diff options
Diffstat (limited to 'sound')
37 files changed, 426 insertions, 167 deletions
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 71ae86ca64a..eb560fa3232 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3222,18 +3222,10 @@ EXPORT_SYMBOL_GPL(snd_pcm_lib_default_mmap); int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_struct *area) { - long size; - unsigned long offset; + struct snd_pcm_runtime *runtime = substream->runtime;; area->vm_page_prot = pgprot_noncached(area->vm_page_prot); - area->vm_flags |= VM_IO; - size = area->vm_end - area->vm_start; - offset = area->vm_pgoff << PAGE_SHIFT; - if (io_remap_pfn_range(area, area->vm_start, - (substream->runtime->dma_addr + offset) >> PAGE_SHIFT, - size, area->vm_page_prot)) - return -EAGAIN; - return 0; + return vm_iomap_memory(area, runtime->dma_addr, runtime->dma_bytes); } EXPORT_SYMBOL(snd_pcm_lib_mmap_iomem); diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index 160b1bd0cd6..24d44b2f61a 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 30bcfe470f8..4ff60a6427d 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 3536b076b52..0aabfedeecb 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 97c68dd24ef..4aba7646dd9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -173,7 +173,7 @@ const char *snd_hda_get_jack_type(u32 cfg) "Line Out", "Speaker", "HP Out", "CD", "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", "Line In", "Aux", "Mic", "Telephony", - "SPDIF In", "Digitial In", "Reserved", "Other" + "SPDIF In", "Digital In", "Reserved", "Other" }; return jack_types[(cfg & AC_DEFCFG_DEVICE) @@ -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_eld.c b/sound/pci/hda/hda_eld.c index 7dd846380a5..d0d7ac1e99d 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -320,7 +320,7 @@ int snd_hdmi_get_eld(struct hda_codec *codec, hda_nid_t nid, unsigned char *buf, int *eld_size) { int i; - int ret; + int ret = 0; int size; /* diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 78897d05d80..2dbe767be16 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -740,7 +740,7 @@ EXPORT_SYMBOL_HDA(snd_hda_activate_path); static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path) { struct hda_gen_spec *spec = codec->spec; - bool changed; + bool changed = false; int i; if (!spec->power_down_unused || path->active) @@ -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 4cea6bb6fad..bcd40ee488e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -134,8 +134,8 @@ MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " * this may give more power-saving, but will take longer time to * wake up. */ -static int power_save_controller = -1; -module_param(power_save_controller, bint, 0644); +static bool power_save_controller = 1; +module_param(power_save_controller, bool, 0644); MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); #endif /* CONFIG_PM */ @@ -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 */ @@ -2846,8 +2931,6 @@ static int azx_runtime_idle(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; - if (power_save_controller > 0) - return 0; if (!power_save_controller || !(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) return -EBUSY; @@ -3481,6 +3564,7 @@ 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 eefc4563b2f..0792b5725f9 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 72ebb8a36b1..0d9c58f1356 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 941bf6c766e..2a89d1eefeb 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_hdmi.c b/sound/pci/hda/patch_hdmi.c index 78e1827d0a9..de8ac5c07fd 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1196,7 +1196,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) _snd_printd(SND_PR_VERBOSE, "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, eld->monitor_present, eld->eld_valid); + codec->addr, pin_nid, pin_eld->monitor_present, eld->eld_valid); if (eld->eld_valid) { if (snd_hdmi_get_eld(codec, pin_nid, eld->eld_buffer, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 563c24df4d6..f15c36bde54 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3440,7 +3440,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec) const hda_nid_t *ssids; if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 || - codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670) + codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670 || + codec->vendor_id == 0x10ec0671) ssids = alc663_ssids; else ssids = alc662_ssids; @@ -3894,6 +3895,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, { .