March 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC:pcm512x: Fix divide by zero issue.
by Howard Mitchell 22 Mar '15

22 Mar '15

If den=1 and pllin_rate>20MHz then den and num are adjusted to 0 causing a divide by zero error a few lines further on. Therefore this patch correctly scales num and den such that pllin_rate/den < 20MHz as required in the device data sheet. Signed-off-by: Howard Mitchell <hm(a)hmbedded.co.uk> --- sound/soc/codecs/pcm512x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 5301c4a..8472099 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -713,8 +713,8 @@ static int pcm512x_find_pll_coeff(struct snd_soc_dai *dai, /* pllin_rate / P (or here, den) cannot be greater than 20 MHz */ if (pllin_rate / den > 20000000 && num < 8) { - num *= 20000000 / (pllin_rate / den); - den *= 20000000 / (pllin_rate / den); + num *= DIV_ROUND_UP(pllin_rate / den, 20000000); + den *= DIV_ROUND_UP(pllin_rate / den, 20000000); } dev_dbg(dev, "num / den = %lu / %lu\n", num, den); -- 1.7.9.5

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Re: [alsa-devel] dummy codec and the dummy platform use the same name: snd-soc-dummy
by Mark Brown 22 Mar '15

22 Mar '15

On Fri, Mar 20, 2015 at 11:27:35PM -0700, Zhao Ye wrote: > Then in > static void soc_init_codec_debugfs(struct snd_soc_codec *codec) > and, > static void soc_init_platform_debugfs(struct snd_soc_platform *platform) > this two place will make two debugfs_create_dir() with the " snd-soc-dummy" name twice. > So I think we should change one of the names. Well, upstream what I'd expect to be happening here is that we merge everything into a single dummy component which avoids the problem another way.

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Re: [alsa-devel] [PATCH] ASoC: wm8741: Fix rates constraints values
by Charles Keepax 22 Mar '15

22 Mar '15

On Fri, Mar 13, 2015 at 10:20:14PM +0100, Sergej Sawazki wrote: > According to table 11 in the datasheet (Rev 4.3): > Example: With a master clock (MCLK) of 11.2896MHz, the following sampling rates (LRCLK) are supported: > 44.1kHz (MCLK/256), 88.2kHz (MCLK/128) > > For the rates lists, we should use 82000 instead of 88235 and 176400 instead of 1764000 (seems to be a typo). > > Signed-off-by: Sergej Sawazki <ce3a(a)gmx.de> > --- Acked-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Thanks, Charles

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[alsa-devel] [PATCH 1/2] ALSA:ControlName.txt: Add 'Analogue' as a prefix for volume controls
by Howard Mitchell 21 Mar '15

21 Mar '15

Currently 'Playback Volume' is the correct way to express an analogue volume control. However, this control name has initialisation defaults applied when using 'alsactl restore' and in some cases this is not appropriate. An example would be a control that has a selection of 0db and -6dB of gain that is intended to set the fullscale ouput voltage of a DAC. The TI pcm512x family of DAcs have such a control. In this case the device/driver reset defaults are preferred. Signed-off-by: Howard Mitchell <hm(a)hmbedded.co.uk> --- Documentation/sound/alsa/ControlNames.txt | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/alsa/ControlNames.txt b/Documentation/sound/alsa/ControlNames.txt index 79a6127..3fc1cf50 100644 --- a/Documentation/sound/alsa/ControlNames.txt +++ b/Documentation/sound/alsa/ControlNames.txt @@ -71,11 +71,11 @@ SOURCE: HDMI/DP (either HDMI or DisplayPort) Exceptions (deprecated): - [Digital] Capture Source - [Digital] Capture Switch (aka input gain switch) - [Digital] Capture Volume (aka input gain volume) - [Digital] Playback Switch (aka output gain switch) - [Digital] Playback Volume (aka output gain volume) + [Analogue|Digital] Capture Source + [Analogue|Digital] Capture Switch (aka input gain switch) + [Analogue|Digital] Capture Volume (aka input gain volume) + [Analogue|Digital] Playback Switch (aka output gain switch) + [Analogue|Digital] Playback Volume (aka output gain volume) Tone Control - Switch Tone Control - Bass Tone Control - Treble -- 1.7.9.5

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[alsa-devel] [PATCH 0/4] More aggressive PM for HD-audio
by Takashi Iwai 21 Mar '15

