diff options
Diffstat (limited to 'sound')
53 files changed, 405 insertions, 89 deletions
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 7ae8e24dc1e6..81624f6e3f33 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -723,6 +723,9 @@ static int snd_compr_stop(struct snd_compr_stream *stream) retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP); if (!retval) { + /* clear flags and stop any drain wait */ + stream->partial_drain = false; + stream->metadata_set = false; snd_compr_drain_notify(stream); stream->runtime->total_bytes_available = 0; stream->runtime->total_bytes_transferred = 0; @@ -880,6 +883,7 @@ static int snd_compr_partial_drain(struct snd_compr_stream *stream) if (stream->next_track == false) return -EPERM; + stream->partial_drain = true; retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); if (retval) { pr_debug("Partial drain returned failure\n"); diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index a73baa1242be..727219f40201 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -229,14 +229,14 @@ static int snd_hwdep_dsp_load(struct snd_hwdep *hw, if (copy_from_user(&info, _info, sizeof(info))) return -EFAULT; /* check whether the dsp was already loaded */ - if (hw->dsp_loaded & (1 << info.index)) + if (hw->dsp_loaded & (1u << info.index)) return -EBUSY; if (!access_ok(VERIFY_READ, info.image, info.length)) return -EFAULT; err = hw->ops.dsp_load(hw, &info); if (err < 0) return err; - hw->dsp_loaded |= (1 << info.index); + hw->dsp_loaded |= (1u << info.index); return 0; } diff --git a/sound/core/info.c b/sound/core/info.c index 5fb00437507b..f15569cd124d 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -634,7 +634,9 @@ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) { int c = -1; - if (snd_BUG_ON(!buffer || !buffer->buffer)) + if (snd_BUG_ON(!buffer)) + return 1; + if (!buffer->buffer) return 1; if (len <= 0 || buffer->stop || buffer->error) return 1; diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c index 3788906421a7..fe27034f2846 100644 --- a/sound/core/oss/mulaw.c +++ b/sound/core/oss/mulaw.c @@ -329,8 +329,8 @@ int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug, snd_BUG(); return -EINVAL; } - if (snd_BUG_ON(!snd_pcm_format_linear(format->format))) - return -ENXIO; + if (!snd_pcm_format_linear(format->format)) + return -EINVAL; err = snd_pcm_plugin_build(plug, "Mu-Law<->linear conversion", src_format, dst_format, diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 14b1ee29509d..071e09c3d855 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1950,6 +1950,11 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) } pcm_file = f.file->private_data; substream1 = pcm_file->substream; + if (substream == substream1) { + res = -EINVAL; + goto _badf; + } + group = kmalloc(sizeof(*group), GFP_KERNEL); if (!group) { res = -ENOMEM; diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index 8cdf489df80e..4b7897959913 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -181,10 +181,16 @@ static long odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { struct seq_oss_devinfo *dp; + long rc; + dp = file->private_data; if (snd_BUG_ON(!dp)) return -ENXIO; - return snd_seq_oss_ioctl(dp, cmd, arg); + + mutex_lock(®ister_mutex); + rc = snd_seq_oss_ioctl(dp, cmd, arg); + mutex_unlock(®ister_mutex); + return rc; } #ifdef CONFIG_COMPAT diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index 42920a243328..3f94746d587a 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -104,6 +104,8 @@ int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file, { struct snd_dm_fm_info info; + memset(&info, 0, sizeof(info)); + info.fm_mode = opl3->fm_mode; info.rhythm = opl3->rhythm; if (copy_to_user(argp, &info, sizeof(struct snd_dm_fm_info))) diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c index ef689997d6a5..bf53e342788e 100644 --- a/sound/firewire/digi00x/digi00x.c +++ b/sound/firewire/digi00x/digi00x.c @@ -15,6 +15,7 @@ MODULE_LICENSE("GPL v2"); #define VENDOR_DIGIDESIGN 0x00a07e #define MODEL_CONSOLE 0x000001 #define MODEL_RACK 0x000002 +#define SPEC_VERSION 0x000001 static int name_card(struct snd_dg00x *dg00x) { @@ -185,14 +186,18 @@ static const struct ieee1394_device_id snd_dg00x_id_table[] = { /* Both of 002/003 use the same ID. */ { .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_VERSION | IEEE1394_MATCH_MODEL_ID, .vendor_id = VENDOR_DIGIDESIGN, + .version = SPEC_VERSION, .model_id = MODEL_CONSOLE, }, { .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_VERSION | IEEE1394_MATCH_MODEL_ID, .vendor_id = VENDOR_DIGIDESIGN, + .version = SPEC_VERSION, .model_id = MODEL_RACK, }, {} diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index d3fdc463a884..1e61cdce2895 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -225,11 +225,39 @@ static void snd_tscm_remove(struct fw_unit *unit) } static const struct ieee1394_device_id snd_tscm_id_table[] = { + // Tascam, FW-1884. { .match_flags = IEEE1394_MATCH_VENDOR_ID | - IEEE1394_MATCH_SPECIFIER_ID, + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, .vendor_id = 0x00022e, .specifier_id = 0x00022e, + .version = 0x800000, + }, + // Tascam, FE-8 (.version = 0x800001) + // This kernel module doesn't support FE-8 because the most of features + // can be implemented in userspace without any specific support of this + // module. + // + // .version = 0x800002 is unknown. + // + // Tascam, FW-1082. + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .vendor_id = 