diff options
| author | Zhiqiang Tu <ztu@codeaurora.org> | 2017-08-24 15:30:08 +0800 |
|---|---|---|
| committer | Zhiqiang Tu <ztu@codeaurora.org> | 2017-08-24 16:39:56 +0800 |
| commit | 9df1d44946b2278f14ee1ba90d3f3f281791e6bc (patch) | |
| tree | d510fb0532267008691b65cca07ebc6324c2222e /sound | |
| parent | 1ebddfae1dd64f54b4da86a81eab1616ef8f2d83 (diff) | |
| parent | cb1c821645fd08ff03c7ed4d62a6e4ca0a6a6ad5 (diff) | |
Merge branch 'msm-4.4' into dev/msm-4.4-8996au
Conflicts:
drivers/iommu/arm-smmu.c
drivers/media/platform/msm/ais/fd/msm_fd_dev.c
drivers/media/platform/msm/camera_v2/fd/msm_fd_dev.c
drivers/soc/qcom/glink.c
include/uapi/linux/msm_ipa.h
Change-Id: Id007a850fa2df09f08c413ffcd447a6532fad83c
Signed-off-by: Zhiqiang Tu <ztu@codeaurora.org>
Diffstat (limited to 'sound')
43 files changed, 3192 insertions, 350 deletions
diff --git a/sound/core/timer.c b/sound/core/timer.c index 30b28e80c6e6..f0675acecc93 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1625,6 +1625,7 @@ static int snd_timer_user_tselect(struct file *file, if (err < 0) goto __err; + tu->qhead = tu->qtail = tu->qused = 0; kfree(tu->queue); tu->queue = NULL; kfree(tu->tqueue); @@ -1962,6 +1963,7 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, tu = file->private_data; unit = tu->tread ? sizeof(struct snd_timer_tread) : sizeof(struct snd_timer_read); + mutex_lock(&tu->ioctl_lock); spin_lock_irq(&tu->qlock); while ((long)count - result >= unit) { while (!tu->qused) { @@ -1977,7 +1979,9 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, add_wait_queue(&tu->qchange_sleep, &wait); spin_unlock_irq(&tu->qlock); + mutex_unlock(&tu->ioctl_lock); schedule(); + mutex_lock(&tu->ioctl_lock); spin_lock_irq(&tu->qlock); remove_wait_queue(&tu->qchange_sleep, &wait); @@ -1997,7 +2001,6 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, tu->qused--; spin_unlock_irq(&tu->qlock); - mutex_lock(&tu->ioctl_lock); if (tu->tread) { if (copy_to_user(buffer, &tu->tqueue[qhead], sizeof(struct snd_timer_tread))) @@ -2007,7 +2010,6 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, sizeof(struct snd_timer_read))) err = -EFAULT; } - mutex_unlock(&tu->ioctl_lock); spin_lock_irq(&tu->qlock); if (err < 0) @@ -2017,6 +2019,7 @@ static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, } _error: spin_unlock_irq(&tu->qlock); + mutex_unlock(&tu->ioctl_lock); return result > 0 ? result : err; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 373fcad840ea..776dffa88aee 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -294,6 +294,8 @@ struct hda_codec { #define list_for_each_codec(c, bus) \ list_for_each_entry(c, &(bus)->core.codec_list, core.list) +#define list_for_each_codec_safe(c, n, bus) \ + list_for_each_entry_safe(c, n, &(bus)->core.codec_list, core.list) /* snd_hda_codec_read/write optional flags */ #define HDA_RW_NO_RESPONSE_FALLBACK (1 << 0) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 5baf8b56b6e7..9c6e10fb479f 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1128,8 +1128,12 @@ EXPORT_SYMBOL_GPL(azx_probe_codecs); /* configure each codec instance */ int azx_codec_configure(struct azx *chip) { - struct hda_codec *codec; - list_for_each_codec(codec, &chip->bus) { + struct hda_codec *codec, *next; + + /* use _safe version here since snd_hda_codec_configure() deregisters + * the device upon error and deletes itself from the bus list. + */ + list_for_each_codec_safe(codec, next, &chip->bus) { snd_hda_codec_configure(codec); } return 0; diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index dc2fa576d60d..689df78f640a 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3190,6 +3190,7 @@ static int check_dyn_adc_switch(struct hda_codec *codec) spec->input_paths[i][nums]); spec->input_paths[i][nums] = spec->input_paths[i][n]; + spec->input_paths[i][n] = 0; } } nums++; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 37b70f8e878f..0abab7926dca 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1537,6 +1537,8 @@ static const struct snd_pci_quirk stac9200_fixup_tbl[] = { "Dell Inspiron 1501", STAC_9200_DELL_M26), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f6, "unknown Dell", STAC_9200_DELL_M26), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0201, + "Dell Latitude D430", STAC_9200_DELL_M22), /* Panasonic */ SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-74", STAC_9200_PANASONIC), /* Gateway machines needs EAPD to be set on resume */ diff --git a/sound/soc/codecs/msm_sdw/msm_sdw_cdc.c b/sound/soc/codecs/msm_sdw/msm_sdw_cdc.c index b91d13c8e010..58df020811f1 100644 --- a/sound/soc/codecs/msm_sdw/msm_sdw_cdc.c +++ b/sound/soc/codecs/msm_sdw/msm_sdw_cdc.c @@ -1039,7 +1039,6 @@ static int msm_sdw_swrm_read(void *handle, int reg) __func__, reg); sdw_rd_addr_base = MSM_SDW_AHB_BRIDGE_RD_ADDR_0; sdw_rd_data_base = MSM_SDW_AHB_BRIDGE_RD_DATA_0; - /* * Add sleep as SWR slave access read takes time. * Allow for RD_DONE to complete for previous register if any. @@ -1054,6 +1053,8 @@ static int msm_sdw_swrm_read(void *handle, int reg) dev_err(msm_sdw->dev, "%s: RD Addr Failure\n", __func__); goto err; } + /* Add sleep for SWR register read value to get updated. */ + usleep_range(100, 105); /* Check for RD value */ ret = regmap_bulk_read(msm_sdw->regmap, sdw_rd_data_base, (u8 *)&val, 4); @@ -1079,12 +1080,12 @@ static int msm_sdw_bulk_write(struct msm_sdw_priv *msm_sdw, sdw_wr_addr_base = MSM_SDW_AHB_BRIDGE_WR_ADDR_0; sdw_wr_data_base = MSM_SDW_AHB_BRIDGE_WR_DATA_0; - /* - * Add sleep as SWR slave write takes time. - * Allow for any previous pending write to complete. - */ - usleep_range(50, 55); for (i = 0; i < len; i += 2) { + /* + * Add sleep as SWR slave write takes time. + * Allow for any previous pending write to complete. + */ + usleep_range(100, 105); /* First Write the Data to register */ ret = regmap_bulk_write(msm_sdw->regmap, sdw_wr_data_base, bulk_reg[i].buf, 4); diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index c1b87c5800b1..b3fddba4c084 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -936,7 +936,8 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, NAU8825_FLL_INTEGER_MASK, fll_param->fll_int); /* FLL pre-scaler */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL4, - NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div); + NAU8825_FLL_REF_DIV_MASK, + fll_param->clk_ref_div << NAU8825_FLL_REF_DIV_SFT); /* select divided VCO input */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, NAU8825_FLL_FILTER_SW_MASK, 0x0000); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index dff8edb83bfd..a0b220726a63 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -114,7 +114,8 @@ #define NAU8825_FLL_INTEGER_MASK (0x3ff << 0) /* FLL4 (0x07) */ -#define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) +#define NAU8825_FLL_REF_DIV_SFT 10 +#define NAU8825_FLL_REF_DIV_MASK (0x3 << NAU8825_FLL_REF_DIV_SFT) /* FLL5 (0x08) */ #define NAU8825_FLL_FILTER_SW_MASK (0x1 << 14) diff --git a/sound/soc/codecs/sdm660_cdc/msm-analog-cdc.c b/sound/soc/codecs/sdm660_cdc/msm-analog-cdc.c index 00f2aa766363..025a592b4015 100644 --- a/sound/soc/codecs/sdm660_cdc/msm-analog-cdc.c +++ b/sound/soc/codecs/sdm660_cdc/msm-analog-cdc.c @@ -1400,8 +1400,26 @@ static int msm_anlg_cdc_codec_enable_on_demand_supply( } switch (event) { case SND_SOC_DAPM_PRE_PMU: - if (atomic_inc_return(&supply->ref) == 1) + if (atomic_inc_return(&supply->ref) == 1) { + ret = regulator_set_voltage(supply->supply, + supply->min_uv, + supply->max_uv); + if (ret) { + dev_err(codec->dev, + "Setting regulator voltage(en) for micbias with err = %d\n", + ret); + goto out; + } + ret = regulator_set_load(supply->supply, + supply->optimum_ua); + if (ret < 0) { + dev_err(codec->dev, + "Setting regulator optimum mode(en) failed for micbias with err = %d\n", + ret); + goto out; + } ret = regulator_enable(supply->supply); + } if (ret) dev_err(codec->dev, "%s: Failed to enable %s\n", __func__, @@ -1413,12 +1431,27 @@ static int msm_anlg_cdc_codec_enable_on_demand_supply( __func__, on_demand_supply_name[w->shift]); goto out; } - if (atomic_dec_return(&supply->ref) == 0) + if (atomic_dec_return(&supply->ref) == 0) { ret = regulator_disable(supply->supply); if (ret) dev_err(codec->dev, "%s: Failed to disable %s\n", __func__, on_demand_supply_name[w->shift]); + ret = regulator_set_voltage(supply->supply, + 0, + supply->max_uv); + if (ret) { + dev_err(codec->dev, + "Setting regulator voltage(dis) failed for micbias with err = %d\n", + ret); + goto out; + } + ret = regulator_set_load(supply->supply, 0); + if (ret < 0) + dev_err(codec->dev, + "Setting regulator optimum mode(dis) failed for micbias with err = %d\n", + ret); + } break; default: break; @@ -2022,6 +2055,9 @@ static const char * const rdac2_mux_text[] = { "ZERO", "RX2", "RX1" }; +static const struct snd_kcontrol_new adc1_switch = + SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 1, 0); + static const struct soc_enum rdac2_mux_enum = SOC_ENUM_SINGLE(MSM89XX_PMIC_DIGITAL_CDC_CONN_HPHR_DAC_CTL, 0, 3, rdac2_mux_text); @@ -3072,7 +3108,8 @@ static const struct snd_soc_dapm_route audio_map[] = { {"ADC2 MUX", "INP2", "ADC2_INP2"}, {"ADC2 MUX", "INP3", "ADC2_INP3"}, - {"ADC1", NULL, "AMIC1"}, + {"ADC1", NULL, "ADC1_INP1"}, + {"ADC1_INP1", "Switch", "AMIC1"}, {"ADC2_INP2", NULL, "AMIC2"}, {"ADC2_INP3", NULL, "AMIC3"}, @@ -3413,6 +3450,8 @@ static const struct snd_soc_dapm_widget msm_anlg_cdc_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", msm_anlg_cdc_codec_enable_spk_ext_pa), + SND_SOC_DAPM_SWITCH("ADC1_INP1", SND_SOC_NOPM, 0, 0, + &adc1_switch), SND_SOC_DAPM_SUPPLY("RX1 CLK", MSM89XX_PMIC_DIGITAL_CDC_DIG_CLK_CTL, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("RX2 CLK", MSM89XX_PMIC_DIGITAL_CDC_DIG_CLK_CTL, @@ -3685,6 +3724,30 @@ static struct regulator *msm_anlg_cdc_find_regulator( return NULL; } +static void msm_anlg_cdc_update_micbias_regulator( + const struct sdm660_cdc_priv *sdm660_cdc, + const char *name, + struct on_demand_supply *micbias_supply) +{ + int i; + struct sdm660_cdc_pdata *pdata = sdm660_cdc->dev->platform_data; + + for (i = 0; i < sdm660_cdc->num_of_supplies; i++) { + if (sdm660_cdc->supplies[i].supply && + !strcmp(sdm660_cdc->supplies[i].supply, name)) { + micbias_supply->supply = + sdm660_cdc->supplies[i].consumer; + micbias_supply->min_uv = pdata->regulator[i].min_uv; + micbias_supply->max_uv = pdata->regulator[i].max_uv; + micbias_supply->optimum_ua = + pdata->regulator[i].optimum_ua; + return; + } + } + + dev_err(sdm660_cdc->dev, "Error: regulator not found:%s\n", name); +} + static int msm_anlg_cdc_device_down(struct snd_soc_codec *codec) { struct msm_asoc_mach_data *pdata = NULL; @@ -4114,10 +4177,10 @@ static int msm_anlg_cdc_soc_probe(struct snd_soc_codec *codec) wcd9xxx_spmi_set_codec(codec); - sdm660_cdc->on_demand_list[ON_DEMAND_MICBIAS].supply = - msm_anlg_cdc_find_regulator( + msm_anlg_cdc_update_micbias_regulator( sdm660_cdc, - on_demand_supply_name[ON_DEMAND_MICBIAS]); + on_demand_supply_name[ON_DEMAND_MICBIAS], + &sdm660_cdc->on_demand_list[ON_DEMAND_MICBIAS]); atomic_set(&sdm660_cdc->on_demand_list[ON_DEMAND_MICBIAS].ref, 0); @@ -4183,7 +4246,7 @@ static int msm_anlg_cdc_enable_static_supplies_to_optimum( if (pdata->regulator[i].ondemand) continue; if (regulator_count_voltages( - sdm660_cdc->supplies[i].consumer) <= 0) + sdm660_cdc->supplies[i].consumer) <= 0) continue; ret = regulator_set_voltage( @@ -4216,7 +4279,7 @@ static int msm_anlg_cdc_disable_static_supplies_to_optimum( if (pdata->regulator[i].ondemand) continue; if (regulator_count_voltages( - sdm660_cdc->supplies[i].consumer) <= 0) + sdm660_cdc->supplies[i].consumer) <= 0) continue; regulator_set_voltage(sdm660_cdc->supplies[i].consumer, 0, pdata->regulator[i].max_uv); @@ -4317,6 +4380,28 @@ static int msm_anlg_cdc_init_supplies(struct sdm660_cdc_priv *sdm660_cdc, if (regulator_count_voltages( sdm660_cdc->supplies[i].consumer) <= 0) continue; + if (pdata->regulator[i].ondemand) { + ret = regulator_set_voltage( + sdm660_cdc->supplies[i].consumer, + 0, pdata->regulator[i].max_uv); + if (ret) { + dev_err(sdm660_cdc->dev, + "Setting regulator voltage failed for regulator %s err = %d\n", + sdm660_cdc->supplies[i].supply, ret); + goto err_supplies; + } + ret = regulator_set_load( + sdm660_cdc->supplies[i].consumer, 0); + if (ret < 0) { + dev_err(sdm660_cdc->dev, + "Setting regulator optimum mode failed for regulator %s err = %d\n", + sdm660_cdc->supplies[i].supply, ret); + goto err_supplies; + } else { + ret = 0; + continue; + } + } ret = regulator_set_voltage(sdm660_cdc->supplies[i].consumer, pdata->regulator[i].min_uv, pdata->regulator[i].max_uv); diff --git a/sound/soc/codecs/sdm660_cdc/msm-analog-cdc.h b/sound/soc/codecs/sdm660_cdc/msm-analog-cdc.h index 0c9e9a6aeb6a..9563565f36d2 100644 --- a/sound/soc/codecs/sdm660_cdc/msm-analog-cdc.h +++ b/sound/soc/codecs/sdm660_cdc/msm-analog-cdc.h @@ -144,6 +144,9 @@ struct sdm660_cdc_regulator { struct on_demand_supply { struct regulator *supply; atomic_t ref; + int min_uv; + int max_uv; + int optimum_ua; }; struct wcd_imped_i_ref { diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index a564759845f9..5a3f544bb3a8 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -126,6 +126,16 @@ static const struct reg_default aic3x_reg[] = { { 108, 0x00 }, { 109, 0x00 }, }; +static bool aic3x_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AIC3X_RESET: + return true; + default: + return false; + } +} + static const struct regmap_config aic3x_regmap = { .reg_bits = 8, .val_bits = 8, @@ -133,6 +143,9 @@ static const struct regmap_config aic3x_regmap = { .max_register = DAC_ICC_ADJ, .reg_defaults = aic3x_reg, .num_reg_defaults = ARRAY_SIZE(aic3x_reg), + + .volatile_reg = aic3x_volatile_reg, + .cache_type = REGCACHE_RBTREE, }; diff --git a/sound/soc/codecs/wcd-dsp-mgr.c b/sound/soc/codecs/wcd-dsp-mgr.c index 9b1c8c98946c..1613c5baa9c7 100644 --- a/sound/soc/codecs/wcd-dsp-mgr.c +++ b/sound/soc/codecs/wcd-dsp-mgr.c @@ -415,22 +415,24 @@ static int wdsp_download_segments(struct wdsp_mgr_priv *wdsp, /* Go through the list of segments and download one by one */ list_for_each_entry(seg, wdsp->seg_list, list) { ret = wdsp_load_each_segment(wdsp, seg); - if (IS_ERR_VALUE(ret)) { - wdsp_broadcast_event_downseq(wdsp, - WDSP_EVENT_DLOAD_FAILED, - NULL); + if (ret) goto dload_error; - } } + /* Flush the list before setting status and notifying components */ + wdsp_flush_segment_list(wdsp->seg_list); + WDSP_SET_STATUS(wdsp, status); /* Notify all components that image is downloaded */ wdsp_broadcast_event_downseq(wdsp, post, NULL); +done: + return ret; dload_error: wdsp_flush_segment_list(wdsp->seg_list); -done: + wdsp_broadcast_event_downseq(wdsp, WDSP_EVENT_DLOAD_FAILED, NULL); + return ret; } @@ -484,10 +486,14 @@ static int wdsp_enable_dsp(struct wdsp_mgr_priv *wdsp) /* Make sure wdsp is in good state */ if (!WDSP_STATUS_IS_SET(wdsp, WDSP_STATUS_CODE_DLOADED)) { WDSP_ERR(wdsp, "WDSP in invalid state 0x%x", wdsp->status); - ret = -EINVAL; - goto done; + return -EINVAL; } + /* + * Acquire SSR mutex lock to make sure enablement of DSP + * does not race with SSR handling. + */ + WDSP_MGR_MUTEX_LOCK(wdsp, wdsp->ssr_mutex); /* Download the read-write sections of image */ ret = wdsp_download_segments(wdsp, WDSP_ELF_FLAG_WRITE); if (IS_ERR_VALUE(ret)) { @@ -508,6 +514,7 @@ static int wdsp_enable_dsp(struct wdsp_mgr_priv *wdsp) wdsp_broadcast_event_downseq(wdsp, WDSP_EVENT_POST_BOOTUP, NULL); WDSP_SET_STATUS(wdsp, WDSP_STATUS_BOOTED); done: + WDSP_MGR_MUTEX_UNLOCK(wdsp, wdsp->ssr_mutex); return ret; } diff --git a/sound/soc/codecs/wcd-mbhc-v2.c b/sound/soc/codecs/wcd-mbhc-v2.c index 44141ae5668e..997a623033fb 100644 --- a/sound/soc/codecs/wcd-mbhc-v2.c +++ b/sound/soc/codecs/wcd-mbhc-v2.c @@ -361,6 +361,7 @@ out_micb_en: /* Disable micbias, pullup & enable cs */ wcd_enable_curr_micbias(mbhc, WCD_MBHC_EN_CS); mutex_unlock(&mbhc->hphl_pa_lock); + clear_bit(WCD_MBHC_ANC0_OFF_ACK, &mbhc->hph_anc_state); break; case WCD_EVENT_PRE_HPHR_PA_OFF: mutex_lock(&mbhc->hphr_pa_lock); @@ -378,6 +379,7 @@ out_micb_en: /* Disable micbias, pullup & enable cs */ wcd_enable_curr_micbias(mbhc, WCD_MBHC_EN_CS); mutex_unlock(&mbhc->hphr_pa_lock); + clear_bit(WCD_MBHC_ANC1_OFF_ACK, &mbhc->hph_anc_state); break; case WCD_EVENT_PRE_HPHL_PA_ON: set_bit(WCD_MBHC_EVENT_PA_HPHL, &mbhc->event_state); @@ -495,6 +497,25 @@ static void wcd_mbhc_clr_and_turnon_hph_padac(struct wcd_mbhc *mbhc) __func__); usleep_range(wg_time * 1000, wg_time * 1000 + 50); } + + if (test_and_clear_bit(WCD_MBHC_ANC0_OFF_ACK, + &mbhc->hph_anc_state)) { + usleep_range(20000, 20100); + pr_debug("%s: HPHL ANC clear flag and enable ANC_EN\n", + __func__); + if (mbhc->mbhc_cb->update_anc_state) + mbhc->mbhc_cb->update_anc_state(mbhc->codec, true, 0); + } + + if (test_and_clear_bit(WCD_MBHC_ANC1_OFF_ACK, + &mbhc->hph_anc_state)) { + usleep_range(20000, 20100); + pr_debug("%s: HPHR ANC clear flag and enable ANC_EN\n", + __func__); + if (mbhc->mbhc_cb->update_anc_state) + mbhc->mbhc_cb->update_anc_state(mbhc->codec, true, 1); + } + } static bool wcd_mbhc_is_hph_pa_on(struct wcd_mbhc *mbhc) @@ -526,6 +547,20 @@ static void wcd_mbhc_set_and_turnoff_hph_padac(struct wcd_mbhc *mbhc) } WCD_MBHC_REG_UPDATE_BITS(WCD_MBHC_HPH_PA_EN, 0); usleep_range(wg_time * 1000, wg_time * 1000 + 50); + + + if (mbhc->mbhc_cb->is_anc_on && mbhc->mbhc_cb->is_anc_on(mbhc)) { + usleep_range(20000, 20100); + pr_debug("%s ANC is on, setting ANC_OFF_ACK\n", __func__); + set_bit(WCD_MBHC_ANC0_OFF_ACK, &mbhc->hph_anc_state); + set_bit(WCD_MBHC_ANC1_OFF_ACK, &mbhc->hph_anc_state); + if (mbhc->mbhc_cb->update_anc_state) { + mbhc->mbhc_cb->update_anc_state(mbhc->codec, false, 0); + mbhc->mbhc_cb->update_anc_state(mbhc->codec, false, 1); + } else { + pr_debug("%s ANC is off\n", __func__); + } + } } int wcd_mbhc_get_impedance(struct wcd_mbhc *mbhc, uint32_t *zl, @@ -633,7 +668,8 @@ static void wcd_mbhc_report_plug(struct wcd_mbhc *mbhc, int insertion, jack_type == SND_JACK_LINEOUT) && (mbhc->hph_status && mbhc->hph_status != jack_type)) { - if (mbhc->micbias_enable) { + if (mbhc->micbias_enable && + mbhc->hph_status == SND_JACK_HEADSET) { if (mbhc->mbhc_cb->mbhc_micbias_control) mbhc->mbhc_cb->mbhc_micbias_control( codec, MIC_BIAS_2, diff --git a/sound/soc/codecs/wcd-mbhc-v2.h b/sound/soc/codecs/wcd-mbhc-v2.h index 32d9801468f9..09b9307ad61d 100644 --- a/sound/soc/codecs/wcd-mbhc-v2.h +++ b/sound/soc/codecs/wcd-mbhc-v2.h @@ -89,6 +89,11 @@ enum pa_dac_ack_flags { WCD_MBHC_HPHR_PA_OFF_ACK, }; +enum anc_ack_flags { + WCD_MBHC_ANC0_OFF_ACK = 0, + WCD_MBHC_ANC1_OFF_ACK, +}; + enum wcd_mbhc_btn_det_mem { WCD_MBHC_BTN_DET_V_BTN_LOW, WCD_MBHC_BTN_DET_V_BTN_HIGH @@ -386,6 +391,9 @@ struct wcd_mbhc_cb { void (*hph_pull_down_ctrl)(struct snd_soc_codec *, bool); void (*mbhc_moisture_config)(struct wcd_mbhc *); bool (*hph_register_recovery)(struct wcd_mbhc *); + void (*update_anc_state)(struct snd_soc_codec *codec, + bool enable, int anc_num); + bool (*is_anc_on)(struct wcd_mbhc *mbhc); }; struct wcd_mbhc { @@ -426,6 +434,7 @@ struct wcd_mbhc { /* track PA/DAC state to sync with userspace */ unsigned long hph_pa_dac_state; + unsigned long hph_anc_state; unsigned long event_state; unsigned long jiffies_atreport; diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 189165ce14c6..10883b0939d6 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -352,7 +352,6 @@ enum { AUDIO_NOMINAL, CPE_NOMINAL, HPH_PA_DELAY, - SB_CLK_GEAR, ANC_MIC_AMIC1, ANC_MIC_AMIC2, ANC_MIC_AMIC3, @@ -854,7 +853,10 @@ struct tasha_priv { int rx_8_count; bool clk_mode; bool clk_internal; - + /* Lock to prevent multiple functions voting at same time */ + struct mutex sb_clk_gear_lock; + /* Count for functions voting or un-voting */ + u32 ref_count; /* Lock to protect mclk enablement */ struct mutex mclk_lock; }; @@ -2217,6 +2219,32 @@ static void tasha_mbhc_moisture_config(struct wcd_mbhc *mbhc) tasha_mbhc_hph_l_pull_up_control(codec, mbhc->moist_iref); } +static void tasha_update_anc_state(struct snd_soc_codec *codec, bool enable, + int anc_num) +{ + if (enable) + snd_soc_update_bits(codec, WCD9335_CDC_RX1_RX_PATH_CFG0 + + (20 * anc_num), 0x10, 0x10); + else + snd_soc_update_bits(codec, WCD9335_CDC_RX1_RX_PATH_CFG0 + + (20 * anc_num), 0x10, 0x00); +} + +static bool tasha_is_anc_on(struct wcd_mbhc *mbhc) +{ + bool anc_on = false; + u16 ancl, ancr; + + ancl = + (snd_soc_read(mbhc->codec, WCD9335_CDC_RX1_RX_PATH_CFG0)) & 0x10; + ancr = + (snd_soc_read(mbhc->codec, WCD9335_CDC_RX2_RX_PATH_CFG0)) & 0x10; + + anc_on = !!(ancl | ancr); + + return anc_on; +} + static const struct wcd_mbhc_cb mbhc_cb = { .request_irq = tasha_mbhc_request_irq, .irq_control = tasha_mbhc_irq_control, @@ -2239,6 +2267,8 @@ static const struct wcd_mbhc_cb mbhc_cb = { .mbhc_gnd_det_ctrl = tasha_mbhc_gnd_det_ctrl, .hph_pull_down_ctrl = tasha_mbhc_hph_pull_down_ctrl, .mbhc_moisture_config = tasha_mbhc_moisture_config, + .update_anc_state = tasha_update_anc_state, + .is_anc_on = tasha_is_anc_on, }; static int tasha_get_anc_slot(struct snd_kcontrol *kcontrol, @@ -3153,10 +3183,7 @@ static int tasha_codec_enable_slimrx(struct snd_soc_dapm_widget *w, &dai->grph); break; case SND_SOC_DAPM_PRE_PMD: - if (!test_bit(SB_CLK_GEAR, &tasha_p->status_mask)) { - tasha_codec_vote_max_bw(codec, true); - set_bit(SB_CLK_GEAR, &tasha_p->status_mask); - } + tasha_codec_vote_max_bw(codec, true); break; case SND_SOC_DAPM_POST_PMD: ret = wcd9xxx_disconnect_port(core, &dai->wcd9xxx_ch_list, @@ -5468,10 +5495,7 @@ static int tasha_codec_enable_interpolator(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - if (!test_bit(SB_CLK_GEAR, &tasha->status_mask)) { - tasha_codec_vote_max_bw(codec, true); - set_bit(SB_CLK_GEAR, &tasha->status_mask); - } + tasha_codec_vote_max_bw(codec, true); /* Reset if needed */ tasha_codec_enable_prim_interpolator(codec, reg, event); break; @@ -11332,11 +11356,8 @@ static void tasha_shutdown(struct snd_pcm_substream *substream, if (tasha->intf_type == WCD9XXX_INTERFACE_TYPE_I2C) return; - if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) && - test_bit(SB_CLK_GEAR, &tasha->status_mask)) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) tasha_codec_vote_max_bw(dai->codec, false); - clear_bit(SB_CLK_GEAR, &tasha->status_mask); - } } static int tasha_set_decimator_rate(struct snd_soc_dai *dai, @@ -11571,15 +11592,11 @@ prim_rate: static int tasha_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct tasha_priv *tasha = snd_soc_codec_get_drvdata(dai->codec); - pr_debug("%s(): substream = %s stream = %d\n" , __func__, substream->name, substream->stream); - if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) && - test_bit(SB_CLK_GEAR, &tasha->status_mask)) { + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) tasha_codec_vote_max_bw(dai->codec, false); - clear_bit(SB_CLK_GEAR, &tasha->status_mask); - } return 0; } @@ -12110,8 +12127,10 @@ static int tasha_dig_core_power_collapse(struct tasha_priv *tasha, goto unlock_mutex; if (tasha->power_active_ref < 0) { - dev_dbg(tasha->dev, "%s: power_active_ref is negative\n", + dev_info(tasha->dev, + "%s: power_active_ref is negative, resetting it\n", __func__); + tasha->power_active_ref = 0; goto unlock_mutex; } @@ -13287,13 +13306,29 @@ static int tasha_codec_vote_max_bw(struct snd_soc_codec *codec, if (tasha->intf_type == WCD9XXX_INTERFACE_TYPE_I2C) return 0; - if (vote) - bw_ops = SLIM_BW_CLK_GEAR_9; - else - bw_ops = SLIM_BW_UNVOTE; + mutex_lock(&tasha->sb_clk_gear_lock); + if (vote) { + tasha->ref_count++; + if (tasha->ref_count == 1) { + bw_ops = SLIM_BW_CLK_GEAR_9; + tasha_codec_slim_reserve_bw(codec, + bw_ops, true); + } + } else if (!vote && tasha->ref_count > 0) { + tasha->ref_count--; + if (tasha->ref_count == 0) { + bw_ops = SLIM_BW_UNVOTE; + tasha_codec_slim_reserve_bw(codec, + bw_ops, true); + } + }; - return tasha_codec_slim_reserve_bw(codec, - bw_ops, true); + dev_dbg(codec->dev, "%s Value of counter after vote or un-vote is %d\n", + __func__, tasha->ref_count); + + mutex_unlock(&tasha->sb_clk_gear_lock); + + return 0; } static int tasha_cpe_err_irq_control(struct snd_soc_codec *codec, @@ -13476,6 +13511,8 @@ static int tasha_post_reset_cb(struct wcd9xxx *wcd9xxx) if (IS_ERR_VALUE(ret)) dev_err(codec->dev, "%s: invalid pdata\n", __func__); + /* Reset reference counter for voting for max bw */ + tasha->ref_count = 0; /* MBHC Init */ wcd_mbhc_deinit(&tasha->mbhc); tasha->mbhc_started = false; @@ -14262,6 +14299,7 @@ static int tasha_probe(struct platform_device *pdev) mutex_init(&tasha->swr_read_lock); mutex_init(&tasha->swr_write_lock); mutex_init(&tasha->swr_clk_lock); + mutex_init(&tasha->sb_clk_gear_lock); mutex_init(&tasha->mclk_lock); cdc_pwr = devm_kzalloc(&pdev->dev, sizeof(struct wcd9xxx_power_region), @@ -14366,6 +14404,7 @@ static int tasha_remove(struct platform_device *pdev) mutex_destroy(&tasha->mclk_lock); devm_kfree(&pdev->dev, tasha); snd_soc_unregister_codec(&pdev->dev); + mutex_destroy(&tasha->sb_clk_gear_lock); return 0; } diff --git a/sound/soc/codecs/wcd934x/wcd934x-dsp-cntl.c b/sound/soc/codecs/wcd934x/wcd934x-dsp-cntl.c index e791bf07ec67..29c218013a07 100644 --- a/sound/soc/codecs/wcd934x/wcd934x-dsp-cntl.c +++ b/sound/soc/codecs/wcd934x/wcd934x-dsp-cntl.c @@ -763,10 +763,6 @@ static int wcd_control_handler(struct device *dev, void *priv_data, case WDSP_EVENT_DLOAD_FAILED: case WDSP_EVENT_POST_SHUTDOWN: - if (event == WDSP_EVENT_POST_DLOAD_CODE) - /* Mark DSP online since code download is complete */ - wcd_cntl_change_online_state(cntl, 1); - /* Disable CPAR */ wcd_cntl_cpar_ctrl(cntl, false); /* Disable all the clocks */ @@ -775,6 +771,11 @@ static int wcd_control_handler(struct device *dev, void *priv_data, dev_err(codec->dev, "%s: Failed to disable clocks, err = %d\n", __func__, ret); + + if (event == WDSP_EVENT_POST_DLOAD_CODE) + /* Mark DSP online since code download is complete */ + wcd_cntl_change_online_state(cntl, 1); + break; case WDSP_EVENT_PRE_DLOAD_DATA: diff --git a/sound/soc/codecs/wcd934x/wcd934x-mbhc.c b/sound/soc/codecs/wcd934x/wcd934x-mbhc.c index 3d032f0a7115..075ab7b2ff2b 100644 --- a/sound/soc/codecs/wcd934x/wcd934x-mbhc.c +++ b/sound/soc/codecs/wcd934x/wcd934x-mbhc.c @@ -812,6 +812,32 @@ static bool tavil_hph_register_recovery(struct wcd_mbhc *mbhc) return wcd934x_mbhc->is_hph_recover; } +static void tavil_update_anc_state(struct snd_soc_codec *codec, bool enable, + int anc_num) +{ + if (enable) + snd_soc_update_bits(codec, WCD934X_CDC_RX1_RX_PATH_CFG0 + + (20 * anc_num), 0x10, 0x10); + else + snd_soc_update_bits(codec, WCD934X_CDC_RX1_RX_PATH_CFG0 + + (20 * anc_num), 0x10, 0x00); +} + +static bool tavil_is_anc_on(struct wcd_mbhc *mbhc) +{ + bool anc_on = false; + u16 ancl, ancr; + + ancl = + (snd_soc_read(mbhc->codec, WCD934X_CDC_RX1_RX_PATH_CFG0)) & 0x10; + ancr = + (snd_soc_read(mbhc->codec, WCD934X_CDC_RX2_RX_PATH_CFG0)) & 0x10; + + anc_on = !!