21 Mar '15

Hi, here is a patchset for supporting more aggressive PM for HD-audio. This allows to change the power state of each widget more dynamically with jack and stream states. It's activated only when the codec driver (or via sysfs or f/w patch) sets codec->power_mgmt flag. In theory, this should work for the recent Realtek codecs, but currently I have no machine for test. David, could you or your team check whether this works for ALC282 or such? Just add like: --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5415,6 +5415,7 @@ static int patch_alc269(struct hda_codec *codec) spec = codec->spec; spec->gen.shared_mic_vref_pin = 0x18; + codec->power_mgmt = 1; snd_hda_pick_fixup(codec, alc269_fixup_models, alc269_fixup_tbl, alc269_fixups); The patchset is for for-next branch of sound git tree, but they might be applicable to 4.0-rc (or even older), too. The current patches are found in topic/hda-power branch. thanks, Takashi === Takashi Iwai (4): ALSA: hda - Simplify PCM setup overrides ALSA: hda - Support advanced power state controls ALSA: hda - Use the new power control for VIA codecs ALSA: hda - Adjust power of beep widget and outputs sound/pci/hda/hda_beep.c | 29 +- sound/pci/hda/hda_beep.h | 1 + sound/pci/hda/hda_codec.c | 4 + sound/pci/hda/hda_codec.h | 2 + sound/pci/hda/hda_generic.c | 480 ++++++++++++++++++++++++------ sound/pci/hda/hda_generic.h | 5 +- sound/pci/hda/patch_realtek.c | 41 --- sound/pci/hda/patch_sigmatel.c | 5 + sound/pci/hda/patch_via.c | 662 +---------------------------------------- 9 files changed, 427 insertions(+), 802 deletions(-) -- 2.3.3

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[alsa-devel] ASoC: Max98090 Codec DAPM Widget setup in Open Embedded
by C. Omer Rafique 20 Mar '15

20 Mar '15

I would like to know which file I can use to connect/enable DPAM widgets permanently so a simple aplay/record can work? My environment is: - Yocto Linux Kernel 3.10 - (DTS based) - Max98090 Codec - imx6SL SoC I thought about doing this in alsa.conf or asound.conf but I couldn't find a single example with DAPM widgets . Thank you Choudhary

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[alsa-devel] [PATCH] ALSA: hda - Rename power_mgmt flag with power_save_node
by Takashi Iwai 20 Mar '15

