 = snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS); - - /* stop recording */ - status1 &= ~DMA_RESUME; - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); - - /* FIXME: clear interrupts or what??? */ - snd_azf3328_codec_outw(chip, IDX_IO_REC_IRQTYPE, 0xffff); - spin_unlock(&chip->reg_lock); - - snd_azf3328_setdmaa(chip, runtime->dma_addr, - snd_pcm_lib_period_bytes(substream), - snd_pcm_lib_buffer_bytes(substream), - AZF_CAPTURE); - - spin_lock(&chip->reg_lock); -#ifdef WIN9X - /* FIXME: enable playback/recording??? */ - status1 |= DMA_PLAY_SOMETHING1 | DMA_PLAY_SOMETHING2; - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); - - /* start capture again */ - /* FIXME: what is this value (0x0010)??? */ - status1 |= DMA_RESUME | DMA_EPILOGUE_SOMETHING; - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); -#else - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, - 0x0000); - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, - DMA_PLAY_SOMETHING1); - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, - DMA_PLAY_SOMETHING1 | - DMA_PLAY_SOMETHING2); - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, - DMA_RESUME | - SOMETHING_ALMOST_ALWAYS_SET | - DMA_EPILOGUE_SOMETHING | - DMA_SOMETHING_ELSE); -#endif - spin_unlock(&chip->reg_lock); - snd_azf3328_codec_activity(chip, AZF_CAPTURE, 1); - - snd_azf3328_dbgplay("STARTED CAPTURE\n"); - break; - case SNDRV_PCM_TRIGGER_RESUME: - snd_azf3328_dbgplay("RESUME CAPTURE\n"); - /* resume recording if we were active */ - spin_lock(&chip->reg_lock); - if (chip->audio_stream[AZF_CAPTURE].running) - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, - snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS) | DMA_RESUME); - spin_unlock(&chip->reg_lock); - break; - case SNDRV_PCM_TRIGGER_STOP: - snd_azf3328_dbgplay("STOP CAPTURE\n"); - - spin_lock(&chip->reg_lock); - /* first, remember current value: */ - status1 = snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS); - - /* stop recording */ - status1 &= ~DMA_RESUME; - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); - - status1 |= DMA_PLAY_SOMETHING1; - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); - - status1 &= ~DMA_PLAY_SOMETHING1; - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); - spin_unlock(&chip->reg_lock); - snd_azf3328_codec_activity(chip, AZF_CAPTURE, 0); + return snd_azf3328_codec_trigger(AZF_CODEC_PLAYBACK, substream, cmd); +} - snd_azf3328_dbgplay("STOPPED CAPTURE\n"); - break; - case SNDRV_PCM_TRIGGER_SUSPEND: - snd_azf3328_dbgplay("SUSPEND CAPTURE\n"); - /* make sure recording is stopped */ - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, - snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS) & ~DMA_RESUME); - break; - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_PUSH NIY!\n"); - break; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_RELEASE NIY!\n"); - break; - default: - printk(KERN_ERR "FIXME: unknown trigger mode!\n"); - return -EINVAL; - } +static int +snd_azf3328_codec_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + return snd_azf3328_codec_trigger(AZF_CODEC_CAPTURE, substream, cmd); +} - snd_azf3328_dbgcallleave(); - return result; +static int +snd_azf3328_codec_i2s_out_trigger(struct snd_pcm_substream *substream, int cmd) +{ + return snd_azf3328_codec_trigger(AZF_CODEC_I2S_OUT, substream, cmd); } static snd_pcm_uframes_t -snd_azf3328_playback_pointer(struct snd_pcm_substream *substream) +snd_azf3328_codec_pointer(struct snd_pcm_substream *substream, + enum snd_azf3328_codec_type codec_type +) { - struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); + const struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); + const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; unsigned long bufptr, result; snd_pcm_uframes_t frmres; #ifdef QUERY_HARDWARE - bufptr = snd_azf3328_codec_inl(chip, IDX_IO_PLAY_DMA_START_1); + bufptr = snd_azf3328_codec_inl(codec, IDX_IO_CODEC_DMA_START_1); #else bufptr = substream->runtime->dma_addr; #endif - result = snd_azf3328_codec_inl(chip, IDX_IO_PLAY_DMA_CURRPOS); + result = snd_azf3328_codec_inl(codec, IDX_IO_CODEC_DMA_CURRPOS); /* calculate offset */ result -= bufptr; frmres = bytes_to_frames( substream->runtime, result); - snd_azf3328_dbgplay("PLAY @ 0x%8lx, frames %8ld\n", result, frmres); + snd_azf3328_dbgplay("%s @ 0x%8lx, frames %8ld\n", + codec->name, result, frmres); return frmres; } static snd_pcm_uframes_t -snd_azf3328_capture_pointer(struct snd_pcm_substream *substream) +snd_azf3328_codec_playback_pointer(struct snd_pcm_substream *substream) { - struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); - unsigned long bufptr, result; - snd_pcm_uframes_t frmres; + return snd_azf3328_codec_pointer(substream, AZF_CODEC_PLAYBACK); +} -#ifdef QUERY_HARDWARE - bufptr = snd_azf3328_codec_inl(chip, IDX_IO_REC_DMA_START_1); -#else - bufptr = substream->runtime->dma_addr; -#endif - result = snd_azf3328_codec_inl(chip, IDX_IO_REC_DMA_CURRPOS); +static snd_pcm_uframes_t +snd_azf3328_codec_capture_pointer(struct snd_pcm_substream *substream) +{ + return snd_azf3328_codec_pointer(substream, AZF_CODEC_CAPTURE); +} - /* calculate offset */ - result -= bufptr; - frmres = bytes_to_frames( substream->runtime, result); - snd_azf3328_dbgplay("REC @ 0x%8lx, frames %8ld\n", result, frmres); - return frmres; +static snd_pcm_uframes_t +snd_azf3328_codec_i2s_out_pointer(struct snd_pcm_substream *substream) +{ + return snd_azf3328_codec_pointer(substream, AZF_CODEC_I2S_OUT); } /******************************************************************/ #ifdef SUPPORT_GAMEPORT static inline void -snd_azf3328_gameport_irq_enable(struct snd_azf3328 *chip, int enable) +snd_azf3328_gameport_irq_enable(struct snd_azf3328 *chip, + bool enable +) { snd_azf3328_io_reg_setb( chip->game_io+IDX_GAME_HWCONFIG, @@ -1400,7 +1397,9 @@ snd_azf3328_gameport_irq_enable(struct snd_azf3328 *chip, int enable) } static inline void -snd_azf3328_gameport_legacy_address_enable(struct snd_azf3328 *chip, int enable) +snd_azf3328_gameport_legacy_address_enable(struct snd_azf3328 *chip, + bool enable +) { snd_azf3328_io_reg_setb( chip->game_io+IDX_GAME_HWCONFIG, @@ -1409,10 +1408,27 @@ snd_azf3328_gameport_legacy_address_enable(struct snd_azf3328 *chip, int enable) ); } +static void +snd_azf3328_gameport_set_counter_frequency(struct snd_azf3328 *chip, + unsigned int freq_cfg +) +{ + snd_azf3328_io_reg_setb( + chip->game_io+IDX_GAME_HWCONFIG, + 0x02, + (freq_cfg & 1) != 0 + ); + snd_azf3328_io_reg_setb( + chip->game_io+IDX_GAME_HWCONFIG, + 0x04, + (freq_cfg & 2) != 0 + ); +} + static inline void -snd_azf3328_gameport_axis_circuit_enable(struct snd_azf3328 *chip, int enable) +snd_azf3328_gameport_axis_circuit_enable(struct snd_azf3328 *chip, bool enable) { - snd_azf3328_codec_reg_6AH_update( + snd_azf3328_ctrl_reg_6AH_update( chip, IO_6A_SOMETHING2_GAMEPORT, enable ); } @@ -1447,6 +1463,8 @@ snd_azf3328_gameport_open(struct gameport *gameport, int mode) break; } + snd_azf3328_gameport_set_counter_frequency(chip, + GAME_HWCFG_ADC_COUNTER_FREQ_STD); snd_azf3328_gameport_axis_circuit_enable(chip, (res == 0)); return res; @@ -1458,6 +1476,8 @@ snd_azf3328_gameport_close(struct gameport *gameport) struct snd_azf3328 *chip = gameport_get_port_data(gameport); snd_azf3328_dbggame("gameport_close\n"); + snd_azf3328_gameport_set_counter_frequency(chip, + GAME_HWCFG_ADC_COUNTER_FREQ_1_200); snd_azf3328_gameport_axis_circuit_enable(chip, 0); } @@ -1491,7 +1511,7 @@ snd_azf3328_gameport_cooked_read(struct gameport *gameport, val = snd_azf3328_game_inb(chip, IDX_GAME_AXES_CONFIG); if (val & GAME_AXES_SAMPLING_READY) { - for (i = 0; i < 4; ++i) { + for (i = 0; i < ARRAY_SIZE(chip->axes); ++i) { /* configure the axis to read */ val = (i << 4) | 0x0f; snd_azf3328_game_outb(chip, IDX_GAME_AXES_CONFIG, val); @@ -1514,7 +1534,7 @@ snd_azf3328_gameport_cooked_read(struct gameport *gameport, snd_azf3328_game_outw(chip, IDX_GAME_AXIS_VALUE, 0xffff); spin_unlock_irqrestore(&chip->reg_lock, flags); - for (i = 0; i < 4; i++) { + for (i = 0; i < ARRAY_SIZE(chip->axes); i++) { axes[i] = chip->axes[i]; if (axes[i] == 0xffff) axes[i] = -1; @@ -1552,6 +1572,8 @@ snd_azf3328_gameport(struct snd_azf3328 *chip, int dev) /* DISABLE legacy address: we don't need it! */ snd_azf3328_gameport_legacy_address_enable(chip, 0); + snd_azf3328_gameport_set_counter_frequency(chip, + GAME_HWCFG_ADC_COUNTER_FREQ_1_200); snd_azf3328_gameport_axis_circuit_enable(chip, 0); gameport_register_port(chip->gameport); @@ -1591,29 +1613,69 @@ snd_azf3328_irq_log_unknown_type(u8 which) ); } +static inline void +snd_azf3328_codec_interrupt(struct snd_azf3328 *chip, u8 status) +{ + u8 which; + enum snd_azf3328_codec_type codec_type; + const struct snd_azf3328_codec_data *codec; + + for (codec_type = AZF_CODEC_PLAYBACK; + codec_type <= AZF_CODEC_I2S_OUT; + ++codec_type) { + + /* skip codec if there's no interrupt for it */ + if (!