Merge tag 'sound-fix-5.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai...

Pull sound fixes from Takashi Iwai:
 "A few small remaining wrap-up for this merge window.

  Most of patches are device-specific (HD-audio and USB-audio quirks,
  FireWire, pcm316a, fsl, rsnd, Atmel, and TI fixes), while there is a
  simple fix (actually two commits) for ASoC core"

* tag 'sound-fix-5.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: usb-audio: Add DSD support for EVGA NU Audio
  ALSA: hda - Add laptop imic fixup for ASUS M9V laptop
  ASoC: ti: fix SND_SOC_DM365_VOICE_CODEC dependencies
  ASoC: pcm3168a: The codec does not support S32_LE
  ASoC: core: use list_del_init and move it back to soc_cleanup_component
  ALSA: hda/realtek - PCI quirk for Medion E4254
  ALSA: hda - Apply AMD controller workaround for Raven platform
  ASoC: rsnd: do error check after rsnd_channel_normalization()
  ASoC: atmel_ssc_dai: Remove wrong spinlock usage
  ASoC: core: delete component->card_list in soc_remove_component only
  ASoC: fsl_sai: Fix noise when using EDMA
  ALSA: usb-audio: Add Hiby device family to quirks for native DSD support
  ALSA: hda/realtek - Fix alienware headset mic
  ALSA: dice: fix wrong packet parameter for Alesis iO26

diff --git a/sound/firewire/dice/dice-alesis.c b/sound/firewire/dice/dice-alesis.c
index 218292bdace6fd63bbc5e94060cd20d69146cdca..f5b325263b6747478a0e78783a8ce1c480e515e3 100644 (file)
--- a/sound/firewire/dice/dice-alesis.c
+++ b/sound/firewire/dice/dice-alesis.c
@@ -15,7 +15,7 @@ alesis_io14_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = {
 
 static const unsigned int
 alesis_io26_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = {
-       {10, 10, 8},    /* Tx0 = Analog + S/PDIF. */
+       {10, 10, 4},    /* Tx0 = Analog + S/PDIF. */
        {16, 8, 0},     /* Tx1 = ADAT1 + ADAT2. */
 };
 

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 91e71be42fa45caeb82b3d8f0ffa239d0a3eda3a..240f4ca76391fbefa80074a182dc9ab88f4394e9 100644 (file)
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2485,8 +2485,7 @@ static const struct pci_device_id azx_ids[] = {
                         AZX_DCAPS_PM_RUNTIME },
        /* AMD Raven */
        { PCI_DEVICE(0x1022, 0x15e3),
-         .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
-                        AZX_DCAPS_PM_RUNTIME },
+         .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB },
        /* ATI HDMI */
        { PCI_DEVICE(0x1002, 0x0002),
          .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index e283966bdbb199d8e88d1f9923cbec8f16ef8c37..bc9dd8e6fd86900659cf10852eeefa5ed0075dcc 100644 (file)
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -357,6 +357,7 @@ static const struct hda_fixup ad1986a_fixups[] = {
 
 static const struct snd_pci_quirk ad1986a_fixup_tbl[] = {
        SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC),
+       SND_PCI_QUIRK(0x1043, 0x1153, "ASUS M9V", AD1986A_FIXUP_LAPTOP_IMIC),
        SND_PCI_QUIRK(0x1043, 0x1443, "ASUS Z99He", AD1986A_FIXUP_EAPD),
        SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8JN", AD1986A_FIXUP_EAPD),
        SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK),

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index da1695418731568fdc557882545fa8c72c615dae..b000b36ac3c627307381d7f7443e29133c077a81 100644 (file)
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5817,6 +5817,7 @@ enum {
        ALC292_FIXUP_DELL_E7X,
        ALC292_FIXUP_DISABLE_AAMIX,
        ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK,
+       ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE,
        ALC298_FIXUP_DELL1_MIC_NO_PRESENCE,
        ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE,
        ALC275_FIXUP_DELL_XPS,
@@ -5871,6 +5872,7 @@ enum {
        ALC256_FIXUP_ASUS_MIC_NO_PRESENCE,
        ALC299_FIXUP_PREDATOR_SPK,
        ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC,
+       ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -6506,6 +6508,15 @@ static const struct hda_fixup alc269_fixups[] = {
                .chained = true,
                .chain_id = ALC292_FIXUP_DISABLE_AAMIX
        },
+       [ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE] = {
+               .type = HDA_FIXUP_PINS,
+               .v.pins = (const struct hda_pintbl[]) {
+                       { 0x18, 0x01a1913c }, /* headset mic w/o jack detect */
+                       { }
+               },
+               .chained_before = true,
+               .chain_id = ALC269_FIXUP_HEADSET_MODE,
+       },
        [ALC298_FIXUP_DELL1_MIC_NO_PRESENCE] = {
                .type = HDA_FIXUP_PINS,
                .v.pins = (const struct hda_pintbl[]) {
@@ -6927,6 +6938,16 @@ static const struct hda_fixup alc269_fixups[] = {
                .chained = true,
                .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
        },
+       [ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE] = {
+               .type = HDA_FIXUP_PINS,
+               .v.pins = (const struct hda_pintbl[]) {
+                       { 0x19, 0x04a11040 },
+                       { 0x21, 0x04211020 },
+                       { }
+               },
+               .chained = true,
+               .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE
+       },
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -7190,6 +7211,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
        SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
        SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS),
        SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
+       SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE),
 