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 }, { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 }, + { .id = 0x10ec0671, .name = "ALC671", .patch = patch_alc662 }, { .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 83d5335ac34..dafe04ae8c7 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/Kconfig b/sound/soc/codecs/Kconfig index 45b72561c61..350b8645897 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -324,7 +324,7 @@ config SND_SOC_TLV320AIC23 tristate config SND_SOC_TLV320AIC26 - tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE + tristate depends on SPI config SND_SOC_TLV320AIC32X4 diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index fc176044994..fc176044994 100755..100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index 7e103f24905..7e103f24905 100755..100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index f2d61a18783..566ea3256e2 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -159,6 +159,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: width = SI476X_PCM_FORMAT_S8; + break; case SNDRV_PCM_FORMAT_S16_LE: width = SI476X_PCM_FORMAT_S16_LE; break; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index b82bbf58414..34d0201d6a7 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -584,7 +584,7 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - struct arizona *arizona = dev_get_drvdata(codec->dev); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); struct regmap *regmap = codec->control_data; const struct reg_default *patch = NULL; int i, patch_size; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 134e41c870b..f8a31ad0b20 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1083,6 +1083,8 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = { { "ROP", NULL, "Right Speaker PGA" }, { "RON", NULL, "Right Speaker PGA" }, + { "Charge Pump", NULL, "CLK_DSP" }, + { "Left Headphone Output PGA", NULL, "Charge Pump" }, { "Right Headphone Output PGA", NULL, "Charge Pump" }, { "Left Line Output PGA", NULL, "Charge Pump" }, diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f3f7e75f862..9af1bddc4c6 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -828,7 +828,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) &buf_list); if (!buf) { adsp_err(dsp, "Out of memory\n"); - return -ENOMEM; + ret = -ENOMEM; + goto out_fw; } adsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n", @@ -865,7 +866,7 @@ out_fw: wm_adsp_buf_free(&buf_list); out: kfree(file); - return 0; + return ret; } int wm_adsp1_init(struct wm_adsp *adsp) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 55464a5b070..810c7eeb7b0 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -496,6 +496,8 @@ static void imx_ssi_ac97_reset(struct snd_ac97 *ac97) if (imx_ssi->ac97_reset) imx_ssi->ac97_reset(ac97); + /* First read sometimes fails, do a dummy read */ + imx_ssi_ac97_read(ac97, 0); } static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) @@ -504,6 +506,9 @@ static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) if (imx_ssi->ac97_warm_reset) imx_ssi->ac97_warm_reset(ac97); + + /* First read sometimes fails, do a dummy read */ + imx_ssi_ac97_read(ac97, 0); } struct snd_ac97_bus_ops soc_ac97_ops = { diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 8e52c1485df..eb4373840bb 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -51,7 +51,7 @@ static struct snd_soc_card pcm030_card = { .num_links = ARRAY_SIZE(pcm030_fabric_dai), }; -static int __init pcm030_fabric_probe(struct platform_device *op) +static int pcm030_fabric_probe(struct platform_device *op) { struct device_node *np = op->dev.of_node; struct device_node *platform_np; diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index d7231e336a7..6bbeb0bf1a7 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -972,6 +972,7 @@ static const struct snd_soc_dai_ops samsung_i2s_dai_ops = { static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) { struct i2s_dai *i2s; + int ret; i2s = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dai), GFP_KERNEL); if (i2s == NULL) @@ -996,15 +997,17 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) i2s->i2s_dai_drv.capture.channels_max = 2; i2s->i2s_dai_drv.capture.rates = SAMSUNG_I2S_RATES; i2s->i2s_dai_drv.capture.formats = SAMSUNG_I2S_FMTS; + dev_set_drvdata(&i2s->pdev->dev, i2s); } else { /* Create a new platform_device for Secondary */ - i2s->pdev = platform_device_register_resndata(NULL, - "samsung-i2s-sec", -1, NULL, 0, NULL, 0); + i2s->pdev = platform_device_alloc("samsung-i2s-sec", -1); if (IS_ERR(i2s->pdev)) return NULL; - } - /* Pre-assign snd_soc_dai_set_drvdata */ - dev_set_drvdata(&i2s->pdev->dev, i2s); + platform_set_drvdata(i2s->pdev, i2s); + ret = platform_device_add(i2s->pdev); + if (ret < 0) + return NULL; + } return i2s; } @@ -1107,6 +1110,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (samsung_dai_type == TYPE_SEC) { sec_dai = dev_get_drvdata(&pdev->dev); + if (!sec_dai) { + dev_err(&pdev->dev, "Unable to get drvdata\n"); + return -EFAULT; + } snd_soc_register_dai(&sec_dai->pdev->dev, &sec_dai->i2s_dai_drv); asoc_dma_platform_register(&pdev->dev); diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 19eff8fc4fd..1a8b03e4b41 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -342,8 +342,8 @@ static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd) return 0; } -static struct snd_soc_platform sh7760_soc_platform = { - .pcm_ops = &camelot_pcm_ops, +static struct snd_soc_platform_driver sh7760_soc_platform = { + .ops = &camelot_pcm_ops, .pcm_new = camelot_pcm_new, .pcm_free = camelot_pcm_free, }; diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index b5b3db71e25..ed0bfb0ddb9 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -211,19 +211,27 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, if (platform->driver->compr_ops && platform->driver->compr_ops->set_params) { ret = platform->driver->compr_ops->set_params(cstream, params); if (ret < 0) - goto out; + goto err; } if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->set_params) { ret = rtd->dai_link->compr_ops->set_params(cstream); if (ret < 0) - goto out; + goto err; } snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_START); -out: + /* cancel any delayed stream shutdown that is pending */ + rtd->pop_wait = 0; + mutex_unlock(&rtd->pcm_mutex); + + cancel_delayed_work_sync(&rtd->delayed_work); + + return ret; + +err: mutex_unlock(&rtd->pcm_mutex); return ret; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b7e84a7cd9e..c70f9e07204 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2963,7 +2963,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, val = val << shift; ret = snd_soc_update_bits_locked(codec, reg, val_mask, val); - if (ret != 0) + if (ret < 0) return ret; if (snd_soc_volsw_is_stereo(mc)) { @@ -3140,7 +3140,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, if (params->mask) { ret = regmap_read(codec->control_data, params->base, &val); if (ret != 0) - return ret; + goto out; val &= params->mask; @@ -3158,13 +3158,15 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, ((u32 *)data)[0] |= cpu_to_be32(val); break; default: - return -EINVAL; + ret = -EINVAL; + goto out; } } ret = regmap_raw_write(codec->control_data, params->base, data, len); +out: kfree(data); return ret; @@ -3906,7 +3908,7 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_dais); * @platform: platform to register */ int snd_soc_register_platform(struct device *dev, - struct snd_soc_platform_driver *platform_drv) + const struct snd_soc_platform_driver *platform_drv) { struct snd_soc_platform *platform; @@ -4022,8 +4024,8 @@ int snd_soc_register_codec(struct device *dev, /* create CODEC component name */ codec->name = fmt_single_name(dev, &codec->id); if (codec->name == NULL) { - kfree(codec); - return -ENOMEM; + ret = -ENOMEM; + goto fail_codec; } if (codec_drv->compress_type) @@ -4062,7 +4064,7 @@ int snd_soc_register_codec(struct device *dev, reg_size, GFP_KERNEL); if (!codec->reg_def_copy) { ret = -ENOMEM; - goto fail; + goto fail_codec_name; } } } @@ -4086,18 +4088,22 @@ int snd_soc_register_codec(struct device *dev, mutex_unlock(&client_mutex); /* register any DAIs */ - if (num_dai) { - ret = snd_soc_register_dais(dev, dai_drv, num_dai); - if (ret < 0) - dev_err(codec->dev, "ASoC: Failed to regster" - " DAIs: %d\n", ret); + ret = snd_soc_register_dais(dev, dai_drv, num_dai); + if (ret < 0) { + dev_err(codec->dev, "ASoC: Failed to regster DAIs: %d\n", ret); + goto fail_codec_name; } dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", codec->name); return 0; -fail: +fail_codec_name: + mutex_lock(&client_mutex); + list_del(&codec->list); + mutex_unlock(&client_mutex); + kfree(codec->name); +fail_codec: kfree(codec); return