0x00022e, + .specifier_id = 0x00022e, + .version = 0x800003, + }, + // Tascam, FW-1804. + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .vendor_id = 0x00022e, + .specifier_id = 0x00022e, + .version = 0x800004, }, /* FE-08 requires reverse-engineering because it just has faders. */ {} diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 714a51721a31..ab9236e4c157 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -155,6 +155,7 @@ static void process_unsol_events(struct work_struct *work) struct hdac_driver *drv; unsigned int rp, caddr, res; + spin_lock_irq(&bus->reg_lock); while (bus->unsol_rp != bus->unsol_wp) { rp = (bus->unsol_rp + 1) % HDA_UNSOL_QUEUE_SIZE; bus->unsol_rp = rp; @@ -166,10 +167,13 @@ static void process_unsol_events(struct work_struct *work) codec = bus->caddr_tbl[caddr & 0x0f]; if (!codec || !codec->dev.driver) continue; + spin_unlock_irq(&bus->reg_lock); drv = drv_to_hdac_driver(codec->dev.driver); if (drv->unsol_event) drv->unsol_event(codec, res); + spin_lock_irq(&bus->reg_lock); } + spin_unlock_irq(&bus->reg_lock); } /** diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 19deb306facb..4a843eb7cc94 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -123,6 +123,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_init); void snd_hdac_device_exit(struct hdac_device *codec) { pm_runtime_put_noidle(&codec->dev); + /* keep balance of runtime PM child_count in parent device */ + pm_runtime_set_suspended(&codec->dev); snd_hdac_bus_remove_device(codec->bus, codec); kfree(codec->vendor_name); kfree(codec->chip_name); diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index a826c138e7f5..8a58ed168756 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -284,8 +284,10 @@ static int snd_es968_pnp_detect(struct pnp_card_link *pcard, return error; } error = snd_es1688_probe(card, dev); - if (error < 0) + if (error < 0) { + snd_card_free(card); return error; + } pnp_set_card_drvdata(pcard, card); snd_es968_pnp_is_probed = 1; return 0; diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 0b1e4b34b299..13c8e6542a2f 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -1175,7 +1175,10 @@ wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header) "alias for %d\n", header->number, header->hdr.a.OriginalSample); - + + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + munge_int32 (header->number, &alias_hdr[0], 2); munge_int32 (header->hdr.a.OriginalSample, &alias_hdr[2], 2); munge_int32 (*((unsigned int *)&header->hdr.a.sampleStartOffset), @@ -1206,6 +1209,9 @@ wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header) int num_samples; unsigned char *msample_hdr; + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + msample_hdr = kmalloc(WF_MSAMPLE_BYTES, GFP_KERNEL); if (! msample_hdr) return -ENOMEM; diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index b1a2a7ea4172..b4ccd9f92400 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -350,7 +350,7 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev, struct hpi_message hm; struct hpi_response hr; struct hpi_adapter adapter; - struct hpi_pci pci; + struct hpi_pci pci = { 0 }; memset(&adapter, 0, sizeof(adapter)); @@ -506,7 +506,7 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev, return 0; err: - for (idx = 0; idx < HPI_MAX_ADAPTER_MEM_SPACES; idx++) { + while (--idx >= 0) { if (pci.ap_mem_base[idx]) { iounmap(pci.ap_mem_base[idx]); pci.ap_mem_base[idx] = NULL; diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index cd27b5536654..675b812e96d6 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -551,7 +551,8 @@ static int snd_ca0106_pcm_power_dac(struct snd_ca0106 *chip, int channel_id, else /* Power down */ chip->spi_dac_reg[reg] |= bit; - return snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]); + if (snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]) != 0) + return -ENXIO; } return 0; } diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 0020fd0efc46..09c547f4cc18 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -780,7 +780,7 @@ static void snd_cs46xx_set_capture_sample_rate(struct snd_cs46xx *chip, unsigned rate = 48000 / 9; /* - * We can not capture at at rate greater than the Input Rate (48000). + * We can not capture at a rate greater than the Input Rate (48000). * Return an error if an attempt is made to stray outside that limit. */ if (rate > 48000) diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 7488e1b7a770..4e726d39b05d 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -1742,7 +1742,7 @@ int cs46xx_iec958_pre_open (struct snd_cs46xx *chip) struct dsp_spos_instance * ins = chip->dsp_spos_instance; if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) { - /* remove AsynchFGTxSCB and and PCMSerialInput_II */ + /* remove AsynchFGTxSCB and PCMSerialInput_II */ cs46xx_dsp_disable_spdif_out (chip); /* save state */ diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index e1f0bcd45c37..b58a098a7270 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2215,7 +2215,6 @@ static int snd_echo_resume(struct device *dev) if (err < 0) { kfree(commpage_bak); dev_err(dev, "resume init_hw err=%d\n", err); - snd_echo_free(chip); return err; } @@ -2242,7 +2241,6 @@ static int snd_echo_resume(struct device *dev) if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { dev_err(chip->card->dev, "cannot grab irq\n"); - snd_echo_free(chip); return -EBUSY; } chip->irq = pci->irq; diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 8b1cf237b96e..c5dc8587d2ac 