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author | Greg Kroah-Hartman <gregkh@linuxfoundation.org> | 2013-03-26 09:19:02 -0700 |
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committer | Greg Kroah-Hartman <gregkh@linuxfoundation.org> | 2013-03-26 09:19:02 -0700 |
commit | e58b9a25eeb89ab2ee05cd093f6d7bc2f34acb21 (patch) | |
tree | 40162c796bc60f00d062b37718dc62adc970ac07 /sound | |
parent | a6025a2a861845447adeb7a11c3043039959d3a1 (diff) | |
parent | df8c3dbee9e6f19ddb0ae8e05cdf76eb2d3b7f00 (diff) | |
download | blackbird-op-linux-e58b9a25eeb89ab2ee05cd093f6d7bc2f34acb21.tar.gz blackbird-op-linux-e58b9a25eeb89ab2ee05cd093f6d7bc2f34acb21.zip |
Merge tag 'arizona-extcon-asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/misc into char-misc-next
Mark writes:
ASoC/extcon: arizona: Fix interaction between HPDET and headphone outputs
This patch series covers both ASoC and extcon subsystems and fixes an
interaction between the HPDET function and the headphone outputs - we
really shouldn't run HPDET while the headphone is active. The first
patch is a refactoring to make the extcon side easier.
Diffstat (limited to 'sound')
-rw-r--r-- | sound/core/seq/seq_timer.c | 8 | ||||
-rw-r--r-- | sound/oss/sequencer.c | 6 | ||||
-rw-r--r-- | sound/pci/asihpi/asihpi.c | 3 | ||||
-rw-r--r-- | sound/pci/hda/hda_codec.c | 26 | ||||
-rw-r--r-- | sound/pci/hda/hda_generic.c | 46 | ||||
-rw-r--r-- | sound/pci/hda/hda_intel.c | 132 | ||||
-rw-r--r-- | sound/pci/hda/patch_ca0132.c | 28 | ||||
-rw-r--r-- | sound/pci/hda/patch_cirrus.c | 8 | ||||
-rw-r--r-- | sound/pci/hda/patch_conexant.c | 16 | ||||
-rw-r--r-- | sound/pci/hda/patch_sigmatel.c | 29 | ||||
-rw-r--r-- | sound/soc/codecs/arizona.c | 33 | ||||
-rw-r--r-- | sound/soc/codecs/arizona.h | 3 | ||||
-rw-r--r-- | sound/soc/codecs/wm5102.c | 8 | ||||
-rw-r--r-- | sound/soc/codecs/wm5110.c | 8 | ||||
-rw-r--r-- | sound/usb/card.c | 15 | ||||
-rw-r--r-- | sound/usb/mixer.c | 21 |
16 files changed, 315 insertions, 75 deletions
diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index 160b1bd0cd62..24d44b2f61ac 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -290,10 +290,10 @@ int snd_seq_timer_open(struct snd_seq_queue *q) tid.device = SNDRV_TIMER_GLOBAL_SYSTEM; err = snd_timer_open(&t, str, &tid, q->queue); } - if (err < 0) { - snd_printk(KERN_ERR "seq fatal error: cannot create timer (%i)\n", err); - return err; - } + } + if (err < 0) { + snd_printk(KERN_ERR "seq fatal error: cannot create timer (%i)\n", err); + return err; } t->callback = snd_seq_timer_interrupt; t->callback_data = q; diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index 30bcfe470f83..4ff60a6427d9 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -545,6 +545,9 @@ static void seq_chn_common_event(unsigned char *event_rec) case MIDI_PGM_CHANGE: if (seq_mode == SEQ_2) { + if (chn > 15) + break; + synth_devs[dev]->chn_info[chn].pgm_num = p1; if ((int) dev >= num_synths) synth_devs[dev]->set_instr(dev, chn, p1); @@ -596,6 +599,9 @@ static void seq_chn_common_event(unsigned char *event_rec) case MIDI_PITCH_BEND: if (seq_mode == SEQ_2) { + if (chn > 15) + break; + synth_devs[dev]->chn_info[chn].bender_value = w14; if ((int) dev < num_synths) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 3536b076b529..0aabfedeecba 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -2549,7 +2549,7 @@ static int snd_asihpi_sampleclock_add(struct snd_card_asihpi *asihpi, static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi) { - struct snd_card *card = asihpi->card; + struct snd_card *card; unsigned int idx = 0; unsigned int subindex = 0; int err; @@ -2557,6 +2557,7 @@ static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi) if (snd_BUG_ON(!asihpi)) return -EINVAL; + card = asihpi->card; strcpy(card->mixername, "Asihpi Mixer"); err = diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 97c68dd24ef5..ecdf30eb5879 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -494,7 +494,7 @@ static unsigned int get_num_conns(struct hda_codec *codec, hda_nid_t nid) int snd_hda_get_num_raw_conns(struct hda_codec *codec, hda_nid_t nid) { - return get_num_conns(codec, nid) & AC_CLIST_LENGTH; + return snd_hda_get_raw_connections(codec, nid, NULL, 0); } /** @@ -517,9 +517,6 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, hda_nid_t prev_nid; int null_count = 0; - if (snd_BUG_ON(!conn_list || max_conns <= 0)) - return -EINVAL; - parm = get_num_conns(codec, nid); if (!parm) return 0; @@ -545,7 +542,8 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, AC_VERB_GET_CONNECT_LIST, 0); if (parm == -1 && codec->bus->rirb_error) return -EIO; - conn_list[0] = parm & mask; + if (conn_list) + conn_list[0] = parm & mask; return 1; } @@ -580,14 +578,20 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, continue; } for (n = prev_nid + 1; n <= val; n++) { + if (conn_list) { + if (conns >= max_conns) + return -ENOSPC; + conn_list[conns] = n; + } + conns++; + } + } else { + if (conn_list) { if (conns >= max_conns) return -ENOSPC; - conn_list[conns++] = n; + conn_list[conns] = val; } - } else { - if (conns >= max_conns) - return -ENOSPC; - conn_list[conns++] = val; + conns++; } prev_nid = val; } @@ -3140,7 +3144,7 @@ static unsigned int convert_to_spdif_status(unsigned short val) if (val & AC_DIG1_PROFESSIONAL) sbits |= IEC958_AES0_PROFESSIONAL; if (sbits & IEC958_AES0_PROFESSIONAL) { - if (sbits & AC_DIG1_EMPHASIS) + if (val & AC_DIG1_EMPHASIS) sbits |= IEC958_AES0_PRO_EMPHASIS_5015; } else { if (val & AC_DIG1_EMPHASIS) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 78897d05d80f..43c2ea539561 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -995,6 +995,8 @@ enum { BAD_NO_EXTRA_SURR_DAC = 0x101, /* Primary DAC shared with main surrounds */ BAD_SHARED_SURROUND = 0x100, + /* No independent HP possible */ + BAD_NO_INDEP_HP = 0x40, /* Primary DAC shared with main CLFE */ BAD_SHARED_CLFE = 0x10, /* Primary DAC shared with extra surrounds */ @@ -1392,6 +1394,43 @@ static int check_aamix_out_path(struct hda_codec *codec, int path_idx) return snd_hda_get_path_idx(codec, path); } +/* check whether the independent HP is available with the current config */ +static bool indep_hp_possible(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct nid_path *path; + int i, idx; + + if (cfg->line_out_type == AUTO_PIN_HP_OUT) + idx = spec->out_paths[0]; + else + idx = spec->hp_paths[0]; + path = snd_hda_get_path_from_idx(codec, idx); + if (!path) + return false; + + /* assume no path conflicts unless aamix is involved */ + if (!spec->mixer_nid || !is_nid_contained(path, spec->mixer_nid)) + return true; + + /* check whether output paths contain aamix */ + for (i = 0; i < cfg->line_outs; i++) { + if (spec->out_paths[i] == idx) + break; + path = snd_hda_get_path_from_idx(codec, spec->out_paths[i]); + if (path && is_nid_contained(path, spec->mixer_nid)) + return false; + } + for (i = 0; i < cfg->speaker_outs; i++) { + path = snd_hda_get_path_from_idx(codec, spec->speaker_paths[i]); + if (path && is_nid_contained(path, spec->mixer_nid)) + return false; + } + + return true; +} + /* fill the empty entries in the dac array for speaker/hp with the * shared dac pointed by the paths */ @@ -1545,6 +1584,9 @@ static int fill_and_eval_dacs(struct hda_codec *codec, badness += BAD_MULTI_IO; } + if (spec->indep_hp && !indep_hp_possible(codec)) + badness += BAD_NO_INDEP_HP; + /* re-fill the shared DAC for speaker / headphone */ if (cfg->line_out_type != AUTO_PIN_HP_OUT) refill_shared_dacs(codec, cfg->hp_outs, @@ -1758,6 +1800,10 @@ static int parse_output_paths(struct hda_codec *codec) cfg->speaker_pins, val); } + /* clear indep_hp flag if