(status & (1 << codec_type))) + continue; + + codec = &chip->codecs[codec_type]; + + spin_lock(&chip->reg_lock); + which = snd_azf3328_codec_inb(codec, IDX_IO_CODEC_IRQTYPE); + /* ack all IRQ types immediately */ + snd_azf3328_codec_outb(codec, IDX_IO_CODEC_IRQTYPE, which); + spin_unlock(&chip->reg_lock); + + if ((chip->pcm[codec_type]) + && (chip->codecs[codec_type].substream)) { + snd_pcm_period_elapsed( + chip->codecs[codec_type].substream + ); + snd_azf3328_dbgplay("%s period done (#%x), @ %x\n", + codec->name, + which, + snd_azf3328_codec_inl( + codec, IDX_IO_CODEC_DMA_CURRPOS + ) + ); + } else + printk(KERN_WARNING "azt3328: irq handler problem!\n"); + if (which & IRQ_SOMETHING) + snd_azf3328_irq_log_unknown_type(which); + } +} + static irqreturn_t snd_azf3328_interrupt(int irq, void *dev_id) { struct snd_azf3328 *chip = dev_id; - u8 status, which; + u8 status; #if DEBUG_PLAY_REC static unsigned long irq_count; #endif - status = snd_azf3328_codec_inb(chip, IDX_IO_IRQSTATUS); + status = snd_azf3328_ctrl_inb(chip, IDX_IO_IRQSTATUS); /* fast path out, to ease interrupt sharing */ if (!(status & - (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_GAMEPORT|IRQ_MPU401|IRQ_TIMER) + (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_I2S_OUT + |IRQ_GAMEPORT|IRQ_MPU401|IRQ_TIMER) )) return IRQ_NONE; /* must be interrupt for another device */ snd_azf3328_dbgplay( - "irq_count %ld! IDX_IO_PLAY_FLAGS %04x, " - "IDX_IO_PLAY_IRQTYPE %04x, IDX_IO_IRQSTATUS %04x\n", + "irq_count %ld! IDX_IO_IRQSTATUS %04x\n", irq_count++ /* debug-only */, - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS), - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE), status ); @@ -1626,63 +1688,24 @@ snd_azf3328_interrupt(int irq, void *dev_id) snd_timer_interrupt(chip->timer, chip->timer->sticks); /* ACK timer */ spin_lock(&chip->reg_lock); - snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x07); + snd_azf3328_ctrl_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x07); spin_unlock(&chip->reg_lock); snd_azf3328_dbgplay("azt3328: timer IRQ\n"); } - if (status & IRQ_PLAYBACK) { - spin_lock(&chip->reg_lock); - which = snd_azf3328_codec_inb(chip, IDX_IO_PLAY_IRQTYPE); - /* ack all IRQ types immediately */ - snd_azf3328_codec_outb(chip, IDX_IO_PLAY_IRQTYPE, which); - spin_unlock(&chip->reg_lock); - if (chip->pcm && chip->audio_stream[AZF_PLAYBACK].substream) { - snd_pcm_period_elapsed( - chip->audio_stream[AZF_PLAYBACK].substream - ); - snd_azf3328_dbgplay("PLAY period done (#%x), @ %x\n", - which, - snd_azf3328_codec_inl( - chip, IDX_IO_PLAY_DMA_CURRPOS - ) - ); - } else - printk(KERN_WARNING "azt3328: irq handler problem!\n"); - if (which & IRQ_PLAY_SOMETHING) - snd_azf3328_irq_log_unknown_type(which); - } - if (status & IRQ_RECORDING) { - spin_lock(&chip->reg_lock); - which = snd_azf3328_codec_inb(chip, IDX_IO_REC_IRQTYPE); - /* ack all IRQ types immediately */ - snd_azf3328_codec_outb(chip, IDX_IO_REC_IRQTYPE, which); - spin_unlock(&chip->reg_lock); + if (status & (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_I2S_OUT)) + snd_azf3328_codec_interrupt(chip, status); - if (chip->pcm && chip->audio_stream[AZF_CAPTURE].substream) { - snd_pcm_period_elapsed( - chip->audio_stream[AZF_CAPTURE].substream - ); - snd_azf3328_dbgplay("REC period done (#%x), @ %x\n", - which, - snd_azf3328_codec_inl( - chip, IDX_IO_REC_DMA_CURRPOS - ) - ); - } else - printk(KERN_WARNING "azt3328: irq handler problem!