Merge branch 'for-linus' into for-next

6 files changed, 104 insertions, 19 deletions

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index b680b4e..8ec5289 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -687,12 +687,45 @@ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid,
 	return val;
 }
 
+/* is this a stereo widget or a stereo-to-mono mix? */
+static bool is_stereo_amps(struct hda_codec *codec, hda_nid_t nid, int dir)
+{
+	unsigned int wcaps = get_wcaps(codec, nid);
+	hda_nid_t conn;
+
+	if (wcaps & AC_WCAP_STEREO)
+		return true;
+	if (dir != HDA_INPUT || get_wcaps_type(wcaps) != AC_WID_AUD_MIX)
+		return false;
+	if (snd_hda_get_num_conns(codec, nid) != 1)
+		return false;
+	if (snd_hda_get_connections(codec, nid, &conn, 1) < 0)
+		return false;
+	return !!(get_wcaps(codec, conn) & AC_WCAP_STEREO);
+}
+
 /* initialize the amp value (only at the first time) */
 static void init_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx)
 {
 	unsigned int caps = query_amp_caps(codec, nid, dir);
 	int val = get_amp_val_to_activate(codec, nid, dir, caps, false);
-	snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val);
+
+	if (is_stereo_amps(codec, nid, dir))
+		snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val);
+	else
+		snd_hda_codec_amp_init(codec, nid, 0, dir, idx, 0xff, val);
+}
+
+/* update the amp, doing in stereo or mono depending on NID */
+static int update_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx,
+		      unsigned int mask, unsigned int val)
+{
+	if (is_stereo_amps(codec, nid, dir))
+		return snd_hda_codec_amp_stereo(codec, nid, dir, idx,
+						mask, val);
+	else
+		return snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
+						mask, val);
 }
 
 /* calculate amp value mask we can modify;
@@ -732,7 +765,7 @@ static void activate_amp(struct hda_codec *codec, hda_nid_t nid, int dir,
 		return;
 
 	val &= mask;
-	snd_hda_codec_amp_stereo(codec, nid, dir, idx, mask, val);
+	update_amp(codec, nid, dir, idx, mask, val);
 }
 
 static void activate_amp_out(struct hda_codec *codec, struct nid_path *path,
@@ -4424,13 +4457,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix)
 	has_amp = nid_has_mute(codec, mix, HDA_INPUT);
 	for (i = 0; i < nums; i++) {
 		if (has_amp)
-			snd_hda_codec_amp_stereo(codec, mix,
-						 HDA_INPUT, i,
-						 0xff, HDA_AMP_MUTE);
+			update_amp(codec, mix, HDA_INPUT, i,
+				   0xff, HDA_AMP_MUTE);
 		else if (nid_has_volume(codec, conn[i], HDA_OUTPUT))
-			snd_hda_codec_amp_stereo(codec, conn[i],
-						 HDA_OUTPUT, 0,
-						 0xff, HDA_AMP_MUTE);
+			update_amp(codec, conn[i], HDA_OUTPUT, 0,
+				   0xff, HDA_AMP_MUTE);
 	}
 }
 
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index ce5a6da..05e19f7 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -134,13 +134,38 @@ static void print_amp_caps(struct snd_info_buffer *buffer,
 		    (caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT);
 }
 
+/* is this a stereo widget or a stereo-to-mono mix? */
+static bool is_stereo_amps(struct hda_codec *codec, hda_nid_t nid,
+			   int dir, unsigned int wcaps, int indices)
+{
+	hda_nid_t conn;
+
+	if (wcaps & AC_WCAP_STEREO)
+		return true;
+	/* check for a stereo-to-mono mix; it must be:
+	 * only a single connection, only for input, and only a mixer widget
+	 */
+	if (indices != 1 || dir != HDA_INPUT ||
+	    get_wcaps_type(wcaps) != AC_WID_AUD_MIX)
+		return false;
+
+	if (snd_hda_get_raw_connections(codec, nid, &conn, 1) < 0)
+		return false;
+	/* the connection source is a stereo? */
+	wcaps = snd_hda_param_read(codec, conn, AC_PAR_AUDIO_WIDGET_CAP);
+	return !!(wcaps & AC_WCAP_STEREO);
+}
+
 static void print_amp_vals(struct snd_info_buffer *buffer,
 			   struct hda_codec *codec, hda_nid_t nid,
-			   int dir, int stereo, int indices)
+			   int dir, unsigned int wcaps, int indices)
 {
 	unsigned int val;
+	bool stereo;
 	int i;
 
