February 2020 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: hda/realtek - Fix silent output on MSI-GL73
by Takashi Iwai 12 Feb '20

12 Feb '20

MSI-GL73 laptop with ALC1220 codec requires a similar workaround for Clevo laptops to enforce the DAC/mixer connection path. Set up a quirk entry for that. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204159 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3ee88adf57e7..6c8cb4ce517e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2447,6 +2447,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), + SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), -- 2.16.4

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[alsa-devel] Add more codec supported Headset Button
by Kailang 12 Feb '20

12 Feb '20

Hi Takashi, Attach patch. Add supported Headset Button for ALC215/ALC285/ALC289. Many Thanks. BR, Kailang

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[alsa-devel] [PATCH] ALSA: hda_codec: Replace zero-length array with flexible-array member
by Gustavo A. R. Silva 12 Feb '20

12 Feb '20

The current codebase makes use of the zero-length array language extension to the C90 standard, but the preferred mechanism to declare variable-length types such as these ones is a flexible array member[1][2], introduced in C99: struct foo { int stuff; struct boo array[]; }; By making use of the mechanism above, we will get a compiler warning in case the flexible array does not occur last in the structure, which will help us prevent some kind of undefined behavior bugs from being inadvertenly introduced[3] to the codebase from now on. This issue was found with the help of Coccinelle. [1] https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html [2] https://github.com/KSPP/linux/issues/21 [3] commit 76497732932f ("cxgb3/l2t: Fix undefined behaviour") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 701a69d856f5..0fbca35509cd 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -102,7 +102,7 @@ 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 */ -- 2.25.0

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[alsa-devel] [PATCH] ALSA: hda/ca0132 - Replace zero-length array with flexible-array member
by Gustavo A. R. Silva 12 Feb '20

12 Feb '20

The current codebase makes use of the zero-length array language extension to the C90 standard, but the preferred mechanism to declare variable-length types such as these ones is a flexible array member[1][2], introduced in C99: struct foo { int stuff; struct boo array[]; }; By making use of the mechanism above, we will get a compiler warning in case the flexible array does not occur last in the structure, which will help us prevent some kind of undefined behavior bugs from being inadvertenly introduced[3] to the codebase from now on. This issue was found with the help of Coccinelle. [1] https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html [2] https://github.com/KSPP/linux/issues/21 [3] commit 76497732932f ("cxgb3/l2t: Fix undefined behaviour") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/pci/hda/patch_ca0132.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index e1ebc6d5f382..ab5632502206 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2708,7 +2708,7 @@ struct dsp_image_seg { u32 magic; u32 chip_addr; u32 count; - u32 data[0]; + u32 data[]; }; static const u32 g_magic_value = 0x4c46584d; -- 2.25.0

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[alsa-devel] [PATCH] ALSA: usb-midi: Replace zero-length array with flexible-array member
by Gustavo A. R. Silva 12 Feb '20

12 Feb '20

The current codebase makes use of the zero-length array language extension to the C90 standard, but the preferred mechanism to declare variable-length types such as these ones is a flexible array member[1][2], introduced in C99: struct foo { int stuff; struct boo array[]; }; By making use of the mechanism above, we will get a compiler warning in case the flexible array does not occur last in the structure, which will help us prevent some kind of undefined behavior bugs from being inadvertenly introduced[3] to the codebase from now on. This issue was found with the help of Coccinelle. [1] https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html [2] https://github.com/KSPP/linux/issues/21 [3] commit 76497732932f ("cxgb3/l2t: Fix undefined behaviour") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/usb/midi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index b737f0ec77d0..f15d36731cbe 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; -- 2.25.0

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[alsa-devel] [PATCH] ALSA: core: Replace zero-length array with flexible-array member
by Gustavo A. R. Silva 12 Feb '20

12 Feb '20

The current codebase makes use of the zero-length array language extension to the C90 standard, but the preferred mechanism to declare variable-length types such as these ones is a flexible array member[1][2], introduced in C99: struct foo { int stuff; struct boo array[]; }; By making use of the mechanism above, we will get a compiler warning in case the flexible array does not occur last in the structure, which will help us prevent some kind of undefined behavior bugs from being inadvertenly introduced[3] to the codebase from now on. This issue was found with the help of Coccinelle. [1] https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html [2] https://github.com/KSPP/linux/issues/21 [3] commit 76497732932f ("cxgb3/l2t: Fix undefined behaviour") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/core/oss/rate.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c index 2fa9299a440d..e9f20fefb117 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) -- 2.25.0

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[alsa-devel] [PATCH 1/2] ALSA: hda/realtek - Fix Lenovo Thinkpad X1 Carbon 7th quirk subdevice id
by Benjamin Poirier 12 Feb '20