(ancl | ancr); + + return anc_on; +} + static const struct wcd_mbhc_cb mbhc_cb = { .request_irq = tavil_mbhc_request_irq, .irq_control = tavil_mbhc_irq_control, @@ -835,6 +861,8 @@ static const struct wcd_mbhc_cb mbhc_cb = { .hph_pull_down_ctrl = tavil_mbhc_hph_pull_down_ctrl, .mbhc_moisture_config = tavil_mbhc_moisture_config, .hph_register_recovery = tavil_hph_register_recovery, + .update_anc_state = tavil_update_anc_state, + .is_anc_on = tavil_is_anc_on, }; static struct regulator *tavil_codec_find_ondemand_regulator( diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index eaaca97e2b8e..058f6c5fa676 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -1123,54 +1123,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wsa881x = { .get_regmap = wsa881x_get_regmap, }; -static int wsa881x_swr_startup(struct swr_device *swr_dev) -{ - int ret = 0; - u8 devnum = 0; - struct wsa881x_priv *wsa881x; - - wsa881x = swr_get_dev_data(swr_dev); - if (!wsa881x) { - dev_err(&swr_dev->dev, "%s: wsa881x is NULL\n", __func__); - return -EINVAL; - } - - /* - * Add 5msec delay to provide sufficient time for - * soundwire auto enumeration of slave devices as - * as per HW requirement. - */ - usleep_range(5000, 5010); - ret = swr_get_logical_dev_num(swr_dev, swr_dev->addr, &devnum); - if (ret) { - dev_dbg(&swr_dev->dev, - "%s get devnum %d for dev addr %lx failed\n", - __func__, devnum, swr_dev->addr); - goto err; - } - swr_dev->dev_num = devnum; - - wsa881x->regmap = devm_regmap_init_swr(swr_dev, - &wsa881x_regmap_config); - if (IS_ERR(wsa881x->regmap)) { - ret = PTR_ERR(wsa881x->regmap); - dev_err(&swr_dev->dev, "%s: regmap_init failed %d\n", - __func__, ret); - goto err; - } - - ret = snd_soc_register_codec(&swr_dev->dev, &soc_codec_dev_wsa881x, - NULL, 0); - if (ret) { - dev_err(&swr_dev->dev, "%s: Codec registration failed\n", - __func__); - goto err; - } - -err: - return ret; -} - static int wsa881x_gpio_ctrl(struct wsa881x_priv *wsa881x, bool enable) { int ret = 0; @@ -1232,6 +1184,8 @@ static int wsa881x_swr_probe(struct swr_device *pdev) { int ret = 0; struct wsa881x_priv *wsa881x; + u8 devnum = 0; + bool pin_state_current = false; wsa881x = devm_kzalloc(&pdev->dev, sizeof(struct wsa881x_priv), GFP_KERNEL); @@ -1265,6 +1219,9 @@ static int wsa881x_swr_probe(struct swr_device *pdev) if (ret) goto err; } + if (wsa881x->wsa_rst_np) + pin_state_current = msm_cdc_pinctrl_get_state( + wsa881x->wsa_rst_np); wsa881x_gpio_ctrl(wsa881x, true); wsa881x->state = WSA881X_DEV_UP; @@ -1291,8 +1248,45 @@ static int wsa881x_swr_probe(struct swr_device *pdev) &codec_debug_ops); } } + + /* + * Add 5msec delay to provide sufficient time for + * soundwire auto enumeration of slave devices as + * as per HW requirement. + */ + usleep_range(5000, 5010); + ret = swr_get_logical_dev_num(pdev, pdev->addr, &devnum); + if (ret) { + dev_dbg(&pdev->dev, + "%s get devnum %d for dev addr %lx failed\n", + __func__, devnum, pdev->addr); + goto dev_err; + } + pdev->dev_num = devnum; + + wsa881x->regmap = devm_regmap_init_swr(pdev, + &wsa881x_regmap_config); + if (IS_ERR(wsa881x->regmap)) { + ret = PTR_ERR(wsa881x->regmap); + dev_err(&pdev->dev, "%s: regmap_init failed %d\n", + __func__, ret); + goto dev_err; + } + + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wsa881x, + NULL, 0); + if (ret) { + dev_err(&pdev->dev, "%s: Codec registration failed\n", + __func__); + goto dev_err; + } + return 0; +dev_err: + if (pin_state_current == false) + wsa881x_gpio_ctrl(wsa881x, false); + swr_remove_device(pdev); err: return ret; } @@ -1371,6 +1365,7 @@ static int wsa881x_swr_reset(struct swr_device *pdev) /* Retry after 1 msec delay */ usleep_range(1000, 1100); } + pdev->dev_num = devnum; regcache_mark_dirty(wsa881x->regmap); regcache_sync(wsa881x->regmap); return 0; @@ -1425,7 +1420,6 @@ static struct swr_driver wsa881x_codec_driver = { .device_up = wsa881x_swr_up, .device_down = wsa881x_swr_down, .reset_device = wsa881x_swr_reset, - .startup = wsa881x_swr_startup, }; static int __init wsa881x_codec_init(void) diff --git a/sound/soc/msm/apq8096-auto.c b/sound/soc/msm/apq8096-auto.c index b6121d75c148..b1dff8764618 100644 --- a/sound/soc/msm/apq8096-auto.c +++ b/sound/soc/msm/apq8096-auto.c @@ -3044,21 +3044,22 @@ static struct snd_soc_dai_link apq8096_common_dai_links[] = { .codec_name = "snd-soc-dummy", }, { - .name = "SLIMBUS_4 Hostless", - .stream_name = "SLIMBUS_4 Hostless", - .cpu_dai_name = "SLIMBUS4_HOSTLESS", - .platform_name = "msm-pcm-hostless", + .name = "MSM8996 HFP RX", + .stream_name = "MultiMedia21", + .cpu_dai_name = "MultiMedia21", + .platform_name = "msm-pcm-loopback", .dynamic = 1, .dpcm_playback = 1, .dpcm_capture = 1, + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .no_host_mode = SND_SOC_DAI_LINK_NO_HOST, .ignore_suspend = 1, - /* this dailink has playback support */ + .no_host_mode = SND_SOC_DAI_LINK_NO_HOST, + /* this dainlink has playback support */ .ignore_pmdown_time = 1, - .codec_dai_name = "snd-soc-dummy-dai", - .codec_name = "snd-soc-dummy", + .be_id = MSM_FRONTEND_DAI_MULTIMEDIA21, }, { .name = "VoLTE", @@ -3480,12 +3481,13 @@ static struct snd_soc_dai_link apq8096_common_dai_links[] = { .be_id = MSM_FRONTEND_DAI_MULTIMEDIA15, }, { - .name = "MSM8996 Compress9", - .stream_name = "Compress9", + .name = "MSM8996 ULL NOIRQ 2", + .stream_name = "MM_NOIRQ_2", .cpu_dai_name = "MultiMedia16", - .platform_name = "msm-compress-dsp", + .platform_name = "msm-pcm-dsp-noirq", .dynamic = 1, .dpcm_playback = 1, + .dpcm_capture = 1, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .codec_dai_name = "snd-soc-dummy-dai", diff --git a/sound/soc/msm/msm-cpe-lsm.c b/sound/soc/msm/msm-cpe-lsm.c index 48f8e22e7faa..2d431a018315 100644 --- a/sound/soc/msm/msm-cpe-lsm.c +++ b/sound/soc/msm/msm-cpe-lsm.c @@ -2672,7 +2672,7 @@ static int msm_cpe_lsm_ioctl_compat(struct snd_pcm_substream *substream, event_status->payload_size; memcpy(udata_32->payload, event_status->payload, - u_pld_size); + event_status->payload_size); } } diff --git a/sound/soc/msm/msm-dai-fe.c b/sound/soc/msm/msm-dai-fe.c index 24dbbedf0be7..5facafdc7729 100644 --- a/sound/soc/msm/msm-dai-fe.c +++ b/sound/soc/msm/msm-dai-fe.c @@ -2496,8 +2496,21 @@ static struct snd_soc_dai_driver msm_fe_dais[] = { .rate_min = 8000, .rate_max = 384000, }, + .capture = { + .stream_name = "MultiMedia16 Capture", + .aif_name = "MM_UL16", + .rates = (SNDRV_PCM_RATE_8000_48000| + SNDRV_PCM_RATE_KNOT), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE), + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 48000, + }, .ops = &msm_fe_Multimedia_dai_ops, - .compress_new = snd_soc_new_compress, .name = "MultiMedia16", .probe = fe_dai_probe, }, @@ -2641,6 +2654,154 @@ static struct snd_soc_dai_driver msm_fe_dais[] = { .name = "MultiMedia20", .probe = fe_dai_probe, }, + { + .playback = { + .stream_name = "MultiMedia21 Playback", + .aif_name = "MM_DL21", + .rates = (SNDRV_PCM_RATE_8000_384000 | + SNDRV_PCM_RATE_KNOT), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE), + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 384000, + }, + .capture = { + .stream_name = "MultiMedia21 Capture", + .aif_name = "MM_UL21", + .rates = (SNDRV_PCM_RATE_8000_48000| + SNDRV_PCM_RATE_KNOT), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE), + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 48000, + }, + .ops = &msm_fe_Multimedia_dai_ops, + .name = "MultiMedia21", + .probe = fe_dai_probe, + }, + { + .playback = { + .stream_name = "MultiMedia22 Playback", + .aif_name = "MM_DL22", + .rates = (SNDRV_PCM_RATE_8000_384000 | + SNDRV_PCM_RATE_KNOT), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE), + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 384000, + }, + .ops = &msm_fe_Multimedia_dai_ops, + .name = "MultiMedia22", + .probe = fe_dai_probe, + }, + { + .playback = { + .stream_name = "MultiMedia23 Playback", + .aif_name = "MM_DL23", + .rates = (SNDRV_PCM_RATE_8000_384000 | + SNDRV_PCM_RATE_KNOT), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE), + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 384000, + }, + .ops = &msm_fe_Multimedia_dai_ops, + .name = "MultiMedia23", + .probe = fe_dai_probe, + }, + { + .playback = { + .stream_name = "MultiMedia24 Playback", + .aif_name = "MM_DL24", + .rates = (SNDRV_PCM_RATE_8000_384000 | + SNDRV_PCM_RATE_KNOT), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE), + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 384000, + }, + .ops = &msm_fe_Multimedia_dai_ops, + .name = "MultiMedia24", + .probe = fe_dai_probe, + }, + { + .playback = { + .stream_name = "MultiMedia25 Playback", + .aif_name = "MM_DL25", + .rates = (SNDRV_PCM_RATE_8000_384000 | + SNDRV_PCM_RATE_KNOT), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE), + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 384000, + }, + .ops = &msm_fe_Multimedia_dai_ops, + .name = "MultiMedia25", + .probe = fe_dai_probe, + }, + { + .playback = { + .stream_name = "MultiMedia26 Playback", + .aif_name = "MM_DL26", + .rates = (SNDRV_PCM_RATE_8000_384000 | + SNDRV_PCM_RATE_KNOT), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE), + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 384000, + }, + .ops = &msm_fe_Multimedia_dai_ops, + .compress_new = snd_soc_new_compress, + .name = "MultiMedia26", + .probe = fe_dai_probe, + }, + { + .capture = { + .stream_name = "MultiMedia27 Capture", + .aif_name = "MM_UL27", + .rates = (SNDRV_PCM_RATE_8000_192000| + SNDRV_PCM_RATE_KNOT), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE), + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, + .ops = &msm_fe_Multimedia_dai_ops, + .compress_new = snd_soc_new_compress, + .name = "MultiMedia27", + .probe = fe_dai_probe, + }, }; static int msm_fe_dai_dev_probe(struct platform_device *pdev) diff --git a/sound/soc/msm/msm8996.c b/sound/soc/msm/msm8996.c index 4eafd322d50d..010dfa3322a0 100644 --- a/sound/soc/msm/msm8996.c +++ b/sound/soc/msm/msm8996.c @@ -1,5 +1,5 @@ /* - * Copyright (c) 2014-2016, The Linux Foundation. All rights reserved. + * Copyright (c) 2014-2017, The Linux Foundation. All rights reserved. * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 and @@ -2910,12 +2910,13 @@ static struct snd_soc_dai_link msm8996_common_dai_links[] = { .be_id = MSM_FRONTEND_DAI_MULTIMEDIA15, }, { - .name = "MSM8996 Compress9", - .stream_name = "Compress9", + .name = "MSM8996 ULL NOIRQ_2", + .stream_name = "MM_NOIRQ_2", .cpu_dai_name = "MultiMedia16", - .platform_name = "msm-compress-dsp", + .platform_name = "msm-pcm-dsp-noirq", .dynamic = 1, .dpcm_playback = 1, + .dpcm_capture = 1, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .codec_dai_name = "snd-soc-dummy-dai", diff --git a/sound/soc/msm/msm8998.c b/sound/soc/msm/msm8998.c index 0b37e7e073aa..9159ea642816 100644 --- a/sound/soc/msm/msm8998.c +++ b/sound/soc/msm/msm8998.c @@ -397,7 +397,9 @@ static struct dev_config aux_pcm_tx_cfg[] = { }; static int msm_vi_feed_tx_ch = 2; -static const char *const slim_rx_ch_text[] = {"One", "Two"}; +static const char *const slim_rx_ch_text[] = {"One", "Two", "Three", "Four", + "Five", "Six", "Seven", + "Eight"}; static const char *const slim_tx_ch_text[] = {"One", "Two", "Three", "Four", "Five", "Six", "Seven", "Eight"}; @@ -432,7 +434,8 @@ static char const *tdm_sample_rate_text[] = {"KHZ_8", "KHZ_16", "KHZ_32", static const char *const auxpcm_rate_text[] = {"KHZ_8", "KHZ_16"}; static char const *mi2s_rate_text[] = {"KHZ_8", "KHZ_16", "KHZ_32", "KHZ_44P1", "KHZ_48", - "KHZ_96", "KHZ_192"}; + "KHZ_88P2", "KHZ_96", "KHZ_176P4", + "KHZ_192"}; static const char *const mi2s_ch_text[] = {"One", "Two", "Three", "Four", "Five", "Six", "Seven", "Eight"}; @@ -555,6 +558,13 @@ static struct snd_soc_dapm_route wcd_audio_paths[] = { {"MIC BIAS4", NULL, "MCLK"}, }; +static u32 mi2s_ebit_clk[MI2S_MAX] = { + Q6AFE_LPASS_CLK_ID_PRI_MI2S_EBIT, + Q6AFE_LPASS_CLK_ID_SEC_MI2S_EBIT, + Q6AFE_LPASS_CLK_ID_TER_MI2S_EBIT, + Q6AFE_LPASS_CLK_ID_QUAD_MI2S_EBIT, +}; + static struct afe_clk_set mi2s_clk[MI2S_MAX] = { { AFE_API_VERSION_I2S_CONFIG, @@ -2219,12 +2229,18 @@ static int mi2s_get_sample_rate_val(int sample_rate) case SAMPLING_RATE_48KHZ: sample_rate_val = 4; break; - case SAMPLING_RATE_96KHZ: + case SAMPLING_RATE_88P2KHZ: sample_rate_val = 5; break; - case SAMPLING_RATE_192KHZ: + case SAMPLING_RATE_96KHZ: sample_rate_val = 6; break; + case SAMPLING_RATE_176P4KHZ: + sample_rate_val = 7; + break; + case SAMPLING_RATE_192KHZ: + sample_rate_val = 8; + break; default: sample_rate_val = 4; break; @@ -2253,9 +2269,15 @@ static int mi2s_get_sample_rate(int value) sample_rate = SAMPLING_RATE_48KHZ; break; case 5: - sample_rate = SAMPLING_RATE_96KHZ; + sample_rate = SAMPLING_RATE_88P2KHZ; break; case 6: + sample_rate = SAMPLING_RATE_96KHZ; + break; + case 7: + sample_rate = SAMPLING_RATE_176P4KHZ; + break; + case 8: sample_rate = SAMPLING_RATE_192KHZ; break; default: @@ -4506,6 +4528,11 @@ static int msm_mi2s_snd_startup(struct snd_pcm_substream *substream) */ mutex_lock(&mi2s_intf_conf[index].lock); if (++mi2s_intf_conf[index].ref_cnt == 1) { + /* Check if msm needs to provide the clock to the interface */ + if (!mi2s_intf_conf[index].msm_is_mi2s_master) { + fmt = SND_SOC_DAIFMT_CBM_CFM; + mi2s_clk[index].clk_id = mi2s_ebit_clk[index]; + } ret = msm_mi2s_set_sclk(substream, true); if (IS_ERR_VALUE(ret)) { dev_err(rtd->card->dev, @@ -4525,9 +4552,6 @@ static int msm_mi2s_snd_startup(struct snd_pcm_substream *substream) ret = -EINVAL; goto clk_off; } - /* Check if msm needs to provide the clock to the interface */ - if (!mi2s_intf_conf[index].msm_is_mi2s_master) - fmt = SND_SOC_DAIFMT_CBM_CFM; ret = snd_soc_dai_set_fmt(cpu_dai, fmt); if (IS_ERR_VALUE(ret)) { pr_err("%s: set fmt cpu dai failed for MI2S (%d), err:%d\n", @@ -5304,12 +5328,13 @@ static struct snd_soc_dai_link msm_common_dai_links[] = { .be_id = MSM_FRONTEND_DAI_MULTIMEDIA15, }, { - .name = MSM_DAILINK_NAME(Compress9), - .stream_name = "Compress9", + .name = MSM_DAILINK_NAME(ULL_NOIRQ_2), + .stream_name = "MM_NOIRQ_2", .cpu_dai_name = "MultiMedia16", - .platform_name = "msm-compress-dsp", + .platform_name = "msm-pcm-dsp-noirq", .dynamic = 1, .dpcm_playback = 1, + .dpcm_capture = 1, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .codec_dai_name = "snd-soc-dummy-dai", @@ -5498,7 +5523,7 @@ static struct snd_soc_dai_link msm_common_misc_fe_dai_links[] = { { .name = MSM_DAILINK_NAME(Transcode Loopback Playback), .stream_name = "Transcode Loopback Playback", - .cpu_dai_name = "MultiMedia14", + .cpu_dai_name = "MultiMedia26", .platform_name = "msm-transcode-loopback", .dynamic = 1, .dpcm_playback = 1, @@ -5509,12 +5534,12 @@ static struct snd_soc_dai_link msm_common_misc_fe_dai_links[] = { .ignore_suspend = 1, .ignore_pmdown_time = 1, /* this dainlink has playback support */ - .be_id = MSM_FRONTEND_DAI_MULTIMEDIA14, + .be_id = MSM_FRONTEND_DAI_MULTIMEDIA26, }, { .name = MSM_DAILINK_NAME(Transcode Loopback Capture), .stream_name = "Transcode Loopback Capture", - .cpu_dai_name = "MultiMedia18", + .cpu_dai_name = "MultiMedia27", .platform_name = "msm-transcode-loopback", .dynamic = 1, .dpcm_capture = 1, @@ -5524,7 +5549,92 @@ static struct snd_soc_dai_link msm_common_misc_fe_dai_links[] = { .codec_name = "snd-soc-dummy", .ignore_suspend = 1, .ignore_pmdown_time = 1, - .be_id = MSM_FRONTEND_DAI_MULTIMEDIA18, + .be_id = MSM_FRONTEND_DAI_MULTIMEDIA27, + }, + { + .name = "MultiMedia21", + .stream_name = "MultiMedia21", + .cpu_dai_name = "MultiMedia21", + .platform_name = "msm-pcm-dsp.0", + .dynamic = 1, + .async_ops = ASYNC_DPCM_SND_SOC_PREPARE, + .dpcm_playback = 1, + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST}, + .ignore_suspend = 1, + /* this dainlink has playback support */ + .ignore_pmdown_time = 1, + .be_id = MSM_FRONTEND_DAI_MULTIMEDIA21, + }, + { + .name = "MultiMedia22", + .stream_name = "MultiMedia22", + .cpu_dai_name = "MultiMedia22", + .platform_name = "msm-pcm-dsp.0", + .dynamic = 1, + .async_ops = ASYNC_DPCM_SND_SOC_PREPARE, + .dpcm_playback = 1, + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST}, + .ignore_suspend = 1, + /* this dainlink has playback support */ + .ignore_pmdown_time = 1, + .be_id = MSM_FRONTEND_DAI_MULTIMEDIA22, + }, + { + .name = "MultiMedia23", + .stream_name = "MultiMedia23", + .cpu_dai_name = "MultiMedia23", + .platform_name = "msm-pcm-dsp.0", + .dynamic = 1, + .async_ops = ASYNC_DPCM_SND_SOC_PREPARE, + .dpcm_playback = 1, + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST}, + .ignore_suspend = 1, + /* this dainlink has playback support */ + .ignore_pmdown_time = 1, + .be_id = MSM_FRONTEND_DAI_MULTIMEDIA23, + }, + { + .name = "MultiMedia24", + .stream_name = "MultiMedia24", + .cpu_dai_name = "MultiMedia24", + .platform_name = "msm-pcm-dsp.0", + .dynamic = 1, + .async_ops = ASYNC_DPCM_SND_SOC_PREPARE, + .dpcm_playback = 1, + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST}, + .ignore_suspend = 1, + /* this dainlink has playback support */ + .ignore_pmdown_time = 1, + .be_id = MSM_FRONTEND_DAI_MULTIMEDIA24, + }, + { + .name = "MultiMedia25", + .stream_name = "MultiMedia25", + .cpu_dai_name = "MultiMedia25", + .platform_name = "msm-pcm-dsp.0", + .dynamic = 1, + .async_ops = ASYNC_DPCM_SND_SOC_PREPARE, + .dpcm_playback = 1, + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST}, + .ignore_suspend = 1, + /* this dainlink has playback support */ + .ignore_pmdown_time = 1, + .be_id = MSM_FRONTEND_DAI_MULTIMEDIA25, }, }; diff --git a/sound/soc/msm/qdsp6v2/audio_calibration.c b/sound/soc/msm/qdsp6v2/audio_calibration.c index 60d09dfaeb7f..2a1b34776b68 100644 --- a/sound/soc/msm/qdsp6v2/audio_calibration.c +++ b/sound/soc/msm/qdsp6v2/audio_calibration.c @@ -453,6 +453,12 @@ static long audio_cal_shared_ioctl(struct file *file, unsigned int cmd, data->cal_type.cal_hdr.buffer_number); ret = -EINVAL; goto done; + } else if ((data->hdr.cal_type_size + sizeof(data->hdr)) > size) { + pr_err("%s: cal type hdr size %zd + cal type size %d is greater than user buffer size %d\n", + __func__, sizeof(data->hdr), data->hdr.cal_type_size, + size); + ret = -EFAULT; + goto done; } @@ -490,13 +496,7 @@ static long audio_cal_shared_ioctl(struct file *file, unsigned int cmd, goto unlock; if (data == NULL) goto unlock; - if ((sizeof(data->hdr) + data->hdr.cal_type_size) > size) { - pr_err("%s: header size %zd plus cal type size %d are greater than data buffer size %d\n", - __func__, sizeof(data->hdr), - data->hdr.cal_type_size, size); - ret = -EFAULT; - goto unlock; - } else if (copy_to_user((void *)arg, data, + if (copy_to_user(arg, data, sizeof(data->hdr) + data->hdr.cal_type_size)) { pr_err("%s: Could not copy cal type to user\n", __func__); diff --git a/sound/soc/msm/qdsp6v2/msm-audio-effects-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-audio-effects-q6-v2.c index e312a879b86a..1286d3185780 100644 --- a/sound/soc/msm/qdsp6v2/msm-audio-effects-q6-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-audio-effects-q6-v2.c @@ -155,7 +155,7 @@ int msm_audio_effects_virtualizer_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "VIRT ENABLE", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_VIRTUALIZER; *updt_params++ = @@ -183,7 +183,7 @@ int msm_audio_effects_virtualizer_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "VIRT STRENGTH", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_VIRTUALIZER; *updt_params++ = @@ -211,7 +211,7 @@ int msm_audio_effects_virtualizer_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "VIRT OUT_TYPE", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_VIRTUALIZER; *updt_params++ = @@ -239,7 +239,7 @@ int msm_audio_effects_virtualizer_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "VIRT GAIN_ADJUST", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_VIRTUALIZER; *updt_params++ = @@ -318,7 +318,7 @@ int msm_audio_effects_reverb_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "REVERB_ENABLE", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_REVERB; *updt_params++ = @@ -346,7 +346,7 @@ int msm_audio_effects_reverb_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "REVERB_MODE", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_REVERB; *updt_params++ = @@ -374,7 +374,7 @@ int msm_audio_effects_reverb_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "REVERB_PRESET", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_REVERB; *updt_params++ = @@ -402,7 +402,7 @@ int msm_audio_effects_reverb_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "REVERB_WET_MIX", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_REVERB; *updt_params++ = @@ -430,7 +430,7 @@ int msm_audio_effects_reverb_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "REVERB_GAIN_ADJUST", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_REVERB; *updt_params++ = @@ -458,7 +458,7 @@ int msm_audio_effects_reverb_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "REVERB_ROOM_LEVEL", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_REVERB; *updt_params++ = @@ -486,7 +486,7 @@ int msm_audio_effects_reverb_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "REVERB_ROOM_HF_LEVEL", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_REVERB; *updt_params++ = @@ -514,7 +514,7 @@ int msm_audio_effects_reverb_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "REVERB_DECAY_TIME", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_REVERB; *updt_params++ = @@ -542,7 +542,7 @@ int msm_audio_effects_reverb_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "REVERB_DECAY_HF_RATIO", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_REVERB; *updt_params++ = @@ -570,7 +570,7 @@ int msm_audio_effects_reverb_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "REVERB_REFLECTIONS_LEVEL", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_REVERB; *updt_params++ = @@ -598,7 +598,7 @@ int msm_audio_effects_reverb_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "REVERB_REFLECTIONS_DELAY", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_REVERB; *updt_params++ = @@ -626,7 +626,7 @@ int msm_audio_effects_reverb_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "REVERB_LEVEL", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_REVERB; *updt_params++ = @@ -654,7 +654,7 @@ int msm_audio_effects_reverb_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "REVERB_DELAY", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_REVERB; *updt_params++ = @@ -682,7 +682,7 @@ int msm_audio_effects_reverb_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "REVERB_DIFFUSION", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_REVERB; *updt_params++ = @@ -710,7 +710,7 @@ int msm_audio_effects_reverb_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "REVERB_DENSITY", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_REVERB; *updt_params++ = @@ -790,7 +790,7 @@ int msm_audio_effects_bass_boost_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "BASS_BOOST_ENABLE", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_BASS_BOOST; *updt_params++ = @@ -818,7 +818,7 @@ int msm_audio_effects_bass_boost_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "BASS_BOOST_MODE", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_BASS_BOOST; *updt_params++ = @@ -846,7 +846,7 @@ int msm_audio_effects_bass_boost_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "BASS_BOOST_STRENGTH", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_BASS_BOOST; *updt_params++ = @@ -924,7 +924,7 @@ int msm_audio_effects_pbe_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "PBE_ENABLE", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_PBE; *updt_params++ = @@ -950,7 +950,7 @@ int msm_audio_effects_pbe_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "PBE_PARAM", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_PBE; *updt_params++ = @@ -1035,7 +1035,7 @@ int msm_audio_effects_popless_eq_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "EQ_ENABLE", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_POPLESS_EQUALIZER; *updt_params++ = @@ -1103,7 +1103,7 @@ int msm_audio_effects_popless_eq_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "EQ_CONFIG", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_POPLESS_EQUALIZER; *updt_params++ = @@ -1154,7 +1154,7 @@ int