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

[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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[alsa-devel] [PATCH 0/2] ASoC: cq93vc: enable compiling on COMPILE_TEST
by Kuninori Morimoto 05 Dec '16

05 Dec '16

Hi Lee, Mark Current cq93vc sound codec indicates that it has MFD_DAVINCI_VOICECODEC dependency, but it doesn't. This means we can compile it from non DAVINCE environment by COMPILE_TEST. These patches solve this issue. I tested this on ARCH=x86 + make allyesconfig, and ARCH=ARM + shmobile_defconfig + COMPILE_TEST. Kuninori Morimoto (2): mfd: davinci_voicecodec: tidyup header difinitions ASoC: cq93vc: remove MFD_DAVINCI_VOICECODEC dependency for CQ0093VC drivers/mfd/davinci_voicecodec.c | 1 + include/linux/mfd/davinci_voicecodec.h | 4 ---- sound/soc/codecs/Kconfig | 2 +- 3 files changed, 2 insertions(+), 5 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: simple_card_utils: tidyup file comment/define
by Kuninori Morimoto 05 Dec '16

05 Dec '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> simple_card_utils was created as simple_card_core in 1st prototype, and current code still have it. Let's tidyup Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/simple_card_utils.h | 8 ++++---- sound/soc/generic/simple-card-utils.c | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index fd641255..64e90ca 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -1,5 +1,5 @@ /* - * simple_card_core.h + * simple_card_utils.h * * Copyright (c) 2016 Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> * @@ -7,8 +7,8 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ -#ifndef __SIMPLE_CARD_CORE_H -#define __SIMPLE_CARD_CORE_H +#ifndef __SIMPLE_CARD_UTILS_H +#define __SIMPLE_CARD_UTILS_H #include <sound/soc.h> @@ -68,4 +68,4 @@ void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, int asoc_simple_card_clean_reference(struct snd_soc_card *card); -#endif /* __SIMPLE_CARD_CORE_H */ +#endif /* __SIMPLE_CARD_UTILS_H */ diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index e5b80f5..cf02625 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -1,5 +1,5 @@ /* - * simple-card-core.c + * simple-card-utils.c * * Copyright (c) 2016 Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> * -- 2.7.4

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[alsa-devel] [PATCH] ASoC: Intel: Add pre & post suspend calls in bxt machine
by Vinod Koul 05 Dec '16

05 Dec '16

We need to disable jack detection before entering suspend and enable it after resume. This was essential as headphone jack widget will always remain ON if a jack is inserted and this results in no audio after system resumes from a suspend. Signed-off-by: Senthilnathan Veppur <senthilnathanx.veppur(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/boards/bxt_rt298.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index 1309405b3808..b272b72e6a77 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -432,6 +432,33 @@ static int bxt_fe_startup(struct snd_pcm_substream *substream) }, }; +static int bxt_suspend_pre(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + if (!strcmp(codec->component.name, "i2c-INT343A:00")) { + rt298_mic_detect(codec, NULL); + break; + } + } + return 0; +} + +static int bxt_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + if (!strcmp(codec->component.name, "i2c-INT343A:00")) { + rt298_mic_detect(codec, &broxton_headset); + break; + } + } + return 0; +} + + /* broxton audio machine driver for SPT + RT298S */ static struct snd_soc_card broxton_rt298 = { .name = "broxton-rt298", @@ -445,6 +472,9 @@ static int bxt_fe_startup(struct snd_pcm_substream *substream) .dapm_routes = broxton_rt298_map, .num_dapm_routes = ARRAY_SIZE(broxton_rt298_map), .fully_routed = true, + .suspend_pre = bxt_suspend_pre, + .resume_post = bxt_resume_post, + }; static int broxton_audio_probe(struct platform_device *pdev) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: cq93vc: enable compiling on COMPILE_TEST
by Kuninori Morimoto 05 Dec '16

05 Dec '16

Hi Mark I posted cq93vc dependency fixup (= no depend to MFD_DAVINCI_VOICECODEC) patch which were 2 patches. [1/2] was tidyup patch for mfd side, and [2/2] was tidyup patch for ASoC Kconfig. [1/2] was accepted by Lee, but [2/2] wasn't. This is for it. This patch is based on Lee's git://git.kernel.org/pub/scm/linux/kernel/git/lee/mfd.git :: for-mfd-next Especially this commit 8ca9edc837932469b81b8b47ea43a074b6add970 ("mfd: davinci_voicecodec: Tidyup header difinitions") Kuninori Morimoto (1): ASoC: cq93vc: remove MFD_DAVINCI_VOICECODEC dependency for CQ0093VC drivers/mfd/davinci_voicecodec.c | 1 + include/linux/mfd/davinci_voicecodec.h | 4 ---- sound/soc/codecs/Kconfig | 2 +- 3 files changed, 2 insertions(+), 5 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/3 v1] usb-misc fix
by Jiada Wang 05 Dec '16

