December 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 0/3] ASoC: intel: atom: Add FW version information
by Sebastien Guiriec 05 Dec '16

05 Dec '16

This patch series is adding FW version information of SST. Version is both log as dev_info and sysfs entry. Update since V1: - sysfs entry is set to "FW not yet loaded" in case we do not have loaded the FW once Sebastien Guiriec (3): ASoC: intel: atom: Add debug information related to FW version ASoC: Intel: atom: save FW version ASoC: Intel: atom: Add sysfs entry in order to store FW version sound/soc/intel/atom/sst/sst.c | 39 ++++++++++++++++++++++++++++++++++++++ sound/soc/intel/atom/sst/sst.h | 1 + sound/soc/intel/atom/sst/sst_ipc.c | 10 +++++++++- 3 files changed, 49 insertions(+), 1 deletion(-) -- 2.9.3 --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

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[alsa-devel] Applied "ASoC: Intel: atom: save FW version" to the asoc tree
by Mark Brown 05 Dec '16

05 Dec '16

The patch ASoC: Intel: atom: save FW version has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c7da1e9d2641f67e8b20703118ac02fe79604f10 Mon Sep 17 00:00:00 2001 From: Sebastien Guiriec <sebastien.guiriec(a)intel.com> Date: Mon, 5 Dec 2016 17:45:01 +0100 Subject: [PATCH] ASoC: Intel: atom: save FW version After the boot of the SST FW the firmware version is send back to the driver. This patch is saving the FW version inside the driver. Signed-off-by: Sebastien Guiriec <sebastien.guiriec(a)intel.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst.h | 1 + sound/soc/intel/atom/sst/sst_ipc.c | 6 ++++++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h index 3f493862e98d..5c9a51cc77aa 100644 --- a/sound/soc/intel/atom/sst/sst.h +++ b/sound/soc/intel/atom/sst/sst.h @@ -436,6 +436,7 @@ struct intel_sst_drv { */ char firmware_name[FW_NAME_SIZE]; + struct snd_sst_fw_version fw_version; struct sst_fw_save *fw_save; }; diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 7934a0047384..374bb61c596d 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -242,6 +242,12 @@ static void process_fw_init(struct intel_sst_drv *sst_drv_ctx, dev_dbg(sst_drv_ctx->dev, "Build date %s Time %s\n", init->build_info.date, init->build_info.time); + /* Save FW version */ + sst_drv_ctx->fw_version.type = init->fw_version.type; + sst_drv_ctx->fw_version.major = init->fw_version.major; + sst_drv_ctx->fw_version.minor = init->fw_version.minor; + sst_drv_ctx->fw_version.build = init->fw_version.build; + ret: sst_wake_up_block(sst_drv_ctx, retval, FW_DWNL_ID, 0 , NULL, 0); } -- 2.10.2

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[alsa-devel] [PATCH v5 0/4] ASoC: intel: atom: Add FW version information
by Sebastien Guiriec 05 Dec '16

05 Dec '16

This patch series is adding FW version information of SST. Version is both log as dev_info and sysfs entry. Update since V1: - sysfs entry is set to "FW not yet loaded" in case we do not have loaded the FW once Update since V2: - Add Vinod ack. https://www.spinics.net/lists/alsa-devel/msg56521.html Update since V3: - Add sysfs entry description. Update since V4: - Add ack from vinod on documentation - Resend on top of v4.9-rc1. Sebastien Guiriec (4): ASoC: intel: atom: Add debug information related to FW version ASoC: Intel: atom: save FW version ASoC: Intel: atom: Add sysfs entry in order to store FW version Documentation: ABI: Add ASoC Intel SST Atom sysfs description Documentation/ABI/testing/sysfs-platform-sst-atom | 17 ++++++++++ sound/soc/intel/atom/sst/sst.c | 39 +++++++++++++++++++++++ sound/soc/intel/atom/sst/sst.h | 1 + sound/soc/intel/atom/sst/sst_ipc.c | 10 +++++- 4 files changed, 66 insertions(+), 1 deletion(-) create mode 100644 Documentation/ABI/testing/sysfs-platform-sst-atom -- 2.9.3 --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

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[alsa-devel] [PATCH v6 0/3] ASoC: intel: atom: Add FW version information
by Sebastien Guiriec 05 Dec '16

