From 99161146d31ea864035442c295271b3f605ce89f Mon Sep 17 00:00:00 2001 From: Laxminath Kasam Date: Thu, 13 Apr 2017 21:46:00 +0530 Subject: ASoC: sdm660_cdc: Update micbias regulator to power saving on use basis In internal codec, MICBIAS regulator uses LDO7B power rail. Add changes to remove voting for voltage and current when this regulator is not in use from audio to save power. CRs-Fixed: 2033776 Change-Id: I3c7cea4afadd293a03787081895d451bcf7bea03 Signed-off-by: Laxminath Kasam --- sound/soc/codecs/sdm660_cdc/msm-analog-cdc.c | 93 +++++++++++++++++++++++++--- sound/soc/codecs/sdm660_cdc/msm-analog-cdc.h | 3 + 2 files changed, 89 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sdm660_cdc/msm-analog-cdc.c b/sound/soc/codecs/sdm660_cdc/msm-analog-cdc.c index 699e7251023f..48deddeb0bef 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; @@ -3685,6 +3718,30 @@ 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; @@ -4128,10 +4185,10 @@ 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); @@ -4197,7 +4254,7 @@ 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( @@ -4230,7 +4287,7 @@ 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); @@ -4331,6 +4388,28 @@ 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 { -- cgit v1.2.3 From 5d64d994ecdfbff7421ae7fb620e7fc5f6ae9315 Mon Sep 17 00:00:00 2001 From: Varun Balaraj Date: Tue, 23 May 2017 15:43:56 +0530 Subject: ASoC: msm8998: modify quat mi2s clock id in slave mode update configuration to set bit clock as EBIT in slave mode. Change-Id: Iec62be7b566f20d4ec2bcd300beb3e4f362b6542 Signed-off-by: Varun Balaraj --- sound/soc/msm/msm8998.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/msm/msm8998.c b/sound/soc/msm/msm8998.c index f6a5d1344568..68126a9a2560 100644 --- a/sound/soc/msm/msm8998.c +++ b/sound/soc/msm/msm8998.c @@ -555,6 +555,13 @@ 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, @@ -4507,6 +4514,11 @@ 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, @@ -4526,9 +4538,6 @@ 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", -- cgit v1.2.3 From d6f50d39ce6cfd4b26a91749d705ac7a0b63654c Mon Sep 17 00:00:00 2001 From: Ben Romberger Date: Thu, 25 May 2017 15:21:03 -0700 Subject: ASoC: msm: qdsp6v2: Clear HDMI channel allocation on shutdown On shutdown of the HDMI Q6 driver clear the channel allocation structure. If the channel allocation is not properly cleared the next session to use the driver may inncorrectly use the channel allocation from the previous configuration. Change-Id: Id033958031dc509c6b3060d892bccf5b1c60524b Signed-off-by: Ben Romberger --- sound/soc/msm/qdsp6v2/msm-dai-q6-hdmi-v2.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') 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)); } -- cgit v1.2.3 From df14423aaa14aaac85eebd7a45b6723367196539 Mon Sep 17 00:00:00 2001 From: Ben Romberger Date: Tue, 23 May 2017 16:47:34 -0700 Subject: ASoC: msm: qdsp6v2: Remove size checks when finding ADM cal Remove size checking when looking for ADM calibration. If no calibration is active for that device a size of 0 is sent. If size is checked while finding calibration while multiple devices on the same path are active a device where no calibration is expected will improperly pull calibration for a different device. Change-Id: Idfc68e879a615873c8b23d3ec4ddf8dc1dfc777c Signed-off-by: Ben Romberger --- sound/soc/msm/qdsp6v2/q6adm.c | 18 ++++++------------ 1 file changed, 6 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/msm/qdsp6v2/q6adm.c b/sound/soc/msm/qdsp6v2/q6adm.c index 9f00e1bc4a3d..416323932452 100644 --- a/sound/soc/msm/qdsp6v2/q6adm.c +++ b/sound/soc/msm/qdsp6v2/q6adm.c @@ -1738,13 +1738,11 @@ 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; } } @@ -1771,14 +1769,12 @@ 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; } } @@ -1809,15 +1805,13 @@ 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; } } -- cgit v1.2.3 From db56317667babb3d196d031ba0aacbeeca8f322a Mon Sep 17 00:00:00 2001 From: Karthikeyan Mani Date: Tue, 4 Apr 2017 14:09:09 -0700 Subject: soundwire: Move device init functionality from master Move from master probe to the slave probe, the device init functionality of swr slave. Provide device remove functionality to remove a given device from the master's list. CRs-fixed: 2050710 Change-Id: Iee95c146d8b148e15dca5a8c10de65368cf3b55a Signed-off-by: Karthikeyan Mani --- drivers/soundwire/soundwire.c | 68 +++++++++++------------------ drivers/soundwire/swr-wcd-ctrl.c | 14 ++---- include/linux/soundwire/soundwire.h | 4 +- sound/soc/codecs/wsa881x.c | 85 ++++++++++++++++--------------------- 4 files changed, 66 insertions(+), 105 deletions(-) (limited to 'sound') diff --git a/drivers/soundwire/soundwire.c b/drivers/soundwire/soundwire.c index 6691418b516e..3e37ac3bda2d 100755 --- a/drivers/soundwire/soundwire.c +++ b/drivers/soundwire/soundwire.c @@ -1,4 +1,4 @@ -/* Copyright (c) 2015-2016, The Linux Foundation. All rights reserved. +/* Copyright (c) 2015-2017, 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 @@ -67,6 +67,27 @@ static void swr_dev_release(struct device *dev) kfree(swr_dev); } +/** + * swr_remove_device - remove a soundwire device + * @swr_dev: soundwire device to remove + * + * Remove a soundwire device. Go through the soundwire + * device list that master has and remove swr_dev from + * it. + */ +void swr_remove_device(struct swr_device *swr_dev) +{ + struct swr_device *swr_dev_loop, *safe; + + list_for_each_entry_safe(swr_dev_loop, safe, + &swr_dev->master->devices, + dev_list) { + if (swr_dev == swr_dev_loop) + list_del(&swr_dev_loop->dev_list); + } +} +EXPORT_SYMBOL(swr_remove_device); + /** * swr_new_device - instantiate a new soundwire device * @master: Controller to which device is connected @@ -128,47 +149,6 @@ err_out: } EXPORT_SYMBOL(swr_new_device); -/** - * swr_startup_devices - perform additional initialization for child devices - * - * @swr_dev: pointer to soundwire slave device - * - * Performs any additional initialization needed for a soundwire slave device. - * This is a optional functionality defined by slave devices. - * Removes the slave node from the list, in case there is any failure. - */ -int swr_startup_devices(struct swr_device *swr_dev) -{ - struct swr_driver *swr_drv; - struct device *dev; - int ret = 0; - - if (!swr_dev) - return -EINVAL; - - dev = &swr_dev->dev; - if (!dev) - return -EINVAL; - - swr_drv = to_swr_driver(dev->driver); - if (!swr_drv) - return -EINVAL; - - if (swr_drv->startup) { - ret = swr_drv->startup(swr_dev); - if (ret) - goto out; - - dev_dbg(&swr_dev->dev, - "%s: startup complete for device %lx\n", - __func__, swr_dev->addr); - } - -out: - return ret; -} -EXPORT_SYMBOL(swr_startup_devices); - /** * of_register_swr_devices - register child devices on to the soundwire bus * @master: pointer to soundwire master device @@ -610,7 +590,7 @@ int swr_device_up(struct swr_device *swr_dev) dev = &swr_dev->dev; sdrv = to_swr_driver(dev->driver); if (!sdrv) - return -EINVAL; + return 0; if (sdrv->device_up) return sdrv->device_up(to_swr_device(dev)); @@ -638,7 +618,7 @@ int swr_device_down(struct swr_device *swr_dev) dev = &swr_dev->dev; sdrv = to_swr_driver(dev->driver); if (!sdrv) - return -EINVAL; + return 0; if (sdrv->device_down) return sdrv->device_down(to_swr_device(dev)); diff --git a/drivers/soundwire/swr-wcd-ctrl.c b/drivers/soundwire/swr-wcd-ctrl.c index e72663bd2138..b13119551cdf 100644 --- a/drivers/soundwire/swr-wcd-ctrl.c +++ b/drivers/soundwire/swr-wcd-ctrl.c @@ -1369,7 +1369,6 @@ static int swrm_probe(struct platform_device *pdev) { struct swr_mstr_ctrl *swrm; struct swr_ctrl_platform_data *pdata; - struct swr_device *swr_dev, *safe; int ret; /* Allocate soundwire master driver structure */ @@ -1470,9 +1469,6 @@ static int swrm_probe(struct platform_device *pdev) goto err_mstr_fail; } - if (pdev->dev.of_node) - of_register_swr_devices(&swrm->master); - /* Add devices registered with board-info as the controller will be up now */ @@ -1489,15 +1485,11 @@ static int swrm_probe(struct platform_device *pdev) } swrm->version = swrm->read(swrm->handle, SWRM_COMP_HW_VERSION); - /* Enumerate slave devices */ - list_for_each_entry_safe(swr_dev, safe, &swrm->master.devices, - dev_list) { - ret = swr_startup_devices(swr_dev); - if (ret) - list_del(&swr_dev->dev_list); - } mutex_unlock(&swrm->mlock); + if (pdev->dev.of_node) + of_register_swr_devices(&swrm->master); + dbgswrm = swrm; debugfs_swrm_dent = debugfs_create_dir(dev_name(&pdev->dev), 0); if (!IS_ERR(debugfs_swrm_dent)) { diff --git a/include/linux/soundwire/soundwire.h b/include/linux/soundwire/soundwire.h index 2083e7b5da25..c3b95eeb37e5 100755 --- a/include/linux/soundwire/soundwire.h +++ b/include/linux/soundwire/soundwire.h @@ -1,4 +1,4 @@ -/* Copyright (c) 2015-2016, The Linux Foundation. All rights reserved. +/* Copyright (c) 2015-2017, 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 @@ -309,4 +309,6 @@ extern int swr_reset_device(struct swr_device *swr_dev); extern int swr_slvdev_datapath_control(struct swr_device *swr_dev, u8 dev_num, bool enable); extern int swr_remove_from_group(struct swr_device *dev, u8 dev_num); + +extern void swr_remove_device(struct swr_device *swr_dev); #endif /* _LINUX_SOUNDWIRE_H */ diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index eaaca97e2b8e..0af656ce48f0 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 +1184,7 @@ static int wsa881x_swr_probe(struct swr_device *pdev) { int ret = 0; struct wsa881x_priv *wsa881x; + u8 devnum = 0; wsa881x = devm_kzalloc(&pdev->dev, sizeof(struct wsa881x_priv), GFP_KERNEL); @@ -1291,8 +1244,43 @@ 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: + swr_remove_device(pdev); err: return ret; } @@ -1425,7 +1413,6 @@ 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) -- cgit v1.2.3 From c848d4ca45ff58fcc406afc9f95be02b054c520d Mon Sep 17 00:00:00 2001 From: Karthikeyan Mani Date: Thu, 4 May 2017 11:39:47 -0700 Subject: ASoC: wsa881x: Fix GPIO leak issue In case of any probe/logical address error, set the pinctrl of wsa881x to the state that it was before the probe entered. Otherwise set it to active state. CRs-fixed: 2050725 Change-Id: I5022885f36111caeac1d25017db8a474e26ca521 Signed-off-by: Karthikeyan Mani --- sound/soc/codecs/wsa881x.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index 0af656ce48f0..fe975a7dbe4e 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -1185,6 +1185,7 @@ 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); @@ -1218,6 +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; @@ -1280,6 +1284,8 @@ static int wsa881x_swr_probe(struct swr_device *pdev) return 0; dev_err: + if (pin_state_current == false) + wsa881x_gpio_ctrl(wsa881x, false); swr_remove_device(pdev); err: return ret; -- cgit v1.2.3 From 023a8b0925bef16bf36d1740f41f6fa74167b0ef Mon Sep 17 00:00:00 2001 From: Alexander Tsoy Date: Mon, 22 May 2017 20:58:11 +0300 Subject: ALSA: hda - apply STAC_9200_DELL_M22 quirk for Dell Latitude D430 commit 1fc2e41f7af4572b07190f9dec28396b418e9a36 upstream. This model is actually called 92XXM2-8 in Windows driver. But since pin configs for M22 and M28 are identical, just reuse M22 quirk. Fixes external microphone (tested) and probably docking station ports (not tested). Signed-off-by: Alexander Tsoy Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') 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 */ -- cgit v1.2.3 From 4282bbbb81023d4016ad0b23fabb841f878a4957 Mon Sep 17 00:00:00 2001 From: Rohit Kumar Date: Fri, 19 May 2017 18:33:19 +0530 Subject: ASoC: msm: qdspv6: Fix wrong smmu sid for ULL playback SMMU sid is set to 0 in q6asm_set_shared_circ_buff and q6asm_set_shared_pos_buff in 32 bit arch. Fix it to send proper SID to ADSP when sharing buffer. Change-Id: I00cc0f881acd7a4a52292e65360ea7b03f2f0212 Signed-off-by: Rohit Kumar --- sound/soc/msm/qdsp6v2/q6asm.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/msm/qdsp6v2/q6asm.c b/sound/soc/msm/qdsp6v2/q6asm.c index da156bf61610..6f097b7e8dd5 100644 --- a/sound/soc/msm/qdsp6v2/q6asm.c +++ b/sound/soc/msm/qdsp6v2/q6asm.c @@ -3399,11 +3399,12 @@ 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 +3446,12 @@ 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: -- cgit v1.2.3 From f5a7555de3bb1da8d7ac70418c6defb515a56b5a Mon Sep 17 00:00:00 2001 From: Manish Dewangan Date: Mon, 12 Jun 2017 15:35:18 +0530 Subject: qdspv2: Add latency calculation support in pcm offload path Update cold and continuous latency debug fs entries for offload path to support latency calculation for pcm offload path. CRs-Fixed: 2059729 Change-Id: Ic59c7b2fec76e682837c89926595fb3262d01aa8 Signed-off-by: Manish Dewangan --- sound/soc/msm/qdsp6v2/q6asm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/msm/qdsp6v2/q6asm.c b/sound/soc/msm/qdsp6v2/q6asm.c index 6f097b7e8dd5..44e5b01b1ccb 100644 --- a/sound/soc/msm/qdsp6v2/q6asm.c +++ b/sound/soc/msm/qdsp6v2/q6asm.c @@ -1952,13 +1952,13 @@ 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: @@ -7750,6 +7750,8 @@ int q6asm_async_write(struct audio_client *ac, } } + config_debug_fs_write(ab); + rc = apr_send_pkt(ac->apr, (uint32_t *) &write); if (rc < 0) { pr_err("%s: write op[0x%x]rc[%d]\n", __func__, -- cgit v1.2.3 From f5bc918760c8100410847a6a6e4c25f24e358e0c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 2 Jun 2017 15:03:38 +0200 Subject: ALSA: timer: Fix race between read and ioctl commit d11662f4f798b50d8c8743f433842c3e40fe3378 upstream. The read from ALSA timer device, the function snd_timer_user_tread(), may access to an uninitialized struct snd_timer_user fields when the read is concurrently performed while the ioctl like snd_timer_user_tselect() is invoked. We have already fixed the races among ioctls via a mutex, but we seem to have forgotten the race between read vs ioctl. This patch simply applies (more exactly extends the already applied range of) tu->ioctl_lock in snd_timer_user_tread() for closing the race window. Reported-by: Alexander Potapenko Tested-by: Alexander Potapenko Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/core/timer.