diff options
Diffstat (limited to 'sound')
54 files changed, 845 insertions, 170 deletions
diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 3fd1a7e24928..552b97afbca5 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -1073,10 +1073,10 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) sdev->pcmid = -1; list_del(&ldev->list); layouts_list_items--; + kfree(ldev); outnodev: of_node_put(sound); layout_device = NULL; - kfree(ldev); return -ENODEV; } diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 86d0caf91b35..62e90b862a0d 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1761,6 +1761,10 @@ static int wait_for_avail(struct snd_pcm_substream *substream, snd_pcm_uframes_t avail = 0; long wait_time, tout; + init_waitqueue_entry(&wait, current); + set_current_state(TASK_INTERRUPTIBLE); + add_wait_queue(&runtime->tsleep, &wait); + if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { @@ -1771,16 +1775,32 @@ static int wait_for_avail(struct snd_pcm_substream *substream, } wait_time = msecs_to_jiffies(wait_time * 1000); } - init_waitqueue_entry(&wait, current); - add_wait_queue(&runtime->tsleep, &wait); + for (;;) { if (signal_pending(current)) { err = -ERESTARTSYS; break; } + + /* + * We need to check if space became available already + * (and thus the wakeup happened already) first to close + * the race of space already having become available. + * This check must happen after been added to the waitqueue + * and having current state be INTERRUPTIBLE. + */ + if (is_playback) + avail = snd_pcm_playback_avail(runtime); + else + avail = snd_pcm_capture_avail(runtime); + if (avail >= runtime->twake) + break; snd_pcm_stream_unlock_irq(substream); - tout = schedule_timeout_interruptible(wait_time); + + tout = schedule_timeout(wait_time); + snd_pcm_stream_lock_irq(substream); + set_current_state(TASK_INTERRUPTIBLE); switch (runtime->status->state) { case SNDRV_PCM_STATE_SUSPENDED: err = -ESTRPIPE; @@ -1806,14 +1826,9 @@ static int wait_for_avail(struct snd_pcm_substream *substream, err = -EIO; break; } - if (is_playback) - avail = snd_pcm_playback_avail(runtime); - else - avail = snd_pcm_capture_avail(runtime); - if (avail >= runtime->twake) - break; } _endloop: + set_current_state(TASK_RUNNING); remove_wait_queue(&runtime->tsleep, &wait); *availp = avail; return err; diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 200c9a1d48b7..a872d0a82976 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -1909,6 +1909,7 @@ static unsigned int ad1981_jacks_whitelist[] = { 0x103c0944, /* HP nc6220 */ 0x103c0934, /* HP nc8220 */ 0x103c006d, /* HP nx9105 */ + 0x103c300d, /* HP Compaq dc5100 SFF(PT003AW) */ 0x17340088, /* FSC Scenic-W */ 0 /* end */ }; diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index e4d76a270c9f..579fc0dce128 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2625,16 +2625,19 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) int err; snd_azf3328_dbgcallenter(); - if (dev >= SNDRV_CARDS) - return -ENODEV; + if (dev >= SNDRV_CARDS) { + err = -ENODEV; + goto out; + } if (!enable[dev]) { dev++; - return -ENOENT; + err = -ENOENT; + goto out; } err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) - return err; + goto out; strcpy(card->driver, "AZF3328"); strcpy(card->shortname, "Aztech AZF3328 (PCI168)"); diff --git a/sound/pci/hda/alc268_quirks.c b/sound/pci/hda/alc268_quirks.c index be58bf2f3aec..2e5876ce71fe 100644 --- a/sound/pci/hda/alc268_quirks.c +++ b/sound/pci/hda/alc268_quirks.c @@ -476,8 +476,8 @@ static const struct snd_pci_quirk alc268_ssid_cfg_tbl[] = { static const struct alc_config_preset alc268_presets[] = { [ALC267_QUANTA_IL1] = { - .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer, - alc268_capture_nosrc_mixer }, + .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer }, + .cap_mixer = alc268_capture_nosrc_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc267_quanta_il1_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -492,8 +492,8 @@ static const struct alc_config_preset alc268_presets[] = { .init_hook = alc_inithook, }, [ALC268_3ST] = { - .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, - alc268_beep_mixer }, + .mixers = { alc268_base_mixer, alc268_beep_mixer }, + .cap_mixer = alc268_capture_alt_mixer, .init_verbs = { alc268_base_init_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), .dac_nids = alc268_dac_nids, @@ -507,8 +507,8 @@ static const struct alc_config_preset alc268_presets[] = { .input_mux = &alc268_capture_source, }, [ALC268_TOSHIBA] = { - .mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer, - alc268_beep_mixer }, + .mixers = { alc268_toshiba_mixer, alc268_beep_mixer }, + .cap_mixer = alc268_capture_alt_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_toshiba_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -525,8 +525,8 @@ static const struct alc_config_preset alc268_presets[] = { .init_hook = alc_inithook, }, [ALC268_ACER] = { - .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer, - alc268_beep_mixer }, + .mixers = { alc268_acer_mixer, alc268_beep_mixer }, + .cap_mixer = alc268_capture_alt_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -543,8 +543,8 @@ static const struct alc_config_preset alc268_presets[] = { .init_hook = alc_inithook, }, [ALC268_ACER_DMIC] = { - .mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer, - alc268_beep_mixer }, + .mixers = { alc268_acer_dmic_mixer, alc268_beep_mixer }, + .cap_mixer = alc268_capture_alt_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -561,9 +561,8 @@ static const struct alc_config_preset alc268_presets[] = { .init_hook = alc_inithook, }, [ALC268_ACER_ASPIRE_ONE] = { - .mixers = { alc268_acer_aspire_one_mixer, - alc268_beep_mixer, - alc268_capture_nosrc_mixer }, + .mixers = { alc268_acer_aspire_one_mixer, alc268_beep_mixer}, + .cap_mixer = alc268_capture_nosrc_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_aspire_one_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -579,8 +578,8 @@ static const struct alc_config_preset alc268_presets[] = { .init_hook = alc_inithook, }, [ALC268_DELL] = { - .mixers = { alc268_dell_mixer, alc268_beep_mixer, - alc268_capture_nosrc_mixer }, + .mixers = { alc268_dell_mixer, alc268_beep_mixer}, + .cap_mixer = alc268_capture_nosrc_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_dell_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -596,8 +595,8 @@ static const struct alc_config_preset alc268_presets[] = { .init_hook = alc_inithook, }, [ALC268_ZEPTO] = { - .