not available */ + if (spec->indep_hp && !indep_hp_possible(codec)) + spec->indep_hp = 0; + kfree(best_cfg); return 0; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4cea6bb6fade..418bfc0eb0a3 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -415,6 +415,8 @@ struct azx_dev { unsigned int opened :1; unsigned int running :1; unsigned int irq_pending :1; + unsigned int prepared:1; + unsigned int locked:1; /* * For VIA: * A flag to ensure DMA position is 0 @@ -426,8 +428,25 @@ struct azx_dev { struct timecounter azx_tc; struct cyclecounter azx_cc; + +#ifdef CONFIG_SND_HDA_DSP_LOADER + struct mutex dsp_mutex; +#endif }; +/* DSP lock helpers */ +#ifdef CONFIG_SND_HDA_DSP_LOADER +#define dsp_lock_init(dev) mutex_init(&(dev)->dsp_mutex) +#define dsp_lock(dev) mutex_lock(&(dev)->dsp_mutex) +#define dsp_unlock(dev) mutex_unlock(&(dev)->dsp_mutex) +#define dsp_is_locked(dev) ((dev)->locked) +#else +#define dsp_lock_init(dev) do {} while (0) +#define dsp_lock(dev) do {} while (0) +#define dsp_unlock(dev) do {} while (0) +#define dsp_is_locked(dev) 0 +#endif + /* CORB/RIRB */ struct azx_rb { u32 *buf; /* CORB/RIRB buffer @@ -527,6 +546,10 @@ struct azx { /* card list (for power_save trigger) */ struct list_head list; + +#ifdef CONFIG_SND_HDA_DSP_LOADER + struct azx_dev saved_azx_dev; +#endif }; #define CREATE_TRACE_POINTS @@ -1793,15 +1816,25 @@ azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream) dev = chip->capture_index_offset; nums = chip->capture_streams; } - for (i = 0; i < nums; i++, dev++) - if (!chip->azx_dev[dev].opened) { - res = &chip->azx_dev[dev]; - if (res->assigned_key == key) - break; + for (i = 0; i < nums; i++, dev++) { + struct azx_dev *azx_dev = &chip->azx_dev[dev]; + dsp_lock(azx_dev); + if (!azx_dev->opened && !dsp_is_locked(azx_dev)) { + res = azx_dev; + if (res->assigned_key == key) { + res->opened = 1; + res->assigned_key = key; + dsp_unlock(azx_dev); + return azx_dev; + } } + dsp_unlock(azx_dev); + } if (res) { + dsp_lock(res); res->opened = 1; res->assigned_key = key; + dsp_unlock(res); } return res; } @@ -2009,6 +2042,12 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream, struct azx_dev *azx_dev = get_azx_dev(substream); int ret; + dsp_lock(azx_dev); + if (dsp_is_locked(azx_dev)) { + ret = -EBUSY; + goto unlock; + } + mark_runtime_wc(chip, azx_dev, substream, false); azx_dev->bufsize = 0; azx_dev->period_bytes = 0; @@ -2016,8 +2055,10 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream, ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (ret < 0) - return ret; + goto unlock; mark_runtime_wc(chip, azx_dev, substream, true); + unlock: + dsp_unlock(azx_dev); return ret; } @@ -2029,16 +2070,21 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; /* reset BDL address */ - azx_sd_writel(azx_dev, SD_BDLPL, 0); - azx_sd_writel(azx_dev, SD_BDLPU, 0); - azx_sd_writel(azx_dev, SD_CTL, 0); - azx_dev->bufsize = 0; - azx_dev->period_bytes = 0; - azx_dev->format_val = 0; + dsp_lock(azx_dev); + if (!dsp_is_locked(azx_dev)) { + azx_sd_writel(azx_dev, SD_BDLPL, 0); + azx_sd_writel(azx_dev, SD_BDLPU, 0); + azx_sd_writel(azx_dev, SD_CTL, 0); + azx_dev->bufsize = 0; + azx_dev->period_bytes = 0; + azx_dev->format_val = 0; + } snd_hda_codec_cleanup(apcm->codec, hinfo, substream); mark_runtime_wc(chip, azx_dev, substream, false); + azx_dev->prepared = 0; + dsp_unlock(azx_dev); return snd_pcm_lib_free_pages(substream); } @@ -2055,6 +2101,12 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) snd_hda_spdif_out_of_nid(apcm->codec, hinfo->nid); unsigned short ctls = spdif ? spdif->ctls : 0; + dsp_lock(azx_dev); + if (dsp_is_locked(azx_dev)) { + err = -EBUSY; + goto unlock; + } + azx_stream_reset(chip, azx_dev); format_val = snd_hda_calc_stream_format(runtime->rate, runtime->channels, @@ -2065,7 +2117,8 