diff options
52 files changed, 1160 insertions, 386 deletions
diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 392875a1b94e..72f97d4b01a7 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -2234,6 +2234,19 @@ use_vmalloc buffers. If mmap is used on such architectures, turn off this option, so that the DMA-coherent buffers are allocated and used instead. +delayed_register + The option is needed for devices that have multiple streams + defined in multiple USB interfaces. The driver may invoke + registrations multiple times (once per interface) and this may + lead to the insufficient device enumeration. + This option receives an array of strings, and you can pass + ID:INTERFACE like ``0123abcd:4`` for performing the delayed + registration to the given device. In this example, when a USB + device 0123:abcd is probed, the driver waits the registration + until the USB interface 4 gets probed. + The driver prints a message like "Found post-registration device + assignment: 1234abcd:04" for such a device, so that user can + notice the need. This module supports multiple devices, autoprobe and hotplugging. diff --git a/include/linux/usb/audio-v2.h b/include/linux/usb/audio-v2.h index ba4b3e3327ff..cb9900b34b67 100644 --- a/include/linux/usb/audio-v2.h +++ b/include/linux/usb/audio-v2.h @@ -156,6 +156,18 @@ struct uac2_feature_unit_descriptor { __u8 bmaControls[0]; /* variable length */ } __attribute__((packed)); +/* 4.7.2.10 Effect Unit Descriptor */ + +struct uac2_effect_unit_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bUnitID; + __le16 wEffectType; + __u8 bSourceID; + __u8 bmaControls[]; /* variable length */ +} __attribute__((packed)); + /* 4.9.2 Class-Specific AS Interface Descriptor */ struct uac2_as_header_descriptor { diff --git a/include/sound/core.h b/include/sound/core.h index ac8b692b69b4..381a010a1bd4 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -266,6 +266,7 @@ void snd_device_disconnect(struct snd_card *card, void *device_data); void snd_device_disconnect_all(struct snd_card *card); void snd_device_free(struct snd_card *card, void *device_data); void snd_device_free_all(struct snd_card *card); +int snd_device_get_state(struct snd_card *card, void *device_data); /* isadma.c */ diff --git a/include/sound/pcm.h b/include/sound/pcm.h index f657ff08f317..31a4b300e4c9 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1415,6 +1415,15 @@ static inline u64 pcm_format_to_bits(snd_pcm_format_t pcm_format) return 1ULL << (__force int) pcm_format; } +/** + * pcm_for_each_format - helper to iterate for each format type + * @f: the iterator variable in snd_pcm_format_t type + */ +#define pcm_for_each_format(f) \ + for ((f) = SNDRV_PCM_FORMAT_FIRST; \ + (__force int)(f) <= (__force int)SNDRV_PCM_FORMAT_LAST; \ + (f) = (__force snd_pcm_format_t)((__force int)(f) + 1)) + /* printk helpers */ #define pcm_err(pcm, fmt, args...) \ dev_err((pcm)->card->dev, fmt, ##args) diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 661450a2095b..36f94735d23d 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -133,6 +133,13 @@ static inline int snd_mask_test(const struct snd_mask *mask, unsigned int val) return mask->bits[MASK_OFS(val)] & MASK_BIT(val); } +/* Most of drivers need only this one */ +static inline int snd_mask_test_format(const struct snd_mask *mask, + snd_pcm_format_t format) +{ + return snd_mask_test(mask, (__force unsigned int)format); +} + static inline int snd_mask_single(const struct snd_mask *mask) { int i, c = 0; diff --git a/sound/core/device.c b/sound/core/device.c index cdc5af526739..bf0b04a7ee79 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -237,3 +237,24 @@ void snd_device_free_all(struct snd_card *card) list_for_each_entry_safe_reverse(dev, next, &card->devices, list) __snd_device_free(dev); } + +/** + * snd_device_get_state - Get the current state of the given device + * @card: the card instance + * @device_data: the data pointer to release + * + * Returns the current state of the given device object. For the valid + * device, either @SNDRV_DEV_BUILD, @SNDRV_DEV_REGISTERED or + * @SNDRV_DEV_DISCONNECTED is returned. + * Or for a non-existing device, -1 is returned as an error. + */ +int snd_device_get_state(struct snd_card *card, void *device_data) +{ + struct snd_device *dev; + + dev = look_for_dev(card, device_data); + if (dev) + return dev->state; + return -1; +} +EXPORT_SYMBOL_GPL(snd_device_get_state); diff --git a/sound/core/info.c b/sound/core/info.c index ca87ae4c30ba..8c6bc5241df5 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -604,7 +604,7 @@ int snd_info_card_free(struct snd_card *card) */ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) { - int c = -1; + int c; if (snd_BUG_ON(!buffer || !buffer->buffer)) return 1; diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 13db77771f0f..930def8201f4 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -884,20 +884,17 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) sformat = snd_pcm_plug_slave_format(format, sformat_mask); if ((__force int)sformat < 0 || - !snd_mask_test(sformat_mask, (__force int)sformat)) { - for (sformat = (__force snd_pcm_format_t)0; - (__force int)sformat <= (__force int)SNDRV_PCM_FORMAT_LAST; - sformat = (__force snd_pcm_format_t)((__force int)sformat + 1)) { - if (snd_mask_test(sformat_mask, (__force int)sformat) && + !snd_mask_test_format(sformat_mask, sformat)) { + pcm_for_each_format(sformat) { + if (snd_mask_test_format(sformat_mask, sformat) && snd_pcm_oss_format_to(sformat) >= 0) - break; - } - if ((__force int)sformat > (__force int)SNDRV_PCM_FORMAT_LAST) { - pcm_dbg(substream->pcm, "Cannot find a format!!!\n"); - err = -EINVAL; - goto failure; + goto format_found; } + pcm_dbg(substream->pcm, "Cannot find a format!!!\n"); + err = -EINVAL; + goto failure; } + format_found: err = _snd_pcm_hw_param_set(sparams, SNDRV_PCM_HW_PARAM_FORMAT, (__force int)sformat, 0); if (err < 0) goto failure; @@ -1220,8 +1217,10 @@ snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const if (ret < 0) break; } + mutex_unlock(&runtime->oss.params_lock); ret = __snd_pcm_lib_xfer(substream, (void *)ptr, true, frames, in_kernel); + mutex_lock(&runtime->oss.params_lock); if (ret != -EPIPE && ret != -ESTRPIPE) break; /* test, if we can't store new data, because the stream */ @@ -1257,8 +1256,10 @@ snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream, char *p ret = snd_pcm_oss_capture_position_fixup(substream, &delay); if (ret < 0) break; + mutex_unlock(&runtime->oss.params_lock); ret = __snd_pcm_lib_xfer(substream, (void *)ptr, true, frames, in_kernel); + mutex_lock(&runtime->oss.params_lock); if (ret == -EPIPE) { if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) { ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL); diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 752d078908e9..fbda4ebf38b3 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -196,82 +196,74 @@ int snd_pcm_plugin_free(struct snd_pcm_plugin *plugin) return 0; } -snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames) +static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug, + snd_pcm_sframes_t frames) { - struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next; - int stream; + struct snd_pcm_plugin *plugin, *plugin_next; - if (snd_BUG_ON(!plug)) - return -ENXIO; - if (drv_frames == 0) - return 0; - stream = snd_pcm_plug_stream(plug); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - plugin = snd_pcm_plug_last(plug); - while (plugin && drv_frames > 0) { - if (drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; - plugin_prev = plugin->prev; - if (plugin->src_frames) - drv_frames = plugin->src_frames(plugin, drv_frames); - plugin = plugin_prev; + plugin = snd_pcm_plug_first(plug); + while (plugin && frames > 0) { + plugin_next = plugin->next; + if (plugin->dst_frames) { + frames = plugin->dst_frames(plugin, frames); + if (frames < 0) + return frames; } - } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { - plugin = snd_pcm_plug_first(plug); - while (plugin && drv_frames > 0) { - plugin_next = plugin->next; - if (plugin->dst_frames) - drv_frames = plugin->dst_frames(plugin, drv_frames); - if (drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; - plugin = plugin_next; + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; + plugin = plugin_next; + } + return frames; +} + +static snd_pcm_sframes_t calc_src_frames(struct snd_pcm_substream *plug, + snd_pcm_sframes_t frames) +{ + struct snd_pcm_plugin *plugin, *plugin_prev; + + plugin = snd_pcm_plug_last(plug); + while (plugin && frames > 0) { + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; + plugin_prev = plugin->prev; + if (plugin->src_frames) { + frames = plugin->src_frames(plugin, frames); + if (frames < 0) + return frames; } - } else + plugin = plugin_prev; + } + return frames; +} + +snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames) +{ + if (snd_BUG_ON(!plug)) + return -ENXIO; + switch (snd_pcm_plug_stream(plug)) { + case SNDRV_PCM_STREAM_PLAYBACK: + return calc_src_frames(plug, drv_frames); + case SNDRV_PCM_STREAM_CAPTURE: + return calc_dst_frames(plug, drv_frames); + default: snd_BUG(); - return drv_frames; + return -EINVAL; + } } snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t clt_frames) { - struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next; - snd_pcm_sframes_t frames; - int stream; - if (snd_BUG_ON(!plug)) return -ENXIO; - if (clt_frames == 0) - return 0; - frames = clt_frames; - stream = snd_pcm_plug_stream(plug); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - plugin = snd_pcm_plug_first(plug); - while (plugin && frames > 0) { - plugin_next = plugin->next; - if (plugin->dst_frames) { - frames = plugin->dst_frames(plugin, frames); - if (frames < 0) - return frames; - } - if (frames > plugin->buf_frames) - frames = plugin->buf_frames; - plugin = plugin_next; - } - } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { - plugin = snd_pcm_plug_last(plug); - while (plugin) { - if (frames > plugin->buf_frames) - frames = plugin->buf_frames; - plugin_prev = plugin->prev; - if (plugin->src_frames) { - frames = plugin->src_frames(plugin, frames); - if (frames < 0) - return frames; - } - plugin = plugin_prev; - } - } else + switch (snd_pcm_plug_stream(plug)) { + case SNDRV_PCM_STREAM_PLAYBACK: + return calc_dst_frames(plug, clt_frames); + case SNDRV_PCM_STREAM_CAPTURE: + return calc_src_frames(plug, clt_frames); + default: snd_BUG(); - return frames; + return -EINVAL; + } } static int snd_pcm_plug_formats(const struct snd_mask *mask, diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c index 7cd09cef6961..d381f4c967c9 100644 --- a/sound/core/oss/rate.c +++ b/sound/core/oss/rate.c @@ -47,7 +47,7 @@ struct rate_priv { unsigned int pos; rate_f func; snd_pcm_sframes_t old_src_frames, old_dst_frames; - struct rate_channel channels[0]; + struct rate_channel channels[]; }; static void rate_init(struct snd_pcm_plugin *plugin) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index a141a301369f..b6d2331a82f7 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1019,7 +1019,7 @@ static ssize_t show_pcm_class(struct device *dev, str = "none"; else str = strs[pcm->dev_class]; - return snprintf(buf, PAGE_SIZE, "%s\n", str); + return sprintf(buf, "%s\n", str); } static DEVICE_ATTR(pcm_class, 0444, show_pcm_class, NULL); diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 5749a8a49784..b37c70864b57 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -426,7 +426,7 @@ int snd_dmaengine_pcm_refine_runtime_hwparams( * default assumption is that it supports 1, 2 and 4 bytes * widths. */ - for (i = SNDRV_PCM_FORMAT_FIRST; i <= SNDRV_PCM_FORMAT_LAST; i++) { + pcm_for_each_format(i) { int bits = snd_pcm_format_physical_width(i); /* diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index a6a541511534..cf3e8c26e00a 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -42,6 +42,11 @@ struct pcm_format_data { /* we do lots of calculations on snd_pcm_format_t; shut up sparse */ #define INT __force int +static bool valid_format(snd_pcm_format_t format) +{ + return (INT)format >= 0 && (INT)format <= (INT)SNDRV_PCM_FORMAT_LAST; +} + static const struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = { [SNDRV_PCM_FORMAT_S8] = { .width = 8, .phys = 8, .le = -1, .signd = 1, @@ -259,7 +264,7 @@ static const struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = int snd_pcm_format_signed(snd_pcm_format_t format) { int val; - if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) + if (!valid_format(format)) return -EINVAL; if ((val = pcm_formats[(INT)format].signd) < 0) return -EINVAL; @@ -307,7 +312,7 @@ EXPORT_SYMBOL(snd_pcm_format_linear); int snd_pcm_format_little_endian(snd_pcm_format_t format) { int val; - if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) + if (!valid_format(format)) return -EINVAL; if ((val = pcm_formats[(INT)format].le) < 0) return -EINVAL; @@ -343,7 +348,7 @@ EXPORT_SYMBOL(snd_pcm_format_big_endian); int snd_pcm_format_width(snd_pcm_format_t format) { int val; - if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) + if (!valid_format(format)) return -EINVAL; if ((val = pcm_formats[(INT)format].width) == 0) return -EINVAL; @@ -361,7 +366,7 @@ EXPORT_SYMBOL(snd_pcm_format_width); int snd_pcm_format_physical_width(snd_pcm_format_t format) { int val; - if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) + if (!valid_format(format)) return -EINVAL; if ((val = pcm_formats[(INT)format].phys) == 0) return -EINVAL; @@ -394,7 +399,7 @@ EXPORT_SYMBOL(snd_pcm_format_size); */ const unsigned char *snd_pcm_format_silence_64(snd_pcm_format_t format) { - if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) + if (!valid_format(format)) return NULL; if (! pcm_formats[(INT)format].phys) return NULL; @@ -418,7 +423,7 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int unsigned char *dst; const unsigned char *pat; - if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) + if (!valid_format(format)) return -EINVAL; if (samples == 0) return 0; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index d5443eeb8b63..8e1060b084f1 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -228,6 +228,9 @@ int snd_pcm_info_user(struct snd_pcm_substream *substream, return err; } +/* macro for simplified cast */ +#define PARAM_MASK_BIT(b) (1U << (__force int)(b)) + static bool hw_support_mmap(struct snd_pcm_substream *substream) { if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP)) @@ -257,7 +260,7 @@ static int constrain_mask_params(struct snd_pcm_substream *substream, return -EINVAL; /* This parameter is not requested to change by a caller. */ - if (!(params->rmask & (1 << k))) + if (!(params->rmask & PARAM_MASK_BIT(k))) continue; if (trace_hw_mask_param_enabled()) @@ -271,7 +274,7 @@ static int constrain_mask_params(struct snd_pcm_substream *substream, /* Set corresponding flag so that the caller gets it. */ trace_hw_mask_param(substream, k, 0, &old_mask, m); - params->cmask |= 1 << k; + params->cmask |= PARAM_MASK_BIT(k); } return 0; @@ -293,7 +296,7 @@ static int constrain_interval_params(struct snd_pcm_substream *substream, return -EINVAL; /* This parameter is not requested to change by a caller. */ - if (!(params->rmask & (1 << k))) + if (!(params->rmask & PARAM_MASK_BIT(k))) continue; if (trace_hw_interval_param_enabled()) @@ -307,7 +310,7 @@ static int constrain_interval_params(struct snd_pcm_substream *substream, /* Set corresponding flag so that the caller gets it. */ trace_hw_interval_param(substream, k, 0, &old_interval, i); - params->cmask |= 1 << k; + params->cmask |= PARAM_MASK_BIT(k); } return 0; @@ -349,7 +352,7 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, * have 0 so that the parameters are never changed anymore. */ for (k = 0; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++) - vstamps[k] = (params->rmask & (1 << k)) ? 1 : 0; + vstamps[k] = (params->rmask & PARAM_MASK_BIT(k)) ? 1 : 0; /* Due to the above design, actual sequence number starts at 2. */ stamp = 2; @@ -417,7 +420,7 @@ retry: hw_param_interval(params, r->var)); } - params->cmask |= (1 << r->var); + params->cmask |= PARAM_MASK_BIT(r->var); vstamps[r->var] = stamp; again = true; } @@ -486,9 +489,9 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, params->info = 0; params->fifo_size = 0; - if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_SAMPLE_BITS)) + if (params->rmask & PARAM_MASK_BIT(SNDRV_PCM_HW_PARAM_SAMPLE_BITS)) params->msbits = 0; - if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_RATE)) { + if (params->rmask & PARAM_MASK_BIT(SNDRV_PCM_HW_PARAM_RATE)) { params->rate_num = 0; params->rate_den = 0; } @@ -2293,21 +2296,21 @@ static int snd_pcm_hw_rule_mulkdiv(struct snd_pcm_hw_params *params, static int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - unsigned int k; + snd_pcm_format_t k; const struct snd_interval *i = hw_param_interval_c(params, rule->deps[0]); struct snd_mask m; struct snd_mask *mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); snd_mask_any(&m); - for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) { + pcm_for_each_format(k) { int bits; - if (! snd_mask_test(mask, k)) + if (!snd_mask_test_format(mask, k)) continue; bits = snd_pcm_format_physical_width(k); if (bits <= 0) continue; /* ignore invalid formats */ if ((unsigned)bits < i->min || (unsigned)bits > i->max) - snd_mask_reset(&m, k); + snd_mask_reset(&m, (__force unsigned)k); } return snd_mask_refine(mask, &m); } @@ -2316,14 +2319,15 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_interval t; - unsigned int k; + snd_pcm_format_t k; + t.min = UINT_MAX; t.max = 0; t.openmin = 0; t.openmax = 0; - for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) { + pcm_for_each_format(k) { int bits; - if (! snd_mask_test(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k)) + if (!snd_mask_test_format(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k)) continue; bits = snd_pcm_format_physical_width(k); if (bits <= 0) @@ -2505,16 +2509,16 @@ static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) unsigned int mask = 0; if (hw->info & SNDRV_PCM_INFO_INTERLEAVED) - mask |= 1 << SNDRV_PCM_ACCESS_RW_INTERLEAVED; + mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_RW_INTERLEAVED); if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED) - mask |= 1 << SNDRV_PCM_ACCESS_RW_NONINTERLEAVED; + mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_RW_NONINTERLEAVED); if (hw_support_mmap(substream)) { if (hw->info & SNDRV_PCM_INFO_INTERLEAVED) - mask |= 1 << SNDRV_PCM_ACCESS_MMAP_INTERLEAVED; + mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_INTERLEAVED); if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED) - mask |= 1 << SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED; + mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED); if (hw->info & SNDRV_PCM_INFO_COMPLEX) - mask |= 1 << SNDRV_PCM_ACCESS_MMAP_COMPLEX; + mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_COMPLEX); } err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_ACCESS, mask); if (err < 0) @@ -2524,7 +2528,8 @@ static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) if (err < 0) return err; - err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_SUBFORMAT, 1 << SNDRV_PCM_SUBFORMAT_STD); + err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_SUBFORMAT, + PARAM_MASK_BIT(SNDRV_PCM_SUBFORMAT_STD)); if (err < 0) return err; diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index d78a27271d6d..251eaf1152e2 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -118,7 +118,7 @@ struct loopback_cable { struct loopback_setup { unsigned int notify: 1; unsigned int rate_shift; - unsigned int format; + snd_pcm_format_t format; unsigned int rate; unsigned int channels; struct snd_ctl_elem_id active_id; @@ -1432,7 +1432,7 @@ static