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, - alc268_beep_mixer }, + .mixers = { alc268_base_mixer, alc268_beep_mixer }, + .cap_mixer = alc268_capture_alt_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_toshiba_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -616,7 +615,8 @@ static const struct alc_config_preset alc268_presets[] = { }, #ifdef CONFIG_SND_DEBUG [ALC268_TEST] = { - .mixers = { alc268_test_mixer, alc268_capture_mixer }, + .mixers = { alc268_test_mixer }, + .cap_mixer = alc268_capture_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_volume_init_verbs, alc268_beep_init_verbs }, diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3e7850c238c3..f3aefef37216 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -579,9 +579,13 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, return -1; } recursive++; - for (i = 0; i < nums; i++) + for (i = 0; i < nums; i++) { + unsigned int type = get_wcaps_type(get_wcaps(codec, conn[i])); + if (type == AC_WID_PIN || type == AC_WID_AUD_OUT) + continue; if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0) return i; + } return -1; } EXPORT_SYMBOL_HDA(snd_hda_get_conn_index); diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 28ce17d09c33..c34f730f4815 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -144,25 +144,17 @@ static int cea_sampling_frequencies[8] = { SNDRV_PCM_RATE_192000, /* 7: 192000Hz */ }; -static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid, +static unsigned int hdmi_get_eld_data(struct hda_codec *codec, hda_nid_t nid, int byte_index) { unsigned int val; val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_ELDD, byte_index); - #ifdef BE_PARANOID printk(KERN_INFO "HDMI: ELD data byte %d: 0x%x\n", byte_index, val); #endif - - if ((val & AC_ELDD_ELD_VALID) == 0) { - snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n", - byte_index); - val = 0; - } - - return val & AC_ELDD_ELD_DATA; + return val; } #define GRAB_BITS(buf, byte, lowbit, bits) \ @@ -344,11 +336,26 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, if (!buf) return -ENOMEM; - for (i = 0; i < size; i++) - buf[i] = hdmi_get_eld_byte(codec, nid, i); + for (i = 0; i < size; i++) { + unsigned int val = hdmi_get_eld_data(codec, nid, i); + if (!(val & AC_ELDD_ELD_VALID)) { + if (!i) { + snd_printd(KERN_INFO + "HDMI: invalid ELD data\n"); + ret = -EINVAL; + goto error; + } + snd_printd(KERN_INFO + "HDMI: invalid ELD data byte %d\n", i); + val = 0; + } else + val &= AC_ELDD_ELD_DATA; + buf[i] = val; + } ret = hdmi_update_eld(eld, buf, size); +error: kfree(buf); return ret; } diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 47d6ffc9b5b5..c45f3e69bcf0 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -375,7 +375,7 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx) static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin, unsigned int *idxp) { - int i; + int i, idx; hda_nid_t nid; nid = codec->start_nid; @@ -384,9 +384,11 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin, type = get_wcaps_type(get_wcaps(codec, nid)); if (type != AC_WID_AUD_IN) continue; - *idxp = snd_hda_get_conn_index(codec, nid, pin, false); - if (*idxp >= 0) + idx = snd_hda_get_conn_index(codec, nid, pin, false); + if (idx >= 0) { + *idxp = idx; return nid; + } } return 0; } @@ -533,7 +535,7 @@ static int add_volume(struct hda_codec *codec, const char *name, int index, unsigned int pval, int dir, struct snd_kcontrol **kctlp) { - char tmp[32]; + char tmp[44]; struct snd_kcontrol_new knew = HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT); knew.private_value = pval; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 502fc9499453..7696d05b9356 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3348,6 +3348,8 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t pin, #define MAX_AUTO_DACS 5 +#define DAC_SLAVE_FLAG 0x8000 /* filled dac is a slave */ + /* fill analog DAC list from the widget tree */ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs) { @@ -3370,16 +3372,26 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs) /* fill pin_dac_pair list from the pin and dac list */ static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins, int num_pins, hda_nid_t *dacs, int *rest, - struct pin_dac_pair *filled, int type) + struct pin_dac_pair *filled, int nums, + int type) { - int i, nums; + int i, start = nums; - nums = 0; - for (i = 0; i < num_pins; i++) { + for (i = 0; i < num_pins; i++, nums++) { filled[nums].pin = pins[i]; filled[nums].type = type; filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest); - nums++; + if (filled[nums].dac) + continue; + if (filled[start].dac && get_connection_index(codec, pins[i], filled[start].dac) >= 0) { + filled[nums].dac = filled[start].dac | DAC_SLAVE_FLAG; + continue; + } + if (filled[0].dac && get_connection_index(codec, pins[i], filled[0].dac) >= 0) { + filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG; + continue; + } + snd_printdd("Failed to find a DAC for pin 0x%x", pins[i]); } return nums; } @@ -3395,19 +3407,19 @@ static void cx_auto_parse_output(struct hda_codec *codec) rest = fill_cx_auto_dacs(codec, dacs); /* parse all analog output pins */ nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs, - dacs, &rest, spec->dac_info, - AUTO_PIN_LINE_OUT); - nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs, - dacs, &rest, spec->dac_info + nums, - AUTO_PIN_HP_OUT); - nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs, - dacs, &rest, spec->dac_info + nums, - AUTO_PIN_SPEAKER_OUT); + dacs, &rest, spec->dac_info, 0, + AUTO_PIN_LINE_OUT); + nums = fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs, + dacs, &rest, spec->dac_info, nums, + AUTO_PIN_HP_OUT); + nums = fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs, + dacs, &rest, spec->dac_info, nums, + AUTO_PIN_SPEAKER_OUT); spec->dac_info_filled = nums; /* fill multiout struct */ for (i = 0; i < nums; i++) { hda_nid_t dac = spec->dac_info[i].dac; - if (!dac) + if (!dac || (dac & DAC_SLAVE_FLAG)) continue; switch (spec->dac_info[i].type) { case AUTO_PIN_LINE_OUT: @@ -3862,7 +3874,7 @@ static void cx_auto_parse_input(struct hda_codec *codec) } if (imux->num_items >= 2 && cfg->num_inputs == imux->num_items) cx_auto_check_auto_mic(codec); - if (imux->num_items > 1 && !spec->auto_mic) { + if (imux->num_items > 1) { for (i = 1; i < imux->num_items; i++) { if (spec->imux_info[i].adc != spec->imux_info[0].adc) { spec->adc_switching = 1; @@ -4035,6 +4047,8 @@ static void cx_auto_init_output(struct hda_codec *codec) nid = spec->dac_info[i].dac; if (!nid) nid = spec->multiout.dac_nids[0]; + else if (nid & DAC_SLAVE_FLAG) + nid &= ~DAC_SLAVE_FLAG; select_connection(codec, spec->dac_info[i].pin, nid); } if (spec->auto_mute) { @@ -4167,9 +4181,11 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac, hda_nid_t pin, const char *name, int idx) { unsigned int caps; - caps = query_amp_caps(codec, dac, HDA_OUTPUT); - if (caps & AC_AMPCAP_NUM_STEPS) - return cx_auto_add_pb_volume(codec, dac, name, idx); + if (dac && !(dac & DAC_SLAVE_FLAG)) { + caps = query_amp_caps(codec, dac, HDA_OUTPUT); + if (caps & AC_AMPCAP_NUM_STEPS) + return cx_auto_add_pb_volume(codec, dac, name, idx); + } caps = query_amp_caps(codec, pin, HDA_OUTPUT); if (caps & AC_AMPCAP_NUM_STEPS) return cx_auto_add_pb_volume(codec, pin, name, idx); @@ -4191,8 +4207,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) for (i = 0; i < spec->dac_info_filled; i++) { const char *label; int idx, type; - if (!spec->dac_info[i].dac) - continue; + hda_nid_t dac = spec->dac_info[i].dac; type = spec->dac_info[i].type; if (type == AUTO_PIN_LINE_OUT) type = spec->autocfg.line_out_type; @@ -4211,7 +4226,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) idx = num_spk++; break; } - err = try_add_pb_volume(codec, spec->dac_info[i].dac, + err = try_add_pb_volume(codec, dac, spec->dac_info[i].pin, label, idx); if (err < 0) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9a1aa09f47fe..0503c999e7d3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -168,7 +168,7 @@ struct alc_spec { unsigned int auto_mic_valid_imux:1; /* valid imux for auto-mic */ unsigned int