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) snd_printk(KERN_ERR SFX "%s: invalid format_val, rate=%d, ch=%d, format=%d\n", pci_name(chip->pci), runtime->rate, runtime->channels, runtime->format); - return -EINVAL; + err = -EINVAL; + goto unlock; } bufsize = snd_pcm_lib_buffer_bytes(substream); @@ -2084,7 +2137,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) azx_dev->no_period_wakeup = runtime->no_period_wakeup; err = azx_setup_periods(chip, substream, azx_dev); if (err < 0) - return err; + goto unlock; } /* wallclk has 24Mhz clock source */ @@ -2101,8 +2154,14 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) if ((chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) && stream_tag > chip->capture_streams) stream_tag -= chip->capture_streams; - return snd_hda_codec_prepare(apcm->codec, hinfo, stream_tag, + err = snd_hda_codec_prepare(apcm->codec, hinfo, stream_tag, azx_dev->format_val, substream); + + unlock: + if (!err) + azx_dev->prepared = 1; + dsp_unlock(azx_dev); + return err; } static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) @@ -2117,6 +2176,9 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) azx_dev = get_azx_dev(substream); trace_azx_pcm_trigger(chip, azx_dev, cmd); + if (dsp_is_locked(azx_dev) || !azx_dev->prepared) + return -EPIPE; + switch (cmd) { case SNDRV_PCM_TRIGGER_START: rstart = 1; @@ -2621,17 +2683,27 @@ static int azx_load_dsp_prepare(struct hda_bus *bus, unsigned int format, struct azx_dev *azx_dev; int err; - if (snd_hda_lock_devices(bus)) - return -EBUSY; + azx_dev = azx_get_dsp_loader_dev(chip); + + dsp_lock(azx_dev); + spin_lock_irq(&chip->reg_lock); + if (azx_dev->running || azx_dev->locked) { + spin_unlock_irq(&chip->reg_lock); + err = -EBUSY; + goto unlock; + } + azx_dev->prepared = 0; + chip->saved_azx_dev = *azx_dev; + azx_dev->locked = 1; + spin_unlock_irq(&chip->reg_lock); err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(chip->pci), byte_size, bufp); if (err < 0) - goto unlock; + goto err_alloc; mark_pages_wc(chip, bufp, true); - azx_dev = azx_get_dsp_loader_dev(chip); azx_dev->bufsize = byte_size; azx_dev->period_bytes = byte_size; azx_dev->format_val = format; @@ -2649,13 +2721,20 @@ static int azx_load_dsp_prepare(struct hda_bus *bus, unsigned int format, goto error; azx_setup_controller(chip, azx_dev); + dsp_unlock(azx_dev); return azx_dev->stream_tag; error: mark_pages_wc(chip, bufp, false); snd_dma_free_pages(bufp); -unlock: - snd_hda_unlock_devices(bus); + err_alloc: + spin_lock_irq(&chip->reg_lock); + if (azx_dev->opened) + *azx_dev = chip->saved_azx_dev; + azx_dev->locked = 0; + spin_unlock_irq(&chip->reg_lock); + unlock: + dsp_unlock(azx_dev); return err; } @@ -2677,9 +2756,10 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus, struct azx *chip = bus->private_data; struct azx_dev *azx_dev = azx_get_dsp_loader_dev(chip); - if (!dmab->area) + if (!dmab->area || !azx_dev->locked) return; + dsp_lock(azx_dev); /* reset BDL address */ azx_sd_writel(azx_dev, SD_BDLPL, 0); azx_sd_writel(azx_dev, SD_BDLPU, 0); @@ -2692,7 +2772,12 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus, snd_dma_free_pages(dmab); dmab->area = NULL; - snd_hda_unlock_devices(bus); + spin_lock_irq(&chip->reg_lock); + if (azx_dev->opened) + *azx_dev = chip->saved_azx_dev; + azx_dev->locked = 0; + spin_unlock_irq(&chip->reg_lock); + dsp_unlock(azx_dev); } #endif /* CONFIG_SND_HDA_DSP_LOADER */ @@ -3481,6 +3566,7 @@ static int azx_first_init(struct azx *chip) } for (i = 0; i < chip->num_streams; i++) { + dsp_lock_init(&chip->azx_dev[i]); /* allocate memory for the BDL for each stream */ err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index eefc4563b2f9..0792b5725f9c 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -3239,7 +3239,7 @@ static int ca0132_set_vipsource(struct hda_codec *codec, int val) struct ca0132_spec *spec = codec->spec; unsigned int tmp; - if (!dspload_is_loaded(codec)) + if (spec->dsp_state != DSP_DOWNLOADED) return 0; /* if CrystalVoice if off, vipsource should be 0 */ @@ -4267,11 +4267,12 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec) */ static void ca0132_setup_defaults(struct hda_codec *codec) { + struct ca0132_spec *spec = codec->spec; unsigned int tmp; int num_fx; int idx, i; - if (!dspload_is_loaded(codec)) + if (spec->dsp_state != DSP_DOWNLOADED) return; /* out, in effects + voicefx */ @@ -4351,12 +4352,16 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec) return false; dsp_os_image = (struct dsp_image_seg *)(fw_entry->data); - dspload_image(codec, dsp_os_image, 0, 0, true, 0); + if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) { + pr_err("ca0132 dspload_image failed.\n"); + goto exit_download; + } + dsp_loaded = dspload_wait_loaded(codec); +exit_download: release_firmware(fw_entry); - return dsp_loaded; } @@ -4367,16 +4372,13 @@ static void ca0132_download_dsp(struct hda_codec *codec) #ifndef CONFIG_SND_HDA_CODEC_CA0132_DSP return; /* NOP */ #endif - spec->dsp_state = DSP_DOWNLOAD_INIT; - if (spec->dsp_state == DSP_DOWNLOAD_INIT) { - chipio_enable_clocks(codec); - spec->dsp_state = DSP_DOWNLOADING; - if (!ca0132_download_dsp_images(codec)) - spec->dsp_state = DSP_DOWNLOAD_FAILED; - else - spec->dsp_state = DSP_DOWNLOADED; - } + chipio_enable_clocks(codec); + spec->dsp_state = DSP_DOWNLOADING; + if (!ca0132_download_dsp_images(codec)) + spec->dsp_state = DSP_DOWNLOAD_FAILED; + else + spec->dsp_state = DSP_DOWNLOADED; if (spec->dsp_state == DSP_DOWNLOADED) ca0132_set_dsp_msr(codec, true); diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 72ebb8a36b13..0d9c58f13560 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -168,10 +168,10 @@ static void cs_automute(struct hda_codec *codec) snd_hda_gen_update_outputs(codec); if (spec->gpio_eapd_hp) { - unsigned int gpio = spec->gen.hp_jack_present ? + spec->gpio_data = spec->gen.hp_jack_present ? spec->gpio_eapd_hp : spec->gpio_eapd_speaker; snd_hda_codec_write(codec, 0x01, 0, - AC_VERB_SET_GPIO_DATA, gpio); + AC_VERB_SET_GPIO_DATA, spec->gpio_data); } } @@ -506,6 +506,8 @@ static int patch_cs420x(struct hda_codec *codec) if (!spec) return -ENOMEM; + spec->gen.automute_hook = cs_automute; + snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl, cs420x_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -893,6 +895,8 @@ static int patch_cs4210(struct hda_codec *codec) if (!spec) return -ENOMEM; + spec->gen.automute_hook = cs_automute; + snd_hda_pick_fixup(codec, cs421x_models, cs421x_fixup_tbl, cs421x_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 941bf6c766ec..2a89d1eefeb6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1142,7 +1142,7 @@ static int patch_cxt5045(struct hda_codec *codec) } if (spec->beep_amp) - snd_hda_attach_beep_device(codec, spec->beep_amp); + snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); return 0; } @@ -1921,7 +1921,7 @@ static int patch_cxt5051(struct hda_codec *codec) } if (spec->beep_amp) - snd_hda_attach_beep_device(codec, spec->beep_amp); + snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); return 0; } @@ -3099,7 +3099,7 @@ static int patch_cxt5066(struct hda_codec *codec) } if (spec->beep_amp) - snd_hda_attach_beep_device(codec, spec->beep_amp); + snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); return 0; } @@ -3191,11 +3191,17 @@ static int cx_auto_build_controls(struct hda_codec *codec) return 0; } +static void cx_auto_free(struct hda_codec *codec) +{ + snd_hda_detach_beep_device(codec); + snd_hda_gen_free(codec); +} + static const struct hda_codec_ops cx_auto_patch_ops = { .build_controls = cx_auto_build_controls, .build_pcms = snd_hda_gen_build_pcms, .init = snd_hda_gen_init, - .free = snd_hda_gen_free, + .free = cx_auto_free, .