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -76,6 +76,12 @@ static int compare_input_type(const void *ap, const void *bp) if (a->type != b->type) return (int)(a->type - b->type); + /* If has both hs_mic and hp_mic, pick the hs_mic ahead of hp_mic. */ + if (a->is_headset_mic && b->is_headphone_mic) + return -1; /* don't swap */ + else if (a->is_headphone_mic && b->is_headset_mic) + return 1; /* swap */ + /* In case one has boost and the other one has not, pick the one with boost first. */ return (int)(b->has_boost_on_pin - a->has_boost_on_pin); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7d65fe31c825..a56f018d586f 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3394,7 +3394,7 @@ EXPORT_SYMBOL_GPL(snd_hda_set_power_save); * @nid: NID to check / update * * Check whether the given NID is in the amp list. If it's in the list, - * check the current AMP status, and update the the power-status according + * check the current AMP status, and update the power-status according * to the mute status. * * This function is supposed to be set or called from the check_power_status diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index fa261b27d858..8198d2e53b7d 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1169,16 +1169,23 @@ irqreturn_t azx_interrupt(int irq, void *dev_id) if (snd_hdac_bus_handle_stream_irq(bus, status, stream_update)) active = true; - /* clear rirb int */ status = azx_readb(chip, RIRBSTS); if (status & RIRB_INT_MASK) { + /* + * Clearing the interrupt status here ensures that no + * interrupt gets masked after the RIRB wp is read in + * snd_hdac_bus_update_rirb. This avoids a possible + * race condition where codec response in RIRB may + * remain unserviced by IRQ, eventually falling back + * to polling mode in azx_rirb_get_response. + */ + azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); active = true; if (status & RIRB_INT_RESPONSE) { if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) udelay(80); snd_hdac_bus_update_rirb(bus); } - azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); } } while (active && ++repeat < 10); diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 28ef409a9e6a..9dee657ce9e2 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -823,7 +823,7 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, } } -/* sync power of each widget in the the given path */ +/* sync power of each widget in the given path */ static hda_nid_t path_power_update(struct hda_codec *codec, struct nid_path *path, bool allow_powerdown) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7779f5460715..e399c5718ee6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1282,6 +1282,7 @@ static void azx_vs_set_state(struct pci_dev *pci, struct snd_card *card = pci_get_drvdata(pci); struct azx *chip = card->private_data; struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct hda_codec *codec; bool disabled; wait_for_completion(&hda->probe_wait); @@ -1306,8 +1307,12 @@ static void azx_vs_set_state(struct pci_dev *pci, dev_info(chip->card->dev, "%s via vga_switcheroo\n", disabled ? "Disabling" : "Enabling"); if (disabled) { - pm_runtime_put_sync_suspend(card->dev); - azx_suspend(card->dev); + list_for_each_codec(codec, &chip->bus) { + pm_runtime_suspend(hda_codec_dev(codec)); + pm_runtime_disable(hda_codec_dev(codec)); + } + pm_runtime_suspend(card->dev); + pm_runtime_disable(card->dev); /* when we get suspended by vga_switcheroo we end up in D3cold, * however we have no ACPI handle, so pci/acpi can't put us there, * put ourselves there */ @@ -1318,9 +1323,12 @@ static void azx_vs_set_state(struct pci_dev *pci, "Cannot lock devices!\n"); } else { snd_hda_unlock_devices(&chip->bus); - pm_runtime_get_noresume(card->dev); chip->disabled = false; - azx_resume(card->dev); + pm_runtime_enable(card->dev); + list_for_each_codec(codec, &chip->bus) { + pm_runtime_enable(hda_codec_dev(codec)); + pm_runtime_resume(hda_codec_dev(codec)); + } } } } @@ -1350,6 +1358,7 @@ static void init_vga_switcheroo(struct azx *chip) dev_info(chip->card->dev, "Handle vga_switcheroo audio client\n"); hda->use_vga_switcheroo = 1; + chip->driver_caps |= AZX_DCAPS_PM_RUNTIME; pci_dev_put(p); } } @@ -1375,9 +1384,6 @@ static int register_vga_switcheroo(struct azx *chip) return err; hda->vga_switcheroo_registered = 1; - /* register as an optimus hdmi audio power domain */ - vga_switcheroo_init_domain_pm_optimus_hdmi_audio(chip->card->dev, - &hda->hdmi_pm_domain); return 0; } #else @@ -1406,10 +1412,8 @@ static int azx_free(struct azx *chip) if (use_vga_switcheroo(hda)) { if (chip->disabled && hda->probe_continued) snd_hda_unlock_devices(&chip->bus); - if (hda->vga_switcheroo_registered) { + if (hda->vga_switcheroo_registered) vga_switcheroo_unregister_client(chip->pci); - vga_switcheroo_fini_domain_pm_ops(chip->card->dev); - } } if (bus->chip_init) { @@ -2301,6 +2305,7 @@ static int azx_probe_continue(struct azx *chip) struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct hdac_bus *bus = azx_bus(chip); struct pci_dev *pci = chip->pci; + struct hda_codec *codec; int dev = chip->dev_index; int val; int err; @@ -2385,6 +2390,14 @@ static int azx_probe_continue(struct azx *chip) chip->running = 1; azx_add_card_list(chip); + /* + * The discrete GPU cannot power down unless the HDA controller runtime + * suspends, so activate runtime PM on codecs even if power_save == 0. + */ + if (use_vga_switcheroo(hda)) + list_for_each_codec(codec, &chip->bus) + codec->auto_runtime_pm = 1; + val = power_save; #ifdef CONFIG_PM if (pm_blacklist) { @@ -2399,7 +2412,7 @@ static int azx_probe_continue(struct azx *chip) } #endif /* CONFIG_PM */ snd_hda_set_power_save(&chip->bus, val * 1000); - if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) + if (azx_has_pm_runtime(chip)) pm_runtime_put_autosuspend(&pci->dev); out_free: diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index ff0c4d617bc1..e3a3d318d2e5 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -40,9 +40,6 @@ struct hda_intel { unsigned int vga_switcheroo_registered:1; unsigned int init_failed:1; /* delayed init failed */ - /* secondary power domain for hdmi audio under vga device */ - struct dev_pm_domain hdmi_pm_domain; - bool need_i915_power:1; /* the hda controller needs i915 power */ }; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 6b4ebaefd8f8..75bdcede04e6 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2546,6 +2546,7 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec, hda_nid_t cvt_nid) { if (per_pin) { + haswell_verify_D0(codec, per_pin->cvt_nid, per_pin->pin_nid); snd_hda_set_dev_select(codec, per_pin->pin_nid, per_pin->dev_id); intel_verify_pin_cvt_connect(codec, per_pin); @@ -3398,6 +3399,7 @@ static int tegra_hdmi_build_pcms(struct hda_codec *codec) static int patch_tegra_hdmi(struct hda_codec *codec) { + struct hdmi_spec *spec; int err; err = patch_generic_hdmi(codec); @@ -3405,6 +3407,10 @@ static int patch_tegra_hdmi(struct hda_codec *codec) return err; codec->patch_ops.build_pcms = tegra_hdmi_build_pcms; + spec = codec->spec; + spec->chmap.ops.chmap_cea_alloc_validate_get_type = + nvhdmi_chmap_cea_alloc_validate_get_type; + spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate; return 0; } @@ -3861,6 +3867,11 @@ HDA_CODEC_ENTRY(0x10de0095, "GPU 95 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0097, "GPU 97 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0098, "GPU 98 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0099, "GPU 99 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de009a, "GPU 9a HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de009d, "GPU 9d HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de009e, "GPU 9e HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de009f, "GPU 9f HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de00a0, "GPU a0 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x10de8067, "MCP67/68 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c4e97b3ba1dd..c27623052264 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -341,6 +341,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0282: case 0x10ec0283: case 0x10ec0286: + case 0x10ec0287: case 0x10ec0288: case 0x10ec0285: case 0x10ec0298: @@ -3153,7 +3154,11 @@ static void alc256_shutup(struct hda_codec *codec) /* 3k pull low control for Headset jack. */ /* NOTE: call this before clearing the pin, otherwise codec stalls */ - alc_update_coef_idx(codec, 0x46, 0, 3 << 12); + /* If disable 3k pulldown control for alc257, the Mic detection will not work correctly + * when booting with headset plugged. So skip setting it for the codec alc257 + */ + if (codec->core.vendor_id != 0x10ec0257) + alc_update_coef_idx(codec, 0x46, 0, 3 << 12); snd_hda_codec_write(codec, hp_pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); @@ -7130,6 +7135,12 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { ALC225_STANDARD_PINS, {0x12, 0xb7a60130}, {0x17, 0x90170110}), + SND_HDA_PIN_QUIRK(0x10ec0623, 0x17aa, "Lenovo", ALC283_FIXUP_HEADSET_MIC, + {0x14, 0x01014010}, + {0x17, 0x90170120}, + {0x18, 0x02a11030}, + {0x19, 0x02a1103f}, + {0x21, 0x0221101f}), {} }; @@ -7290,6 +7301,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0245: case 0x10ec0285: + case 0x10ec0287: case 0x10ec0289: spec->codec_variant = ALC269_TYPE_ALC215; spec->gen.mixer_nid = 0; @@ -8389,6 +8401,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0284, "ALC284", patch_alc269), HDA_CODEC_ENTRY(0x10ec0285, "ALC285", patch_alc269), HDA_CODEC_ENTRY(0x10ec0286, "ALC286", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0287, "ALC287", patch_alc269), HDA_CODEC_ENTRY(0x10ec0288, "ALC288", patch_alc269), HDA_CODEC_ENTRY(0x10ec0289, "ALC289", patch_alc269), HDA_CODEC_ENTRY(0x10ec0290, "ALC290", patch_alc269), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7cd147411b22..f7896a9ae3d6 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -863,7 +863,7 @@ static int stac_auto_create_beep_ctls(struct hda_codec *codec, static struct snd_kcontrol_new beep_vol_ctl = HDA_CODEC_VOLUME(NULL, 0, 0, 0); - /* check for mute support for the the amp */ + /* check for mute support for the amp */ if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { const struct snd_kcontrol_new *temp; if (spec->anabeep_nid == nid) diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 3919aed39ca0..5e52086d7b98 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -31,7 +31,7 @@ * Experimentally I found out that only a combination of * OCKS0=1, OCKS1=1 (128fs, 64fs output) and ice1724 - * VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct - * sampling rate. That means the the FPGA doubles the + * sampling rate. That means that the FPGA doubles the * MCK01 rate. * * Copyright (c) 2003 Takashi Iwai <tiwai@suse.de> diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index 4cf3200e988b..df44135e1b0c 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -39,7 +39,7 @@ * GPIO 4 <- headphone detect * GPIO 5 -> enable ADC analog circuit for the left channel * GPIO 6 -> enable ADC analog circuit for the right channel - * GPIO 7 -> switch green rear output jack between CS4245 and and the first + * GPIO 7 -> switch green rear output jack between CS4245 and the first * channel of CS4361 (mechanical relay) * GPIO 8 -> enable output to speakers * diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 3633eb30dd13..4f949ad50d6a 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -16,8 +16,8 @@ #define CDC_D_REVISION1 (0xf000) #define CDC_D_PERPH_SUBTYPE (0xf005) -#define CDC_D_INT_EN_SET (0x015) -#define CDC_D_INT_EN_CLR (0x016) +#define CDC_D_INT_EN_SET (0xf015) +#define CDC_D_INT_EN_CLR (0xf016) #define MBHC_SWITCH_INT BIT(7) #define MBHC_MIC_ELECTRICAL_INS_REM_DET BIT(6) #define MBHC_BUTTON_PRESS_DET BIT(5) diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 5ba485cae4e6..06d7c0aaeb61 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -760,7 +760,7 @@ #define RT5670_PWR_VREF2_BIT 4 #define RT5670_PWR_FV2 (0x1 << 3) #define RT5670_PWR_FV2_BIT 3 -#define RT5670_LDO_SEL_MASK (0x3) +#define RT5670_LDO_SEL_MASK (0x7) #define RT5670_LDO_SEL_SFT 0 /* Power Management for Analog 2 (0x64) */ diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e10e03800cce..6991718d7c8a 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1747,8 +1747,10 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) PTR_ERR(chan)); return PTR_ERR(chan); } - if (WARN_ON(!chan->device || !chan->device->dev)) + if (WARN_ON(!chan->device || !chan->device->dev)) { + dma_release_channel(chan); return -EINVAL; + } if (chan->device->dev->of_node) ret = of_property_read_string(chan->device->dev->of_node, diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index e1b97e59275a..15d7e6da0555 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -243,6 +243,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream, ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be); if (ret) { dev_err(dev, "failed to config DMA channel for Back-End\n"); + dma_release_channel(pair->dma_chan[dir]); return ret; } diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 4558c8b93036..3a645fc425cd 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -339,7 +339,7 @@ static int sst_media_open(struct snd_pcm_substream *substream, ret_val = power_up_sst(stream); if (ret_val < 0) - return ret_val; + goto out_power_up; /* Make sure, that the period size is always even */ snd_pcm_hw_constraint_step(substream->runtime, 0, @@ -348,8 +348,9 @@ static int sst_media_open(struct snd_pcm_substream *substream, return snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); out_ops: - kfree(stream); mutex_unlock(&sst_lock); +out_power_up: + kfree(stream); return ret_val; } diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index 7843104fadcb..1b01bc318fd2 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -529,6 +529,7 @@ static int bxt_card_late_probe(struct snd_soc_card *card) /* broxton audio machine driver for SPT + RT298S */ static struct snd_soc_card broxton_rt298 = { .name = "broxton-rt298", + .owner = THIS_MODULE, .dai_link = broxton_rt298_dais, .num_links = ARRAY_SIZE(broxton_rt298_dais), .controls = broxton_controls, @@ -544,6 +545,7 @@ static struct snd_soc_card broxton_rt298 = { static struct snd_soc_card geminilake_rt298 = { .name = "geminilake-rt298", + .owner = THIS_MODULE, .dai_link = broxton_rt298_dais, .num_links = ARRAY_SIZE(broxton_rt298_dais), .controls = broxton_controls, diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index cf23af159acf..35ca8e8bb5e5 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -136,7 +136,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED, "kirkwood-i2s", priv); if (err) - return -EBUSY; + return err; /* * Enable Error interrupts. We're only ack'ing them but diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 8a2e3bbce3a1..ad16c8310dd3 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -478,8 +478,10 @@ static int rockchip_pdm_resume(struct device *dev) int ret; ret = pm_runtime_get_sync(dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put(dev); return ret; + } ret = regcache_sync(pdm->regmap); diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 43679aeeb12b..88e838ac937d 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -655,8 +655,10 @@ static int tegra30_ahub_resume(struct device *dev) int ret; ret = pm_runtime_get_sync(dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put(dev); return ret; + } ret = regcache_sync(ahub->regmap_ahub); ret |= regcache_sync(ahub->regmap_apbif); pm_runtime_put(dev); diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 0b176ea24914..bf155c5092f0 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -551,8 +551,10 @@ static int tegra30_i2s_resume(struct device *dev) int ret; ret = pm_runtime_get_sync(dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put(dev); return ret; + } ret = regcache_sync(i2s->regmap); pm_runtime_put(dev); diff --git a/sound/usb/card.c b/sound/usb/card.c index 4169c71f8a32..721f91f5766d 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -768,9 +768,6 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) if (chip == (void *)-1L) return 0; - chip->autosuspended = !!PMSG_IS_AUTO(message); - if (!chip->autosuspended) - snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot); if (!chip->num_suspended_intf++) { list_for_each_entry(as, &chip->pcm_list, list) { snd_pcm_suspend_all(as->pcm); @@ -783,6 +780,11 