\n"); - if (which & IRQ_REC_SOMETHING) - snd_azf3328_irq_log_unknown_type(which); - } if (status & IRQ_GAMEPORT) snd_azf3328_gameport_interrupt(chip); + /* MPU401 has less critical IRQ requirements * than timer and playback/recording, right? */ if (status & IRQ_MPU401) { snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data); /* hmm, do we have to ack the IRQ here somehow? - * If so, then I don't know how... */ + * If so, then I don't know how yet... */ snd_azf3328_dbgplay("azt3328: MPU401 IRQ\n"); } return IRQ_HANDLED; @@ -1690,7 +1713,11 @@ snd_azf3328_interrupt(int irq, void *dev_id) /*****************************************************************/ -static const struct snd_pcm_hardware snd_azf3328_playback = +/* as long as we think we have identical snd_pcm_hardware parameters + for playback, capture and i2s out, we can use the same physical struct + since the struct is simply being copied into a member. +*/ +static const struct snd_pcm_hardware snd_azf3328_hardware = { /* FIXME!! Correct? */ .info = SNDRV_PCM_INFO_MMAP | @@ -1718,31 +1745,6 @@ static const struct snd_pcm_hardware snd_azf3328_playback = .fifo_size = 0, }; -static const struct snd_pcm_hardware snd_azf3328_capture = -{ - /* FIXME */ - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID, - .formats = SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_U16_LE, - .rates = SNDRV_PCM_RATE_5512 | - SNDRV_PCM_RATE_8000_48000 | - SNDRV_PCM_RATE_KNOT, - .rate_min = AZF_FREQ_4000, - .rate_max = AZF_FREQ_66200, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = 65536, - .period_bytes_min = 64, - .period_bytes_max = 65536, - .periods_min = 1, - .periods_max = 1024, - .fifo_size = 0, -}; - static unsigned int snd_azf3328_fixed_rates[] = { AZF_FREQ_4000, @@ -1770,55 +1772,72 @@ static struct snd_pcm_hw_constraint_list snd_azf3328_hw_constraints_rates = { /*****************************************************************/ static int -snd_azf3328_playback_open(struct snd_pcm_substream *substream) +snd_azf3328_pcm_open(struct snd_pcm_substream *substream, + enum snd_azf3328_codec_type codec_type +) { struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; snd_azf3328_dbgcallenter(); - chip->audio_stream[AZF_PLAYBACK].substream = substream; - runtime->hw = snd_azf3328_playback; + chip->codecs[codec_type].substream = substream; + + /* same parameters for all our codecs - at least we think so... */ + runtime->hw = snd_azf3328_hardware; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_azf3328_hw_constraints_rates); snd_azf3328_dbgcallleave(); return 0; } +static int +snd_azf3328_playback_open(struct snd_pcm_substream *substream) +{ + return snd_azf3328_pcm_open(substream, AZF_CODEC_PLAYBACK); +} + static int snd_azf3328_capture_open(struct snd_pcm_substream *substream) { - struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; + return snd_azf3328_pcm_open(substream, AZF_CODEC_CAPTURE); +} - snd_azf3328_dbgcallenter(); - chip->audio_stream[AZF_CAPTURE].substream = substream; - runtime->hw = snd_azf3328_capture; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - &snd_azf3328_hw_constraints_rates); - snd_azf3328_dbgcallleave(); - return 0; +static int +snd_azf3328_i2s_out_open(struct snd_pcm_substream *substream) +{ + return snd_azf3328_pcm_open(substream, AZF_CODEC_I2S_OUT); } static int -snd_azf3328_playback_close(struct snd_pcm_substream *substream) +snd_azf3328_pcm_close(struct snd_pcm_substream *substream, + enum snd_azf3328_codec_type codec_type +) { struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); snd_azf3328_dbgcallenter(); - chip->audio_stream[AZF_PLAYBACK].substream = NULL; + chip->codecs[codec_type].substream = NULL; snd_azf3328_dbgcallleave(); return 0; } +static int +snd_azf3328_playback_close(struct snd_pcm_substream *substream) +{ + return snd_azf3328_pcm_close(substream, AZF_CODEC_PLAYBACK); +} + static int snd_azf3328_capture_close(struct snd_pcm_substream *substream) { - struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); + return snd_azf3328_pcm_close(substream, AZF_CODEC_CAPTURE); +} - snd_azf3328_dbgcallenter(); - chip->audio_stream[AZF_CAPTURE].substream = NULL; - snd_azf3328_dbgcallleave(); - return 0; +static int +snd_azf3328_i2s_out_close(struct snd_pcm_substream *substream) +{ + return snd_azf3328_pcm_close(substream, AZF_CODEC_I2S_OUT); } /******************************************************************/ @@ -1829,9 +1848,9 @@ static struct snd_pcm_ops snd_azf3328_playback_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_azf3328_hw_params, .hw_free = snd_azf3328_hw_free, - .prepare = snd_azf3328_playback_prepare, - .trigger = snd_azf3328_playback_trigger, - .pointer = snd_azf3328_playback_pointer + .prepare = snd_azf3328_codec_prepare, + .trigger = snd_azf3328_codec_playback_trigger, + .pointer = snd_azf3328_codec_playback_pointer }; static