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Date: Wed, 18 Aug 2010 17:30:09 +0200
From: Takashi Iwai <tiwai@...e.de>
To: Linus Torvalds <torvalds@...ux-foundation.org>
Cc: Andrew Morton <akpm@...ux-foundation.org>,
linux-kernel@...r.kernel.org
Subject: [GIT PULL] sound fixes for 2.6.36-rc2
Linus,
please pull sound fixes for v2.6.36-rc2 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
Except for a bit lengthy fix for ALC680 codec in hda/patch_realtek.c,
all small and local fixes.
Thanks!
Takashi
===
Jaroslav Kysela (1):
ALSA: emu10k1 - delay the PCM interrupts (add pcm_irq_delay parameter)
Kailang Yang (1):
ALSA: hda - Fix ALC680 base model capture
Mark Brown (1):
ASoC: Remove DSP mode support for WM8776
Takashi Iwai (2):
ALSA: riptide - Fix detection / load of firmware files
ALSA: hda - Add quirk for Dell Vostro 1220
---
include/sound/emu10k1.h | 1 +
sound/core/pcm_native.c | 4 +
sound/pci/emu10k1/emu10k1.c | 4 +
sound/pci/emu10k1/emupcm.c | 30 ++++++-
sound/pci/emu10k1/memory.c | 4 +-
sound/pci/hda/patch_conexant.c | 1 +
sound/pci/hda/patch_realtek.c | 176 ++++++++++++++++++++++++++++++++-------
sound/pci/riptide/riptide.c | 11 +--
sound/soc/codecs/wm8776.c | 7 --
9 files changed, 188 insertions(+), 50 deletions(-)
diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h
index 6a664c3..7dc97d1 100644
--- a/include/sound/emu10k1.h
+++ b/include/sound/emu10k1.h
@@ -1707,6 +1707,7 @@ struct snd_emu10k1 {
unsigned int card_type; /* EMU10K1_CARD_* */
unsigned int ecard_ctrl; /* ecard control bits */
unsigned long dma_mask; /* PCI DMA mask */
+ unsigned int delay_pcm_irq; /* in samples */
int max_cache_pages; /* max memory size / PAGE_SIZE */
struct snd_dma_buffer silent_page; /* silent page */
struct snd_dma_buffer ptb_pages; /* page table pages */
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index a3b2a64..134fc6c 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -978,6 +978,10 @@ static int snd_pcm_do_pause(struct snd_pcm_substream *substream, int push)
{
if (substream->runtime->trigger_master != substream)
return 0;
+ /* some drivers might use hw_ptr to recover from the pause -
+ update the hw_ptr now */
+ if (push)
+ snd_pcm_update_hw_ptr(substream);
/* The jiffies check in snd_pcm_update_hw_ptr*() is done by
* a delta betwen the current jiffies, this gives a large enough
* delta, effectively to skip the check once.
diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
index 4203782..aff8387 100644
--- a/sound/pci/emu10k1/emu10k1.c
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -52,6 +52,7 @@ static int max_synth_voices[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 64};
static int max_buffer_size[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 128};
static int enable_ir[SNDRV_CARDS];
static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */
+static uint delay_pcm_irq[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2};
module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index value for the EMU10K1 soundcard.");
@@ -73,6 +74,8 @@ module_param_array(enable_ir, bool, NULL, 0444);
MODULE_PARM_DESC(enable_ir, "Enable IR.");
module_param_array(subsystem, uint, NULL, 0444);
MODULE_PARM_DESC(subsystem, "Force card subsystem model.");
+module_param_array(delay_pcm_irq, uint, NULL, 0444);
+MODULE_PARM_DESC(delay_pcm_irq, "Delay PCM interrupt by specified number of samples (default 0).");
/*