12 Feb '20

fixes the pci subsystem device ids for the "Thinkpad X1 Carbon 7th" and "Thinkpad X1 Yoga 7th" quirks. My machine reports the following: dmidecode -t system Manufacturer: LENOVO Product Name: 20QDCTO1WW Version: ThinkPad X1 Carbon 7th lspci -s 1f.3 -vnn 00:1f.3 Audio device [0403]: Intel Corporation Cannon Point-LP High Definition Audio Controller [8086:9dc8] (rev 11) (prog-if 80) Subsystem: Lenovo Cannon Point-LP High Definition Audio Controller [17aa:2292] /proc/asound/card0/codec#0 Subsystem Id: 0x17aa2293 Notice the different subsystem device ids between pci info and codec info. commit d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen") added a quirk meant for the X1 Carbon but used device id 0x2293. Note that this does not match the PCI SSID but it matches the codec SSID. commit 54a6a7dc107d ("ALSA: hda/realtek - Add quirk for the bass speaker on Lenovo Yoga X1 7th gen") added a quirk meant for the X1 Yoga but used subdevice id 0x2292, the PCI SSID used on the X1 Carbon. Given that in snd_hdac_device_init() quirks are first matched by PCI SSID and then, if there is no match, by codec SSID, the net result is that the quirk labelled "Thinkpad X1 Yoga 7th" now gets applied on the X1 Carbon. Example from my machine (an X1 Carbon, not Yoga): [ 15.817637] snd_hda_codec_realtek hdaudioC0D0: ALC285: picked fixup Thinkpad X1 Yoga 7th (PCI SSID) Therefore, fix the subdevice id for the "ThinkPad X1 Carbon 7th" quirk. Note that looking through the lspci outputs collected at https://github.com/linuxhw/LsPCI/tree/master/Notebook/Lenovo/ThinkPad all X1 Carbon there have an Audio device with PCI SSID 0x2292, which matches with the output from my machine. This leaves the question of what to do with the quirk labelled "Thinkpad X1 Yoga 7th". >From email discussions, it seems that the author of commit 54a6a7dc107d ("ALSA: hda/realtek - Add quirk for the bass speaker on Lenovo Yoga X1 7th gen") did not have a device to test the changes. I don't have an X1 Yoga either and I did not find a sample lspci listing for it online. Therefore, the best course of action seems to be to remove that quirk. In the best case, the quirk for the X1 Carbon will match the X1 Yoga (via PCI SSID or codec SSID). In the worst case, it will not and someone who actually has such a machine should come forth with concrete data about subsystem ids and needed quirks. Fixes: d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen") Fixes: 54a6a7dc107d ("ALSA: hda/realtek - Add quirk for the bass speaker on Lenovo Yoga X1 7th gen") Link: https://lore.kernel.org/alsa-devel/20200210025249.GA2700@f3/ Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Kailang Yang <kailang(a)realtek.com> Signed-off-by: Benjamin Poirier <benjamin.poirier(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4770fb3f51fb..05d44df2008e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7268,8 +7268,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_SPEAKER2_TO_DAC1), - SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_SPEAKER2_TO_DAC1), + SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_SPEAKER2_TO_DAC1), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), -- 2.25.0

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[alsa-devel] MIDI 2.0 Universal MIDI Packet format (UMP)
by GitHub issues - edited 12 Feb '20

12 Feb '20

alsa-project/alsa-lib issue #24 was edited from symdeb: Since MIDI 2.0 only support USB and not over DIN (for now), will MIDI 2.0 need a new or updated OS class driver for the 32/64 bit universal packet interface and device descriptor for devices ? Page 16 of the USB MIDI class driver spec defines support for 1,2 or 3 bytes transfers only, How will this have to be implemented by MIDI USB device manufacturers and well as on the Linux OSes side ? How will this impact Linux ALSA ? Would the first byte of the message with b7=0 not confuse the driver of running status ? Link universal packet format https://www.midi.org/articles-old/details-about-midi-2-0-midi-ci-profiles-a…) Link first MIDI 2.0 Ready product: Roland A-88MKII https://www.midi.org/articles-old/roland-announces Issue URL : https://github.com/alsa-project/alsa-lib/issues/24 Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] [PATCH v3 5/6] ASoC: qcom: apq8096: add support for primary and quaternary I2S/PCM
by Adam Serbinski 12 Feb '20