int loopback_format_info(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = SNDRV_PCM_FORMAT_LAST; + uinfo->value.integer.max = (__force int)SNDRV_PCM_FORMAT_LAST; uinfo->value.integer.step = 1; return 0; } @@ -1443,7 +1443,7 @@ static int loopback_format_get(struct snd_kcontrol *kcontrol, struct loopback *loopback = snd_kcontrol_chip(kcontrol); ucontrol->value.integer.value[0] = - loopback->setup[kcontrol->id.subdevice] + (__force int)loopback->setup[kcontrol->id.subdevice] [kcontrol->id.device].format; return 0; } diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 02ac3f4e0c02..b5486de08b97 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -901,10 +901,10 @@ static int snd_card_dummy_new_mixer(struct snd_dummy *dummy) static void print_formats(struct snd_dummy *dummy, struct snd_info_buffer *buffer) { - int i; + snd_pcm_format_t i; - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - if (dummy->pcm_hw.formats & (1ULL << i)) + pcm_for_each_format(i) { + if (dummy->pcm_hw.formats & pcm_format_to_bits(i)) snd_iprintf(buffer, " %s", snd_pcm_format_name(i)); } } diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 976d8cb9a34f..2c8e3392a490 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -509,7 +509,7 @@ MODULE_DEVICE_TABLE(ieee1394, bebob_id_table); static struct fw_driver bebob_driver = { .driver = { .owner = THIS_MODULE, - .name = "snd-bebob", + .name = KBUILD_MODNAME, .bus = &fw_bus_type, }, .probe = bebob_probe, diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c index 1f5fc0e7c024..c84b913a9fe0 100644 --- a/sound/firewire/digi00x/digi00x.c +++ b/sound/firewire/digi00x/digi00x.c @@ -192,7 +192,7 @@ MODULE_DEVICE_TABLE(ieee1394, snd_dg00x_id_table); static struct fw_driver dg00x_driver = { .driver = { .owner = THIS_MODULE, - .name = "snd-firewire-digi00x", + .name = KBUILD_MODNAME, .bus = &fw_bus_type, }, .probe = snd_dg00x_probe, diff --git a/sound/firewire/fireface/ff.c b/sound/firewire/fireface/ff.c index f5a016560eb8..b62a4fd22407 100644 --- a/sound/firewire/fireface/ff.c +++ b/sound/firewire/fireface/ff.c @@ -224,7 +224,7 @@ MODULE_DEVICE_TABLE(ieee1394, snd_ff_id_table); static struct fw_driver ff_driver = { .driver = { .owner = THIS_MODULE, - .name = "snd-fireface", + .name = KBUILD_MODNAME, .bus = &fw_bus_type, }, .probe = snd_ff_probe, diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 134fc9ee26b9..b1cc013a3540 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -362,7 +362,7 @@ MODULE_DEVICE_TABLE(ieee1394, efw_id_table); static struct fw_driver efw_driver = { .driver = { .owner = THIS_MODULE, - .name = "snd-fireworks", + .name = KBUILD_MODNAME, .bus = &fw_bus_type, }, .probe = efw_probe, diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c index c29a97f6f638..6f38335fe10b 100644 --- a/sound/firewire/tascam/tascam-hwdep.c +++ b/sound/firewire/tascam/tascam-hwdep.c @@ -17,6 +17,7 @@ static long tscm_hwdep_read_locked(struct snd_tscm *tscm, char __user *buf, long count, loff_t *offset) + __releases(&tscm->lock) { struct snd_firewire_event_lock_status event = { .type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS, @@ -36,6 +37,7 @@ static long tscm_hwdep_read_locked(struct snd_tscm *tscm, char __user *buf, static long tscm_hwdep_read_queue(struct snd_tscm *tscm, char __user *buf, long remained, loff_t *offset) + __releases(&tscm->lock) { char __user *pos = buf; unsigned int type = SNDRV_FIREWIRE_EVENT_TASCAM_CONTROL; diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index addc464503bc..5dac0d9fc58e 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -224,7 +224,7 @@ MODULE_DEVICE_TABLE(ieee1394, snd_tscm_id_table); static struct fw_driver tscm_driver = { .driver = { .owner = THIS_MODULE, - .name = "snd-firewire-tascam", + .name = KBUILD_MODNAME, .bus = &fw_bus_type, }, .probe = snd_tscm_probe, diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 9a526aeef8da..e3119f5cb0d5 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -204,7 +204,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_set_chip_name); */ int snd_hdac_codec_modalias(struct hdac_device *codec, char *buf, size_t size) { - return snprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n", + return scnprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n", codec->vendor_id, codec->revision_id, codec->type); } EXPORT_SYMBOL_GPL(snd_hdac_codec_modalias); diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c index e377ac93f37f..8e8257c574b0 100644 --- a/sound/isa/sb/emu8000_pcm.c +++ b/sound/isa/sb/emu8000_pcm.c @@ -435,7 +435,7 @@ enum { #define LOOP_WRITE(rec, offset, _buf, count, mode) \ do { \ struct snd_emu8000 *emu = (rec)->emu; \ - unsigned short *buf = (unsigned short *)(_buf); \ + unsigned short *buf = (__force unsigned short *)(_buf); \ snd_emu8000_write_wait(emu, 1); \ EMU8000_SMALW_WRITE(emu, offset); \ while (count > 0) { \ @@ -492,7 +492,7 @@ static int emu8k_pcm_silence(struct snd_pcm_substream *subs, #define LOOP_WRITE(rec, pos, _buf, count, mode) \ do { \ struct snd_emu8000 *emu = rec->emu; \ - unsigned short *buf = (unsigned short *)(_buf); \ + unsigned short *buf = (__force unsigned short *)(_buf); \ snd_emu8000_write_wait(emu, 1); \ EMU8000_SMALW_WRITE(emu, pos + rec->loop_start[0]); \ if (rec->voices > 1) \ diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 4f524a9dbbca..4462375d2d82 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -1070,7 +1070,7 @@ static int snd_ali_trigger(struct snd_pcm_substream *substream, { struct snd_ali *codec = snd_pcm_substream_chip(substream); struct snd_pcm_substream *s; - unsigned int what, whati, capture_flag; + unsigned int what, whati; struct snd_ali_voice *pvoice, *evoice; unsigned int val; int do_start; @@ -1088,7 +1088,7 @@ static int snd_ali_trigger(struct snd_pcm_substream *substream, return -EINVAL; } - what = whati = capture_flag = 0; + what = whati = 0; snd_pcm_group_for_each_entry(s, substream) { if ((struct snd_ali *) snd_pcm_substream_chip(s) == codec) { pvoice = s->runtime->private_data; @@ -1110,8 +1110,6 @@ static int snd_ali_trigger(struct snd_pcm_substream *substream, evoice->running = 0; } snd_pcm_trigger_done(s, substream); - if (pvoice->mode) - capture_flag = 1; } } spin_lock(&codec->reg_lock); diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index a89a7e603ca8..6ff581733a19 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1789,6 +1789,7 @@ int snd_emu10k1_create(struct snd_card *card, int idx, err; int is_audigy; size_t page_table_size; + __le32 *pgtbl; unsigned int silent_page; const struct snd_emu_chip_details *c; static const struct snd_device_ops ops = { @@ -2009,8 +2010,9 @@ int snd_emu10k1_create(struct snd_card *card, /* Clear silent pages and set up pointers */ memset(emu->silent_page.area, 0, emu->silent_page.bytes); silent_page = emu->silent_page.addr << emu->address_mode; + pgtbl = (__le32 *)emu->ptb_pages.area; for (idx = 0; idx < (emu->address_mode ? MAXPAGES1 : MAXPAGES0); idx++) - ((u32 *)emu->ptb_pages.area)[idx] = cpu_to_le32(silent_page | idx); + pgtbl[idx] = cpu_to_le32(silent_page | idx); /* set up voice indices */ for (idx = 0; idx < NUM_G; idx++) { diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index bd48335d09d7..e1d3082a4fe9 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -184,6 +184,7 @@ comment "Set to Y if you want auto-loading the codec driver" config SND_HDA_CODEC_CA0132_DSP bool "Support new DSP code for CA0132 codec" depends on SND_HDA_CODEC_CA0132 + default y select SND_HDA_DSP_LOADER select FW_LOADER help diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 53e7732ef752..a34a2c9f4bcf 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -88,7 +88,7 @@ struct hda_conn_list { struct list_head list; int len; hda_nid_t nid; - hda_nid_t conns[0]; + hda_nid_t conns[]; }; /* look up the cached results */ diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 2609e391ce54..9765652a73d7 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -373,7 +373,7 @@ static int azx_get_sync_time(ktime_t *device, u32 wallclk_ctr, wallclk_cycles; bool direction; u32 dma_select; - u32 timeout = 200; + u32 timeout; u32 retry_count = 0; runtime = substream->runtime; diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 10223e080d59..34fe753a46fb 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2699,7 +2699,7 @@ struct dsp_image_seg { u32 magic; u32 chip_addr; u32 count; - u32 data[0]; + u32 data[]; }; static const u32 g_magic_value = 0x4c46584d; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5119a9ae3d8a..bb287a916dae 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -154,7 +154,6 @@ struct hdmi_spec { struct hda_multi_out multiout; struct hda_pcm_stream pcm_playback; - bool use_jack_detect; /* jack detection enabled */ bool use_acomp_notifier; /* use eld_notify callback for hotplug */ bool acomp_registered; /* audio component registered in this driver */ struct drm_audio_component_audio_ops drm_audio_ops; @@ -753,7 +752,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, * Unsolicited events */ -static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll); +static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll); static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid, int dev_id) @@ -764,8 +763,7 @@ static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid, if (pin_idx < 0) return; mutex_lock(&spec->pcm_lock); - if (hdmi_present_sense(get_pin(spec, pin_idx), 1)) - snd_hda_jack_report_sync(codec); + hdmi_present_sense(get_pin(spec, pin_idx), 1); mutex_unlock(&spec->pcm_lock); } @@ -779,21 +777,9 @@ static void jack_callback(struct hda_codec *codec, check_presence_and_report(codec, jack->nid, jack->dev_id); } -static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) +static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res, + struct hda_jack_tbl *jack) { - int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - struct hda_jack_tbl *jack; - - if (codec->dp_mst) { - int dev_entry = - (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT; - - jack = snd_hda_jack_tbl_get_from_tag(codec, tag, dev_entry); - } else { - jack = snd_hda_jack_tbl_get_from_tag(codec, tag, 0); - } - if (!jack) - return; jack->jack_dirty = 1; codec_dbg(codec, @@ -853,7 +839,7 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) } if (subtag == 0) - hdmi_intrinsic_event(codec, res); + hdmi_intrinsic_event(codec, res, jack); else hdmi_non_intrinsic_event(codec, res); } @@ -1480,21 +1466,60 @@ static void hdmi_pcm_reset_pin(struct hdmi_spec *spec, per_pin->channels = 0; } +static struct snd_jack *pin_idx_to_pcm_jack(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin) +{ + struct hdmi_spec *spec = codec->spec; + + if (per_pin->pcm_idx >= 0) + return spec->pcm_rec[per_pin->pcm_idx].jack; + else + return NULL; +} + /* update per_pin ELD from the given new ELD; * setup info frame and notification accordingly + * also notify ELD kctl and report jack status changes */ -static bool update_eld(struct hda_codec *codec, +static void update_eld(struct hda_codec *codec, struct hdmi_spec_per_pin *per_pin, - struct hdmi_eld *eld) + struct hdmi_eld *eld, + int repoll) { struct hdmi_eld *pin_eld = &per_pin->sink_eld; struct hdmi_spec *spec = codec->spec; + struct snd_jack *pcm_jack; bool old_eld_valid = pin_eld->eld_valid; bool eld_changed; int pcm_idx; + if (eld->eld_valid) { + if (eld->eld_size <= 0 || + snd_hdmi_parse_eld(codec, &eld->info, eld->eld_buffer, + eld->eld_size) < 0) { + eld->eld_valid = false; + if (repoll) { + schedule_delayed_work(&per_pin->work, + msecs_to_jiffies(300)); + return; + } + } + } + + if (!eld->eld_valid || eld->eld_size <= 0) { + eld->eld_valid = false; + eld->eld_size = 0; + } + /* for monitor disconnection, save pcm_idx firstly */ pcm_idx = per_pin->pcm_idx; + + /* + * pcm_idx >=0 before update_eld() means it is in monitor + * disconnected event. Jack must be fetched before update_eld(). + */ + pcm_jack = pin_idx_to_pcm_jack(codec, per_pin); + if (spec->dyn_pcm_assign) { if (eld->eld_valid) { hdmi_attach_hda_pcm(spec, per_pin); @@ -1509,6 +1534,8 @@ static bool update_eld(struct hda_codec *codec, */ if (pcm_idx == -1) pcm_idx = per_pin->pcm_idx; + if (!pcm_jack) + pcm_jack = pin_idx_to_pcm_jack(codec, per_pin); if (eld->eld_valid) snd_hdmi_show_eld(codec, &eld->info); @@ -1547,42 +1574,17 @@ static bool update_eld(struct hda_codec *codec, SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, &get_hdmi_pcm(spec, pcm_idx)->eld_ctl->id); - return eld_changed; -} -static struct snd_jack *pin_idx_to_pcm_jack(struct hda_codec *codec, - struct hdmi_spec_per_pin *per_pin) -{ - struct hdmi_spec *spec = codec->spec; - struct snd_jack *jack = NULL; - struct hda_jack_tbl *jack_tbl; - - /* if !dyn_pcm_assign, get jack from hda_jack_tbl - * in !dyn_pcm_assign case, spec->pcm_rec[].jack is not - * NULL even after snd_hda_jack_tbl_clear() is called to - * free snd_jack. This may cause access invalid memory - * when calling snd_jack_report - */ - if (per_pin->pcm_idx >= 0 && spec->dyn_pcm_assign) { - jack = spec->pcm_rec[per_pin->pcm_idx].jack; - } else if (!spec->dyn_pcm_assign) { - /* - * jack tbl doesn't support DP MST - * DP MST will use dyn_pcm_assign, - * so DP MST will never come here - */ - jack_tbl = snd_hda_jack_tbl_get_mst(codec, per_pin->pin_nid, - per_pin->dev_id); - if (jack_tbl) - jack = jack_tbl->jack; - } - return jack; + if (eld_changed && pcm_jack) + snd_jack_report(pcm_jack, + (eld->monitor_present && eld->eld_valid) ? + SND_JACK_AVOUT : 0); } + /* update ELD and jack state via HD-audio verbs */ -static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, +static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, int repoll) { - struct hda_jack_tbl *jack; struct hda_codec *codec = per_pin->codec; struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->temp_eld; @@ -1597,9 +1599,11 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, * the unsolicited response to avoid custom WARs. */ int present; - bool ret; - bool do_repoll = false; - struct snd_jack *pcm_jack = NULL; + int ret; + + ret = snd_hda_power_up_pm(codec); + if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) + goto out; present = snd_hda_jack_pin_sense(codec, pin_nid, dev_id); @@ -1618,62 +1622,12 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, if (spec->ops.pin_get_eld(codec, pin_nid, dev_id, eld->eld_buffer, &eld->eld_size) < 0) eld->eld_valid = false; - else { - if (snd_hdmi_parse_eld(codec, &eld->info, eld->eld_buffer, - eld->eld_size) < 0) - eld->eld_valid = false; - } - if (!eld->eld_valid && repoll) - do_repoll = true; } - if (do_repoll) { - schedule_delayed_work(&per_pin->work, msecs_to_jiffies(300)); - } else { - /* - * pcm_idx >=0 before update_eld() means it is in monitor - * disconnected event. Jack must be fetched before - * update_eld(). - */ - pcm_jack = pin_idx_to_pcm_jack(codec, per_pin); - update_eld(codec, per_pin, eld); - if (!pcm_jack) - pcm_jack = pin_idx_to_pcm_jack(codec, per_pin); - } - - ret = !repoll || !eld->monitor_present || eld->eld_valid; - - jack = snd_hda_jack_tbl_get_mst(codec, pin_nid, per_pin->dev_id); - if (jack) { - jack->block_report = !ret; - jack->pin_sense = (eld->monitor_present && eld->eld_valid) ? - AC_PINSENSE_PRESENCE : 0; - - if (spec->dyn_pcm_assign && pcm_jack && !do_repoll) { - int state = 0; - - if (jack->pin_sense & AC_PINSENSE_PRESENCE) - state = SND_JACK_AVOUT; - snd_jack_report(pcm_jack, state); - } - - /* - * snd_hda_jack_pin_sense() call at the beginning of this - * function, updates jack->pins_sense and clears - * jack->jack_dirty, therefore snd_hda_jack_report_sync() will - * not override the jack->pin_sense. - * - * snd_hda_jack_report_sync() is superfluous for dyn_pcm_assign - * case. The jack->pin_sense update was already performed, and - * hda_jack->jack is NULL for dyn_pcm_assign. - * - * Don't call snd_hda_jack_report_sync() for - * dyn_pcm_assign. - */ - ret = ret && !spec->dyn_pcm_assign; - } + update_eld(codec, per_pin, eld, repoll); mutex_unlock(&per_pin->lock); - return ret; + out: + snd_hda_power_down_pm(codec); } /* update ELD and jack state via audio component */ @@ -1682,64 +1636,25 @@ static void sync_eld_via_acomp(struct hda_codec *codec, { struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->temp_eld; - struct snd_jack *jack = NULL; - bool changed; - int size; mutex_lock(&per_pin->lock); eld->monitor_present = false; - size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, + eld->eld_size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, per_pin->dev_id, &eld->monitor_present, eld->eld_buffer, ELD_MAX_SIZE); - if (size > 0) { - size = min(size, ELD_MAX_SIZE); - if (snd_hdmi_parse_eld(codec, &eld->info, - eld->eld_buffer, size) < 0) - size = -EINVAL; - } - - if (size > 0) { - eld->eld_valid = true; - eld->eld_size = size; - } else { - eld->eld_valid = false; - eld->eld_size = 0; - } - - /* pcm_idx >=0 before update_eld() means it is in monitor - * disconnected event. Jack must be fetched before update_eld() - */ - jack = pin_idx_to_pcm_jack(codec, per_pin); - changed = update_eld(codec, per_pin, eld); - if (jack == NULL) - jack = pin_idx_to_pcm_jack(codec, per_pin); - if (changed && jack) - snd_jack_report(jack, - (eld->monitor_present && eld->eld_valid) ? - SND_JACK_AVOUT : 0); + eld->eld_valid = (eld->eld_size > 0); + update_eld(codec, per_pin, eld, 0); mutex_unlock(&per_pin->lock); } -static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) +static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) { struct hda_codec *codec = per_pin->codec; - int ret; - /* no temporary power up/down needed for component notifier */ - if (!codec_has_acomp(codec)) { - ret = snd_hda_power_up_pm(codec); - if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) { - snd_hda_power_down_pm(codec); - return false; - } - ret = hdmi_present_sense_via_verbs(per_pin, repoll); - snd_hda_power_down_pm(codec); - } else { + if (!codec_has_acomp(codec)) + hdmi_present_sense_via_verbs(per_pin, repoll); + else sync_eld_via_acomp(codec, per_pin); - ret = false; /* don't call snd_hda_jack_report_sync() */ - } - - return ret; } static void hdmi_repoll_eld(struct work_struct *work) @@ -1759,8 +1674,7 @@ static void hdmi_repoll_eld(struct work_struct *work) per_pin->repoll_count = 0; mutex_lock(&spec->pcm_lock); - if (hdmi_present_sense(per_pin, per_pin->repoll_count)) - snd_hda_jack_report_sync(per_pin->codec); + hdmi_present_sense(per_pin, per_pin->repoll_count); mutex_unlock(&spec->pcm_lock); } @@ -2206,15 +2120,23 @@ static void free_hdmi_jack_priv(struct snd_jack *jack) pcm->jack = NULL; } -static int add_hdmi_jack_kctl(struct hda_codec *codec, - struct hdmi_spec *spec, - int pcm_idx, - const char *name) +static int generic_hdmi_build_jack(struct hda_codec *codec, int pcm_idx) { + char hdmi_str[32] = "HDMI/DP"; + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pcm_idx); struct snd_jack *jack; + int pcmdev = get_pcm_rec(spec, pcm_idx)->device; int err; - err = snd_jack_new(codec->card, name, SND_JACK_AVOUT, &jack, + if (pcmdev > 0) + sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev); + if (!spec->dyn_pcm_assign && + !is_jack_detectable(codec, per_pin->pin_nid)) + strncat(hdmi_str, " Phantom", + sizeof(hdmi_str) - strlen(hdmi_str) - 1); + + err = snd_jack_new(codec->card, hdmi_str, SND_JACK_AVOUT, &jack, true, false); if (err < 0) return err; @@ -2225,48 +2147,6 @@ static int add_hdmi_jack_kctl(struct hda_codec *codec, return 0; } -static int generic_hdmi_build_jack(struct hda_codec *codec, int pcm_idx) -{ - char hdmi_str[32] = "HDMI/DP"; - struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_pin *per_pin; - struct hda_jack_tbl *jack; - int pcmdev = get_pcm_rec(spec, pcm_idx)->device; - bool phantom_jack; - int ret; - - if (pcmdev > 0) - sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev); - - if (spec->dyn_pcm_assign) - return