* Class 0401: 1102:0008 (rev 00) Subsystem: 1102:1001 -> Audigy2 Value Model:SB0400
*/
@@ -127,6 +130,7 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci,
&emu)) < 0)
goto error;
card->private_data = emu;
+ emu->delay_pcm_irq = delay_pcm_irq[dev] & 0x1f;
if ((err = snd_emu10k1_pcm(emu, 0, NULL)) < 0)
goto error;
if ((err = snd_emu10k1_pcm_mic(emu, 1, NULL)) < 0)
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index 55b83ef..622bace 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -332,7 +332,7 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu,
evoice->epcm->ccca_start_addr = start_addr + ccis;
if (extra) {
start_addr += ccis;
- end_addr += ccis;
+ end_addr += ccis + emu->delay_pcm_irq;
}
if (stereo && !extra) {
snd_emu10k1_ptr_write(emu, CPF, voice, CPF_STEREO_MASK);
@@ -360,7 +360,9 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu,
/* Assumption that PT is already 0 so no harm overwriting */
snd_emu10k1_ptr_write(emu, PTRX, voice, (send_amount[0] << 8) | send_amount[1]);
snd_emu10k1_ptr_write(emu, DSL, voice, end_addr | (send_amount[3] << 24));
- snd_emu10k1_ptr_write(emu, PSST, voice, start_addr | (send_amount[2] << 24));
+ snd_emu10k1_ptr_write(emu, PSST, voice,
+ (start_addr + (extra ? emu->delay_pcm_irq : 0)) |
+ (send_amount[2] << 24));
if (emu->card_capabilities->emu_model)
pitch_target = PITCH_48000; /* Disable interpolators on emu1010 card */
else
@@ -732,6 +734,23 @@ static void snd_emu10k1_playback_stop_voice(struct snd_emu10k1 *emu, struct snd_
snd_emu10k1_ptr_write(emu, IP, voice, 0);
}
+static inline void snd_emu10k1_playback_mangle_extra(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_pcm *epcm,
+ struct snd_pcm_substream *substream,
+ struct snd_pcm_runtime *runtime)
+{
+ unsigned int ptr, period_pos;
+
+ /* try to sychronize the current position for the interrupt
+ source voice */
+ period_pos = runtime->status->hw_ptr - runtime->hw_ptr_interrupt;
+ period_pos %= runtime->period_size;
+ ptr = snd_emu10k1_ptr_read(emu, CCCA, epcm->extra->number);
+ ptr &= ~0x00ffffff;
+ ptr |= epcm->ccca_start_addr + period_pos;
+ snd_emu10k1_ptr_write(emu, CCCA, epcm->extra->number, ptr);
+}
+
static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream,
int cmd)
{
@@ -753,6 +772,8 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream,
/* follow thru */
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
case SNDRV_PCM_TRIGGER_RESUME:
+ if (cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE)
+ snd_emu10k1_playback_mangle_extra(emu, epcm, substream, runtime);
mix = &emu->pcm_mixer[substream->number];
snd_emu10k1_playback_prepare_voice(emu, epcm->voices[0], 1, 0, mix);
snd_emu10k1_playback_prepare_voice(emu, epcm->voices[1], 0, 0, mix);
@@ -869,8 +890,9 @@ static snd_pcm_uframes_t snd_emu10k1_playback_pointer(struct snd_pcm_substream *
#endif
/*
printk(KERN_DEBUG
- "ptr = 0x%x, buffer_size = 0x%x, period_size = 0x%x\n",
- ptr, runtime->buffer_size, runtime->period_size);
+ "ptr = 0x%lx, buffer_size = 0x%lx, period_size = 0x%lx\n",
+ (long)ptr, (long)runtime->buffer_size,
+ (long)runtime->period_size);
*/
return ptr;
}
diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c
index ffb1ddb..957a311 100644
--- a/sound/pci/emu10k1/memory.c
+++ b/sound/pci/emu10k1/memory.c
@@ -310,8 +310,10 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst
if (snd_BUG_ON(!hdr))
return NULL;
+ idx = runtime->period_size >= runtime->buffer_size ?