automute:1; /* HP automute enabled */ unsigned int detect_line:1; /* Line-out detection enabled */ - unsigned int automute_lines:1; /* automute line-out as well */ + unsigned int automute_lines:1; /* automute line-out as well; NOP when automute_hp_lo isn't set */ unsigned int automute_hp_lo:1; /* both HP and LO available */ /* other flags */ @@ -551,7 +551,7 @@ static void update_speakers(struct hda_codec *codec) if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0] || spec->autocfg.line_out_pins[0] == spec->autocfg.speaker_pins[0]) return; - if (!spec->automute_lines || !spec->automute) + if (!spec->automute || (spec->automute_hp_lo && !spec->automute_lines)) on = 0; else on = spec->jack_present; @@ -565,11 +565,11 @@ static void alc_hp_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (!spec->automute) - return; spec->jack_present = detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins), spec->autocfg.hp_pins); + if (!spec->automute) + return; update_speakers(codec); } @@ -578,11 +578,11 @@ static void alc_line_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (!spec->automute || !spec->detect_line) - return; spec->line_jack_present = detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins), spec->autocfg.line_out_pins); + if (!spec->automute || !spec->detect_line) + return; update_speakers(codec); } @@ -803,7 +803,7 @@ static int alc_automute_mode_get(struct snd_kcontrol *kcontrol, unsigned int val; if (!spec->automute) val = 0; - else if (!spec->automute_lines) + else if (!spec->automute_hp_lo || !spec->automute_lines) val = 1; else val = 2; @@ -824,7 +824,8 @@ static int alc_automute_mode_put(struct snd_kcontrol *kcontrol, spec->automute = 0; break; case 1: - if (spec->automute && !spec->automute_lines) + if (spec->automute && + (!spec->automute_hp_lo || !spec->automute_lines)) return 0; spec->automute = 1; spec->automute_lines = 0; @@ -1784,6 +1785,7 @@ static const char * const alc_slave_vols[] = { "Speaker Playback Volume", "Mono Playback Volume", "Line-Out Playback Volume", + "PCM Playback Volume", NULL, }; @@ -1798,6 +1800,7 @@ static const char * const alc_slave_sws[] = { "Mono Playback Switch", "IEC958 Playback Switch", "Line-Out Playback Switch", + "PCM Playback Switch", NULL, }; @@ -3081,16 +3084,22 @@ static void alc_auto_init_multi_out(struct hda_codec *codec) static void alc_auto_init_extra_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t pin; + hda_nid_t pin, dac; pin = spec->autocfg.hp_pins[0]; - if (pin) - alc_auto_set_output_and_unmute(codec, pin, PIN_HP, - spec->multiout.hp_nid); + if (pin) { + dac = spec->multiout.hp_nid; + if (!dac) + dac = spec->multiout.dac_nids[0]; + alc_auto_set_output_and_unmute(codec, pin, PIN_HP, dac); + } pin = spec->autocfg.speaker_pins[0]; - if (pin) - alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, - spec->multiout.extra_out_nid[0]); + if (pin) { + dac = spec->multiout.extra_out_nid[0]; + if (!dac) + dac = spec->multiout.dac_nids[0]; + alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac); + } } /* diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index aa376b59c006..1b7c11432aa7 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -673,6 +673,7 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol, return 0; } +#ifdef CONFIG_SND_HDA_POWER_SAVE static int stac_vrefout_set(struct hda_codec *codec, hda_nid_t nid, unsigned int new_vref) { @@ -696,6 +697,7 @@ static int stac_vrefout_set(struct hda_codec *codec, return 1; } +#endif static unsigned int stac92xx_vref_set(struct hda_codec *codec, hda_nid_t nid, unsigned int new_vref) @@ -6571,6 +6573,7 @@ static const struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx }, { .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx }, { .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76df, .name = "92HD93BXX", .patch = patch_stac92hd83xxx}, { .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx}, { .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx}, { .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx}, diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index d6651c033cb7..5956584ea3a4 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -56,7 +56,7 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 48000: - clk = 12288000; + clk = 24576000; break; } @@ -103,7 +103,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { .cpu_dai_name = "bfin-tdm.0", .codec_dai_name ="ad193x-hifi", .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad193x.5", + .codec_name = "spi0.5", .ops = &bf5xx_ad193x_ops, }, { @@ -112,7 +112,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { .cpu_dai_name = "bfin-tdm.1", .codec_dai_name ="ad193x-hifi", .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad193x.5", + .codec_name = "spi0.5", .ops = &bf5xx_ad193x_ops, }, }; diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 2374ca5ffe68..eedb6f5e5823 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -27,11 +27,6 @@ struct ad193x_priv { int sysclk; }; -/* ad193x register cache & default register settings */ -static const u8 ad193x_reg[AD193X_NUM_REGS] = { - 0, 0, 0, 0, 0, 0, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0, 0, 0, -}; - /* * AD193X volume/mute/de-emphasis etc. controls */ @@ -307,7 +302,8 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg); reg = snd_soc_read(codec, AD193X_DAC_CTRL2); - reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len; + reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) + | (word_len << AD193X_DAC_WORD_LEN_SHFT); snd_soc_write(codec, AD193X_DAC_CTRL2, reg); reg = snd_soc_read(codec, AD193X_ADC_CTRL1); @@ -389,9 +385,6 @@ static int ad193x_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_ad193x = { .probe = ad193x_probe, - .reg_cache_default = ad193x_reg, - .reg_cache_size = AD193X_NUM_REGS, - .reg_word_size = sizeof(u16), }; #if defined(CONFIG_SPI_MASTER) diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index 9747b5497877..cccc2e8e5fbd 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -34,7 +34,8 @@ #define AD193X_DAC_LEFT_HIGH (1 << 3) #define AD193X_DAC_BCLK_INV (1 << 7) #define AD193X_DAC_CTRL2 0x804 -#define AD193X_DAC_WORD_LEN_MASK 0xC +#define AD193X_DAC_WORD_LEN_SHFT 3 +#define AD193X_DAC_WORD_LEN_MASK 0x18 #define AD193X_DAC_MASTER_MUTE 1 #define AD193X_DAC_CHNL_MUTE 0x805 #define AD193X_DACL1_MUTE 0 @@ -63,7 +64,7 @@ #define AD193X_ADC_CTRL1 0x80f #define AD193X_ADC_SERFMT_MASK 0x60 #define AD193X_ADC_SERFMT_STEREO (0 << 5) -#define AD193X_ADC_SERFMT_TDM (1 << 2) +#define AD193X_ADC_SERFMT_TDM (1 << 5) #define AD193X_ADC_SERFMT_AUX (2 << 5) #define AD193X_ADC_WORD_LEN_MASK 0x3 #define AD193X_ADC_CTRL2 0x810 diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 409d89d1f34c..fbd7eb9e61ce 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -857,6 +857,7 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1); if (ret != 0) { dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); + kfree(sta32x); return ret; } diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 38f38fddd190..d0003cc3bcd6 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -778,11 +778,19 @@ static int __devexit wm8750_spi_remove(struct spi_device *spi) return 0; } +static const struct spi_device_id wm8750_spi_ids[] = { + { "wm8750", 0 }, + { "wm8987", 0 }, + { }, +}; +MODULE_DEVICE_TABLE(spi, wm8750_spi_ids); + static