diff options
Diffstat (limited to 'sound')
-rw-r--r-- | sound/core/seq/oss/seq_oss_event.c | 14 | ||||
-rw-r--r-- | sound/core/seq/seq_timer.c | 8 | ||||
-rw-r--r-- | sound/core/vmaster.c | 5 | ||||
-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 | 37 | ||||
-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 | 36 | ||||
-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_realtek.c | 2 | ||||
-rw-r--r-- | sound/pci/hda/patch_sigmatel.c | 29 | ||||
-rw-r--r-- | sound/pci/ice1712/ice1712.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/wm5102.c | 15 | ||||
-rw-r--r-- | sound/soc/codecs/wm5110.c | 16 | ||||
-rw-r--r-- | sound/soc/codecs/wm8350.c | 4 | ||||
-rw-r--r-- | sound/soc/codecs/wm8960.c | 8 | ||||
-rw-r--r-- | sound/soc/tegra/tegra20_i2s.h | 2 | ||||
-rw-r--r-- | sound/soc/tegra/tegra30_i2s.h | 2 | ||||
-rw-r--r-- | sound/usb/card.c | 15 | ||||
-rw-r--r-- | sound/usb/mixer.c | 21 |
22 files changed, 335 insertions, 92 deletions
diff --git a/sound/core/seq/oss/seq_oss_event.c b/sound/core/seq/oss/seq_oss_event.c index 066f5f3e3f4c..c3908862bc8b 100644 --- a/sound/core/seq/oss/seq_oss_event.c +++ b/sound/core/seq/oss/seq_oss_event.c @@ -285,7 +285,12 @@ local_event(struct seq_oss_devinfo *dp, union evrec *q, struct snd_seq_event *ev static int note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev) { - struct seq_oss_synthinfo *info = &dp->synths[dev]; + struct seq_oss_synthinfo *info; + + if (!snd_seq_oss_synth_is_valid(dp, dev)) + return -ENXIO; + + info = &dp->synths[dev]; switch (info->arg.event_passing) { case SNDRV_SEQ_OSS_PROCESS_EVENTS: if (! info->ch || ch < 0 || ch >= info->nr_voices) { @@ -340,7 +345,12 @@ note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, st static int note_off_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev) { - struct seq_oss_synthinfo *info = &dp->synths[dev]; + struct seq_oss_synthinfo *info; + + if (!snd_seq_oss_synth_is_valid(dp, dev)) + return -ENXIO; + + info = &dp->synths[dev]; switch (info->arg.event_passing) { case SNDRV_SEQ_OSS_PROCESS_EVENTS: if (! info->ch || ch < 0 || ch >= info->nr_voices) { 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/core/vmaster.c b/sound/core/vmaster.c index 857586135d18..0097f3619faa 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -213,7 +213,10 @@ static int slave_put(struct snd_kcontrol *kcontrol, } if (!changed) return 0; - return slave_put_val(slave, ucontrol); + err = slave_put_val(slave, ucontrol); + if (err < 0) + return err; + return 1; } static int slave_tlv_cmd(struct snd_kcontrol *kcontrol, 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 04b57383e8cb..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) @@ -3334,6 +3338,8 @@ int snd_hda_create_dig_out_ctls(struct hda_codec *codec, return -EBUSY; } spdif = snd_array_new(&codec->spdif_out); + if (!spdif) + return -ENOMEM; for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) { kctl = snd_ctl_new1(dig_mix, codec); if (!kctl) @@ -3431,11 +3437,16 @@ static struct snd_kcontrol_new spdif_share_sw = { int snd_hda_create_spdif_share_sw(struct hda_codec *codec, struct hda_multi_out *mout) { + struct snd_kcontrol *kctl; + if (!mout->dig_out_nid) return 0; + + kctl = snd_ctl_new1(&spdif_share_sw, mout); + if (!kctl) + return -ENOMEM; /* ATTENTION: here mout is passed as private_data, instead of codec */ - return snd_hda_ctl_add(codec, mout->dig_out_nid, - snd_ctl_new1(&spdif_share_sw, mout)); + return snd_hda_ctl_add(codec, mout->dig_out_nid, kctl); } EXPORT_SYMBOL_HDA(snd_hda_create_spdif_share_sw); 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 db02c1e96b08..0792b5725f9c 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2298,6 +2298,11 @@ static int dspxfr_one_seg(struct hda_codec *codec, hda_frame_size_words = ((sample_rate_div == 0) ? 0 : (num_chans * sample_rate_mul / sample_rate_div)); + if (hda_frame_size_words == 0) { + snd_printdd(KERN_ERR "frmsz zero\n"); + return -EINVAL; + } + buffer_size_words = min(buffer_size_words, (unsigned int)(UC_RANGE(chip_addx, 1) ? 65536 : 32768)); @@ -2308,8 +2313,7 @@ static int dspxfr_one_seg(struct hda_codec *codec, chip_addx, hda_frame_size_words, num_chans, sample_rate_mul, sample_rate_div, buffer_size_words); - if ((buffer_addx == NULL) || (hda_frame_size_words == 0) || - (buffer_size_words < hda_frame_size_words)) { + if (buffer_size_words < hda_frame_size_words) { snd_printdd(KERN_ERR "dspxfr_one_seg:failed\n"); return -EINVAL; } @@ -3235,7 +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 */ @@ -4263,11 +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 */ @@ -4347,12 +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; } @@ -4363,16 +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_realtek.c b/sound/pci/hda/patch_realtek.c index 2d4237bc0d8e..563c24df4d6f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3163,6 +3163,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0290: spec->codec_variant = ALC269_TYPE_ALC280; break; + case 0x10ec0233: case 0x10ec0282: case 0x10ec0283: spec->codec_variant = ALC269_TYPE_ALC282; @@ -3862,6 +3863,7 @@ static int patch_alc680(struct hda_codec *codec) */ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 }, + { .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 }, { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 }, { .