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM .check_power_status = snd_hda_gen_check_power_status, @@ -3391,7 +3397,7 @@ static int patch_conexant_auto(struct hda_codec *codec) codec->patch_ops = cx_auto_patch_ops; if (spec->beep_amp) - snd_hda_attach_beep_device(codec, spec->beep_amp); + snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); /* Some laptops with Conexant chips show stalls in S3 resume, * which falls into the single-cmd mode. diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 83d5335ac348..dafe04ae8c72 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -815,6 +815,29 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) return 0; } +/* check whether a built-in speaker is included in parsed pins */ +static bool has_builtin_speaker(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + hda_nid_t *nid_pin; + int nids, i; + + if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) { + nid_pin = spec->gen.autocfg.line_out_pins; + nids = spec->gen.autocfg.line_outs; + } else { + nid_pin = spec->gen.autocfg.speaker_pins; + nids = spec->gen.autocfg.speaker_outs; + } + + for (i = 0; i < nids; i++) { + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid_pin[i]); + if (snd_hda_get_input_pin_attr(def_conf) == INPUT_PIN_ATTR_INT) + return true; + } + return false; +} + /* * PC beep controls */ @@ -3890,6 +3913,12 @@ static int patch_stac92hd73xx(struct hda_codec *codec) return err; } + /* Don't GPIO-mute speakers if there are no internal speakers, because + * the GPIO might be necessary for Headphone + */ + if (spec->eapd_switch && !has_builtin_speaker(codec)) + spec->eapd_switch = 0; + codec->proc_widget_hook = stac92hd7x_proc_hook; snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index ac948a671ea6..e7d34711412c 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -364,6 +364,39 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(arizona_out_ev); +int arizona_hp_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec); + unsigned int mask = 1 << w->shift; + unsigned int val; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + val = mask; + break; + case SND_SOC_DAPM_PRE_PMD: + val = 0; + break; + default: + return -EINVAL; + } + + /* Store the desired state for the HP outputs */ + priv->arizona->hp_ena &= ~mask; + priv->arizona->hp_ena |= val; + + /* Force off if HPDET magic is active */ + if (priv->arizona->hpdet_magic) + val = 0; + + snd_soc_update_bits(w->codec, ARIZONA_OUTPUT_ENABLES_1, mask, val); + + return arizona_out_ev(w, kcontrol, event); +} +EXPORT_SYMBOL_GPL(arizona_hp_ev); + static unsigned int arizona_sysclk_48k_rates[] = { 6144000, 12288000, diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 116372c91f5d..13dd2916b721 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -184,6 +184,9 @@ extern int arizona_in_ev(struct snd_soc_dapm_widget *w, extern int arizona_out_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +extern int arizona_hp_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event); extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index b82bbf584146..2657aad3f8b1 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1131,11 +1131,11 @@ ARIZONA_DSP_WIDGETS(DSP1, "DSP1"), SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux), -SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1, - ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, +SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, + ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA_E("OUT1R", ARIZONA_OUTPUT_ENABLES_1, - ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, +SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM, + ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index cdeb301da1f6..7841b42a819c 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -551,11 +551,11 @@ SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0, SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0), -SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1, - ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, +SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, + ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA_E("OUT1R", ARIZONA_OUTPUT_ENABLES_1, - ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, +SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM, + ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, diff --git a/sound/usb/card.c b/sound/usb/card.c index 803953a9bff3..2da8ad75fd96 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -244,6 +244,21 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) usb_ifnum_to_if(dev, ctrlif)->intf_assoc; if (!assoc) { + /* + * Firmware writers cannot count to three. So to find + * the IAD on the NuForce UDH-100, also check the next + * interface. + */ + struct usb_interface *iface = + usb_ifnum_to_if(dev, ctrlif + 1); + if (iface && + iface->intf_assoc && + iface->intf_assoc->bFunctionClass == USB_CLASS_AUDIO && + iface->intf_assoc->bFunctionProtocol == UAC_VERSION_2) + assoc = iface->intf_assoc; + } + + if (!assoc) { snd_printk(KERN_ERR "Audio class v2 interfaces need an interface association\n"); return -EINVAL; } diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 638e7f738018..ca4739c3f650 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -715,8 +715,9 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ case UAC2_CLOCK_SELECTOR: { struct uac_selector_unit_descriptor *d = p1; /* call recursively to retrieve the channel info */ - if (check_input_term(state, d->baSourceID[0], term) < 0) - return -ENODEV; + err = check_input_term(state, d->baSourceID[0], term); + if (err < 0) + return err; term->type = d->bDescriptorSubtype << 16; /* virtual type */ term->id = id; term->name = uac_selector_unit_iSelector(d); @@ -725,7 +726,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ case UAC1_PROCESSING_UNIT: case UAC1_EXTENSION_UNIT: /* UAC2_PROCESSING_UNIT_V2 */ - /* UAC2_EFFECT_UNIT */ { + /* UAC2_EFFECT_UNIT */ + case UAC2_EXTENSION_UNIT_V2: { struct uac_processing_unit_descriptor *d = p1; if (state->mixer->protocol == UAC_VERSION_2 && @@ -1356,8 +1358,9 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void return err; /* determine the input source type and name */ - if (check_input_term(state, hdr->bSourceID, &iterm) < 0) - return -EINVAL; + err = check_input_term(state, hdr->bSourceID, &iterm); + if (err < 0) + return err; master_bits = snd_usb_combine_bytes(bmaControls, csize); /* master configuration quirks */ @@ -2052,6 +2055,8 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) return parse_audio_extension_unit(state, unitid, p1); else /* UAC_VERSION_2 */ return parse_audio_processing_unit(state, unitid, p1); + case UAC2_EXTENSION_UNIT_V2: + return parse_audio_extension_unit(state, unitid, p1); default: snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]); return -EINVAL; @@ -2118,7 +2123,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) state.oterm.type = le16_to_cpu(desc->wTerminalType); state.oterm.name = desc->iTerminal; err = parse_audio_unit(&state, desc->bSourceID); - if (err < 0) + if (err < 0 && err != -EINVAL) return err; } else { /* UAC_VERSION_2 */ struct uac2_output_terminal_descriptor *desc = p; @@ -2130,12 +2135,12 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) state.oterm.type = le16_to_cpu(desc->wTerminalType); state.oterm.name = desc->iTerminal; err = parse_audio_unit(&state, desc->bSourceID); - if (err < 0) + if (err < 0 && err != -EINVAL) return err; /* for UAC2, use the same approach to also add the clock selectors */ err = parse_audio_unit(&state, desc->bCSourceID); - if (err < 0) + if (err < 0 && err != -EINVAL) return err; } } |