struct snd_pcm_ops snd_azf3328_capture_ops = { @@ -1840,30 +1859,67 @@ static struct snd_pcm_ops snd_azf3328_capture_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_azf3328_hw_params, .hw_free = snd_azf3328_hw_free, - .prepare = snd_azf3328_capture_prepare, - .trigger = snd_azf3328_capture_trigger, - .pointer = snd_azf3328_capture_pointer + .prepare = snd_azf3328_codec_prepare, + .trigger = snd_azf3328_codec_capture_trigger, + .pointer = snd_azf3328_codec_capture_pointer +}; + +static struct snd_pcm_ops snd_azf3328_i2s_out_ops = { + .open = snd_azf3328_i2s_out_open, + .close = snd_azf3328_i2s_out_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_azf3328_hw_params, + .hw_free = snd_azf3328_hw_free, + .prepare = snd_azf3328_codec_prepare, + .trigger = snd_azf3328_codec_i2s_out_trigger, + .pointer = snd_azf3328_codec_i2s_out_pointer }; static int __devinit -snd_azf3328_pcm(struct snd_azf3328 *chip, int device) +snd_azf3328_pcm(struct snd_azf3328 *chip) { +enum { AZF_PCMDEV_STD, AZF_PCMDEV_I2S_OUT, NUM_AZF_PCMDEVS }; /* pcm devices */ + struct snd_pcm *pcm; int err; snd_azf3328_dbgcallenter(); - if ((err = snd_pcm_new(chip->card, "AZF3328 DSP", device, 1, 1, &pcm)) < 0) + + err = snd_pcm_new(chip->card, "AZF3328 DSP", AZF_PCMDEV_STD, + 1, 1, &pcm); + if (err < 0) return err; - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_azf3328_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_azf3328_capture_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_azf3328_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_azf3328_capture_ops); pcm->private_data = chip; pcm->info_flags = 0; strcpy(pcm->name, chip->card->shortname); - chip->pcm = pcm; + /* same pcm object for playback/capture (see snd_pcm_new() above) */ + chip->pcm[AZF_CODEC_PLAYBACK] = pcm; + chip->pcm[AZF_CODEC_CAPTURE] = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), 64*1024, 64*1024); + snd_dma_pci_data(chip->pci), + 64*1024, 64*1024); + + err = snd_pcm_new(chip->card, "AZF3328 I2S OUT", AZF_PCMDEV_I2S_OUT, + 1, 0, &pcm); + if (err < 0) + return err; + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_azf3328_i2s_out_ops); + + pcm->private_data = chip; + pcm->info_flags = 0; + strcpy(pcm->name, chip->card->shortname); + chip->pcm[AZF_CODEC_I2S_OUT] = pcm; + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + 64*1024, 64*1024); snd_azf3328_dbgcallleave(); return 0; @@ -1902,7 +1958,7 @@ snd_azf3328_timer_start(struct snd_timer *timer) snd_azf3328_dbgtimer("setting timer countdown value %d, add COUNTDOWN|IRQ\n", delay); delay |= TIMER_COUNTDOWN_ENABLE | TIMER_IRQ_ENABLE; spin_lock_irqsave(&chip->reg_lock, flags); - snd_azf3328_codec_outl(chip, IDX_IO_TIMER_VALUE, delay); + snd_azf3328_ctrl_outl(chip, IDX_IO_TIMER_VALUE, delay); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_azf3328_dbgcallleave(); return 0; @@ -1919,7 +1975,7 @@ snd_azf3328_timer_stop(struct snd_timer *timer) spin_lock_irqsave(&chip->reg_lock, flags); /* disable timer countdown and interrupt */ /* FIXME: should we write TIMER_IRQ_ACK here? */ - snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0); + snd_azf3328_ctrl_outb(chip, IDX_IO_TIMER_VALUE + 3, 0); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_azf3328_dbgcallleave(); return 0; @@ -2048,9 +2104,9 @@ snd_azf3328_debug_show_ports(const struct snd_azf3328 *chip) u16 tmp; snd_azf3328_dbgmisc( - "codec_io 0x%lx, game_io 0x%lx, mpu_io 0x%lx, " + "ctrl_io 0x%lx, game_io 0x%lx, mpu_io 0x%lx, " "opl3_io 0x%lx, mixer_io 0x%lx, irq %d\n", - chip->codec_io, chip->game_io, chip->mpu_io, + chip->ctrl_io, chip->game_io, chip->mpu_io, chip->opl3_io, chip->mixer_io, chip->irq ); @@ -2083,9 +2139,9 @@ snd_azf3328_debug_show_ports(const struct snd_azf3328 *chip) inb(0x38c + tmp) ); - for (tmp = 0; tmp < AZF_IO_SIZE_CODEC; tmp += 2) - snd_azf3328_dbgmisc("codec 0x%02x: 0x%04x\n", - tmp, snd_azf3328_codec_inw(chip, tmp) + for (tmp = 0; tmp < AZF_IO_SIZE_CTRL; tmp += 2) + snd_azf3328_dbgmisc("ctrl 0x%02x: 0x%04x\n", + tmp, snd_azf3328_ctrl_inw(chip, tmp) ); for (tmp = 0; tmp < AZF_IO_SIZE_MIXER; tmp += 2) @@ -2106,7 +2162,8 @@ snd_azf3328_create(struct snd_card *card, static struct snd_device_ops ops = { .dev_free = snd_azf3328_dev_free, }; - u16 tmp; + u8 dma_init; + enum snd_azf3328_codec_type codec_type; *rchip = NULL; @@ -2138,14 +2195,21 @@ snd_azf3328_create(struct snd_card *card, if (err < 0) goto out_err; - chip->codec_io = pci_resource_start(pci, 0); + chip->ctrl_io = pci_resource_start(pci, 0); chip->game_io = pci_resource_start(pci, 1); chip->mpu_io = pci_resource_start(pci, 2); - chip->opl3_io = pci_resource_start(pci, 3); + chip->opl3_io = pci_resource_start(pci, 3); chip->mixer_io = pci_resource_start(pci, 4); - chip->audio_stream[AZF_PLAYBACK].portbase = chip->codec_io + 0x00; - chip->audio_stream[AZF_CAPTURE].portbase = chip->codec_io + 0x20; + chip->codecs[AZF_CODEC_PLAYBACK].io_base = + chip->ctrl_io + AZF_IO_OFFS_CODEC_PLAYBACK; + chip->codecs[AZF_CODEC_PLAYBACK].name = "PLAYBACK"; + chip->codecs[AZF_CODEC_CAPTURE].io_base = + chip->ctrl_io + AZF_IO_OFFS_CODEC_CAPTURE; + chip->codecs[AZF_CODEC_CAPTURE].name = "CAPTURE"; + chip->codecs[AZF_CODEC_I2S_OUT].io_base = + chip->ctrl_io + AZF_IO_OFFS_CODEC_I2S_OUT; + chip->codecs[AZF_CODEC_I2S_OUT].name = "I2S_OUT"; if (request_irq(pci->irq, snd_azf3328_interrupt, IRQF_SHARED, card->shortname, chip)) { @@ -2168,20 +2232,25 @@ snd_azf3328_create(struct snd_card *card, if (err < 0) goto out_err; - /* shutdown codecs to save power */ - /* have snd_azf3328_codec_activity() act properly */ - chip->audio_stream[AZF_PLAYBACK].running = 1; - snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 0); + /* standard codec init stuff */ + /* default DMA init value */ + dma_init = DMA_RUN_SOMETHING2|DMA_EPILOGUE_SOMETHING|DMA_SOMETHING_ELSE; + + for (codec_type = AZF_CODEC_PLAYBACK; + codec_type <= AZF_CODEC_I2S_OUT; ++codec_type) { + struct snd_azf3328_codec_data *codec = + &chip->codecs[codec_type]; - /* standard chip init stuff */ - /* default IRQ init value */ - tmp = DMA_PLAY_SOMETHING2|DMA_EPILOGUE_SOMETHING|DMA_SOMETHING_ELSE; + /* shutdown codecs to save power */ + /* have ...ctrl_codec_activity() act properly */ + codec->running = 1; + snd_azf3328_ctrl_codec_activity(chip, codec_type, 0); - spin_lock_irq(&chip->reg_lock); - snd_azf3328_codec_outb(chip, IDX_IO_PLAY_FLAGS, tmp); - snd_azf3328_codec_outb(chip, IDX_IO_REC_FLAGS, tmp); - snd_azf3328_codec_outb(chip, IDX_IO_SOMETHING_FLAGS, tmp); - spin_unlock_irq(&chip->reg_lock); + spin_lock_irq(&chip->reg_lock); + snd_azf3328_codec_outb(codec, IDX_IO_CODEC_DMA_FLAGS, + dma_init); + spin_unlock_irq(&chip->reg_lock); + } snd_card_set_dev(card, &pci->dev); @@ -2244,7 +2313,7 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) if (err < 0) goto out_err; - err = snd_azf3328_pcm(chip, 0); + err = snd_azf3328_pcm(chip); if (err < 0) goto out_err; @@ -2266,7 +2335,7 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) opl3->private_data = chip; sprintf(card->longname, "%s at 0x%lx, irq %i", - card->shortname, chip->codec_io, chip->irq); + card->shortname, chip->ctrl_io, chip->irq); err = snd_card_register(card); if (err < 0) @@ -2317,7 +2386,8 @@ snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state) snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - snd_pcm_suspend_all(chip->pcm); + snd_pcm_suspend_all(chip->pcm[AZF_CODEC_PLAYBACK]); + snd_pcm_suspend_all(chip->pcm[AZF_CODEC_I2S_OUT]); for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; ++reg) chip->saved_regs_mixer[reg] = inw(chip->mixer_io + reg * 2); @@ -2326,11 +2396,11 @@ snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state) snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1); snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); - for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg) - chip->saved_regs_codec[reg] = inw(chip->codec_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_CTRL_PM / 2; ++reg) + chip->saved_regs_ctrl[reg] = inw(chip->ctrl_io + reg * 2); /* manually store the one currently relevant write-only reg, too */ - chip->saved_regs_codec[IDX_IO_6AH / 2] = chip->shadow_reg_codec_6AH; + chip->saved_regs_ctrl[IDX_IO_6AH / 2] = chip->shadow_reg_ctrl_6AH; for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg) chip->saved_regs_game[reg] = inw(chip->game_io + reg * 2); @@ -2349,7 +2419,7 @@ static int snd_azf3328_resume(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); - struct snd_azf3328 *chip = card->private_data; + const struct snd_azf3328 *chip = card->private_data; unsigned reg; pci_set_power_state(pci, PCI_D0); @@ -2370,8 +2440,8 @@ snd_azf3328_resume(struct pci_dev *pci) outw(chip->saved_regs_opl3[reg], chip->opl3_io + reg * 2); for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; ++reg) outw(chip->saved_regs_mixer[reg], chip->mixer_io + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg) - outw(chip->saved_regs_codec[reg], chip->codec_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_CTRL_PM / 2; ++reg) + outw(chip->saved_regs_ctrl[reg], chip->ctrl_io + reg * 2); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; diff --git a/sound/pci/azt3328.h b/sound/pci/azt3328.h index 974e05122f00..11d4b108b8db 100644 --- a/sound/pci/azt3328.h +++ b/sound/pci/azt3328.h @@ -6,50 +6,59 @@ /*** main I/O area port indices ***/ /* (only 0x70 of 0x80 bytes saved/restored by Windows driver) */ -#define AZF_IO_SIZE_CODEC 0x80 -#define AZF_IO_SIZE_CODEC_PM 0x70 +#define AZF_IO_SIZE_CTRL 0x80 +#define AZF_IO_SIZE_CTRL_PM 0x70 -/* the driver initialisation suggests a layout of 4 main areas: - * from 0x00 (playback), from 0x20 (recording) and from 0x40 (maybe MPU401??). +/* the driver initialisation suggests a layout of 4 areas + * within the main card control I/O: + * from 0x00 (playback codec), from 0x20 (recording codec) + * and from 0x40 (most certainly I2S out codec). * And another area from 0x60 to 0x6f (DirectX timer, IRQ management, * power management etc.???). */ -/** playback area **/ -#define IDX_IO_PLAY_FLAGS 0x00 /* PU:0x0000 */ +#define AZF_IO_OFFS_CODEC_PLAYBACK 0x00 +#define AZF_IO_OFFS_CODEC_CAPTURE 0x20 +#define AZF_IO_OFFS_CODEC_I2S_OUT 0x40 + +#define IDX_IO_CODEC_DMA_FLAGS 0x00 /* PU:0x0000 */ /* able to reactivate output after output muting due to 8/16bit * output change, just like 0x0002. * 0x0001 is the only bit that's able to start the DMA counter */ - #define DMA_RESUME 0x0001 /* paused if cleared ? */ + #define DMA_RESUME 0x0001 /* paused if cleared? */ /* 0x0002 *temporarily* set during DMA stopping. hmm * both 0x0002 and 0x0004 set in playback setup. */ /* able to reactivate output after output muting due to 8/16bit * output change, just like 0x0001. */ - #define DMA_PLAY_SOMETHING1 0x0002 /* \ alternated (toggled) */ + #define DMA_RUN_SOMETHING1 0x0002 /* \ alternated (toggled) */ /* 0x0004: NOT able to reactivate output */ - #define DMA_PLAY_SOMETHING2 0x0004 /* / bits */ + #define DMA_RUN_SOMETHING2 0x0004 /* / bits */ #define SOMETHING_ALMOST_ALWAYS_SET 0x0008 /* ???; can be modified */ #define DMA_EPILOGUE_SOMETHING 0x0010 #define DMA_SOMETHING_ELSE 0x0020 /* ??? */ - #define SOMETHING_UNMODIFIABLE 0xffc0 /* unused ? not modifiable */ -#define IDX_IO_PLAY_IRQTYPE 0x02 /* PU:0x0001 */ + #define SOMETHING_UNMODIFIABLE 0xffc0 /* unused? not modifiable */ +#define IDX_IO_CODEC_IRQTYPE 0x02 /* PU:0x0001 */ /* write back to flags in case flags are set, in order to ACK IRQ in handler * (bit 1 of port 0x64 indicates interrupt for one of these three types) * sometimes in this case it just writes 0xffff to globally ACK all IRQs * settings written are not reflected when reading back, though. - * seems to be IRQ, too (frequently used: port |= 0x07 !), but who knows ? */ - #define IRQ_PLAY_SOMETHING 0x0001 /* something & ACK */ - #define IRQ_FINISHED_PLAYBUF_1 0x0002 /* 1st dmabuf finished & ACK */ - #define IRQ_FINISHED_PLAYBUF_2 0x0004 /* 2nd dmabuf finished & ACK */ + * seems to be IRQ, too (frequently used: port |= 0x07 !), but who knows? */ + #define IRQ_SOMETHING 0x0001 /* something & ACK */ + #define IRQ_FINISHED_DMABUF_1 0x0002 /* 1st dmabuf finished & ACK */ + #define IRQ_FINISHED_DMABUF_2 0x0004 /* 2nd dmabuf finished & ACK */ #define IRQMASK_SOME_STATUS_1 0x0008 /* \ related bits */ #define IRQMASK_SOME_STATUS_2 0x0010 /* / (checked together in loop) */ - #define IRQMASK_UNMODIFIABLE 0xffe0 /* unused ? not modifiable */ -#define IDX_IO_PLAY_DMA_START_1 0x04 /* start address of 1st DMA play area, PU:0x00000000 */ -#define IDX_IO_PLAY_DMA_START_2 0x08 /* start address of 2nd DMA play area, PU:0x00000000 */ -#define IDX_IO_PLAY_DMA_LEN_1 0x0c /* length of 1st DMA play area, PU:0x0000 */ -#define IDX_IO_PLAY_DMA_LEN_2 0x0e /* length of 2nd DMA play area, PU:0x0000 */ -#define IDX_IO_PLAY_DMA_CURRPOS 0x10 /* current DMA position, PU:0x00000000 */ -#define IDX_IO_PLAY_DMA_CURROFS 0x14 /* offset within current DMA play