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index 278a332f97bd..7f381739706a 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1958,6 +1958,7 @@ 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) { @@ -1973,7 +1974,9 @@ 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); @@ -1993,7 +1996,6 @@ 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))) @@ -2003,7 +2005,6 @@ 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) @@ -2013,6 +2014,7 @@ 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; } -- cgit v1.2.3 From 54d12fbf54d4d40d2a47200150344bc001a29e96 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 2 Jun 2017 17:26:56 +0200 Subject: ALSA: timer: Fix missing queue indices reset at SNDRV_TIMER_IOCTL_SELECT commit ba3021b2c79b2fa9114f92790a99deb27a65b728 upstream. snd_timer_user_tselect() reallocates the queue buffer dynamically, but it forgot to reset its indices. Since the read may happen concurrently with ioctl and snd_timer_user_tselect() allocates the buffer via kmalloc(), this may lead to the leak of uninitialized kernel-space data, as spotted via KMSAN: BUG: KMSAN: use of unitialized memory in snd_timer_user_read+0x6c4/0xa10 CPU: 0 PID: 1037 Comm: probe Not tainted 4.11.0-rc5+ #2739 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:16 dump_stack+0x143/0x1b0 lib/dump_stack.c:52 kmsan_report+0x12a/0x180 mm/kmsan/kmsan.c:1007 kmsan_check_memory+0xc2/0x140 mm/kmsan/kmsan.c:1086 copy_to_user ./arch/x86/include/asm/uaccess.h:725 snd_timer_user_read+0x6c4/0xa10 sound/core/timer.c:2004 do_loop_readv_writev fs/read_write.c:716 __do_readv_writev+0x94c/0x1380 fs/read_write.c:864 do_readv_writev fs/read_write.c:894 vfs_readv fs/read_write.c:908 do_readv+0x52a/0x5d0 fs/read_write.c:934 SYSC_readv+0xb6/0xd0 fs/read_write.c:1021 SyS_readv+0x87/0xb0 fs/read_write.c:1018 This patch adds the missing reset of queue indices. Together with the previous fix for the ioctl/read race, we cover the whole problem. Reported-by: Alexander Potapenko Tested-by: Alexander Potapenko Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/core/timer.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index 7f381739706a..48eaccba82a3 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1621,6 +1621,7 @@ 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); -- cgit v1.2.3 From 9a9388953bdcd416f94991f3bd19ea9bc2b31930 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 24 May 2017 10:19:45 +0200 Subject: ASoC: Fix use-after-free at card unregistration commit 4efda5f2130da033aeedc5b3205569893b910de2 upstream. soc_cleanup_card_resources() call snd_card_free() at the last of its procedure. This turned out to lead to a use-after-free. PCM runtimes have been already removed via soc_remove_pcm_runtimes(), while it's dereferenced later in soc_pcm_free() called via snd_card_free(). The fix is simple: just move the snd_card_free() call to the beginning of the whole procedure. This also gives another benefit: it guarantees that all operations have been shut down before actually releasing the resources, which was racy until now. Reported-and-tested-by: Robert Jarzmik Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown Signed-off-by: Greg Kroah-Hartman --- sound/soc/soc-core.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a1305f827a98..fa6b74a304a7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1775,6 +1775,9 @@ 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); @@ -1786,9 +1789,7 @@ 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 */ -- cgit v1.2.3 From 41144e80b67849f3700273bcff48de5b653f2b6e Mon Sep 17 00:00:00 2001 From: Siddartha Shaik Date: Wed, 14 Jun 2017 11:43:06 +0530 Subject: ASoC: msm: Update mixer control for QUAT MI2S TX Format DSP loopback uses QUAT MI2S TX as source interface so mixer control is updated for QUAT MI2S TX Format to support compressed and pcm input data formats. CRs-Fixed: 2061759 Change-Id: I8c283de767f938ae8f4e1180a9dffc79a1da7ff0 Signed-off-by: Siddartha Shaik --- sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c index 64b1961794b9..fa8bdddacef2 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, -- cgit v1.2.3 From 04b7e499f885100ce8dd6ad3c19acfea5f62383a Mon Sep 17 00:00:00 2001 From: Siddartha Shaik Date: Wed, 14 Jun 2017 12:03:35 +0530 Subject: ASoC: msm: Add Compressed app type in transcode loopback driver Pick proper app type for compressed input data port in transcode loopback driver to support compressed input in DSP loopback. CRs-Fixed: 2061759 Change-Id: Ib2f66b467456787e08265a0da2bd85bb881a5d61 Signed-off-by: Siddartha Shaik --- sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) (limited to 'sound') 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..6bb85ca8e84e 100644 --- a/sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c @@ -112,7 +112,8 @@ 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__); @@ -449,17 +450,17 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream, trans->audio_client->perf_mode, 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, + true, trans->session_id, SNDRV_PCM_STREAM_PLAYBACK); pr_debug("%s: Successfully opened ADM sessions\n", __func__); @@ -913,10 +914,21 @@ static int msm_transcode_loopback_probe(struct snd_soc_platform *platform) 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) -- cgit v1.2.3 From e0085c2429cfabc3db0de2d23e783c45d9d2aa14 Mon Sep 17 00:00:00 2001 From: Meng Wang Date: Thu, 22 Jun 2017 15:53:46 +0800 Subject: ASoC: wcd-mbhc: Disconnect ANC from RX chain during plug removal Disconnect ANC from RX when headphone plug is removed from the jack to prevent any audible pop. Change-Id: I7a3f6e876f8560580965762c7393c83833ef3b47 Signed-off-by: Meng Wang --- sound/soc/codecs/wcd-mbhc-v2.c | 35 +++++++++++++++++++++++++++++++++ sound/soc/codecs/wcd-mbhc-v2.h | 9 +++++++++ sound/soc/codecs/wcd9335.c | 28 ++++++++++++++++++++++++++ sound/soc/codecs/wcd934x/wcd934x-mbhc.c | 28 ++++++++++++++++++++++++++ 4 files changed, 100 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wcd-mbhc-v2.c b/sound/soc/codecs/wcd-mbhc-v2.c index 44141ae5668e..3d131ed463c9 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, 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..4b9ab630b9aa 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -2217,6 +2217,32 @@ 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 +2265,8 @@ 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, 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( -- cgit v1.2.3 From 04282fcc795edab73cffb4a1c63459657815bd9d Mon Sep 17 00:00:00 2001 From: Ajay Agarwal Date: Thu, 22 Jun 2017 11:09:33 +0530 Subject: sound: usb: Add quirk to issue SET_CUR volume on 2nd channel Some audio devices with two channels require the host to issue SET_CUR command on the Volume control on both the channels in order to function (playback or record). Currently, the Linux Host driver issues SET_CUR only for the 1st channel. Hence, add the quirk for concerned devices for SET_CUR on 2nd channel. Change-Id: I6f0bbcdfd3a8b3ccb33a1d56c252c701849a442f Signed-off-by: Ajay Agarwal --- sound/usb/mixer.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') 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; + } } -- cgit v1.2.3 From 72856792e1e6ca39cffbb8fccf7643450a57437a Mon Sep 17 00:00:00 2001 From: Vatsal Bucha Date: Thu, 18 May 2017 11:37:39 +0530 Subject: ASoC: wcd9335: Add counter to maintain count of functions voting for max bw During voice calls slimbus underflow is observed. This is because some functions vote without updating status mask. Thus adding a reference counter in tasha which would keep track of voting and unvoting instances and unvote only when count is 0. CRs-Fixed: 2047164 Change-Id: I50710a6dfde8bcdf750b90dc7f45e9632e8634fe Signed-off-by: Vatsal Bucha --- sound/soc/codecs/wcd9335.c | 61 ++++++++++++++++++++++++++-------------------- 1 file changed, 35 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 7ad3aeaa8fb7..a0836bc50e59 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; }; @@ -3153,10 +3155,7 @@ 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 +5467,7 @@ 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; @@ -11335,11 +11331,8 @@ 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, @@ -11574,15 +11567,11 @@ 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; } @@ -13290,13 +13279,29 @@ 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, @@ -13479,6 +13484,8 @@ 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; @@ -14265,6 +14272,7 @@ 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), @@ -14369,6 +14377,7 @@ 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; } -- cgit v1.2.3 From 3a94db7a1c9cfa03d187439848822f46a66929da Mon Sep 17 00:00:00 2001 From: Vatsal Bucha Date: Mon, 5 Jun 2017 19:35:45 +0530 Subject: ASoC: sdm660_cdc: Prevent MICBIAS1 enable during headset record Add a switch to prevent default powering up of AMIC1 thereby preventing MICBIAS1 from getting on all the time while headset record. CRs-Fixed: 2049029 Change-Id: Iab5771b62e454fac25df42c53c35b3ec67fec5c9 Signed-off-by: Vatsal Bucha --- sound/soc/codecs/sdm660_cdc/msm-analog-cdc.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sdm660_cdc/msm-analog-cdc.c b/sound/soc/codecs/sdm660_cdc/msm-analog-cdc.c index 00f2aa766363..6afae4eb7af2 100644 --- a/sound/soc/codecs/sdm660_cdc/msm-analog-cdc.c +++ b/sound/soc/codecs/sdm660_cdc/msm-analog-cdc.c @@ -2022,6 +2022,9 @@ 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 +3075,8 @@ 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 +3417,8 @@ 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, -- cgit v1.2.3 From 8914f7b8616daec323016b0e5a9920e7501ef6e9 Mon Sep 17 00:00:00 2001 From: Vidyakumar Athota Date: Tue, 20 Jun 2017 16:39:00 -0700 Subject: ASoC: msm: qdsp6v2: add size check to fix out of bounds issue Before calling audio calibration ioctl functions, compare the allocated buffer size to the size of the header and cal type header to ensure the buffer is big enough. Change-Id: I601bb37ddcc34d459c207cf579f29744fe912d7b Signed-off-by: Vidyakumar Athota --- sound/soc/msm/qdsp6v2/audio_calibration.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') 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__); -- cgit v1.2.3 From 88fad2e0e16924da2864ed4f3f017672a8165d14 Mon Sep 17 00:00:00 2001 From: Ben Romberger Date: Tue, 13 Jun 2017 17:42:17 -0700 Subject: ASoC: msm: qdsp6v2: Add matrix limiter support Add matrix limiter to ADM. Limiter is used to prevent saturation when mixing multiple audio streams. It should be applied after ADM open but before ADM matrix map. Change-Id: I6787fe869e8ceee13694245b744ecd74c3a49682 CRs-Fixed: 2064258 Signed-off-by: Ben Romberger --- include/sound/apr_audio-v2.h | 83 +++++++++++++++++++++- include/sound/q6adm-v2.h | 8 +++ sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c | 98 +++++++++++++++----------- sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h | 18 +++-- sound/soc/msm/qdsp6v2/q6adm.c | 108 ++++++++++++++++++++++++++--- 5 files changed, 262 insertions(+), 53 deletions(-) (limited to 'sound') diff --git a/include/sound/apr_audio-v2.h b/include/sound/apr_audio-v2.h index 74995a0cdbad..f519f9ae6606 100644 --- a/include/sound/apr_audio-v2.h +++ b/include/sound/apr_audio-v2.h @@ -446,6 +446,88 @@ struct adm_param_data_v5 { */ } __packed; + +struct param_data_v6 { + /* Unique ID of the module. */ + u32 module_id; + /* Unique ID of the instance. */ + u16 instance_id; + /* Reserved for future enhancements. + * This field must be set to zero. + */ + u16 reserved; + /* Unique ID of the parameter. */ + u32 param_id; + /* Data size of the param_id/module_id combination. + * This value is a + * multiple of 4 bytes. + */ + u32 param_size; +} __packed; + +/* ADM_CMD_SET_MTMX_STRTR_DEV_PARAMS_V1 command is used to set + * calibration data to the ADSP Matrix Mixer the payload is + * of struct adm_cmd_set_mtmx_params_v1. + * + * ADM_CMD_GET_MTMX_STRTR_DEV_PARAMS_V1 can be used to get + * the calibration data from the ADSP Matrix Mixer and + * ADM_CMDRSP_GET_MTMX_STRTR_DEV_PARAMS_V1 is the response + * ioctl to ADM_CMD_GET_MTMX_STRTR_DEV_PARAMS_V1. + */ +#define ADM_CMD_SET_MTMX_STRTR_DEV_PARAMS_V1 0x00010367 +#define ADM_CMD_GET_MTMX_STRTR_DEV_PARAMS_V1 0x00010368 +#define ADM_CMDRSP_GET_MTMX_STRTR_DEV_PARAMS_V1 0x00010369 + +/* Payload of the #define ADM_CMD_SET_MTMX_STRTR_DEV_PARAMS_V1 command. + * If the data_payload_addr_lsw and data_payload_addr_msw element + * are NULL, a series of struct param_data_v6 structures immediately + * follows, whose total size is payload_size bytes. + */ +struct adm_cmd_set_mtmx_params_v1 { + struct apr_hdr hdr; + /* LSW of parameter data payload address.