add_hdmi_jack_kctl(codec, spec, pcm_idx, hdmi_str); - - /* for !dyn_pcm_assign, we still use hda_jack for compatibility */ - /* if !dyn_pcm_assign, it must be non-MST mode. - * This means pcms and pins are statically mapped. - * And pcm_idx is pin_idx. - */ - per_pin = get_pin(spec, pcm_idx); - phantom_jack = !is_jack_detectable(codec, per_pin->pin_nid); - if (phantom_jack) - strncat(hdmi_str, " Phantom", - sizeof(hdmi_str) - strlen(hdmi_str) - 1); - ret = snd_hda_jack_add_kctl_mst(codec, per_pin->pin_nid, - per_pin->dev_id, hdmi_str, phantom_jack, - 0, NULL); - if (ret < 0) - return ret; - jack = snd_hda_jack_tbl_get_mst(codec, per_pin->pin_nid, - per_pin->dev_id); - if (jack == NULL) - return 0; - /* assign jack->jack to pcm_rec[].jack to - * align with dyn_pcm_assign mode - */ - spec->pcm_rec[pcm_idx].jack = jack->jack; - return 0; -} - static int generic_hdmi_build_controls(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -2355,7 +2235,6 @@ static int generic_hdmi_init(struct hda_codec *codec) int pin_idx; mutex_lock(&spec->bind_lock); - spec->use_jack_detect = !codec->jackpoll_interval; for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); hda_nid_t pin_nid = per_pin->pin_nid; @@ -2365,12 +2244,8 @@ static int generic_hdmi_init(struct hda_codec *codec) hdmi_init_pin(codec, pin_nid); if (codec_has_acomp(codec)) continue; - if (spec->use_jack_detect) - snd_hda_jack_detect_enable(codec, pin_nid, dev_id); - else - snd_hda_jack_detect_enable_callback_mst(codec, pin_nid, - dev_id, - jack_callback); + snd_hda_jack_detect_enable_callback_mst(codec, pin_nid, dev_id, + jack_callback); } mutex_unlock(&spec->bind_lock); return 0; @@ -2532,12 +2407,6 @@ static void reprogram_jack_detect(struct hda_codec *codec, hda_nid_t nid, unsigned int val = use_acomp ? 0 : (AC_USRSP_EN | tbl->tag); snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, val); - } else { - /* if no jack entry was defined beforehand, create a new one - * at need (i.e. only when notifier is cleared) - */ - if (!use_acomp) - snd_hda_jack_detect_enable(codec, nid, dev_id); } } @@ -2553,13 +2422,11 @@ static void generic_acomp_notifier_set(struct drm_audio_component *acomp, spec->use_acomp_notifier = use_acomp; spec->codec->relaxed_resume = use_acomp; /* reprogram each jack detection logic depending on the notifier */ - if (spec->use_jack_detect) { - for (i = 0; i < spec->num_pins; i++) - reprogram_jack_detect(spec->codec, - get_pin(spec, i)->pin_nid, - get_pin(spec, i)->dev_id, - use_acomp); - } + for (i = 0; i < spec->num_pins; i++) + reprogram_jack_detect(spec->codec, + get_pin(spec, i)->pin_nid, + get_pin(spec, i)->dev_id, + use_acomp); mutex_unlock(&spec->bind_lock); } diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 21ab9cc50c71..65a887b217ee 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -30,7 +30,7 @@ #if K1212_DEBUG_LEVEL > 0 #define K1212_DEBUG_PRINTK(fmt,args...) printk(KERN_DEBUG fmt,##args) #else -#define K1212_DEBUG_PRINTK(fmt,...) +#define K1212_DEBUG_PRINTK(fmt,...) do { } while (0) #endif #if K1212_DEBUG_LEVEL > 1 #define K1212_DEBUG_PRINTK_VERBOSE(fmt,args...) printk(KERN_DEBUG fmt,##args) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index cc06f0a1a7e4..227aece17e39 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -3353,7 +3353,8 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) return; } } else { - int err = -EINVAL; + int err; + err = hdsp_request_fw_loader(hdsp); if (err < 0) { snd_iprintf(buffer, diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 799789c8eea9..8b03e2dc503f 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -414,6 +414,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre { unsigned int i, idx, ofs, rest; struct via82xx *chip = snd_pcm_substream_chip(substream); + __le32 *pgtbl; if (dev->table.area == NULL) { /* the start of each lists must be aligned to 8 bytes, @@ -435,6 +436,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre /* fill the entries */ idx = 0; ofs = 0; + pgtbl = (__le32 *)dev->table.area; for (i = 0; i < periods; i++) { rest = fragsize; /* fill descriptors for a period. @@ -451,7 +453,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre return -EINVAL; } addr = snd_pcm_sgbuf_get_addr(substream, ofs); - ((u32 *)dev->table.area)[idx << 1] = cpu_to_le32(addr); + pgtbl[idx << 1] = cpu_to_le32(addr); r = snd_pcm_sgbuf_get_chunk_size(substream, ofs, rest); rest -= r; if (! rest) { @@ -466,7 +468,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre "tbl %d: at %d size %d (rest %d)\n", idx, ofs, r, rest); */ - ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag); + pgtbl[(idx<<1) + 1] = cpu_to_le32(r | flag); dev->idx_table[idx].offset = ofs; dev->idx_table[idx].size = r; ofs += r; diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 84e589803e2e..607b7100db1c 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -267,6 +267,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre { unsigned int i, idx, ofs, rest; struct via82xx_modem *chip = snd_pcm_substream_chip(substream); + __le32 *pgtbl; if (dev->table.area == NULL) { /* the start of each lists must be aligned to 8 bytes, @@ -288,6 +289,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre /* fill the entries */ idx = 0; ofs = 0; + pgtbl = (__le32 *)dev->table.area; for (i = 0; i < periods; i++) { rest = fragsize; /* fill descriptors for a period. @@ -304,7 +306,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre return -EINVAL; } addr = snd_pcm_sgbuf_get_addr(substream, ofs); - ((u32 *)dev->table.area)[idx << 1] = cpu_to_le32(addr); + pgtbl[idx << 1] = cpu_to_le32(addr); r = PAGE_SIZE - (ofs % PAGE_SIZE); if (rest < r) r = rest; @@ -321,7 +323,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre "tbl %d: at %d size %d (rest %d)\n", idx, ofs, r, rest); */ - ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag); + pgtbl[(idx<<1) + 1] = cpu_to_le32(r | flag); dev->idx_table[idx].offset = ofs; dev->idx_table[idx].size = r; ofs += r; diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index 093806d735c6..9554a0c506af 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -40,6 +40,7 @@ static int keywest_probe(struct i2c_client *client, static int keywest_attach_adapter(struct i2c_adapter *adapter) { struct i2c_board_info info; + struct i2c_client *client; if (! keywest_ctx) return -EINVAL; @@ -50,9 +51,11 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter) memset(&info, 0, sizeof(struct i2c_board_info)); strlcpy(info.type, "keywest", I2C_NAME_SIZE); info.addr = keywest_ctx->addr; - keywest_ctx->client = i2c_new_device(adapter, &info); - if (!keywest_ctx->client) - return -ENODEV; + client = i2c_new_client_device(adapter, &info); + if (IS_ERR(client)) + return PTR_ERR(client); + keywest_ctx->client = client; + /* * We know the driver is already loaded, so the device should be * already bound. If not it means binding failed, and then there diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 78edd7d2f418..56031026b113 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -13,6 +13,7 @@ snd-usb-audio-objs := card.o \ mixer_scarlett.o \ mixer_scarlett_gen2.o \ mixer_us16x08.o \ + mixer_s1810c.o \ pcm.o \ power.o \ proc.o \ diff --git a/sound/usb/card.c b/sound/usb/card.c index 827fb0bc8b56..fd6fd1726ea0 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -72,6 +72,7 @@ static int device_setup[SNDRV_CARDS]; /* device parameter for this card */ static bool ignore_ctl_error; static bool autoclock = true; static char *quirk_alias[SNDRV_CARDS]; +static char *delayed_register[SNDRV_CARDS]; bool snd_usb_use_vmalloc = true; bool snd_usb_skip_validation; @@ -95,6 +96,8 @@ module_param(autoclock, bool, 0444); MODULE_PARM_DESC(autoclock, "Enable auto-clock selection for UAC2 devices (default: yes)."); module_param_array(quirk_alias, charp, NULL, 0444); MODULE_PARM_DESC(quirk_alias, "Quirk aliases, e.g. 0123abcd:5678beef."); +module_param_array(delayed_register, charp, NULL, 0444); +MODULE_PARM_DESC(delayed_register, "Quirk for delayed registration, given by id:iface, e.g. 0123abcd:4."); module_param_named(use_vmalloc, snd_usb_use_vmalloc, bool, 0444); MODULE_PARM_DESC(use_vmalloc, "Use vmalloc for PCM intermediate buffers (default: yes)."); module_param_named(skip_validation, snd_usb_skip_validation, bool, 0444); @@ -525,6 +528,21 @@ static bool get_alias_id(struct usb_device *dev, unsigned int *id) return false; } +static bool check_delayed_register_option(struct snd_usb_audio *chip, int iface) +{ + int i; + unsigned int id, inum; + + for (i = 0; i < ARRAY_SIZE(delayed_register); i++) { + if (delayed_register[i] && + sscanf(delayed_register[i], "%x:%x", &id, &inum) == 2 && + id == chip->usb_id) + return inum != iface; + } + + return false; +} + static const struct usb_device_id usb_audio_ids[]; /* defined below */ /* look for the corresponding quirk */ @@ -662,10 +680,22 @@ static int usb_audio_probe(struct usb_interface *intf, goto __error; } - /* we are allowed to call snd_card_register() many times */ - err = snd_card_register(chip->card); - if (err < 0) - goto __error; + if (chip->need_delayed_register) { + dev_info(&dev->dev, + "Found post-registration device assignment: %08x:%02x\n", + chip->usb_id, ifnum); + chip->need_delayed_register = false; /* clear again */ + } + + /* we are allowed to call snd_card_register() many times, but first + * check to see if a device needs to skip it or do anything special + */ + if (!snd_usb_registration_quirk(chip, ifnum) && + !check_delayed_register_option(chip, ifnum)) { + err = snd_card_register(chip->card); + if (err < 0) + goto __error; + } if (quirk && quirk->shares_media_device) { /* don't want to fail when snd_media_device_create() fails */ diff --git a/sound/usb/clock.c b/sound/usb/clock.c index a48313dfa967..b118cf97607f 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -151,16 +151,15 @@ static int uac_clock_selector_set_val(struct snd_usb_audio *chip, int selector_i