12 Feb '20

This adds support to primary and quarternary I2S and PCM ports. Signed-off-by: Adam Serbinski <adam(a)serbinski.com> CC: Andy Gross <agross(a)kernel.org> CC: Mark Rutland <mark.rutland(a)arm.com> CC: Liam Girdwood <lgirdwood(a)gmail.com> CC: Patrick Lai <plai(a)codeaurora.org> CC: Banajit Goswami <bgoswami(a)codeaurora.org> CC: Jaroslav Kysela <perex(a)perex.cz> CC: Takashi Iwai <tiwai(a)suse.com> CC: alsa-devel(a)alsa-project.org CC: linux-arm-msm(a)vger.kernel.org CC: devicetree(a)vger.kernel.org CC: linux-kernel(a)vger.kernel.org --- sound/soc/qcom/apq8096.c | 86 +++++++++++++++++++++++++++++++++------- 1 file changed, 71 insertions(+), 15 deletions(-) diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 94363fd6846a..0dd58cdf34dd 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -8,24 +8,13 @@ #include <sound/soc-dapm.h> #include <sound/pcm.h> #include "common.h" +#include "qdsp6/q6afe.h" #define SLIM_MAX_TX_PORTS 16 #define SLIM_MAX_RX_PORTS 16 #define WCD9335_DEFAULT_MCLK_RATE 9600000 - -static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - - rate->min = rate->max = 48000; - channels->min = channels->max = 2; - - return 0; -} +#define MI2S_BCLK_RATE 1536000 +#define PCM_BCLK_RATE 1024000 static int msm_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -33,10 +22,32 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS]; u32 rx_ch_cnt = 0, tx_ch_cnt = 0; int ret = 0; + switch (cpu_dai->id) { + case PRIMARY_PCM_RX: + case PRIMARY_PCM_TX: + case QUATERNARY_PCM_RX: + case QUATERNARY_PCM_TX: + rate->min = 16000; + rate->max = 16000; + channels->min = 1; + channels->max = 1; + break; + default: + rate->min = 48000; + rate->max = 48000; + channels->min = 2; + channels->max = 2; + break; + } + ret = snd_soc_dai_get_channel_map(codec_dai, &tx_ch_cnt, tx_ch, &rx_ch_cnt, rx_ch); if (ret != 0 && ret != -ENOTSUPP) { @@ -60,8 +71,54 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream, return ret; } +static int msm_snd_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + switch (cpu_dai->id) { + case PRIMARY_MI2S_RX: + case PRIMARY_MI2S_TX: + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT, + MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS); + snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS); + break; + case QUATERNARY_MI2S_RX: + case QUATERNARY_MI2S_TX: + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_QUAD_MI2S_IBIT, + MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS); + snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS); + break; + case PRIMARY_PCM_RX: + case PRIMARY_PCM_TX: + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_PRI_PCM_IBIT, + PCM_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS); + snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS); + break; + case QUATERNARY_PCM_RX: + case QUATERNARY_PCM_TX: + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_QUAD_PCM_IBIT, + PCM_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS); + snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBS_CFS); + break; + default: + return -1; + } + return 0; +} + static struct snd_soc_ops apq8096_ops = { .hw_params = msm_snd_hw_params, + .startup = msm_snd_startup, }; static int apq8096_init(struct snd_soc_pcm_runtime *rtd) @@ -96,7 +153,6 @@ static void apq8096_add_be_ops(struct snd_soc_card *card) for_each_card_prelinks(card, i, link) { if (link->no_pcm == 1) { - link->be_hw_params_fixup = apq8096_be_hw_params_fixup; link->init = apq8096_init; link->ops = &apq8096_ops; } -- 2.21.1

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[alsa-devel] [PATCH v3 4/6] ASoC: qdsp6: q6routing: add pcm port routing
by Adam Serbinski 12 Feb '20