05 Dec '16

This patch series is adding FW version information of SST. Version is both log as dev_info and sysfs entry. Update since V1: - sysfs entry is set to "FW not yet loaded" in case we do not have loaded the FW once Update since V2: - Add Vinod ack. https://www.spinics.net/lists/alsa-devel/msg56521.html Update since V3: - Add sysfs entry description. Update since V4: - Add ack from vinod on documentation - Resend on top of v4.9-rc1. Update since V5: - Resend on top of topic/intel - Remove first patch as the first version was already applied inside topic/intel without getting "Applied" notifiaction. https://www.spinics.net/lists/alsa-devel/msg56122.html Sebastien Guiriec (3): ASoC: Intel: atom: save FW version ASoC: Intel: atom: Add sysfs entry in order to store FW version Documentation: ABI: Add ASoC Intel SST Atom sysfs description Documentation/ABI/testing/sysfs-platform-sst-atom | 17 ++++++++++ sound/soc/intel/atom/sst/sst.c | 39 +++++++++++++++++++++++ sound/soc/intel/atom/sst/sst.h | 1 + sound/soc/intel/atom/sst/sst_ipc.c | 6 ++++ 4 files changed, 63 insertions(+) create mode 100644 Documentation/ABI/testing/sysfs-platform-sst-atom -- 2.9.3 --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

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Re: [alsa-devel] [PATCH v3] ASoC: omap-mcbsp: Add PM QoS support for McBSP to prevent glitches
by Peter Ujfalusi 05 Dec '16

05 Dec '16

Hi, On 12/01/2016 03:19 AM, Matt Ranostay wrote: > We can get audio errors if hitting deeper idle states on omaps: > > [alsa.c:230] error: Fatal problem with alsa output, error -5. > [audio.c:614] error: Error in writing audio (Input/output error?)! > > This seems to happen with off mode idle enabled as power for the > whole SoC may get cut off between filling the McBSP fifo using DMA. > While active DMA blocks deeper idle states in hardware, McBSP > activity does not seem to do so. > > Basing the QoS latency on the FIFO size with the approximate delay > required being 2.3 milliseconds per buffer location. > > Note that this is different from snd_pcm_hw_constraint_step() as > that can configure things based on the buffer and period size. > However, that does not help to block idle between the fifo fills. > > Note that in theory some omaps are capable of playing audio while > hitting deeper idle states. If somebody needs that and gets it > working, we can set the latency requirements accordingly. > > Based on the original patch by Tony Lindgren > Link: https://patchwork.kernel.org/patch/9305867/ > > Cc: Tony Lindgren <tony(a)atomide.com> > Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> > Signed-off-by: Matt Ranostay <matt(a)ranostay.consulting> > --- > Changes from v1: > * add calculations for latency per number of FIFO locations > > Changes from v2: > * add missing mcbsp.h header change > > sound/soc/omap/mcbsp.c | 15 +++++++++++++++ > sound/soc/omap/mcbsp.h | 2 ++ > 2 files changed, 17 insertions(+) > > diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c > index 06fec5699cc8..b4a773ca2bac 100644 > --- a/sound/soc/omap/mcbsp.c > +++ b/sound/soc/omap/mcbsp.c > @@ -25,6 +25,7 @@ > #include <linux/io.h> > #include <linux/slab.h> > #include <linux/pm_runtime.h> > +#include <linux/pm_qos.h> > > #include <linux/platform_data/asoc-ti-mcbsp.h> > > @@ -637,12 +638,21 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) > * Here we start the McBSP, by enabling transmitter, receiver or both. > * If no transmitter or receiver is active prior calling, then sample-rate > * generator and frame sync are started. > + * > + * Also setting of the QoS latency for the FIFO which varies upon the buffer > + * size. Approximately 2.3 milliseconds per FIFO location. > */ > void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx) > { > int enable_srg = 0; > + int latency = mcbsp->pdata->buffer_size * 23; I think this is not correct. The McBSP FIFO time depends on the sample rate and on the number of channels the audio is using. With 8KHz mono you have 12 times more time per FIFO element compared to 48KHz stereo. As it has been discussed with Tony we should calculate the QoS latency in hw_params: latency_ms = ((FIFOsize - FIFOthreshold) / channels) * 1000/sampling-rate On OMAP3.McBSP2 for example (44.1KHz, stereo): FIFO threshold 128 - DMA request will be triggered when 128 slots are free in the FIFO - at that point we have still 1152 words in the FIFO. - if the C wakeup latency is longer then what it takes to play out the samples from the FIFO (13.06ms), we will drain the FIFO and got underflow. - in this case the QOS should be set as 13.06ms FIFO threshold 1024 - DMA request will be triggered when 1024 slots are free in the FIFO - at that point we have still 256 words in the FIFO. - if the C wakeup latency is longer then what it takes to play out the samples from the FIFO (2.9ms), we will drain the FIFO and got underflow. - in this case the QOS should be 2.9ms On other McBSPs with 128 word FIFO the required latency is shorter to ensure we don't drain the FIFO. > u16 w; > > + /* Prevent omap hardware from hitting off between fifo fills */ > + if (latency) > + pm_qos_add_request(&mcbsp->pm_qos_req, > + PM_QOS_CPU_DMA_LATENCY, latency); > + > if (mcbsp->st_data) > omap_st_start(mcbsp); > > @@ -731,6 +741,8 @@ void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx) > > if (mcbsp->st_data) > omap_st_stop(mcbsp); > + > + pm_qos_remove_request(&mcbsp->pm_qos_req); > } > > int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) > @@ -1098,6 +1110,9 @@ int omap_mcbsp_init(struct platform_device *pdev) > > void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp) > { > + if (pm_qos_request_active(&mcbsp->pm_qos_req)) > + pm_qos_remove_request(&mcbsp->pm_qos_req); > + > if (mcbsp->pdata->buffer_size) > sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); > > diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h > index 61e93b1c185d..e603f33f4082 100644 > --- a/sound/soc/omap/mcbsp.h > +++ b/sound/soc/omap/mcbsp.h > @@ -325,6 +325,8 @@ struct omap_mcbsp { > unsigned int in_freq; > int clk_div; > int wlen; > + > + struct pm_qos_request pm_qos_req; > }; > > void omap_mcbsp_config(struct omap_mcbsp *mcbsp, > -- Péter