struct spi_driver wm8750_spi_driver = { .driver = { .name = "wm8750-codec", .owner = THIS_MODULE, }, + .id_table = wm8750_spi_ids, .probe = wm8750_spi_probe, .remove = __devexit_p(wm8750_spi_remove), }; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 43e3d760766f..4ad8ebd290e3 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2046,8 +2046,13 @@ static int wm8903_probe(struct snd_soc_codec *codec) /* power down chip */ static int wm8903_remove(struct snd_soc_codec *codec) { + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + wm8903_free_gpio(codec); wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); + if (wm8903->irq) + free_irq(wm8903->irq, codec); + return 0; } diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 60d740ebeb5b..1725550c293e 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2221,6 +2221,8 @@ static int sysclk_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: if (fll) { + try_wait_for_completion(&wm8962->fll_lock); + snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_ENA, WM8962_FLL_ENA); if (wm8962->irq) { @@ -2927,10 +2929,6 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec, WM8962_BIAS_ENA | 0x180); msleep(5); - - snd_soc_update_bits(codec, WM8962_CLOCKING2, - WM8962_CLKREG_OVD, - WM8962_CLKREG_OVD); } /* VMID 2*250k */ @@ -3288,6 +3286,8 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_write(codec, WM8962_FLL_CONTROL_7, fll_div.lambda); snd_soc_write(codec, WM8962_FLL_CONTROL_8, fll_div.n); + try_wait_for_completion(&wm8962->fll_lock); + snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK | WM8962_FLL_ENA, fll1); @@ -3868,6 +3868,10 @@ static int wm8962_probe(struct snd_soc_codec *codec) */ snd_soc_update_bits(codec, WM8962_CLOCKING2, WM8962_SYSCLK_ENA, 0); + /* Ensure we have soft control over all registers */ + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_CLKREG_OVD, WM8962_CLKREG_OVD); + regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); if (pdata) { diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 09e680ae88b2..b393f9fac97a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2981,6 +2981,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.dcs_readback_mode = 1; break; } + break; case WM8958: wm8994->hubs.dcs_readback_mode = 1; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index ab8e9d1aaff0..0cdb9d105671 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -420,7 +420,7 @@ static const char *sidetone_hpf_text[] = { }; static const struct soc_enum sidetone_hpf = - SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 6, sidetone_hpf_text); + SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 7, sidetone_hpf_text); static const char *hpf_mode_text[] = { "HiFi", "Custom", "Voice" @@ -988,15 +988,10 @@ SND_SOC_DAPM_MICBIAS("MICB1", WM8996_POWER_MANAGEMENT_1, 8, 0), SND_SOC_DAPM_PGA("IN1L PGA", WM8996_POWER_MANAGEMENT_2, 5, 0, NULL, 0), SND_SOC_DAPM_PGA("IN1R PGA", WM8996_POWER_MANAGEMENT_2, 4, 0, NULL, 0), -SND_SOC_DAPM_MUX("IN1L Mux", SND_SOC_NOPM, 0, 0, &in1_mux), -SND_SOC_DAPM_MUX("IN1R Mux", SND_SOC_NOPM, 0, 0, &in1_mux), -SND_SOC_DAPM_MUX("IN2L Mux", SND_SOC_NOPM, 0, 0, &in2_mux), -SND_SOC_DAPM_MUX("IN2R Mux", SND_SOC_NOPM, 0, 0, &in2_mux), - -SND_SOC_DAPM_PGA("IN1L", WM8996_POWER_MANAGEMENT_7, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA("IN1R", WM8996_POWER_MANAGEMENT_7, 3, 0, NULL, 0), -SND_SOC_DAPM_PGA("IN2L", WM8996_POWER_MANAGEMENT_7, 6, 0, NULL, 0), -SND_SOC_DAPM_PGA("IN2R", WM8996_POWER_MANAGEMENT_7, 7, 0, NULL, 0), +SND_SOC_DAPM_MUX("IN1L Mux", WM8996_POWER_MANAGEMENT_7, 2, 0, &in1_mux), +SND_SOC_DAPM_MUX("IN1R Mux", WM8996_POWER_MANAGEMENT_7, 3, 0, &in1_mux), +SND_SOC_DAPM_MUX("IN2L Mux", WM8996_POWER_MANAGEMENT_7, 6, 0, &in2_mux), +SND_SOC_DAPM_MUX("IN2R Mux", WM8996_POWER_MANAGEMENT_7, 7, 0, &in2_mux), SND_SOC_DAPM_SUPPLY("DMIC2", WM8996_POWER_MANAGEMENT_7, 9, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DMIC1", WM8996_POWER_MANAGEMENT_7, 8, 0, NULL, 0), @@ -1213,6 +1208,16 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "AIF2RX0", NULL, "AIFCLK" }, { "AIF2RX1", NULL, "AIFCLK" }, + { "AIF1TX0", NULL, "AIFCLK" }, + { "AIF1TX1", NULL, "AIFCLK" }, + { "AIF1TX2", NULL, "AIFCLK" }, + { "AIF1TX3", NULL, "AIFCLK" }, + { "AIF1TX4", NULL, "AIFCLK" }, + { "AIF1TX5", NULL, "AIFCLK" }, + + { "AIF2TX0", NULL, "AIFCLK" }, + { "AIF2TX1", NULL, "AIFCLK" }, + { "DSP1RXL", NULL, "SYSDSPCLK" }, { "DSP1RXR", NULL, "SYSDSPCLK" }, { "DSP2RXL", NULL, "SYSDSPCLK" }, @@ -2106,6 +2111,9 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_write(codec, WM8996_FLL_EFS_1, fll_div.lambda); + /* Clear any pending completions (eg, from failed startups) */ + try_wait_for_completion(&wm8996->fll_lock); + snd_soc_update_bits(codec, WM8996_FLL_CONTROL_1, WM8996_FLL_ENA, WM8996_FLL_ENA); diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 56efa0c1c9a9..099614e16651 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -385,14 +385,14 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { err = -ENODEV; - goto fail; + goto fail_free_info; } info->mem = request_mem_region(res->start, resource_size(res), pdev->name); if (!info->mem) { err = -EBUSY; - goto fail; + goto fail_free_info; } info->regs = ioremap(info->mem->start, resource_size(info->mem)); @@ -435,6 +435,7 @@ fail_unmap_mem: iounmap(info->regs); fail_release_mem: release_mem_region(info->mem->start, resource_size(info->mem)); +fail_free_info: kfree(info); fail: return err; diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 732208c8c0b4..cb50598338e9 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -879,10 +879,12 @@ static struct device_node *find_ssi_node(struct device_node *dma_channel_np) * assume that device_node pointers are a valid comparison. */ np = of_parse_phandle(ssi_np, "fsl,playback-dma", 0); + of_node_put(np); if (np == dma_channel_np) return ssi_np; np = of_parse_phandle(ssi_np, "fsl,capture-dma", 0); + of_node_put(np); if (np == dma_channel_np) return ssi_np; } diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index fd0dc46afc34..5c6c2457386e 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -369,7 +369,7 @@ static struct snd_soc_platform_driver mpc5200_audio_dma_platform = { .pcm_free = &psc_dma_free, }; -static int mpc5200_hpcd_probe(struct of_device *op) +static int mpc5200_hpcd_probe(struct platform_device *op) { phys_addr_t fifo; struct psc_dma *psc_dma; @@ -487,7 +487,7 @@ out_unmap: return ret; } -static int mpc5200_hpcd_remove(struct of_device *op) +static int mpc5200_hpcd_remove(struct platform_device *op) { struct psc_dma *psc_dma = dev_get_drvdata(&op->dev); @@ -519,7 +519,7 @@ MODULE_DEVICE_TABLE(of, mpc5200_hpcd_match); static struct platform_driver mpc5200_hpcd_of_driver = { .probe = mpc5200_hpcd_probe, .remove = mpc5200_hpcd_remove, - .dev = { + .driver = { .owner = THIS_MODULE, .name = "mpc5200-pcm-audio", .of_match_table = mpc5200_hpcd_match, diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index a19297959587..358f0baaf71b 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -345,8 +345,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) } machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); - if (!machine_data) - return -ENOMEM; + if (!machine_data) { + ret = -ENOMEM; + goto error_alloc; + } machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev); machine_data->dai[0].ops = &mpc8610_hpcd_ops; @@ -494,7 +496,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) ret = platform_device_add(sound_device); if (ret) { dev_err(&pdev->dev, "platform device add failed\n"); - goto error; + goto error_sound; } dev_set_drvdata(&pdev->dev, sound_device); @@ -502,14 +504,12 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) return 0; +error_sound: + platform_device_unregister(sound_device); error: - of_node_put(codec_np); - - if (sound_device) - platform_device_unregister(sound_device); - kfree(machine_data); - +error_alloc: + of_node_put(codec_np); return ret; } diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 8fa4d5f8eda1..fcb862eb0c73 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -297,8 +297,10 @@ static int get_dma_channel(struct device_node *ssi_np, * dai->platform name should already point to an allocated buffer. */ ret = of_address_to_resource(dma_channel_np, 0, &res); - if (ret) + if (ret) { + of_node_put(dma_channel_np); return ret; + } snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", (unsigned long long) res.start, dma_channel_np->name); diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 309c59e6fb6c..7945625e0e08 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -240,7 +240,6 @@ static int ssi_irq = 0; static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) { - struct snd_card *card = rtd->card->snd_card; struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int ret; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index a33fc51f363b..d0bcf3fcea01 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -424,7 +424,7 @@ static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev) if (!priv->mem) { dev_err(&pdev->dev, "request_mem_region failed\n"); err = -EBUSY; - goto error; + goto err_alloc; } priv->io = ioremap(priv->mem->start, SZ_16K); diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 30fe0d0efe1c..0aa475f92efa 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -514,7 +514,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) } /* Set codec bias level */ - ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY); + ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY); /* Add hook switch - can be used to control the codec from userspace * even if line discipline fails */ @@ -649,7 +649,9 @@ static void __exit ams_delta_module_exit(void) ams_delta_hook_switch_gpios); /* Keep modem power on */ - ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY); + ams_delta_set_bias_level(&ams_delta_audio_card, + &ams_delta_audio_card.rtd[0].codec->dapm, + SND_SOC_BIAS_STANDBY); platform_device_unregister(cx20442_platform_device); platform_device_unregister(ams_delta_audio_platform_device); diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 83d213bfd3d1..62e292f49313 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jhnikula@gmail.com> + * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -402,6 +402,6 @@ static void __exit n810_soc_exit(void) module_init(n810_soc_init); module_exit(n810_soc_exit); -MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>"); +MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>"); MODULE_DESCRIPTION("ALSA SoC Nokia N810"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 07b772357244..ebcc2d4d2b18 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jhnikula@gmail.com> + * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com> * Peter Ujfalusi <peter.ujfalusi@ti.com> * * This program is free software; you can redistribute it and/or @@ -780,6 +780,6 @@ static void __exit snd_omap_mcbsp_exit(void) } module_exit(snd_omap_mcbsp_exit); -MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>"); +MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>"); MODULE_DESCRIPTION("OMAP I2S SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index 9a7dedd6f5a9..65cde9d3807b 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jhnikula@gmail.com> + * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com> * Peter Ujfalusi <peter.ujfalusi@ti.com> * * This program is free software; you can redistribute it and/or diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index b2f5751edae3..9b5c88ac35b9 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jhnikula@gmail.com> + * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com> * Peter Ujfalusi <peter.ujfalusi@ti.com> * * This program is free software; you can redistribute it and/or @@ -436,6 +436,6 @@ static void __exit snd_omap_pcm_exit(void) } module_exit(snd_omap_pcm_exit); -MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>"); +MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>"); MODULE_DESCRIPTION("OMAP PCM DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index a0ed1dbb52d6..f95fe3064172 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jhnikula@gmail.com> + * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com> * Peter Ujfalusi <peter.ujfalusi@ti.com> * * This program is free software; you can redistribute it and/or diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 0aae998b6540..893300a53bab 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -5,7 +5,7 @@ * * Contact: Peter Ujfalusi <peter.ujfalusi@ti.com> * Eduardo Valentin <eduardo.valentin@nokia.com> - * Jarkko Nikula <jhnikula@gmail.com> + * Jarkko Nikula <jarkko.nikula@bitmer.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index b99091fc34eb..65f980ef2870 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -185,6 +185,7 @@ config SND_SOC_SPEYSIDE select SND_SAMSUNG_I2S select SND_SOC_WM8996 select SND_SOC_WM9081 + select SND_SOC_WM1250_EV1 config SND_SOC_SPEYSIDE_WM8962 tristate "Audio support for Wolfson Speyside with WM8962" diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 9eb3b12eb72f..8509d3c4366e 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -1,5 +1,6 @@ # S3c24XX Platform Support snd-soc-s3c24xx-objs := dma.o +snd-soc-idma-objs := idma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-ac97-objs := ac97.o @@ -16,6 +17,7 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o obj-$(CONFIG_SND_SAMSUNG_SPDIF) += snd-soc-samsung-spdif.o obj-$(CONFIG_SND_SAMSUNG_PCM) += snd-soc-pcm.o obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-i2s.o +obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-idma.o # S3C24XX Machine Support snd-soc-jive-wm8750-objs := jive_wm8750.o diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 241f55d00660..c6c65892294e 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -13,6 +13,7 @@ * */ +#include <linux/types.h> #include <linux/gpio.h> #include <sound/soc.h> diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c new file mode 100644 index 000000000000..ebde0740ab19 --- /dev/null +++ b/sound/soc/samsung/idma.c @@ -0,0 +1,453 @@ +/* + * sound/soc/samsung/idma.c + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd. + * http://www.samsung.com + * + * I2S0's Internal DMA driver + * + * 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 the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "i2s.h" +#include "idma.h" +#include "dma.h" +#include "i2s-regs.h" + +#define ST_RUNNING (1<<0) +#define ST_OPENED (1<<1) + +static const struct snd_pcm_hardware idma_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_U24_LE | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = MAX_IDMA_BUFFER, + .period_bytes_min = 128, + .period_bytes_max = MAX_IDMA_PERIOD, + .periods_min = 1, + .periods_max = 2, +}; + +struct idma_ctrl { + spinlock_t lock; + int state; + dma_addr_t start; + dma_addr_t pos; + dma_addr_t end; + dma_addr_t period; + dma_addr_t periodsz; + void *token; + void (*cb)(void *dt, int bytes_xfer); +}; + +static struct idma_info { + spinlock_t lock; + void __iomem *regs; + dma_addr_t lp_tx_addr; +} idma; + +static void idma_getpos(dma_addr_t *src) +{ + *src = idma.lp_tx_addr + + (readl(idma.regs + I2STRNCNT) & 0xffffff) * 4; +} + +static int idma_enqueue(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = substream->runtime->private_data; + u32 val; + + spin_lock(&prtd->lock); + prtd->token = (void *) substream; + spin_unlock(&prtd->lock); + + /* Internal DMA Level0 Interrupt Address */ + val = idma.lp_tx_addr + prtd->periodsz; + writel(val, idma.regs + I2SLVL0ADDR); + + /* Start address0 of I2S internal DMA operation. */ + val = idma.lp_tx_addr; + writel(val, idma.regs + I2SSTR0); + + /* + * Transfer block size for I2S internal DMA. + * Should decide transfer size before start dma operation + */ + val = readl(idma.regs + I2SSIZE); + val &= ~(I2SSIZE_TRNMSK << I2SSIZE_SHIFT); + val |= (((runtime->dma_bytes >> 2) & + I2SSIZE_TRNMSK) << I2SSIZE_SHIFT); + writel(val, idma.regs + I2SSIZE); + + val = readl(idma.regs + I2SAHB); + val |= AHB_INTENLVL0; + writel(val, idma.regs + I2SAHB); + + return 0; +} + +static void idma_setcallbk(struct snd_pcm_substream *substream, + void (*cb)(void *, int)) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + + spin_lock(&prtd->lock); + prtd->cb = cb; + spin_unlock(&prtd->lock); +} + +static void idma_control(int op) +{ + u32 val = readl(idma.regs + I2SAHB); + + spin_lock(&idma.lock); + + switch (op) { + case LPAM_DMA_START: + val |= (AHB_INTENLVL0 | AHB_DMAEN); + break; + case LPAM_DMA_STOP: + val &= ~(AHB_INTENLVL0 | AHB_DMAEN); + break; + default: + spin_unlock(&idma.lock); + return; + } + + writel(val, idma.regs + I2SAHB); + spin_unlock(&idma.lock); +} + +static void idma_done(void *id, int bytes_xfer) +{ + struct snd_pcm_substream *substream = id; + struct idma_ctrl *prtd = substream->runtime->private_data; + + if (prtd && (prtd->state & ST_RUNNING)) + snd_pcm_period_elapsed(substream); +} + +static int idma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = substream->runtime->private_data; + u32 mod = readl(idma.regs + I2SMOD); + u32 ahb = readl(idma.regs + I2SAHB); + + ahb |= (AHB_DMARLD | AHB_INTMASK); + mod |= MOD_TXS_IDMA; + writel(ahb, idma.regs + I2SAHB); + writel(mod, idma.regs + I2SMOD); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->start = prtd->pos = runtime->dma_addr; + prtd->period = params_periods(params); + prtd->periodsz = params_period_bytes(params); + prtd->end = runtime->dma_addr + runtime->dma_bytes; + + idma_setcallbk(substream, idma_done); + + return 0; +} + +static int idma_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + + return 0; +} + +static int idma_prepare(struct snd_pcm_substream *substream) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + + prtd->pos = prtd->start; + + /* flush the DMA channel */ + idma_control(LPAM_DMA_STOP); + idma_enqueue(substream); + + return 0; +} + +static int idma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + int ret = 0; + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + prtd->state |= ST_RUNNING; + idma_control(LPAM_DMA_START); + break; + + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + prtd->state &= ~ST_RUNNING; + idma_control(LPAM_DMA_STOP); + break; + + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static snd_pcm_uframes_t + idma_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = runtime->private_data; + dma_addr_t src; + unsigned long res; + + spin_lock(&prtd->lock); + + idma_getpos(&src); + res = src - prtd->start; + + spin_unlock(&prtd->lock); + + return bytes_to_frames(substream->runtime, res); +} + +static int idma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long size, offset; + int ret; + + /* From snd_pcm_lib_mmap_iomem */ + vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + vma->vm_flags |= VM_IO; + size = vma->vm_end - vma->vm_start; + offset = vma->vm_pgoff << PAGE_SHIFT; + ret = io_remap_pfn_range(vma, vma->vm_start, + (runtime->dma_addr + offset) >> PAGE_SHIFT, + size, vma->vm_page_prot); + + return ret; +} + +static irqreturn_t iis_irq(int irqno, void *dev_id) +{ + struct idma_ctrl *prtd = (struct idma_ctrl *)dev_id; + u32 iiscon, iisahb, val, addr; + + iisahb = readl(idma.regs + I2SAHB); + iiscon = readl(idma.regs + I2SCON); + + val = (iisahb & AHB_LVL0INT) ? AHB_CLRLVL0INT : 0; + + if (val) { + iisahb |= val; + writel(iisahb, idma.regs + I2SAHB); + + addr = readl(idma.regs + I2SLVL0ADDR) - idma.lp_tx_addr; + addr += prtd->periodsz; + addr %= (prtd->end - prtd->start); + addr += idma.lp_tx_addr; + + writel(addr, idma.regs + I2SLVL0ADDR); + + if (prtd->cb) + prtd->cb(prtd->token, prtd->period); + } + + return IRQ_HANDLED; +} + +static int idma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd; + int ret; + + snd_soc_set_runtime_hwparams(substream, &idma_hardware); + + prtd = kzalloc(sizeof(struct idma_ctrl), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + ret = request_irq(IRQ_I2S0, iis_irq, 0, "i2s", prtd); + if (ret < 0) { + pr_err("fail to claim i2s irq , ret = %d\n", ret); + kfree(prtd); + return ret; + } + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + + return 0; +} + +static int idma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = runtime->private_data; + + free_irq(IRQ_I2S0, prtd); + + if (!prtd) + pr_err("idma_close called with prtd == NULL\n"); + + kfree(prtd); + + return 0; +} + +static struct snd_pcm_ops idma_ops = { + .open = idma_open, + .close = idma_close, + .ioctl = snd_pcm_lib_ioctl, + .trigger = idma_trigger, + .pointer = idma_pointer, + .mmap = idma_mmap, + .hw_params = idma_hw_params, + .hw_free = idma_hw_free, + .prepare = idma_prepare, +}; + +static void idma_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + + substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + if (!substream) + return; + + buf = &substream->dma_buffer; + if (!buf->area) + return; + + iounmap(buf->area); + + buf->area = NULL; + buf->addr = 0; +} + +static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + /* Assign PCM buffer pointers */ + buf->dev.type = SNDRV_DMA_TYPE_CONTINUOUS; + buf->addr = idma.lp_tx_addr; + buf->bytes = idma_hardware.buffer_bytes_max; + buf->area = (unsigned char *)ioremap(buf->addr, buf->bytes); + + return 0; +} + +static u64 idma_mask = DMA_BIT_MASK(32); + +static int idma_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &idma_mask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (dai->driver->playback.channels_min) + ret = preallocate_idma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + + return ret; +} + +void idma_reg_addr_init(void *regs, dma_addr_t addr) +{ + spin_lock_init(&idma.lock); + idma.regs = regs; + idma.lp_tx_addr = addr; +} + +struct snd_soc_platform_driver asoc_idma_platform = { + .ops = &idma_ops, + .pcm_new = idma_new, + .pcm_free = idma_free, +}; + +static int __devinit asoc_idma_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &asoc_idma_platform); +} + +static