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) snd_usb_mixer_suspend(mixer); } + if (!PMSG_IS_AUTO(message) && !chip->system_suspend) { + snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot); + chip->system_suspend = chip->num_suspended_intf; + } + return 0; } @@ -795,10 +797,11 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) if (chip == (void *)-1L) return 0; - if (--chip->num_suspended_intf) - return 0; atomic_inc(&chip->active); /* avoid autopm */ + if (chip->num_suspended_intf > 1) + goto out; + /* * ALSA leaves material resumption to user space * we just notify and restart the mixers @@ -813,9 +816,12 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) snd_usbmidi_resume(p); } - if (!chip->autosuspended) + out: + if (chip->num_suspended_intf == chip->system_suspend) { snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0); - chip->autosuspended = 0; + chip->system_suspend = 0; + } + chip->num_suspended_intf--; err_out: atomic_dec(&chip->active); /* allow autopm after this point */ diff --git a/sound/usb/card.h b/sound/usb/card.h index ed87cc83eb47..0cde519bfa42 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -126,6 +126,7 @@ struct snd_usb_substream { unsigned int tx_length_quirk:1; /* add length specifier to transfers */ unsigned int fmt_type; /* USB audio format type (1-3) */ unsigned int pkt_offset_adj; /* Bytes to drop from beginning of packets (for non-compliant devices) */ + unsigned int stream_offset_adj; /* Bytes to drop from beginning of stream (for non-compliant devices) */ unsigned int running: 1; /* running status */ diff --git a/sound/usb/line6/capture.c b/sound/usb/line6/capture.c index 7c812565f90d..a65a82d5791d 100644 --- a/sound/usb/line6/capture.c +++ b/sound/usb/line6/capture.c @@ -291,6 +291,8 @@ int line6_create_audio_in_urbs(struct snd_line6_pcm *line6pcm) urb->interval = LINE6_ISO_INTERVAL; urb->error_count = 0; urb->complete = audio_in_callback; + if (usb_urb_ep_type_check(urb)) + return -EINVAL; } return 0; diff --git a/sound/usb/line6/playback.c b/sound/usb/line6/playback.c index 812d18191e01..1736eb3ee98e 100644 --- a/sound/usb/line6/playback.c +++ b/sound/usb/line6/playback.c @@ -436,6 +436,8 @@ int line6_create_audio_out_urbs(struct snd_line6_pcm *line6pcm) urb->interval = LINE6_ISO_INTERVAL; urb->error_count = 0; urb->complete = audio_out_callback; + if (usb_urb_ep_type_check(urb)) + return -EINVAL; } return 0; diff --git a/sound/usb/midi.c b/sound/usb/midi.c index a92e2b2a91ec..a3d1c0c1b4a6 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1477,6 +1477,8 @@ void snd_usbmidi_disconnect(struct list_head *p) spin_unlock_irq(&umidi->disc_lock); up_write(&umidi->disc_rwsem); + del_timer_sync(&umidi->error_timer); + for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { struct snd_usb_midi_endpoint *ep = &umidi->endpoints[i]; if (ep->out) @@ -1503,7 +1505,6 @@ void snd_usbmidi_disconnect(struct list_head *p) ep->in = NULL; } } - del_timer_sync(&umidi->error_timer); } EXPORT_SYMBOL(snd_usbmidi_disconnect); @@ -1804,6 +1805,28 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi *umidi, return 0; } +static struct usb_ms_endpoint_descriptor *find_usb_ms_endpoint_descriptor( + struct usb_host_endpoint *hostep) +{ + unsigned char *extra = hostep->extra; + int extralen = hostep->extralen; + + while (extralen > 3) { + struct usb_ms_endpoint_descriptor *ms_ep = + (struct usb_ms_endpoint_descriptor *)extra; + + if (ms_ep->bLength > 3 && + ms_ep->bDescriptorType == USB_DT_CS_ENDPOINT && + ms_ep->bDescriptorSubtype == UAC_MS_GENERAL) + return ms_ep; + if (!extra[0]) + break; + extralen -= extra[0]; + extra += extra[0]; + } + return NULL; +} + /* * Returns MIDIStreaming device capabilities. */ @@ -1841,11 +1864,8 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, ep = get_ep_desc(hostep); if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep)) continue; - ms_ep = (struct usb_ms_endpoint_descriptor *)hostep->extra; - if (hostep->extralen < 4 || - ms_ep->bLength < 4 || - ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT || - ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) + ms_ep = find_usb_ms_endpoint_descriptor(hostep); + if (!ms_ep) continue; if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { @@ -2260,16 +2280,22 @@ void snd_usbmidi_input_stop(struct list_head *p) } EXPORT_SYMBOL(snd_usbmidi_input_stop); -static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint *ep) +static void snd_usbmidi_input_start_ep(struct snd_usb_midi *umidi, + struct snd_usb_midi_in_endpoint *ep) { unsigned int i; + unsigned long flags; if (!ep) return; for (i = 0; i < INPUT_URBS; ++i) { struct urb *urb = ep->urbs[i]; - urb->dev = ep->umidi->dev; - snd_usbmidi_submit_urb(urb, GFP_KERNEL); + spin_lock_irqsave(&umidi->disc_lock, flags); + if (!atomic_read(&urb->use_count)) { + urb->dev = ep->umidi->dev; + snd_usbmidi_submit_urb(urb, GFP_ATOMIC); + } + spin_unlock_irqrestore(&umidi->disc_lock, flags); } } @@ -2285,7 +2311,7 @@ void snd_usbmidi_input_start(struct list_head *p) if (umidi->input_running || !umidi->opened[1]) return; for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) - snd_usbmidi_input_start_ep(umidi->endpoints[i].in); + snd_usbmidi_input_start_ep(umidi, umidi->endpoints[i].in); umidi->input_running = 1; } EXPORT_SYMBOL(snd_usbmidi_input_start); diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 7b75208d5cea..b29a3546ab6a 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -591,8 +591,9 @@ static int