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[alsa-devel] [PATCH] ASoC: cs42l56: Fix misuse of regmap_update_bits
by Florian Vaussard 05 Dec '16

05 Dec '16

Using regmap_update_bits(..., mask, 1) with 'mask' following (1 << k) and k greater than 0 is wrong. Indeed, _regmap_update_bits will perform (mask & 1), which results in 0 if LSB of mask is 0. Thus the call regmap_update_bits(..., mask, 1) is in reality equivalent to regmap_update_bits(..., mask, 0). In such a case, the correct use is regmap_update_bits(..., mask, mask). This driver is performing such a mistake with the CS42L56_AIN*_REF_MASK masks, which equal 0x10, 0x20, 0x40 and 0x80. Fix the driver to make it consistent with the API. Please note that this change is untested, as I do not have this piece of hardware. Testers are welcome! Signed-off-by: Florian Vaussard <florian.vaussard(a)heig-vd.ch> --- sound/soc/codecs/cs42l56.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 3e2c046..cb6ca85 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -1277,19 +1277,23 @@ static int cs42l56_i2c_probe(struct i2c_client *i2c_client, if (cs42l56->pdata.ain1a_ref_cfg) regmap_update_bits(cs42l56->regmap, CS42L56_AIN_REFCFG_ADC_MUX, - CS42L56_AIN1A_REF_MASK, 1); + CS42L56_AIN1A_REF_MASK, + CS42L56_AIN1A_REF_MASK); if (cs42l56->pdata.ain1b_ref_cfg) regmap_update_bits(cs42l56->regmap, CS42L56_AIN_REFCFG_ADC_MUX, - CS42L56_AIN1B_REF_MASK, 1); + CS42L56_AIN1B_REF_MASK, + CS42L56_AIN1B_REF_MASK); if (cs42l56->pdata.ain2a_ref_cfg) regmap_update_bits(cs42l56->regmap, CS42L56_AIN_REFCFG_ADC_MUX, - CS42L56_AIN2A_REF_MASK, 1); + CS42L56_AIN2A_REF_MASK, + CS42L56_AIN2A_REF_MASK); if (cs42l56->pdata.ain2b_ref_cfg) regmap_update_bits(cs42l56->regmap, CS42L56_AIN_REFCFG_ADC_MUX, - CS42L56_AIN2B_REF_MASK, 1); + CS42L56_AIN2B_REF_MASK, + CS42L56_AIN2B_REF_MASK); if (cs42l56->pdata.micbias_lvl) regmap_update_bits(cs42l56->regmap, CS42L56_GAIN_BIAS_CTL, -- 2.5.5