return ret; } -/* - * Assume the clock is valid if clock source supports only one single sample - * rate, the terminal is connected directly to it (there is no clock selector) - * and clock type is internal. This is to deal with some Denon DJ controllers - * that always reports that clock is invalid. - */ static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip, struct audioformat *fmt, int source_id) { + bool ret = false; + int count; + unsigned char data; + struct usb_device *dev = chip->dev; + if (fmt->protocol == UAC_VERSION_2) { struct uac_clock_source_descriptor *cs_desc = snd_usb_find_clock_source(chip->ctrl_intf, source_id); @@ -168,13 +167,51 @@ static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip, if (!cs_desc) return false; - return (fmt->nr_rates == 1 && - (fmt->clock & 0xff) == cs_desc->bClockID && - (cs_desc->bmAttributes & 0x3) != - UAC_CLOCK_SOURCE_TYPE_EXT); + /* + * Assume the clock is valid if clock source supports only one + * single sample rate, the terminal is connected directly to it + * (there is no clock selector) and clock type is internal. + * This is to deal with some Denon DJ controllers that always + * reports that clock is invalid. + */ + if (fmt->nr_rates == 1 && + (fmt->clock & 0xff) == cs_desc->bClockID && + (cs_desc->bmAttributes & 0x3) != + UAC_CLOCK_SOURCE_TYPE_EXT) + return true; + } + + /* + * MOTU MicroBook IIc + * Sample rate changes takes more than 2 seconds for this device. Clock + * validity request returns false during that period. + */ + if (chip->usb_id == USB_ID(0x07fd, 0x0004)) { + count = 0; + + while ((!ret) && (count < 50)) { + int err; + + msleep(100); + + 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_CLOCK_VALID << 8, + snd_usb_ctrl_intf(chip) | (source_id << 8), + &data, sizeof(data)); + if (err < 0) { + dev_warn(&dev->dev, + "%s(): cannot get clock validity for id %d\n", + __func__, source_id); + return false; + } + + ret = !!data; + count++; + } } - return false; + return ret; } static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, diff --git a/sound/usb/format.c b/sound/usb/format.c index 9f5cb4ed3a0c..50e1874c847c 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -247,6 +247,36 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof return 0; } + +/* + * Presonus Studio 1810c supports a limited set of sampling + * rates per altsetting but reports the full set each time. + * If we don't filter out the unsupported rates and attempt + * to configure the card, it will hang refusing to do any + * further audio I/O until a hard reset is performed. + * + * The list of supported rates per altsetting (set of available + * I/O channels) is described in the owner's manual, section 2.2. + */ +static bool s1810c_valid_sample_rate(struct audioformat *fp, + unsigned int rate) +{ + switch (fp->altsetting) { + case 1: + /* All ADAT ports available */ + return rate <= 48000; + case 2: + /* Half of ADAT ports available */ + return (rate == 88200 || rate == 96000); + case 3: + /* Analog I/O only (no S/PDIF nor ADAT) */ + return rate >= 176400; + default: + return false; + } + return false; +} + /* * Helper function to walk the array of sample rate triplets reported by * the device. The problem is that we need to parse whole array first to @@ -283,6 +313,12 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, } for (rate = min; rate <= max; rate += res) { + + /* Filter out invalid rates on Presonus Studio 1810c */ + if (chip->usb_id == USB_ID(0x0194f, 0x010c) && + !s1810c_valid_sample_rate(fp, rate)) + goto skip_rate; + if (fp->rate_table) fp->rate_table[nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) @@ -297,6 +333,7 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, break; } +skip_rate: /* avoid endless loop */ if (res == 0) break; diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 392e5fda680c..be5c597ed40c 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -91,7 +91,7 @@ struct usb_ms_endpoint_descriptor { __u8 bDescriptorType; __u8 bDescriptorSubtype; __u8 bNumEmbMIDIJack; - __u8 baAssocJackID[0]; + __u8 baAssocJackID[]; } __attribute__ ((packed)); struct snd_usb_midi_in_endpoint; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 81b2db0edd5f..721d12130d0c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -292,6 +292,11 @@ static int uac2_ctl_value_size(int val_type) * retrieve a mixer value */ +static inline int mixer_ctrl_intf(struct usb_mixer_interface *mixer) +{ + return get_iface_desc(mixer->hostif)->bInterfaceNumber; +} + static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { @@ -306,7 +311,7 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, return -EIO; while (timeout-- > 0) { - idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8); + idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8); err = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, idx, buf, val_len); @@ -354,7 +359,7 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, if (ret) goto error; - idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8); + idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8); ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), bRequest, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, idx, buf, size); @@ -479,7 +484,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, return -EIO; while (timeout-- > 0) { - idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8); + idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8); err = snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, @@ -901,6 +906,12 @@ static int parse_term_effect_unit(struct mixer_build *state, struct usb_audio_term *term, void *p1, int id) { + struct uac2_effect_unit_descriptor *d = p1; + int err; + + err = __check_input_term(state, d->bSourceID, term); + if (err < 0) + return err; term->type = UAC3_EFFECT_UNIT << 16; /* virtual type */ term->id = id; return 0; @@ -1203,7 +1214,7 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, get_ctl_value(cval, UAC_GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) { usb_audio_err(cval->head.mixer->chip, "%d:%d: cannot get min/max values for control %d (id %d)\n", - cval->head.id, snd_usb_ctrl_intf(cval->head.mixer->chip), + cval->head.id, mixer_ctrl_intf(cval->head.mixer), cval->control, cval->head.id); return -EINVAL; } @@ -1422,7 +1433,7 @@ static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, if (ret) goto error; - idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8); + idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8); if (cval->head.mixer->protocol == UAC_VERSION_2) { struct uac2_connectors_ctl_blk uac2_conn; @@ -1674,6 +1685,16 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer, /* get min/max values */ get_min_max_with_quirks(cval, 0, kctl); + /* skip a bogus volume range */ + if (cval->max <= cval->min) { + usb_audio_dbg(mixer->chip, + "[%d] FU [%s] skipped due to invalid volume\n", + cval->head.id, kctl->id.name); + snd_ctl_free_one(kctl); + return; + } + + if (control == UAC_FU_VOLUME) { check_mapped_dB(map, cval); if (cval->dBmin < cval->dBmax || !cval->initialized) { @@ -3203,7 +3224,7 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, list_for_each_entry(mixer, &chip->mixer_list, list) { snd_iprintf(buffer, "USB Mixer: usb_id=0x%08x, ctrlif=%i, ctlerr=%i\n", - chip->usb_id, snd_usb_ctrl_intf(chip), + chip->usb_id, mixer_ctrl_intf(mixer), mixer->ignore_ctl_error); snd_iprintf(buffer, "Card: %s\n", chip->card->longname); for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) { diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index c237e24f08d9..02b036b2aefb 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -34,6 +34,7 @@ #include "mixer_scarlett.h" #include "mixer_scarlett_gen2.h" #include "mixer_us16x08.h" +#include "mixer_s1810c.h" #include "helper.h" struct std_mono_table { @@ -2277,6 +2278,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2a39, 0x3fd4): /* RME */ err = snd_rme_controls_create(mixer); break; + + case USB_ID(0x0194f, 0x010c): /* Presonus Studio 1810c */ + err = snd_sc1810_init_mixer(mixer); + break; } return err; diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c new file mode 100644 index 000000000000..6483e47bafd0 --- /dev/null +++ b/sound/usb/mixer_s1810c.c @@ -0,0 +1,595 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Presonus Studio 1810c driver for ALSA + * Copyright (C) 2019 Nick Kossifidis <mickflemm@gmail.com> + * + * Based on reverse engineering of the communication protocol + * between the windows driver / Univeral Control (UC) program + * and the device, through usbmon. + * + * For now this bypasses the mixer, with all channels split, + * so that the software can mix with greater flexibility. + * It also adds controls for the 4 buttons on the front of + * the device. + */ + +#include <linux/usb.h> +#include <linux/usb/audio-v2.