check_matrix_bitmap(unsigned char *bmap, * if failed, give up and free the control instance. */ -int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list, - struct snd_kcontrol *kctl) +int snd_usb_mixer_add_list(struct usb_mixer_elem_list *list, + struct snd_kcontrol *kctl, + bool is_std_info) { struct usb_mixer_interface *mixer = list->mixer; int err; @@ -605,6 +606,7 @@ int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list, return err; } list->kctl = kctl; + list->is_std_info = is_std_info; list->next_id_elem = mixer->id_elems[list->id]; mixer->id_elems[list->id] = list; return 0; @@ -986,6 +988,14 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->res = 384; } break; + case USB_ID(0x0495, 0x3042): /* ESS Technology Asus USB DAC */ + if ((strstr(kctl->id.name, "Playback Volume") != NULL) || + strstr(kctl->id.name, "Capture Volume") != NULL) { + cval->min >>= 8; + cval->max = 0; + cval->res = 1; + } + break; } } @@ -2395,15 +2405,23 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid) { struct usb_mixer_elem_list *list; - for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) + for_each_mixer_elem(list, mixer, unitid) { + struct usb_mixer_elem_info *info; + + if (!list->is_std_info) + continue; + info = mixer_elem_list_to_info(list); + /* invalidate cache, so the value is read from the device */ + info->cached = 0; snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &list->kctl->id); + } } static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer, struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *cval = (struct usb_mixer_elem_info *)list; + struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); static char *val_types[] = {"BOOLEAN", "INV_BOOLEAN", "S8", "U8", "S16", "U16"}; snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%x, " @@ -2429,8 +2447,7 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, mixer->ignore_ctl_error); snd_iprintf(buffer, "Card: %s\n", chip->card->longname); for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) { - for (list = mixer->id_elems[unitid]; list; - list = list->next_id_elem) { + for_each_mixer_elem(list, mixer, unitid) { snd_iprintf(buffer, " Unit: %i\n", list->id); if (list->kctl) snd_iprintf(buffer, @@ -2460,19 +2477,21 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, return; } - for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) + for_each_mixer_elem(list, mixer, unitid) count++; if (count == 0) return; - for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) { + for_each_mixer_elem(list, mixer, unitid) { struct usb_mixer_elem_info *info; if (!list->kctl) continue; + if (!list->is_std_info) + continue; - info = (struct usb_mixer_elem_info *)list; + info = mixer_elem_list_to_info(list); if (count > 1 && info->control != control) continue; @@ -2692,7 +2711,7 @@ int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer) static int restore_mixer_value(struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *cval = (struct usb_mixer_elem_info *)list; + struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); int c, err, idx; if (cval->cmask) { @@ -2728,8 +2747,7 @@ int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) if (reset_resume) { /* restore cached mixer values */ for (id = 0; id < MAX_ID_ELEMS; id++) { - for (list = mixer->id_elems[id]; list; - list = list->next_id_elem) { + for_each_mixer_elem(list, mixer, id) { if (list->resume) { err = list->resume(list); if (err < 0) diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index ba27f7ade670..7c824a44589b 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -49,10 +49,17 @@ struct usb_mixer_elem_list { struct usb_mixer_elem_list *next_id_elem; /* list of controls with same id */ struct snd_kcontrol *kctl; unsigned int id; + bool is_std_info; usb_mixer_elem_dump_func_t dump; usb_mixer_elem_resume_func_t resume; }; +/* iterate over mixer element list of the given unit id */ +#define for_each_mixer_elem(list, mixer, id) \ + for ((list) = (mixer)->id_elems[id]; (list); (list) = (list)->next_id_elem) +#define mixer_elem_list_to_info(list) \ + container_of(list, struct usb_mixer_elem_info, head) + struct usb_mixer_elem_info { struct usb_mixer_elem_list head; unsigned int control; /* CS or ICN (high byte) */ @@ -80,8 +87,12 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid); int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set); -int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list, - struct snd_kcontrol *kctl); +int snd_usb_mixer_add_list(struct usb_mixer_elem_list *list, + struct snd_kcontrol *kctl, + bool is_std_info); + +#define snd_usb_mixer_add_control(list, kctl) \ + snd_usb_mixer_add_list(list, kctl, true) void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, struct usb_mixer_interface *mixer, diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index b9ea4a42aee4..d7878ed5ecc0 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -169,7 +169,8 @@ static int add_single_ctl_with_resume(struct usb_mixer_interface *mixer, return -ENOMEM; } kctl->private_free = snd_usb_mixer_elem_free; - return snd_usb_mixer_add_control(list, kctl); + /* don't use snd_usb_mixer_add_control() here, this is a special list element */ + return snd_usb_mixer_add_list(list, kctl, false); } /* @@ -195,6 +196,7 @@ static const struct rc_config { { USB_ID(0x041e, 0x3042), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 */ { USB_ID(0x041e, 0x30df), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 Pro */ { USB_ID(0x041e, 