Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This contains pretty many small commits covering fairly large range of
  files in sound/ directory.  Partly because of additional API support
  and partly because of constantly developed ASoC and ARM stuff.

  Some highlights:

   - Introduced the helper function and documentation for exposing the
     channel map via control API, as discussed in Plumbers; most of PCI
     drivers are covered, will follow more drivers later

   - Most of drivers have been replaced with the new PM callbacks (if
     the bus is supported)

   - HD-audio controller got the support of runtime PM and the support
     of D3 clock-stop.  Also changing the power_save option in sysfs
     kicks off immediately to enable / disable the power-save mode.

   - Another significant code change in HD-audio is the rewrite of
     firmware loading code.  Other than that, most of changes in
     HD-audio are continued cleanups and standardization for the generic
     auto parser and bug fixes (HBR, device-specific fixups), in
     addition to the support of channel-map API.

   - Addition of ASoC bindings for the compressed API, used by the
     mid-x86 drivers.

   - Lots of cleanups and API refreshes for ASoC codec drivers and
     DaVinci.

   - Conversion of OMAP to dmaengine.

   - New machine driver for Wolfson Microelectronics Bells.

   - New CODEC driver for Wolfson Microelectronics WM0010.

   - Enhancements to the ux500 and wm2000 drivers

   - A new driver for DA9055 and the support for regulator bypass mode."

Fix up various arm soc header file reorg conflicts.

* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits)
  ALSA: hda - Add new codec ALC283 ALC290 support
  ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
  ALSA: hda - fix indices on boost volume on Conexant
  ALSA: aloop - add locking to timer access
  ALSA: hda - Fix hang caused by race during suspend.
  sound: Remove unnecessary semicolon
  ALSA: hda/realtek - Fix detection of ALC271X codec
  ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
  ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event
  ALSA: hda - make a generic unsol event handler
  ASoC: codecs: Add DA9055 codec driver
  ASoC: eukrea-tlv320: Convert it to platform driver
  ALSA: ASoC: add DT bindings for CS4271
  ASoC: wm_hubs: Ensure volume updates are handled during class W startup
  ASoC: wm5110: Adding missing volume update bits
  ASoC: wm5110: Add OUT3R support
  ASoC: wm5110: Add AEC loopback support
  ASoC: wm5110: Rename EPOUT to HPOUT3
  ASoC: arizona: Add more clock rates
  ASoC: arizona: Add more DSP options for mixer input muxes
  ...

1 files changed, 78 insertions, 48 deletions

diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index f782ce19bf5a..55e19e1b80ec 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -82,8 +82,7 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream
 /*
  * find a matching audio format
  */
-static struct audioformat *find_format(struct snd_usb_substream *subs, unsigned int format,
-				       unsigned int rate, unsigned int channels)
+static struct audioformat *find_format(struct snd_usb_substream *subs)
 {
 	struct list_head *p;
 	struct audioformat *found = NULL;
@@ -92,16 +91,17 @@ static struct audioformat *find_format(struct snd_usb_substream *subs, unsigned
 	list_for_each(p, &subs->fmt_list) {
 		struct audioformat *fp;
 		fp = list_entry(p, struct audioformat, list);
-		if (!(fp->formats & (1uLL << format)))
+		if (!(fp->formats & (1uLL << subs->pcm_format)))
 			continue;
-		if (fp->channels != channels)
+		if (fp->channels != subs->channels)
 			continue;
-		if (rate < fp->rate_min || rate > fp->rate_max)
+		if (subs->cur_rate < fp->rate_min ||
+		    subs->cur_rate > fp->rate_max)
 			continue;
 		if (! (fp->rates & SNDRV_PCM_RATE_CONTINUOUS)) {
 			unsigned int i;
 			for (i = 0; i < fp->nr_rates; i++)
-				if (fp->rate_table[i] == rate)
+				if (fp->rate_table[i] == subs->cur_rate)
 					break;
 			if (i >= fp->nr_rates)
 				continue;
@@ -436,6 +436,42 @@ add_sync_ep:
 }
 