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/control.h> + +#include "usbaudio.h" +#include "mixer.h" +#include "mixer_quirks.h" +#include "helper.h" +#include "mixer_s1810c.h" + +#define SC1810C_CMD_REQ 160 +#define SC1810C_CMD_REQTYPE \ + (USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT) +#define SC1810C_CMD_F1 0x50617269 +#define SC1810C_CMD_F2 0x14 + +/* + * DISCLAIMER: These are just guesses based on the + * dumps I got. + * + * It seems like a selects between + * device (0), mixer (0x64) and output (0x65) + * + * For mixer (0x64): + * * b selects an input channel (see below). + * * c selects an output channel pair (see below). + * * d selects left (0) or right (1) of that pair. + * * e 0-> disconnect, 0x01000000-> connect, + * 0x0109-> used for stereo-linking channels, + * e is also used for setting volume levels + * in which case b is also set so I guess + * this way it is possible to set the volume + * level from the specified input to the + * specified output. + * + * IN Channels: + * 0 - 7 Mic/Inst/Line (Analog inputs) + * 8 - 9 S/PDIF + * 10 - 17 ADAT + * 18 - 35 DAW (Inputs from the host) + * + * OUT Channels (pairs): + * 0 -> Main out + * 1 -> Line1/2 + * 2 -> Line3/4 + * 3 -> S/PDIF + * 4 -> ADAT? + * + * For device (0): + * * b and c are not used, at least not on the + * dumps I got. + * * d sets the control id to be modified + * (see below). + * * e sets the setting for that control. + * (so for the switches I was interested + * in it's 0/1) + * + * For output (0x65): + * * b is the output channel (see above). + * * c is zero. + * * e I guess the same as with mixer except 0x0109 + * which I didn't see in my dumps. + * + * The two fixed fields have the same values for + * mixer and output but a different set for device. + */ +struct s1810c_ctl_packet { + u32 a; + u32 b; + u32 fixed1; + u32 fixed2; + u32 c; + u32 d; + u32 e; +}; + +#define SC1810C_CTL_LINE_SW 0 +#define SC1810C_CTL_MUTE_SW 1 +#define SC1810C_CTL_AB_SW 3 +#define SC1810C_CTL_48V_SW 4 + +#define SC1810C_SET_STATE_REQ 161 +#define SC1810C_SET_STATE_REQTYPE SC1810C_CMD_REQTYPE +#define SC1810C_SET_STATE_F1 0x64656D73 +#define SC1810C_SET_STATE_F2 0xF4 + +#define SC1810C_GET_STATE_REQ 162 +#define SC1810C_GET_STATE_REQTYPE \ + (USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN) +#define SC1810C_GET_STATE_F1 SC1810C_SET_STATE_F1 +#define SC1810C_GET_STATE_F2 SC1810C_SET_STATE_F2 + +#define SC1810C_STATE_F1_IDX 2 +#define SC1810C_STATE_F2_IDX 3 + +/* + * This packet includes mixer volumes and + * various other fields, it's an extended + * version of ctl_packet, with a and b + * being zero and different f1/f2. + */ +struct s1810c_state_packet { + u32 fields[63]; +}; + +#define SC1810C_STATE_48V_SW 58 +#define SC1810C_STATE_LINE_SW 59 +#define SC1810C_STATE_MUTE_SW 60 +#define SC1810C_STATE_AB_SW 62 + +struct s1810_mixer_state { + uint16_t seqnum; + struct mutex usb_mutex; + struct mutex data_mutex; +}; + +static int +snd_s1810c_send_ctl_packet(struct usb_device *dev, u32 a, + u32 b, u32 c, u32 d, u32 e) +{ + struct s1810c_ctl_packet pkt = { 0 }; + int ret = 0; + + pkt.fixed1 = SC1810C_CMD_F1; + pkt.fixed2 = SC1810C_CMD_F2; + + pkt.a = a; + pkt.b = b; + pkt.c = c; + pkt.d = d; + /* + * Value for settings 0/1 for this + * output channel is always 0 (probably because + * there is no ADAT output on 1810c) + */ + pkt.e = (c == 4) ? 0 : e; + + ret = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + SC1810C_CMD_REQ, + SC1810C_CMD_REQTYPE, 0, 0, &pkt, sizeof(pkt)); + if (ret < 0) { + dev_warn(&dev->dev, "could not send ctl packet\n"); + return ret; + } + return 0; +} + +/* + * When opening Universal Control the program periodicaly + * sends and receives state packets for syncinc state between + * the device and the host. + * + * Note that if we send only the request to get data back we'll + * get an error, we need to first send an empty state packet and + * then ask to receive a filled. Their seqnumbers must also match. + */ +static int +snd_sc1810c_get_status_field(struct usb_device *dev, + u32 *field, int field_idx, uint16_t *seqnum) +{ + struct s1810c_state_packet pkt_out = { { 0 } }; + struct s1810c_state_packet pkt_in = { { 0 } }; + int ret = 0; + + pkt_out.fields[SC1810C_STATE_F1_IDX] = SC1810C_SET_STATE_F1; + pkt_out.fields[SC1810C_STATE_F2_IDX] = SC1810C_SET_STATE_F2; + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + SC1810C_SET_STATE_REQ, + SC1810C_SET_STATE_REQTYPE, + (*seqnum), 0, &pkt_out, sizeof(pkt_out)); + if (ret < 0) { + dev_warn(&dev->dev, "could not send state packet (%d)\n", ret); + return ret; + } + + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + SC1810C_GET_STATE_REQ, + SC1810C_GET_STATE_REQTYPE, + (*seqnum), 0, &pkt_in, sizeof(pkt_in)); + if (ret < 0) { + dev_warn(&dev->dev, "could not get state field %u (%d)\n", + field_idx, ret); + return ret; + } + + (*field) = pkt_in.fields[field_idx]; + (*seqnum)++; + return 0; +} + +/* + * This is what I got when bypassing the mixer with + * all channels split. I'm not 100% sure of what's going + * on, I could probably clean this up based on my observations + * but I prefer to keep the same behavior as the windows driver. + */ +static int snd_s1810c_init_mixer_maps(struct snd_usb_audio *chip) +{ + u32 a, b, c, e, n, off; + struct usb_device *dev = chip->dev; + + /* Set initial volume levels ? */ + a = 0x64; + e = 0xbc; + for (n = 0; n < 2; n++) { + off = n * 18; + for (b = off, c = 0; b < 18 + off; b++) { + /* This channel to all outputs ? */ + for (c = 0; c <= 8; c++) { + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } + /* This channel to main output (again) */ + snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e); + } + /* + * I noticed on UC that DAW channels have different + * initial volumes, so this makes sense. + */ + e = 0xb53bf0; + } + + /* Connect analog outputs ? */ + a = 0x65; + e = 0x01000000; + for (b = 1; b < 3; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e); + } + snd_s1810c_send_ctl_packet(dev, a, 0, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 0, 0, 1, e); + + /* Set initial volume levels for S/PDIF mappings ? */ + a = 0x64; + e = 0xbc; + c = 3; + for (n = 0; n < 2; n++) { + off = n * 18; + for (b = off; b < 18 + off; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } + e = 0xb53bf0; + } + + /* Connect S/PDIF output ? */ + a = 0x65; + e = 0x01000000; + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e); + + /* Connect all outputs (again) ? */ + a = 0x65; + e = 0x01000000; + for (b = 0; b < 4; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e); + } + + /* Basic routing to get sound out of the device */ + a = 0x64; + e = 0x01000000; + for (c = 0; c < 4; c++) { + for (b = 0; b < 36; b++) { + if ((c == 0 && b == 18) || /* DAW1/2 -> Main */ + (c == 1 && b == 20) || /* DAW3/4 -> Line3/4 */ + (c == 2 && b == 22) || /* DAW4/5 -> Line5/6 */ + (c == 3 && b == 24)) { /* DAW5/6 -> S/PDIF */ + /* Left */ + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0); + b++; + /* Right */ + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } else { + /* Leave the rest disconnected */ + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0); + } + } + } + + /* Set initial volume levels for S/PDIF (again) ? */ + a = 0x64; + e = 0xbc; + c = 3; + for (n = 0; n < 2; n++) { + off = n * 18; + for (b = off; b < 18 + off; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } + e = 0xb53bf0; + } + + /* Connect S/PDIF outputs (again) ? */ + a = 0x65; + e = 0x01000000; + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e); + + /* Again ? */ + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e); + + return 0; +} + +/* + * Sync state with the device and retrieve the requested field, + * whose index is specified in (kctl->private_value & 0xFF), + * from the received fields array. + */ +static int +snd_s1810c_get_switch_state(struct usb_mixer_interface *mixer, + struct snd_kcontrol *kctl, u32 *state) +{ + struct snd_usb_audio *chip = mixer->chip; + struct s1810_mixer_state *private = mixer->private_data; + u32 field = 0; + u32 ctl_idx = (u32) (kctl->private_value & 0xFF); + int ret = 0; + + mutex_lock(&private->usb_mutex); + ret = snd_sc1810c_get_status_field(chip->dev, &field, + ctl_idx, &private->seqnum); + if (ret < 0) + goto unlock; + + *state = field; + unlock: + mutex_unlock(&private->usb_mutex); + return ret ? ret : 0; +} + +/* + * Send a control packet to the device for the control id + * specified in (kctl->private_value >> 8) with value + * specified in (kctl->private_value >> 16). + */ +static int +snd_s1810c_set_switch_state(struct usb_mixer_interface *mixer, + struct snd_kcontrol *kctl) +{ + struct snd_usb_audio *chip = mixer->chip; + struct s1810_mixer_state *private = mixer->private_data; + u32 pval = (u32) kctl->private_value; + u32 ctl_id = (pval >> 8) & 0xFF; + u32 ctl_val = (pval >> 16) & 0x1; + int ret = 0; + + mutex_lock(&private->usb_mutex); + ret = snd_s1810c_send_ctl_packet(chip->dev, 0, 0, 0, ctl_id, ctl_val); + mutex_unlock(&private->usb_mutex); + return ret; +} + +/* Generic get/set/init functions for switch controls */ + +static int +snd_s1810c_switch_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ctl_elem) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + struct s1810_mixer_state *private = mixer->private_data; + u32 pval = (u32) kctl->private_value; + u32 ctl_idx = pval & 0xFF; + u32 state = 0; + int ret = 0; + + mutex_lock(&private->data_mutex); + ret = snd_s1810c_get_switch_state(mixer, kctl, &state); + if (ret < 0) + goto unlock; + + switch (ctl_idx) { + case SC1810C_STATE_LINE_SW: + case SC1810C_STATE_AB_SW: + ctl_elem->value.enumerated.item[0] = (int)state; + break; + default: + ctl_elem->value.integer.value[0] = (long)state; + } + + unlock: + mutex_unlock(&private->data_mutex); + return (ret < 0) ? ret : 0; +} + +static int +snd_s1810c_switch_set(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ctl_elem) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + struct s1810_mixer_state *private = mixer->private_data; + u32 pval = (u32) kctl->private_value; + u32 ctl_idx = pval & 0xFF; + u32 curval = 0; + u32 newval = 0; + int ret = 0; + + mutex_lock(&private->data_mutex); + ret = snd_s1810c_get_switch_state(mixer, kctl, &curval); + if (ret < 0) + goto unlock; + + switch (ctl_idx) { + case SC1810C_STATE_LINE_SW: + case SC1810C_STATE_AB_SW: + newval = (u32) ctl_elem->value.enumerated.item[0]; + break; + default: + newval = (u32) ctl_elem->value.integer.value[0]; + } + + if (curval == newval) + goto unlock; + + kctl->private_value &= ~(0x1 << 16); + kctl->private_value |= (unsigned int)(newval & 0x1) << 16; + ret = snd_s1810c_set_switch_state(mixer, kctl); + + unlock: + mutex_unlock(&private->data_mutex); + return (ret < 0) ? 