msm_audio_effects_popless_eq_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "EQ_BAND_INDEX", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_POPLESS_EQUALIZER; *updt_params++ = @@ -1186,7 +1186,7 @@ int msm_audio_effects_popless_eq_handler(struct audio_client *ac, MAX_INBAND_PARAM_SZ, "EQ_SINGLE_BAND_FREQ", rc); if (rc != 0) - break; + goto invalid_config; *updt_params++ = AUDPROC_MODULE_ID_POPLESS_EQUALIZER; *updt_params++ = @@ -1276,7 +1276,7 @@ static int __msm_audio_effects_volume_handler(struct audio_client *ac, "VOLUME/VOLUME2_GAIN_2CH", rc); if (rc != 0) - break; + goto invalid_config; if (instance == SOFT_VOLUME_INSTANCE_2) *updt_params++ = ASM_MODULE_ID_VOL_CTRL2; @@ -1325,7 +1325,7 @@ static int __msm_audio_effects_volume_handler(struct audio_client *ac, "VOLUME/VOLUME2_GAIN_MASTER", rc); if (rc != 0) - break; + goto invalid_config; if (instance == SOFT_VOLUME_INSTANCE_2) *updt_params++ = ASM_MODULE_ID_VOL_CTRL2; diff --git a/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c index 0acf6e8ffe49..076dbed207a9 100644 --- a/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c @@ -39,6 +39,7 @@ #include <sound/apr_audio-v2.h> #include <sound/q6asm-v2.h> +#include <sound/q6core.h> #include <sound/compress_params.h> #include <sound/compress_offload.h> #include <sound/compress_driver.h> @@ -103,6 +104,7 @@ struct msm_compr_pdata { bool use_legacy_api; /* indicates use older asm apis*/ struct msm_compr_dec_params *dec_params[MSM_FRONTEND_DAI_MAX]; struct msm_compr_ch_map *ch_map[MSM_FRONTEND_DAI_MAX]; + int32_t ion_fd[MSM_FRONTEND_DAI_MAX]; }; struct msm_compr_audio { @@ -156,6 +158,8 @@ struct msm_compr_audio { uint32_t start_delay_lsw; uint32_t start_delay_msw; + int32_t shm_ion_fd; + uint64_t marker_timestamp; struct msm_compr_gapless_state gapless_state; @@ -1506,6 +1510,40 @@ static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream) return ret; } +static int msm_compr_map_unmap_fd(int fd, bool add_pages) +{ + ion_phys_addr_t paddr; + size_t pa_len = 0; + int ret = 0; + u8 assign_type; + + if (add_pages) + assign_type = HLOS_TO_ADSP; + else + assign_type = ADSP_TO_HLOS; + + ret = msm_audio_ion_phys_assign("audio_lib_mem_client", fd, + &paddr, &pa_len, assign_type); + if (ret) { + pr_err("%s: audio lib ION phys failed, rc = %d\n", + __func__, ret); + goto done; + } + + ret = q6core_add_remove_pool_pages(paddr, pa_len, + ADSP_MEMORY_MAP_HLOS_PHYSPOOL, add_pages); + if (ret) { + pr_err("%s: add remove pages failed, rc = %d\n", __func__, ret); + /* Assign back to HLOS if add pages cmd failed */ + if (add_pages) + msm_audio_ion_phys_assign("audio_lib_mem_client", fd, + &paddr, &pa_len, ADSP_TO_HLOS); + } + +done: + return ret; +} + static int msm_compr_playback_open(struct snd_compr_stream *cstream) { struct snd_compr_runtime *runtime = cstream->runtime; @@ -1513,6 +1551,7 @@ static int msm_compr_playback_open(struct snd_compr_stream *cstream) struct msm_compr_audio *prtd; struct msm_compr_pdata *pdata = snd_soc_platform_get_drvdata(rtd->platform); + int ret = 0; pr_debug("%s\n", __func__); prtd = kzalloc(sizeof(struct msm_compr_audio), GFP_KERNEL); @@ -1528,19 +1567,16 @@ static int msm_compr_playback_open(struct snd_compr_stream *cstream) kzalloc(sizeof(struct msm_compr_audio_effects), GFP_KERNEL); if (!pdata->audio_effects[rtd->dai_link->be_id]) { pr_err("%s: Could not allocate memory for effects\n", __func__); - pdata->cstream[rtd->dai_link->be_id] = NULL; - kfree(prtd); - return -ENOMEM; + ret = -ENOMEM; + goto effect_err; } pdata->dec_params[rtd->dai_link->be_id] = kzalloc(sizeof(struct msm_compr_dec_params), GFP_KERNEL); if (!pdata->dec_params[rtd->dai_link->be_id]) { pr_err("%s: Could not allocate memory for dec params\n", __func__); - kfree(pdata->audio_effects[rtd->dai_link->be_id]); - pdata->cstream[rtd->dai_link->be_id] = NULL; - kfree(prtd); - return -ENOMEM; + ret = -ENOMEM; + goto param_err; } prtd->codec = FORMAT_MP3; prtd->bytes_received = 0; @@ -1584,19 +1620,32 @@ static int msm_compr_playback_open(struct snd_compr_stream *cstream) (app_cb)compr_event_handler, prtd); if (!prtd->audio_client) { pr_err("%s: Could not allocate memory for client\n", __func__); - kfree(pdata->audio_effects[rtd->dai_link->be_id]); - kfree(pdata->dec_params[rtd->dai_link->be_id]); - pdata->cstream[rtd->dai_link->be_id] = NULL; - runtime->private_data = NULL; - kfree(prtd); - return -ENOMEM; + ret = -ENOMEM; + goto ac_err; } pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session); prtd->audio_client->perf_mode = false; prtd->session_id = prtd->audio_client->session; msm_adsp_init_mixer_ctl_pp_event_queue(rtd); - + if (pdata->ion_fd[rtd->dai_link->be_id] > 0) { + ret = msm_compr_map_unmap_fd( + pdata->ion_fd[rtd->dai_link->be_id], true); + if (ret < 0) + goto map_err; + } return 0; + +map_err: + q6asm_audio_client_free(prtd->audio_client); +ac_err: + kfree(pdata->dec_params[rtd->dai_link->be_id]); +param_err: + kfree(pdata->audio_effects[rtd->dai_link->be_id]); +effect_err: + pdata->cstream[rtd->dai_link->be_id] = NULL; + runtime->private_data = NULL; + kfree(prtd); + return ret; } static int msm_compr_capture_open(struct snd_compr_stream *cstream) @@ -1675,6 +1724,8 @@ static int msm_compr_playback_free(struct snd_compr_stream *cstream) int dir = IN, ret = 0, stream_id; unsigned long flags; uint32_t stream_index; + ion_phys_addr_t paddr; + size_t pa_len = 0; pr_debug("%s\n", __func__); @@ -1748,6 +1799,15 @@ static int msm_compr_playback_free(struct snd_compr_stream *cstream) } q6asm_audio_client_buf_free_contiguous(dir, ac); + if (prtd->shm_ion_fd > 0) + msm_audio_ion_phys_assign("audio_shm_mem_client", + prtd->shm_ion_fd, + &paddr, &pa_len, ADSP_TO_HLOS); + if (pdata->ion_fd[soc_prtd->dai_link->be_id] > 0) { + msm_compr_map_unmap_fd(pdata->ion_fd[soc_prtd->dai_link->be_id], + false); + pdata->ion_fd[soc_prtd->dai_link->be_id] = 0; + } q6asm_audio_client_free(ac); msm_adsp_clean_mixer_ctl_pp_event_queue(soc_prtd); @@ -3655,7 +3715,7 @@ done: return ret; } -static int msm_compr_ion_fd_map_put(struct snd_kcontrol *kcontrol, +static int msm_compr_shm_ion_fd_map_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); @@ -3664,7 +3724,6 @@ static int msm_compr_ion_fd_map_put(struct snd_kcontrol *kcontrol, snd_soc_component_get_drvdata(comp); struct snd_compr_stream *cstream = NULL; struct msm_compr_audio *prtd; - int fd; int ret = 0; if (fe_id >= MSM_FRONTEND_DAI_MAX) { @@ -3694,10 +3753,34 @@ static int msm_compr_ion_fd_map_put(struct snd_kcontrol *kcontrol, goto done; } - memcpy(&fd, ucontrol->value.bytes.data, sizeof(fd)); - ret = q6asm_send_ion_fd(prtd->audio_client, fd); + memcpy(&prtd->shm_ion_fd, ucontrol->value.bytes.data, + sizeof(prtd->shm_ion_fd)); + ret = q6asm_audio_map_shm_fd(prtd->audio_client, prtd->shm_ion_fd); if (ret < 0) - pr_err("%s: failed to register ion fd\n", __func__); + pr_err("%s: failed to map shm mem\n", __func__); +done: + return ret; +} + +static int msm_compr_lib_ion_fd_map_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + unsigned long fe_id = kcontrol->private_value; + struct msm_compr_pdata *pdata = (struct msm_compr_pdata *) + snd_soc_component_get_drvdata(comp); + int ret = 0; + + if (fe_id >= MSM_FRONTEND_DAI_MAX) { + pr_err("%s Received out of bounds invalid fe_id %lu\n", + __func__, fe_id); + ret = -EINVAL; + goto done; + } + + memcpy(&pdata->ion_fd[fe_id], ucontrol->value.bytes.data, + sizeof(pdata->ion_fd[fe_id])); + done: return ret; } @@ -4329,7 +4412,7 @@ static int msm_compr_add_channel_map_control(struct snd_soc_pcm_runtime *rtd) return 0; } -static int msm_compr_add_io_fd_cmd_control(struct snd_soc_pcm_runtime *rtd) +static int msm_compr_add_shm_ion_fd_cmd_control(struct snd_soc_pcm_runtime *rtd) { const char *mixer_ctl_name = "Playback ION FD"; const char *deviceNo = "NN"; @@ -4341,7 +4424,52 @@ static int msm_compr_add_io_fd_cmd_control(struct snd_soc_pcm_runtime *rtd) .name = "?", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, .info = msm_adsp_stream_cmd_info, - .put = msm_compr_ion_fd_map_put, + .put = msm_compr_shm_ion_fd_map_put, + .private_value = 0, + } + }; + + if (!rtd) { + pr_err("%s NULL rtd\n", __func__); + ret = -EINVAL; + goto done; + } + + ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1; + mixer_str = kzalloc(ctl_len, GFP_KERNEL); + if (!mixer_str) { + ret = -ENOMEM; + goto done; + } + + snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device); + fe_ion_fd_config_control[0].name = mixer_str; + fe_ion_fd_config_control[0].private_value = rtd->dai_link->be_id; + pr_debug("%s: Registering new mixer ctl %s\n", __func__, mixer_str); + ret = snd_soc_add_platform_controls(rtd->platform, + fe_ion_fd_config_control, + ARRAY_SIZE(fe_ion_fd_config_control)); + if (ret < 0) + pr_err("%s: failed to add ctl %s\n", __func__, mixer_str); + + kfree(mixer_str); +done: + return ret; +} + +static int msm_compr_add_lib_ion_fd_cmd_control(struct snd_soc_pcm_runtime *rtd) +{ + const char *mixer_ctl_name = "Playback ION LIB FD"; + const char *deviceNo = "NN"; + char *mixer_str = NULL; + int ctl_len = 0, ret = 0; + struct snd_kcontrol_new fe_ion_fd_config_control[1] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "?", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = msm_adsp_stream_cmd_info, + .put = msm_compr_lib_ion_fd_map_put, .private_value = 0, } }; @@ -4442,11 +4570,16 @@ static int msm_compr_new(struct snd_soc_pcm_runtime *rtd) pr_err("%s: Could not add Compr ADSP Stream Callback Control\n", __func__); - rc = msm_compr_add_io_fd_cmd_control(rtd); + rc = msm_compr_add_shm_ion_fd_cmd_control(rtd); if (rc) pr_err("%s: Could not add Compr ion fd Control\n", __func__); + rc = msm_compr_add_lib_ion_fd_cmd_control(rtd); + if (rc) + pr_err("%s: Could not add Compr ion lib fd Control\n", + __func__); + rc = msm_compr_add_event_ack_cmd_control(rtd); if (rc) pr_err("%s: Could not add Compr event ack Control\n", diff --git a/sound/soc/msm/qdsp6v2/msm-dai-q6-hdmi-v2.c b/sound/soc/msm/qdsp6v2/msm-dai-q6-hdmi-v2.c index a71fb74d35bc..6b1c150f1ee4 100644 --- a/sound/soc/msm/qdsp6v2/msm-dai-q6-hdmi-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-dai-q6-hdmi-v2.c @@ -283,6 +283,7 @@ static void msm_dai_q6_hdmi_shutdown(struct snd_pcm_substream *substream, *dai_data->status_mask); clear_bit(STATUS_PORT_STARTED, dai_data->status_mask); + memset(&dai_data->ca, 0, sizeof(dai_data->ca)); } diff --git a/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c index 64b1961794b9..292b3d04f7d5 100644 --- a/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c @@ -3586,6 +3586,9 @@ static int msm_dai_q6_dai_mi2s_probe(struct snd_soc_dai *dai) ctrl = &mi2s_config_controls[11]; } + if (dai->id == MSM_QUAT_MI2S) + ctrl = &mi2s_config_controls[8]; + if (ctrl) { rc = snd_ctl_add(dai->component->card->snd_card, snd_ctl_new1(ctrl, @@ -4149,7 +4152,8 @@ static struct snd_soc_dai_driver msm_dai_q6_mi2s_dai[] = { .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE, .rate_min = 8000, @@ -5144,6 +5148,27 @@ static int msm_dai_tdm_q6_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "%s: Clk Rate from DT file %d\n", __func__, tdm_clk_set.clk_freq_in_hz); + /* initialize static tdm clk attribute to default value */ + tdm_clk_set.clk_attri = Q6AFE_LPASS_CLK_ATTRIBUTE_INVERT_COUPLE_NO; + + /* extract tdm clk attribute into static */ + if (of_find_property(pdev->dev.of_node, + "qcom,msm-cpudai-tdm-clk-attribute", NULL)) { + rc = of_property_read_u16(pdev->dev.of_node, + "qcom,msm-cpudai-tdm-clk-attribute", + &tdm_clk_set.clk_attri); + if (rc) { + dev_err(&pdev->dev, "%s: clk attribute from DT file %s\n", + __func__, "qcom,msm-cpudai-tdm-clk-attribute"); + goto rtn; + } + dev_dbg(&pdev->dev, "%s: clk attribute from DT file %d\n", + __func__, tdm_clk_set.clk_attri); + } else { + dev_dbg(&pdev->dev, "%s: No optional clk attribute found\n", + __func__); + } + /* extract tdm clk src master/slave info into static */ rc = of_property_read_u32(pdev->dev.of_node, "qcom,msm-cpudai-tdm-clk-internal", @@ -6223,6 +6248,41 @@ static int msm_dai_q6_tdm_set_tdm_slot(struct snd_soc_dai *dai, return rc; } +static int msm_dai_q6_tdm_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct msm_dai_q6_tdm_dai_data *dai_data = + dev_get_drvdata(dai->dev); + + switch (dai->id) { + case AFE_PORT_ID_PRIMARY_TDM_RX: + case AFE_PORT_ID_PRIMARY_TDM_RX_1: + case AFE_PORT_ID_PRIMARY_TDM_RX_2: + case AFE_PORT_ID_PRIMARY_TDM_RX_3: + case AFE_PORT_ID_PRIMARY_TDM_RX_4: + case AFE_PORT_ID_PRIMARY_TDM_RX_5: + case AFE_PORT_ID_PRIMARY_TDM_RX_6: + case AFE_PORT_ID_PRIMARY_TDM_RX_7: + case AFE_PORT_ID_PRIMARY_TDM_TX: + case AFE_PORT_ID_PRIMARY_TDM_TX_1: + case AFE_PORT_ID_PRIMARY_TDM_TX_2: + case AFE_PORT_ID_PRIMARY_TDM_TX_3: + case AFE_PORT_ID_PRIMARY_TDM_TX_4: + case AFE_PORT_ID_PRIMARY_TDM_TX_5: + case AFE_PORT_ID_PRIMARY_TDM_TX_6: + case AFE_PORT_ID_PRIMARY_TDM_TX_7: + dai_data->clk_set.clk_freq_in_hz = freq; + break; + default: + return 0; + } + + dev_dbg(dai->dev, "%s: dai id = 0x%x group clk_freq %d\n", + __func__, dai->id, freq); + return 0; +} + + static int msm_dai_q6_tdm_set_channel_map(struct snd_soc_dai *dai, unsigned int tx_num, unsigned int *tx_slot, unsigned int rx_num, unsigned int *rx_slot) @@ -6650,6 +6710,7 @@ static struct snd_soc_dai_ops msm_dai_q6_tdm_ops = { .hw_params = msm_dai_q6_tdm_hw_params, .set_tdm_slot = msm_dai_q6_tdm_set_tdm_slot, .set_channel_map = msm_dai_q6_tdm_set_channel_map, + .set_sysclk = msm_dai_q6_tdm_set_sysclk, .shutdown = msm_dai_q6_tdm_shutdown, }; diff --git a/sound/soc/msm/qdsp6v2/msm-lsm-client.c b/sound/soc/msm/qdsp6v2/msm-lsm-client.c index 3e72aa130c18..35270e3340ec 100644 --- a/sound/soc/msm/qdsp6v2/msm-lsm-client.c +++ b/sound/soc/msm/qdsp6v2/msm-lsm-client.c @@ -1683,7 +1683,7 @@ static int msm_lsm_ioctl(struct snd_pcm_substream *substream, dev_err(rtd->dev, "%s REG_SND_MODEL failed err %d\n", __func__, err); - return err; + goto done; } break; case SNDRV_LSM_SET_PARAMS: { @@ -1855,13 +1855,15 @@ static int msm_lsm_ioctl(struct snd_pcm_substream *substream, dev_err(rtd->dev, "%s: Invalid params event_status_v3\n", __func__); - return -EINVAL; + err = -EINVAL; + goto done; } if (copy_from_user(&userarg, arg, sizeof(userarg))) { dev_err(rtd->dev, "%s: err copyuser event_status_v3\n", __func__); - return -EFAULT; + err = -EFAULT; + goto done; } if (userarg.payload_size > @@ -1869,7 +1871,8 @@ static int msm_lsm_ioctl(struct snd_pcm_substream *substream, pr_err("%s: payload_size %d is invalid, max allowed = %d\n", __func__, userarg.payload_size, LISTEN_MAX_STATUS_PAYLOAD_SIZE); - return -EINVAL; + err = -EINVAL; + goto done; } size = sizeof(struct snd_lsm_event_status_v3) + @@ -1879,7 +1882,8 @@ static int msm_lsm_ioctl(struct snd_pcm_substream *substream, dev_err(rtd->dev, "%s: Allocation failed event status size %d\n", __func__, size); - return -EFAULT; + err = -EFAULT; + goto done; } user->payload_size = userarg.payload_size; err = msm_lsm_ioctl_shared(substream, cmd, user); diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-q6-noirq.c b/sound/soc/msm/qdsp6v2/msm-pcm-q6-noirq.c index 6b026bafa276..276270258771 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-q6-noirq.c +++ b/sound/soc/msm/qdsp6v2/msm-pcm-q6-noirq.c @@ -30,9 +30,12 @@ #include <sound/control.h> #include <sound/q6audio-v2.h> #include <sound/timer.h> +#include <sound/hwdep.h> + #include <asm/dma.h> #include <sound/tlv.h> #include <sound/pcm_params.h> +#include <sound/devdep_params.h> #include "msm-pcm-q6-v2.h" #include "msm-pcm-routing-v2.h" @@ -421,6 +424,42 @@ static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } + +static int msm_pcm_mmap_fd(struct snd_pcm_substream *substream, + struct snd_pcm_mmap_fd *mmap_fd) +{ + struct msm_audio *prtd; + struct audio_port_data *apd; + struct audio_buffer *ab; + int dir = -1; + + if (!substream->runtime) { + pr_err("%s substream runtime not found\n", __func__); + return -EFAULT; + } + + prtd = substream->runtime->private_data; + if (!prtd || !prtd->audio_client || !prtd->mmap_flag) { + pr_err("%s no audio client or not an mmap session\n", __func__); + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = IN; + else + dir = OUT; + + apd = prtd->audio_client->port; + ab = &(apd[dir].buf[0]); + mmap_fd->fd = ion_share_dma_buf_fd(ab->client, ab->handle); + if (mmap_fd->fd >= 0) { + mmap_fd->dir = dir; + mmap_fd->actual_size = ab->actual_size; + mmap_fd->size = ab->size; + } + return mmap_fd->fd < 0 ? -EFAULT : 0; +} + static int msm_pcm_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg) { @@ -445,6 +484,15 @@ static int msm_pcm_ioctl(struct snd_pcm_substream *substream, return snd_pcm_lib_ioctl(substream, cmd, arg); } +#ifdef CONFIG_COMPAT +static int msm_pcm_compat_ioctl(struct snd_pcm_substream *substream, + unsigned int cmd, void *arg) +{ + /* we only handle RESET which is common for both modes */ + return msm_pcm_ioctl(substream, cmd, arg); +} +#endif + static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -994,6 +1042,101 @@ static int msm_pcm_add_app_type_controls(struct snd_soc_pcm_runtime *rtd) return 0; } +static int msm_pcm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, + unsigned int cmd, unsigned long arg) +{ + int ret = 0; + struct snd_pcm *pcm = hw->private_data; + struct snd_pcm_mmap_fd __user *_mmap_fd = NULL; + struct snd_pcm_mmap_fd mmap_fd; + struct snd_pcm_substream *substream = NULL; + int32_t dir = -1; + + switch (cmd) { + case SNDRV_PCM_IOCTL_MMAP_DATA_FD: + _mmap_fd = (struct snd_pcm_mmap_fd __user *)arg; + if (get_user(dir, (int32_t __user *)&(_mmap_fd->dir))) { + pr_err("%s: error copying mmap_fd from user\n", + __func__); + ret = -EFAULT; + break; + } + if (dir != OUT && dir != IN) { + pr_err("%s invalid stream dir\n", __func__); + ret = -EINVAL; + break; + } + substream = pcm->streams[dir].substream; + if (!substream) { + pr_err("%s substream not found\n", __func__); + ret = -ENODEV; + break; + } + pr_debug("%s : %s MMAP Data fd\n", __func__, + dir == 0 ? "P" : "C"); + if (msm_pcm_mmap_fd(substream, &mmap_fd) < 0) { + pr_err("%s: error getting fd\n", + __func__); + ret = -EFAULT; + break; + } + if (put_user(mmap_fd.fd, &_mmap_fd->fd) || + put_user(mmap_fd.size, &_mmap_fd->size) || + put_user(mmap_fd.actual_size, &_mmap_fd->actual_size)) { + pr_err("%s: error copying fd\n", __func__); + return -EFAULT; + } + break; + default: + ret = -EINVAL; + break; + } + return ret; +} + +#ifdef CONFIG_COMPAT +static int msm_pcm_hwdep_compat_ioctl(struct snd_hwdep *hw, + struct file *file, + unsigned int cmd, + unsigned long arg) +{ + /* we only support mmap fd. Handling is common in both modes */ + return msm_pcm_hwdep_ioctl(hw, file, cmd, arg); +} +#else +static int msm_pcm_hwdep_compat_ioctl(struct snd_hwdep *hw, + struct file *file, + unsigned int cmd, + unsigned long arg) +{ + return -EINVAL; +} +#endif + +static int msm_pcm_add_hwdep_dev(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_hwdep *hwdep; + int rc; + char id[] = "NOIRQ_NN"; + + snprintf(id, sizeof(id), "NOIRQ_%d", runtime->pcm->device); + pr_debug("%s: pcm dev %d\n", __func__, runtime->pcm->device); + rc = snd_hwdep_new(runtime->card->snd_card, + &id[0], + HWDEP_FE_BASE + runtime->pcm->device, + &hwdep); + if (!hwdep || rc < 0) { + pr_err("%s: hwdep intf failed to create %s - hwdep\n", __func__, + id); + return rc; + } + + hwdep->iface = SNDRV_HWDEP_IFACE_AUDIO_BE; /* for lack of a FE iface */ + hwdep->private_data = runtime->pcm; /* of type struct snd_pcm */ + hwdep->ops.ioctl = msm_pcm_hwdep_ioctl; + hwdep->ops.ioctl_compat = msm_pcm_hwdep_compat_ioctl; + return 0; +} static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd) { @@ -1027,7 +1170,9 @@ static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd) pr_err("%s: Could not add app type controls failed %d\n", __func__, ret); } - + ret = msm_pcm_add_hwdep_dev(rtd); + if (ret) + pr_err("%s: Could not add hw dep node\n", __func__); pcm->nonatomic = true; exit: return ret; @@ -1040,6 +1185,9 @@ static struct snd_pcm_ops msm_pcm_ops = { .copy = msm_pcm_copy, .hw_params = msm_pcm_hw_params, .ioctl = msm_pcm_ioctl, +#ifdef CONFIG_COMPAT + .compat_ioctl = msm_pcm_compat_ioctl, +#endif .trigger = msm_pcm_trigger, .pointer = msm_pcm_pointer, .mmap = msm_pcm_mmap, diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c index 46a3324d2d6b..b94eb6fbfeea 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c @@ -1603,6 +1603,266 @@ done: return ret; } +static int msm_pcm_playback_pan_scale_ctl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int ret = 0; + int len = 0; + int i = 0; + struct snd_pcm_usr *usr_info = snd_kcontrol_chip(kcontrol); + struct snd_pcm_substream *substream; + struct msm_audio *prtd; + struct asm_stream_pan_ctrl_params pan_param; + + if (!usr_info) { + pr_err("%s: usr_info is null\n", __func__); + ret = -EINVAL; + goto done; + } + + substream = usr_info->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + if (!substream) { + pr_err("%s substream not found\n", __func__); + ret = -EINVAL; + goto done; + } + + if (!substream->runtime) { + pr_err("%s substream runtime not found\n", __func__); + ret = -EINVAL; + goto done; + } + prtd = substream->runtime->private_data; + if (!prtd) { + ret = -EINVAL; + goto done; + } + pan_param.num_output_channels = + ucontrol->value.integer.value[len++]; + if (pan_param.num_output_channels > + PCM_FORMAT_MAX_NUM_CHANNEL) { + ret = -EINVAL; + goto done; + } + pan_param.num_input_channels = + ucontrol->value.integer.value[len++]; + if (pan_param.num_input_channels > + PCM_FORMAT_MAX_NUM_CHANNEL) { + ret = -EINVAL; + goto done; + } + + if (ucontrol->value.integer.value[len++]) { + for (i = 0; i < pan_param.num_output_channels; i++) { + pan_param.output_channel_map[i] = + ucontrol->value.integer.value[len++]; + } + } + if (ucontrol->value.integer.value[len++]) { + for (i = 0; i < pan_param.num_input_channels; i++) { + pan_param.input_channel_map[i] = + ucontrol->value.integer.value[len++]; + } + } + if (ucontrol->value.integer.value[len++]) { + for (i = 0; i < pan_param.num_output_channels * + pan_param.num_input_channels; i++) { + pan_param.gain[i] = + !