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 }, 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/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 2ffdc35d5ffd..806407a3973e 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2594,6 +2594,8 @@ static int snd_ice1712_create(struct snd_card *card, snd_ice1712_proc_init(ice); synchronize_irq(pci->irq); + card->private_data = ice; + err = pci_request_regions(pci, "ICE1712"); if (err < 0) { kfree(ice); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index b8d461db369f..b82bbf584146 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -573,6 +573,13 @@ static const struct reg_default wm5102_sysclk_reva_patch[] = { { 0x025e, 0x0112 }, }; +static const struct reg_default wm5102_sysclk_revb_patch[] = { + { 0x3081, 0x08FE }, + { 0x3083, 0x00ED }, + { 0x30C1, 0x08FE }, + { 0x30C3, 0x00ED }, +}; + static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -587,6 +594,10 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, patch = wm5102_sysclk_reva_patch; patch_size = ARRAY_SIZE(wm5102_sysclk_reva_patch); break; + default: + patch = wm5102_sysclk_revb_patch; + patch_size = ARRAY_SIZE(wm5102_sysclk_revb_patch); + break; } switch (event) { @@ -755,7 +766,7 @@ SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, +SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1), SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), @@ -767,7 +778,7 @@ SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, +SOC_DOUBLE_R_TLV("HPOUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index cd17b477781d..cdeb301da1f6 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -213,9 +213,9 @@ ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE), SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L, ARIZONA_OUT1_OSR_SHIFT, 1, 0), -SOC_SINGLE("OUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L, +SOC_SINGLE("HPOUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L, ARIZONA_OUT2_OSR_SHIFT, 1, 0), -SOC_SINGLE("OUT3 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L, +SOC_SINGLE("HPOUT3 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L, ARIZONA_OUT3_OSR_SHIFT, 1, 0), SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, ARIZONA_OUT4_OSR_SHIFT, 1, 0), @@ -226,9 +226,9 @@ SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L, SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, +SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("OUT3 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, +SOC_DOUBLE_R("HPOUT3 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, ARIZONA_DAC_DIGITAL_VOLUME_3R, ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), @@ -240,10 +240,10 @@ SOC_DOUBLE_R("SPKDAT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_6L, SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, +SOC_DOUBLE_R_TLV("HPOUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT, 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("OUT3 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, +SOC_DOUBLE_R_TLV("HPOUT3 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, ARIZONA_DAC_DIGITAL_VOLUME_3R, ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, @@ -260,11 +260,11 @@ SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L, ARIZONA_OUTPUT_PATH_CONFIG_1R, ARIZONA_OUT1L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("OUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L, +SOC_DOUBLE_R_RANGE_TLV("HPOUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L, ARIZONA_OUTPUT_PATH_CONFIG_2R, ARIZONA_OUT2L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("OUT3 Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L, +SOC_DOUBLE_R_RANGE_TLV("HPOUT3 Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L, ARIZONA_OUTPUT_PATH_CONFIG_3R, ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv), diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index ec0efc1443ba..0e8b3aaf6c8d 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1301,7 +1301,7 @@ static irqreturn_t wm8350_hpl_jack_handler(int irq, void *data) if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&priv->hpl.work, 200); + schedule_delayed_work(&priv->hpl.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } @@ -1318,7 +1318,7 @@ static irqreturn_t wm8350_hpr_jack_handler(int irq, void *data) if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&priv->hpr.work, 200); + schedule_delayed_work(&priv->hpr.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 9bb927325993..a64b93425ae3 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -53,8 +53,8 @@ * using 2 wire for device control, so we cache them instead. */ static const struct reg_default wm8960_reg_defaults[] = { - { 0x0, 0x0097 }, - { 0x1, 0x0097 }, + { 0x0, 0x00a7 }, + { 0x1, 0x00a7 }, { 0x2, 0x0000 }, { 0x3, 0x0000 }, { 0x4, 0x0000 }, @@ -323,8 +323,8 @@ SND_SOC_DAPM_MIXER("Left Input Mixer", WM8960_POWER3, 5, 0, SND_SOC_DAPM_MIXER("Right Input Mixer", WM8960_POWER3, 4, 0, wm8960_rin, ARRAY_SIZE(wm8960_rin)), -SND_SOC_DAPM_ADC("Left ADC", "Capture", WM8960_POWER2, 3, 0), -SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER2, 2, 0), +SND_SOC_DAPM_ADC("Left ADC", "Capture", WM8960_POWER1, 3, 0), +SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER1, 2, 0), SND_SOC_DAPM_DAC("Left DAC", "Playback", WM8960_POWER2, 8, 0), SND_SOC_DAPM_DAC("Right DAC", "Playback", WM8960_POWER2, 7, 0), diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h index c27069d24d77..729958713cd4 100644 --- a/sound/soc/tegra/tegra20_i2s.h +++ b/sound/soc/tegra/tegra20_i2s.h @@ -121,7 +121,7 @@ #define TEGRA20_I2S_TIMING_NON_SYM_ENABLE (1 << 12) #define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 -#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff +#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7ff #define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) /* Fields in TEGRA20_I2S_FIFO_SCR */ diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h index 34dc47b9581c..a294d942b9f7 100644 --- a/sound/soc/tegra/tegra30_i2s.h +++ b/sound/soc/tegra/tegra30_i2s.h @@ -110,7 +110,7 @@ #define TEGRA30_I2S_TIMING_NON_SYM_ENABLE (1 << 12) #define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 -#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff +#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7ff #define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) /* Fields in TEGRA30_I2S_OFFSET */ 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; } } |