*/ + u32 payload_addr_lsw; + + /* MSW of parameter data payload address.*/ + u32 payload_addr_msw; + + /* Memory map handle returned by ADM_CMD_SHARED_MEM_MAP_REGIONS + * command. + * If mem_map_handle is zero it implies the message is in + * the payload + */ + u32 mem_map_handle; + + /* Size in bytes of the variable payload accompanying this + * message or in shared memory. This is used for parsing + * the parameter payload. + */ + u32 payload_size; + + /* COPP ID/Device ID */ + u16 copp_id; + + /* For alignment, must be set to 0 */ + u16 reserved; +} __packed; + +struct enable_param_v6 { + /* + * Specifies whether the Audio processing module is enabled. + * This parameter is generic/common parameter to configure or + * determine the state of any audio processing module. + */ + struct param_data_v6 param; + + /* @values 0 : Disable 1: Enable */ + uint32_t enable; +} __packed; + +/* Defined in ADSP as VOICE_MODULE_TX_STREAM_LIMITER but + * used for RX stream limiter on matrix input to ADM. + */ +#define ADM_MTMX_MODULE_STREAM_LIMITER 0x00010F15 + #define ASM_STREAM_CMD_REGISTER_PP_EVENTS 0x00013213 #define ASM_STREAM_PP_EVENT 0x00013214 #define ASM_STREAM_CMD_REGISTER_IEC_61937_FMT_UPDATE 0x13333 @@ -9151,7 +9233,6 @@ struct srs_trumedia_params { } __packed; /* SRS TruMedia end */ -#define AUDPROC_PARAM_ID_ENABLE 0x00010904 #define ASM_STREAM_POSTPROC_TOPO_ID_SA_PLUS 0x1000FFFF /* DTS Eagle */ #define AUDPROC_MODULE_ID_DTS_HPX_PREMIX 0x0001077C diff --git a/include/sound/q6adm-v2.h b/include/sound/q6adm-v2.h index e689e9357012..abf11d056c6e 100644 --- a/include/sound/q6adm-v2.h +++ b/include/sound/q6adm-v2.h @@ -51,6 +51,13 @@ enum { ADM_CLIENT_ID_MAX, }; +/* ENUM for adm_status & route_status */ +enum adm_status_flags { + ADM_STATUS_CALIBRATION_REQUIRED = 0, + ADM_STATUS_LIMITER, + ADM_STATUS_MAX, +}; + #define MAX_COPPS_PER_PORT 0x8 #define ADM_MAX_CHANNELS 8 @@ -61,6 +68,7 @@ struct route_payload { int app_type[MAX_COPPS_PER_PORT]; int acdb_dev_id[MAX_COPPS_PER_PORT]; int sample_rate[MAX_COPPS_PER_PORT]; + unsigned long route_status[MAX_COPPS_PER_PORT]; unsigned short num_copps; unsigned int session_id; }; diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c index 974d4a582540..e53909e40e7a 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c @@ -118,17 +118,13 @@ 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 +138,7 @@ 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); @@ -953,7 +950,7 @@ 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 +980,11 @@ 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++; } } @@ -1204,6 +1206,11 @@ 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 +1441,11 @@ 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++; } } @@ -15168,7 +15180,7 @@ 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 +15204,6 @@ 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 +15215,11 @@ 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 +15229,7 @@ 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 +15238,47 @@ 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 +15290,6 @@ 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 +15312,20 @@ 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 +15337,7 @@ static int msm_routing_put_device_pp_params_mixer(struct snd_kcontrol *kcontrol, } } } +done: return 0; } diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h index 19e726001d25..15125982408d 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h +++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h @@ -392,12 +392,20 @@ 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 diff --git a/sound/soc/msm/qdsp6v2/q6adm.c b/sound/soc/msm/qdsp6v2/q6adm.c index 61dd478b97e4..713ca9898206 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]; @@ -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] @@ -2701,6 +2696,97 @@ 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 +2883,11 @@ 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 +3088,8 @@ 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 +4906,7 @@ 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; } } -- cgit v1.2.3 From 35f63de1e1251aa54445e1434981668c777d3b76 Mon Sep 17 00:00:00 2001 From: Revathi Uddaraju Date: Fri, 30 Jun 2017 16:08:43 +0530 Subject: ASoC: msm-cpe: Resolve memory out of bound access Resolve memory out of bound access by correcting the length of the buffer to be copied. Change-Id: I2cc74a664399913acf67464a5f6827b100522676 Signed-off-by: Revathi Uddaraju --- sound/soc/msm/msm-cpe-lsm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') 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); } } -- cgit v1.2.3 From 21472bb1589242b53e50d738c0e85f843cc74e83 Mon Sep 17 00:00:00 2001 From: Rohit Kumar Date: Thu, 29 Jun 2017 10:56:23 +0530 Subject: ASoC: wcd9335: Reset power_active_ref to 0 when it is negative Reset power_active_ref counter to 0 when it is negative to take digital core out of reset during next session. CRs-Fixed: 2054140 Change-Id: I4ca4c465e7d15f8a0ccd0546db7b0cc5c6ea701e Signed-off-by: Rohit Kumar --- sound/soc/codecs/wcd9335.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 189165ce14c6..12006b32c3b1 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -12110,8 +12110,10 @@ 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; } -- cgit v1.2.3 From 267f48418a7fe010a53b8f8a96738df3eab9c60f Mon Sep 17 00:00:00 2001 From: Vatsal Bucha Date: Thu, 29 Jun 2017 13:09:04 +0530 Subject: ASoC: mbhc: Fix audio mute for special headset MICBIAS is disabled extra time during playback session resulting in mclk also being disabled. This results in slimbus overflow and audio mute. This can be fixed by checking whether device connected is special headset before disabling micbias during report_plug so that micbias is not disabled again. CRs-Fixed: 2056268 Change-Id: Iab94ca2d6a3d69a5fa122da2c4f1e8d42ce75155 Signed-off-by: Vatsal Bucha --- sound/soc/codecs/wcd-mbhc-v2.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wcd-mbhc-v2.c b/sound/soc/codecs/wcd-mbhc-v2.c index 44141ae5668e..353f14c643cb 100644 --- a/sound/soc/codecs/wcd-mbhc-v2.c +++ b/sound/soc/codecs/wcd-mbhc-v2.c @@ -633,7 +633,8 @@ 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, -- cgit v1.2.3 From 1a2b67c241a1a33f384c3394c9237beb79b509e3 Mon Sep 17 00:00:00 2001 From: Satish Babu Patakokila Date: Thu, 8 Jun 2017 17:51:43 +0530 Subject: ASoc: msm8998: Add multichannel support for SLIM_0_RX Add 8 channel support for SLIM_0_RX dai. Change-Id: I1acff62ef74dd18b78fb9862102a535212c84e5e Signed-off-by: Satish Babu Patakokila --- sound/soc/msm/msm8998.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/msm/msm8998.c b/sound/soc/msm/msm8998.c index 0b37e7e073aa..4fac5e02c75d 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"}; -- cgit v1.2.3 From 11327be3570e4640bf9c19e60b724f9fbb21eeb6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 28 Jun 2017 12:02:02 +0200 Subject: ALSA: hda - Fix endless loop of codec configure commit d94815f917da770d42c377786dc428f542e38f71 upstream. azx_codec_configure() loops over the codecs found on the given controller via a linked list. The code used to work in the past, but in the current version, this may lead to an endless loop when a codec binding returns an error. The culprit is that the snd_hda_codec_configure() unregisters the device upon error, and this eventually deletes the given codec object from the bus. Since the list is initialized via list_del_init(), the next object points to the same device itself. This behavior change was introduced at splitting the HD-audio code code, and forgotten to adapt it here. For fixing this bug, just use a *_safe() version of list iteration. Fixes: d068ebc25e6e ("ALSA: hda - Move some codes up to hdac_bus struct") Reported-by: Daniel Vetter Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/hda/hda_codec.h | 2 ++ sound/pci/hda/hda_controller.c | 8 ++++++-- 2 files changed, 8 insertions(+), 2 deletions(-) (limited to 'sound') 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; -- cgit v1.2.3 From c70e2006d06a868fb53d922e8b4057561774b081 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Wed, 28 Jun 2017 08:59:16 +0800 Subject: ALSA: hda - set input_path bitmap to zero after moving it to new place commit a8f20fd25bdce81a8e41767c39f456d346b63427 upstream. Recently we met a problem, the codec has valid adcs and input pins, and they can form valid input paths, but the driver does not build valid controls for them like "Mic boost", "Capture Volume" and "Capture Switch". Through debugging, I found the driver needs to shrink the invalid adcs and input paths for this machine, so it will move the whole column bitmap value to the previous column, after moving it, the driver forgets to set the original column bitmap value to zero, as a result, the driver will invalidate the path whose index value is the original colume bitmap value. After executing this function, all valid input paths are invalidated by a mistake, there are no any valid input paths, so the driver won't build controls for them. Fixes: 3a65bcdc577a ("ALSA: hda - Fix inconsistent input_paths after ADC reduction") Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai Signed-off-by: Greg Kroah-Hartman --- sound/pci/hda/hda_generic.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') 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++; -- cgit v1.2.3 From a1170b1aecf33507bbf1a479367a86c9d83d0080 Mon Sep 17 00:00:00 2001 From: Haynes Mathew George Date: Thu, 27 Apr 2017 17:30:03 -0700 Subject: ASoC: msm: Add hwdep node for NOIRQ FE ALSA lacks support to accept shared memory supplied from userspace. But some usecases need to get a shareable file descriptor for the dma buffer. As an alternate to adding custom ioctls to the ALSA framework, use a hwdep node to get a file descriptor to the underlying dma buffer. Maintain compat mode as well. CRs-Fixed: 2041151 Change-Id: Id783bb84e8ef59b28e42d982903971625577b9a3 Signed-off-by: Haynes Mathew George --- include/uapi/sound/devdep_params.h | 12 ++- sound/soc/msm/qdsp6v2/msm-pcm-q6-noirq.c | 150 ++++++++++++++++++++++++++++++- 2 files changed, 160 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/include/uapi/sound/devdep_params.h b/include/uapi/sound/devdep_params.h index 5061ec0da356..9e3133b76c68 100644 --- a/include/uapi/sound/devdep_params.h +++ b/include/uapi/sound/devdep_params.h @@ -1,5 +1,5 @@ /* - * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved. + * Copyright (c) 2013-2015,2017, 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 @@ -66,4 +66,14 @@ struct dts_eagle_param_desc { uint32_t device; } __packed; +#define HWDEP_FE_BASE 3000 /*unique base for FE hw dep nodes*/ +struct snd_pcm_mmap_fd { + int32_t dir; + int32_t fd; + int32_t size; + int32_t actual_size; +}; + +#define SNDRV_PCM_IOCTL_MMAP_DATA_FD _IOWR('U', 0xd2, struct snd_pcm_mmap_fd) + #endif 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 #include #include +#include + #include #include #include +#include #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, -- cgit v1.2.3 From ea135cb129b0f0a7bbf98746f9c71c74ea697c77 Mon Sep 17 00:00:00 2001 From: Cong Tang Date: Tue, 6 Jun 2017 17:27:01 +0800 Subject: ASoc: msm: Support PRI_TDM_TX routing to QUAT_TDM_RX port mixer. Support PRI_TDM_TX routing to QUAT_TDM_RX port mixer for HFP downlink usecase. Change-Id: I18babd519c2a623898f4d3913e9e7b900e86c22c CRs-Fixed: 2054358 Signed-off-by: Cong Tang --- sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c | 80 ++++++++++++++++++++++++++++++ 1 file changed, 80 insertions(+) (limited to 'sound') diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c index 974d4a582540..660db8930b9c 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c @@ -9797,6 +9797,22 @@ 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 +9880,22 @@ 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 +9963,22 @@ 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 +10046,22 @@ 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, @@ -14460,6 +14524,10 @@ 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 +14546,10 @@ 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 +14568,10 @@ 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 +14590,10 @@ 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"}, -- cgit v1.2.3 From 93f219af2df8c09e889d2123a6057ac618cce10f Mon Sep 17 00:00:00 2001 From: Karthikeyan Mani Date: Wed, 26 Apr 2017 14:30:31 -0700 Subject: ASoC: wsa881x: Assign device number in reset Assign the logical device number in the swr_reset function so that the dev_num of the speaker device is assigned to the correct one after a reset event. CRs-fixed: 2039206 Change-Id: Ief3c65c3b36c93e7dcf775413e527e92d9ec7b0c Signed-off-by: Karthikeyan Mani --- sound/soc/codecs/wsa881x.