(ucontrol->value.integer.value[len++] > 0) ? + 0 : 2 << 13; + } + } + + ret = q6asm_set_mfc_panning_params(prtd->audio_client, + &pan_param); + len -= pan_param.num_output_channels * + pan_param.num_input_channels; + for (i = 0; i < pan_param.num_output_channels * + pan_param.num_input_channels; i++) { + /* + * The data userspace passes is already in Q14 format. + * For volume gain is in Q28. + */ + pan_param.gain[i] = + ucontrol->value.integer.value[len++] << 14; + } + ret = q6asm_set_vol_ctrl_gain_pair(prtd->audio_client, + &pan_param); + +done: + return ret; +} + +static int msm_pcm_playback_pan_scale_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return 0; +} + +static int msm_add_stream_pan_scale_controls(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm; + struct snd_pcm_usr *pan_ctl_info; + struct snd_kcontrol *kctl; + const char *playback_mixer_ctl_name = "Audio Stream"; + const char *deviceNo = "NN"; + const char *suffix = "Pan Scale Control"; + int ctl_len, ret = 0; + + if (!rtd) { + pr_err("%s: rtd is NULL\n", __func__); + ret = -EINVAL; + goto done; + } + + pcm = rtd->pcm; + ctl_len = strlen(playback_mixer_ctl_name) + 1 + strlen(deviceNo) + 1 + + strlen(suffix) + 1; + + ret = snd_pcm_add_usr_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, + NULL, 1, ctl_len, rtd->dai_link->be_id, + &pan_ctl_info); + + if (ret < 0) { + pr_err("%s: failed add ctl %s. err = %d\n", + __func__, suffix, ret); + goto done; + } + kctl = pan_ctl_info->kctl; + snprintf(kctl->id.name, ctl_len, "%s %d %s", playback_mixer_ctl_name, + rtd->pcm->device, suffix); + kctl->put = msm_pcm_playback_pan_scale_ctl_put; + kctl->get = msm_pcm_playback_pan_scale_ctl_get; + pr_debug("%s: Registering new mixer ctl = %s\n", __func__, + kctl->id.name); +done: + return ret; + +} + +static int msm_pcm_playback_dnmix_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return 0; +} + +static int msm_pcm_playback_dnmix_ctl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int ret = 0; + int len = 0; + int i = 0; + struct snd_pcm_usr *usr_info = snd_kcontrol_chip(kcontrol); + struct snd_pcm_substream *substream; + struct msm_audio *prtd; + struct asm_stream_pan_ctrl_params dnmix_param; + + int be_id = ucontrol->value.integer.value[len++]; + int stream_id = 0; + + if (!usr_info) { + pr_err("%s usr_info is null\n", __func__); + ret = -EINVAL; + goto done; + } + substream = usr_info->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + if (!substream) { + pr_err("%s substream not found\n", __func__); + ret = -EINVAL; + goto done; + } + if (!substream->runtime) { + pr_err("%s substream runtime not found\n", __func__); + ret = -EINVAL; + goto done; + } + prtd = substream->runtime->private_data; + if (!prtd) { + ret = -EINVAL; + goto done; + } + stream_id = prtd->audio_client->session; + dnmix_param.num_output_channels = + ucontrol->value.integer.value[len++]; + if (dnmix_param.num_output_channels > + PCM_FORMAT_MAX_NUM_CHANNEL) { + ret = -EINVAL; + goto done; + } + dnmix_param.num_input_channels = + ucontrol->value.integer.value[len++]; + if (dnmix_param.num_input_channels > + PCM_FORMAT_MAX_NUM_CHANNEL) { + ret = -EINVAL; + goto done; + } + + if (ucontrol->value.integer.value[len++]) { + for (i = 0; i < dnmix_param.num_output_channels; i++) { + dnmix_param.output_channel_map[i] = + ucontrol->value.integer.value[len++]; + } + } + if (ucontrol->value.integer.value[len++]) { + for (i = 0; i < dnmix_param.num_input_channels; i++) { + dnmix_param.input_channel_map[i] = + ucontrol->value.integer.value[len++]; + } + } + if (ucontrol->value.integer.value[len++]) { + for (i = 0; i < dnmix_param.num_output_channels * + dnmix_param.num_input_channels; i++) { + dnmix_param.gain[i] = + ucontrol->value.integer.value[len++]; + } + } + msm_routing_set_downmix_control_data(be_id, + stream_id, + &dnmix_param); + +done: + return ret; +} + +static int msm_add_device_down_mix_controls(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm; + struct snd_pcm_usr *usr_info; + struct snd_kcontrol *kctl; + const char *playback_mixer_ctl_name = "Audio Device"; + const char *deviceNo = "NN"; + const char *suffix = "Downmix Control"; + int ctl_len, ret = 0; + + if (!rtd) { + pr_err("%s: rtd is NULL\n", __func__); + ret = -EINVAL; + goto done; + } + + pcm = rtd->pcm; + ctl_len = strlen(playback_mixer_ctl_name) + 1 + + strlen(deviceNo) + 1 + strlen(suffix) + 1; + ret = snd_pcm_add_usr_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, + NULL, 1, ctl_len, rtd->dai_link->be_id, + &usr_info); + if (ret < 0) { + pr_err("%s: downmix control add failed: %d\n", + __func__, ret); + goto done; + } + + kctl = usr_info->kctl; + snprintf(kctl->id.name, ctl_len, "%s %d %s", + playback_mixer_ctl_name, rtd->pcm->device, suffix); + kctl->put = msm_pcm_playback_dnmix_ctl_put; + kctl->get = msm_pcm_playback_dnmix_ctl_get; + pr_debug("%s: downmix control name = %s\n", + __func__, playback_mixer_ctl_name); +done: + return ret; +} + static int msm_pcm_capture_app_type_cfg_ctl_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1719,6 +1979,14 @@ static int msm_pcm_add_controls(struct snd_soc_pcm_runtime *rtd) if (ret) pr_err("%s: pcm add app type controls failed:%d\n", __func__, ret); + ret = msm_add_stream_pan_scale_controls(rtd); + if (ret) + pr_err("%s: pcm add pan scale controls failed:%d\n", + __func__, ret); + ret = msm_add_device_down_mix_controls(rtd); + if (ret) + pr_err("%s: pcm add dnmix controls failed:%d\n", + __func__, ret); return ret; } diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c index 974d4a582540..7326e658c947 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c @@ -35,6 +35,8 @@ #include <sound/audio_cal_utils.h> #include <sound/audio_effects.h> #include <sound/hwdep.h> +#include <sound/q6adm-v2.h> +#include <sound/apr_audio-v2.h> #include "msm-pcm-routing-v2.h" #include "msm-pcm-routing-devdep.h" @@ -118,17 +120,13 @@ static const char * const lsm_port_text[] = { }; struct msm_pcm_route_bdai_pp_params { - u16 port_id; /* AFE port ID */ unsigned long pp_params_config; bool mute_on; int latency; }; static struct msm_pcm_route_bdai_pp_params - msm_bedais_pp_params[MSM_BACKEND_DAI_PP_PARAMS_REQ_MAX] = { - {HDMI_RX, 0, 0, 0}, - {DISPLAY_PORT_RX, 0, 0, 0}, -}; + msm_bedais_pp_params[MSM_BACKEND_DAI_MAX]; /* * The be_dai_name_table is passed to HAL so that it can specify the @@ -142,6 +140,7 @@ struct msm_pcm_route_bdai_name { }; static struct msm_pcm_route_bdai_name be_dai_name_table[MSM_BACKEND_DAI_MAX]; +static bool msm_pcm_routing_test_pp_param(int be_idx, long param_bit); static int msm_routing_send_device_pp_params(int port_id, int copp_idx, int fe_id); @@ -606,6 +605,27 @@ static struct msm_pcm_routing_fdai_data /* MULTIMEDIA20 */ {{0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} }, {0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} } }, + /* MULTIMEDIA21 */ + {{0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} }, + {0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} } }, + /* MULTIMEDIA22 */ + {{0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} }, + {0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} } }, + /* MULTIMEDIA23 */ + {{0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} }, + {0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} } }, + /* MULTIMEDIA24 */ + {{0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} }, + {0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} } }, + /* MULTIMEDIA25 */ + {{0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} }, + {0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} } }, + /* MULTIMEDIA26 */ + {{0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} }, + {0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} } }, + /* MULTIMEDIA27 */ + {{0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} }, + {0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} } }, /* CS_VOICE */ {{0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} }, {0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} } }, @@ -953,7 +973,7 @@ static void msm_pcm_routing_build_matrix(int fedai_id, int sess_type, uint32_t passthr_mode) { int i, port_type, j, num_copps = 0; - struct route_payload payload; + struct route_payload payload = { {0} }; port_type = ((path_type == ADM_PATH_PLAYBACK || path_type == ADM_PATH_COMPRESSED_RX) ? @@ -983,6 +1003,11 @@ static void msm_pcm_routing_build_matrix(int fedai_id, int sess_type, fe_dai_app_type_cfg [fedai_id][sess_type][i] .sample_rate; + if (msm_pcm_routing_test_pp_param(i, + ADM_PP_PARAM_LIMITER_BIT)) + set_bit(ADM_STATUS_LIMITER, + &payload.route_status + [num_copps]); num_copps++; } } @@ -1088,7 +1113,7 @@ int msm_pcm_routing_reg_phy_compr_stream(int fe_id, int perf_mode, port_type = MSM_AFE_PORT_TYPE_RX; } else if (stream_type == SNDRV_PCM_STREAM_CAPTURE) { session_type = SESSION_TYPE_TX; - if (passthr_mode != LEGACY_PCM) + if ((passthr_mode != LEGACY_PCM) && (passthr_mode != LISTEN)) path_type = ADM_PATH_COMPRESSED_TX; else path_type = ADM_PATH_LIVE_REC; @@ -1204,6 +1229,11 @@ int msm_pcm_routing_reg_phy_compr_stream(int fe_id, int perf_mode, fe_dai_app_type_cfg [fe_id][session_type][i] .sample_rate; + if (msm_pcm_routing_test_pp_param(i, + ADM_PP_PARAM_LIMITER_BIT)) + set_bit(ADM_STATUS_LIMITER, + &payload.route_status + [num_copps]); num_copps++; } } @@ -1434,6 +1464,11 @@ int msm_pcm_routing_reg_phy_stream(int fedai_id, int perf_mode, fe_dai_app_type_cfg [fedai_id][session_type] [i].sample_rate; + if (msm_pcm_routing_test_pp_param(i, + ADM_PP_PARAM_LIMITER_BIT)) + set_bit(ADM_STATUS_LIMITER, + &payload.route_status + [num_copps]); num_copps++; } } @@ -3640,6 +3675,11 @@ static const struct snd_kcontrol_new ext_ec_ref_mux_ul9 = msm_route_ec_ref_rx_enum[0], msm_routing_ec_ref_rx_get, msm_routing_ec_ref_rx_put); +static const struct snd_kcontrol_new ext_ec_ref_mux_ul16 = + SOC_DAPM_ENUM_EXT("AUDIO_REF_EC_UL16 MUX Mux", + msm_route_ec_ref_rx_enum[0], + msm_routing_ec_ref_rx_get, msm_routing_ec_ref_rx_put); + static const struct snd_kcontrol_new ext_ec_ref_mux_ul17 = SOC_DAPM_ENUM_EXT("AUDIO_REF_EC_UL17 MUX Mux", msm_route_ec_ref_rx_enum[0], @@ -3798,6 +3838,9 @@ static const struct snd_kcontrol_new pri_i2s_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_PRI_I2S_RX, MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_PRI_I2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new sec_i2s_rx_mixer_controls[] = { @@ -3858,6 +3901,9 @@ static const struct snd_kcontrol_new sec_i2s_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_SEC_I2S_RX, MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_SEC_I2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new spdif_rx_mixer_controls[] = { @@ -3918,7 +3964,9 @@ static const struct snd_kcontrol_new spdif_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_SPDIF_RX, MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), - + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_SPDIF_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new slimbus_2_rx_mixer_controls[] = { @@ -3970,6 +4018,9 @@ static const struct snd_kcontrol_new slimbus_2_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_SLIMBUS_2_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_SLIMBUS_2_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new slimbus_5_rx_mixer_controls[] = { @@ -4030,6 +4081,9 @@ static const struct snd_kcontrol_new slimbus_5_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_SLIMBUS_5_RX, MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_SLIMBUS_5_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = { @@ -4090,6 +4144,24 @@ static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_SLIMBUS_0_RX, MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_SLIMBUS_0_RX, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia22", MSM_BACKEND_DAI_SLIMBUS_0_RX, + MSM_FRONTEND_DAI_MULTIMEDIA22, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia23", MSM_BACKEND_DAI_SLIMBUS_0_RX, + MSM_FRONTEND_DAI_MULTIMEDIA23, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia24", MSM_BACKEND_DAI_SLIMBUS_0_RX, + MSM_FRONTEND_DAI_MULTIMEDIA24, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia25", MSM_BACKEND_DAI_SLIMBUS_0_RX, + MSM_FRONTEND_DAI_MULTIMEDIA25, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_SLIMBUS_0_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new mi2s_rx_mixer_controls[] = { @@ -4150,6 +4222,9 @@ static const struct snd_kcontrol_new mi2s_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_MI2S_RX, MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new quaternary_mi2s_rx_mixer_controls[] = { @@ -4210,6 +4285,9 @@ static const struct snd_kcontrol_new quaternary_mi2s_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_QUATERNARY_MI2S_RX, MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_QUATERNARY_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new quinary_mi2s_rx_mixer_controls[] = { @@ -4270,6 +4348,9 @@ static const struct snd_kcontrol_new quinary_mi2s_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_QUINARY_MI2S_RX, MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_QUINARY_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new tertiary_mi2s_rx_mixer_controls[] = { @@ -4324,6 +4405,9 @@ static const struct snd_kcontrol_new tertiary_mi2s_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_TERTIARY_MI2S_RX, MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_TERTIARY_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new secondary_mi2s_rx2_mixer_controls[] = { @@ -4390,6 +4474,9 @@ static const struct snd_kcontrol_new secondary_mi2s_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_SECONDARY_MI2S_RX, MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_SECONDARY_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new primary_mi2s_rx_mixer_controls[] = { @@ -4450,6 +4537,9 @@ static const struct snd_kcontrol_new primary_mi2s_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_PRI_MI2S_RX, MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_PRI_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new int0_mi2s_rx_mixer_controls[] = { @@ -4501,6 +4591,9 @@ static const struct snd_kcontrol_new int0_mi2s_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_INT0_MI2S_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_INT0_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new int4_mi2s_rx_mixer_controls[] = { @@ -4552,6 +4645,9 @@ static const struct snd_kcontrol_new int4_mi2s_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_INT4_MI2S_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_INT4_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new hdmi_mixer_controls[] = { @@ -4612,6 +4708,24 @@ static const struct snd_kcontrol_new hdmi_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_HDMI_RX, MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_HDMI_RX, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia22", MSM_BACKEND_DAI_HDMI_RX, + MSM_FRONTEND_DAI_MULTIMEDIA22, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia23", MSM_BACKEND_DAI_HDMI_RX, + MSM_FRONTEND_DAI_MULTIMEDIA23, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia24", MSM_BACKEND_DAI_HDMI_RX, + MSM_FRONTEND_DAI_MULTIMEDIA24, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia25", MSM_BACKEND_DAI_HDMI_RX, + MSM_FRONTEND_DAI_MULTIMEDIA25, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_HDMI_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new display_port_mixer_controls[] = { @@ -4663,6 +4777,9 @@ static const struct snd_kcontrol_new display_port_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_DISPLAY_PORT_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_DISPLAY_PORT_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; /* incall music delivery mixer */ @@ -4760,6 +4877,24 @@ static const struct snd_kcontrol_new slimbus_6_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_SLIMBUS_6_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_SLIMBUS_6_RX, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia22", MSM_BACKEND_DAI_SLIMBUS_6_RX, + MSM_FRONTEND_DAI_MULTIMEDIA22, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia23", MSM_BACKEND_DAI_SLIMBUS_6_RX, + MSM_FRONTEND_DAI_MULTIMEDIA23, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia24", MSM_BACKEND_DAI_SLIMBUS_6_RX, + MSM_FRONTEND_DAI_MULTIMEDIA24, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia25", MSM_BACKEND_DAI_SLIMBUS_6_RX, + MSM_FRONTEND_DAI_MULTIMEDIA25, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_SLIMBUS_6_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new slimbus_7_rx_mixer_controls[] = { @@ -4811,6 +4946,24 @@ static const struct snd_kcontrol_new slimbus_7_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_SLIMBUS_7_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_SLIMBUS_7_RX, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia22", MSM_BACKEND_DAI_SLIMBUS_7_RX, + MSM_FRONTEND_DAI_MULTIMEDIA22, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia23", MSM_BACKEND_DAI_SLIMBUS_7_RX, + MSM_FRONTEND_DAI_MULTIMEDIA23, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia24", MSM_BACKEND_DAI_SLIMBUS_7_RX, + MSM_FRONTEND_DAI_MULTIMEDIA24, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia25", MSM_BACKEND_DAI_SLIMBUS_7_RX, + MSM_FRONTEND_DAI_MULTIMEDIA25, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_SLIMBUS_7_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new usb_audio_rx_mixer_controls[] = { @@ -4862,6 +5015,9 @@ static const struct snd_kcontrol_new usb_audio_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_USB_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_USB_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new int_bt_sco_rx_mixer_controls[] = { @@ -4922,6 +5078,9 @@ static const struct snd_kcontrol_new int_bt_sco_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_INT_BT_SCO_RX, MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_INT_BT_SCO_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new int_bt_a2dp_rx_mixer_controls[] = { @@ -4973,6 +5132,9 @@ static const struct snd_kcontrol_new int_bt_a2dp_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_INT_BT_A2DP_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_INT_BT_A2DP_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new int_fm_rx_mixer_controls[] = { @@ -5033,6 +5195,9 @@ static const struct snd_kcontrol_new int_fm_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_INT_FM_RX, MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_INT_FM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new afe_pcm_rx_mixer_controls[] = { @@ -5093,6 +5258,9 @@ static const struct snd_kcontrol_new afe_pcm_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_AFE_PCM_RX, MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_AFE_PCM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new auxpcm_rx_mixer_controls[] = { @@ -5153,6 +5321,12 @@ static const struct snd_kcontrol_new auxpcm_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_AUXPCM_RX, MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_AUXPCM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_AUXPCM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new sec_auxpcm_rx_mixer_controls[] = { @@ -5213,6 +5387,12 @@ static const struct snd_kcontrol_new sec_auxpcm_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_SEC_AUXPCM_RX, MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_SEC_AUXPCM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_SEC_AUXPCM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new tert_auxpcm_rx_mixer_controls[] = { @@ -5264,6 +5444,9 @@ static const struct snd_kcontrol_new tert_auxpcm_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_TERT_AUXPCM_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_TERT_AUXPCM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new quat_auxpcm_rx_mixer_controls[] = { @@ -5315,6 +5498,9 @@ static const struct snd_kcontrol_new quat_auxpcm_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_QUAT_AUXPCM_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_QUAT_AUXPCM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new pri_tdm_rx_0_mixer_controls[] = { @@ -5366,6 +5552,12 @@ static const struct snd_kcontrol_new pri_tdm_rx_0_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_PRI_TDM_RX_0, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_PRI_TDM_RX_0, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_PRI_TDM_RX_0, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new pri_tdm_rx_1_mixer_controls[] = { @@ -5417,6 +5609,12 @@ static const struct snd_kcontrol_new pri_tdm_rx_1_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_PRI_TDM_RX_1, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_PRI_TDM_RX_1, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_PRI_TDM_RX_1, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new pri_tdm_rx_2_mixer_controls[] = { @@ -5468,6 +5666,12 @@ static const struct snd_kcontrol_new pri_tdm_rx_2_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_PRI_TDM_RX_2, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_PRI_TDM_RX_2, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_PRI_TDM_RX_2, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new pri_tdm_rx_3_mixer_controls[] = { @@ -5519,6 +5723,12 @@ static const struct snd_kcontrol_new pri_tdm_rx_3_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_PRI_TDM_RX_3, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_PRI_TDM_RX_3, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_PRI_TDM_RX_3, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new pri_tdm_tx_0_mixer_controls[] = { @@ -5621,6 +5831,12 @@ static const struct snd_kcontrol_new sec_tdm_rx_0_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_SEC_TDM_RX_0, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_SEC_TDM_RX_0, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_SEC_TDM_RX_0, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new sec_tdm_rx_1_mixer_controls[] = { @@ -5672,6 +5888,12 @@ static const struct snd_kcontrol_new sec_tdm_rx_1_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_SEC_TDM_RX_1, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_SEC_TDM_RX_1, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_SEC_TDM_RX_1, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new sec_tdm_rx_2_mixer_controls[] = { @@ -5723,6 +5945,12 @@ static const struct snd_kcontrol_new sec_tdm_rx_2_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_SEC_TDM_RX_2, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_SEC_TDM_RX_2, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_SEC_TDM_RX_2, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new sec_tdm_rx_3_mixer_controls[] = { @@ -5774,6 +6002,12 @@ static const struct snd_kcontrol_new sec_tdm_rx_3_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_SEC_TDM_RX_3, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_SEC_TDM_RX_3, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_SEC_TDM_RX_3, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new sec_tdm_tx_0_mixer_controls[] = { @@ -5876,6 +6110,12 @@ static const struct snd_kcontrol_new tert_tdm_rx_0_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_TERT_TDM_RX_0, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_TERT_TDM_RX_0, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_TERT_TDM_RX_0, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new tert_tdm_tx_0_mixer_controls[] = { @@ -5978,6 +6218,12 @@ static const struct snd_kcontrol_new tert_tdm_rx_1_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_TERT_TDM_RX_1, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_TERT_TDM_RX_1, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_TERT_TDM_RX_1, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new tert_tdm_rx_2_mixer_controls[] = { @@ -6029,6 +6275,12 @@ static const struct snd_kcontrol_new tert_tdm_rx_2_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_TERT_TDM_RX_2, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_TERT_TDM_RX_2, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_TERT_TDM_RX_2, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new tert_tdm_rx_3_mixer_controls[] = { @@ -6080,6 +6332,12 @@ static const struct snd_kcontrol_new tert_tdm_rx_3_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_TERT_TDM_RX_3, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_TERT_TDM_RX_3, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_TERT_TDM_RX_3, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new tert_tdm_rx_4_mixer_controls[] = { @@ -6131,6 +6389,12 @@ static const struct snd_kcontrol_new tert_tdm_rx_4_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_TERT_TDM_RX_4, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_TERT_TDM_RX_4, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_TERT_TDM_RX_4, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new quat_tdm_rx_0_mixer_controls[] = { @@ -6185,6 +6449,12 @@ static const struct snd_kcontrol_new quat_tdm_rx_0_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia20", MSM_BACKEND_DAI_QUAT_TDM_RX_0, MSM_FRONTEND_DAI_MULTIMEDIA20, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_QUAT_TDM_RX_0, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_QUAT_TDM_RX_0, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new quat_tdm_tx_0_mixer_controls[] = { @@ -6290,6 +6560,12 @@ static const struct snd_kcontrol_new quat_tdm_rx_1_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia20", MSM_BACKEND_DAI_QUAT_TDM_RX_1, MSM_FRONTEND_DAI_MULTIMEDIA20, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_QUAT_TDM_RX_1, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_QUAT_TDM_RX_1, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new quat_tdm_rx_2_mixer_controls[] = { @@ -6344,6 +6620,12 @@ static const struct snd_kcontrol_new quat_tdm_rx_2_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia20", MSM_BACKEND_DAI_QUAT_TDM_RX_2, MSM_FRONTEND_DAI_MULTIMEDIA20, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_QUAT_TDM_RX_2, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_QUAT_TDM_RX_2, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new quat_tdm_rx_3_mixer_controls[] = { @@ -6398,6 +6680,12 @@ static const struct snd_kcontrol_new quat_tdm_rx_3_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia20", MSM_BACKEND_DAI_QUAT_TDM_RX_3, MSM_FRONTEND_DAI_MULTIMEDIA20, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia21", MSM_BACKEND_DAI_QUAT_TDM_RX_3, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia26", MSM_BACKEND_DAI_QUAT_TDM_RX_3, + MSM_FRONTEND_DAI_MULTIMEDIA26, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new mmul1_mixer_controls[] = { @@ -7087,6 +7375,114 @@ static const struct snd_kcontrol_new mmul8_mixer_controls[] = { msm_routing_put_audio_mixer), }; +static const struct snd_kcontrol_new mmul16_mixer_controls[] = { + SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_SLIMBUS_0_TX, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("PRI_MI2S_TX", MSM_BACKEND_DAI_PRI_MI2S_TX, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("SEC_MI2S_TX", MSM_BACKEND_DAI_SECONDARY_MI2S_TX, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("TERT_MI2S_TX", MSM_BACKEND_DAI_TERTIARY_MI2S_TX, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("INT2_MI2S_TX", MSM_BACKEND_DAI_INT2_MI2S_TX, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("INT3_MI2S_TX", MSM_BACKEND_DAI_INT3_MI2S_TX, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("QUAT_MI2S_TX", MSM_BACKEND_DAI_QUATERNARY_MI2S_TX, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("INTERNAL_FM_TX", MSM_BACKEND_DAI_INT_FM_TX, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX", MSM_BACKEND_DAI_INT_BT_SCO_TX, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("AFE_PCM_TX", MSM_BACKEND_DAI_AFE_PCM_TX, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("VOC_REC_DL", MSM_BACKEND_DAI_INCALL_RECORD_RX, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("VOC_REC_UL", MSM_BACKEND_DAI_INCALL_RECORD_TX, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("SLIM_6_TX", MSM_BACKEND_DAI_SLIMBUS_6_TX, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_0", MSM_BACKEND_DAI_PRI_TDM_TX_0, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_1", MSM_BACKEND_DAI_PRI_TDM_TX_1, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_2", MSM_BACKEND_DAI_PRI_TDM_TX_2, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_3", MSM_BACKEND_DAI_PRI_TDM_TX_3, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("SEC_TDM_TX_0", MSM_BACKEND_DAI_SEC_TDM_TX_0, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("SEC_TDM_TX_1", MSM_BACKEND_DAI_SEC_TDM_TX_1, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("SEC_TDM_TX_2", MSM_BACKEND_DAI_SEC_TDM_TX_2, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("SEC_TDM_TX_3", MSM_BACKEND_DAI_SEC_TDM_TX_3, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("TERT_TDM_TX_0", MSM_BACKEND_DAI_TERT_TDM_TX_0, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("TERT_TDM_TX_1", MSM_BACKEND_DAI_TERT_TDM_TX_1, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("TERT_TDM_TX_2", MSM_BACKEND_DAI_TERT_TDM_TX_2, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("TERT_TDM_TX_3", MSM_BACKEND_DAI_TERT_TDM_TX_3, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("QUAT_TDM_TX_0", MSM_BACKEND_DAI_QUAT_TDM_TX_0, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("QUAT_TDM_TX_1", MSM_BACKEND_DAI_QUAT_TDM_TX_1, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("QUAT_TDM_TX_2", MSM_BACKEND_DAI_QUAT_TDM_TX_2, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("QUAT_TDM_TX_3", MSM_BACKEND_DAI_QUAT_TDM_TX_3, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("SLIM_7_TX", MSM_BACKEND_DAI_SLIMBUS_7_TX, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("USB_AUDIO_TX", MSM_BACKEND_DAI_USB_TX, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MI2S_TX", MSM_BACKEND_DAI_MI2S_TX, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("AUX_PCM_TX", MSM_BACKEND_DAI_AUXPCM_TX, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("SEC_AUX_PCM_TX", MSM_BACKEND_DAI_SEC_AUXPCM_TX, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("QUAT_AUX_PCM_TX", MSM_BACKEND_DAI_QUAT_AUXPCM_TX, + MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), +}; + static