 /*
+ * configure endpoint params
+ *
+ * called  during initial setup and upon resume
+ */
+static int configure_endpoint(struct snd_usb_substream *subs)
+{
+	int ret;
+
+	mutex_lock(&subs->stream->chip->shutdown_mutex);
+	/* format changed */
+	stop_endpoints(subs, 0, 0, 0);
+	ret = snd_usb_endpoint_set_params(subs->data_endpoint,
+					  subs->pcm_format,
+					  subs->channels,
+					  subs->period_bytes,
+					  subs->cur_rate,
+					  subs->cur_audiofmt,
+					  subs->sync_endpoint);
+	if (ret < 0)
+		goto unlock;
+
+	if (subs->sync_endpoint)
+		ret = snd_usb_endpoint_set_params(subs->data_endpoint,
+						  subs->pcm_format,
+						  subs->channels,
+						  subs->period_bytes,
+						  subs->cur_rate,
+						  subs->cur_audiofmt,
+						  NULL);
+
+unlock:
+	mutex_unlock(&subs->stream->chip->shutdown_mutex);
+	return ret;
+}
+
+/*
  * hw_params callback
  *
  * allocate a buffer and set the given audio format.
@@ -450,63 +486,33 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_usb_substream *subs = substream->runtime->private_data;
 	struct audioformat *fmt;
-	unsigned int channels, rate, format;
-	int ret, changed;
+	int ret;
 
 	ret = snd_pcm_lib_alloc_vmalloc_buffer(substream,
 					       params_buffer_bytes(hw_params));
 	if (ret < 0)
 		return ret;
 
-	format = params_format(hw_params);
-	rate = params_rate(hw_params);
-	channels = params_channels(hw_params);
-	fmt = find_format(subs, format, rate, channels);
+	subs->pcm_format = params_format(hw_params);
+	subs->period_bytes = params_period_bytes(hw_params);
+	subs->channels = params_channels(hw_params);
+	subs->cur_rate = params_rate(hw_params);
+
+	fmt = find_format(subs);
 	if (!fmt) {
 		snd_printd(KERN_DEBUG "cannot set format: format = %#x, rate = %d, channels = %d\n",
-			   format, rate, channels);
+			   subs->pcm_format, subs->cur_rate, subs->channels);
 		return -EINVAL;
 	}
 
-	changed = subs->cur_audiofmt != fmt ||
-		subs->period_bytes != params_period_bytes(hw_params) ||
-		subs->cur_rate != rate;
 	if ((ret = set_format(subs, fmt)) < 0)
 		return ret;
 
-	if (subs->cur_rate != rate) {
-		struct usb_host_interface *alts;
-		struct usb_interface *iface;
-		iface = usb_ifnum_to_if(subs->dev, fmt->iface);
-		alts = &iface->altsetting[fmt->altset_idx];
-		ret = snd_usb_init_sample_rate(subs->stream->chip, fmt->iface, alts, fmt, rate);
-		if (ret < 0)
-			return ret;
-		subs->cur_rate = rate;
-	}
-
-	if (changed) {
-		mutex_lock(&subs->stream->chip->shutdown_mutex);
-		/* format changed */
-		stop_endpoints(subs, 0, 0, 0);
-		ret = snd_usb_endpoint_set_params(subs->data_endpoint, hw_params, fmt,
-						  subs->sync_endpoint);
-		if (ret < 0)
-			goto unlock;
+	subs->interface = fmt->iface;
+	subs->altset_idx = fmt->altset_idx;
+	subs->need_setup_ep = true;
 
-		if (subs->sync_endpoint)
-			ret = snd_usb_endpoint_set_params(subs->sync_endpoint,
-							  hw_params, fmt, NULL);
-unlock:
-		mutex_unlock(&subs->stream->chip->shutdown_mutex);
-	}
-
-	if (ret == 0) {
-		subs->interface = fmt->iface;
-		subs->altset_idx = fmt->altset_idx;
-	}
-
-	return ret;
+	return 0;
 }
 
 /*
@@ -537,6 +543,9 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_usb_substream *subs = runtime->private_data;
+	struct usb_host_interface *alts;
+	struct usb_interface *iface;
+	int ret;
 
 	if (! subs->cur_audiofmt) {
 		snd_printk(KERN_ERR "usbaudio: no format is specified!\n");
@@ -546,6 +555,27 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
 	if (snd_BUG_ON(!subs->data_endpoint))
 		return -EIO;
 
+	ret = set_format(subs, subs->cur_audiofmt);
+	if (ret < 0)
+		return ret;
+
+	iface = usb_ifnum_to_if(subs->dev, subs->cur_audiofmt->iface);
+	alts = &iface->altsetting[subs->cur_audiofmt->altset_idx];
+	ret = snd_usb_init_sample_rate(subs->stream->chip,
+				       subs->cur_audiofmt->iface,
+				       alts,
+				       subs->cur_audiofmt,
+				       subs->cur_rate);
+	if (ret < 0)
+		return ret;
+
+	if (subs->need_setup_ep) {
+		ret = configure_endpoint(subs);
+		if (ret < 0)
+			return ret;
+		subs->need_setup_ep = false;
+	}
+
 	/* some unit conversions in runtime */
 	subs->data_endpoint->maxframesize =
 		bytes_to_frames(runtime, subs->data_endpoint->maxpacksize);