 #if 0
        /* Below is a quirk table taken from the old code.
@@ -7358,6 +7380,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
        {.id = ALC295_FIXUP_CHROME_BOOK, .name = "alc-chrome-book"},
        {.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"},
        {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"},
+       {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"},
        {}
 };
 #define ALC225_STANDARD_PINS \
@@ -7770,6 +7793,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
                {0x17, 0x90170110},
                {0x1a, 0x03011020},
                {0x21, 0x03211030}),
+       SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE,
+               {0x12, 0xb7a60140},
+               {0x17, 0x90170110},
+               {0x1a, 0x03a11030},
+               {0x21, 0x03211020}),
        SND_HDA_PIN_QUIRK(0x10ec0299, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE,
                ALC225_STANDARD_PINS,
                {0x12, 0xb7a60130},

diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 48e9eef34c0fead794794ed987597d2bb37ba83a..ca603397651c0f19376fc4543269b6d38e233e76 100644 (file)
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -116,19 +116,16 @@ static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = {
 static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = {
        {
        .name           = "ssc0",
-       .lock           = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock),
        .dir_mask       = SSC_DIR_MASK_UNUSED,
        .initialized    = 0,
        },
        {
        .name           = "ssc1",
-       .lock           = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock),
        .dir_mask       = SSC_DIR_MASK_UNUSED,
        .initialized    = 0,
        },
        {
        .name           = "ssc2",
-       .lock           = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock),
        .dir_mask       = SSC_DIR_MASK_UNUSED,
        .initialized    = 0,
        },
@@ -317,13 +314,10 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream,
 
        snd_soc_dai_set_dma_data(dai, substream, dma_params);
 
-       spin_lock_irq(&ssc_p->lock);
-       if (ssc_p->dir_mask & dir_mask) {
-               spin_unlock_irq(&ssc_p->lock);
+       if (ssc_p->dir_mask & dir_mask)
                return -EBUSY;
-       }
+
        ssc_p->dir_mask |= dir_mask;
-       spin_unlock_irq(&ssc_p->lock);
 
        return 0;
 }
@@ -355,7 +349,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream,
 
        dir_mask = 1 << dir;
 
-       spin_lock_irq(&ssc_p->lock);
        ssc_p->dir_mask &= ~dir_mask;
        if (!ssc_p->dir_mask) {
                if (ssc_p->initialized) {
@@ -369,7 +362,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream,
                ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0;
                ssc_p->forced_divider = 0;
        }
-       spin_unlock_irq(&ssc_p->lock);
 
        /* Shutdown the SSC clock. */
        pr_debug("atmel_ssc_dai: Stopping clock\n");

diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h
index ae764cb541c78cba6c1e87cec190a54f5c5c96cd..3470b966e44971a1126e4356ea544356f7a674c4 100644 (file)
--- a/sound/soc/atmel/atmel_ssc_dai.h
+++ b/sound/soc/atmel/atmel_ssc_dai.h
@@ -93,7 +93,6 @@ struct atmel_ssc_state {
 struct atmel_ssc_info {
        char *name;
        struct ssc_device *ssc;
-       spinlock_t lock;        /* lock for dir_mask */
        unsigned short dir_mask;        /* 0=unused, 1=playback, 2=capture */
        unsigned short initialized;     /* true if SSC has been initialized */
        unsigned short daifmt;

diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c
index 50ed86d45c26fd390feb5f11ec2044288cfeb7ad..88b75695fbf7f582a46d837877d9522a385b61c4 100644 (file)
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -21,8 +21,7 @@
 
 #define PCM3168A_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
                         SNDRV_PCM_FMTBIT_S24_3LE | \
-                        SNDRV_PCM_FMTBIT_S24_LE | \
-                        SNDRV_PCM_FMTBIT_S32_LE)
+                        SNDRV_PCM_FMTBIT_S24_LE)
 
 #define PCM3168A_FMT_I2S               0x0
 #define PCM3168A_FMT_LEFT_J            0x1

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index ef0b74693093ba2f5137084f2c6b1ec98e06a6cd..b517e4bc1b8737f8821b51d33701a49242689c45 100644 (file)
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -628,6 +628,16 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream,
                           FSL_SAI_CR3_TRCE_MASK,
                           FSL_SAI_CR3_TRCE);
 