const struct snd_kcontrol_new mmul9_mixer_controls[] = { SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_SLIMBUS_0_TX, MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer, @@ -7276,6 +7672,75 @@ static const struct snd_kcontrol_new mmul20_mixer_controls[] = { msm_routing_put_audio_mixer), }; +static const struct snd_kcontrol_new mmul21_mixer_controls[] = { + SOC_SINGLE_EXT("AUX_PCM_UL_TX", MSM_BACKEND_DAI_AUXPCM_TX, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("SEC_AUX_PCM_UL_TX", MSM_BACKEND_DAI_SEC_AUXPCM_TX, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_0", MSM_BACKEND_DAI_PRI_TDM_TX_0, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_1", MSM_BACKEND_DAI_PRI_TDM_TX_1, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_2", MSM_BACKEND_DAI_PRI_TDM_TX_2, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_3", MSM_BACKEND_DAI_PRI_TDM_TX_3, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("SEC_TDM_TX_0", MSM_BACKEND_DAI_SEC_TDM_TX_0, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("SEC_TDM_TX_1", MSM_BACKEND_DAI_SEC_TDM_TX_1, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("SEC_TDM_TX_2", MSM_BACKEND_DAI_SEC_TDM_TX_2, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("SEC_TDM_TX_3", MSM_BACKEND_DAI_SEC_TDM_TX_3, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("TERT_TDM_TX_0", MSM_BACKEND_DAI_TERT_TDM_TX_0, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("TERT_TDM_TX_1", MSM_BACKEND_DAI_TERT_TDM_TX_1, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("TERT_TDM_TX_2", MSM_BACKEND_DAI_TERT_TDM_TX_2, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("TERT_TDM_TX_3", MSM_BACKEND_DAI_TERT_TDM_TX_3, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("QUAT_TDM_TX_0", MSM_BACKEND_DAI_QUAT_TDM_TX_0, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("QUAT_TDM_TX_1", MSM_BACKEND_DAI_QUAT_TDM_TX_1, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("QUAT_TDM_TX_2", MSM_BACKEND_DAI_QUAT_TDM_TX_2, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("QUAT_TDM_TX_3", MSM_BACKEND_DAI_QUAT_TDM_TX_3, + MSM_FRONTEND_DAI_MULTIMEDIA21, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), +}; + +static const struct snd_kcontrol_new mmul27_mixer_controls[] = { + SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_SLIMBUS_0_TX, + MSM_FRONTEND_DAI_MULTIMEDIA27, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("PRI_MI2S_TX", MSM_BACKEND_DAI_PRI_MI2S_TX, + MSM_FRONTEND_DAI_MULTIMEDIA27, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("QUAT_MI2S_TX", MSM_BACKEND_DAI_QUATERNARY_MI2S_TX, + MSM_FRONTEND_DAI_MULTIMEDIA27, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), +}; + static const struct snd_kcontrol_new pri_rx_voice_mixer_controls[] = { SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_PRI_I2S_RX, MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer, @@ -9797,6 +10262,22 @@ static const struct snd_kcontrol_new quat_tdm_rx_0_port_mixer_controls[] = { MSM_BACKEND_DAI_SEC_AUXPCM_TX, 1, 0, msm_routing_get_port_mixer, msm_routing_put_port_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_0", MSM_BACKEND_DAI_QUAT_TDM_RX_0, + MSM_BACKEND_DAI_PRI_TDM_TX_0, 1, 0, + msm_routing_get_port_mixer, + msm_routing_put_port_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_1", MSM_BACKEND_DAI_QUAT_TDM_RX_0, + MSM_BACKEND_DAI_PRI_TDM_TX_1, 1, 0, + msm_routing_get_port_mixer, + msm_routing_put_port_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_2", MSM_BACKEND_DAI_QUAT_TDM_RX_0, + MSM_BACKEND_DAI_PRI_TDM_TX_2, 1, 0, + msm_routing_get_port_mixer, + msm_routing_put_port_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_3", MSM_BACKEND_DAI_QUAT_TDM_RX_0, + MSM_BACKEND_DAI_PRI_TDM_TX_3, 1, 0, + msm_routing_get_port_mixer, + msm_routing_put_port_mixer), SOC_SINGLE_EXT("TERT_TDM_TX_0", MSM_BACKEND_DAI_QUAT_TDM_RX_0, MSM_BACKEND_DAI_TERT_TDM_TX_0, 1, 0, msm_routing_get_port_mixer, @@ -9864,6 +10345,22 @@ static const struct snd_kcontrol_new quat_tdm_rx_1_port_mixer_controls[] = { MSM_BACKEND_DAI_SEC_AUXPCM_TX, 1, 0, msm_routing_get_port_mixer, msm_routing_put_port_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_0", MSM_BACKEND_DAI_QUAT_TDM_RX_1, + MSM_BACKEND_DAI_PRI_TDM_TX_0, 1, 0, + msm_routing_get_port_mixer, + msm_routing_put_port_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_1", MSM_BACKEND_DAI_QUAT_TDM_RX_1, + MSM_BACKEND_DAI_PRI_TDM_TX_1, 1, 0, + msm_routing_get_port_mixer, + msm_routing_put_port_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_2", MSM_BACKEND_DAI_QUAT_TDM_RX_1, + MSM_BACKEND_DAI_PRI_TDM_TX_2, 1, 0, + msm_routing_get_port_mixer, + msm_routing_put_port_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_3", MSM_BACKEND_DAI_QUAT_TDM_RX_1, + MSM_BACKEND_DAI_PRI_TDM_TX_3, 1, 0, + msm_routing_get_port_mixer, + msm_routing_put_port_mixer), SOC_SINGLE_EXT("TERT_TDM_TX_0", MSM_BACKEND_DAI_QUAT_TDM_RX_1, MSM_BACKEND_DAI_TERT_TDM_TX_0, 1, 0, msm_routing_get_port_mixer, @@ -9931,6 +10428,22 @@ static const struct snd_kcontrol_new quat_tdm_rx_2_port_mixer_controls[] = { MSM_BACKEND_DAI_SEC_AUXPCM_TX, 1, 0, msm_routing_get_port_mixer, msm_routing_put_port_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_0", MSM_BACKEND_DAI_QUAT_TDM_RX_2, + MSM_BACKEND_DAI_PRI_TDM_TX_0, 1, 0, + msm_routing_get_port_mixer, + msm_routing_put_port_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_1", MSM_BACKEND_DAI_QUAT_TDM_RX_2, + MSM_BACKEND_DAI_PRI_TDM_TX_1, 1, 0, + msm_routing_get_port_mixer, + msm_routing_put_port_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_2", MSM_BACKEND_DAI_QUAT_TDM_RX_2, + MSM_BACKEND_DAI_PRI_TDM_TX_2, 1, 0, + msm_routing_get_port_mixer, + msm_routing_put_port_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_3", MSM_BACKEND_DAI_QUAT_TDM_RX_2, + MSM_BACKEND_DAI_PRI_TDM_TX_3, 1, 0, + msm_routing_get_port_mixer, + msm_routing_put_port_mixer), SOC_SINGLE_EXT("TERT_TDM_TX_0", MSM_BACKEND_DAI_QUAT_TDM_RX_2, MSM_BACKEND_DAI_TERT_TDM_TX_0, 1, 0, msm_routing_get_port_mixer, @@ -9998,6 +10511,22 @@ static const struct snd_kcontrol_new quat_tdm_rx_3_port_mixer_controls[] = { MSM_BACKEND_DAI_SEC_AUXPCM_TX, 1, 0, msm_routing_get_port_mixer, msm_routing_put_port_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_0", MSM_BACKEND_DAI_QUAT_TDM_RX_3, + MSM_BACKEND_DAI_PRI_TDM_TX_0, 1, 0, + msm_routing_get_port_mixer, + msm_routing_put_port_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_1", MSM_BACKEND_DAI_QUAT_TDM_RX_3, + MSM_BACKEND_DAI_PRI_TDM_TX_1, 1, 0, + msm_routing_get_port_mixer, + msm_routing_put_port_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_2", MSM_BACKEND_DAI_QUAT_TDM_RX_3, + MSM_BACKEND_DAI_PRI_TDM_TX_2, 1, 0, + msm_routing_get_port_mixer, + msm_routing_put_port_mixer), + SOC_SINGLE_EXT("PRI_TDM_TX_3", MSM_BACKEND_DAI_QUAT_TDM_RX_3, + MSM_BACKEND_DAI_PRI_TDM_TX_3, 1, 0, + msm_routing_get_port_mixer, + msm_routing_put_port_mixer), SOC_SINGLE_EXT("TERT_TDM_TX_0", MSM_BACKEND_DAI_QUAT_TDM_RX_3, MSM_BACKEND_DAI_TERT_TDM_TX_0, 1, 0, msm_routing_get_port_mixer, @@ -11314,6 +11843,12 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { SND_SOC_DAPM_AIF_IN("MM_DL15", "MultiMedia15 Playback", 0, 0, 0, 0), SND_SOC_DAPM_AIF_IN("MM_DL16", "MultiMedia16 Playback", 0, 0, 0, 0), SND_SOC_DAPM_AIF_IN("MM_DL20", "MultiMedia20 Playback", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL21", "MultiMedia21 Playback", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL22", "MultiMedia22 Playback", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL23", "MultiMedia23 Playback", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL24", "MultiMedia24 Playback", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL25", "MultiMedia25 Playback", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL26", "MultiMedia26 Playback", 0, 0, 0, 0), SND_SOC_DAPM_AIF_IN("VOIP_DL", "VoIP Playback", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, 0, 0, 0), @@ -11323,10 +11858,13 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { SND_SOC_DAPM_AIF_OUT("MM_UL6", "MultiMedia6 Capture", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("MM_UL8", "MultiMedia8 Capture", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("MM_UL9", "MultiMedia9 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL16", "MultiMedia16 Capture", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("MM_UL17", "MultiMedia17 Capture", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("MM_UL18", "MultiMedia18 Capture", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("MM_UL19", "MultiMedia19 Capture", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("MM_UL20", "MultiMedia20 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL21", "MultiMedia21 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL27", "MultiMedia27 Capture", 0, 0, 0, 0), SND_SOC_DAPM_AIF_IN("CS-VOICE_DL1", "CS-VOICE Playback", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("CS-VOICE_UL1", "CS-VOICE Capture", 0, 0, 0, 0), SND_SOC_DAPM_AIF_IN("VOICE2_DL", "Voice2 Playback", 0, 0, 0, 0), @@ -12059,6 +12597,8 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { mmul8_mixer_controls, ARRAY_SIZE(mmul8_mixer_controls)), SND_SOC_DAPM_MIXER("MultiMedia9 Mixer", SND_SOC_NOPM, 0, 0, mmul9_mixer_controls, ARRAY_SIZE(mmul9_mixer_controls)), + SND_SOC_DAPM_MIXER("MultiMedia16 Mixer", SND_SOC_NOPM, 0, 0, + mmul16_mixer_controls, ARRAY_SIZE(mmul16_mixer_controls)), SND_SOC_DAPM_MIXER("MultiMedia17 Mixer", SND_SOC_NOPM, 0, 0, mmul17_mixer_controls, ARRAY_SIZE(mmul17_mixer_controls)), SND_SOC_DAPM_MIXER("MultiMedia18 Mixer", SND_SOC_NOPM, 0, 0, @@ -12067,6 +12607,10 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { mmul19_mixer_controls, ARRAY_SIZE(mmul19_mixer_controls)), SND_SOC_DAPM_MIXER("MultiMedia20 Mixer", SND_SOC_NOPM, 0, 0, mmul20_mixer_controls, ARRAY_SIZE(mmul20_mixer_controls)), + SND_SOC_DAPM_MIXER("MultiMedia21 Mixer", SND_SOC_NOPM, 0, 0, + mmul21_mixer_controls, ARRAY_SIZE(mmul21_mixer_controls)), + SND_SOC_DAPM_MIXER("MultiMedia27 Mixer", SND_SOC_NOPM, 0, 0, + mmul27_mixer_controls, ARRAY_SIZE(mmul27_mixer_controls)), SND_SOC_DAPM_MIXER("AUX_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0, auxpcm_rx_mixer_controls, ARRAY_SIZE(auxpcm_rx_mixer_controls)), SND_SOC_DAPM_MIXER("SEC_AUX_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0, @@ -12389,6 +12933,8 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { &ext_ec_ref_mux_ul8), SND_SOC_DAPM_MUX("AUDIO_REF_EC_UL9 MUX", SND_SOC_NOPM, 0, 0, &ext_ec_ref_mux_ul9), + SND_SOC_DAPM_MUX("AUDIO_REF_EC_UL16 MUX", SND_SOC_NOPM, 0, 0, + &ext_ec_ref_mux_ul16), SND_SOC_DAPM_MUX("AUDIO_REF_EC_UL17 MUX", SND_SOC_NOPM, 0, 0, &ext_ec_ref_mux_ul17), SND_SOC_DAPM_MUX("AUDIO_REF_EC_UL18 MUX", SND_SOC_NOPM, 0, 0, @@ -12414,6 +12960,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"PRI_RX Audio Mixer", "MultiMedia14", "MM_DL14"}, {"PRI_RX Audio Mixer", "MultiMedia15", "MM_DL15"}, {"PRI_RX Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"PRI_RX Audio Mixer", "MultiMedia26", "MM_DL26"}, {"PRI_I2S_RX", NULL, "PRI_RX Audio Mixer"}, {"SEC_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12432,6 +12979,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"SEC_RX Audio Mixer", "MultiMedia14", "MM_DL14"}, {"SEC_RX Audio Mixer", "MultiMedia15", "MM_DL15"}, {"SEC_RX Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"SEC_RX Audio Mixer", "MultiMedia26", "MM_DL26"}, {"SEC_I2S_RX", NULL, "SEC_RX Audio Mixer"}, {"SLIMBUS_0_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12450,6 +12998,12 @@ static const struct snd_soc_dapm_route intercon[] = { {"SLIMBUS_0_RX Audio Mixer", "MultiMedia14", "MM_DL14"}, {"SLIMBUS_0_RX Audio Mixer", "MultiMedia15", "MM_DL15"}, {"SLIMBUS_0_RX Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"SLIMBUS_0_RX Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"SLIMBUS_0_RX Audio Mixer", "MultiMedia22", "MM_DL22"}, + {"SLIMBUS_0_RX Audio Mixer", "MultiMedia23", "MM_DL23"}, + {"SLIMBUS_0_RX Audio Mixer", "MultiMedia24", "MM_DL24"}, + {"SLIMBUS_0_RX Audio Mixer", "MultiMedia25", "MM_DL25"}, + {"SLIMBUS_0_RX Audio Mixer", "MultiMedia26", "MM_DL26"}, {"SLIMBUS_0_RX", NULL, "SLIMBUS_0_RX Audio Mixer"}, {"SLIMBUS_2_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12468,6 +13022,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"SLIMBUS_2_RX Audio Mixer", "MultiMedia14", "MM_DL14"}, {"SLIMBUS_2_RX Audio Mixer", "MultiMedia15", "MM_DL15"}, {"SLIMBUS_2_RX Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"SLIMBUS_2_RX Audio Mixer", "MultiMedia26", "MM_DL26"}, {"SLIMBUS_2_RX", NULL, "SLIMBUS_2_RX Audio Mixer"}, {"SLIMBUS_5_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12486,6 +13041,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"SLIMBUS_5_RX Audio Mixer", "MultiMedia14", "MM_DL14"}, {"SLIMBUS_5_RX Audio Mixer", "MultiMedia15", "MM_DL15"}, {"SLIMBUS_5_RX Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"SLIMBUS_5_RX Audio Mixer", "MultiMedia26", "MM_DL26"}, {"SLIMBUS_5_RX", NULL, "SLIMBUS_5_RX Audio Mixer"}, {"HDMI Mixer", "MultiMedia1", "MM_DL1"}, @@ -12504,6 +13060,12 @@ static const struct snd_soc_dapm_route intercon[] = { {"HDMI Mixer", "MultiMedia14", "MM_DL14"}, {"HDMI Mixer", "MultiMedia15", "MM_DL15"}, {"HDMI Mixer", "MultiMedia16", "MM_DL16"}, + {"HDMI Mixer", "MultiMedia21", "MM_DL21"}, + {"HDMI Mixer", "MultiMedia22", "MM_DL22"}, + {"HDMI Mixer", "MultiMedia23", "MM_DL23"}, + {"HDMI Mixer", "MultiMedia24", "MM_DL24"}, + {"HDMI Mixer", "MultiMedia25", "MM_DL25"}, + {"HDMI Mixer", "MultiMedia26", "MM_DL26"}, {"HDMI", NULL, "HDMI Mixer"}, {"DISPLAY_PORT Mixer", "MultiMedia1", "MM_DL1"}, @@ -12522,6 +13084,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"DISPLAY_PORT Mixer", "MultiMedia14", "MM_DL14"}, {"DISPLAY_PORT Mixer", "MultiMedia15", "MM_DL15"}, {"DISPLAY_PORT Mixer", "MultiMedia16", "MM_DL16"}, + {"DISPLAY_PORT Mixer", "MultiMedia26", "MM_DL26"}, {"DISPLAY_PORT", NULL, "DISPLAY_PORT Mixer"}, {"SPDIF_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12540,6 +13103,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"SPDIF_RX Audio Mixer", "MultiMedia14", "MM_DL14"}, {"SPDIF_RX Audio Mixer", "MultiMedia15", "MM_DL15"}, {"SPDIF_RX Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"SPDIF_RX Audio Mixer", "MultiMedia26", "MM_DL26"}, {"SPDIF_RX", NULL, "SPDIF_RX Audio Mixer"}, /* incall */ @@ -12575,6 +13139,12 @@ static const struct snd_soc_dapm_route intercon[] = { {"SLIMBUS_6_RX Audio Mixer", "MultiMedia14", "MM_DL14"}, {"SLIMBUS_6_RX Audio Mixer", "MultiMedia15", "MM_DL15"}, {"SLIMBUS_6_RX Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"SLIMBUS_6_RX Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"SLIMBUS_6_RX Audio Mixer", "MultiMedia22", "MM_DL22"}, + {"SLIMBUS_6_RX Audio Mixer", "MultiMedia23", "MM_DL23"}, + {"SLIMBUS_6_RX Audio Mixer", "MultiMedia24", "MM_DL24"}, + {"SLIMBUS_6_RX Audio Mixer", "MultiMedia25", "MM_DL25"}, + {"SLIMBUS_6_RX Audio Mixer", "MultiMedia26", "MM_DL26"}, {"SLIMBUS_6_RX", NULL, "SLIMBUS_6_RX Audio Mixer"}, {"SLIMBUS_7_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12593,6 +13163,12 @@ static const struct snd_soc_dapm_route intercon[] = { {"SLIMBUS_7_RX Audio Mixer", "MultiMedia14", "MM_DL14"}, {"SLIMBUS_7_RX Audio Mixer", "MultiMedia15", "MM_DL15"}, {"SLIMBUS_7_RX Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"SLIMBUS_7_RX Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"SLIMBUS_7_RX Audio Mixer", "MultiMedia22", "MM_DL22"}, + {"SLIMBUS_7_RX Audio Mixer", "MultiMedia23", "MM_DL23"}, + {"SLIMBUS_7_RX Audio Mixer", "MultiMedia24", "MM_DL24"}, + {"SLIMBUS_7_RX Audio Mixer", "MultiMedia25", "MM_DL25"}, + {"SLIMBUS_7_RX Audio Mixer", "MultiMedia26", "MM_DL26"}, {"SLIMBUS_7_RX", NULL, "SLIMBUS_7_RX Audio Mixer"}, {"USB_AUDIO_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12611,6 +13187,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"USB_AUDIO_RX Audio Mixer", "MultiMedia14", "MM_DL14"}, {"USB_AUDIO_RX Audio Mixer", "MultiMedia15", "MM_DL15"}, {"USB_AUDIO_RX Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"USB_AUDIO_RX Audio Mixer", "MultiMedia26", "MM_DL26"}, {"USB_AUDIO_RX", NULL, "USB_AUDIO_RX Audio Mixer"}, {"MultiMedia1 Mixer", "VOC_REC_UL", "INCALL_RECORD_TX"}, @@ -12640,6 +13217,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"MultiMedia8 Mixer", "INT3_MI2S_TX", "INT3_MI2S_TX"}, {"MultiMedia3 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"}, {"MultiMedia5 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"}, + {"MultiMedia16 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"}, {"MultiMedia5 Mixer", "SLIM_7_TX", "SLIMBUS_7_TX"}, {"MultiMedia5 Mixer", "SLIM_8_TX", "SLIMBUS_8_TX"}, {"MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12658,6 +13236,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"MI2S_RX Audio Mixer", "MultiMedia14", "MM_DL14"}, {"MI2S_RX Audio Mixer", "MultiMedia15", "MM_DL15"}, {"MI2S_RX Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"MI2S_RX Audio Mixer", "MultiMedia26", "MM_DL26"}, {"MI2S_RX", NULL, "MI2S_RX Audio Mixer"}, {"QUAT_MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12675,6 +13254,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"QUAT_MI2S_RX Audio Mixer", "MultiMedia14", "MM_DL14"}, {"QUAT_MI2S_RX Audio Mixer", "MultiMedia15", "MM_DL15"}, {"QUAT_MI2S_RX Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"QUAT_MI2S_RX Audio Mixer", "MultiMedia26", "MM_DL26"}, {"QUAT_MI2S_RX", NULL, "QUAT_MI2S_RX Audio Mixer"}, {"TERT_MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12691,6 +13271,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"TERT_MI2S_RX Audio Mixer", "MultiMedia14", "MM_DL14"}, {"TERT_MI2S_RX Audio Mixer", "MultiMedia15", "MM_DL15"}, {"TERT_MI2S_RX Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"TERT_MI2S_RX Audio Mixer", "MultiMedia26", "MM_DL26"}, {"TERT_MI2S_RX", NULL, "TERT_MI2S_RX Audio Mixer"}, {"SEC_MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12707,6 +13288,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"SEC_MI2S_RX Audio Mixer", "MultiMedia14", "MM_DL14"}, {"SEC_MI2S_RX Audio Mixer", "MultiMedia15", "MM_DL15"}, {"SEC_MI2S_RX Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"SEC_MI2S_RX Audio Mixer", "MultiMedia26", "MM_DL26"}, {"SEC_MI2S_RX", NULL, "SEC_MI2S_RX Audio Mixer"}, {"SEC_MI2S_RX_SD1 Audio Mixer", "MultiMedia6", "MM_DL6"}, @@ -12730,6 +13312,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"PRI_MI2S_RX Audio Mixer", "MultiMedia14", "MM_DL14"}, {"PRI_MI2S_RX Audio Mixer", "MultiMedia15", "MM_DL15"}, {"PRI_MI2S_RX Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"PRI_MI2S_RX Audio Mixer", "MultiMedia26", "MM_DL26"}, {"PRI_MI2S_RX", NULL, "PRI_MI2S_RX Audio Mixer"}, {"INT0_MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12747,6 +13330,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"INT0_MI2S_RX Audio Mixer", "MultiMedia14", "MM_DL14"}, {"INT0_MI2S_RX Audio Mixer", "MultiMedia15", "MM_DL15"}, {"INT0_MI2S_RX Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"INT0_MI2S_RX Audio Mixer", "MultiMedia26", "MM_DL26"}, {"INT0_MI2S_RX", NULL, "INT0_MI2S_RX Audio Mixer"}, {"INT4_MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12764,6 +13348,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"INT4_MI2S_RX Audio Mixer", "MultiMedia14", "MM_DL14"}, {"INT4_MI2S_RX Audio Mixer", "MultiMedia15", "MM_DL15"}, {"INT4_MI2S_RX Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"INT4_MI2S_RX Audio Mixer", "MultiMedia26", "MM_DL26"}, {"INT4_MI2S_RX", NULL, "INT4_MI2S_RX Audio Mixer"}, {"QUIN_MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12783,6 +13368,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"QUIN_MI2S_RX Audio Mixer", "MultiMedia14", "MM_DL14"}, {"QUIN_MI2S_RX Audio Mixer", "MultiMedia15", "MM_DL15"}, {"QUIN_MI2S_RX Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"QUIN_MI2S_RX Audio Mixer", "MultiMedia26", "MM_DL26"}, {"QUIN_MI2S_RX", NULL, "QUIN_MI2S_RX Audio Mixer"}, {"PRI_TDM_RX_0 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12801,6 +13387,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"PRI_TDM_RX_0 Audio Mixer", "MultiMedia14", "MM_DL14"}, {"PRI_TDM_RX_0 Audio Mixer", "MultiMedia15", "MM_DL15"}, {"PRI_TDM_RX_0 Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"PRI_TDM_RX_0 Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"PRI_TDM_RX_0 Audio Mixer", "MultiMedia26", "MM_DL26"}, {"PRI_TDM_RX_0", NULL, "PRI_TDM_RX_0 Audio Mixer"}, {"PRI_TDM_RX_1 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12819,6 +13407,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"PRI_TDM_RX_1 Audio Mixer", "MultiMedia14", "MM_DL14"}, {"PRI_TDM_RX_1 Audio Mixer", "MultiMedia15", "MM_DL15"}, {"PRI_TDM_RX_1 Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"PRI_TDM_RX_1 Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"PRI_TDM_RX_1 Audio Mixer", "MultiMedia26", "MM_DL26"}, {"PRI_TDM_RX_1", NULL, "PRI_TDM_RX_1 Audio Mixer"}, {"PRI_TDM_RX_2 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12837,6 +13427,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"PRI_TDM_RX_2 Audio Mixer", "MultiMedia14", "MM_DL14"}, {"PRI_TDM_RX_2 Audio Mixer", "MultiMedia15", "MM_DL15"}, {"PRI_TDM_RX_2 Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"PRI_TDM_RX_2 Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"PRI_TDM_RX_2 Audio Mixer", "MultiMedia26", "MM_DL26"}, {"PRI_TDM_RX_2", NULL, "PRI_TDM_RX_2 Audio Mixer"}, {"PRI_TDM_RX_3 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12855,6 +13447,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"PRI_TDM_RX_3 Audio Mixer", "MultiMedia14", "MM_DL14"}, {"PRI_TDM_RX_3 Audio Mixer", "MultiMedia15", "MM_DL15"}, {"PRI_TDM_RX_3 Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"PRI_TDM_RX_3 Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"PRI_TDM_RX_3 Audio Mixer", "MultiMedia26", "MM_DL26"}, {"PRI_TDM_RX_3", NULL, "PRI_TDM_RX_3 Audio Mixer"}, {"PRI_TDM_TX_0 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12891,6 +13485,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"SEC_TDM_RX_0 Audio Mixer", "MultiMedia14", "MM_DL14"}, {"SEC_TDM_RX_0 Audio Mixer", "MultiMedia15", "MM_DL15"}, {"SEC_TDM_RX_0 Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"SEC_TDM_RX_0 Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"SEC_TDM_RX_0 Audio Mixer", "MultiMedia26", "MM_DL26"}, {"SEC_TDM_RX_0", NULL, "SEC_TDM_RX_0 Audio Mixer"}, {"SEC_TDM_RX_1 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12909,6 +13505,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"SEC_TDM_RX_1 Audio Mixer", "MultiMedia14", "MM_DL14"}, {"SEC_TDM_RX_1 Audio Mixer", "MultiMedia15", "MM_DL15"}, {"SEC_TDM_RX_1 Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"SEC_TDM_RX_1 Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"SEC_TDM_RX_1 Audio Mixer", "MultiMedia26", "MM_DL26"}, {"SEC_TDM_RX_1", NULL, "SEC_TDM_RX_1 Audio Mixer"}, {"SEC_TDM_RX_2 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12927,6 +13525,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"SEC_TDM_RX_2 Audio Mixer", "MultiMedia14", "MM_DL14"}, {"SEC_TDM_RX_2 Audio Mixer", "MultiMedia15", "MM_DL15"}, {"SEC_TDM_RX_2 Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"SEC_TDM_RX_2 Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"SEC_TDM_RX_2 Audio Mixer", "MultiMedia26", "MM_DL26"}, {"SEC_TDM_RX_2", NULL, "SEC_TDM_RX_2 Audio Mixer"}, {"SEC_TDM_RX_3 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12945,6 +13545,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"SEC_TDM_RX_3 Audio Mixer", "MultiMedia14", "MM_DL14"}, {"SEC_TDM_RX_3 Audio Mixer", "MultiMedia15", "MM_DL15"}, {"SEC_TDM_RX_3 Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"SEC_TDM_RX_3 Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"SEC_TDM_RX_3 Audio Mixer", "MultiMedia26", "MM_DL26"}, {"SEC_TDM_RX_3", NULL, "SEC_TDM_RX_3 Audio Mixer"}, {"SEC_TDM_TX_0 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12981,6 +13583,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"TERT_TDM_RX_0 Audio Mixer", "MultiMedia14", "MM_DL14"}, {"TERT_TDM_RX_0 Audio Mixer", "MultiMedia15", "MM_DL15"}, {"TERT_TDM_RX_0 Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"TERT_TDM_RX_0 Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"TERT_TDM_RX_0 Audio Mixer", "MultiMedia26", "MM_DL26"}, {"TERT_TDM_RX_0", NULL, "TERT_TDM_RX_0 Audio Mixer"}, {"TERT_TDM_TX_0 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13017,6 +13621,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"TERT_TDM_RX_1 Audio Mixer", "MultiMedia14", "MM_DL14"}, {"TERT_TDM_RX_1 Audio Mixer", "MultiMedia15", "MM_DL15"}, {"TERT_TDM_RX_1 Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"TERT_TDM_RX_1 Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"TERT_TDM_RX_1 Audio Mixer", "MultiMedia26", "MM_DL26"}, {"TERT_TDM_RX_1", NULL, "TERT_TDM_RX_1 Audio Mixer"}, {"TERT_TDM_RX_2 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13035,6 +13641,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"TERT_TDM_RX_2 Audio Mixer", "MultiMedia14", "MM_DL14"}, {"TERT_TDM_RX_2 Audio Mixer", "MultiMedia15", "MM_DL15"}, {"TERT_TDM_RX_2 Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"TERT_TDM_RX_2 Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"TERT_TDM_RX_2 Audio Mixer", "MultiMedia26", "MM_DL26"}, {"TERT_TDM_RX_2", NULL, "TERT_TDM_RX_2 Audio Mixer"}, {"TERT_TDM_RX_3 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13053,6 +13661,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"TERT_TDM_RX_3 Audio Mixer", "MultiMedia14", "MM_DL14"}, {"TERT_TDM_RX_3 Audio Mixer", "MultiMedia15", "MM_DL15"}, {"TERT_TDM_RX_3 Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"TERT_TDM_RX_3 Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"TERT_TDM_RX_3 Audio Mixer", "MultiMedia26", "MM_DL26"}, {"TERT_TDM_RX_3", NULL, "TERT_TDM_RX_3 Audio Mixer"}, {"TERT_TDM_RX_4 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13071,6 +13681,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"TERT_TDM_RX_4 Audio Mixer", "MultiMedia14", "MM_DL14"}, {"TERT_TDM_RX_4 Audio Mixer", "MultiMedia15", "MM_DL15"}, {"TERT_TDM_RX_4 Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"TERT_TDM_RX_4 Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"TERT_TDM_RX_4 Audio Mixer", "MultiMedia26", "MM_DL26"}, {"TERT_TDM_RX_4", NULL, "TERT_TDM_RX_4 Audio Mixer"}, {"QUAT_TDM_RX_0 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13090,6 +13702,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"QUAT_TDM_RX_0 Audio Mixer", "MultiMedia15", "MM_DL15"}, {"QUAT_TDM_RX_0 Audio Mixer", "MultiMedia16", "MM_DL16"}, {"QUAT_TDM_RX_0 Audio Mixer", "MultiMedia20", "MM_DL20"}, + {"QUAT_TDM_RX_0 Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"QUAT_TDM_RX_0 Audio Mixer", "MultiMedia26", "MM_DL26"}, {"QUAT_TDM_RX_0", NULL, "QUAT_TDM_RX_0 Audio Mixer"}, {"PRI_TDM_RX_0 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13108,6 +13722,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"PRI_TDM_RX_0 Audio Mixer", "MultiMedia14", "MM_DL14"}, {"PRI_TDM_RX_0 Audio Mixer", "MultiMedia15", "MM_DL15"}, {"PRI_TDM_RX_0 Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"PRI_TDM_RX_0 Audio Mixer", "MultiMedia26", "MM_DL26"}, {"PRI_TDM_RX_0", NULL, "PRI_TDM_RX_0 Audio Mixer"}, {"SEC_TDM_RX_0 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13126,6 +13741,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"SEC_TDM_RX_0 Audio Mixer", "MultiMedia14", "MM_DL14"}, {"SEC_TDM_RX_0 Audio Mixer", "MultiMedia15", "MM_DL15"}, {"SEC_TDM_RX_0 Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"SEC_TDM_RX_0 Audio Mixer", "MultiMedia26", "MM_DL26"}, {"SEC_TDM_RX_0", NULL, "SEC_TDM_RX_0 Audio Mixer"}, {"QUAT_TDM_TX_0 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13163,6 +13779,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"QUAT_TDM_RX_1 Audio Mixer", "MultiMedia15", "MM_DL15"}, {"QUAT_TDM_RX_1 Audio Mixer", "MultiMedia16", "MM_DL16"}, {"QUAT_TDM_RX_1 Audio Mixer", "MultiMedia20", "MM_DL20"}, + {"QUAT_TDM_RX_1 Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"QUAT_TDM_RX_1 Audio Mixer", "MultiMedia26", "MM_DL26"}, {"QUAT_TDM_RX_1", NULL, "QUAT_TDM_RX_1 Audio Mixer"}, {"QUAT_TDM_RX_2 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13182,6 +13800,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"QUAT_TDM_RX_2 Audio Mixer", "MultiMedia15", "MM_DL15"}, {"QUAT_TDM_RX_2 Audio Mixer", "MultiMedia16", "MM_DL16"}, {"QUAT_TDM_RX_2 Audio Mixer", "MultiMedia20", "MM_DL20"}, + {"QUAT_TDM_RX_2 Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"QUAT_TDM_RX_2 Audio Mixer", "MultiMedia26", "MM_DL26"}, {"QUAT_TDM_RX_2", NULL, "QUAT_TDM_RX_2 Audio Mixer"}, {"QUAT_TDM_RX_3 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13201,6 +13821,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"QUAT_TDM_RX_3 Audio Mixer", "MultiMedia15", "MM_DL15"}, {"QUAT_TDM_RX_3 Audio Mixer", "MultiMedia16", "MM_DL16"}, {"QUAT_TDM_RX_3 Audio Mixer", "MultiMedia20", "MM_DL20"}, + {"QUAT_TDM_RX_3 Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"QUAT_TDM_RX_3 Audio Mixer", "MultiMedia26", "MM_DL26"}, {"QUAT_TDM_RX_3", NULL, "QUAT_TDM_RX_3 Audio Mixer"}, {"MultiMedia1 Mixer", "PRI_TX", "PRI_I2S_TX"}, @@ -13208,9 +13830,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"MultiMedia2 Mixer", "MI2S_TX", "MI2S_TX"}, {"MultiMedia3 Mixer", "MI2S_TX", "MI2S_TX"}, {"MultiMedia5 Mixer", "MI2S_TX", "MI2S_TX"}, + {"MultiMedia16 Mixer", "MI2S_TX", "MI2S_TX"}, {"MultiMedia1 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"}, {"MultiMedia2 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"}, {"MultiMedia6 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"}, + {"MultiMedia27 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"}, {"MultiMedia1 Mixer", "QUIN_MI2S_TX", "QUIN_MI2S_TX"}, {"MultiMedia2 Mixer", "QUIN_MI2S_TX", "QUIN_MI2S_TX"}, {"MultiMedia1 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"}, @@ -13220,24 +13844,29 @@ static const struct snd_soc_dapm_route intercon[] = { {"MultiMedia1 Mixer", "INT3_MI2S_TX", "INT3_MI2S_TX"}, {"MultiMedia2 Mixer", "INT3_MI2S_TX", "INT3_MI2S_TX"}, {"MultiMedia1 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"}, + {"MultiMedia27 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"}, {"MultiMedia1 Mixer", "AUX_PCM_UL_TX", "AUX_PCM_TX"}, {"MultiMedia3 Mixer", "AUX_PCM_TX", "AUX_PCM_TX"}, {"MultiMedia5 Mixer", "AUX_PCM_UL_TX", "AUX_PCM_TX"}, {"MultiMedia1 Mixer", "SEC_AUX_PCM_UL_TX", "SEC_AUX_PCM_TX"}, {"MultiMedia3 Mixer", "SEC_AUX_PCM_TX", "SEC_AUX_PCM_TX"}, {"MultiMedia5 Mixer", "SEC_AUX_PCM_TX", "SEC_AUX_PCM_TX"}, + {"MultiMedia16 Mixer", "AUX_PCM_TX", "AUX_PCM_TX"}, + {"MultiMedia16 Mixer", "SEC_AUX_PCM_TX", "SEC_AUX_PCM_TX"}, {"MultiMedia1 Mixer", "TERT_AUXPCM_UL_TX", "TERT_AUX_PCM_TX"}, {"MultiMedia3 Mixer", "TERT_AUX_PCM_TX", "TERT_AUX_PCM_TX"}, {"MultiMedia5 Mixer", "TERT_AUX_PCM_TX", "TERT_AUX_PCM_TX"}, {"MultiMedia1 Mixer", "QUAT_AUXPCM_UL_TX", "QUAT_AUX_PCM_TX"}, {"MultiMedia3 Mixer", "QUAT_AUX_PCM_TX", "QUAT_AUX_PCM_TX"}, {"MultiMedia5 Mixer", "QUAT_AUX_PCM_TX", "QUAT_AUX_PCM_TX"}, + {"MultiMedia16 Mixer", "QUAT_AUX_PCM_TX", "QUAT_AUX_PCM_TX"}, {"MultiMedia2 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"}, {"MultiMedia2 Mixer", "SLIM_6_TX", "SLIMBUS_6_TX"}, {"MultiMedia2 Mixer", "SLIM_1_TX", "SLIMBUS_1_TX"}, {"MultiMedia2 Mixer", "SLIM_8_TX", "SLIMBUS_8_TX"}, {"MultiMedia1 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"}, {"MultiMedia1 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"}, + {"MultiMedia27 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"}, {"MultiMedia2 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"}, {"MultiMedia6 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"}, {"MultiMedia6 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"}, @@ -13246,9 +13875,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"MultiMedia6 Mixer", "INT2_MI2S_TX", "INT2_MI2S_TX"}, {"MultiMedia3 Mixer", "INT2_MI2S_TX", "INT2_MI2S_TX"}, {"MultiMedia5 Mixer", "INT2_MI2S_TX", "INT2_MI2S_TX"}, + {"MultiMedia16 Mixer", "INT2_MI2S_TX", "INT2_MI2S_TX"}, {"MultiMedia6 Mixer", "INT3_MI2S_TX", "INT3_MI2S_TX"}, {"MultiMedia3 Mixer", "INT3_MI2S_TX", "INT3_MI2S_TX"}, {"MultiMedia5 Mixer", "INT3_MI2S_TX", "INT3_MI2S_TX"}, + {"MultiMedia16 Mixer", "INT3_MI2S_TX", "INT3_MI2S_TX"}, {"MultiMedia6 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"}, {"MultiMedia6 Mixer", "AUX_PCM_UL_TX", "AUX_PCM_TX"}, {"MultiMedia6 Mixer", "SEC_AUX_PCM_UL_TX", "SEC_AUX_PCM_TX"}, @@ -13403,6 +14034,25 @@ static const struct snd_soc_dapm_route intercon[] = { {"MultiMedia20 Mixer", "QUAT_TDM_TX_2", "QUAT_TDM_TX_2"}, {"MultiMedia20 Mixer", "QUAT_TDM_TX_3", "QUAT_TDM_TX_3"}, + {"MultiMedia21 Mixer", "PRI_TDM_TX_0", "PRI_TDM_TX_0"}, + {"MultiMedia21 Mixer", "PRI_TDM_TX_1", "PRI_TDM_TX_1"}, + {"MultiMedia21 Mixer", "PRI_TDM_TX_2", "PRI_TDM_TX_2"}, + {"MultiMedia21 Mixer", "PRI_TDM_TX_3", "PRI_TDM_TX_3"}, + {"MultiMedia21 Mixer", "SEC_TDM_TX_0", "SEC_TDM_TX_0"}, + {"MultiMedia21 Mixer", "SEC_TDM_TX_1", "SEC_TDM_TX_1"}, + {"MultiMedia21 Mixer", "SEC_TDM_TX_2", "SEC_TDM_TX_2"}, + {"MultiMedia21 Mixer", "SEC_TDM_TX_3", "SEC_TDM_TX_3"}, + {"MultiMedia21 Mixer", "TERT_TDM_TX_0", "TERT_TDM_TX_0"}, + {"MultiMedia21 Mixer", "TERT_TDM_TX_1", "TERT_TDM_TX_1"}, + {"MultiMedia21 Mixer", "TERT_TDM_TX_2", "TERT_TDM_TX_2"}, + {"MultiMedia21 Mixer", "TERT_TDM_TX_3", "TERT_TDM_TX_3"}, + {"MultiMedia21 Mixer", "QUAT_TDM_TX_0", "QUAT_TDM_TX_0"}, + {"MultiMedia21 Mixer", "QUAT_TDM_TX_1", "QUAT_TDM_TX_1"}, + {"MultiMedia21 Mixer", "QUAT_TDM_TX_2", "QUAT_TDM_TX_2"}, + {"MultiMedia21 Mixer", "QUAT_TDM_TX_3", "QUAT_TDM_TX_3"}, + {"MultiMedia21 Mixer", "AUX_PCM_UL_TX", "AUX_PCM_TX"}, + {"MultiMedia21 Mixer", "SEC_AUX_PCM_UL_TX", "SEC_AUX_PCM_TX"}, + {"MultiMedia1 Mixer", "USB_AUDIO_TX", "USB_AUDIO_TX"}, {"MultiMedia2 Mixer", "USB_AUDIO_TX", "USB_AUDIO_TX"}, {"MultiMedia4 Mixer", "USB_AUDIO_TX", "USB_AUDIO_TX"}, @@ -13410,6 +14060,24 @@ static const struct snd_soc_dapm_route intercon[] = { {"MultiMedia6 Mixer", "USB_AUDIO_TX", "USB_AUDIO_TX"}, {"MultiMedia8 Mixer", "USB_AUDIO_TX", "USB_AUDIO_TX"}, + {"MultiMedia16 Mixer", "PRI_TDM_TX_0", "PRI_TDM_TX_0"}, + {"MultiMedia16 Mixer", "PRI_TDM_TX_1", "PRI_TDM_TX_1"}, + {"MultiMedia16 Mixer", "PRI_TDM_TX_2", "PRI_TDM_TX_2"}, + {"MultiMedia16 Mixer", "PRI_TDM_TX_3", "PRI_TDM_TX_3"}, + {"MultiMedia16 Mixer", "SEC_TDM_TX_0", "SEC_TDM_TX_0"}, + {"MultiMedia16 Mixer", "SEC_TDM_TX_1", "SEC_TDM_TX_1"}, + {"MultiMedia16 Mixer", "SEC_TDM_TX_2", "SEC_TDM_TX_2"}, + {"MultiMedia16 Mixer", "SEC_TDM_TX_3", "SEC_TDM_TX_3"}, + {"MultiMedia16 Mixer", "TERT_TDM_TX_0", "TERT_TDM_TX_0"}, + {"MultiMedia16 Mixer", "TERT_TDM_TX_1", "TERT_TDM_TX_1"}, + {"MultiMedia16 Mixer", "TERT_TDM_TX_2", "TERT_TDM_TX_2"}, + {"MultiMedia16 Mixer", "TERT_TDM_TX_3", "TERT_TDM_TX_3"}, + {"MultiMedia16 Mixer", "QUAT_TDM_TX_0", "QUAT_TDM_TX_0"}, + {"MultiMedia16 Mixer", "QUAT_TDM_TX_1", "QUAT_TDM_TX_1"}, + {"MultiMedia16 Mixer", "QUAT_TDM_TX_2", "QUAT_TDM_TX_2"}, + {"MultiMedia16 Mixer", "QUAT_TDM_TX_3", "QUAT_TDM_TX_3"}, + {"MultiMedia16 Mixer", "USB_AUDIO_TX", "USB_AUDIO_TX"}, + {"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, {"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia2", "MM_DL2"}, {"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia3", "MM_DL3"}, @@ -13492,8 +14160,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"MultiMedia19 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"}, {"MultiMedia5 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"}, {"MultiMedia8 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"}, + {"MultiMedia16 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"}, {"MultiMedia1 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"}, {"MultiMedia4 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"}, + {"MultiMedia16 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"}, {"MultiMedia17 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"}, {"MultiMedia18 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"}, {"MultiMedia19 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"}, @@ -13509,6 +14179,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"MultiMedia19 Mixer", "AFE_PCM_TX", "PCM_TX"}, {"MultiMedia5 Mixer", "AFE_PCM_TX", "PCM_TX"}, {"MultiMedia8 Mixer", "AFE_PCM_TX", "PCM_TX"}, + {"MultiMedia16 Mixer", "AFE_PCM_TX", "PCM_TX"}, {"MM_UL1", NULL, "MultiMedia1 Mixer"}, {"MultiMedia2 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"}, {"MM_UL2", NULL, "MultiMedia2 Mixer"}, @@ -13518,10 +14189,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"MM_UL6", NULL, "MultiMedia6 Mixer"}, {"MM_UL8", NULL, "MultiMedia8 Mixer"}, {"MM_UL9", NULL, "MultiMedia9 Mixer"}, + {"MM_UL16", NULL, "MultiMedia16 Mixer"}, {"MM_UL17", NULL, "MultiMedia17 Mixer"}, {"MM_UL18", NULL, "MultiMedia18 Mixer"}, {"MM_UL19", NULL, "MultiMedia19 Mixer"}, {"MM_UL20", NULL, "MultiMedia20 Mixer"}, + {"MM_UL21", NULL, "MultiMedia21 Mixer"}, + {"MM_UL27", NULL, "MultiMedia27 Mixer"}, {"AUX_PCM_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, {"AUX_PCM_RX Audio Mixer", "MultiMedia2", "MM_DL2"}, @@ -13539,6 +14213,9 @@ static const struct snd_soc_dapm_route intercon[] = { {"AUX_PCM_RX Audio Mixer", "MultiMedia14", "MM_DL14"}, {"AUX_PCM_RX Audio Mixer", "MultiMedia15", "MM_DL15"}, {"AUX_PCM_RX Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"AUX_PCM_RX Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"AUX_PCM_RX Audio Mixer", "MultiMedia6", "MM_UL6"}, + {"AUX_PCM_RX Audio Mixer", "MultiMedia21", "MM_UL21"}, {"AUX_PCM_RX", NULL, "AUX_PCM_RX Audio Mixer"}, {"SEC_AUX_PCM_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13557,6 +14234,9 @@ static const struct snd_soc_dapm_route intercon[] = { {"SEC_AUX_PCM_RX Audio Mixer", "MultiMedia14", "MM_DL14"}, {"SEC_AUX_PCM_RX Audio Mixer", "MultiMedia15", "MM_DL15"}, {"SEC_AUX_PCM_RX Audio Mixer", "MultiMedia16", "MM_DL16"}, + {"SEC_AUX_PCM_RX Audio Mixer", "MultiMedia21", "MM_DL21"}, + {"SEC_AUX_PCM_RX Audio Mixer", "MultiMedia6", "MM_UL6"}, + {"SEC_AUX_PCM_RX Audio Mixer", "MultiMedia21", "MM_UL21"}, {"SEC_AUX_PCM_RX", NULL, "SEC_AUX_PCM_RX Audio Mixer"}, {"TERT_AUX_PCM_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13928,6 +14608,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"MM_UL6", NULL, "AUDIO_REF_EC_UL6 MUX"}, {"MM_UL8", NULL, "AUDIO_REF_EC_UL8 MUX"}, {"MM_UL9", NULL, "AUDIO_REF_EC_UL9 MUX"}, + {"MM_UL16", NULL, "AUDIO_REF_EC_UL16 MUX"}, {"MM_UL17", NULL, "AUDIO_REF_EC_UL17 MUX"}, {"MM_UL18", NULL, "AUDIO_REF_EC_UL18 MUX"}, {"MM_UL19", NULL, "AUDIO_REF_EC_UL19 MUX"}, @@ -14460,6 +15141,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"QUAT_TDM_RX_0 Port Mixer", "AFE_PCM_TX", "PCM_TX"}, {"QUAT_TDM_RX_0 Port Mixer", "AUX_PCM_UL_TX", "AUX_PCM_TX"}, {"QUAT_TDM_RX_0 Port Mixer", "SEC_AUX_PCM_UL_TX", "SEC_AUX_PCM_TX"}, + {"QUAT_TDM_RX_0 Port Mixer", "PRI_TDM_TX_0", "PRI_TDM_TX_0"}, + {"QUAT_TDM_RX_0 Port Mixer", "PRI_TDM_TX_1", "PRI_TDM_TX_1"}, + {"QUAT_TDM_RX_0 Port Mixer", "PRI_TDM_TX_2", "PRI_TDM_TX_2"}, + {"QUAT_TDM_RX_0 Port Mixer", "PRI_TDM_TX_3", "PRI_TDM_TX_3"}, {"QUAT_TDM_RX_0 Port Mixer", "TERT_TDM_TX_0", "TERT_TDM_TX_0"}, {"QUAT_TDM_RX_0 Port Mixer", "TERT_TDM_TX_1", "TERT_TDM_TX_1"}, {"QUAT_TDM_RX_0 Port Mixer", "TERT_TDM_TX_2", "TERT_TDM_TX_2"}, @@ -14478,6 +15163,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"QUAT_TDM_RX_1 Port Mixer", "AFE_PCM_TX", "PCM_TX"}, {"QUAT_TDM_RX_1 Port Mixer", "AUX_PCM_UL_TX", "AUX_PCM_TX"}, {"QUAT_TDM_RX_1 Port Mixer", "SEC_AUX_PCM_UL_TX", "SEC_AUX_PCM_TX"}, + {"QUAT_TDM_RX_1 Port Mixer", "PRI_TDM_TX_0", "PRI_TDM_TX_0"}, + {"QUAT_TDM_RX_1 Port Mixer", "PRI_TDM_TX_1", "PRI_TDM_TX_1"}, + {"QUAT_TDM_RX_1 Port Mixer", "PRI_TDM_TX_2", "PRI_TDM_TX_2"}, + {"QUAT_TDM_RX_1 Port Mixer", "PRI_TDM_TX_3", "PRI_TDM_TX_3"}, {"QUAT_TDM_RX_1 Port Mixer", "TERT_TDM_TX_0", "TERT_TDM_TX_0"}, {"QUAT_TDM_RX_1 Port Mixer", "TERT_TDM_TX_1", "TERT_TDM_TX_1"}, {"QUAT_TDM_RX_1 Port Mixer", "TERT_TDM_TX_2", "TERT_TDM_TX_2"}, @@ -14496,6 +15185,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"QUAT_TDM_RX_2 Port Mixer", "AFE_PCM_TX", "PCM_TX"}, {"QUAT_TDM_RX_2 Port Mixer", "AUX_PCM_UL_TX", "AUX_PCM_TX"}, {"QUAT_TDM_RX_2 Port Mixer", "SEC_AUX_PCM_UL_TX", "SEC_AUX_PCM_TX"}, + {"QUAT_TDM_RX_2 Port Mixer", "PRI_TDM_TX_0", "PRI_TDM_TX_0"}, + {"QUAT_TDM_RX_2 Port Mixer", "PRI_TDM_TX_1", "PRI_TDM_TX_1"}, + {"QUAT_TDM_RX_2 Port Mixer", "PRI_TDM_TX_2", "PRI_TDM_TX_2"}, + {"QUAT_TDM_RX_2 Port Mixer", "PRI_TDM_TX_3", "PRI_TDM_TX_3"}, {"QUAT_TDM_RX_2 Port Mixer", "TERT_TDM_TX_0", "TERT_TDM_TX_0"}, {"QUAT_TDM_RX_2 Port Mixer", "TERT_TDM_TX_1", "TERT_TDM_TX_1"}, {"QUAT_TDM_RX_2 Port Mixer", "TERT_TDM_TX_2", "TERT_TDM_TX_2"}, @@ -14514,6 +15207,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"QUAT_TDM_RX_3 Port Mixer", "AFE_PCM_TX", "PCM_TX"}, {"QUAT_TDM_RX_3 Port Mixer", "AUX_PCM_UL_TX", "AUX_PCM_TX"}, {"QUAT_TDM_RX_3 Port Mixer", "SEC_AUX_PCM_UL_TX", "SEC_AUX_PCM_TX"}, + {"QUAT_TDM_RX_3 Port Mixer", "PRI_TDM_TX_0", "PRI_TDM_TX_0"}, + {"QUAT_TDM_RX_3 Port Mixer", "PRI_TDM_TX_1", "PRI_TDM_TX_1"}, + {"QUAT_TDM_RX_3 Port Mixer", "PRI_TDM_TX_2", "PRI_TDM_TX_2"}, + {"QUAT_TDM_RX_3 Port Mixer", "PRI_TDM_TX_3", "PRI_TDM_TX_3"}, {"QUAT_TDM_RX_3 Port Mixer", "TERT_TDM_TX_0", "TERT_TDM_TX_0"}, {"QUAT_TDM_RX_3 Port Mixer", "TERT_TDM_TX_1", "TERT_TDM_TX_1"}, {"QUAT_TDM_RX_3 Port Mixer", "TERT_TDM_TX_2", "TERT_TDM_TX_2"}, @@ -14962,10 +15659,6 @@ static int msm_pcm_routing_close(struct snd_pcm_substream *substream) bedai->active = 0; bedai->sample_rate = 0; bedai->channel = 0; - for (i = 0; i < MSM_FRONTEND_DAI_MAX; i++) { - if (bedai->passthr_mode[i] != LISTEN) - bedai->passthr_mode[i] = LEGACY_PCM; - } mutex_unlock(&routing_lock); return 0; @@ -15168,7 +15861,7 @@ done: static int msm_routing_send_device_pp_params(int port_id, int copp_idx, int fe_id) { - int index, topo_id, be_idx; + int topo_id, be_idx; unsigned long pp_config = 0; bool mute_on; int latency; @@ -15192,16 +15885,6 @@ static int msm_routing_send_device_pp_params(int port_id, int copp_idx, return -EINVAL; } - for (index = 0; index < MSM_BACKEND_DAI_PP_PARAMS_REQ_MAX; index++) { - if (msm_bedais_pp_params[index].port_id == port_id) - break; - } - if (index >= MSM_BACKEND_DAI_PP_PARAMS_REQ_MAX) { - pr_err("%s: Invalid backend pp params index %d\n", - __func__, index); - return -EINVAL; - } - topo_id = adm_get_topology_for_port_copp_idx(port_id, copp_idx); if (topo_id != COMPRESSED_PASSTHROUGH_DEFAULT_TOPOLOGY) { pr_err("%s: Invalid passthrough topology 0x%x\n", @@ -15213,11 +15896,11 @@ static int msm_routing_send_device_pp_params(int port_id, int copp_idx, (msm_bedais[be_idx].passthr_mode[fe_id] == LISTEN)) compr_passthr_mode = false; - pp_config = msm_bedais_pp_params[index].pp_params_config; + pp_config = msm_bedais_pp_params[be_idx].pp_params_config; if (test_bit(ADM_PP_PARAM_MUTE_BIT, &pp_config)) { pr_debug("%s: ADM_PP_PARAM_MUTE\n", __func__); clear_bit(ADM_PP_PARAM_MUTE_BIT, &pp_config); - mute_on = msm_bedais_pp_params[index].mute_on; + mute_on = msm_bedais_pp_params[be_idx].mute_on; if ((msm_bedais[be_idx].active) && compr_passthr_mode) adm_send_compressed_device_mute(port_id, copp_idx, @@ -15227,7 +15910,7 @@ static int msm_routing_send_device_pp_params(int port_id, int copp_idx, pr_debug("%s: ADM_PP_PARAM_LATENCY\n", __func__); clear_bit(ADM_PP_PARAM_LATENCY_BIT, &pp_config); - latency = msm_bedais_pp_params[index].latency; + latency = msm_bedais_pp_params[be_idx].latency; if ((msm_bedais[be_idx].active) && compr_passthr_mode) adm_send_compressed_device_latency(port_id, copp_idx, @@ -15236,18 +15919,47 @@ static int msm_routing_send_device_pp_params(int port_id, int copp_idx, return 0; } +static bool msm_pcm_routing_test_pp_param(int be_idx, long param_bit) +{ + return test_bit(param_bit, + &msm_bedais_pp_params[be_idx].pp_params_config); +} + +static void msm_routing_set_pp_param(long param_bit, int value) +{ + int be_idx; + + if (value) { + for (be_idx = 0; be_idx < MSM_BACKEND_DAI_MAX; be_idx++) + set_bit(param_bit, + &msm_bedais_pp_params[be_idx]. + pp_params_config); + } else { + for (be_idx = 0; be_idx < MSM_BACKEND_DAI_MAX; be_idx++) + clear_bit(param_bit, + &msm_bedais_pp_params[be_idx]. + pp_params_config); + } +} + static int msm_routing_put_device_pp_params_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { int pp_id = ucontrol->value.integer.value[0]; + int value = ucontrol->value.integer.value[1]; int port_id = 0; - int index, be_idx, i, topo_id, idx; + int be_idx, i, topo_id, idx; bool mute; int latency; bool compr_passthr_mode = true; pr_debug("%s: pp_id: 0x%x\n", __func__, pp_id); + if (pp_id == ADM_PP_PARAM_LIMITER_ID) { + msm_routing_set_pp_param(ADM_PP_PARAM_LIMITER_BIT, value); + goto done; + } + for (be_idx = 0; be_idx < MSM_BACKEND_DAI_MAX; be_idx++) { port_id = msm_bedais[be_idx].port_id; if (port_id == HDMI_RX || port_id == DISPLAY_PORT_RX) @@ -15259,16 +15971,6 @@ static int msm_routing_put_device_pp_params_mixer(struct snd_kcontrol *kcontrol, return -EINVAL; } - for (index = 0; index < MSM_BACKEND_DAI_PP_PARAMS_REQ_MAX; index++) { - if (msm_bedais_pp_params[index].port_id == port_id) - break; - } - if (index >= MSM_BACKEND_DAI_PP_PARAMS_REQ_MAX) { - pr_err("%s: Invalid pp params backend index %d\n", - __func__, index); - return -EINVAL; - } - for_each_set_bit(i, &msm_bedais[be_idx].fe_sessions[0], MSM_FRONTEND_DAI_MM_SIZE) { if ((msm_bedais[be_idx].passthr_mode[i] == LEGACY_PCM) || @@ -15291,22 +15993,20 @@ static int msm_routing_put_device_pp_params_mixer(struct snd_kcontrol *kcontrol, switch (pp_id) { case ADM_PP_PARAM_MUTE_ID: pr_debug("%s: ADM_PP_PARAM_MUTE\n", __func__); - mute = ucontrol->value.integer.value[1] ? true : false; - msm_bedais_pp_params[index].mute_on = mute; + mute = value ? true : false; + msm_bedais_pp_params[be_idx].mute_on = mute; set_bit(ADM_PP_PARAM_MUTE_BIT, - &msm_bedais_pp_params[index].pp_params_config); + &msm_bedais_pp_params[be_idx].pp_params_config); if ((msm_bedais[be_idx].active) && compr_passthr_mode) adm_send_compressed_device_mute(port_id, idx, mute); break; case ADM_PP_PARAM_LATENCY_ID: pr_debug("%s: ADM_PP_PARAM_LATENCY\n", __func__); - msm_bedais_pp_params[index].latency = - ucontrol->value.integer.value[1]; + msm_bedais_pp_params[be_idx].latency = value; set_bit(ADM_PP_PARAM_LATENCY_BIT, - &msm_bedais_pp_params[index].pp_params_config); - latency = msm_bedais_pp_params[index].latency = - ucontrol->value.integer.value[1]; + &msm_bedais_pp_params[be_idx].pp_params_config); + latency = msm_bedais_pp_params[be_idx].latency = value; if ((msm_bedais[be_idx].active) && compr_passthr_mode) adm_send_compressed_device_latency(port_id, idx, latency); @@ -15318,6 +16018,7 @@ static int msm_routing_put_device_pp_params_mixer(struct snd_kcontrol *kcontrol, } } } +done: return 0; } @@ -15486,6 +16187,96 @@ static struct snd_pcm_ops msm_routing_pcm_ops = { .prepare = msm_pcm_routing_prepare, }; +int msm_routing_set_downmix_control_data(int be_id, int session_id, + struct asm_stream_pan_ctrl_params *dnmix_param) +{ + int i, rc = 0; + struct adm_pspd_param_data_t data; + struct audproc_chmixer_param_coeff dnmix_cfg; + uint16_t variable_payload = 0; + char *adm_params = NULL; + int port_id, copp_idx = 0; + uint32_t params_length = 0; + uint16_t ii; + uint16_t *dst_gain_ptr = NULL; + + if (be_id >= MSM_BACKEND_DAI_MAX) { + rc = -EINVAL; + return rc; + } + port_id = msm_bedais[be_id].port_id; + copp_idx = adm_get_default_copp_idx(port_id); + pr_debug("%s: port_id - %d, copp_idx %d session id - %d\n", + __func__, port_id, copp_idx, session_id); + + variable_payload = dnmix_param->num_output_channels * sizeof(uint16_t)+ + dnmix_param->num_input_channels * sizeof(uint16_t) + + dnmix_param->num_output_channels * + dnmix_param->num_input_channels * sizeof(uint16_t); + i = (variable_payload % sizeof(uint32_t)); + variable_payload += (i == 0) ? 