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[alsa-devel] [PATCH 1/2] ASoC: topology: Allow a widget to have multiple enum controls
by mengdong.lin＠linux.intel.com 05 Dec '16

05 Dec '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This patch can create multiple enumerated mixer controls for a widget. Previously topology kernel driver assumes a widget can have only one emumerated mixer control. We need to remove this restriction for Broxton. Its firmware modules (widgets) may need multiple enum controls based on the channel and MIC combination. No ABI change is needed. The ABI allows a widget to embed multiple controls. Reported-by: G Kranthi <gudishax.kranthikumar(a)intel.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 4dfdc65..63e1a50 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -486,21 +486,24 @@ static void remove_widget(struct snd_soc_component *comp, dobj->ops->widget_unload(comp, dobj); /* - * Dynamic Widgets either have 1 enum kcontrol or 1..N mixers. + * Dynamic Widgets either have 1..N enum kcontrols or mixers. * The enum may either have an array of values or strings. */ if (dobj->widget.kcontrol_enum) { /* enumerated widget mixer */ - struct soc_enum *se = - (struct soc_enum *)w->kcontrols[0]->private_value; + for (i = 0; i < w->num_kcontrols; i++) { + struct snd_kcontrol *kcontrol = w->kcontrols[i]; + struct soc_enum *se = + (struct soc_enum *)kcontrol->private_value; - snd_ctl_remove(card, w->kcontrols[0]); + snd_ctl_remove(card, kcontrol); - kfree(se->dobj.control.dvalues); - for (i = 0; i < se->items; i++) - kfree(se->dobj.control.dtexts[i]); + kfree(se->dobj.control.dvalues); + for (i = 0; i < se->items; i++) + kfree(se->dobj.control.dtexts[i]); - kfree(se); + kfree(se); + } kfree(w->kcontrol_news); } else { /* non enumerated widget mixer */ @@ -1256,98 +1259,105 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( } static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( - struct soc_tplg *tplg) + struct soc_tplg *tplg, int num_kcontrols) { struct snd_kcontrol_new *kc; struct snd_soc_tplg_enum_control *ec; struct soc_enum *se; - int i, err; - - ec = (struct snd_soc_tplg_enum_control *)tplg->pos; - tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + - ec->priv.size); - - /* validate kcontrol */ - if (strnlen(ec->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == - SNDRV_CTL_ELEM_ID_NAME_MAXLEN) - return NULL; + int i, j, err; - kc = kzalloc(sizeof(*kc), GFP_KERNEL); + kc = kcalloc(num_kcontrols, sizeof(*kc), GFP_KERNEL); if (kc == NULL) return NULL; - se = kzalloc(sizeof(*se), GFP_KERNEL); - if (se == NULL) - goto err; + for (i = 0; i < num_kcontrols; i++) { + ec = (struct snd_soc_tplg_enum_control *)tplg->pos; + /* validate kcontrol */ + if (strnlen(ec->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + return NULL; + + se = kzalloc(sizeof(*se), GFP_KERNEL); + if (se == NULL) + goto err; - dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", - ec->hdr.name); + dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", + ec->hdr.name); - kc->name = ec->hdr.name; - kc->private_value = (long)se; - kc->iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc->access = ec->hdr.access; + kc[i].name = ec->hdr.name; + kc[i].private_value = (long)se; + kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kc[i].access = ec->hdr.access; - /* we only support FL/FR channel mapping atm */ - se->reg = tplc_chan_get_reg(tplg, ec->channel, SNDRV_CHMAP_FL); - se->shift_l = tplc_chan_get_shift(tplg, ec->channel, SNDRV_CHMAP_FL); - se->shift_r = tplc_chan_get_shift(tplg, ec->channel, SNDRV_CHMAP_FR); + /* we