Merge branch 'for-linus' into for-next

Back-merge for cleaning up usb-audio code the recent commit modified,
and further UAC2 autoclock patches.

13 files changed, 73 insertions, 27 deletions

diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 7dd846380a50..d0d7ac1e99d2 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/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 3e1159dd98c3..ede82156da0f 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1194,7 +1194,7 @@ 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/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index fc176044994d..fc176044994d 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 7e103f249053..7e103f249053 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 f2d61a187830..566ea3256e2d 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/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index f3f7e75f8628..9af1bddc4c62 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 55464a5b0706..810c7eeb7b03 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 8e52c1485df3..eb4373840bb6 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/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 19eff8fc4fdd..1a8b03e4b41b 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-core.c b/sound/soc/soc-core.c
index b7e84a7cd9ee..507d251916af 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -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;
@@ -4197,7 +4199,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
                         dev_err(card->dev,
                                 "ASoC: Property '%s' index %d could not be read: %d\n",
                                 propname, 2 * i, ret);
-                        kfree(routes);
                         return -EINVAL;
                 }
                 ret = of_property_read_string_index(np, propname,
@@ -4206,7 +4207,6 @@ 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 1d6a9b3ceb27..d6d9ba2e6916 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 9b76cc5a1148..5e7aebe1e664 100644
--- a/sound/soc/spear/spear_pcm.c
+++ b/sound/soc/spear/spear_pcm.c
@@ -149,9 +149,9 @@ static void spear_pcm_free(struct snd_pcm *pcm)
 
 static u64 spear_pcm_dmamask = DMA_BIT_MASK(32);
 
-static int spear_pcm_new(struct snd_card *card,
-                struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int spear_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+        struct snd_card *card = rtd->card->snd_card;
         int ret;
 
         if (!card->dev->dma_mask)
@@ -159,16 +159,16 @@ static int spear_pcm_new(struct snd_card *card,
         if (!card->dev->coherent_dma_mask)
                 card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
 
-        if (dai->driver->playback.channels_min) {
-                ret = spear_pcm_preallocate_dma_buffer(pcm,
+        if (rtd->cpu_dai->driver->playback.channels_min) {
+                ret = spear_pcm_preallocate_dma_buffer(rtd->pcm,
                                 SNDRV_PCM_STREAM_PLAYBACK,
                                 spear_pcm_hardware.buffer_bytes_max);
                 if (ret)
                         return ret;
         }
 
-        if (dai->driver->capture.channels_min) {
-                ret = spear_pcm_preallocate_dma_buffer(pcm,
+        if (rtd->cpu_dai->driver->capture.channels_min) {
+                ret = spear_pcm_preallocate_dma_buffer(rtd->pcm,
                                 SNDRV_PCM_STREAM_CAPTURE,
                                 spear_pcm_hardware.buffer_bytes_max);
                 if (ret)
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 5e634a2eb282..9e2703a25156 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;
 }