20 Mar '15

David suggested that the name "power_mgmt" is too ambiguous. Rename the flag with a bit clearer one "power_save_node". Also, add the corresponding description to HD-Audio.txt, too. Reported-by: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa/HD-Audio.txt | 6 +++++- sound/pci/hda/hda_codec.h | 2 +- sound/pci/hda/hda_generic.c | 30 +++++++++++++++--------------- sound/pci/hda/patch_sigmatel.c | 8 ++++---- sound/pci/hda/patch_via.c | 12 ++++++------ 5 files changed, 31 insertions(+), 27 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 42a0a39b77e6..e7193aac669c 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -466,7 +466,11 @@ The generic parser supports the following hints: - add_jack_modes (bool): add "xxx Jack Mode" enum controls to each I/O jack for allowing to change the headphone amp and mic bias VREF capabilities -- power_down_unused (bool): power down the unused widgets +- power_save_node (bool): advanced power management for each widget, + controlling the power sate (D0/D3) of each widget node depending on + the actual pin and stream states +- power_down_unused (bool): power down the unused widgets, a subset of + power_save_node, and will be dropped in future - add_hp_mic (bool): add the headphone to capture source if possible - hp_mic_detect (bool): enable/disable the hp/mic shared input for a single built-in mic case; default true diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 148e84ce61cf..ccf355d4a8fa 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -371,7 +371,7 @@ struct hda_codec { unsigned int cached_write:1; /* write only to caches */ unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ - unsigned int power_mgmt:1; /* advanced PM for each widget */ + unsigned int power_save_node:1; /* advanced PM for each widget */ #ifdef CONFIG_PM unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ atomic_t in_pm; /* suspend/resume being performed */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 94c7f43ad182..de4024db27b4 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -140,9 +140,9 @@ static void parse_user_hints(struct hda_codec *codec) val = snd_hda_get_bool_hint(codec, "single_adc_amp"); if (val >= 0) codec->single_adc_amp = !!val; - val = snd_hda_get_bool_hint(codec, "power_mgmt"); + val = snd_hda_get_bool_hint(codec, "power_save_node"); if (val >= 0) - codec->power_mgmt = !!val; + codec->power_save_node = !!val; val = snd_hda_get_bool_hint(codec, "auto_mute"); if (val >= 0) @@ -661,7 +661,7 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, struct nid_path *path = snd_array_elem(&spec->paths, n); if (!path->active) continue; - if (codec->power_mgmt) { + if (codec->power_save_node) { if (!path->stream_enabled && !path->pin_fixed) continue; /* ignore unplugged paths except for DAC/ADC */ @@ -879,8 +879,8 @@ void snd_hda_activate_path(struct hda_codec *codec, struct nid_path *path, path->active = false; /* make sure the widget is powered up */ - if (enable && (spec->power_down_unused || codec->power_mgmt)) - path_power_update(codec, path, codec->power_mgmt); + if (enable && (spec->power_down_unused || codec->power_save_node)) + path_power_update(codec, path, codec->power_save_node); for (i = path->depth - 1; i >= 0; i--) { hda_nid_t nid = path->path[i]; @@ -905,7 +905,7 @@ static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path) { struct hda_gen_spec *spec = codec->spec; - if (!(spec->power_down_unused || codec->power_mgmt) || path->active) + if (!(spec->power_down_unused || codec->power_save_node) || path->active) return; sync_power_state_change(codec, path_power_update(codec, path, true)); } @@ -3981,7 +3981,7 @@ static hda_nid_t set_pin_power_jack(struct hda_codec *codec, hda_nid_t pin, { bool on; - if (!codec->power_mgmt) + if (!codec->power_save_node) return 0; on = snd_hda_jack_detect_state(codec, pin) != HDA_JACK_NOT_PRESENT; @@ -4038,7 +4038,7 @@ static void add_all_pin_power_ctls(struct hda_codec *codec, bool on) struct auto_pin_cfg *cfg = &spec->autocfg; int i; - if (!codec->power_mgmt) + if (!codec->power_save_node) return; add_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins, on); if (cfg->line_out_type != AUTO_PIN_HP_OUT) @@ -4067,7 +4067,7 @@ static void sync_all_pin_power_ctls(struct hda_codec *codec) struct auto_pin_cfg *cfg = &spec->autocfg; int i; - if (!codec->power_mgmt) + if (!codec->power_save_node) return; sync_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins); if (cfg->line_out_type != AUTO_PIN_HP_OUT) @@ -4111,7 +4111,7 @@ static int add_fake_beep_paths(struct hda_codec *codec) hda_nid_t nid = spec->beep_nid; int err; - if (!codec->power_mgmt || !nid) + if (!codec->power_save_node || !nid) return 0; err = add_fake_paths(codec, nid, cfg->line_outs, cfg->line_out_pins); if (err < 0) @@ -4232,7 +4232,7 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, } set_pin_eapd(codec, nid, !mute); - if (codec->power_mgmt) { + if (codec->power_save_node) { bool on = !mute; if (on) on = snd_hda_jack_detect_state(codec, nid) @@ -4740,11 +4740,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) * @nid: audio widget * @on: power on/off flag * - * Set this in patch_ops.stream_pm. Only valid with power_mgmt flag. + * Set this in patch_ops.stream_pm. Only valid with power_save_node flag. */ void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on) { - if (codec->power_mgmt) + if (codec->power_save_node) set_path_power(codec, nid, -1, on); } EXPORT_SYMBOL_GPL(snd_hda_gen_stream_pm); @@ -4915,14 +4915,14 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, dig_only: parse_digital(codec); - if (spec->power_down_unused || codec->power_mgmt) + if (spec->power_down_unused || codec->power_save_node) codec->power_filter = snd_hda_gen_path_power_filter; if (!spec->no_analog && spec->beep_nid) { err = snd_hda_attach_beep_device(codec, spec->beep_nid); if (err < 0) return err; - if (codec->beep && codec->power_mgmt) { + if (codec->beep && codec->power_save_node) { err = add_fake_beep_paths(codec); if (err < 0) return err; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7e531d5cde51..5b7c173adcb8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4494,7 +4494,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec) return err; spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.mixer_nid = 0x1d; spec->have_spdif_mux = 1; @@ -4600,7 +4600,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) codec->epss = 0; /* longer delay needed for D3 */ spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; @@ -4650,7 +4650,7 @@ static int patch_stac92hd95(struct hda_codec *codec) codec->epss = 0; /* longer delay needed for D3 */ spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; @@ -4692,7 +4692,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) return err; spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index d5d1dca4f11b..485663bb9101 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -133,7 +133,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) spec->gen.keep_eapd_on = 1; spec->gen.pcm_playback_hook = via_playback_pcm_hook; spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->gen.power_down_unused = 1; return spec; } @@ -236,7 +236,7 @@ static int via_pin_power_ctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = codec->power_mgmt; + ucontrol->value.enumerated.item[0] = codec->power_save_node; return 0; } @@ -247,9 +247,9 @@ static int via_pin_power_ctl_put(struct snd_kcontrol *kcontrol, struct via_spec *spec = codec->spec; bool val = !!ucontrol->value.enumerated.item[0]; - if (val == codec->power_mgmt) + if (val == codec->power_save_node) return 0; - codec->power_mgmt = val; + codec->power_save_node = val; spec->gen.power_down_unused = val; analog_low_current_mode(codec); return 1; @@ -295,7 +295,7 @@ static void __analog_low_current_mode(struct hda_codec *codec, bool force) bool enable; unsigned int verb, parm; - if (!codec->power_mgmt) + if (!codec->power_save_node) enable = false; else enable = is_aa_path_mute(codec) && !spec->gen.active_streams; @@ -517,7 +517,7 @@ static int via_parse_auto_config(struct hda_codec *codec) return err; /* disable widget PM at start for compatibility */ - codec->power_mgmt = 0; + codec->power_save_node = 0; spec->gen.power_down_unused = 0; return 0; } -- 2.3.3