only support FL/FR channel mapping atm */ + se->reg = tplc_chan_get_reg(tplg, ec->channel, SNDRV_CHMAP_FL); + se->shift_l = tplc_chan_get_shift(tplg, ec->channel, + SNDRV_CHMAP_FL); + se->shift_r = tplc_chan_get_shift(tplg, ec->channel, + SNDRV_CHMAP_FR); - se->items = ec->items; - se->mask = ec->mask; - se->dobj.index = tplg->index; + se->items = ec->items; + se->mask = ec->mask; + se->dobj.index = tplg->index; - switch (ec->hdr.ops.info) { - case SND_SOC_TPLG_CTL_ENUM_VALUE: - case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: - err = soc_tplg_denum_create_values(se, ec); - if (err < 0) { - dev_err(tplg->dev, "ASoC: could not create values for %s\n", - ec->hdr.name); + switch (ec->hdr.ops.info) { + case SND_SOC_TPLG_CTL_ENUM_VALUE: + case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: + err = soc_tplg_denum_create_values(se, ec); + if (err < 0) { + dev_err(tplg->dev, "ASoC: could not create values for %s\n", + ec->hdr.name); + goto err_se; + } + /* fall through to create texts */ + case SND_SOC_TPLG_CTL_ENUM: + case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: + case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: + err = soc_tplg_denum_create_texts(se, ec); + if (err < 0) { + dev_err(tplg->dev, "ASoC: could not create texts for %s\n", + ec->hdr.name); + goto err_se; + } + break; + default: + dev_err(tplg->dev, "ASoC: invalid enum control type %d for %s\n", + ec->hdr.ops.info, ec->hdr.name); goto err_se; } - /* fall through to create texts */ - case SND_SOC_TPLG_CTL_ENUM: - case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: - case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: - err = soc_tplg_denum_create_texts(se, ec); + + /* map io handlers */ + err = soc_tplg_kcontrol_bind_io(&ec->hdr, &kc[i], tplg); + if (err) { + soc_control_err(tplg, &ec->hdr, ec->hdr.name); + goto err_se; + } + + /* pass control to driver for optional further init */ + err = soc_tplg_init_kcontrol(tplg, &kc[i], + (struct snd_soc_tplg_ctl_hdr *)ec); if (err < 0) { - dev_err(tplg->dev, "ASoC: could not create texts for %s\n", + dev_err(tplg->dev, "ASoC: failed to init %s\n", ec->hdr.name); goto err_se; } - break; - default: - dev_err(tplg->dev, "ASoC: invalid enum control type %d for %s\n", - ec->hdr.ops.info, ec->hdr.name); - goto err_se; - } - /* map io handlers */ - err = soc_tplg_kcontrol_bind_io(&ec->hdr, kc, tplg); - if (err) { - soc_control_err(tplg, &ec->hdr, ec->hdr.name); - goto err_se; - } - - /* pass control to driver for optional further init */ - err = soc_tplg_init_kcontrol(tplg, kc, - (struct snd_soc_tplg_ctl_hdr *)ec); - if (err < 0) { - dev_err(tplg->dev, "ASoC: failed to init %s\n", - ec->hdr.name); - goto err_se; + tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + + ec->priv.size); } return kc; err_se: - /* free values and texts */ - kfree(se->dobj.control.dvalues); - for (i = 0; i < ec->items; i++) - kfree(se->dobj.control.dtexts[i]); + for (; i >= 0; i--) { + /* free values and texts */ + se = (struct soc_enum *)kc[i].private_value; + kfree(se->dobj.control.dvalues); + for (j = 0; j < ec->items; j++) + kfree(se->dobj.control.dtexts[j]); - kfree(se); + kfree(se); + } err: kfree(kc); @@ -1499,9 +1509,10 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: template.dobj.widget.kcontrol_enum = 1; - template.num_kcontrols = 1; + template.num_kcontrols = w->num_kcontrols; template.kcontrol_news = - soc_tplg_dapm_widget_denum_create(tplg); + soc_tplg_dapm_widget_denum_create(tplg, + template.num_kcontrols); if (!template.kcontrol_news) { ret = -ENOMEM; goto hdr_err; -- 2.5.0