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index eaaca97e2b8e..f4eeb26f00c3 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -1371,6 +1371,7 @@ 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; -- cgit v1.2.3 From 15903eb66ada34ca5d26d73d6ed2e075883a8278 Mon Sep 17 00:00:00 2001 From: Cong Tang Date: Fri, 26 May 2017 12:06:50 +0800 Subject: ASoc: msm: Support TDM dai clk attribute and freq configuration TDM dai driver support optional clk attribute configuration from device tree. To configure TDM interface as PCM mode, the clk attribute should be couple no. Implement set_sysclk callback to update TDM interface clk freq. CRs-Fixed: 2054358 Change-Id: I41edaa8d99325e9582e04ddb81a6ad5b5e4435bc Signed-off-by: Cong Tang --- .../devicetree/bindings/sound/qcom-audio-dev.txt | 7 +++ sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c | 57 ++++++++++++++++++++++ 2 files changed, 64 insertions(+) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/qcom-audio-dev.txt b/Documentation/devicetree/bindings/sound/qcom-audio-dev.txt index 38e056cdc0ee..db21a2b58c2b 100644 --- a/Documentation/devicetree/bindings/sound/qcom-audio-dev.txt +++ b/Documentation/devicetree/bindings/sound/qcom-audio-dev.txt @@ -1311,6 +1311,13 @@ Optional properties: - pinctrl-x: Defines pinctrl state for each pin group. + - qcom,msm-cpudai-tdm-clk-attribute: Clock attribute for tdm. + 0 - Clk invalid attribute + 1 - Clk attribute couple no + 2 - Clk attribute couple dividend + 3 - Clk attribute couple divisor + 4 - Clk attribute invert couple no + Example: qcom,msm-dai-tdm-quat-rx { diff --git a/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c index fa8bdddacef2..3610901addea 100644 --- a/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c @@ -5147,6 +5147,27 @@ 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", @@ -6226,6 +6247,41 @@ 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) @@ -6653,6 +6709,7 @@ 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, }; -- cgit v1.2.3 From aea170507758700faf5f10b97466b6d829567fb0 Mon Sep 17 00:00:00 2001 From: Revathi Uddaraju Date: Thu, 6 Jul 2017 19:23:21 +0530 Subject: ASoC: msm: qdsp6v2: Set path type to LIVE_REC in LISTEN passthrough case ADSP SVA needs LIVE_REC path type for LISTEN pass through mode for successful loading of SVA App. Hence set path type to LIVE_REC in LISTEN pass through case. Change-Id: I4404f2a64f0add62782beab3a5d8d63615d8ac01 Signed-off-by: Revathi Uddaraju --- sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c index 974d4a582540..17174769495b 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c @@ -1088,7 +1088,7 @@ 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; -- cgit v1.2.3 From ddb173d8ad550ddb18d5b13450a982de69e491b4 Mon Sep 17 00:00:00 2001 From: Laxminath Kasam Date: Fri, 26 May 2017 20:09:52 +0530 Subject: ASoC: msm_sdw: Move the delay logic inside bulk write loop On consecutive writes in bulk write API, ensure delay is provided for atleast 100us between each soundwire master write for WR_DONE status update and reflect current register value. Also ensure delay in soundwire master read is present after register address update and before register value read. CRs-Fixed: 2035787 Change-Id: I8399c5ca32328abdd4e90b46d6f8d6a6c0225905 Signed-off-by: Laxminath Kasam --- sound/soc/codecs/msm_sdw/msm_sdw_cdc.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) (limited to 'sound') 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); -- cgit v1.2.3 From 701ac49eb764851f08de0ca5ce46ed355fc6ee9c Mon Sep 17 00:00:00 2001 From: Varun Balaraj Date: Thu, 13 Jul 2017 13:30:15 +0530 Subject: ASoC: q6dspv2: Support for pan-scale and downmix set param Add support for set params on ASM/ADM for MFC and Volume modules. Make PSPD mitrix set param api generic. Change-Id: I75a5b9e3fd2316b75be41439848f89190944bc36 Signed-off-by: Varun Balaraj --- include/sound/apr_audio-v2.h | 54 ++++++ include/sound/q6adm-v2.h | 10 +- include/sound/q6asm-v2.h | 8 + sound/soc/msm/qdsp6v2/msm-qti-pp-config.c | 10 +- sound/soc/msm/qdsp6v2/q6adm.c | 6 +- sound/soc/msm/qdsp6v2/q6asm.c | 292 +++++++++++++++++++++++++++++- 6 files changed, 368 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/include/sound/apr_audio-v2.h b/include/sound/apr_audio-v2.h index 74995a0cdbad..00cba88b6082 100644 --- a/include/sound/apr_audio-v2.h +++ b/include/sound/apr_audio-v2.h @@ -625,6 +625,10 @@ struct audproc_softvolume_params { */ #define AUDPROC_PARAM_ID_MFC_OUTPUT_MEDIA_FORMAT 0x00010913 +/* ID of the Coefficient parameter used by AUDPROC_MODULE_ID_CHMIXER to + *configure the channel mixer weighting coefficients. + */ +#define AUDPROC_CHMIXER_PARAM_ID_COEFF 0x00010342 struct audproc_mfc_output_media_fmt { struct adm_cmd_set_pp_params_v5 params; @@ -3902,6 +3906,14 @@ struct asm_softvolume_params { u32 rampingcurve; } __packed; +struct asm_stream_pan_ctrl_params { + uint16_t num_output_channels; + uint16_t num_input_channels; + uint16_t output_channel_map[8]; + uint16_t input_channel_map[8]; + uint16_t gain[64]; +} __packed; + #define ASM_END_POINT_DEVICE_MATRIX 0 #define PCM_CHANNEL_NULL 0 @@ -7802,6 +7814,48 @@ struct asm_volume_ctrl_lr_chan_gain { /*< Linear gain in Q13 format for the right channel.*/ } __packed; +struct audproc_chmixer_param_coeff { + uint32_t index; + uint16_t num_output_channels; + uint16_t num_input_channels; +} __packed; + + +/* ID of the Multichannel Volume Control parameters used by + * AUDPROC_MODULE_ID_VOL_CTRL. + */ +#define AUDPROC_PARAM_ID_MULTICHANNEL_GAIN 0x00010713 + +/* Payload of the AUDPROC_PARAM_ID_MULTICHANNEL_GAIN channel type/gain + * pairs used by the Volume Control module. + * This structure immediately follows the + * audproc_volume_ctrl_multichannel_gain_t structure. + */ +struct audproc_volume_ctrl_channel_type_gain_pair { + uint8_t channel_type; + /* Channel type for which the gain setting is to be applied. */ + + uint8_t reserved1; + uint8_t reserved2; + uint8_t reserved3; + + uint32_t gain; + /* Gain value for this channel in Q28 format. */ +} __packed; + +/* Payload of the AUDPROC_PARAM_ID_MULTICHANNEL_MUTE parameters used by + * the Volume Control module. + */ +struct audproc_volume_ctrl_multichannel_gain { + uint32_t num_channels; + /* Number of channels for which mute configuration is provided. Any + * channels present in the data for which mute configuration is not + * provided are set to unmute. + */ + + struct audproc_volume_ctrl_channel_type_gain_pair *gain_data; + /* Array of channel type/mute setting pairs. */ +} __packed; /* Structure for the mute configuration parameter for a volume control module. */ diff --git a/include/sound/q6adm-v2.h b/include/sound/q6adm-v2.h index e689e9357012..8c53bb9762ce 100644 --- a/include/sound/q6adm-v2.h +++ b/include/sound/q6adm-v2.h @@ -138,9 +138,13 @@ int adm_get_topology_for_port_copp_idx(int port_id, int copp_idx); int adm_get_indexes_from_copp_id(int copp_id, int *port_idx, int *copp_idx); -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); + +int adm_set_downmix_params(int port_id, int copp_idx, + unsigned int session_id, char *params, + uint32_t params_length); int adm_get_pp_topo_module_list(int port_id, int copp_idx, int32_t param_length, char *params); diff --git a/include/sound/q6asm-v2.h b/include/sound/q6asm-v2.h index d0dffbd15923..177c2f4da32e 100644 --- a/include/sound/q6asm-v2.h +++ b/include/sound/q6asm-v2.h @@ -611,6 +611,14 @@ int q6asm_set_softvolume(struct audio_client *ac, int q6asm_set_softvolume_v2(struct audio_client *ac, struct asm_softvolume_params *param, int instance); +/* Set panning and MFC params */ +int q6asm_set_mfc_panning_params(struct audio_client *ac, + struct asm_stream_pan_ctrl_params *pan_param); + +/* Set vol gain pair */ +int q6asm_set_vol_ctrl_gain_pair(struct audio_client *ac, + struct asm_stream_pan_ctrl_params *pan_param); + /* Send left-right channel gain */ int q6asm_set_lrgain(struct audio_client *ac, int left_gain, int right_gain); 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/q6adm.c b/sound/soc/msm/qdsp6v2/q6adm.c index 28aaf2172221..4aad16db3e0f 100644 --- a/sound/soc/msm/qdsp6v2/q6adm.c +++ b/sound/soc/msm/qdsp6v2/q6adm.c @@ -781,9 +781,9 @@ 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; diff --git a/sound/soc/msm/qdsp6v2/q6asm.c b/sound/soc/msm/qdsp6v2/q6asm.c index 44e5b01b1ccb..d3cff9fccf59 100644 --- a/sound/soc/msm/qdsp6v2/q6asm.c +++ b/sound/soc/msm/qdsp6v2/q6asm.c @@ -44,7 +44,7 @@ #define TRUE 0x01 #define FALSE 0x00 #define SESSION_MAX 8 - +#define ASM_MAX_CHANNELS 8 enum { ASM_TOPOLOGY_CAL = 0, ASM_CUSTOM_TOP_CAL, @@ -7419,6 +7419,296 @@ 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; + uint16_t *asm_params = 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 * sizeof(uint16_t) * + 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)); + 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->num_input_channels * sizeof(uint16_t))), + pan_param->gain, + (pan_param->num_output_channels * sizeof(uint16_t)) * + (pan_param->num_input_channels * sizeof(uint16_t))); + + 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; -- cgit v1.2.3 From a48f734c3b1655d532a3c4795729cc13a7faddd6 Mon Sep 17 00:00:00 2001 From: Varun Balaraj Date: Thu, 13 Jul 2017 13:42:12 +0530 Subject: ASoC: msm: Support mixer controls for Pan/scale & downmix Add changes to register and receive parameters for pan/scale and downmix operations. Change-Id: If0e747304595f9ed3bd19b25e3f4eab2db382a67 Signed-off-by: Varun Balaraj --- include/sound/apr_audio-v2.h | 19 +++ sound/soc/msm/msm-dai-fe.c | 95 +++++++++++ sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c | 264 +++++++++++++++++++++++++++++ sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c | 174 +++++++++++++++++++ sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h | 11 +- 5 files changed, 561 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/include/sound/apr_audio-v2.h b/include/sound/apr_audio-v2.h index 00cba88b6082..a1d8b04d08ac 100644 --- a/include/sound/apr_audio-v2.h +++ b/include/sound/apr_audio-v2.h @@ -625,11 +625,30 @@ struct audproc_softvolume_params { */ #define AUDPROC_PARAM_ID_MFC_OUTPUT_MEDIA_FORMAT 0x00010913 +/* ID of the Channel Mixer module, which is used to configure + * channel-mixer related parameters. + * This module supports the AUDPROC_CHMIXER_PARAM_ID_COEFF parameter ID. + */ +#define AUDPROC_MODULE_ID_CHMIXER 0x00010341 + /* ID of the Coefficient parameter used by AUDPROC_MODULE_ID_CHMIXER to *configure the channel mixer weighting coefficients. */ #define AUDPROC_CHMIXER_PARAM_ID_COEFF 0x00010342 +/* Payload of the per-session, per-device parameter data of the + * #ADM_CMD_SET_PSPD_MTMX_STRTR_PARAMS_V5 command or + * #ADM_CMD_SET_PSPD_MTMX_STRTR_PARAMS_V6 command. + * Immediately following this structure are param_size bytes of parameter + * data. The structure and size depend on the module_id/param_id pair. + */ +struct adm_pspd_param_data_t { + uint32_t module_id; + uint32_t param_id; + uint16_t param_size; + uint16_t reserved; +} __packed; + struct audproc_mfc_output_media_fmt { struct adm_cmd_set_pp_params_v5 params; struct adm_param_data_v5 data; diff --git a/sound/soc/msm/msm-dai-fe.c b/sound/soc/msm/msm-dai-fe.c index 24dbbedf0be7..cc89408fcb39 100644 --- a/sound/soc/msm/msm-dai-fe.c +++ b/sound/soc/msm/msm-dai-fe.c @@ -2641,6 +2641,101 @@ 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, + }, + .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, + }, }; static int msm_fe_dai_dev_probe(struct platform_device *pdev) diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c index 46a3324d2d6b..de769e8b806c 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c @@ -1603,6 +1603,262 @@ 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++) { + pan_param.gain[i] = + ucontrol->value.integer.value[len++]; + } + 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 +1975,14 @@ 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..bf7a9ef07ed3 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 #include #include +#include +#include #include "msm-pcm-routing-v2.h" #include "msm-pcm-routing-devdep.h" @@ -4090,6 +4092,21 @@ 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), }; static const struct snd_kcontrol_new mi2s_rx_mixer_controls[] = { @@ -4612,6 +4629,21 @@ 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), }; static const struct snd_kcontrol_new display_port_mixer_controls[] = { @@ -4760,6 +4792,21 @@ 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), }; static const struct snd_kcontrol_new slimbus_7_rx_mixer_controls[] = { @@ -4811,6 +4858,21 @@ 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), }; static const struct snd_kcontrol_new usb_audio_rx_mixer_controls[] = { @@ -11314,6 +11376,11 @@ 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("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), @@ -12450,6 +12517,11 @@ 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", NULL, "SLIMBUS_0_RX Audio Mixer"}, {"SLIMBUS_2_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12504,6 +12576,11 @@ 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", NULL, "HDMI Mixer"}, {"DISPLAY_PORT Mixer", "MultiMedia1", "MM_DL1"}, @@ -12575,6 +12652,11 @@ 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", NULL, "SLIMBUS_6_RX Audio Mixer"}, {"SLIMBUS_7_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12593,6 +12675,11 @@ 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", NULL, "SLIMBUS_7_RX Audio Mixer"}, {"USB_AUDIO_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -15486,6 +15573,93 @@ 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; + + 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 * sizeof(uint16_t) * + 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)); + 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->num_input_channels * sizeof(uint16_t))), + dnmix_param->gain, + (dnmix_param->num_output_channels * sizeof(uint16_t)) * + (dnmix_param->num_input_channels * sizeof(uint16_t))); + + 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..fa948abd1b4c 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h +++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h @@ -193,6 +193,11 @@ 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_CS_VOICE, MSM_FRONTEND_DAI_VOIP, MSM_FRONTEND_DAI_AFE_RX, @@ -218,8 +223,8 @@ 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_MULTIMEDIA25 + 1) +#define MSM_FRONTEND_DAI_MM_MAX_ID MSM_FRONTEND_DAI_MULTIMEDIA25 enum { MSM_BACKEND_DAI_PRI_I2S_RX = 0, @@ -483,4 +488,6 @@ 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*/ -- cgit v1.2.3 From f829f461285235daa54976a4600cf4c210c5c8a4 Mon Sep 17 00:00:00 2001 From: Varun Balaraj Date: Mon, 3 Jul 2017 10:34:56 +0530 Subject: ASoC: msm8998: Add MM front end DAI links for interactive streams Add MM front end DAI links to support concurrent interactive streams Change-Id: I9cd95d855f2ddc3345453807e4be235c430b9ef3 Signed-off-by: Varun Balaraj --- sound/soc/msm/msm8998.c | 85 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 85 insertions(+) (limited to 'sound') diff --git a/sound/soc/msm/msm8998.c b/sound/soc/msm/msm8998.c index 0b37e7e073aa..82bdb4af68d8 100644 --- a/sound/soc/msm/msm8998.c +++ b/sound/soc/msm/msm8998.c @@ -5526,6 +5526,91 @@ static struct snd_soc_dai_link msm_common_misc_fe_dai_links[] = { .ignore_pmdown_time = 1, .be_id = MSM_FRONTEND_DAI_MULTIMEDIA18, }, + { + .