05 Dec '16

This patch set contains the following patches Andreas Pape (1): ALSA: usb-audio: more tolerant packetsize Daniel Girnus (1): ALSA: usb-audio: avoid setting of sample rate multiple times on bus Mark Craske (1): ALSA: usb-audio: fix race in snd_usb_endpoint_stop sound/usb/endpoint.c | 24 ++++++++++++++++-------- sound/usb/pcm.c | 21 +++++++++++---------- 2 files changed, 27 insertions(+), 18 deletions(-) -- 2.9.3

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[alsa-devel] [RFC PATCH v4 7/7] ALSA: x86: hdmi: continue playback even when display resolution changes
by Jerome Anand 02 Dec '16

02 Dec '16

When the display resolution changes, the drm disables the display pipes due to which audio rendering stops. At this time, we need to ensure the existing audio pointers and buffers are cleared out so that the playback can restarted once the display pipe is enabled with a different N/CTS values Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Jerome Anand <jerome.anand(a)intel.com> --- sound/x86/intel_hdmi_audio.c | 21 ++++++++++++++++++--- 1 file changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 9249521..d6fd638 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -43,6 +43,7 @@ static DEFINE_MUTEX(had_mutex); static int hdmi_card_index = SNDRV_DEFAULT_IDX1; static char *hdmi_card_id = SNDRV_DEFAULT_STR1; static struct snd_intelhad *had_data; +static int underrun_count; module_param(hdmi_card_index, int, 0444); MODULE_PARM_DESC(hdmi_card_index, @@ -1114,6 +1115,7 @@ static int snd_intelhad_open(struct snd_pcm_substream *substream) intelhaddata = snd_pcm_substream_chip(substream); had_stream = intelhaddata->private_data; runtime = substream->runtime; + underrun_count = 0; pm_runtime_get(intelhaddata->dev); @@ -1506,10 +1508,23 @@ static snd_pcm_uframes_t snd_intelhad_pcm_pointer( buf_id = intelhaddata->curr_buf % 4; had_read_register(AUD_BUF_A_LENGTH + (buf_id * HAD_REG_WIDTH), &t); - if (t == 0) { - pr_debug("discovered buffer done for buf %d\n", buf_id); - /* had_process_buffer_done(intelhaddata); */ + + if ((t == 0) || (t == ((u32)-1L))) { + underrun_count++; + pr_debug("discovered buffer done for buf %d, count = %d\n", + buf_id, underrun_count); + + if (underrun_count > (HAD_MIN_PERIODS/2)) { + pr_debug("assume audio_codec_reset, underrun = %d - do xrun\n", + underrun_count); + underrun_count = 0; + return SNDRV_PCM_POS_XRUN; + } + } else { + /* Reset Counter */ + underrun_count = 0; } + t = intelhaddata->buf_info[buf_id].buf_size - t; if (intelhaddata->stream_info.buffer_rendered) -- 2.9.3

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[alsa-devel] [RFC PATCH v4 6/7] ALSA: x86: hdmi: Fixup some monitor
by Jerome Anand 02 Dec '16

02 Dec '16

This change was given to Canonical apparently to fix an issue with on some monitor brand. It's not clear what this patch does but it doesn't seem to have side effects. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Jerome Anand <jerome.anand(a)intel.com> --- sound/x86/intel_hdmi_audio.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index d9ce750..9249521 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -337,6 +337,7 @@ static void snd_intelhad_reset_audio_v2(u8 reset) static int had_prog_status_reg(struct snd_pcm_substream *substream, struct snd_intelhad *intelhaddata) { + union aud_cfg cfg_val = {.cfg_regval = 0}; union aud_ch_status_0 ch_stat0 = {.status_0_regval = 0}; union aud_ch_status_1 ch_stat1 = {.status_1_regval = 0}; int format; @@ -347,6 +348,7 @@ static int had_prog_status_reg(struct snd_pcm_substream *substream, IEC958_AES0_NONAUDIO)>>1; ch_stat0.status_0_regx.clk_acc = (intelhaddata->aes_bits & IEC958_AES3_CON_CLOCK)>>4; + cfg_val.cfg_regx.val_bit = ch_stat0.status_0_regx.lpcm_id; switch (substream->runtime->rate) { case AUD_SAMPLE_RATE_32: @@ -426,7 +428,6 @@ int snd_intelhad_prog_audio_ctrl_v2(struct snd_pcm_substream *substream, else cfg_val.cfg_regx_v2.layout = LAYOUT1; - cfg_val.cfg_regx_v2.val_bit = 1; had_write_register(AUD_CONFIG, cfg_val.cfg_regval); return 0; } @@ -482,7 +483,6 @@ int snd_intelhad_prog_audio_ctrl_v1(struct snd_pcm_substream *substream, } - cfg_val.cfg_regx.val_bit = 1; had_write_register(AUD_CONFIG, cfg_val.cfg_regval); return 0; } -- 2.9.3

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