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[alsa-devel] [PATCH 2/2] ASoC: topology: Only free TLV for volume mixers of a widget
by mengdong.lin＠linux.intel.com 05 Dec '16

05 Dec '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This patch will check the type of embedded controls for a widget, and only free the TLV of volume mixers. Bytes controls don't have TLV. Just free the private value which is used as struct soc_mixer_control for volume mixers or soc_bytes_ext for bytes controls. No need to cast to these types before freeing it. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index d318fe4..86c884a 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -53,7 +53,7 @@ struct snd_soc_dobj_control { /* dynamic widget object */ struct snd_soc_dobj_widget { - unsigned int kcontrol_enum:1; /* this widget is an enum kcontrol */ + unsigned int kcontrol_type; /* kcontrol type: mixer, enum, bytes */ }; /* generic dynamic object - all dynamic objects belong to this struct */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 63e1a50..11feb19 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -489,7 +489,7 @@ static void remove_widget(struct snd_soc_component *comp, * Dynamic Widgets either have 1..N enum kcontrols or mixers. * The enum may either have an array of values or strings. */ - if (dobj->widget.kcontrol_enum) { + if (dobj->widget.kcontrol_type == SND_SOC_TPLG_TYPE_ENUM) { /* enumerated widget mixer */ for (i = 0; i < w->num_kcontrols; i++) { struct snd_kcontrol *kcontrol = w->kcontrols[i]; @@ -506,16 +506,21 @@ static void remove_widget(struct snd_soc_component *comp, } kfree(w->kcontrol_news); } else { - /* non enumerated widget mixer */ + /* volume mixer or bytes controls */ for (i = 0; i < w->num_kcontrols; i++) { struct snd_kcontrol *kcontrol = w->kcontrols[i]; - struct soc_mixer_control *sm = - (struct soc_mixer_control *) kcontrol->private_value; - kfree(w->kcontrols[i]->tlv.p); + if (dobj->widget.kcontrol_type + == SND_SOC_TPLG_TYPE_MIXER) + kfree(kcontrol->tlv.p); - snd_ctl_remove(card, w->kcontrols[i]); - kfree(sm); + snd_ctl_remove(card, kcontrol); + + /* Private value is used as struct soc_mixer_control + * for volume mixers or soc_bytes_ext for bytes + * controls. + */ + kfree((void *)kcontrol->private_value); } kfree(w->kcontrol_news); } @@ -1439,6 +1444,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, struct snd_soc_dapm_widget template, *widget; struct snd_soc_tplg_ctl_hdr *control_hdr; struct snd_soc_card *card = tplg->comp->card; + unsigned int kcontrol_type; int ret = 0; if (strnlen(w->name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == @@ -1494,6 +1500,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, case SND_SOC_TPLG_CTL_VOLSW_XR_SX: case SND_SOC_TPLG_CTL_RANGE: case SND_SOC_TPLG_DAPM_CTL_VOLSW: + kcontrol_type = SND_SOC_TPLG_TYPE_MIXER; /* volume mixer */ template.num_kcontrols = w->num_kcontrols; template.kcontrol_news = soc_tplg_dapm_widget_dmixer_create(tplg, @@ -1508,7 +1515,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: - template.dobj.widget.kcontrol_enum = 1; + kcontrol_type = SND_SOC_TPLG_TYPE_ENUM; /* enumerated mixer */ template.num_kcontrols = w->num_kcontrols; template.kcontrol_news = soc_tplg_dapm_widget_denum_create(tplg, @@ -1519,6 +1526,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, } break; case SND_SOC_TPLG_CTL_BYTES: + kcontrol_type = SND_SOC_TPLG_TYPE_BYTES; /* bytes control */ template.num_kcontrols = w->num_kcontrols; template.kcontrol_news = soc_tplg_dapm_widget_dbytes_create(tplg, @@ -1555,6 +1563,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, } widget->dobj.type = SND_SOC_DOBJ_WIDGET; + widget->dobj.widget.kcontrol_type = kcontrol_type; widget->dobj.ops = tplg->ops; widget->dobj.index = tplg->index; kfree(template.sname); -- 2.5.0

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Re: [alsa-devel] ProjectMix IO Feedback
by Takashi Sakamoto 05 Dec '16