0 : sizeof(uint32_t) - i; + + params_length = variable_payload + + sizeof(struct adm_pspd_param_data_t) + + sizeof(struct audproc_chmixer_param_coeff); + adm_params = kzalloc(params_length, GFP_KERNEL); + if (!adm_params) { + rc = -ENOMEM; + goto end; + } + + data.module_id = AUDPROC_MODULE_ID_CHMIXER; + data.param_id = AUDPROC_CHMIXER_PARAM_ID_COEFF; + data.param_size = sizeof(struct audproc_chmixer_param_coeff) + + variable_payload; + data.reserved = 0; + memcpy((u8 *)adm_params, &data, sizeof(struct adm_pspd_param_data_t)); + + dnmix_cfg.index = 0; + dnmix_cfg.num_output_channels = dnmix_param->num_output_channels; + dnmix_cfg.num_input_channels = dnmix_param->num_input_channels; + memcpy(((u8 *)adm_params + + sizeof(struct adm_pspd_param_data_t)), + &dnmix_cfg, sizeof(struct audproc_chmixer_param_coeff)); + + memcpy(((u8 *)adm_params + + sizeof(struct adm_pspd_param_data_t) + + sizeof(struct audproc_chmixer_param_coeff)), + dnmix_param->output_channel_map, + dnmix_param->num_output_channels * sizeof(uint16_t)); + memcpy(((u8 *)adm_params + + sizeof(struct adm_pspd_param_data_t) + + sizeof(struct audproc_chmixer_param_coeff) + + dnmix_param->num_output_channels * sizeof(uint16_t)), + dnmix_param->input_channel_map, + dnmix_param->num_input_channels * sizeof(uint16_t)); + + dst_gain_ptr = (uint16_t *) ((u8 *)adm_params + + sizeof(struct adm_pspd_param_data_t) + + sizeof(struct audproc_chmixer_param_coeff) + + (dnmix_param->num_output_channels * sizeof(uint16_t)) + + (dnmix_param->num_input_channels * sizeof(uint16_t))); + for (ii = 0; ii < dnmix_param->num_output_channels * + dnmix_param->num_input_channels; ii++) + dst_gain_ptr[ii] = (uint16_t) dnmix_param->gain[ii]; + + if (params_length) { + rc = adm_set_pspd_matrix_params(port_id, + copp_idx, + session_id, + adm_params, + params_length); + if (rc) { + pr_err("%s: send params failed rc=%d\n", __func__, rc); + rc = -EINVAL; + } + } +end: + kfree(adm_params); + return rc; +} + + /* Not used but frame seems to require it */ static int msm_routing_probe(struct snd_soc_platform *platform) { diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h index 19e726001d25..21dfa48308c3 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h +++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h @@ -56,8 +56,8 @@ #define LPASS_BE_SEC_MI2S_TX "SEC_MI2S_TX" #define LPASS_BE_PRI_MI2S_RX "PRI_MI2S_RX" #define LPASS_BE_PRI_MI2S_TX "PRI_MI2S_TX" -#define LPASS_BE_TERT_MI2S_RX "TERTIARY_MI2S_RX" -#define LPASS_BE_TERT_MI2S_TX "TERTIARY_MI2S_TX" +#define LPASS_BE_TERT_MI2S_RX "TERT_MI2S_RX" +#define LPASS_BE_TERT_MI2S_TX "TERT_MI2S_TX" #define LPASS_BE_AUDIO_I2S_RX "AUDIO_I2S_RX" #define LPASS_BE_STUB_RX "STUB_RX" #define LPASS_BE_STUB_TX "STUB_TX" @@ -193,6 +193,13 @@ enum { MSM_FRONTEND_DAI_MULTIMEDIA18, MSM_FRONTEND_DAI_MULTIMEDIA19, MSM_FRONTEND_DAI_MULTIMEDIA20, + MSM_FRONTEND_DAI_MULTIMEDIA21, + MSM_FRONTEND_DAI_MULTIMEDIA22, + MSM_FRONTEND_DAI_MULTIMEDIA23, + MSM_FRONTEND_DAI_MULTIMEDIA24, + MSM_FRONTEND_DAI_MULTIMEDIA25, + MSM_FRONTEND_DAI_MULTIMEDIA26, + MSM_FRONTEND_DAI_MULTIMEDIA27, MSM_FRONTEND_DAI_CS_VOICE, MSM_FRONTEND_DAI_VOIP, MSM_FRONTEND_DAI_AFE_RX, @@ -218,8 +225,8 @@ enum { MSM_FRONTEND_DAI_MAX, }; -#define MSM_FRONTEND_DAI_MM_SIZE (MSM_FRONTEND_DAI_MULTIMEDIA20 + 1) -#define MSM_FRONTEND_DAI_MM_MAX_ID MSM_FRONTEND_DAI_MULTIMEDIA20 +#define MSM_FRONTEND_DAI_MM_SIZE (MSM_FRONTEND_DAI_MULTIMEDIA27 + 1) +#define MSM_FRONTEND_DAI_MM_MAX_ID MSM_FRONTEND_DAI_MULTIMEDIA27 enum { MSM_BACKEND_DAI_PRI_I2S_RX = 0, @@ -392,12 +399,20 @@ enum { #define RELEASE_LOCK 0 #define ACQUIRE_LOCK 1 -#define MSM_BACKEND_DAI_PP_PARAMS_REQ_MAX 2 #define HDMI_RX_ID 0x8001 -#define ADM_PP_PARAM_MUTE_ID 0 -#define ADM_PP_PARAM_MUTE_BIT 1 -#define ADM_PP_PARAM_LATENCY_ID 1 -#define ADM_PP_PARAM_LATENCY_BIT 2 + +enum { + ADM_PP_PARAM_MUTE_ID, + ADM_PP_PARAM_LATENCY_ID, + ADM_PP_PARAM_LIMITER_ID +}; + +enum { + ADM_PP_PARAM_MUTE_BIT = 0x1, + ADM_PP_PARAM_LATENCY_BIT = 0x2, + ADM_PP_PARAM_LIMITER_BIT = 0x4 +}; + #define BE_DAI_PORT_SESSIONS_IDX_MAX 4 #define BE_DAI_FE_SESSIONS_IDX_MAX 2 @@ -483,4 +498,6 @@ int msm_pcm_routing_reg_stream_app_type_cfg( int msm_pcm_routing_get_stream_app_type_cfg( int fedai_id, int session_type, int *be_id, struct msm_pcm_stream_app_type_cfg *cfg_data); +int msm_routing_set_downmix_control_data(int be_id, int session_id, + struct asm_stream_pan_ctrl_params *pan_param); #endif /*_MSM_PCM_H*/ diff --git a/sound/soc/msm/qdsp6v2/msm-qti-pp-config.c b/sound/soc/msm/qdsp6v2/msm-qti-pp-config.c index cac28f43e5ae..65f5167d9dee 100644 --- a/sound/soc/msm/qdsp6v2/msm-qti-pp-config.c +++ b/sound/soc/msm/qdsp6v2/msm-qti-pp-config.c @@ -298,11 +298,11 @@ int msm_qti_pp_send_stereo_to_custom_stereo_cmd(int port_id, int copp_idx, *update_params_value16++ = op_FR_ip_FR_weight; avail_length = avail_length - (4 * sizeof(uint16_t)); if (params_length) { - rc = adm_set_stereo_to_custom_stereo(port_id, - copp_idx, - session_id, - params_value, - params_length); + rc = adm_set_pspd_matrix_params(port_id, + copp_idx, + session_id, + params_value, + params_length); if (rc) { pr_err("%s: send params failed rc=%d\n", __func__, rc); kfree(params_value); diff --git a/sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c index dbda90c3616d..fdeb8a15ffee 100644 --- a/sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c @@ -27,6 +27,8 @@ #include <sound/initval.h> #include <sound/control.h> #include <sound/q6asm-v2.h> +#include <sound/q6core.h> +#include <sound/q6audio-v2.h> #include <sound/pcm_params.h> #include <sound/timer.h> #include <sound/tlv.h> @@ -40,11 +42,21 @@ #include "msm-qti-pp-config.h" #define LOOPBACK_SESSION_MAX_NUM_STREAMS 2 +/* Max volume corresponding to 24dB */ +#define TRANSCODE_LR_VOL_MAX_STEPS 0xFFFF + +#define APP_TYPE_CONFIG_IDX_APP_TYPE 0 +#define APP_TYPE_CONFIG_IDX_ACDB_ID 1 +#define APP_TYPE_CONFIG_IDX_SAMPLE_RATE 2 +#define APP_TYPE_CONFIG_IDX_BE_ID 3 static DEFINE_MUTEX(transcode_loopback_session_lock); struct trans_loopback_pdata { struct snd_compr_stream *cstream[MSM_FRONTEND_DAI_MAX]; + int32_t ion_fd[MSM_FRONTEND_DAI_MAX]; + uint32_t master_gain; + int perf_mode; }; struct loopback_stream { @@ -79,6 +91,7 @@ struct msm_transcode_loopback { int session_id; struct audio_client *audio_client; + int32_t shm_ion_fd; }; /* Transcode loopback global info struct */ @@ -100,7 +113,7 @@ static void loopback_event_handler(uint32_t opcode, return; } - cstream = trans->source.cstream; + cstream = trans->sink.cstream; ac = trans->audio_client; /* @@ -112,7 +125,8 @@ static void loopback_event_handler(uint32_t opcode, switch (opcode) { case ASM_STREAM_CMD_ENCDEC_EVENTS: case ASM_IEC_61937_MEDIA_FMT_EVENT: - pr_debug("%s: ASM_IEC_61937_MEDIA_FMT_EVENT\n", __func__); + pr_debug("%s: Handling stream event : 0X%x\n", + __func__, opcode); rtd = cstream->private_data; if (!rtd) { pr_err("%s: rtd is NULL\n", __func__); @@ -178,6 +192,40 @@ static void populate_codec_list(struct msm_transcode_loopback *trans, } } +static int msm_transcode_map_unmap_fd(int fd, bool add_pages) +{ + ion_phys_addr_t paddr; + size_t pa_len = 0; + int ret = 0; + u8 assign_type; + + if (add_pages) + assign_type = HLOS_TO_ADSP; + else + assign_type = ADSP_TO_HLOS; + + ret = msm_audio_ion_phys_assign("audio_lib_mem_client", fd, + &paddr, &pa_len, assign_type); + if (ret) { + pr_err("%s: audio lib ION phys failed, rc = %d\n", __func__, + ret); + goto done; + } + + ret = q6core_add_remove_pool_pages(paddr, pa_len, + ADSP_MEMORY_MAP_HLOS_PHYSPOOL, add_pages); + if (ret) { + pr_err("%s: add remove pages failed, rc = %d\n", __func__, ret); + /* Assign back to HLOS if add pages cmd failed */ + if (add_pages) + msm_audio_ion_phys_assign("audio_lib_mem_client", fd, + &paddr, &pa_len, ADSP_TO_HLOS); + } + +done: + return ret; +} + static int msm_transcode_loopback_open(struct snd_compr_stream *cstream) { int ret = 0; @@ -222,6 +270,14 @@ static int msm_transcode_loopback_open(struct snd_compr_stream *cstream) ret = -EINVAL; goto exit; } + msm_adsp_init_mixer_ctl_pp_event_queue(rtd); + if (pdata->ion_fd[rtd->dai_link->be_id] > 0) { + ret = msm_transcode_map_unmap_fd( + pdata->ion_fd[rtd->dai_link->be_id], + true); + if (ret < 0) + goto exit; + } } pr_debug("%s: num stream%d, stream name %s\n", __func__, @@ -236,8 +292,7 @@ static int msm_transcode_loopback_open(struct snd_compr_stream *cstream) } runtime->private_data = trans; - if (trans->num_streams == 1) - msm_adsp_init_mixer_ctl_pp_event_queue(rtd); + exit: mutex_unlock(&trans->lock); return ret; @@ -273,7 +328,11 @@ static int msm_transcode_loopback_free(struct snd_compr_stream *cstream) struct snd_compr_runtime *runtime = cstream->runtime; struct msm_transcode_loopback *trans = runtime->private_data; struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(cstream); + struct trans_loopback_pdata *pdata = snd_soc_platform_get_drvdata( + rtd->platform); int ret = 0; + ion_phys_addr_t paddr; + size_t pa_len = 0; mutex_lock(&trans->lock); @@ -282,14 +341,27 @@ static int msm_transcode_loopback_free(struct snd_compr_stream *cstream) trans->num_streams--; stop_transcoding(trans); - if (cstream->direction == SND_COMPRESS_PLAYBACK) + if (cstream->direction == SND_COMPRESS_PLAYBACK) { memset(&trans->sink, 0, sizeof(struct loopback_stream)); - else if (cstream->direction == SND_COMPRESS_CAPTURE) + msm_adsp_clean_mixer_ctl_pp_event_queue(rtd); + if (trans->shm_ion_fd > 0) { + msm_audio_ion_phys_assign("audio_shm_mem_client", + trans->shm_ion_fd, + &paddr, &pa_len, + ADSP_TO_HLOS); + trans->shm_ion_fd = 0; + } + if (pdata->ion_fd[rtd->dai_link->be_id] > 0) { + msm_transcode_map_unmap_fd( + pdata->ion_fd[rtd->dai_link->be_id], + false); + pdata->ion_fd[rtd->dai_link->be_id] = 0; + } + } else if (cstream->direction == SND_COMPRESS_CAPTURE) { memset(&trans->source, 0, sizeof(struct loopback_stream)); + } trans->session_state = LOOPBACK_SESSION_CLOSE; - if (trans->num_streams == 1) - msm_adsp_clean_mixer_ctl_pp_event_queue(rtd); mutex_unlock(&trans->lock); return ret; } @@ -336,6 +408,8 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream, struct msm_transcode_loopback *trans = runtime->private_data; struct snd_soc_pcm_runtime *soc_pcm_rx; struct snd_soc_pcm_runtime *soc_pcm_tx; + struct snd_soc_pcm_runtime *rtd; + struct trans_loopback_pdata *pdata; uint32_t bit_width = 16; int ret = 0; @@ -346,6 +420,9 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream, mutex_lock(&trans->lock); + rtd = snd_pcm_substream_chip(cstream); + pdata = snd_soc_platform_get_drvdata(rtd->platform); + if (cstream->direction == SND_COMPRESS_PLAYBACK) { if (codec_param->codec.id == SND_AUDIOCODEC_PCM) { trans->sink.codec_format = @@ -427,7 +504,7 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream, pr_debug("%s: ASM client allocated, callback %pK\n", __func__, loopback_event_handler); trans->session_id = trans->audio_client->session; - trans->audio_client->perf_mode = false; + trans->audio_client->perf_mode = pdata->perf_mode; ret = q6asm_open_transcode_loopback(trans->audio_client, bit_width, trans->source.codec_format, @@ -446,17 +523,17 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream, if (trans->source.codec_format != FORMAT_LINEAR_PCM) msm_pcm_routing_reg_phy_compr_stream( soc_pcm_tx->dai_link->be_id, - trans->audio_client->perf_mode, + false, trans->session_id, SNDRV_PCM_STREAM_CAPTURE, - true); + COMPRESSED_PASSTHROUGH_GEN); else msm_pcm_routing_reg_phy_stream( soc_pcm_tx->dai_link->be_id, trans->audio_client->perf_mode, trans->session_id, SNDRV_PCM_STREAM_CAPTURE); - + /* Opening Rx ADM in LOW_LATENCY mode by default */ msm_pcm_routing_reg_phy_stream( soc_pcm_rx->dai_link->be_id, trans->audio_client->perf_mode, @@ -492,6 +569,46 @@ static int msm_transcode_loopback_get_caps(struct snd_compr_stream *cstream, return 0; } +static int msm_transcode_loopback_set_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct snd_soc_pcm_runtime *rtd; + struct trans_loopback_pdata *pdata; + + if (!metadata || !cstream) { + pr_err("%s: Invalid arguments\n", __func__); + return -EINVAL; + } + + rtd = snd_pcm_substream_chip(cstream); + pdata = snd_soc_platform_get_drvdata(rtd->platform); + + switch (metadata->key) { + case SNDRV_COMPRESS_LATENCY_MODE: + { + switch (metadata->value[0]) { + case SNDRV_COMPRESS_LEGACY_LATENCY_MODE: + pdata->perf_mode = LEGACY_PCM_MODE; + break; + case SNDRV_COMPRESS_LOW_LATENCY_MODE: + pdata->perf_mode = LOW_LATENCY_PCM_MODE; + break; + default: + pr_debug("%s: Unsupported latency mode %d, default to Legacy\n", + __func__, metadata->value[0]); + pdata->perf_mode = LEGACY_PCM_MODE; + break; + } + } + break; + default: + pr_debug("%s: Unsupported metadata %d\n", + __func__, metadata->key); + break; + } + return 0; +} + static int msm_transcode_stream_cmd_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -556,7 +673,7 @@ done: return ret; } -static int msm_transcode_ion_fd_map_put(struct snd_kcontrol *kcontrol, +static int msm_transcode_shm_ion_fd_map_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); @@ -565,7 +682,6 @@ static int msm_transcode_ion_fd_map_put(struct snd_kcontrol *kcontrol, snd_soc_component_get_drvdata(comp); struct snd_compr_stream *cstream = NULL; struct msm_transcode_loopback *prtd; - int fd; int ret = 0; if (fe_id >= MSM_FRONTEND_DAI_MAX) { @@ -595,10 +711,34 @@ static int msm_transcode_ion_fd_map_put(struct snd_kcontrol *kcontrol, goto done; } - memcpy(&fd, ucontrol->value.bytes.data, sizeof(fd)); - ret = q6asm_send_ion_fd(prtd->audio_client, fd); + memcpy(&prtd->shm_ion_fd, ucontrol->value.bytes.data, + sizeof(prtd->shm_ion_fd)); + ret = q6asm_audio_map_shm_fd(prtd->audio_client, prtd->shm_ion_fd); if (ret < 0) - pr_err("%s: failed to register ion fd\n", __func__); + pr_err("%s: failed to map shm mem\n", __func__); +done: + return ret; +} + + +static int msm_transcode_lib_ion_fd_map_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + unsigned long fe_id = kcontrol->private_value; + struct trans_loopback_pdata *pdata = (struct trans_loopback_pdata *) + snd_soc_component_get_drvdata(comp); + int ret = 0; + + if (fe_id >= MSM_FRONTEND_DAI_MAX) { + pr_err("%s Received out of bounds invalid fe_id %lu\n", + __func__, fe_id); + ret = -EINVAL; + goto done; + } + + memcpy(&pdata->ion_fd[fe_id], ucontrol->value.bytes.data, + sizeof(pdata->ion_fd[fe_id])); done: return ret; } @@ -662,6 +802,141 @@ done: return ret; } +static int msm_transcode_playback_app_type_cfg_put( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u64 fe_id = kcontrol->private_value; + int session_type = SESSION_TYPE_RX; + int be_id = ucontrol->value.integer.value[APP_TYPE_CONFIG_IDX_BE_ID]; + struct msm_pcm_stream_app_type_cfg cfg_data = {0, 0, 48000}; + int ret = 0; + + cfg_data.app_type = ucontrol->value.integer.value[ + APP_TYPE_CONFIG_IDX_APP_TYPE]; + cfg_data.acdb_dev_id = ucontrol->value.integer.value[ + APP_TYPE_CONFIG_IDX_ACDB_ID]; + if (ucontrol->value.integer.value[APP_TYPE_CONFIG_IDX_SAMPLE_RATE] != 0) + cfg_data.sample_rate = ucontrol->value.integer.value[ + APP_TYPE_CONFIG_IDX_SAMPLE_RATE]; + pr_debug("%s: fe_id %llu session_type %d be_id %d app_type %d acdb_dev_id %d sample_rate- %d\n", + __func__, fe_id, session_type, be_id, + cfg_data.app_type, cfg_data.acdb_dev_id, cfg_data.sample_rate); + ret = msm_pcm_routing_reg_stream_app_type_cfg(fe_id, session_type, + be_id, &cfg_data); + if (ret < 0) + pr_err("%s: msm_transcode_playback_stream_app_type_cfg set failed returned %d\n", + __func__, ret); + + return ret; +} + +static int msm_transcode_playback_app_type_cfg_get( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u64 fe_id = kcontrol->private_value; + int session_type = SESSION_TYPE_RX; + int be_id = 0; + struct msm_pcm_stream_app_type_cfg cfg_data = {0}; + int ret = 0; + + ret = msm_pcm_routing_get_stream_app_type_cfg(fe_id, session_type, + &be_id, &cfg_data); + if (ret < 0) { + pr_err("%s: msm_transcode_playback_stream_app_type_cfg get failed returned %d\n", + __func__, ret); + goto done; + } + + ucontrol->value.integer.value[APP_TYPE_CONFIG_IDX_APP_TYPE] = + cfg_data.app_type; + ucontrol->value.integer.value[APP_TYPE_CONFIG_IDX_ACDB_ID] = + cfg_data.acdb_dev_id; + ucontrol->value.integer.value[APP_TYPE_CONFIG_IDX_SAMPLE_RATE] = + cfg_data.sample_rate; + ucontrol->value.integer.value[APP_TYPE_CONFIG_IDX_BE_ID] = be_id; + pr_debug("%s: fedai_id %llu, session_type %d, be_id %d, app_type %d, acdb_dev_id %d, sample_rate %d\n", + __func__, fe_id, session_type, be_id, + cfg_data.app_type, cfg_data.acdb_dev_id, cfg_data.sample_rate); +done: + return ret; +} + +static int msm_transcode_set_volume(struct snd_compr_stream *cstream, + uint32_t master_gain) +{ + int rc = 0; + struct msm_transcode_loopback *prtd; + struct snd_soc_pcm_runtime *rtd; + + pr_debug("%s: master_gain %d\n", __func__, master_gain); + if (!cstream || !cstream->runtime) { + pr_err("%s: session not active\n", __func__); + return -EPERM; + } + rtd = cstream->private_data; + prtd = cstream->runtime->private_data; + + if (!rtd || !rtd->platform || !prtd || !prtd->audio_client) { + pr_err("%s: invalid rtd, prtd or audio client", __func__); + return -EINVAL; + } + + rc = q6asm_set_volume(prtd->audio_client, master_gain); + if (rc < 0) + pr_err("%s: Send vol gain command failed rc=%d\n", + __func__, rc); + + return rc; +} + +static int msm_transcode_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + unsigned long fe_id = kcontrol->private_value; + struct trans_loopback_pdata *pdata = (struct trans_loopback_pdata *) + snd_soc_component_get_drvdata(comp); + struct snd_compr_stream *cstream = NULL; + uint32_t ret = 0; + + if (fe_id >= MSM_FRONTEND_DAI_MAX) { + pr_err("%s Received out of bounds fe_id %lu\n", + __func__, fe_id); + return -EINVAL; + } + + cstream = pdata->cstream[fe_id]; + pdata->master_gain = ucontrol->value.integer.value[0]; + + pr_debug("%s: fe_id %lu master_gain %d\n", + __func__, fe_id, pdata->master_gain); + if (cstream) + ret = msm_transcode_set_volume(cstream, pdata->master_gain); + return ret; +} + +static int msm_transcode_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + unsigned long fe_id = kcontrol->private_value; + + struct trans_loopback_pdata *pdata = (struct trans_loopback_pdata *) + snd_soc_component_get_drvdata(comp); + + if (fe_id >= MSM_FRONTEND_DAI_MAX) { + pr_err("%s Received out of bound fe_id %lu\n", __func__, fe_id); + return -EINVAL; + } + + pr_debug("%s: fe_id %lu\n", __func__, fe_id); + ucontrol->value.integer.value[0] = pdata->master_gain; + + return 0; +} + static int msm_transcode_stream_cmd_control( struct snd_soc_pcm_runtime *rtd) { @@ -772,7 +1047,8 @@ done: return ret; } -static int msm_transcode_add_ion_fd_cmd_control(struct snd_soc_pcm_runtime *rtd) +static int msm_transcode_add_shm_ion_fd_cmd_control( + struct snd_soc_pcm_runtime *rtd) { const char *mixer_ctl_name = "Playback ION FD"; const char *deviceNo = "NN"; @@ -784,7 +1060,53 @@ static int msm_transcode_add_ion_fd_cmd_control(struct snd_soc_pcm_runtime *rtd) .name = "?", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, .info = msm_adsp_stream_cmd_info, - .put = msm_transcode_ion_fd_map_put, + .put = msm_transcode_shm_ion_fd_map_put, + .private_value = 0, + } + }; + + if (!rtd) { + pr_err("%s NULL rtd\n", __func__); + ret = -EINVAL; + goto done; + } + + ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1; + mixer_str = kzalloc(ctl_len, GFP_KERNEL); + if (!mixer_str) { + ret = -ENOMEM; + goto done; + } + + snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device); + fe_ion_fd_config_control[0].name = mixer_str; + fe_ion_fd_config_control[0].private_value = rtd->dai_link->be_id; + pr_debug("%s: Registering new mixer ctl %s\n", __func__, mixer_str); + ret = snd_soc_add_platform_controls(rtd->platform, + fe_ion_fd_config_control, + ARRAY_SIZE(fe_ion_fd_config_control)); + if (ret < 0) + pr_err("%s: failed to add ctl %s\n", __func__, mixer_str); + + kfree(mixer_str); +done: + return ret; +} + +static int msm_transcode_add_lib_ion_fd_cmd_control( + struct snd_soc_pcm_runtime *rtd) +{ + const char *mixer_ctl_name = "Playback ION LIB FD"; + const char *deviceNo = "NN"; + char *mixer_str = NULL; + int ctl_len = 0, ret = 0; + struct snd_kcontrol_new fe_ion_fd_config_control[1] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "?", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = msm_adsp_stream_cmd_info, + .put = msm_transcode_lib_ion_fd_map_put, .private_value = 0, } }; @@ -863,6 +1185,99 @@ done: return ret; } +static int msm_transcode_app_type_cfg_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 5; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0xFFFFFFFF; + return 0; +} + +static int msm_transcode_add_app_type_cfg_control( + struct snd_soc_pcm_runtime *rtd) +{ + char mixer_str[32]; + int rc = 0; + struct snd_kcontrol_new fe_app_type_cfg_control[1] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = msm_transcode_app_type_cfg_info, + .put = msm_transcode_playback_app_type_cfg_put, + .get = msm_transcode_playback_app_type_cfg_get, + .private_value = 0, + } + }; + + if (!rtd) { + pr_err("%s NULL rtd\n", __func__); + + return -EINVAL; + } + + if (rtd->compr->direction == SND_COMPRESS_PLAYBACK) { + + snprintf(mixer_str, sizeof(mixer_str), + "Audio Stream %d App Type Cfg", + rtd->pcm->device); + + fe_app_type_cfg_control[0].name = mixer_str; + fe_app_type_cfg_control[0].private_value = rtd->dai_link->be_id; + + fe_app_type_cfg_control[0].put = + msm_transcode_playback_app_type_cfg_put; + fe_app_type_cfg_control[0].get = + msm_transcode_playback_app_type_cfg_get; + + pr_debug("Registering new mixer ctl %s", mixer_str); + snd_soc_add_platform_controls(rtd->platform, + fe_app_type_cfg_control, + ARRAY_SIZE(fe_app_type_cfg_control)); + } + + return rc; +} +static int msm_transcode_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = TRANSCODE_LR_VOL_MAX_STEPS; + return 0; +} + +static int msm_transcode_add_volume_control(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_kcontrol_new fe_volume_control[1] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Transcode Loopback Rx Volume", + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = msm_transcode_volume_info, + .get = msm_transcode_volume_get, + .put = msm_transcode_volume_put, + .private_value = 0, + } + }; + + if (!rtd) { + pr_err("%s NULL rtd\n", __func__); + return -EINVAL; + } + if (rtd->compr->direction == SND_COMPRESS_PLAYBACK) { + fe_volume_control[0].private_value = rtd->dai_link->be_id; + pr_debug("Registering new mixer ctl %s", + fe_volume_control[0].name); + snd_soc_add_platform_controls(rtd->platform, fe_volume_control, + ARRAY_SIZE(fe_volume_control)); + } + return 0; +} + static int msm_transcode_loopback_new(struct snd_soc_pcm_runtime *rtd) { int rc; @@ -876,9 +1291,14 @@ static int msm_transcode_loopback_new(struct snd_soc_pcm_runtime *rtd) pr_err("%s: ADSP Stream callback Control open failed\n", __func__); - rc = msm_transcode_add_ion_fd_cmd_control(rtd); + rc = msm_transcode_add_shm_ion_fd_cmd_control(rtd); if (rc) - pr_err("%s: Could not add transcode ion fd Control\n", + pr_err("%s: Could not add transcode shm ion fd Control\n", + __func__); + + rc = msm_transcode_add_lib_ion_fd_cmd_control(rtd); + if (rc) + pr_err("%s: Could not add transcode lib ion fd Control\n", __func__); rc = msm_transcode_add_event_ack_cmd_control(rtd); @@ -886,6 +1306,16 @@ static int msm_transcode_loopback_new(struct snd_soc_pcm_runtime *rtd) pr_err("%s: Could not add transcode event ack Control\n", __func__); + rc = msm_transcode_add_app_type_cfg_control(rtd); + if (rc) + pr_err("%s: Could not add Compr App Type Cfg Control\n", + __func__); + + rc = msm_transcode_add_volume_control(rtd); + if (rc) + pr_err("%s: Could not add transcode volume Control\n", + __func__); + return 0; } @@ -895,6 +1325,7 @@ static struct snd_compr_ops msm_transcode_loopback_ops = { .trigger = msm_transcode_loopback_trigger, .set_params = msm_transcode_loopback_set_params, .get_caps = msm_transcode_loopback_get_caps, + .set_metadata = msm_transcode_loopback_set_metadata, }; @@ -909,14 +1340,26 @@ static int msm_transcode_loopback_probe(struct snd_soc_platform *platform) if (!pdata) return -ENOMEM; + pdata->perf_mode = LOW_LATENCY_PCM_MODE; snd_soc_platform_set_drvdata(platform, pdata); return 0; } +static int msm_transcode_loopback_remove(struct snd_soc_platform *platform) +{ + struct trans_loopback_pdata *pdata = NULL; + + pdata = (struct trans_loopback_pdata *) + snd_soc_platform_get_drvdata(platform); + kfree(pdata); + return 0; +} + static struct snd_soc_platform_driver msm_soc_platform = { .probe = msm_transcode_loopback_probe, .compr_ops = &msm_transcode_loopback_ops, .pcm_new = msm_transcode_loopback_new, + .remove = msm_transcode_loopback_remove, }; static int msm_transcode_dev_probe(struct platform_device *pdev) diff --git a/sound/soc/msm/qdsp6v2/q6adm.c b/sound/soc/msm/qdsp6v2/q6adm.c index 61dd478b97e4..018681309f2e 100644 --- a/sound/soc/msm/qdsp6v2/q6adm.c +++ b/sound/soc/msm/qdsp6v2/q6adm.c @@ -48,12 +48,6 @@ #define DS2_ADM_COPP_TOPOLOGY_ID 0xFFFFFFFF #endif -/* ENUM for adm_status */ -enum adm_cal_status { - ADM_STATUS_CALIBRATION_REQUIRED = 0, - ADM_STATUS_MAX, -}; - struct adm_copp { atomic_t id[AFE_MAX_PORTS][MAX_COPPS_PER_PORT]; @@ -781,9 +775,9 @@ fail_cmd: return ret; } -int adm_set_stereo_to_custom_stereo(int port_id, int copp_idx, - unsigned int session_id, char *params, - uint32_t params_length) +int adm_set_pspd_matrix_params(int port_id, int copp_idx, + unsigned int session_id, char *params, + uint32_t params_length) { struct adm_cmd_set_pspd_mtmx_strtr_params_v5 *adm_params = NULL; int sz, rc = 0, port_idx; @@ -1413,6 +1407,7 @@ static int32_t adm_callback(struct apr_client_data *data, void *priv) case ADM_CMD_DEVICE_OPEN_V5: case ADM_CMD_DEVICE_CLOSE_V5: case ADM_CMD_DEVICE_OPEN_V6: + case ADM_CMD_SET_MTMX_STRTR_DEV_PARAMS_V1: pr_debug("%s: Basic callback received, wake up.