area, PU:0x0000 */ -#define IDX_IO_PLAY_SOUNDFORMAT 0x16 /* PU:0x0010 */ + #define IRQMASK_UNMODIFIABLE 0xffe0 /* unused? not modifiable */ + /* start address of 1st DMA transfer area, PU:0x00000000 */ +#define IDX_IO_CODEC_DMA_START_1 0x04 + /* start address of 2nd DMA transfer area, PU:0x00000000 */ +#define IDX_IO_CODEC_DMA_START_2 0x08 + /* both lengths of DMA transfer areas, PU:0x00000000 + length1: offset 0x0c, length2: offset 0x0e */ +#define IDX_IO_CODEC_DMA_LENGTHS 0x0c +#define IDX_IO_CODEC_DMA_CURRPOS 0x10 /* current DMA position, PU:0x00000000 */ + /* offset within current DMA transfer area, PU:0x0000 */ +#define IDX_IO_CODEC_DMA_CURROFS 0x14 +#define IDX_IO_CODEC_SOUNDFORMAT 0x16 /* PU:0x0010 */ /* all unspecified bits can't be modified */ #define SOUNDFORMAT_FREQUENCY_MASK 0x000f #define SOUNDFORMAT_XTAL1 0x00 @@ -76,6 +85,7 @@ #define SOUNDFORMAT_FLAG_16BIT 0x0010 #define SOUNDFORMAT_FLAG_2CHANNELS 0x0020 + /* define frequency helpers, for maximum value safety */ enum azf_freq_t { #define AZF_FREQ(rate) AZF_FREQ_##rate = rate @@ -96,29 +106,6 @@ enum azf_freq_t { #undef AZF_FREQ }; -/** recording area (see also: playback bit flag definitions) **/ -#define IDX_IO_REC_FLAGS 0x20 /* ??, PU:0x0000 */ -#define IDX_IO_REC_IRQTYPE 0x22 /* ??, PU:0x0000 */ - #define IRQ_REC_SOMETHING 0x0001 /* something & ACK */ - #define IRQ_FINISHED_RECBUF_1 0x0002 /* 1st dmabuf finished & ACK */ - #define IRQ_FINISHED_RECBUF_2 0x0004 /* 2nd dmabuf finished & ACK */ - /* hmm, maybe these are just the corresponding *recording* flags ? - * but OTOH they are most likely at port 0x22 instead */ - #define IRQMASK_SOME_STATUS_1 0x0008 /* \ related bits */ - #define IRQMASK_SOME_STATUS_2 0x0010 /* / (checked together in loop) */ -#define IDX_IO_REC_DMA_START_1 0x24 /* PU:0x00000000 */ -#define IDX_IO_REC_DMA_START_2 0x28 /* PU:0x00000000 */ -#define IDX_IO_REC_DMA_LEN_1 0x2c /* PU:0x0000 */ -#define IDX_IO_REC_DMA_LEN_2 0x2e /* PU:0x0000 */ -#define IDX_IO_REC_DMA_CURRPOS 0x30 /* PU:0x00000000 */ -#define IDX_IO_REC_DMA_CURROFS 0x34 /* PU:0x00000000 */ -#define IDX_IO_REC_SOUNDFORMAT 0x36 /* PU:0x0000 */ - -/** hmm, what is this I/O area for? MPU401?? or external DAC via I2S?? (after playback, recording, ???, timer) **/ -#define IDX_IO_SOMETHING_FLAGS 0x40 /* gets set to 0x34 just like port 0x0 and 0x20 on card init, PU:0x0000 */ -/* general */ -#define IDX_IO_42H 0x42 /* PU:0x0001 */ - /** DirectX timer, main interrupt area (FIXME: and something else?) **/ #define IDX_IO_TIMER_VALUE 0x60 /* found this timer area by pure luck :-) */ /* timer countdown value; triggers IRQ when timer is finished */ @@ -138,7 +125,7 @@ enum azf_freq_t { #define IRQ_PLAYBACK 0x0001 #define IRQ_RECORDING 0x0002 - #define IRQ_UNKNOWN1 0x0004 /* most probably I2S port */ + #define IRQ_I2S_OUT 0x0004 /* this IS I2S, right!? (untested) */ #define IRQ_GAMEPORT 0x0008 /* Interrupt of Digital(ly) Enhanced Game Port */ #define IRQ_MPU401 0x0010 #define IRQ_TIMER 0x0020 /* DirectX timer */ @@ -272,6 +259,12 @@ enum { * 11 --> 1/200: */ #define GAME_HWCFG_ADC_COUNTER_FREQ_MASK 0x06 + /* FIXME: these values might be reversed... */ + #define GAME_HWCFG_ADC_COUNTER_FREQ_STD 0 + #define GAME_HWCFG_ADC_COUNTER_FREQ_1_2 1 + #define GAME_HWCFG_ADC_COUNTER_FREQ_1_20 2 + #define GAME_HWCFG_ADC_COUNTER_FREQ_1_200 3 + /* enable gameport legacy I/O address (0x200) * I was unable to locate any configurability for a different address: */ #define GAME_HWCFG_LEGACY_ADDRESS_ENABLE 0x08 -- cgit v1.2.3 From 9983aa62c321a22774e47cf701b6d8b16d92a822 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 Jul 2009 14:31:59 +0200 Subject: ALSA: info - Use krealloc() Use krealloc() to resize the buffer in sound/core/info.c. Signed-off-by: Takashi Iwai --- sound/core/info.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/core/info.c b/sound/core/info.c index 35df614f6c55..3d1f5137420a 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -88,12 +88,10 @@ static int resize_info_buffer(struct snd_info_buffer *buffer, char *nbuf; nsize = PAGE_ALIGN(nsize); - nbuf = kmalloc(nsize, GFP_KERNEL); + nbuf = krealloc(buffer->buffer, nsize, GFP_KERNEL); if (! nbuf) return -ENOMEM; - memcpy(nbuf, buffer->buffer, buffer->len); - kfree(buffer->buffer); buffer->buffer = nbuf; buffer->len = nsize; return 0; -- cgit v1.2.3