+       /*
+        * EDMA controller needs period size to be a multiple of
+        * tx/rx maxburst
+        */
+       if (sai->soc_data->use_edma)
+               snd_pcm_hw_constraint_step(substream->runtime, 0,
+                                          SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+                                          tx ? sai->dma_params_tx.maxburst :
+                                          sai->dma_params_rx.maxburst);
+
        ret = snd_pcm_hw_constraint_list(substream->runtime, 0,
                        SNDRV_PCM_HW_PARAM_RATE, &fsl_sai_rate_constraints);
 
@@ -1026,30 +1036,35 @@ static int fsl_sai_remove(struct platform_device *pdev)
 
 static const struct fsl_sai_soc_data fsl_sai_vf610_data = {
        .use_imx_pcm = false,
+       .use_edma = false,
        .fifo_depth = 32,
        .reg_offset = 0,
 };
 
 static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = {
        .use_imx_pcm = true,
+       .use_edma = false,
        .fifo_depth = 32,
        .reg_offset = 0,
 };
 
 static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = {
        .use_imx_pcm = true,
+       .use_edma = false,
        .fifo_depth = 16,
        .reg_offset = 8,
 };
 
 static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = {
        .use_imx_pcm = true,
+       .use_edma = false,
        .fifo_depth = 128,
        .reg_offset = 8,
 };
 
 static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = {
        .use_imx_pcm = true,
+       .use_edma = true,
        .fifo_depth = 64,
        .reg_offset = 0,
 };

diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index b12cb578f6d07042bea0e42efe093353674fd418..76b15deea80c7afeec9a37276c37ba54033a8686 100644 (file)
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -157,6 +157,7 @@
 
 struct fsl_sai_soc_data {
        bool use_imx_pcm;
+       bool use_edma;
        unsigned int fifo_depth;
        unsigned int reg_offset;
 };

diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index f6a7466622ea6b610e9c12d4a40658ed80cbd093..fc5d089868dfc084cfa4e3234259a57402cd8d5e 100644 (file)
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -286,6 +286,11 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
        if (rsnd_ssi_is_multi_slave(mod, io))
                return 0;
 
+       if (rsnd_runtime_is_tdm_split(io))
+               chan = rsnd_io_converted_chan(io);
+
+       chan = rsnd_channel_normalization(chan);
+
        if (ssi->usrcnt > 0) {
                if (ssi->rate != rate) {
                        dev_err(dev, "SSI parent/child should use same rate\n");
@@ -300,11 +305,6 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
                return 0;
        }
 
-       if (rsnd_runtime_is_tdm_split(io))
-               chan = rsnd_io_converted_chan(io);
-
-       chan = rsnd_channel_normalization(chan);
-
        main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx);
        if (!main_rate) {
                dev_err(dev, "unsupported clock rate\n");

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 35f48e9c5ead69a8075ed1133b13167461b08013..88978a3036c4e6754b405cb1ccb5f987bca45deb 100644 (file)
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -978,7 +978,7 @@ static void soc_cleanup_component(struct snd_soc_component *component)
        /* For framework level robustness */
        snd_soc_component_set_jack(component, NULL, NULL);
 
-       list_del(&component->card_list);
+       list_del_init(&component->card_list);
        snd_soc_dapm_free(snd_soc_component_get_dapm(component));
        soc_cleanup_component_debugfs(component);
        component->card = NULL;

diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig
index 87a9b9dd4e98f8a4b4373cc4452b911d88f813d2..29f61053ab62e357d5f5cbc8a873728fc6c8dac3 100644 (file)
--- a/sound/soc/ti/Kconfig
+++ b/sound/soc/ti/Kconfig
@@ -200,11 +200,18 @@ config SND_SOC_DM365_AIC3X_CODEC
 
 config SND_SOC_DM365_VOICE_CODEC
        bool "Voice Codec - CQ93VC"
-       select MFD_DAVINCI_VOICECODEC
-       select SND_SOC_CQ0093VC
        help
          Say Y if you want to add support for SoC On-chip voice codec
 endchoice
 
+config SND_SOC_DM365_VOICE_CODEC_MODULE
+       def_tristate y
+       depends on SND_SOC_DM365_VOICE_CODEC && SND_SOC
+       select MFD_DAVINCI_VOICECODEC
+       select SND_SOC_CQ0093VC
+       help
+         The is an internal symbol needed to ensure that the codec
+         and MFD driver can be built as loadable modules if necessary.
+
 endmenu
 

diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 25faf2d3c639d9984ee16e82f5fe76e9bdfe3c56..fbfde996fee76df28b56439c0030fb1cb0de7da3 100644 (file)
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1658,6 +1658,8 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
        case 0x25ce:  /* Mytek devices */
        case 0x278b:  /* Rotel? */
        case 0x2ab6:  /* T+A devices */
+       case 0x3842:  /* EVGA */
+       case 0xc502:  /* HiBy devices */
                if (fp->dsd_raw)
                        return SNDRV_PCM_FMTBIT_DSD_U32_BE;
                break;