+	stereo = is_stereo_amps(codec, nid, dir, wcaps, indices);
+
 	dir = dir == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT;
 	for (i = 0; i < indices; i++) {
 		snd_iprintf(buffer, " [");
@@ -757,12 +782,10 @@ static void print_codec_info(struct snd_info_entry *entry,
 			    (codec->single_adc_amp &&
 			     wid_type == AC_WID_AUD_IN))
 				print_amp_vals(buffer, codec, nid, HDA_INPUT,
-					       wid_caps & AC_WCAP_STEREO,
-					       1);
+					       wid_caps, 1);
 			else
 				print_amp_vals(buffer, codec, nid, HDA_INPUT,
-					       wid_caps & AC_WCAP_STEREO,
-					       conn_len);
+					       wid_caps, conn_len);
 		}
 		if (wid_caps & AC_WCAP_OUT_AMP) {
 			snd_iprintf(buffer, "  Amp-Out caps: ");
@@ -771,11 +794,10 @@ static void print_codec_info(struct snd_info_entry *entry,
 			if (wid_type == AC_WID_PIN &&
 			    codec->pin_amp_workaround)
 				print_amp_vals(buffer, codec, nid, HDA_OUTPUT,
-					       wid_caps & AC_WCAP_STEREO,
-					       conn_len);
+					       wid_caps, conn_len);
 			else
 				print_amp_vals(buffer, codec, nid, HDA_OUTPUT,
-					       wid_caps & AC_WCAP_STEREO, 1);
+					       wid_caps, 1);
 		}
 
 		switch (wid_type) {
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 1589c9b..dd2b3d9 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -393,6 +393,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81),
 	SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122),
 	SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101),
+	SND_PCI_QUIRK(0x106b, 0x5600, "MacBookAir 5,2", CS420X_MBP81),
 	SND_PCI_QUIRK(0x106b, 0x5b00, "MacBookAir 4,2", CS420X_MBA42),
 	SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE),
 	{} /* terminator */
@@ -584,6 +585,7 @@ static int patch_cs420x(struct hda_codec *codec)
 		return -ENOMEM;
 
 	spec->gen.automute_hook = cs_automute;
+	codec->single_adc_amp = 1;
 
 	snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl,
 			   cs420x_fixups);
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 75870c0..91eb3ae 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -1049,7 +1049,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv,
 				enum spdif_txrate index, bool round)
 {
 	const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 };
-	bool is_sysclk = clk == spdif_priv->sysclk;
+	bool is_sysclk = clk_is_match(clk, spdif_priv->sysclk);
 	u64 rate_ideal, rate_actual, sub;
 	u32 sysclk_dfmin, sysclk_dfmax;
 	u32 txclk_df, sysclk_df, arate;
@@ -1143,7 +1143,7 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv,
 			spdif_priv->txclk_src[index], rate[index]);
 	dev_dbg(&pdev->dev, "use txclk df %d for %dHz sample rate\n",
 			spdif_priv->txclk_df[index], rate[index]);
-	if (spdif_priv->txclk[index] == spdif_priv->sysclk)
+	if (clk_is_match(spdif_priv->txclk[index], spdif_priv->sysclk))
 		dev_dbg(&pdev->dev, "use sysclk df %d for %dHz sample rate\n",
 				spdif_priv->sysclk_df[index], rate[index]);
 	dev_dbg(&pdev->dev, "the best rate for %dHz sample rate is %dHz\n",
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index def7d82..d194830 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -579,7 +579,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 		if (PTR_ERR(priv->extclk) == -EPROBE_DEFER)
 			return -EPROBE_DEFER;
 	} else {
-		if (priv->extclk == priv->clk) {
+		if (clk_is_match(priv->extclk, priv->clk)) {
 			devm_clk_put(&pdev->dev, priv->extclk);
 			priv->extclk = ERR_PTR(-EINVAL);
 		} else {
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 67d4765..07f984d 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1773,6 +1773,36 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		}
 	}
 },
+{
+	USB_DEVICE(0x0582, 0x0159),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		/* .vendor_name = "Roland", */
+		/* .product_name = "UA-22", */
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 1,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_MIDI_FIXED_ENDPOINT,
+				.data = & (const struct snd_usb_midi_endpoint_info) {
+					.out_cables = 0x0001,
+					.in_cables = 0x0001
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
 /* this catches most recent vendor-specific Roland devices */
 {
 	.match_flags = USB_DEVICE_ID_MATCH_VENDOR |