diff options
Diffstat (limited to 'sound')
-rw-r--r-- | sound/pci/hda/hda_codec.c | 2 | ||||
-rw-r--r-- | sound/pci/hda/hda_eld.c | 2 | ||||
-rw-r--r-- | sound/pci/hda/hda_generic.c | 2 | ||||
-rw-r--r-- | sound/pci/hda/hda_intel.c | 6 | ||||
-rw-r--r-- | sound/pci/hda/patch_hdmi.c | 2 | ||||
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 4 | ||||
-rw-r--r--[-rwxr-xr-x] | sound/soc/codecs/max98090.c | 0 | ||||
-rw-r--r--[-rwxr-xr-x] | sound/soc/codecs/max98090.h | 0 | ||||
-rw-r--r-- | sound/soc/codecs/si476x.c | 1 | ||||
-rw-r--r-- | sound/soc/codecs/wm5102.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/wm8903.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/wm_adsp.c | 5 | ||||
-rw-r--r-- | sound/soc/fsl/imx-ssi.c | 5 | ||||
-rw-r--r-- | sound/soc/fsl/pcm030-audio-fabric.c | 2 | ||||
-rw-r--r-- | sound/soc/samsung/i2s.c | 17 | ||||
-rw-r--r-- | sound/soc/sh/dma-sh7760.c | 4 | ||||
-rw-r--r-- | sound/soc/soc-compress.c | 14 | ||||
-rw-r--r-- | sound/soc/soc-core.c | 10 | ||||
-rw-r--r-- | sound/soc/soc-dapm.c | 14 | ||||
-rw-r--r-- | sound/soc/spear/spear_pcm.c | 12 | ||||
-rw-r--r-- | sound/soc/tegra/tegra_pcm.c | 24 | ||||
-rw-r--r-- | sound/usb/clock.c | 45 | ||||
-rw-r--r-- | sound/usb/mixer_quirks.c | 4 | ||||
-rw-r--r-- | sound/usb/quirks.c | 2 |
24 files changed, 111 insertions, 70 deletions
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ecdf30e..4aba764 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) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 7dd8463..d0d7ac1 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 43c2ea5..2dbe767 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) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 418bfc0..bcd40ee 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 */ @@ -2931,8 +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; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 78e1827..de8ac5c 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 563c24d..f15c36b 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/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index fc17604..fc17604 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 7e103f2..7e103f2 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 f2d61a1..566ea32 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 b82bbf5..34d0201 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 134e41c..f8a31ad 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 f3f7e75..9af1bdd 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 dce05b6..7c84eb1 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 8e52c14..eb43738 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 d7231e3..6bbeb0b 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 19eff8f..1a8b03e 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 b5b3db7..ed0bfb0 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 0e64af1..78468c6 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; @@ -4237,7 +4239,6 @@ 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, @@ -4246,7 +4247,6 @@ 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 625d4824..33acd8b 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; } } diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index ba66a3f..d653763 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -136,9 +136,9 @@ 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) @@ -146,16 +146,16 @@ 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 32d0811..f9f247c 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, @@ -111,26 +109,6 @@ 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) { @@ -148,7 +126,7 @@ 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/clock.c b/sound/usb/clock.c index 5e634a2..9e2703a 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_quirks.c b/sound/usb/mixer_quirks.c index 497d274..ebe9144 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 5325a38..9c5ab22 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; |