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, + }, }; static struct snd_soc_dai_link msm_common_be_dai_links[] = { -- cgit v1.2.3 From 39cdcfaa36202e960970d99442a184b34dff9860 Mon Sep 17 00:00:00 2001 From: Vatsal Bucha Date: Tue, 11 Jul 2017 12:45:29 +0530 Subject: ASoC: msm: qdsp6v2: Modify name for Tertiary mi2s RX and TX Change name from "TERTIARY_MI2S" to "TERT_MI2S" for RX and TX in order to maintain consistency with other Mi2S backends. CRs-Fixed: 2075701 Change-Id: Iceb789307306183ffe2eaae7f58ecc2a7590d627 Signed-off-by: Vatsal Bucha --- sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h index 19e726001d25..d73624a40cb7 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" -- cgit v1.2.3 From ac162df12cca48d4a0daf025bc98be14250c5b7e Mon Sep 17 00:00:00 2001 From: Manish Dewangan Date: Wed, 5 Jul 2017 16:13:14 +0530 Subject: ASoc: msm: qdspv2: Fix latency calculation in pcm offload path Continuous latency for pcm offload is incorrect as buffer offset is not getting considered while pattern matching. Fix this by considering buffer offset while calculating continuous latency. CRs-Fixed: 2059729 Change-Id: I943da364f99a4152119d34a98b343e6e1cbad5b8 Signed-off-by: Manish Dewangan --- sound/soc/msm/qdsp6v2/q6asm.c | 28 ++++++++++++++++++++-------- 1 file changed, 20 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/msm/qdsp6v2/q6asm.c b/sound/soc/msm/qdsp6v2/q6asm.c index d3cff9fccf59..a44569530846 100644 --- a/sound/soc/msm/qdsp6v2/q6asm.c +++ b/sound/soc/msm/qdsp6v2/q6asm.c @@ -423,14 +423,22 @@ 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 +448,8 @@ 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 +496,7 @@ 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; } @@ -7979,6 +7987,7 @@ 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__); @@ -8040,7 +8049,10 @@ int q6asm_async_write(struct audio_client *ac, } } - config_debug_fs_write(ab); + 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) { @@ -8187,7 +8199,7 @@ 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) { -- cgit v1.2.3 From 7357058c09a8a923f6698cf045e9ab5c677bb32b Mon Sep 17 00:00:00 2001 From: Haynes Mathew George Date: Wed, 5 Jul 2017 15:39:25 -0700 Subject: ASoC: msm: qdsp6v2: Reserve MultiMedia16 for NOIRQ Reserve MultiMedia16 for second instance of NOIRQ CRs-Fixed: 2071838 Change-Id: I6a75d9a5e85585f65262ce486c992402ac58bf5f Signed-off-by: Haynes Mathew George --- sound/soc/msm/apq8096-auto.c | 7 +- sound/soc/msm/msm-dai-fe.c | 15 ++- sound/soc/msm/msm8996.c | 9 +- sound/soc/msm/msm8998.c | 7 +- sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c | 148 +++++++++++++++++++++++++++++ sound/soc/msm/sdm660-ext-dai-links.c | 6 +- sound/soc/msm/sdm660-internal.c | 6 +- 7 files changed, 181 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/msm/apq8096-auto.c b/sound/soc/msm/apq8096-auto.c index b6121d75c148..1d0fc93eb77f 100644 --- a/sound/soc/msm/apq8096-auto.c +++ b/sound/soc/msm/apq8096-auto.c @@ -3480,12 +3480,13 @@ 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-dai-fe.c b/sound/soc/msm/msm-dai-fe.c index 24dbbedf0be7..a0ff2e2c64c8 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, }, 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..c287e31844a8 100644 --- a/sound/soc/msm/msm8998.c +++ b/sound/soc/msm/msm8998.c @@ -5304,12 +5304,13 @@ 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", diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c index 974d4a582540..d20434bf8835 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c @@ -3640,6 +3640,11 @@ 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], @@ -7087,6 +7092,114 @@ 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, @@ -11323,6 +11436,7 @@ 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), @@ -12059,6 +12173,8 @@ 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, @@ -12389,6 +12505,8 @@ 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, @@ -12640,6 +12758,7 @@ 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"}, @@ -13208,6 +13327,7 @@ 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"}, @@ -13226,12 +13346,15 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13246,9 +13369,11 @@ 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"}, @@ -13410,6 +13535,24 @@ 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 +13635,10 @@ 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 +13654,7 @@ 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"}, @@ -13518,6 +13664,7 @@ 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"}, @@ -13928,6 +14075,7 @@ 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"}, 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, -- cgit v1.2.3 From 778f131a018d07de191694d17f33044066355885 Mon Sep 17 00:00:00 2001 From: Derek Chen Date: Fri, 26 May 2017 19:17:03 -0400 Subject: ASoC: msm: qdsp6v2: Add MultiMedia21 capture and routing Add MultiMedia21 capture and routing support for ASM loopback from AUX PCM TX to QUAT TDM RX. CRs-fixed: 2069348 Signed-off-by: Derek Chen Change-Id: I632392d98092ff31fa9bdb13c02eef8c2a0a54a3 --- sound/soc/msm/msm-dai-fe.c | 14 +++ sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c | 175 +++++++++++++++++++++++++++++ 2 files changed, 189 insertions(+) (limited to 'sound') diff --git a/sound/soc/msm/msm-dai-fe.c b/sound/soc/msm/msm-dai-fe.c index cc89408fcb39..468afdc81424 100644 --- a/sound/soc/msm/msm-dai-fe.c +++ b/sound/soc/msm/msm-dai-fe.c @@ -2656,6 +2656,20 @@ static struct snd_soc_dai_driver msm_fe_dais[] = { .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, diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c index 2bf61521ad52..b99549d62e8a 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c @@ -603,6 +603,21 @@ static struct msm_pcm_routing_fdai_data {{0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} }, {0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} } }, /* 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} } }, /* CS_VOICE */ @@ -5227,6 +5242,9 @@ 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), }; static const struct snd_kcontrol_new sec_auxpcm_rx_mixer_controls[] = { @@ -5287,6 +5305,9 @@ 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), }; static const struct snd_kcontrol_new tert_auxpcm_rx_mixer_controls[] = { @@ -5440,6 +5461,9 @@ 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), }; static const struct snd_kcontrol_new pri_tdm_rx_1_mixer_controls[] = { @@ -5491,6 +5515,9 @@ 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), }; static const struct snd_kcontrol_new pri_tdm_rx_2_mixer_controls[] = { @@ -5542,6 +5569,9 @@ 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), }; static const struct snd_kcontrol_new pri_tdm_rx_3_mixer_controls[] = { @@ -5593,6 +5623,9 @@ 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), }; static const struct snd_kcontrol_new pri_tdm_tx_0_mixer_controls[] = { @@ -5695,6 +5728,9 @@ 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), }; static const struct snd_kcontrol_new sec_tdm_rx_1_mixer_controls[] = { @@ -5746,6 +5782,9 @@ 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), }; static const struct snd_kcontrol_new sec_tdm_rx_2_mixer_controls[] = { @@ -5797,6 +5836,9 @@ 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), }; static const struct snd_kcontrol_new sec_tdm_rx_3_mixer_controls[] = { @@ -5848,6 +5890,9 @@ 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), }; static const struct snd_kcontrol_new sec_tdm_tx_0_mixer_controls[] = { @@ -5950,6 +5995,9 @@ 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), }; static const struct snd_kcontrol_new tert_tdm_tx_0_mixer_controls[] = { @@ -6052,6 +6100,9 @@ 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), }; static const struct snd_kcontrol_new tert_tdm_rx_2_mixer_controls[] = { @@ -6103,6 +6154,9 @@ 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), }; static const struct snd_kcontrol_new tert_tdm_rx_3_mixer_controls[] = { @@ -6154,6 +6208,9 @@ 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), }; static const struct snd_kcontrol_new tert_tdm_rx_4_mixer_controls[] = { @@ -6205,6 +6262,9 @@ 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), }; static const struct snd_kcontrol_new quat_tdm_rx_0_mixer_controls[] = { @@ -6259,6 +6319,9 @@ 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), }; static const struct snd_kcontrol_new quat_tdm_tx_0_mixer_controls[] = { @@ -6364,6 +6427,9 @@ 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), }; static const struct snd_kcontrol_new quat_tdm_rx_2_mixer_controls[] = { @@ -6418,6 +6484,9 @@ 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), }; static const struct snd_kcontrol_new quat_tdm_rx_3_mixer_controls[] = { @@ -6472,6 +6541,9 @@ 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), }; static const struct snd_kcontrol_new mmul1_mixer_controls[] = { @@ -7350,6 +7422,63 @@ 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 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, @@ -11470,6 +11599,7 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { 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_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), @@ -12210,6 +12340,8 @@ 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("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, @@ -12964,6 +13096,7 @@ 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", "MultiMedia21", "MM_DL21"}, {"PRI_TDM_RX_0", NULL, "PRI_TDM_RX_0 Audio Mixer"}, {"PRI_TDM_RX_1 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -12982,6 +13115,7 @@ 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", NULL, "PRI_TDM_RX_1 Audio Mixer"}, {"PRI_TDM_RX_2 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13000,6 +13134,7 @@ 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", NULL, "PRI_TDM_RX_2 Audio Mixer"}, {"PRI_TDM_RX_3 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13018,6 +13153,7 @@ 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", NULL, "PRI_TDM_RX_3 Audio Mixer"}, {"PRI_TDM_TX_0 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13054,6 +13190,7 @@ 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", "MultiMedia21", "MM_DL21"}, {"SEC_TDM_RX_0", NULL, "SEC_TDM_RX_0 Audio Mixer"}, {"SEC_TDM_RX_1 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13072,6 +13209,7 @@ 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", NULL, "SEC_TDM_RX_1 Audio Mixer"}, {"SEC_TDM_RX_2 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13090,6 +13228,7 @@ 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", NULL, "SEC_TDM_RX_2 Audio Mixer"}, {"SEC_TDM_RX_3 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13108,6 +13247,7 @@ 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", NULL, "SEC_TDM_RX_3 Audio Mixer"}, {"SEC_TDM_TX_0 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13144,6 +13284,7 @@ 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", NULL, "TERT_TDM_RX_0 Audio Mixer"}, {"TERT_TDM_TX_0 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13180,6 +13321,7 @@ 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", NULL, "TERT_TDM_RX_1 Audio Mixer"}, {"TERT_TDM_RX_2 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13198,6 +13340,7 @@ 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", NULL, "TERT_TDM_RX_2 Audio Mixer"}, {"TERT_TDM_RX_3 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13216,6 +13359,7 @@ 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", NULL, "TERT_TDM_RX_3 Audio Mixer"}, {"TERT_TDM_RX_4 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13234,6 +13378,7 @@ 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", NULL, "TERT_TDM_RX_4 Audio Mixer"}, {"QUAT_TDM_RX_0 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13253,6 +13398,7 @@ 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", NULL, "QUAT_TDM_RX_0 Audio Mixer"}, {"PRI_TDM_RX_0 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13326,6 +13472,7 @@ 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", NULL, "QUAT_TDM_RX_1 Audio Mixer"}, {"QUAT_TDM_RX_2 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13345,6 +13492,7 @@ 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", NULL, "QUAT_TDM_RX_2 Audio Mixer"}, {"QUAT_TDM_RX_3 Audio Mixer", "MultiMedia1", "MM_DL1"}, @@ -13364,6 +13512,7 @@ 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", NULL, "QUAT_TDM_RX_3 Audio Mixer"}, {"MultiMedia1 Mixer", "PRI_TX", "PRI_I2S_TX"}, @@ -13566,6 +13715,25 @@ 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"}, @@ -13685,6 +13853,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"MM_UL18", NULL, "MultiMedia18 Mixer"}, {"MM_UL19", NULL, "MultiMedia19 Mixer"}, {"MM_UL20", NULL, "MultiMedia20 Mixer"}, + {"MM_UL21", NULL, "MultiMedia21 Mixer"}, {"AUX_PCM_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, {"AUX_PCM_RX Audio Mixer", "MultiMedia2", "MM_DL2"}, @@ -13702,6 +13871,9 @@ 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"}, @@ -13720,6 +13892,9 @@ 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"}, -- cgit v1.2.3 From c4295314273810a9ecea527f29c3d29d4469d213 Mon Sep 17 00:00:00 2001 From: Derek Chen Date: Fri, 26 May 2017 19:14:20 -0400 Subject: ASoC: msm: Add FE dai for MultiMedia21 for ASM Loopback Adding ASM Loopback FE Dai, MultiMedia21, for HFP call downlink. CRs-fixed: 2069348 Signed-off-by: Derek Chen Change-Id: I38f0c1ea71568eff2ba789cbe2d41b950c77f232 --- sound/soc/msm/apq8096-auto.