05 Dec '16

\Hi Darren, (C.C.ed to alsa-devel and ffado-user) On Dec 1 2016 19:47, Darren Anderson wrote: > Hi Takashi, > > Just wanted to give you some feedback on how the ProjectMix IO is working > with your changes that were merged into the Linux Kernel. > > Out of the box, on Ubuntu with kernel version 4.4.0-31, on boot, the > interface makes its distinctive popping sound as it sets the clock rate, > showing that the driver is loading fine. It is also now detected by > PulseAudio (without using a jack sink, as I previously had to). I'm then > able to use ffado-mixer to control the device. Thanks for your feedback to ALSA bebob driver. But it's better to post this kind of report to public mailing list, instead of personal address. There might be people who need the information. > There are, however, three issues. > > Firstly, the sample rate seems to be set every time an audio stream is > about to begin. This can be quite irritating at times, due to the > aforementioned distinctive popping sound and lag in the starting of audio > streams. Would it be possible to set the sample rate once at boot time, and > leave it at this unless the user specifically instructs otherwise? Applications can set sampling rate as they want, by ALSA PCM interface. I think it better to keep it what it is unless there's few special cases. For example, ALSA dice driver performs what you expected, due to hardware restriction. Application can't set sampling rate by ALSA PCM interface and they require the other application for this purpose. > Secondly, the main audio seems to be routed from Pulse through the "Aux > 1/2" channel, as opposed to "Mixer 1/2". The device will output no audio > until the "Analog 1/2" output is set to use "Aux 1/2" in FFADO mixer. > However in fixing this issue, a separate issue may be created. As most > distributions play some sort of "start-up sound", and the device seems to > initialize to full volume unless instructed otherwise, the fact that the > output channel needs to be changed may actually save some ear drums. It may > be wise to ask the device to reduce to something reasonable like 25% volume > when the driver is loaded. Or perhaps just mute all outputs by default. Just after booting up, your device has no routing configuration, therefore it generates no sound till configured with use space applications. Default settings of gain/volume are 0dB, perhaps. But this can also be configured from user space applications. What I can do is to produce a way to control devices for applications in user land, and actual way to use the device is left to application developers. If users have something to do, they would do it. > And finally, I've have found that the quality of the audio output from the > device is of a somewhat lower fidelity than the equivalent output on > Windows. I'm currently testing the interface using my KRK Rokit monitor > speakers and finding that the output is lacking in low frequencies. It > sounds as if there is some high pass filtering going on. ALSA bebob driver allows applications to use the device with both of 16 bit sample and 24bit sample with adjusted 8 bit, for playbacking. I guess that your trial is to playback 16 bit PCM sample and looks with highpass filter. Please check the property of PCM substream. > I hope that you find this feedback useful to your ongoing efforts. You're > doing great work. > > If I can do anything else to help, don't hesitate to ask. Cheers Takashi Sakamoto

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[alsa-devel] [PATCH] ASoC: rsnd: rsnd_get_dalign() needs to care SSIU, not SSI
by Kuninori Morimoto 05 Dec '16

05 Dec '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> SSIU was controlled by SSI before, but commit c7f69ab53("ASoC: rsnd: use mod base common method on SSIU") separated it into ssiu.c But, it didn't care about rsnd_get_dalign() for judging SSI_BUSIF_DALIGN register value which changes the stream data order. This function will be called from cmd/src/ssiu now, but current code still cares ssi, not ssiu. This patch fix it up Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 6 +++--- sound/soc/sh/rcar/rsnd.h | 1 + 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index f181410..28ef9ab 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -306,7 +306,7 @@ u32 rsnd_get_adinr_bit(struct rsnd_mod *mod, struct rsnd_dai_stream *io) */ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { - struct rsnd_mod *ssi = rsnd_io_to_mod_ssi(io); + struct rsnd_mod *ssiu = rsnd_io_to_mod_ssiu(io); struct rsnd_mod *target; struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); u32 val = 0x76543210; @@ -315,11 +315,11 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) if (rsnd_io_is_play(io)) { struct rsnd_mod *src = rsnd_io_to_mod_src(io); - target = src ? src : ssi; + target = src ? src : ssiu; } else { struct rsnd_mod *cmd = rsnd_io_to_mod_cmd(io); - target = cmd ? cmd : ssi; + target = cmd ? cmd : ssiu; } mask <<= runtime->channels * 4; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index a8f61d7..8ca1705 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -382,6 +382,7 @@ struct rsnd_dai_stream { }; #define rsnd_io_to_mod(io, i) ((i) < RSND_MOD_MAX ? (io)->mod[(i)] : NULL) #define rsnd_io_to_mod_ssi(io) rsnd_io_to_mod((io), RSND_MOD_SSI) +#define rsnd_io_to_mod_ssiu(io) rsnd_io_to_mod((io), RSND_MOD_SSIU) #define rsnd_io_to_mod_ssip(io) rsnd_io_to_mod((io), RSND_MOD_SSIP) #define rsnd_io_to_mod_src(io) rsnd_io_to_mod((io), RSND_MOD_SRC) #define rsnd_io_to_mod_ctu(io) rsnd_io_to_mod((io), RSND_MOD_CTU) -- 2.7.4

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