ret; } @@ -4111,7 +4117,6 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); void snd_soc_unregister_codec(struct device *dev) { struct snd_soc_codec *codec; - int i; list_for_each_entry(codec, &codec_list, list) { if (dev == codec->dev) @@ -4120,9 +4125,7 @@ void snd_soc_unregister_codec(struct device *dev) return; found: - if (codec->num_dai) - for (i = 0; i < codec->num_dai; i++) - snd_soc_unregister_dai(dev); + snd_soc_unregister_dais(dev, codec->num_dai); mutex_lock(&client_mutex); list_del(&codec->list); @@ -4197,7 +4200,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", propname, 2 * i, ret); - kfree(routes); return -EINVAL; } ret = of_property_read_string_index(np, propname, @@ -4206,7 +4208,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", propname, (2 * i) + 1, ret); - kfree(routes); return -EINVAL; } } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1d6a9b3ceb2..33acd8b892d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -831,6 +831,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, if (path->weak) continue; + if (path->walking) + return 1; + if (path->walked) continue; @@ -838,6 +841,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, if (path->sink && path->connect) { path->walked = 1; + path->walking = 1; /* do we need to add this widget to the list ? */ if (list) { @@ -847,11 +851,14 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, dev_err(widget->dapm->dev, "ASoC: could not add widget %s\n", widget->name); + path->walking = 0; return con; } } con += is_connected_output_ep(path->sink, list); + + path->walking = 0; } } @@ -931,6 +938,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, if (path->weak) continue; + if (path->walking) + return 1; + if (path->walked) continue; @@ -938,6 +948,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, if (path->source && path->connect) { path->walked = 1; + path->walking = 1; /* do we need to add this widget to the list ? */ if (list) { @@ -947,11 +958,14 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, dev_err(widget->dapm->dev, "ASoC: could not add widget %s\n", widget->name); + path->walking = 0; return con; } } con += is_connected_input_ep(path->source, list); + + path->walking = 0; } } @@ -3123,7 +3137,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, break; } - dapm->n_widgets++; w->dapm = dapm; w->codec = dapm->codec; w->platform = dapm->platform; diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 29183ef2b93..8ca9ecc5ac5 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -158,10 +158,7 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, return -EINVAL; } - if (IS_ERR(codec->control_data)) - return PTR_ERR(codec->control_data); - - return 0; + return PTR_RET(codec->control_data); } EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); #else diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index fe4541df498..4b3be6c3c91 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -90,8 +90,33 @@ static struct snd_soc_platform_driver dummy_platform = { }; static struct snd_soc_codec_driver dummy_codec; + +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE | \ + SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) static struct snd_soc_dai_driver dummy_dai = { .name = "snd-soc-dummy-dai", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, }; static int snd_soc_dummy_probe(struct platform_device *pdev) diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 9b76cc5a114..5e7aebe1e66 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -149,9 +149,9 @@ static void spear_pcm_free(struct snd_pcm *pcm) static u64 spear_pcm_dmamask = DMA_BIT_MASK(32); -static int spear_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int spear_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; int ret; if (!card->dev->dma_mask) @@ -159,16 +159,16 @@ static int spear_pcm_new(struct snd_card *card, if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (dai->driver->playback.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(pcm, + if (rtd->cpu_dai->driver->playback.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, spear_pcm_hardware.buffer_bytes_max); if (ret) return ret; } - if (dai->driver->capture.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(pcm, + if (rtd->cpu_dai->driver->capture.