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/msm/apq8096-auto.c b/sound/soc/msm/apq8096-auto.c index b6121d75c148..2ae78f75c340 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", -- cgit v1.2.3 From e29ec858f7408620eac5d0ae0994075f86fcf6e1 Mon Sep 17 00:00:00 2001 From: Siddartha Shaik Date: Fri, 14 Jul 2017 09:51:10 +0530 Subject: ASoC: msm: mixer control queue handling in sink stream Move mixer control queue handling to be part of playback stream in DSP transcode loopback driver. CRs-Fixed: 2076617 Change-Id: Ia8067237f80d509f969cf454f5afa08aaa6b0e52 Signed-off-by: Siddartha Shaik --- sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') 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 6bb85ca8e84e..5c5f7bc482c8 100644 --- a/sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c @@ -100,7 +100,7 @@ static void loopback_event_handler(uint32_t opcode, return; } - cstream = trans->source.cstream; + cstream = trans->sink.cstream; ac = trans->audio_client; /* @@ -223,6 +223,7 @@ static int msm_transcode_loopback_open(struct snd_compr_stream *cstream) ret = -EINVAL; goto exit; } + msm_adsp_init_mixer_ctl_pp_event_queue(rtd); } pr_debug("%s: num stream%d, stream name %s\n", __func__, @@ -237,8 +238,7 @@ 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; @@ -283,14 +283,14 @@ 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); + } 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; } -- cgit v1.2.3 From 50b40109fadad5af5194536eb0ac90cc00dbc465 Mon Sep 17 00:00:00 2001 From: Satish Babu Patakokila Date: Fri, 16 Jun 2017 17:33:40 -0700 Subject: ASoC: compress: Derive substream from stream based on direction commit 01b8cedfd0422326caae308641dcadaa85e0ca72 upstream. Currently compress driver hardcodes direction as playback to get substream from the stream. This results in getting the incorrect substream for compressed capture usecase. To fix this, remove the hardcoding and derive substream based on the stream direction. Signed-off-by: Satish Babu Patakokila Signed-off-by: Banajit Goswami Acked-By: Vinod Koul Signed-off-by: Mark Brown Signed-off-by: Greg Kroah-Hartman --- sound/soc/soc-compress.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index bb82bb966000..c1addf49c4f2 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -68,7 +68,8 @@ out: static int soc_compr_open_fe(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *fe = cstream->private_data; - struct snd_pcm_substream *fe_substream = fe->pcm->streams[0].substream; + struct snd_pcm_substream *fe_substream = + fe->pcm->streams[cstream->direction].substream; struct snd_soc_platform *platform = fe->platform; struct snd_soc_dpcm *dpcm; struct snd_soc_dapm_widget_list *list; @@ -412,7 +413,8 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, struct snd_compr_params *params) { struct snd_soc_pcm_runtime *fe = cstream->private_data; - struct snd_pcm_substream *fe_substream = fe->pcm->streams[0].substream; + struct snd_pcm_substream *fe_substream = + fe->pcm->streams[cstream->direction].substream; struct snd_soc_platform *platform = fe->platform; int ret = 0, stream; -- cgit v1.2.3 From 0665a04d950d33758c2a9407cb8e504fd5a13532 Mon Sep 17 00:00:00 2001 From: Weiyin Jiang Date: Fri, 28 Jul 2017 11:01:40 +0800 Subject: SoC: msm: audio-effects: return directly to avoid integer overflow Return error code directly to avoid further integer overflow leading to buffer overflow. Change-Id: I8b74efda227726494724f4387c45b5b6fa04637b CRs-Fixed: 2077909 Signed-off-by: Weiyin Jiang --- sound/soc/msm/qdsp6v2/msm-audio-effects-q6-v2.c | 60 ++++++++++++------------- 1 file changed, 30 insertions(+), 30 deletions(-) (limited to 'sound') 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; -- cgit v1.2.3 From e788a3c6f53fe2e86ebe6012218159a7e261b5f3 Mon Sep 17 00:00:00 2001 From: Siddartha Shaik Date: Wed, 19 Jul 2017 15:40:12 +0530 Subject: ASoC: msm: Add new FE Dais for DSP HW transcode loopback Add new frontend DAIs MultiMedia26 and MultiMedia27 for DSP HW transcode loopback. CRs-Fixed: 2077700 Change-Id: Iedbd7da5e83a556749d9dc03cf39b0f942d634fa Signed-off-by: Siddartha Shaik --- sound/soc/msm/msm-dai-fe.c | 39 ++++++ sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c | 197 ++++++++++++++++++++++++++++- sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h | 6 +- 3 files changed, 239 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/msm/msm-dai-fe.c b/sound/soc/msm/msm-dai-fe.c index 468afdc81424..9be628cc2351 100644 --- a/sound/soc/msm/msm-dai-fe.c +++ b/sound/soc/msm/msm-dai-fe.c @@ -2750,6 +2750,45 @@ static struct snd_soc_dai_driver msm_fe_dais[] = { .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/qdsp6v2/msm-pcm-routing-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c index b99549d62e8a..e92065d0412d 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c @@ -618,6 +618,12 @@ static struct msm_pcm_routing_fdai_data {{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 */ @@ -3827,6 +3833,9 @@ 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[] = { @@ -3887,6 +3896,9 @@ 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[] = { @@ -3947,7 +3959,9 @@ 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[] = { @@ -3999,6 +4013,9 @@ 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[] = { @@ -4059,6 +4076,9 @@ 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[] = { @@ -4134,6 +4154,9 @@ static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = { 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[] = { @@ -4194,6 +4217,9 @@ 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[] = { @@ -4254,6 +4280,9 @@ 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[] = { @@ -4314,6 +4343,9 @@ 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[] = { @@ -4368,6 +4400,9 @@ 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[] = { @@ -4434,6 +4469,9 @@ 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[] = { @@ -4494,6 +4532,9 @@ 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[] = { @@ -4545,6 +4586,9 @@ 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[] = { @@ -4596,6 +4640,9 @@ 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[] = { @@ -4671,6 +4718,9 @@ static const struct snd_kcontrol_new hdmi_mixer_controls[] = { 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[] = { @@ -4722,6 +4772,9 @@ 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 */ @@ -4834,6 +4887,9 @@ static const struct snd_kcontrol_new slimbus_6_rx_mixer_controls[] = { 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[] = { @@ -4900,6 +4956,9 @@ static const struct snd_kcontrol_new slimbus_7_rx_mixer_controls[] = { 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[] = { @@ -4951,6 +5010,9 @@ 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[] = { @@ -5011,6 +5073,9 @@ 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[] = { @@ -5062,6 +5127,9 @@ 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[] = { @@ -5122,6 +5190,9 @@ 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[] = { @@ -5182,6 +5253,9 @@ 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[] = { @@ -5245,6 +5319,9 @@ static const struct snd_kcontrol_new auxpcm_rx_mixer_controls[] = { 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[] = { @@ -5308,6 +5385,9 @@ static const struct snd_kcontrol_new sec_auxpcm_rx_mixer_controls[] = { 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[] = { @@ -5359,6 +5439,9 @@ 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[] = { @@ -5410,6 +5493,9 @@ 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[] = { @@ -5464,6 +5550,9 @@ static const struct snd_kcontrol_new pri_tdm_rx_0_mixer_controls[] = { 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[] = { @@ -5518,6 +5607,9 @@ static const struct snd_kcontrol_new pri_tdm_rx_1_mixer_controls[] = { 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[] = { @@ -5572,6 +5664,9 @@ static const struct snd_kcontrol_new pri_tdm_rx_2_mixer_controls[] = { 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[] = { @@ -5626,6 +5721,9 @@ static const struct snd_kcontrol_new pri_tdm_rx_3_mixer_controls[] = { 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[] = { @@ -5731,6 +5829,9 @@ static const struct snd_kcontrol_new sec_tdm_rx_0_mixer_controls[] = { 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[] = { @@ -5785,6 +5886,9 @@ static const struct snd_kcontrol_new sec_tdm_rx_1_mixer_controls[] = { 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[] = { @@ -5839,6 +5943,9 @@ static const struct snd_kcontrol_new sec_tdm_rx_2_mixer_controls[] = { 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[] = { @@ -5893,6 +6000,9 @@ static const struct snd_kcontrol_new sec_tdm_rx_3_mixer_controls[] = { 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[] = { @@ -5998,6 +6108,9 @@ static const struct snd_kcontrol_new tert_tdm_rx_0_mixer_controls[] = { 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[] = { @@ -6103,6 +6216,9 @@ static const struct snd_kcontrol_new tert_tdm_rx_1_mixer_controls[] = { 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[] = { @@ -6157,6 +6273,9 @@ static const struct snd_kcontrol_new tert_tdm_rx_2_mixer_controls[] = { 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[] = { @@ -6211,6 +6330,9 @@ static const struct snd_kcontrol_new tert_tdm_rx_3_mixer_controls[] = { 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[] = { @@ -6265,6 +6387,9 @@ static const struct snd_kcontrol_new tert_tdm_rx_4_mixer_controls[] = { 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[] = { @@ -6322,6 +6447,9 @@ static const struct snd_kcontrol_new quat_tdm_rx_0_mixer_controls[] = { 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[] = { @@ -6430,6 +6558,9 @@ static const struct snd_kcontrol_new quat_tdm_rx_1_mixer_controls[] = { 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[] = { @@ -6487,6 +6618,9 @@ static const struct snd_kcontrol_new quat_tdm_rx_2_mixer_controls[] = { 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[] = { @@ -6544,6 +6678,9 @@ static const struct snd_kcontrol_new quat_tdm_rx_3_mixer_controls[] = { 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[] = { @@ -7479,6 +7616,18 @@ static const struct snd_kcontrol_new mmul21_mixer_controls[] = { 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, @@ -11586,6 +11735,7 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { 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), @@ -11600,6 +11750,7 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { 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), @@ -12342,6 +12493,8 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { 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, @@ -12689,6 +12842,7 @@ 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"}, @@ -12707,6 +12861,7 @@ 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"}, @@ -12730,6 +12885,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -12748,6 +12904,7 @@ 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"}, @@ -12766,6 +12923,7 @@ 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"}, @@ -12789,6 +12947,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -12807,6 +12966,7 @@ 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"}, @@ -12825,6 +12985,7 @@ 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 */ @@ -12865,6 +13026,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -12888,6 +13050,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -12906,6 +13069,7 @@ 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"}, @@ -12953,6 +13117,7 @@ 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"}, @@ -12970,6 +13135,7 @@ 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"}, @@ -12986,6 +13152,7 @@ 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"}, @@ -13002,6 +13169,7 @@ 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"}, @@ -13025,6 +13193,7 @@ 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"}, @@ -13042,6 +13211,7 @@ 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"}, @@ -13059,6 +13229,7 @@ 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"}, @@ -13078,6 +13249,7 @@ 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"}, @@ -13097,6 +13269,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13116,6 +13289,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13135,6 +13309,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13154,6 +13329,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13191,6 +13367,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13210,6 +13387,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13229,6 +13407,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13248,6 +13427,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13285,6 +13465,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13322,6 +13503,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13341,6 +13523,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13360,6 +13543,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13379,6 +13563,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13399,6 +13584,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13417,6 +13603,7 @@ 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"}, @@ -13435,6 +13622,7 @@ 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"}, @@ -13473,6 +13661,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13493,6 +13682,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13513,6 +13703,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13523,6 +13714,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13532,6 +13724,7 @@ 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"}, @@ -13550,6 +13743,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, @@ -13854,6 +14048,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"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"}, diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h index a8daff09db58..c640be343e10 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h +++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h @@ -198,6 +198,8 @@ enum { 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, @@ -223,8 +225,8 @@ enum { MSM_FRONTEND_DAI_MAX, }; -#define MSM_FRONTEND_DAI_MM_SIZE (MSM_FRONTEND_DAI_MULTIMEDIA25 + 1) -#define MSM_FRONTEND_DAI_MM_MAX_ID MSM_FRONTEND_DAI_MULTIMEDIA25 +#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, -- cgit v1.2.3 From e2e5a2e1222dcc68c29bb1a531b3bb926fd0fa04 Mon Sep 17 00:00:00 2001 From: Siddartha Shaik Date: Wed, 19 Jul 2017 19:22:47 +0530 Subject: ASoC: msm8998: Update DAI IDs for DSP HW Transcode loopback Replace Multimedia14 and Multimedia18 with Multimedia26 and Multimedia27 respectively for transcode loopback CRs-Fixed: 2077700 Change-Id: I40e6666882859a5699e86856a0b8769eb295f748 Signed-off-by: Siddartha Shaik --- sound/soc/msm/msm8998.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/msm/msm8998.c b/sound/soc/msm/msm8998.c index 8d737c3d3351..87ad98ec1913 100644 --- a/sound/soc/msm/msm8998.c +++ b/sound/soc/msm/msm8998.c @@ -5509,7 +5509,7 @@ 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, @@ -5520,12 +5520,12 @@ 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, @@ -5535,7 +5535,7 @@ 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", -- cgit v1.2.3 From aa5d0fc24044bec2dadd0007fd881fd4a09f7f6a Mon Sep 17 00:00:00 2001 From: Haynes Mathew George Date: Tue, 25 Jul 2017 14:02:13 -0700 Subject: ASoC: msm: qdsp6v2: Open unique COPP port for concurrent ULL streams 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. We need to open a separate COPP port for concurrent ULL streams CRs-Fixed: 2083105 Change-Id: I569b32830145d6dae99449d0bc4148b2f60b101d Signed-off-by: Haynes Mathew George Acked-by: Shiv Maliyappanahalli --- sound/soc/msm/qdsp6v2/q6adm.c | 16 +++++++++++++--- 1 file changed, 13 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/msm/qdsp6v2/q6adm.c b/sound/soc/msm/qdsp6v2/q6adm.c index e1bfc950d0e3..018681309f2e 100644 --- a/sound/soc/msm/qdsp6v2/q6adm.c +++ b/sound/soc/msm/qdsp6v2/q6adm.c @@ -2364,7 +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", @@ -2418,8 +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) { -- cgit v1.2.3 From dc19e98eb86e37371a0c7817abd1f2a19480187c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 23 Dec 2016 11:21:10 +0200 Subject: ASoC: tlv320aic3x: Mark the RESET register as volatile [ Upstream commit 63c3194b82530bd71fd49db84eb7ab656b8d404a ] The RESET register only have one self clearing bit and it should not be cached. If it is cached, when we sync the registers back to the chip we will initiate a software reset as well, which is not desirable. Signed-off-by: Peter Ujfalusi Reviewed-by: Jarkko Nikula Signed-off-by: Mark Brown Signed-off-by: Sasha Levin Signed-off-by: Greg Kroah-Hartman --- sound/soc/codecs/tlv320aic3x.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') 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, }; -- cgit v1.2.3 From 1a734b39857e0321573e11b1d3f762f3551fb5a8 Mon Sep 17 00:00:00 2001 From: John Hsu Date: Tue, 20 Dec 2016 12:03:09 +0800 Subject: ASoC: nau8825: fix invalid configuration in Pre-Scalar of FLL [ Upstream commit a1792cda51300e15b03549cccf0b09f3be82e697 ] The clk_ref_div is not configured in the correct position of the register. The patch fixes that clk_ref_div, Pre-Scalar, is assigned the wrong value. Signed-off-by: John Hsu Signed-off-by: Mark Brown Signed-off-by: Sasha Levin Signed-off-by: Greg Kroah-Hartman --- sound/soc/codecs/nau8825.c | 3 ++- sound/soc/codecs/nau8825.h | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') 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) -- cgit v1.2.3 From cab7c045f5fd2c269490c50469966e5fdff1aa53 Mon Sep 17 00:00:00 2001 From: Ioan-Adrian Ratiu Date: Thu, 5 Jan 2017 00:37:47 +0200 Subject: ALSA: usb-audio: test EP_FLAG_RUNNING at urb completion [ Upstream commit 13a6c8328e6056932dc680e447d4c5e8ad9add17 ] Testing EP_FLAG_RUNNING in snd_complete_urb() before running the completion logic allows us to save a few cpu cycles by returning early, skipping the pending urb in case the stream was stopped; the stop logic handles the urb and sets the completion callbacks to NULL. Signed-off-by: Ioan-Adrian Ratiu Signed-off-by: Takashi Iwai Signed-off-by: Sasha Levin Signed-off-by: Greg Kroah-Hartman --- sound/usb/endpoint.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index be1f511e4f54..ae2981460cd8 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 */ -- cgit v1.2.3 From 2f8e6140bb6ad439ec3e32e4bd4e06af9c3f43cd Mon Sep 17 00:00:00 2001 From: Patrick Lai Date: Sat, 31 Dec 2016 22:44:39 -0800 Subject: ASoC: dpcm: Avoid putting stream state to STOP when FE stream is paused [ Upstream commit 9f169b9f52a4afccdab7a7d2311b0c53a78a1e6b ] When multiple front-ends are using the same back-end, putting state of a front-end to STOP state upon receiving pause command will result in backend stream getting released by DPCM framework unintentionally. In order to avoid backend to be released when another active front-end stream is present, put the stream state to PAUSED state instead of STOP state. Signed-off-by: Patrick Lai Signed-off-by: Mark Brown Signed-off-by: Sasha Levin Signed-off-by: Greg Kroah-Hartman --- sound/soc/soc-pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 65b936e251ea..a1e605bbc465 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2073,9 +2073,11 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: fe->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; + break; } out: -- cgit v1.2.3 From 3ec77b8214c65a62701e80de4d457231f7a18860 Mon Sep 17 00:00:00 2001 From: Vidyakumar Athota Date: Mon, 26 Jun 2017 14:13:06 -0700 Subject: ASoC: msm: qdsp6v2: assign decoder memory to ADSP Get the physical address of memory from ion FD and assign it to ADSP for secure playback scenario. Change-Id: I51019f5da6fb5fedefc81504b192f985b851ab12 Signed-off-by: Vidyakumar Athota --- include/sound/q6asm-v2.h | 2 +- include/sound/q6core.h | 16 +- sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c | 177 ++++++++++++++++++--- .../soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c | 154 ++++++++++++++++-- sound/soc/msm/qdsp6v2/q6asm.c | 21 +-- sound/soc/msm/qdsp6v2/q6core.c | 62 ++++++++ 6 files changed, 383 insertions(+), 49 deletions(-) (limited to 'sound') diff --git a/include/sound/q6asm-v2.h b/include/sound/q6asm-v2.h index 177c2f4da32e..dcf7dcb4f6e4 100644 --- a/include/sound/q6asm-v2.h +++ b/include/sound/q6asm-v2.h @@ -639,7 +639,7 @@ int q6asm_send_audio_effects_params(struct audio_client *ac, char *params, int q6asm_send_stream_cmd(struct audio_client *ac, struct msm_adsp_event_data *data); -int q6asm_send_ion_fd(struct audio_client *ac, int fd); +int q6asm_audio_map_shm_fd(struct audio_client *ac, int fd); int q6asm_send_rtic_event_ack(struct audio_client *ac, void *param, uint32_t params_length); diff --git a/include/sound/q6core.h b/include/sound/q6core.h index 1d81bda4b513..4f55880d410f 100644 --- a/include/sound/q6core.h +++ b/include/sound/q6core.h @@ -1,4 +1,4 @@ -/* Copyright (c) 2012-2016, The Linux Foundation. All rights reserved. +/* Copyright (c) 2012-2016, 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 @@ -20,6 +20,8 @@ #define AVCS_CMDRSP_ADSP_EVENT_GET_STATE 0x0001290D bool q6core_is_adsp_ready(void); +int q6core_add_remove_pool_pages(phys_addr_t buf_add, uint32_t bufsz, + uint32_t mempool_id, bool add_pages); #define ADSP_CMD_SET_DTS_EAGLE_DATA_ID 0x00012919 #define DTS_EAGLE_LICENSE_ID 0x00028346 @@ -153,4 +155,16 @@ struct avcs_cmd_deregister_topologies { int32_t core_set_license(uint32_t key, uint32_t module_id); int32_t core_get_license_status(uint32_t module_id); +#define ADSP_MEMORY_MAP_HLOS_PHYSPOOL 4 +#define AVCS_CMD_ADD_POOL_PAGES 0x0001292E +#define AVCS_CMD_REMOVE_POOL_PAGES 0x0001292F + +struct avs_mem_assign_region { + struct apr_hdr hdr; + u32 pool_id; + u32 size; + u32 addr_lsw; + u32 addr_msw; +} __packed; + #endif /* __Q6CORE_H__ */ 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 #include +#include #include #include #include @@ -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-transcode-loopback-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c index 5c5f7bc482c8..a75de6d1d7c0 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 +27,7 @@ #include #include #include +#include #include #include #include @@ -45,6 +46,7 @@ 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]; }; struct loopback_stream { @@ -79,6 +81,7 @@ struct msm_transcode_loopback { int session_id; struct audio_client *audio_client; + int32_t shm_ion_fd; }; /* Transcode loopback global info struct */ @@ -179,6 +182,40 @@ 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; @@ -224,6 +261,13 @@ static int msm_transcode_loopback_open(struct snd_compr_stream *cstream) 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__, @@ -274,7 +318,11 @@ 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); @@ -286,6 +334,19 @@ static int msm_transcode_loopback_free(struct snd_compr_stream *cstream) if (cstream->direction == SND_COMPRESS_PLAYBACK) { memset(&trans->sink, 0, sizeof(struct loopback_stream)); 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)); } @@ -557,7 +618,7 @@ 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); @@ -566,7 +627,6 @@ 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) { @@ -596,10 +656,34 @@ 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; } @@ -773,7 +857,8 @@ 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"; @@ -785,7 +870,53 @@ 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, } }; @@ -877,9 +1008,14 @@ 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 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 ion fd Control\n", + pr_err("%s: Could not add transcode lib ion fd Control\n", __func__); rc = msm_transcode_add_event_ack_cmd_control(rtd); diff --git a/sound/soc/msm/qdsp6v2/q6asm.c b/sound/soc/msm/qdsp6v2/q6asm.c index a44569530846..d297acd62665 100644 --- a/sound/soc/msm/qdsp6v2/q6asm.c +++ b/sound/soc/msm/qdsp6v2/q6asm.c @@ -36,6 +36,7 @@ #include #include +#include #include #include #include @@ -7117,13 +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; @@ -7139,19 +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 */ 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; -- cgit v1.2.3 From 2fff7d7ae65ce4356ab75e22ebfdfd4ae05a2b65 Mon Sep 17 00:00:00 2001 From: Tanya Dixit Date: Thu, 3 Aug 2017 18:49:03 +0530 Subject: ASoC: msm: qdspv2: Fix missing mutex unlock in LSM client driver Fix missing mutex unlock in LSM client driver. CRs-fixed: 2059497 Change-Id: If5da5ab2eeb424b86c0e6476b69aa6ed36e7ea72 Signed-off-by: Tanya Dixit --- sound/soc/msm/qdsp6v2/msm-lsm-client.