\n", __func__); atomic_set(&this_adm.copp.stat[port_idx] @@ -1999,13 +1994,11 @@ static struct cal_block_data *adm_find_cal_by_path(int cal_index, int path) if (cal_index == ADM_AUDPROC_CAL) { audproc_cal_info = cal_block->cal_info; - if ((audproc_cal_info->path == path) && - (cal_block->cal_data.size > 0)) + if (audproc_cal_info->path == path) return cal_block; } else if (cal_index == ADM_AUDVOL_CAL) { audvol_cal_info = cal_block->cal_info; - if ((audvol_cal_info->path == path) && - (cal_block->cal_data.size > 0)) + if (audvol_cal_info->path == path) return cal_block; } } @@ -2032,14 +2025,12 @@ static struct cal_block_data *adm_find_cal_by_app_type(int cal_index, int path, if (cal_index == ADM_AUDPROC_CAL) { audproc_cal_info = cal_block->cal_info; if ((audproc_cal_info->path == path) && - (audproc_cal_info->app_type == app_type) && - (cal_block->cal_data.size > 0)) + (audproc_cal_info->app_type == app_type)) return cal_block; } else if (cal_index == ADM_AUDVOL_CAL) { audvol_cal_info = cal_block->cal_info; if ((audvol_cal_info->path == path) && - (audvol_cal_info->app_type == app_type) && - (cal_block->cal_data.size > 0)) + (audvol_cal_info->app_type == app_type)) return cal_block; } } @@ -2070,15 +2061,13 @@ static struct cal_block_data *adm_find_cal(int cal_index, int path, if ((audproc_cal_info->path == path) && (audproc_cal_info->app_type == app_type) && (audproc_cal_info->acdb_id == acdb_id) && - (audproc_cal_info->sample_rate == sample_rate) && - (cal_block->cal_data.size > 0)) + (audproc_cal_info->sample_rate == sample_rate)) return cal_block; } else if (cal_index == ADM_AUDVOL_CAL) { audvol_cal_info = cal_block->cal_info; if ((audvol_cal_info->path == path) && (audvol_cal_info->app_type == app_type) && - (audvol_cal_info->acdb_id == acdb_id) && - (cal_block->cal_data.size > 0)) + (audvol_cal_info->acdb_id == acdb_id)) return cal_block; } } @@ -2375,7 +2364,8 @@ int adm_open(int port_id, int path, int rate, int channel_mode, int topology, struct adm_cmd_device_open_v5 open; struct adm_cmd_device_open_v6 open_v6; int ret = 0; - int port_idx, copp_idx, flags; + int port_idx, flags; + int copp_idx = -1; int tmp_port = q6audio_get_port_id(port_id); pr_debug("%s:port %#x path:%d rate:%d mode:%d perf_mode:%d,topo_id %d\n", @@ -2429,8 +2419,17 @@ int adm_open(int port_id, int path, int rate, int channel_mode, int topology, (topology == VPM_TX_DM_RFECNS_COPP_TOPOLOGY)) rate = 16000; - copp_idx = adm_get_idx_if_copp_exists(port_idx, topology, perf_mode, - rate, bit_width, app_type); + /* + * Routing driver reuses the same adm for streams with the same + * app_type, sample_rate etc. + * This isn't allowed for ULL streams as per the DSP interface + */ + if (perf_mode != ULTRA_LOW_LATENCY_PCM_MODE) + copp_idx = adm_get_idx_if_copp_exists(port_idx, topology, + perf_mode, + rate, bit_width, + app_type); + if (copp_idx < 0) { copp_idx = adm_get_next_available_copp(port_idx); if (copp_idx >= MAX_COPPS_PER_PORT) { @@ -2701,6 +2700,97 @@ fail_cmd: return; } + +static int adm_set_mtmx_params_v1(int port_idx, int copp_idx, + int params_length, void *params) +{ + struct adm_cmd_set_mtmx_params_v1 *adm_params = NULL; + int rc = 0; + int sz; + + sz = sizeof(*adm_params) + params_length; + adm_params = kzalloc(sz, GFP_KERNEL); + if (!adm_params) + return -ENOMEM; + + memcpy(((u8 *)adm_params + sizeof(*adm_params)), + params, params_length); + adm_params->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD, + APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER); + adm_params->hdr.pkt_size = sz; + adm_params->hdr.src_svc = APR_SVC_ADM; + adm_params->hdr.src_domain = APR_DOMAIN_APPS; + adm_params->hdr.src_port = 0; + adm_params->hdr.dest_svc = APR_SVC_ADM; + adm_params->hdr.dest_domain = APR_DOMAIN_ADSP; + adm_params->hdr.dest_port = + atomic_read(&this_adm.copp.id[port_idx][copp_idx]); + adm_params->hdr.token = port_idx << 16 | copp_idx; + adm_params->hdr.opcode = ADM_CMD_SET_MTMX_STRTR_DEV_PARAMS_V1; + adm_params->payload_addr_lsw = 0; + adm_params->payload_addr_msw = 0; + adm_params->mem_map_handle = 0; + adm_params->payload_size = params_length; + adm_params->copp_id = atomic_read(&this_adm.copp. + id[port_idx][copp_idx]); + + atomic_set(&this_adm.copp.stat[port_idx][copp_idx], -1); + rc = apr_send_pkt(this_adm.apr, (uint32_t *)adm_params); + if (rc < 0) { + pr_err("%s: Set params failed port_idx = 0x%x rc %d\n", + __func__, port_idx, rc); + rc = -EINVAL; + goto send_param_return; + } + /* Wait for the callback */ + rc = wait_event_timeout(this_adm.copp.wait[port_idx][copp_idx], + atomic_read(&this_adm.copp.stat[port_idx][copp_idx]) >= 0, + msecs_to_jiffies(TIMEOUT_MS)); + if (!rc) { + pr_err("%s: Set params timed out port_idx = 0x%x\n", + __func__, port_idx); + rc = -EINVAL; + goto send_param_return; + } else if (atomic_read(&this_adm.copp.stat + [port_idx][copp_idx]) > 0) { + pr_err("%s: DSP returned error[%s]\n", + __func__, adsp_err_get_err_str( + atomic_read(&this_adm.copp.stat + [port_idx][copp_idx]))); + rc = adsp_err_get_lnx_err_code( + atomic_read(&this_adm.copp.stat + [port_idx][copp_idx])); + goto send_param_return; + } + rc = 0; +send_param_return: + kfree(adm_params); + return rc; +} + +static void adm_enable_mtmx_limiter(int port_idx, int copp_idx) +{ + int rc; + struct enable_param_v6 adm_param = { {0} }; + + adm_param.param.module_id = ADM_MTMX_MODULE_STREAM_LIMITER; + adm_param.param.param_id = AUDPROC_PARAM_ID_ENABLE; + adm_param.param.param_size = sizeof(adm_param.enable); + adm_param.enable = 1; + + rc = adm_set_mtmx_params_v1(port_idx, copp_idx, + sizeof(adm_param), &adm_param); + if (rc < 0) { + pr_err("%s: adm_set_mtmx_params_v1 failed port_idx = 0x%x rc %d\n", + __func__, port_idx, rc); + goto done; + } + set_bit(ADM_STATUS_LIMITER, + (void *)&this_adm.copp.adm_status[port_idx][copp_idx]); +done: + return; +} + static void route_set_opcode_matrix_id( struct adm_cmd_matrix_map_routings_v5 **route_addr, int path, uint32_t passthr_mode) @@ -2797,6 +2887,11 @@ int adm_matrix_map(int path, struct route_payload payload_map, int perf_mode, copp_idx = payload_map.copp_idx[i]; copps_list[i] = atomic_read(&this_adm.copp.id[port_idx] [copp_idx]); + if (test_bit(ADM_STATUS_LIMITER, + (void *)&payload_map.route_status) && + ((path == ADM_PATH_PLAYBACK) || + (path == ADM_PATH_COMPRESSED_RX))) + adm_enable_mtmx_limiter(port_idx, copp_idx); } atomic_set(&this_adm.matrix_map_stat, -1); @@ -2997,6 +3092,8 @@ int adm_close(int port_id, int perf_mode, int copp_idx) clear_bit(ADM_STATUS_CALIBRATION_REQUIRED, (void *)&this_adm.copp.adm_status[port_idx][copp_idx]); + clear_bit(ADM_STATUS_LIMITER, + (void *)&this_adm.copp.adm_status[port_idx][copp_idx]); ret = apr_send_pkt(this_adm.apr, (uint32_t *)&close); if (ret < 0) { @@ -4813,8 +4910,7 @@ static int __init adm_init(void) &this_adm.copp.adm_delay_wait[i][j]); atomic_set(&this_adm.copp.topology[i][j], 0); this_adm.copp.adm_delay[i][j] = 0; - this_adm.copp.adm_status[i][j] = - ADM_STATUS_CALIBRATION_REQUIRED; + this_adm.copp.adm_status[i][j] = 0; } } diff --git a/sound/soc/msm/qdsp6v2/q6asm.c b/sound/soc/msm/qdsp6v2/q6asm.c index da156bf61610..609ae8cd82ac 100644 --- a/sound/soc/msm/qdsp6v2/q6asm.c +++ b/sound/soc/msm/qdsp6v2/q6asm.c @@ -36,6 +36,7 @@ #include <sound/apr_audio-v2.h> #include <sound/q6asm-v2.h> +#include <sound/q6core.h> #include <sound/q6audio-v2.h> #include <sound/audio_cal_utils.h> #include <sound/adsp_err.h> @@ -44,7 +45,7 @@ #define TRUE 0x01 #define FALSE 0x00 #define SESSION_MAX 8 - +#define ASM_MAX_CHANNELS 8 enum { ASM_TOPOLOGY_CAL = 0, ASM_CUSTOM_TOP_CAL, @@ -423,14 +424,22 @@ static void config_debug_fs_run(void) } } -static void config_debug_fs_write(struct audio_buffer *ab) +static void config_debug_fs_write(struct audio_buffer *ab, int offset) { if (out_enable_flag) { char zero_pattern[2] = {0x00, 0x00}; + char *data; + + if ((offset < 0) || (offset > ab->size)) { + pr_err("Invalid offset %d", offset); + return; + } + + data = (char *)ab->data + offset; /* If First two byte is non zero and last two byte is zero then it is warm output pattern */ - if ((strncmp(((char *)ab->data), zero_pattern, 2)) && - (!strncmp(((char *)ab->data + 2), zero_pattern, 2))) { + if ((strncmp(data, zero_pattern, 2)) && + (!strncmp((data + 2), zero_pattern, 2))) { do_gettimeofday(&out_warm_tv); pr_debug("%s: WARM:apr_send_pkt at %ld sec %ld microsec\n", __func__, @@ -440,8 +449,8 @@ static void config_debug_fs_write(struct audio_buffer *ab) } /* If First two byte is zero and last two byte is non zero then it is cont ouput pattern */ - else if ((!strncmp(((char *)ab->data), zero_pattern, 2)) - && (strncmp(((char *)ab->data + 2), zero_pattern, 2))) { + else if ((!strncmp(data, zero_pattern, 2)) + && (strncmp((data + 2), zero_pattern, 2))) { do_gettimeofday(&out_cont_tv); pr_debug("%s: CONT:apr_send_pkt at %ld sec %ld microsec\n", __func__, @@ -488,7 +497,7 @@ outbuf_fail: return; } #else -static void config_debug_fs_write(struct audio_buffer *ab) +static void config_debug_fs_write(struct audio_buffer *ab, int offset) { return; } @@ -1952,13 +1961,13 @@ static int32_t q6asm_callback(struct apr_client_data *data, void *priv) port->buf[buf_index].used = 1; spin_unlock_irqrestore(&port->dsp_lock, dsp_flags); - config_debug_fs_write_cb(); for (i = 0; i < port->max_buf_cnt; i++) dev_vdbg(ac->dev, "%s %d\n", __func__, port->buf[i].used); } + config_debug_fs_write_cb(); break; } case ASM_STREAM_CMDRSP_GET_PP_PARAMS_V2: @@ -3399,11 +3408,12 @@ int q6asm_set_shared_circ_buff(struct audio_client *ac, open->shared_circ_buf_start_phy_addr_lsw = lower_32_bits(buf_circ->phys); open->shared_circ_buf_start_phy_addr_msw = - upper_32_bits(buf_circ->phys); + msm_audio_populate_upper_32_bits(buf_circ->phys); open->shared_circ_buf_size = bufsz * bufcnt; open->map_region_circ_buf.shm_addr_lsw = lower_32_bits(buf_circ->phys); - open->map_region_circ_buf.shm_addr_msw = upper_32_bits(buf_circ->phys); + open->map_region_circ_buf.shm_addr_msw = + msm_audio_populate_upper_32_bits(buf_circ->phys); open->map_region_circ_buf.mem_size_bytes = bytes_to_alloc; mutex_unlock(&ac->cmd_lock); @@ -3445,10 +3455,12 @@ int q6asm_set_shared_pos_buff(struct audio_client *ac, open->shared_pos_buf_num_regions = 1; open->shared_pos_buf_property_flag = 0x00; open->shared_pos_buf_phy_addr_lsw = lower_32_bits(buf_pos->phys); - open->shared_pos_buf_phy_addr_msw = upper_32_bits(buf_pos->phys); + open->shared_pos_buf_phy_addr_msw = + msm_audio_populate_upper_32_bits(buf_pos->phys); open->map_region_pos_buf.shm_addr_lsw = lower_32_bits(buf_pos->phys); - open->map_region_pos_buf.shm_addr_msw = upper_32_bits(buf_pos->phys); + open->map_region_pos_buf.shm_addr_msw = + msm_audio_populate_upper_32_bits(buf_pos->phys); open->map_region_pos_buf.mem_size_bytes = bytes_to_alloc; done: @@ -7106,13 +7118,10 @@ fail_cmd: return rc; } -int q6asm_send_ion_fd(struct audio_client *ac, int fd) +int q6asm_audio_map_shm_fd(struct audio_client *ac, int fd) { - struct ion_client *client; - struct ion_handle *handle; ion_phys_addr_t paddr; size_t pa_len = 0; - void *vaddr; int ret; int sz = 0; struct avs_rtic_shared_mem_addr shm; @@ -7128,19 +7137,10 @@ int q6asm_send_ion_fd(struct audio_client *ac, int fd) goto fail_cmd; } - ret = msm_audio_ion_import("audio_mem_client", - &client, - &handle, - fd, - NULL, - 0, - &paddr, - &pa_len, - &vaddr); + ret = msm_audio_ion_phys_assign("audio_shm_mem_client", fd, + &paddr, &pa_len, HLOS_TO_ADSP); if (ret) { - pr_err("%s: audio ION import failed, rc = %d\n", - __func__, ret); - ret = -ENOMEM; + pr_err("%s: shm ION phys failed, rc = %d\n", __func__, ret); goto fail_cmd; } /* get payload length */ @@ -7416,6 +7416,299 @@ int q6asm_set_softvolume_v2(struct audio_client *ac, return __q6asm_set_softvolume(ac, softvol_param, instance); } +int q6asm_set_vol_ctrl_gain_pair(struct audio_client *ac, + struct asm_stream_pan_ctrl_params *pan_param) +{ + int sz = 0; + int rc = 0; + int i = 0; + int32_t ch = 0; + struct apr_hdr hdr; + struct audproc_volume_ctrl_channel_type_gain_pair + gain_data[ASM_MAX_CHANNELS]; + struct asm_stream_cmd_set_pp_params_v2 payload_params; + struct asm_stream_param_data_v2 data; + uint16_t *asm_params = NULL; + + if (ac == NULL) { + pr_err("%s: ac is NULL\n", __func__); + rc = -EINVAL; + goto fail; + } + if (ac->apr == NULL) { + dev_err(ac->dev, "%s: ac apr handle NULL\n", __func__); + rc = -EINVAL; + goto fail; + } + + sz = sizeof(struct apr_hdr) + + sizeof(struct asm_stream_cmd_set_pp_params_v2) + + sizeof(struct asm_stream_param_data_v2) + + sizeof(uint32_t) + + (sizeof(struct audproc_volume_ctrl_channel_type_gain_pair) * + ASM_MAX_CHANNELS); + asm_params = kzalloc(sz, GFP_KERNEL); + if (!asm_params) { + rc = -ENOMEM; + goto fail; + } + + q6asm_add_hdr_async(ac, &hdr, sz, TRUE); + atomic_set(&ac->cmd_state_pp, -1); + + hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V2; + memcpy(((u8 *)asm_params), &hdr, sizeof(struct apr_hdr)); + + payload_params.data_payload_addr_lsw = 0; + payload_params.data_payload_addr_msw = 0; + payload_params.mem_map_handle = 0; + payload_params.data_payload_size = + sizeof(struct asm_stream_param_data_v2) + + sizeof(uint32_t) + + (sizeof(struct audproc_volume_ctrl_channel_type_gain_pair) * + ASM_MAX_CHANNELS); + memcpy(((u8 *)asm_params + sizeof(struct apr_hdr)), + &payload_params, + sizeof(struct asm_stream_cmd_set_pp_params_v2)); + + data.module_id = AUDPROC_MODULE_ID_VOL_CTRL; + data.param_id = AUDPROC_PARAM_ID_MULTICHANNEL_GAIN; + data.param_size = sizeof(uint32_t) + + (sizeof(struct audproc_volume_ctrl_channel_type_gain_pair) * + ASM_MAX_CHANNELS); + data.reserved = 0; + memcpy(((u8 *)asm_params + sizeof(struct apr_hdr) + + sizeof(struct asm_stream_cmd_set_pp_params_v2)), + &data, sizeof(struct asm_stream_param_data_v2)); + + ch = pan_param->num_output_channels; + memcpy(((u8 *)asm_params + sizeof(struct apr_hdr) + + sizeof(struct asm_stream_cmd_set_pp_params_v2) + + sizeof(struct asm_stream_param_data_v2)), + &ch, + sizeof(uint32_t)); + + memset(gain_data, 0, + ASM_MAX_CHANNELS * + sizeof(struct audproc_volume_ctrl_channel_type_gain_pair)); + for (i = 0; i < pan_param->num_output_channels; i++) { + gain_data[i].channel_type = + pan_param->output_channel_map[i]; + gain_data[i].gain = pan_param->gain[i]; + } + memcpy(((u8 *)asm_params + sizeof(struct apr_hdr) + + sizeof(struct asm_stream_cmd_set_pp_params_v2) + + sizeof(struct asm_stream_param_data_v2) + + sizeof(uint32_t)), + gain_data, + ASM_MAX_CHANNELS * + sizeof(struct audproc_volume_ctrl_channel_type_gain_pair)); + + rc = apr_send_pkt(ac->apr, (uint32_t *) asm_params); + if (rc < 0) { + pr_err("%s: set-params send failed paramid[0x%x] rc %d\n", + __func__, data.param_id, rc); + goto done; + } + + rc = wait_event_timeout(ac->cmd_wait, + (atomic_read(&ac->cmd_state_pp) >= 0), 5 * HZ); + if (!rc) { + pr_err("%s: timeout, set-params paramid[0x%x]\n", __func__, + data.param_id); + rc = -EINVAL; + goto done; + } + if (atomic_read(&ac->cmd_state_pp) > 0) { + pr_err("%s: DSP returned error[%d], set-params paramid[0x%x]\n", + __func__, atomic_read(&ac->cmd_state_pp), + data.param_id); + rc = -EINVAL; + goto done; + } + rc = 0; +done: + kfree(asm_params); +fail: + return rc; +} + +int q6asm_set_mfc_panning_params(struct audio_client *ac, + struct asm_stream_pan_ctrl_params *pan_param) +{ + int sz, rc, i; + struct audproc_mfc_output_media_fmt mfc_cfg; + struct apr_hdr hdr; + struct asm_stream_cmd_set_pp_params_v2 payload_params; + struct asm_stream_param_data_v2 data; + struct audproc_chmixer_param_coeff pan_cfg; + uint16_t variable_payload = 0; + char *asm_params = NULL; + uint16_t ii; + uint16_t *dst_gain_ptr = NULL; + + sz = rc = i = 0; + if (ac == NULL) { + pr_err("%s: ac handle NULL\n", __func__); + rc = -EINVAL; + goto fail_cmd1; + } + if (ac->apr == NULL) { + pr_err("%s: ac apr handle NULL\n", __func__); + rc = -EINVAL; + goto fail_cmd1; + } + + sz = sizeof(struct audproc_mfc_output_media_fmt); + q6asm_add_hdr_async(ac, &mfc_cfg.params.hdr, sz, TRUE); + atomic_set(&ac->cmd_state_pp, -1); + mfc_cfg.params.hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V2; + mfc_cfg.params.payload_addr_lsw = 0; + mfc_cfg.params.payload_addr_msw = 0; + mfc_cfg.params.mem_map_handle = 0; + mfc_cfg.params.payload_size = sizeof(mfc_cfg) - sizeof(mfc_cfg.params); + mfc_cfg.data.module_id = AUDPROC_MODULE_ID_MFC; + mfc_cfg.data.param_id = AUDPROC_PARAM_ID_MFC_OUTPUT_MEDIA_FORMAT; + mfc_cfg.data.param_size = mfc_cfg.params.payload_size - + sizeof(mfc_cfg.data); + mfc_cfg.data.reserved = 0; + mfc_cfg.sampling_rate = 0; + mfc_cfg.bits_per_sample = 0; + mfc_cfg.num_channels = pan_param->num_output_channels; + for (i = 0; i < mfc_cfg.num_channels; i++) + mfc_cfg.channel_type[i] = pan_param->output_channel_map[i]; + + rc = apr_send_pkt(ac->apr, (uint32_t *) &mfc_cfg); + if (rc < 0) { + pr_err("%s: set-params send failed paramid[0x%x] rc %d\n", + __func__, mfc_cfg.data.param_id, rc); + rc = -EINVAL; + goto fail_cmd1; + } + + rc = wait_event_timeout(ac->cmd_wait, + (atomic_read(&ac->cmd_state_pp) >= 0), 5*HZ); + if (!rc) { + pr_err("%s: timeout, set-params paramid[0x%x]\n", __func__, + mfc_cfg.data.param_id); + rc = -ETIMEDOUT; + goto fail_cmd1; + } + if (atomic_read(&ac->cmd_state_pp) > 0) { + pr_err("%s: DSP returned error[%s] set-params paramid[0x%x]\n", + __func__, adsp_err_get_err_str( + atomic_read(&ac->cmd_state_pp)), + mfc_cfg.data.param_id); + rc = adsp_err_get_lnx_err_code( + atomic_read(&ac->cmd_state_pp)); + goto fail_cmd1; + } + + variable_payload = pan_param->num_output_channels * sizeof(uint16_t)+ + pan_param->num_input_channels * sizeof(uint16_t) + + pan_param->num_output_channels * + pan_param->num_input_channels * sizeof(uint16_t); + i = (variable_payload % sizeof(uint32_t)); + variable_payload += (i == 0) ? 0 : sizeof(uint32_t) - i; + sz = sizeof(struct apr_hdr) + + sizeof(struct asm_stream_cmd_set_pp_params_v2) + + sizeof(struct asm_stream_param_data_v2) + + sizeof(struct audproc_chmixer_param_coeff) + + variable_payload; + + asm_params = kzalloc(sz, GFP_KERNEL); + if (!asm_params) { + rc = -ENOMEM; + goto fail_cmd1; + } + + q6asm_add_hdr_async(ac, &hdr, sz, TRUE); + atomic_set(&ac->cmd_state_pp, -1); + hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V2; + memcpy(((u8 *)asm_params), &hdr, sizeof(struct apr_hdr)); + + payload_params.data_payload_addr_lsw = 0; + payload_params.data_payload_addr_msw = 0; + payload_params.mem_map_handle = 0; + payload_params.data_payload_size = + sizeof(struct audproc_chmixer_param_coeff) + + variable_payload + sizeof(struct asm_stream_param_data_v2); + memcpy(((u8 *)asm_params + sizeof(struct apr_hdr)), + &payload_params, + sizeof(struct asm_stream_cmd_set_pp_params_v2)); + + data.module_id = AUDPROC_MODULE_ID_MFC; + data.param_id = AUDPROC_CHMIXER_PARAM_ID_COEFF; + data.param_size = sizeof(struct audproc_chmixer_param_coeff) + + variable_payload; + data.reserved = 0; + memcpy(((u8 *)asm_params + sizeof(struct apr_hdr) + + sizeof(struct asm_stream_cmd_set_pp_params_v2)), + &data, sizeof(struct asm_stream_param_data_v2)); + + pan_cfg.index = 0; + pan_cfg.num_output_channels = pan_param->num_output_channels; + pan_cfg.num_input_channels = pan_param->num_input_channels; + memcpy(((u8 *)asm_params + sizeof(struct apr_hdr) + + sizeof(struct asm_stream_cmd_set_pp_params_v2) + + sizeof(struct asm_stream_param_data_v2)), + &pan_cfg, sizeof(struct audproc_chmixer_param_coeff)); + memcpy(((u8 *)asm_params + sizeof(struct apr_hdr) + + sizeof(struct asm_stream_cmd_set_pp_params_v2) + + sizeof(struct asm_stream_param_data_v2) + + sizeof(struct audproc_chmixer_param_coeff)), + pan_param->output_channel_map, + pan_param->num_output_channels * sizeof(uint16_t)); + memcpy(((u8 *)asm_params + sizeof(struct apr_hdr) + + sizeof(struct asm_stream_cmd_set_pp_params_v2) + + sizeof(struct asm_stream_param_data_v2) + + sizeof(struct audproc_chmixer_param_coeff) + + pan_param->num_output_channels * sizeof(uint16_t)), + pan_param->input_channel_map, + pan_param->num_input_channels * sizeof(uint16_t)); + + dst_gain_ptr = (uint16_t *) ((u8 *)asm_params + sizeof(struct apr_hdr) + + sizeof(struct asm_stream_cmd_set_pp_params_v2) + + sizeof(struct asm_stream_param_data_v2) + + sizeof(struct audproc_chmixer_param_coeff) + + (pan_param->num_output_channels * sizeof(uint16_t)) + + (pan_param->num_input_channels * sizeof(uint16_t))); + for (ii = 0; ii < pan_param->num_output_channels * + pan_param->num_input_channels; ii++) + dst_gain_ptr[ii] = (uint16_t) pan_param->gain[ii]; + + rc = apr_send_pkt(ac->apr, (uint32_t *) asm_params); + if (rc < 0) { + pr_err("%s: set-params send failed paramid[0x%x] rc %d\n", + __func__, data.param_id, rc); + rc = -EINVAL; + goto fail_cmd2; + } + + rc = wait_event_timeout(ac->cmd_wait, + (atomic_read(&ac->cmd_state_pp) >= 0), 5*HZ); + if (!rc) { + pr_err("%s: timeout, set-params paramid[0x%x]\n", __func__, + data.param_id); + rc = -ETIMEDOUT; + goto fail_cmd2; + } + if (atomic_read(&ac->cmd_state_pp) > 0) { + pr_err("%s: DSP returned error[%s] set-params paramid[0x%x]\n", + __func__, adsp_err_get_err_str( + atomic_read(&ac->cmd_state_pp)), + data.param_id); + rc = adsp_err_get_lnx_err_code( + atomic_read(&ac->cmd_state_pp)); + goto fail_cmd2; + } + rc = 0; +fail_cmd2: + kfree(asm_params); +fail_cmd1: + return rc; +} + int q6asm_equalizer(struct audio_client *ac, void *eq_p) { struct asm_eq_params eq; @@ -7686,6 +7979,7 @@ int q6asm_async_write(struct audio_client *ac, u32 liomode; u32 io_compressed; u32 io_compressed_stream; + int offset = 0; if (ac == NULL) { pr_err("%s: APR handle NULL\n", __func__); @@ -7747,6 +8041,11 @@ int q6asm_async_write(struct audio_client *ac, } } + if (ab != NULL) { + offset = lbuf_phys_addr - ab->phys; + config_debug_fs_write(ab, offset); + } + rc = apr_send_pkt(ac->apr, (uint32_t *) &write); if (rc < 0) { pr_err("%s: write op[0x%x]rc[%d]\n", __func__, @@ -7892,7 +8191,7 @@ int q6asm_write(struct audio_client *ac, uint32_t len, uint32_t msw_ts, write.mem_map_handle); mutex_unlock(&port->lock); - config_debug_fs_write(ab); + config_debug_fs_write(ab, 0); rc = apr_send_pkt(ac->apr, (uint32_t *) &write); if (rc < 0) { diff --git a/sound/soc/msm/qdsp6v2/q6core.c b/sound/soc/msm/qdsp6v2/q6core.c index 4340d31c218c..6fed443186e5 100644 --- a/sound/soc/msm/qdsp6v2/q6core.c +++ b/sound/soc/msm/qdsp6v2/q6core.c @@ -119,6 +119,18 @@ static int32_t aprv2_core_fn_q(struct apr_client_data *data, void *priv) q6core_lcl.bus_bw_resp_received = 1; wake_up(&q6core_lcl.bus_bw_req_wait); break; + case AVCS_CMD_ADD_POOL_PAGES: + pr_debug("%s: Cmd = AVCS_CMD_ADD_POOL_PAGES status[0x%x]\n", + __func__, payload1[1]); + q6core_lcl.bus_bw_resp_received = 1; + wake_up(&q6core_lcl.bus_bw_req_wait); + break; + case AVCS_CMD_REMOVE_POOL_PAGES: + pr_debug("%s: Cmd = AVCS_CMD_REMOVE_POOL_PAGES status[0x%x]\n", + __func__, payload1[1]); + q6core_lcl.bus_bw_resp_received = 1; + wake_up(&q6core_lcl.bus_bw_req_wait); + break; default: pr_err("%s: Invalid cmd rsp[0x%x][0x%x] opcode %d\n", __func__, @@ -542,6 +554,56 @@ done: return ret; } +int q6core_add_remove_pool_pages(ion_phys_addr_t buf_add, uint32_t bufsz, + uint32_t mempool_id, bool add_pages) +{ + struct avs_mem_assign_region mem_pool; + int ret = 0, sz; + + if (add_pages) + mem_pool.hdr.opcode = AVCS_CMD_ADD_POOL_PAGES; + else + mem_pool.hdr.opcode = AVCS_CMD_REMOVE_POOL_PAGES; + + /* get payload length */ + sz = sizeof(struct avs_mem_assign_region); + mem_pool.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD, + APR_HDR_LEN(sizeof(struct apr_hdr)), + APR_PKT_VER); + mem_pool.hdr.src_port = 0; + mem_pool.hdr.dest_port = 0; + mem_pool.hdr.token = 0; + mem_pool.hdr.pkt_size = sz; + mem_pool.pool_id = mempool_id; + mem_pool.size = bufsz; + mem_pool.addr_lsw = lower_32_bits(buf_add); + mem_pool.addr_msw = msm_audio_populate_upper_32_bits(buf_add); + pr_debug("%s: sending memory map, size %d\n", + __func__, bufsz); + + q6core_lcl.bus_bw_resp_received = 0; + ret = apr_send_pkt(q6core_lcl.core_handle_q, (uint32_t *)&mem_pool); + if (ret < 0) { + pr_err("%s: library map region failed %d\n", + __func__, ret); + ret = -EINVAL; + goto done; + } + + ret = wait_event_timeout(q6core_lcl.bus_bw_req_wait, + (q6core_lcl.bus_bw_resp_received == 1), + msecs_to_jiffies(TIMEOUT_MS)); + if (!ret) { + pr_err("%s: timeout. waited for library memory map\n", + __func__); + ret = -ETIME; + goto done; + } + ret = 0; +done: + return ret; +} + static int q6core_dereg_all_custom_topologies(void) { int ret = 0; diff --git a/sound/soc/msm/sdm660-ext-dai-links.c b/sound/soc/msm/sdm660-ext-dai-links.c index 1c03d8c9e797..30c3ffe2347d 100644 --- a/sound/soc/msm/sdm660-ext-dai-links.c +++ b/sound/soc/msm/sdm660-ext-dai-links.c @@ -1270,10 +1270,10 @@ static struct snd_soc_dai_link msm_ext_common_fe_dai[] = { .be_id = MSM_FRONTEND_DAI_MULTIMEDIA15, }, {/* hw:x,33 */ - .name = MSM_DAILINK_NAME(Compress9), - .stream_name = "Compress9", + .name = MSM_DAILINK_NAME(ULL_NOIRQ_2), + .stream_name = "MM_NOIRQ_2", .cpu_dai_name = "MultiMedia16", - .platform_name = "msm-compress-dsp", + .platform_name = "msm-pcm-dsp-noirq", .dynamic = 1, .dpcm_capture = 1, .dpcm_playback = 1, diff --git a/sound/soc/msm/sdm660-internal.c b/sound/soc/msm/sdm660-internal.c index f4219148e81c..0f28e100f535 100644 --- a/sound/soc/msm/sdm660-internal.c +++ b/sound/soc/msm/sdm660-internal.c @@ -2188,10 +2188,10 @@ static struct snd_soc_dai_link msm_int_dai[] = { .be_id = MSM_FRONTEND_DAI_MULTIMEDIA15, }, {/* hw:x,33 */ - .name = MSM_DAILINK_NAME(Compress9), - .stream_name = "Compress9", + .name = MSM_DAILINK_NAME(ULL_NOIRQ_2), + .stream_name = "MM_NOIRQ_2", .cpu_dai_name = "MultiMedia16", - .platform_name = "msm-compress-dsp", + .platform_name = "msm-pcm-dsp-noirq", .dynamic = 1, .dpcm_capture = 1, .dpcm_playback = 1, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cac2d4975a15..e63b4346e82e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1792,6 +1792,9 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) for (i = 0; i < card->num_aux_devs; i++) soc_remove_aux_dev(card, i); + /* free the ALSA card at first; this syncs with pending operations */ + snd_card_free(card->snd_card); + /* remove and free each DAI */ soc_remove_dai_links(card); @@ -1803,9 +1806,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) snd_soc_dapm_free(&card->dapm); - snd_card_free(card->snd_card); return 0; - } /* removes a socdev */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 2bba7dbc35f9..61f64dcd5374 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -384,6 +384,9 @@ static void snd_complete_urb(struct urb *urb) if (unlikely(atomic_read(&ep->chip->shutdown))) goto exit_clear; + if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags))) + goto exit_clear; + if (usb_pipeout(ep->pipe)) { retire_outbound_urb(ep, ctx); /* can be stopped during retire callback */ diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 70dfdd22102e..d5cc315a5eb4 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1012,6 +1012,17 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->res = 384; } break; + + case USB_ID(0x1130, 0x1620): /* Logitech Speakers S150 */ + /* This audio device has 2 channels and it explicitly requires the + * host to send SET_CUR command on the volume control of both the + * channels. 7936 = 0x1F00 is the default value. + */ + if (cval->channels == 2) + snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, + (cval->control << 8) | 2, 7936); + break; + } } |