int __devexit asoc_idma_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver asoc_idma_driver = { + .driver = { + .name = "samsung-idma", + .owner = THIS_MODULE, + }, + + .probe = asoc_idma_platform_probe, + .remove = __devexit_p(asoc_idma_platform_remove), +}; + +static int __init asoc_idma_init(void) +{ + return platform_driver_register(&asoc_idma_driver); +} +module_init(asoc_idma_init); + +static void __exit asoc_idma_exit(void) +{ + platform_driver_unregister(&asoc_idma_driver); +} +module_exit(asoc_idma_exit); + +MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar@gmail.com>"); +MODULE_DESCRIPTION("Samsung ASoC IDMA Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/idma.h b/sound/soc/samsung/idma.h new file mode 100644 index 000000000000..48273216166e --- /dev/null +++ b/sound/soc/samsung/idma.h @@ -0,0 +1,26 @@ +/* + * sound/soc/samsung/idma.h + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd + * http://www.samsung.com + * + * 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 the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef __SND_SOC_SAMSUNG_IDMA_H_ +#define __SND_SOC_SAMSUNG_IDMA_H_ + +extern void idma_reg_addr_init(void *regs, dma_addr_t addr); + +/* dma_state */ +#define LPAM_DMA_STOP 0 +#define LPAM_DMA_START 1 + +#define MAX_IDMA_PERIOD (128 * 1024) +#define MAX_IDMA_BUFFER (160 * 1024) + +#endif /* __SND_SOC_SAMSUNG_IDMA_H_ */ diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 3b53ad54bc33..14eb6ea69e7c 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -131,7 +131,7 @@ static struct snd_soc_dai_link jive_dai = { .cpu_dai_name = "s3c2412-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "samsung-audio", - .codec_name = "wm8750-codec.0-0x1a", + .codec_name = "wm8750-codec.0-001a", .init = jive_wm8750_init, .ops = &jive_ops, }; diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 1e574a5d440d..bc8c1676459f 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -17,6 +17,7 @@ * */ +#include <linux/types.h> #include <linux/gpio.h> #include <sound/soc.h> diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 8ac42bf82090..72535f2daaf2 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -23,6 +23,9 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; int ret; + if (dapm->dev != codec_dai->dev) + return 0; + switch (level) { case SND_SOC_BIAS_PREPARE: if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { @@ -37,7 +40,7 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, 44100 * 256, SND_SOC_CLOCK_IN); if (ret < 0) { - pr_err("Failed to set SYSCLK: %d\n"); + pr_err("Failed to set SYSCLK: %d\n", ret); return ret; } } @@ -57,6 +60,9 @@ static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; int ret; + if (dapm->dev != codec_dai->dev) + return 0; + switch (level) { case SND_SOC_BIAS_STANDBY: ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index d9f8aded51f3..20b7f3b003a3 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -203,14 +203,14 @@ static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec) rbnode = rb_entry(node, struct snd_soc_rbtree_node, node); for (i = 0; i < rbnode->blklen; ++i) { regtmp = rbnode->base_reg + i; - WARN_ON(codec->writable_register && - codec->writable_register(codec, regtmp)); val = snd_soc_rbtree_get_register(rbnode, i); def = snd_soc_get_cache_val(codec->reg_def_copy, i, rbnode->word_size); if (val == def) continue; + WARN_ON(!snd_soc_codec_writable_register(codec, regtmp)); + codec->cache_bypass = 1; ret = snd_soc_write(codec, regtmp, val); codec->cache_bypass = 0; @@ -563,8 +563,7 @@ static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec) lzo_blocks = codec->reg_cache; for_each_set_bit(i, lzo_blocks[0]->sync_bmp, lzo_blocks[0]->sync_bmp_nbits) { - WARN_ON(codec->writable_register && - codec->writable_register(codec, i)); + WARN_ON(!snd_soc_codec_writable_register(codec, i)); ret = snd_soc_cache_read(codec, i, &val); if (ret) return ret; @@ -823,8 +822,6 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) codec_drv = codec->driver; for (i = 0; i < codec_drv->reg_cache_size; ++i) { - WARN_ON(codec->writable_register && - codec->writable_register(codec, i)); ret = snd_soc_cache_read(codec, i, &val); if (ret) return ret; @@ -832,6 +829,9 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) if (snd_soc_get_cache_val(codec->reg_def_copy, i, codec_drv->reg_word_size) == val) continue; + + WARN_ON(!snd_soc_codec_writable_register(codec, i)); + ret = snd_soc_write(codec, i, val); if (ret) return ret; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 83ad8ca27490..d2ef014af215 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1633,7 +1633,7 @@ int snd_soc_codec_readable_register(struct snd_soc_codec *codec, if (codec->readable_register) return codec->readable_register(codec, reg); else - return 0; + return 1; } EXPORT_SYMBOL_GPL(snd_soc_codec_readable_register); @@ -1651,7 +1651,7 @@ int snd_soc_codec_writable_register(struct snd_soc_codec *codec, if (codec->writable_register) return codec->writable_register(codec, reg); else - return 0; + return 1; } EXPORT_SYMBOL_GPL(snd_soc_codec_writable_register); @@ -1913,7 +1913,7 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, if (prefix) { name_len = strlen(long_name) + strlen(prefix) + 2; - name = kmalloc(name_len, GFP_ATOMIC); + name = kmalloc(name_len, GFP_KERNEL); if (!name) return NULL; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7e15914b3633..d67c637557a7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2763,7 +2763,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend); /** * snd_soc_dapm_free - free dapm resources - * @card: SoC device + * @dapm: DAPM context * * Free all dapm widgets and resources. */ diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index cca490c80589..a62f7dd4ba96 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -205,6 +205,25 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, #define snd_soc_16_8_read_i2c NULL #endif +#if defined(CONFIG_SPI_MASTER) +static unsigned int snd_soc_16_8_read_spi(struct snd_soc_codec *codec, + unsigned int r) +{ + struct spi_device *spi = codec->control_data; + + const u16 reg = cpu_to_be16(r | 0x100); + u8 data; + int ret; + + ret = spi_write_then_read(spi, ®, 2, &data, 1); + if (ret < 0) + return 0; + return data; +} +#else +#define snd_soc_16_8_read_spi NULL +#endif + static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -295,6 +314,7 @@ static struct { int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); + unsigned int (*spi_read)(struct snd_soc_codec *, unsigned int); } io_types[] = { { .addr_bits = 4, .data_bits = 12, @@ -318,6 +338,7 @@ static struct { .addr_bits = 16, .data_bits = 8, .write = snd_soc_16_8_write, .i2c_read = snd_soc_16_8_read_i2c, + .spi_read = snd_soc_16_8_read_spi, }, { .addr_bits = 16, .data_bits = 16, @@ -383,6 +404,8 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, #ifdef CONFIG_SPI_MASTER codec->hw_write = do_spi_write; #endif + if (io_types[i].spi_read) + codec->hw_read = io_types[i].spi_read; codec->control_data = container_of(codec->dev, struct spi_device, diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 7c17b98d5846..fa31d9c2abd8 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -105,7 +105,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) snd_soc_dapm_sync(dapm); - snd_jack_report(jack->jack, status); + snd_jack_report(jack->jack, jack->status); out: mutex_unlock(&codec->mutex); @@ -327,7 +327,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, IRQF_TRIGGER_FALLING, gpios[i].name, &gpios[i]); - if (ret) + if (ret < 0) goto err; if (gpios[i].wake) { diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b5759397afa3..2879c883eebc 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -290,6 +290,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec_dai->active--; codec->active--; + if (!cpu_dai->active && !codec_dai->active) + rtd->rate = 0; + /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. */ diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index ff86e5e3db68..c7cfd96e991e 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -309,9 +309,14 @@ static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream) { - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + + substream = pcm->streams[stream].substream; + if (!substream) + return; + buf = &substream->dma_buffer; if (!buf->area) return; diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index a42e9ac30f28..be27f1d229af 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -56,6 +56,7 @@ #define GPIO_HP_MUTE BIT(1) #define GPIO_INT_MIC_EN BIT(2) #define GPIO_EXT_MIC_EN BIT(3) +#define GPIO_HP_DET BIT(4) struct tegra_wm8903 { struct tegra_asoc_utils_data util_data; @@ -304,6 +305,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack, 1, &tegra_wm8903_hp_jack_gpio); + machine->gpio_requested |= GPIO_HP_DET; } snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, @@ -317,7 +319,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); /* FIXME: Calculate automatically based on DAPM routes? */ - if (!machine_is_harmony() && !machine_is_ventana()) + if (!machine_is_harmony()) snd_soc_dapm_nc_pin(dapm, "IN1L"); if (!machine_is_seaboard() && !machine_is_aebl()) snd_soc_dapm_nc_pin(dapm, "IN1R"); @@ -393,7 +395,7 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); - if (machine_is_harmony() || machine_is_ventana()) { + if (machine_is_harmony()) { card->dapm_routes = harmony_audio_map; card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map); } else if (machine_is_seaboard()) { @@ -429,10 +431,10 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev) struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); struct tegra_wm8903_platform_data *pdata = machine->pdata; - snd_soc_unregister_card(card); - - tegra_asoc_utils_fini(&machine->util_data); - + if (machine->gpio_requested & GPIO_HP_DET) + snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack, + 1, + &tegra_wm8903_hp_jack_gpio); if (machine->gpio_requested & GPIO_EXT_MIC_EN) gpio_free(pdata->gpio_ext_mic_en); if (machine->gpio_requested & GPIO_INT_MIC_EN) @@ -441,6 +443,11 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev) gpio_free(pdata->gpio_hp_mute); if (machine->gpio_requested & GPIO_SPKR_EN) gpio_free(pdata->gpio_spkr_en); + machine->gpio_requested = 0; + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&machine->util_data); kfree(machine); diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index d0d493ca28ae..2cf87f5afed4 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -139,8 +139,12 @@ static void stream_stop(struct snd_usb_caiaqdev *dev) for (i = 0; i < N_URBS; i++) { usb_kill_urb(dev->data_urbs_in[i]); - usb_kill_urb(dev->data_urbs_out[i]); + + if (test_bit(i, &dev->outurb_active_mask)) + usb_kill_urb(dev->data_urbs_out[i]); } + + dev->outurb_active_mask = 0; } static int snd_usb_caiaq_substream_open(struct snd_pcm_substream *substream) @@ -612,8 +616,9 @@ static void read_completed(struct urb *urb) { struct snd_usb_caiaq_cb_info *info = urb->context; struct snd_usb_caiaqdev *dev; - struct urb *out; - int frame, len, send_it = 0, outframe = 0; + struct urb *out = NULL; + int i, frame, len, send_it = 0, outframe = 0; + size_t offset = 0; if (urb->status || !info) return; @@ -623,7 +628,17 @@ static void read_completed(struct urb *urb) if (!dev->streaming) return; - out = dev->data_urbs_out[info->index]; + /* find an unused output urb that is unused */ + for (i = 0; i < N_URBS; i++) + if (test_and_set_bit(i, &dev->outurb_active_mask) == 0) { + out = dev->data_urbs_out[i]; + break; + } + + if (!out) { + log("Unable to find an output urb to use\n"); + goto requeue; + } /* read the recently received packet and send back one which has * the same layout */ @@ -634,7 +649,8 @@ static void read_completed(struct urb *urb) len = urb->iso_frame_desc[outframe].actual_length; out->iso_frame_desc[outframe].length = len; out->iso_frame_desc[outframe].actual_length = 0; - out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame; + out->iso_frame_desc[outframe].offset = offset; + offset += len; if (len > 0) { spin_lock(&dev->spinlock); @@ -650,11 +666,15 @@ static void read_completed(struct urb *urb) } if (send_it) { - out->number_of_packets = FRAMES_PER_URB; + out->number_of_packets = outframe; out->transfer_flags = URB_ISO_ASAP; usb_submit_urb(out, GFP_ATOMIC); + } else { + struct snd_usb_caiaq_cb_info *oinfo = out->context; + clear_bit(oinfo->index, &dev->outurb_active_mask); } +requeue: /* re-submit inbound urb */ for (frame = 0; frame < FRAMES_PER_URB; frame++) { urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame; @@ -676,6 +696,8 @@ static void write_completed(struct urb *urb) dev->output_running = 1; wake_up(&dev->prepare_wait_queue); } + + clear_bit(info->index, &dev->outurb_active_mask); } static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) @@ -827,6 +849,9 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) if (!dev->data_cb_info) return -ENOMEM; + dev->outurb_active_mask = 0; + BUILD_BUG_ON(N_URBS > (sizeof(dev->outurb_active_mask) * 8)); + for (i = 0; i < N_URBS; i++) { dev->data_cb_info[i].dev = dev; dev->data_cb_info[i].index = i; diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index b2b310194ffa..3f9c6339ae90 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -96,6 +96,7 @@ struct snd_usb_caiaqdev { int input_panic, output_panic, warned; char *audio_in_buf, *audio_out_buf; unsigned int samplerates, bpp; + unsigned long outurb_active_mask; struct snd_pcm_substream *sub_playback[MAX_STREAMS]; struct snd_pcm_substream *sub_capture[MAX_STREAMS]; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c04d7c71ac88..cdd19d7fe500 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -152,6 +152,7 @@ static inline void check_mapped_dB(const struct usbmix_name_map *p, if (p && p->dB) { cval->dBmin = p->dB->min; cval->dBmax = p->dB->max; + cval->initialized = 1; } } @@ -1092,7 +1093,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, " Switch" : " Volume"); if (control == UAC_FU_VOLUME) { check_mapped_dB(map, cval); - if (cval->dBmin < cval->dBmax) { + if (cval->dBmin < cval->dBmax || !cval->initialized) { kctl->tlv.c = mixer_vol_tlv; kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ | diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 4d4f86552a23..a42e3ef3832d 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1707,6 +1707,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE(0x0582, 0x0130), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "BOSS", */ + /* .product_name = "MICRO BR-80", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* Guillemot devices */ { |