Merge branch 'for-linus' into for-next

author: Takashi Iwai <tiwai@suse.de> 2015-03-16 09:48:05 -0400
committer: Takashi Iwai <tiwai@suse.de> 2015-03-16 09:48:05 -0400
commit: 34e72afe73c40d9974c1f230c3b62fc43f5c5b28 (patch)
tree: fee2be2e3453e246ee55ffb711a36a8ec053361a /sound
parent: e79d74ab25339437447478e4dfe2b35c5b560512 (diff)
parent: cc261738add93947d138d2fabad9f4dbed4e5c00 (diff)
6 files changed, 104 insertions, 19 deletions
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index b680b4ec6331..8ec5289f8e05 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,
+                        update_amp(codec, mix, HDA_INPUT, i,
-                                                 HDA_INPUT, i,
+                                   0xff, HDA_AMP_MUTE);
-                                                 0xff, HDA_AMP_MUTE);
                else if (nid_has_volume(codec, conn[i], HDA_OUTPUT))
-                        snd_hda_codec_amp_stereo(codec, conn[i],
+                        update_amp(codec, conn[i], HDA_OUTPUT, 0,
-                                                 HDA_OUTPUT, 0,
+                                   0xff, HDA_AMP_MUTE);
-                                                 0xff, HDA_AMP_MUTE);
        }
 }
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index ce5a6da83419..05e19f78b4cb 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,
+                                               wid_caps, 1);
-                                               1);
                        else
                                print_amp_vals(buffer, codec, nid, HDA_INPUT,
-                                               wid_caps & AC_WCAP_STEREO,
+                                               wid_caps, conn_len);
-                                               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,
+                                               wid_caps, conn_len);
-                                               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 1589c9bcce3e..dd2b3d92071f 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 75870c0ea2c9..91eb3aef7f02 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 def7d8260c4e..d19483081f9b 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 67d476548dcf..07f984d5f516 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 |
author	Takashi Iwai <tiwai@suse.de>	2015-03-16 09:48:05 -0400
committer	Takashi Iwai <tiwai@suse.de>	2015-03-16 09:48:05 -0400
commit	34e72afe73c40d9974c1f230c3b62fc43f5c5b28 (patch)
tree	fee2be2e3453e246ee55ffb711a36a8ec053361a /sound
parent	e79d74ab25339437447478e4dfe2b35c5b560512 (diff)
parent	cc261738add93947d138d2fabad9f4dbed4e5c00 (diff)

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b680b4ec6331..8ec5289f8e05 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,
687	return val;	687	return val;
688	}	688	}
689		689
		690	/* is this a stereo widget or a stereo-to-mono mix? */
		691	static bool is_stereo_amps(struct hda_codec *codec, hda_nid_t nid, int dir)
		692	{
		693	unsigned int wcaps = get_wcaps(codec, nid);
		694	hda_nid_t conn;
		695
		696	if (wcaps & AC_WCAP_STEREO)
		697	return true;
		698	if (dir != HDA_INPUT \|\| get_wcaps_type(wcaps) != AC_WID_AUD_MIX)
		699	return false;
		700	if (snd_hda_get_num_conns(codec, nid) != 1)
		701	return false;
		702	if (snd_hda_get_connections(codec, nid, &conn, 1) < 0)
		703	return false;
		704	return !!(get_wcaps(codec, conn) & AC_WCAP_STEREO);
		705	}
		706
690	/* initialize the amp value (only at the first time) */	707	/* initialize the amp value (only at the first time) */
691	static void init_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx)	708	static void init_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx)
692	{	709	{
693	unsigned int caps = query_amp_caps(codec, nid, dir);	710	unsigned int caps = query_amp_caps(codec, nid, dir);
694	int val = get_amp_val_to_activate(codec, nid, dir, caps, false);	711	int val = get_amp_val_to_activate(codec, nid, dir, caps, false);
695	snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val);	712
		713	if (is_stereo_amps(codec, nid, dir))
		714	snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val);
		715	else
		716	snd_hda_codec_amp_init(codec, nid, 0, dir, idx, 0xff, val);
		717	}
		718
		719	/* update the amp, doing in stereo or mono depending on NID */
		720	static int update_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx,
		721	unsigned int mask, unsigned int val)
		722	{
		723	if (is_stereo_amps(codec, nid, dir))
		724	return snd_hda_codec_amp_stereo(codec, nid, dir, idx,
		725	mask, val);
		726	else
		727	return snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
		728	mask, val);
696	}	729	}
697		730
698	/* calculate amp value mask we can modify;	731	/* 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,
732	return;	765	return;
733		766
734	val &= mask;	767	val &= mask;
735	snd_hda_codec_amp_stereo(codec, nid, dir, idx, mask, val);	768	update_amp(codec, nid, dir, idx, mask, val);
736	}	769	}
737		770
738	static void activate_amp_out(struct hda_codec codec, struct nid_path path,	771	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)
4424	has_amp = nid_has_mute(codec, mix, HDA_INPUT);	4457	has_amp = nid_has_mute(codec, mix, HDA_INPUT);
4425	for (i = 0; i < nums; i++) {	4458	for (i = 0; i < nums; i++) {
4426	if (has_amp)	4459	if (has_amp)
4427	snd_hda_codec_amp_stereo(codec, mix,	4460	update_amp(codec, mix, HDA_INPUT, i,
4428	HDA_INPUT, i,	4461	0xff, HDA_AMP_MUTE);
4429	0xff, HDA_AMP_MUTE);
4430	else if (nid_has_volume(codec, conn[i], HDA_OUTPUT))	4462	else if (nid_has_volume(codec, conn[i], HDA_OUTPUT))
4431	snd_hda_codec_amp_stereo(codec, conn[i],	4463	update_amp(codec, conn[i], HDA_OUTPUT, 0,
4432	HDA_OUTPUT, 0,	4464	0xff, HDA_AMP_MUTE);
4433	0xff, HDA_AMP_MUTE);
4434	}	4465	}
4435	}	4466	}
4436		4467


diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index ce5a6da83419..05e19f78b4cb 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,
134	(caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT);	134	(caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT);
135	}	135	}
136		136
		137	/* is this a stereo widget or a stereo-to-mono mix? */
		138	static bool is_stereo_amps(struct hda_codec *codec, hda_nid_t nid,
		139	int dir, unsigned int wcaps, int indices)
		140	{
		141	hda_nid_t conn;
		142
		143	if (wcaps & AC_WCAP_STEREO)
		144	return true;
		145	/* check for a stereo-to-mono mix; it must be:
		146	* only a single connection, only for input, and only a mixer widget
		147	*/
		148	if (indices != 1 \|\| dir != HDA_INPUT \|\|
		149	get_wcaps_type(wcaps) != AC_WID_AUD_MIX)
		150	return false;
		151
		152	if (snd_hda_get_raw_connections(codec, nid, &conn, 1) < 0)
		153	return false;
		154	/* the connection source is a stereo? */
		155	wcaps = snd_hda_param_read(codec, conn, AC_PAR_AUDIO_WIDGET_CAP);
		156	return !!(wcaps & AC_WCAP_STEREO);
		157	}
		158
137	static void print_amp_vals(struct snd_info_buffer *buffer,	159	static void print_amp_vals(struct snd_info_buffer *buffer,
138	struct hda_codec *codec, hda_nid_t nid,	160	struct hda_codec *codec, hda_nid_t nid,
139	int dir, int stereo, int indices)	161	int dir, unsigned int wcaps, int indices)
140	{	162	{
141	unsigned int val;	163	unsigned int val;
		164	bool stereo;
142	int i;	165	int i;
143		166
		167	stereo = is_stereo_amps(codec, nid, dir, wcaps, indices);
		168
144	dir = dir == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT;	169	dir = dir == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT;
145	for (i = 0; i < indices; i++) {	170	for (i = 0; i < indices; i++) {
146	snd_iprintf(buffer, " [");	171	snd_iprintf(buffer, " [");
@@ -757,12 +782,10 @@ static void print_codec_info(struct snd_info_entry *entry,
757	(codec->single_adc_amp &&	782	(codec->single_adc_amp &&
758	wid_type == AC_WID_AUD_IN))	783	wid_type == AC_WID_AUD_IN))
759	print_amp_vals(buffer, codec, nid, HDA_INPUT,	784	print_amp_vals(buffer, codec, nid, HDA_INPUT,
760	wid_caps & AC_WCAP_STEREO,	785	wid_caps, 1);
761	1);
762	else	786	else
763	print_amp_vals(buffer, codec, nid, HDA_INPUT,	787	print_amp_vals(buffer, codec, nid, HDA_INPUT,
764	wid_caps & AC_WCAP_STEREO,	788	wid_caps, conn_len);
765	conn_len);
766	}	789	}
767	if (wid_caps & AC_WCAP_OUT_AMP) {	790	if (wid_caps & AC_WCAP_OUT_AMP) {
768	snd_iprintf(buffer, " Amp-Out caps: ");	791	snd_iprintf(buffer, " Amp-Out caps: ");
@@ -771,11 +794,10 @@ static void print_codec_info(struct snd_info_entry *entry,
771	if (wid_type == AC_WID_PIN &&	794	if (wid_type == AC_WID_PIN &&
772	codec->pin_amp_workaround)	795	codec->pin_amp_workaround)
773	print_amp_vals(buffer, codec, nid, HDA_OUTPUT,	796	print_amp_vals(buffer, codec, nid, HDA_OUTPUT,
774	wid_caps & AC_WCAP_STEREO,	797	wid_caps, conn_len);
775	conn_len);
776	else	798	else
777	print_amp_vals(buffer, codec, nid, HDA_OUTPUT,	799	print_amp_vals(buffer, codec, nid, HDA_OUTPUT,
778	wid_caps & AC_WCAP_STEREO, 1);	800	wid_caps, 1);
779	}	801	}
780		802
781	switch (wid_type) {	803	switch (wid_type) {


diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 1589c9bcce3e..dd2b3d92071f 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[] = {
393	SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81),	393	SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81),
394	SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122),	394	SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122),
395	SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101),	395	SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101),
		396	SND_PCI_QUIRK(0x106b, 0x5600, "MacBookAir 5,2", CS420X_MBP81),
396	SND_PCI_QUIRK(0x106b, 0x5b00, "MacBookAir 4,2", CS420X_MBA42),	397	SND_PCI_QUIRK(0x106b, 0x5b00, "MacBookAir 4,2", CS420X_MBA42),
397	SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE),	398	SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE),
398	{} /* terminator */	399	{} /* terminator */
@@ -584,6 +585,7 @@ static int patch_cs420x(struct hda_codec *codec)
584	return -ENOMEM;	585	return -ENOMEM;
585		586
586	spec->gen.automute_hook = cs_automute;	587	spec->gen.automute_hook = cs_automute;
		588	codec->single_adc_amp = 1;
587		589
588	snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl,	590	snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl,
589	cs420x_fixups);	591	cs420x_fixups);


diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 75870c0ea2c9..91eb3aef7f02 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,
1049	enum spdif_txrate index, bool round)	1049	enum spdif_txrate index, bool round)
1050	{	1050	{
1051	const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 };	1051	const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 };
1052	bool is_sysclk = clk == spdif_priv->sysclk;	1052	bool is_sysclk = clk_is_match(clk, spdif_priv->sysclk);
1053	u64 rate_ideal, rate_actual, sub;	1053	u64 rate_ideal, rate_actual, sub;
1054	u32 sysclk_dfmin, sysclk_dfmax;	1054	u32 sysclk_dfmin, sysclk_dfmax;
1055	u32 txclk_df, sysclk_df, arate;	1055	u32 txclk_df, sysclk_df, arate;
@@ -1143,7 +1143,7 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv,
1143	spdif_priv->txclk_src[index], rate[index]);	1143	spdif_priv->txclk_src[index], rate[index]);
1144	dev_dbg(&pdev->dev, "use txclk df %d for %dHz sample rate\n",	1144	dev_dbg(&pdev->dev, "use txclk df %d for %dHz sample rate\n",
1145	spdif_priv->txclk_df[index], rate[index]);	1145	spdif_priv->txclk_df[index], rate[index]);
1146	if (spdif_priv->txclk[index] == spdif_priv->sysclk)	1146	if (clk_is_match(spdif_priv->txclk[index], spdif_priv->sysclk))
1147	dev_dbg(&pdev->dev, "use sysclk df %d for %dHz sample rate\n",	1147	dev_dbg(&pdev->dev, "use sysclk df %d for %dHz sample rate\n",
1148	spdif_priv->sysclk_df[index], rate[index]);	1148	spdif_priv->sysclk_df[index], rate[index]);
1149	dev_dbg(&pdev->dev, "the best rate for %dHz sample rate is %dHz\n",	1149	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 def7d8260c4e..d19483081f9b 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)
579	if (PTR_ERR(priv->extclk) == -EPROBE_DEFER)	579	if (PTR_ERR(priv->extclk) == -EPROBE_DEFER)
580	return -EPROBE_DEFER;	580	return -EPROBE_DEFER;
581	} else {	581	} else {
582	if (priv->extclk == priv->clk) {	582	if (clk_is_match(priv->extclk, priv->clk)) {
583	devm_clk_put(&pdev->dev, priv->extclk);	583	devm_clk_put(&pdev->dev, priv->extclk);
584	priv->extclk = ERR_PTR(-EINVAL);	584	priv->extclk = ERR_PTR(-EINVAL);
585	} else {	585	} else {


diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 67d476548dcf..07f984d5f516 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h
@@ -1773,6 +1773,36 @@ YAMAHA_DEVICE(0x7010, "UB99"),
1773	}	1773	}
1774	}	1774	}
1775	},	1775	},
		1776	{
		1777	USB_DEVICE(0x0582, 0x0159),
		1778	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
		1779	/* .vendor_name = "Roland", */
		1780	/* .product_name = "UA-22", */
		1781	.ifnum = QUIRK_ANY_INTERFACE,
		1782	.type = QUIRK_COMPOSITE,
		1783	.data = (const struct snd_usb_audio_quirk[]) {
		1784	{
		1785	.ifnum = 0,
		1786	.type = QUIRK_AUDIO_STANDARD_INTERFACE
		1787	},
		1788	{
		1789	.ifnum = 1,
		1790	.type = QUIRK_AUDIO_STANDARD_INTERFACE
		1791	},
		1792	{
		1793	.ifnum = 2,
		1794	.type = QUIRK_MIDI_FIXED_ENDPOINT,
		1795	.data = & (const struct snd_usb_midi_endpoint_info) {
		1796	.out_cables = 0x0001,
		1797	.in_cables = 0x0001
		1798	}
		1799	},
		1800	{
		1801	.ifnum = -1
		1802	}
		1803	}
		1804	}
		1805	},
1776	/* this catches most recent vendor-specific Roland devices */	1806	/* this catches most recent vendor-specific Roland devices */
1777	{	1807	{
1778	.match_flags = USB_DEVICE_ID_MATCH_VENDOR \|	1808	.match_flags = USB_DEVICE_ID_MATCH_VENDOR \|