0x3237), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 Pro */ + { USB_ID(0x041e, 0x3263), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 Pro */ { USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */ }; @@ -1171,7 +1173,7 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, int unitid = 12; /* SamleRate ExtensionUnit ID */ list_for_each_entry(mixer, &chip->mixer_list, list) { - cval = (struct usb_mixer_elem_info *)mixer->id_elems[unitid]; + cval = mixer_elem_list_to_info(mixer->id_elems[unitid]); if (cval) { snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, cval->control << 8, diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index c33e2378089d..4aeb9488a0c9 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -287,8 +287,7 @@ static int scarlett_ctl_switch_put(struct snd_kcontrol *kctl, static int scarlett_ctl_resume(struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *elem = - container_of(list, struct usb_mixer_elem_info, head); + struct usb_mixer_elem_info *elem = mixer_elem_list_to_info(list); int i; for (i = 0; i < elem->channels; i++) @@ -447,8 +446,7 @@ static int scarlett_ctl_enum_put(struct snd_kcontrol *kctl, static int scarlett_ctl_enum_resume(struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *elem = - container_of(list, struct usb_mixer_elem_info, head); + struct usb_mixer_elem_info *elem = mixer_elem_list_to_info(list); if (elem->cached) snd_usb_set_cur_mix_value(elem, 0, 0, *elem->cache_val); diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index ff38fca1781b..f27213b846e6 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1313,6 +1313,12 @@ static void retire_capture_urb(struct snd_usb_substream *subs, // continue; } bytes = urb->iso_frame_desc[i].actual_length; + if (subs->stream_offset_adj > 0) { + unsigned int adj = min(subs->stream_offset_adj, bytes); + cp += adj; + bytes -= adj; + subs->stream_offset_adj -= adj; + } frames = bytes / stride; if (!subs->txfr_quirk) bytes = frames * stride; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c892b4d1e733..a917b7e02d31 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3323,4 +3323,118 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +/* + * MacroSilicon MS2109 based HDMI capture cards + * + * These claim 96kHz 1ch in the descriptors, but are actually 48kHz 2ch. + * They also need QUIRK_AUDIO_ALIGN_TRANSFER, which makes one wonder if + * they pretend to be 96kHz mono as a workaround for stereo being broken + * by that... + * + * They also have an issue with initial stream alignment that causes the + * channels to be swapped and out of phase, which is dealt with in quirks.c. + */ +{ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x534d, + .idProduct = 0x2109, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "MacroSilicon", + .product_name = "MS2109", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 2, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 2, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 48000, + .rate_max = 48000, + } + }, + { + .ifnum = -1 + } + } + } +}, +{ + /* + * PIONEER DJ DDJ-RB + * PCM is 4 channels out, 2 dummy channels in @ 44.1 fixed + * The feedback for the output is the dummy input. + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x000e), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 2, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = -1 + } + } + } +}, + #undef USB_DEVICE_VENDOR_SPEC diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index f29a8ed4f856..224e0a760428 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1120,6 +1120,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; + case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ + subs->stream_offset_adj = 2; + break; } } @@ -1164,6 +1167,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) static bool is_itf_usb_dsd_2alts_dac(unsigned int id) { switch (id) { + case USB_ID(0x154e, 0x1002): /* Denon DCD-1500RE */ case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */ case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ @@ -1318,12 +1322,13 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); - /* Zoom R16/24, Logitech H650e, Jabra 550a, Kingston HyperX needs a tiny - * delay here, otherwise requests like get/set frequency return as - * failed despite actually succeeding. + /* Zoom R16/24, Logitech H650e/H570e, Jabra 550a, Kingston HyperX + * needs a tiny delay here, otherwise requests like get/set + * frequency return as failed despite actually succeeding. */ if ((chip->usb_id == USB_ID(0x1686, 0x00dd) || chip->usb_id == USB_ID(0x046d, 0x0a46) || + chip->usb_id == USB_ID(0x046d, 0x0a56) || chip->usb_id == USB_ID(0x0b0e, 0x0349) || chip->usb_id == USB_ID(0x0951, 0x16ad)) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index d1776e5517ff..452646959586 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -95,6 +95,7 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, subs->tx_length_quirk = as->chip->tx_length_quirk; subs->speed = snd_usb_get_speed(subs->dev); subs->pkt_offset_adj = 0; + subs->stream_offset_adj = 0; snd_usb_set_pcm_ops(as->pcm, stream); diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 4d5c89a7ba2b..f4ee83c8e0b2 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -37,7 +37,7 @@ struct snd_usb_audio { struct usb_interface *pm_intf; u32 usb_id; struct mutex mutex; - unsigned int autosuspended:1; + unsigned int system_suspend; atomic_t active; atomic_t shutdown; atomic_t usage_count; |