+ (emu->delay_pcm_irq * 2) : 0;
mutex_lock(&hdr->block_mutex);
- blk = search_empty(emu, runtime->dma_bytes);
+ blk = search_empty(emu, runtime->dma_bytes + idx);
if (blk == NULL) {
mutex_unlock(&hdr->block_mutex);
return NULL;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 31b5d9e..c424952 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3049,6 +3049,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x02f5, "Dell",
CXT5066_DELL_LAPTOP),
SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
+ SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTO),
SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO),
SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2cd1ae8..a4dd045 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -19030,6 +19030,7 @@ static int patch_alc888(struct hda_codec *codec)
/*
* ALC680 support
*/
+#define ALC680_DIGIN_NID ALC880_DIGIN_NID
#define ALC680_DIGOUT_NID ALC880_DIGOUT_NID
#define alc680_modes alc260_modes
@@ -19044,23 +19045,93 @@ static hda_nid_t alc680_adc_nids[3] = {
0x07, 0x08, 0x09
};
+/*
+ * Analog capture ADC cgange
+ */
+static int alc680_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct alc_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ unsigned int pre_mic, pre_line;
+
+ pre_mic = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_MIC]);
+ pre_line = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_LINE]);
+
+ spec->cur_adc_stream_tag = stream_tag;
+ spec->cur_adc_format = format;
+
+ if (pre_mic || pre_line) {
+ if (pre_mic)
+ snd_hda_codec_setup_stream(codec, 0x08, stream_tag, 0,
+ format);
+ else
+ snd_hda_codec_setup_stream(codec, 0x09, stream_tag, 0,
+ format);
+ } else
+ snd_hda_codec_setup_stream(codec, 0x07, stream_tag, 0, format);
+ return 0;
+}
+
+static int alc680_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ snd_hda_codec_cleanup_stream(codec, 0x07);
+ snd_hda_codec_cleanup_stream(codec, 0x08);
+ snd_hda_codec_cleanup_stream(codec, 0x09);
+ return 0;
+}
+
+static struct hda_pcm_stream alc680_pcm_analog_auto_capture = {
+ .substreams = 1, /* can be overridden */
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in alc_build_pcms */
+ .ops = {
+ .prepare = alc680_capture_pcm_prepare,
+ .cleanup = alc680_capture_pcm_cleanup
+ },
+};
+
static struct snd_kcontrol_new alc680_base_mixer[] = {
/* output mixer control */
HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x4, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Int Mic Boost", 0x12, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line In Boost", 0x19, 0, HDA_INPUT),
{ }
};
-static struct snd_kcontrol_new alc680_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT),
+static struct hda_bind_ctls alc680_bind_cap_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+ 0
+ },
+};
+
+static struct hda_bind_ctls alc680_bind_cap_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+ 0
+ },
+};
+
+static struct snd_kcontrol_new alc680_master_capture_mixer[] = {
+ HDA_BIND_VOL("Capture Volume", &alc680_bind_cap_vol),
+ HDA_BIND_SW("Capture Switch", &alc680_bind_cap_switch),
{ } /* end */
};
@@ -19068,25 +19139,73 @@ static struct snd_kcontrol_new alc680_capture_mixer[] = {
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc680_init_verbs[] = {
- /* Unmute DAC0-1 and set vol = 0 */
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
+
{ }
};
+/* toggle speaker-output according to the hp-jack state */
+static void alc680_base_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x16;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x15;
+ spec->autocfg.input_pins[AUTO_PIN_MIC] = 0x18;
+ spec->autocfg.input_pins[AUTO_PIN_LINE] = 0x19;
+}
+
+static void alc680_rec_autoswitch(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ unsigned int present;
+ hda_nid_t new_adc;
+
+ present = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_MIC]);
+
+ new_adc = present ? 0x8 : 0x7;
+ __snd_hda_codec_cleanup_stream(codec, !present ? 0x8 : 0x7, 1);