0 : 1; +} + +static int +snd_s1810c_switch_init(struct usb_mixer_interface *mixer, + const struct snd_kcontrol_new *new_kctl) +{ + struct snd_kcontrol *kctl; + struct usb_mixer_elem_info *elem; + + elem = kzalloc(sizeof(struct usb_mixer_elem_info), GFP_KERNEL); + if (!elem) + return -ENOMEM; + + elem->head.mixer = mixer; + elem->control = 0; + elem->head.id = 0; + elem->channels = 1; + + kctl = snd_ctl_new1(new_kctl, elem); + if (!kctl) { + kfree(elem); + return -ENOMEM; + } + kctl->private_free = snd_usb_mixer_elem_free; + + return snd_usb_mixer_add_control(&elem->head, kctl); +} + +static int +snd_s1810c_line_sw_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const texts[2] = { + "Preamp On (Mic/Inst)", + "Preamp Off (Line in)" + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static const struct snd_kcontrol_new snd_s1810c_line_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line 1/2 Source Type", + .info = snd_s1810c_line_sw_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_LINE_SW | SC1810C_CTL_LINE_SW << 8) +}; + +static const struct snd_kcontrol_new snd_s1810c_mute_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mute Main Out Switch", + .info = snd_ctl_boolean_mono_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_MUTE_SW | SC1810C_CTL_MUTE_SW << 8) +}; + +static const struct snd_kcontrol_new snd_s1810c_48v_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "48V Phantom Power On Mic Inputs Switch", + .info = snd_ctl_boolean_mono_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_48V_SW | SC1810C_CTL_48V_SW << 8) +}; + +static int +snd_s1810c_ab_sw_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const texts[2] = { + "1/2", + "3/4" + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static const struct snd_kcontrol_new snd_s1810c_ab_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone 1 Source Route", + .info = snd_s1810c_ab_sw_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_AB_SW | SC1810C_CTL_AB_SW << 8) +}; + +static void snd_sc1810_mixer_state_free(struct usb_mixer_interface *mixer) +{ + struct s1810_mixer_state *private = mixer->private_data; + kfree(private); + mixer->private_data = NULL; +} + +/* Entry point, called from mixer_quirks.c */ +int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer) +{ + struct s1810_mixer_state *private = NULL; + struct snd_usb_audio *chip = mixer->chip; + struct usb_device *dev = chip->dev; + int ret = 0; + + /* Run this only once */ + if (!list_empty(&chip->mixer_list)) + return 0; + + dev_info(&dev->dev, + "Presonus Studio 1810c, device_setup: %u\n", chip->setup); + if (chip->setup == 1) + dev_info(&dev->dev, "(8out/18in @ 48KHz)\n"); + else if (chip->setup == 2) + dev_info(&dev->dev, "(6out/8in @ 192KHz)\n"); + else + dev_info(&dev->dev, "(8out/14in @ 96KHz)\n"); + + ret = snd_s1810c_init_mixer_maps(chip); + if (ret < 0) + return ret; + + private = kzalloc(sizeof(struct s1810_mixer_state), GFP_KERNEL); + if (!private) + return -ENOMEM; + + mutex_init(&private->usb_mutex); + mutex_init(&private->data_mutex); + + mixer->private_data = private; + mixer->private_free = snd_sc1810_mixer_state_free; + + private->seqnum = 1; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_line_sw); + if (ret < 0) + return ret; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_mute_sw); + if (ret < 0) + return ret; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_48v_sw); + if (ret < 0) + return ret; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_ab_sw); + if (ret < 0) + return ret; + return ret; +} diff --git a/sound/usb/mixer_s1810c.h b/sound/usb/mixer_s1810c.h new file mode 100644 index 000000000000..a79a3743cff3 --- /dev/null +++ b/sound/usb/mixer_s1810c.h @@ -0,0 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Presonus Studio 1810c driver for ALSA + * Copyright (C) 2019 Nick Kossifidis <mickflemm@gmail.com> + */ + +int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer); diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index bd258f1ec2dd..a4e4064f9aee 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -357,7 +357,12 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; ifnum = 1; goto add_sync_ep_from_ifnum; - case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */ + case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II/IIc */ + /* MicroBook IIc */ + if (altsd->bInterfaceClass == USB_CLASS_AUDIO) + return 0; + + /* MicroBook II */ ep = 0x84; ifnum = 0; goto add_sync_ep_from_ifnum; diff --git a/sound/usb/proc.c b/sound/usb/proc.c index ffbf4bd9208c..4174ad11fca6 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -70,7 +70,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s snd_iprintf(buffer, " Interface %d\n", fp->iface); snd_iprintf(buffer, " Altset %d\n", fp->altsetting); snd_iprintf(buffer, " Format:"); - for (fmt = 0; fmt <= SNDRV_PCM_FORMAT_LAST; ++fmt) + pcm_for_each_format(fmt) if (fp->formats & pcm_format_to_bits(fmt)) snd_iprintf(buffer, " %s", snd_pcm_format_name(fmt)); diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d187aa6d50db..1c8719292eee 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3472,7 +3472,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), }, /* MOTU Microbook II */ { - USB_DEVICE(0x07fd, 0x0004), + USB_DEVICE_VENDOR_SPEC(0x07fd, 0x0004), .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "MOTU", .product_name = "MicroBookII", diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 7f558f4b4520..86f192a3043d 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1252,6 +1252,38 @@ static int fasttrackpro_skip_setting_quirk(struct snd_usb_audio *chip, return 0; /* keep this altsetting */ } +static int s1810c_skip_setting_quirk(struct snd_usb_audio *chip, + int iface, int altno) +{ + /* + * Altno settings: + * + * Playback (Interface 1): + * 1: 6 Analog + 2 S/PDIF + * 2: 6 Analog + 2 S/PDIF + * 3: 6 Analog + * + * Capture (Interface 2): + * 1: 8 Analog + 2 S/PDIF + 8 ADAT + * 2: 8 Analog + 2 S/PDIF + 4 ADAT + * 3: 8 Analog + */ + + /* + * I'll leave 2 as the default one and + * use device_setup to switch to the + * other two. + */ + if ((chip->setup == 0 || chip->setup > 2) && altno != 2) + return 1; + else if (chip->setup == 1 && altno != 1) + return 1; + else if (chip->setup == 2 && altno != 3) + return 1; + + return 0; +} + int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, int iface, int altno) @@ -1265,6 +1297,10 @@ int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, /* fasttrackpro usb: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x0763, 0x2012)) return fasttrackpro_skip_setting_quirk(chip, iface, altno); + /* presonus studio 1810c: skip altsets incompatible with device_setup */ + if (chip->usb_id == USB_ID(0x0194f, 0x010c)) + return s1810c_skip_setting_quirk(chip, iface, altno); + return 0; } @@ -1316,7 +1352,15 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx 3 */ return snd_usb_axefx3_boot_quirk(dev); case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */ - return snd_usb_motu_microbookii_boot_quirk(dev); + /* + * For some reason interface 3 with vendor-spec class is + * detected on MicroBook IIc. + */ + if (get_iface_desc(intf->altsetting)->bInterfaceClass == + USB_CLASS_VENDOR_SPEC && + get_iface_desc(intf->altsetting)->bInterfaceNumber < 3) + return snd_usb_motu_microbookii_boot_quirk(dev); + break; } return 0; @@ -1754,5 +1798,47 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, else fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC; break; + case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook IIc */ + /* + * MaxPacketsOnly attribute is erroneously set in endpoint + * descriptors. As a result this card produces noise with + * all sample rates other than 96 KHz. + */ + fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX; + break; } } + +/* + * registration quirk: + * the registration is skipped if a device matches with the given ID, + * unless the interface reaches to the defined one. This is for delaying + * the registration until the last known interface, so that the card and + * devices appear at the same time. + */ + +struct registration_quirk { + unsigned int usb_id; /* composed via USB_ID() */ + unsigned int interface; /* the interface to trigger register */ +}; + +#define REG_QUIRK_ENTRY(vendor, product, iface) \ + { .usb_id = USB_ID(vendor, product), .interface = (iface) } + +static const struct registration_quirk registration_quirks[] = { + REG_QUIRK_ENTRY(0x0951, 0x16d8, 2), /* Kingston HyperX AMP */ + { 0 } /* terminator */ +}; + +/* return true if skipping registration */ +bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface) +{ + const struct registration_quirk *q; + + for (q = registration_quirks; q->usb_id; q++) + if (chip->usb_id == q->usb_id) + return iface != q->interface; + + /* Register as normal */ + return false; +} diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index df0355843a4c..c76cf24a640a 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -51,4 +51,6 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, struct audioformat *fp, int stream); +bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface); + #endif /* __USBAUDIO_QUIRKS_H */ diff --git a/sound/usb/stream.c b/sound/usb/stream.c index afd5aa574611..15296f2c902c 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -502,6 +502,9 @@ static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip, subs = &as->substream[stream]; if (subs->ep_num) continue; + if (snd_device_get_state(chip->card, as->pcm) != + SNDRV_DEV_BUILD) + chip->need_delayed_register = true; err = snd_pcm_new_stream(as->pcm, stream, 1); if (err < 0) return err; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 6fe3ab582ec6..1c892c7f14d7 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -34,6 +34,7 @@ struct snd_usb_audio { unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */ unsigned int tx_length_quirk:1; /* Put length specifier in transfers */ unsigned int setup_fmt_after_resume_quirk:1; /* setup the format to interface after resume */ + unsigned int need_delayed_register:1; /* warn for delayed registration */ int num_interfaces; int num_suspended_intf; int sample_rate_read_error; |