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) (limited to 'sound') 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); -- cgit v1.2.3 From 159d8f287092e6a9d3dabd77911624629bfe069c Mon Sep 17 00:00:00 2001 From: Bhalchandra Gajare Date: Mon, 22 May 2017 14:50:55 -0700 Subject: ASoC: wcd-dsp-mgr: fix race during subsystem restart Whenever subsystem restart or DSP boot occurs, the manager driver parses the dsp image and stores the segments in list. During certain back to back regression tests, it is observed that SSR is failing due to race between SSR handling and DSP enablement. Fix this by acquiring the ssr_mutex during enabling DSP and sequence to list operations such that there is no race conditions. Change-Id: I15c55bf96737e4ffd7e0faf571a1109ba6c38163 Signed-off-by: Bhalchandra Gajare --- sound/soc/codecs/wcd-dsp-mgr.c | 23 +++++++++++++++-------- 1 file changed, 15 insertions(+), 8 deletions(-) (limited to 'sound') 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; } -- cgit v1.2.3 From 9f408076d4ab63ee15f4ad1f4e3c0d16c93e3b28 Mon Sep 17 00:00:00 2001 From: Bhalchandra Gajare Date: Mon, 15 May 2017 19:14:49 -0700 Subject: ASoc: wcd934x-dsp-cntl: notify online event after clocks are disabled The moment the online event is notified to userspace, it may happen that the userspace might enable the WDSP. This causes race between the enabling of WDSP and SSR handling of WDSP. Change the sequence to notify online event after all SSR handling is completed. Change-Id: I3cb5d40034884cdfc35de957fab4dafd42d0697c Signed-off-by: Bhalchandra Gajare --- sound/soc/codecs/wcd934x/wcd934x-dsp-cntl.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') 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: -- cgit v1.2.3 From 9aa8e1cd38cec9e7f49e614f7e23a2a4f5e130e4 Mon Sep 17 00:00:00 2001 From: Siddartha Shaik Date: Fri, 11 Aug 2017 18:18:15 +0530 Subject: ASoC: msm8998: Add 88.2 and 176.4 khz support for MI2S Interface Extend MI2S supported sampling rates to support HDMI In feature requirements using DSP HW transcode loopback. CRs-Fixed: 2091424 Change-Id: Ic73cbec2473ab509f6a85bd7f6f60869b7986d7b Signed-off-by: Siddartha Shaik --- sound/soc/msm/msm8998.c | 21 +++++++++++++++++---- 1 file changed, 17 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/msm/msm8998.c b/sound/soc/msm/msm8998.c index b8bd32e046a3..9159ea642816 100644 --- a/sound/soc/msm/msm8998.c +++ b/sound/soc/msm/msm8998.c @@ -434,7 +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"}; @@ -2228,12 +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; @@ -2262,9 +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: -- cgit v1.2.3 From bcf5264d73e37e69001dcd0bcd46068656a294dd Mon Sep 17 00:00:00 2001 From: Siddartha Shaik Date: Fri, 11 Aug 2017 18:24:42 +0530 Subject: ASoC: msm: Quat MI2S capture dai update for 88.2 and 176.4 Khz rates Quaternary MI2S capture dai update to support HDMI In feature requirement using DSP HW transcode loopback. CRs-Fixed: 2091424 Change-Id: I144bd8ad8b0e2f708acf78be00e8a93a25a88f2e Signed-off-by: Siddartha Shaik --- sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c index 3610901addea..292b3d04f7d5 100644 --- a/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c @@ -4152,7 +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, -- cgit v1.2.3 From 4ef6a54e2c51d9a60365e16955e6f06c38e537d0 Mon Sep 17 00:00:00 2001 From: Siddartha Shaik Date: Thu, 10 Aug 2017 14:17:45 +0530 Subject: ASoC: msm: enable app type config for transcode loopback Add app type configuration support for DSP transcode loopback to enable required PP topologies. CRs-Fixed: 2092562 Change-Id: I60ee50d78ba3c0edc9df042a2a86e691d62f24b7 Signed-off-by: Siddartha Shaik --- .../soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c | 125 +++++++++++++++++++++ 1 file changed, 125 insertions(+) (limited to 'sound') 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 a75de6d1d7c0..55ae0638d0c7 100644 --- a/sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c @@ -42,6 +42,11 @@ #define LOOPBACK_SESSION_MAX_NUM_STREAMS 2 +#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 { @@ -747,6 +752,67 @@ 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_stream_cmd_control( struct snd_soc_pcm_runtime *rtd) { @@ -995,6 +1061,60 @@ 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_loopback_new(struct snd_soc_pcm_runtime *rtd) { int rc; @@ -1023,6 +1143,11 @@ 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__); + return 0; } -- cgit v1.2.3 From dd32d9226c5b818c78c9a3e28ddd17a1533269e5 Mon Sep 17 00:00:00 2001 From: Siddartha Shaik Date: Mon, 14 Aug 2017 00:57:46 +0530 Subject: ASoC: msm: volume control support for DSP transcode loopback Add stream volume control support for DSP transcode loopback to support mute configuration requirement. CRs-Fixed: 2092562 Change-Id: I2003e40c9888245c1b12f0e7fbd364170d5008cf Signed-off-by: Siddartha Shaik --- .../soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c | 121 +++++++++++++++++++++ 1 file changed, 121 insertions(+) (limited to 'sound') 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 55ae0638d0c7..cf558f2e0e1f 100644 --- a/sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c @@ -41,6 +41,8 @@ #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 @@ -52,6 +54,7 @@ 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; }; struct loopback_stream { @@ -813,6 +816,80 @@ 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) { @@ -1089,6 +1166,7 @@ static int msm_transcode_add_app_type_cfg_control( if (!rtd) { pr_err("%s NULL rtd\n", __func__); + return -EINVAL; } @@ -1114,6 +1192,44 @@ static int msm_transcode_add_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) { @@ -1148,6 +1264,11 @@ static int msm_transcode_loopback_new(struct snd_soc_pcm_runtime *rtd) 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; } -- cgit v1.2.3 From 97390bee20f5bfc16501c0d18f28cb5160b24f67 Mon Sep 17 00:00:00 2001 From: Siddartha Shaik Date: Mon, 14 Aug 2017 04:18:35 +0530 Subject: ASoC: msm: qdsp6v2: latency mode support for transcode loopback Add metadata to configure latency mode for DSP transcode loopback to set legacy or low latency path. Userspace API introduced to configure the same. CRs-Fixed: 2092562 Change-Id: I914c68a9e9d8647530b72c42548e571b7508c423 Signed-off-by: Siddartha Shaik --- include/uapi/sound/compress_offload.h | 7 +++ .../soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c | 55 ++++++++++++++++++++-- 2 files changed, 59 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 866ec3d2af69..7845fdd556fa 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -138,6 +138,11 @@ struct snd_compr_audio_info { #define SNDRV_COMPRESS_CLK_REC_MODE_NONE 0 #define SNDRV_COMPRESS_CLK_REC_MODE_AUTO 1 +enum sndrv_compress_latency_mode { + SNDRV_COMPRESS_LEGACY_LATENCY_MODE = 0, + SNDRV_COMPRESS_LOW_LATENCY_MODE = 1, +}; + /** * enum sndrv_compress_encoder * @SNDRV_COMPRESS_ENCODER_PADDING: no of samples appended by the encoder at the @@ -164,6 +169,7 @@ enum sndrv_compress_encoder { SNDRV_COMPRESS_START_DELAY = 9, SNDRV_COMPRESS_ENABLE_ADJUST_SESSION_CLOCK = 10, SNDRV_COMPRESS_ADJUST_SESSION_CLOCK = 11, + SNDRV_COMPRESS_LATENCY_MODE = 12, }; #define SNDRV_COMPRESS_PATH_DELAY SNDRV_COMPRESS_PATH_DELAY @@ -174,6 +180,7 @@ enum sndrv_compress_encoder { #define SNDRV_COMPRESS_ENABLE_ADJUST_SESSION_CLOCK \ SNDRV_COMPRESS_ENABLE_ADJUST_SESSION_CLOCK #define SNDRV_COMPRESS_ADJUST_SESSION_CLOCK SNDRV_COMPRESS_ADJUST_SESSION_CLOCK +#define SNDRV_COMPRESS_LATENCY_MODE SNDRV_COMPRESS_LATENCY_MODE /** * struct snd_compr_metadata - compressed stream metadata 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 cf558f2e0e1f..fdeb8a15ffee 100644 --- a/sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include #include @@ -55,6 +56,7 @@ 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 { @@ -406,6 +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; @@ -416,6 +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 = @@ -497,7 +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, @@ -516,7 +523,7 @@ 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, COMPRESSED_PASSTHROUGH_GEN); @@ -529,7 +536,7 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream, /* Opening Rx ADM in LOW_LATENCY mode by default */ msm_pcm_routing_reg_phy_stream( soc_pcm_rx->dai_link->be_id, - true, + trans->audio_client->perf_mode, trans->session_id, SNDRV_PCM_STREAM_PLAYBACK); pr_debug("%s: Successfully opened ADM sessions\n", __func__); @@ -562,6 +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) { @@ -1278,6 +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, }; @@ -1292,6 +1340,7 @@ 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; } -- cgit v1.2.3 From a935e0c8f8ea49fbb1d887d772700d0d541a205b Mon Sep 17 00:00:00 2001 From: Varun Balaraj Date: Tue, 15 Aug 2017 16:40:27 +0530 Subject: ASoC: msm: Fix pan control param gain to accommodate Q28 format Fix pan control gain param to accommodate Q28 format. Convert 32bit gain to be inlined with 16bit mixer coefficients of ADSP. Change-Id: I85ebd11c49b4af982f34091937acabb9e6580e8f Signed-off-by: Varun Balaraj --- include/sound/apr_audio-v2.h | 2 +- sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c | 12 ++++++++---- sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c | 13 ++++++++----- sound/soc/msm/qdsp6v2/q6asm.c | 15 +++++++++------ 4 files changed, 26 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/include/sound/apr_audio-v2.h b/include/sound/apr_audio-v2.h index 2680a13721c2..21b30bf7c74c 100644 --- a/include/sound/apr_audio-v2.h +++ b/include/sound/apr_audio-v2.h @@ -4012,7 +4012,7 @@ struct asm_stream_pan_ctrl_params { uint16_t num_input_channels; uint16_t output_channel_map[8]; uint16_t input_channel_map[8]; - uint16_t gain[64]; + uint32_t gain[64]; } __packed; #define ASM_END_POINT_DEVICE_MATRIX 0 diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c index de769e8b806c..b94eb6fbfeea 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c @@ -1668,7 +1668,7 @@ static int msm_pcm_playback_pan_scale_ctl_put(struct snd_kcontrol *kcontrol, 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) ? + !(ucontrol->value.integer.value[len++] > 0) ? 0 : 2 << 13; } } @@ -1679,8 +1679,12 @@ static int msm_pcm_playback_pan_scale_ctl_put(struct snd_kcontrol *kcontrol, 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++]; + ucontrol->value.integer.value[len++] << 14; } ret = q6asm_set_vol_ctrl_gain_pair(prtd->audio_client, &pan_param); @@ -1753,7 +1757,7 @@ static int msm_pcm_playback_dnmix_ctl_put(struct snd_kcontrol *kcontrol, struct msm_audio *prtd; struct asm_stream_pan_ctrl_params dnmix_param; - int be_id = ucontrol->value.integer.value[len]; + int be_id = ucontrol->value.integer.value[len++]; int stream_id = 0; if (!usr_info) { @@ -1809,7 +1813,7 @@ static int msm_pcm_playback_dnmix_ctl_put(struct snd_kcontrol *kcontrol, for (i = 0; i < dnmix_param.num_output_channels * dnmix_param.num_input_channels; i++) { dnmix_param.gain[i] = - ucontrol->value.integer.value[len++]; + ucontrol->value.integer.value[len++]; } } msm_routing_set_downmix_control_data(be_id, diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c index 38937612dcce..9e2e8b37ec10 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c @@ -16201,6 +16201,8 @@ int msm_routing_set_downmix_control_data(int be_id, int session_id, 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; @@ -16213,7 +16215,7 @@ int msm_routing_set_downmix_control_data(int be_id, int 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 * 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; @@ -16252,14 +16254,15 @@ int msm_routing_set_downmix_control_data(int be_id, int session_id, dnmix_param->num_output_channels * sizeof(uint16_t)), dnmix_param->input_channel_map, dnmix_param->num_input_channels * sizeof(uint16_t)); - memcpy(((u8 *)adm_params + + + 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))), - dnmix_param->gain, - (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, diff --git a/sound/soc/msm/qdsp6v2/q6asm.c b/sound/soc/msm/qdsp6v2/q6asm.c index d297acd62665..609ae8cd82ac 100644 --- a/sound/soc/msm/qdsp6v2/q6asm.c +++ b/sound/soc/msm/qdsp6v2/q6asm.c @@ -7543,7 +7543,9 @@ int q6asm_set_mfc_panning_params(struct audio_client *ac, struct asm_stream_param_data_v2 data; struct audproc_chmixer_param_coeff pan_cfg; uint16_t variable_payload = 0; - uint16_t *asm_params = NULL; + char *asm_params = NULL; + uint16_t ii; + uint16_t *dst_gain_ptr = NULL; sz = rc = i = 0; if (ac == NULL) { @@ -7604,7 +7606,7 @@ int q6asm_set_mfc_panning_params(struct audio_client *ac, variable_payload = pan_param->num_output_channels * sizeof(uint16_t)+ pan_param->num_input_channels * sizeof(uint16_t) + - pan_param->num_output_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; @@ -7664,15 +7666,16 @@ int q6asm_set_mfc_panning_params(struct audio_client *ac, pan_param->num_output_channels * sizeof(uint16_t)), pan_param->input_channel_map, pan_param->num_input_channels * sizeof(uint16_t)); - memcpy(((u8 *)asm_params + sizeof(struct apr_hdr) + + + 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))), - pan_param->gain, - (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) { -- cgit v1.2.3 From aa91dc149facde1d65739c56d52fda63c02d12d2 Mon Sep 17 00:00:00 2001 From: Manish Dewangan Date: Sat, 19 Aug 2017 16:54:06 +0530 Subject: ASoC: msm: qdsp6v2: Do not reset backend pass-through mode Backend pass-through mode is reset to LEGACY_PCM during routing close. This results in opening of adm with non pass-through path type for pass-through session if device switch is done during pass-through session. Fix this by avoid resetting Backend passthrough-mode. Change-Id: Ia1ea10ce68ba04a72416827661b6fdc2abd48b3c Signed-off-by: Manish Dewangan --- sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c index 9e2e8b37ec10..7326e658c947 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c @@ -15659,10 +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; -- cgit v1.2.3