+ snd_hda_codec_setup_stream(codec, new_adc,
+ spec->cur_adc_stream_tag, 0,
+ spec->cur_adc_format);
+
+}
+
+static void alc680_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc_automute_amp(codec);
+ if ((res >> 26) == ALC880_MIC_EVENT)
+ alc680_rec_autoswitch(codec);
+}
+
+static void alc680_inithook(struct hda_codec *codec)
+{
+ alc_automute_amp(codec);
+ alc680_rec_autoswitch(codec);
+}
+
/* create input playback/capture controls for the given pin */
static int alc680_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
const char *ctlname, int idx)
@@ -19197,13 +19316,7 @@ static void alc680_auto_init_hp_out(struct hda_codec *codec)
#define alc680_pcm_analog_capture alc880_pcm_analog_capture
#define alc680_pcm_analog_alt_capture alc880_pcm_analog_alt_capture
#define alc680_pcm_digital_playback alc880_pcm_digital_playback
-
-static struct hda_input_mux alc680_capture_source = {
- .num_items = 1,
- .items = {
- { "Mic", 0x0 },
- },
-};
+#define alc680_pcm_digital_capture alc880_pcm_digital_capture
/*
* BIOS auto configuration
@@ -19218,6 +19331,7 @@ static int alc680_parse_auto_config(struct hda_codec *codec)
alc680_ignore);
if (err < 0)
return err;
+
if (!spec->autocfg.line_outs) {
if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) {
spec->multiout.max_channels = 2;
@@ -19239,8 +19353,6 @@ static int alc680_parse_auto_config(struct hda_codec *codec)
add_mixer(spec, spec->kctls.list);
add_verb(spec, alc680_init_verbs);
- spec->num_mux_defs = 1;
- spec->input_mux = &alc680_capture_source;
err = alc_auto_add_mic_boost(codec);
if (err < 0)
@@ -19279,17 +19391,17 @@ static struct snd_pci_quirk alc680_cfg_tbl[] = {
static struct alc_config_preset alc680_presets[] = {
[ALC680_BASE] = {
.mixers = { alc680_base_mixer },
- .cap_mixer = alc680_capture_mixer,
+ .cap_mixer = alc680_master_capture_mixer,
.init_verbs = { alc680_init_verbs },
.num_dacs = ARRAY_SIZE(alc680_dac_nids),
.dac_nids = alc680_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc680_adc_nids),
- .adc_nids = alc680_adc_nids,
- .hp_nid = 0x04,
.dig_out_nid = ALC680_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc680_modes),
.channel_mode = alc680_modes,
- .input_mux = &alc680_capture_source,
+ .unsol_event = alc680_unsol_event,
+ .setup = alc680_base_setup,
+ .init_hook = alc680_inithook,
+
},
};
@@ -19333,9 +19445,9 @@ static int patch_alc680(struct hda_codec *codec)
setup_preset(codec, &alc680_presets[board_config]);
spec->stream_analog_playback = &alc680_pcm_analog_playback;
- spec->stream_analog_capture = &alc680_pcm_analog_capture;
- spec->stream_analog_alt_capture = &alc680_pcm_analog_alt_capture;
+ spec->stream_analog_capture = &alc680_pcm_analog_auto_capture;
spec->stream_digital_playback = &alc680_pcm_digital_playback;
+ spec->stream_digital_capture = &alc680_pcm_digital_capture;
if (!spec->adc_nids) {
spec->adc_nids = alc680_adc_nids;
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index f64fb7d..ad5202e 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1224,15 +1224,14 @@ static int try_to_load_firmware(struct cmdif *cif, struct snd_riptide *chip)
firmware.firmware.ASIC, firmware.firmware.CODEC,
firmware.firmware.AUXDSP, firmware.firmware.PROG);
+ if (!chip)
+ return 1;
+
for (i = 0; i < FIRMWARE_VERSIONS; i++) {
if (!memcmp(&firmware_versions[i], &firmware, sizeof(firmware)))
- break;
- }
- if (i >= FIRMWARE_VERSIONS)
- return 0; /* no match */
+ return 1; /* OK */
- if (!chip)
- return 1; /* OK */
+ }
snd_printdd("Writing Firmware\n");
if (!chip->fw_entry) {
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index 4e212ed..f8154e6 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -178,13 +178,6 @@ static int wm8776_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
case SND_SOC_DAIFMT_LEFT_J:
iface |= 0x0001;
break;
- /* FIXME: CHECK A/B */
- case SND_SOC_DAIFMT_DSP_A:
- iface |= 0x0003;
- break;
- case SND_SOC_DAIFMT_DSP_B:
- iface |= 0x0007;
- break;
default:
return -EINVAL;
}
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