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(rtd->pcm, SNDRV_PCM_STREAM_CAPTURE, spear_pcm_hardware.buffer_bytes_max); if (ret) diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index c925ab0adeb..5e2c55c5b25 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -43,8 +43,6 @@ static const struct snd_pcm_hardware tegra_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_INTERLEAVED, .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels_min = 2, @@ -127,26 +125,6 @@ static int tegra_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int tegra_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - return snd_dmaengine_pcm_trigger(substream, - SNDRV_PCM_TRIGGER_START); - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - return snd_dmaengine_pcm_trigger(substream, - SNDRV_PCM_TRIGGER_STOP); - default: - return -EINVAL; - } - return 0; -} - static int tegra_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) { @@ -164,7 +142,7 @@ static struct snd_pcm_ops tegra_pcm_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = tegra_pcm_hw_params, .hw_free = tegra_pcm_hw_free, - .trigger = tegra_pcm_trigger, + .trigger = snd_dmaengine_pcm_trigger, .pointer = snd_dmaengine_pcm_pointer, .mmap = tegra_pcm_mmap, }; diff --git a/sound/usb/card.c b/sound/usb/card.c index 803953a9bff..2da8ad75fd9 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/clock.c b/sound/usb/clock.c index 5e634a2eb28..9e2703a2515 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -253,7 +253,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, { struct usb_device *dev = chip->dev; unsigned char data[4]; - int err, crate; + int err, cur_rate, prev_rate; int clock = snd_usb_clock_find_source(chip, fmt->clock); if (clock < 0) @@ -266,6 +266,19 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, return -ENXIO; } + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + data, sizeof(data)); + if (err < 0) { + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", + dev->devnum, iface, fmt->altsetting); + prev_rate = 0; + } else { + prev_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); + } + data[0] = rate; data[1] = rate >> 8; data[2] = rate >> 16; @@ -280,19 +293,31 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, return err; } - if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, - UAC2_CS_CONTROL_SAM_FREQ << 8, - snd_usb_ctrl_intf(chip) | (clock << 8), - data, sizeof(data))) < 0) { + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + data, sizeof(data)); + if (err < 0) { snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", dev->devnum, iface, fmt->altsetting); - return err; + cur_rate = 0; + } else { + cur_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); } - crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); - if (crate != rate) - snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); + if (cur_rate != rate) { + snd_printd(KERN_WARNING + "current rate %d is different from the runtime rate %d\n", + cur_rate, rate); + } + + /* Some devices doesn't respond to sample rate changes while the + * interface is active. */ + if (rate != prev_rate) { + usb_set_interface(dev, iface, 0); + usb_set_interface(dev, iface, fmt->altsetting); + } return 0; } diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 638e7f73801..ca4739c3f65 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; } } diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 497d2741d11..ebe91440a06 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -509,7 +509,7 @@ static int snd_nativeinstruments_control_get(struct snd_kcontrol *kcontrol, else ret = usb_control_msg(dev, usb_rcvctrlpipe(dev, 0), bRequest, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, - 0, cpu_to_le16(wIndex), + 0, wIndex, &tmp, sizeof(tmp), 1000); up_read(&mixer->chip->shutdown_rwsem); @@ -540,7 +540,7 @@ static int snd_nativeinstruments_control_put(struct snd_kcontrol *kcontrol, else ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), bRequest, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, - cpu_to_le16(wValue), cpu_to_le16(wIndex), + wValue, wIndex, NULL, 0, 1000); up_read(&mixer->chip->shutdown_rwsem); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5325a3869bb..9c5ab22358b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -486,7 +486,7 @@ static int snd_usb_nativeinstruments_boot_quirk(struct usb_device *dev) { int ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), 0xaf, USB_TYPE_VENDOR | USB_RECIP_DEVICE, - cpu_to_le16(1), 0, NULL, 0, 1000); + 1, 0, NULL, 0, 1000); if (ret < 0) return ret; |