12 Feb '20

This patch adds support to PCM_PORT mixers required to select path between ASM stream and AFE ports. Signed-off-by: Adam Serbinski <adam(a)serbinski.com> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> CC: Andy Gross <agross(a)kernel.org> CC: Mark Rutland <mark.rutland(a)arm.com> CC: Liam Girdwood <lgirdwood(a)gmail.com> CC: Patrick Lai <plai(a)codeaurora.org> CC: Banajit Goswami <bgoswami(a)codeaurora.org> CC: Jaroslav Kysela <perex(a)perex.cz> CC: Takashi Iwai <tiwai(a)suse.com> CC: alsa-devel(a)alsa-project.org CC: linux-arm-msm(a)vger.kernel.org CC: devicetree(a)vger.kernel.org CC: linux-kernel(a)vger.kernel.org --- sound/soc/qcom/qdsp6/q6routing.c | 44 ++++++++++++++++++++++++++++++++ 1 file changed, 44 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 20724102e85a..3a81d2161707 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -67,6 +67,10 @@ { mix_name, "SEC_MI2S_TX", "SEC_MI2S_TX" }, \ { mix_name, "QUAT_MI2S_TX", "QUAT_MI2S_TX" }, \ { mix_name, "TERT_MI2S_TX", "TERT_MI2S_TX" }, \ + { mix_name, "PRI_PCM_TX", "PRI_PCM_TX" }, \ + { mix_name, "SEC_PCM_TX", "SEC_PCM_TX" }, \ + { mix_name, "TERT_PCM_TX", "TERT_PCM_TX" }, \ + { mix_name, "QUAT_PCM_TX", "QUAT_PCM_TX" }, \ { mix_name, "SLIMBUS_0_TX", "SLIMBUS_0_TX" }, \ { mix_name, "SLIMBUS_1_TX", "SLIMBUS_1_TX" }, \ { mix_name, "SLIMBUS_2_TX", "SLIMBUS_2_TX" }, \ @@ -128,6 +132,18 @@ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX, \ id, 1, 0, msm_routing_get_audio_mixer, \ msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("PRI_PCM_TX", PRIMARY_PCM_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("SEC_PCM_TX", SECONDARY_PCM_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("TERT_PCM_TX", TERTIARY_PCM_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ + SOC_SINGLE_EXT("QUAT_PCM_TX", QUATERNARY_PCM_TX, \ + id, 1, 0, msm_routing_get_audio_mixer, \ + msm_routing_put_audio_mixer), \ SOC_SINGLE_EXT("SLIMBUS_0_TX", SLIMBUS_0_TX, \ id, 1, 0, msm_routing_get_audio_mixer, \ msm_routing_put_audio_mixer), \ @@ -468,6 +484,18 @@ static const struct snd_kcontrol_new quaternary_mi2s_rx_mixer_controls[] = { static const struct snd_kcontrol_new tertiary_mi2s_rx_mixer_controls[] = { Q6ROUTING_RX_MIXERS(TERTIARY_MI2S_RX) }; +static const struct snd_kcontrol_new primary_pcm_rx_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(PRIMARY_PCM_RX) }; + +static const struct snd_kcontrol_new secondary_pcm_rx_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(SECONDARY_PCM_RX) }; + +static const struct snd_kcontrol_new tertiary_pcm_rx_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(TERTIARY_PCM_RX) }; + +static const struct snd_kcontrol_new quaternary_pcm_rx_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(QUATERNARY_PCM_RX) }; + static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = { Q6ROUTING_RX_MIXERS(SLIMBUS_0_RX) }; @@ -695,6 +723,18 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { SND_SOC_DAPM_MIXER("TERT_MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0, tertiary_mi2s_rx_mixer_controls, ARRAY_SIZE(tertiary_mi2s_rx_mixer_controls)), + SND_SOC_DAPM_MIXER("PRI_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + primary_pcm_rx_mixer_controls, + ARRAY_SIZE(primary_pcm_rx_mixer_controls)), + SND_SOC_DAPM_MIXER("SEC_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + secondary_pcm_rx_mixer_controls, + ARRAY_SIZE(secondary_pcm_rx_mixer_controls)), + SND_SOC_DAPM_MIXER("TERT_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + tertiary_pcm_rx_mixer_controls, + ARRAY_SIZE(tertiary_pcm_rx_mixer_controls)), + SND_SOC_DAPM_MIXER("QUAT_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + quaternary_pcm_rx_mixer_controls, + ARRAY_SIZE(quaternary_pcm_rx_mixer_controls)), SND_SOC_DAPM_MIXER("PRIMARY_TDM_RX_0 Audio Mixer", SND_SOC_NOPM, 0, 0, pri_tdm_rx_0_mixer_controls, ARRAY_SIZE(pri_tdm_rx_0_mixer_controls)), @@ -853,6 +893,10 @@ static const struct snd_soc_dapm_route intercon[] = { Q6ROUTING_RX_DAPM_ROUTE("TERT_MI2S_RX Audio Mixer", "TERT_MI2S_RX"), Q6ROUTING_RX_DAPM_ROUTE("SEC_MI2S_RX Audio Mixer", "SEC_MI2S_RX"), Q6ROUTING_RX_DAPM_ROUTE("PRI_MI2S_RX Audio Mixer", "PRI_MI2S_RX"), + Q6ROUTING_RX_DAPM_ROUTE("PRI_PCM_RX Audio Mixer", "PRI_PCM_RX"), + Q6ROUTING_RX_DAPM_ROUTE("SEC_PCM_RX Audio Mixer", "SEC_PCM_RX"), + Q6ROUTING_RX_DAPM_ROUTE("TERT_PCM_RX Audio Mixer", "TERT_PCM_RX"), + Q6ROUTING_RX_DAPM_ROUTE("QUAT_PCM_RX Audio Mixer", "QUAT_PCM_RX"), Q6ROUTING_RX_DAPM_ROUTE("PRIMARY_TDM_RX_0 Audio Mixer", "PRIMARY_TDM_RX_0"), Q6ROUTING_RX_DAPM_ROUTE("PRIMARY_TDM_RX_1 Audio Mixer", -- 2.21.1

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