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[alsa-devel] [PATCH] ASoC: max98925: Constify regmap config and other codec data
by Krzysztof Kozlowski 20 Mar '15

20 Mar '15

Constify local structures (snd_soc_dai_ops, snd_soc_codec_driver, regmap_config) and array (reg_defaults) which are not modified by the driver and passed to core as pointer to const. Signed-off-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> --- sound/soc/codecs/max98925.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index 1f8e80315749..0796dfaeb139 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -34,7 +34,7 @@ static const char *const hpf_text[] = { "Disable", "DC Block", "100Hz", "200Hz", "400Hz", "800Hz", }; -static struct reg_default max98925_reg[] = { +static const struct reg_default max98925_reg[] = { { 0x0B, 0x00 }, /* IRQ Enable0 */ { 0x0C, 0x00 }, /* IRQ Enable1 */ { 0x0D, 0x00 }, /* IRQ Enable2 */ @@ -491,7 +491,7 @@ static int max98925_dai_set_sysclk(struct snd_soc_dai *dai, #define MAX98925_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops max98925_dai_ops = { +static const struct snd_soc_dai_ops max98925_dai_ops = { .set_sysclk = max98925_dai_set_sysclk, .set_fmt = max98925_dai_set_fmt, .hw_params = max98925_dai_hw_params, @@ -541,7 +541,7 @@ static int max98925_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_max98925 = { +static const struct snd_soc_codec_driver soc_codec_dev_max98925 = { .probe = max98925_probe, .controls = max98925_snd_controls, .num_controls = ARRAY_SIZE(max98925_snd_controls), @@ -551,7 +551,7 @@ static struct snd_soc_codec_driver soc_codec_dev_max98925 = { .num_dapm_widgets = ARRAY_SIZE(max98925_dapm_widgets), }; -static struct regmap_config max98925_regmap = { +static const struct regmap_config max98925_regmap = { .reg_bits = 8, .val_bits = 8, .max_register = MAX98925_REV_VERSION, -- 1.9.1

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[alsa-devel] [RFC][PATCH 00/13] alsa-tools: libhinawa for control applications of FireWire devices
by Takashi Sakamoto 20 Mar '15

20 Mar '15

This is RFC for a new library into alsa-tools. ALSA in Linux 3.16 or later extends a support for FireWire sound devices. Currently ALSA drivers supports streaming functionality only, while most of these devices require software implementation to control its internal DSP. The way to achieve this is to transfer byte sequence to the unit and wait byte sequence which the unit transfers if required. This library, libhinawa, just support these operations, nothing others. The supported types of transactions are: - IEEE 1394 read transaction - IEEE 1394 write transaction - IEEE 1394 lock transaction - IEC 61883-1 FCP transaction - Echo Fireworks transaction (with a help of snd-fireworks kernel driver) - Dice notification (with a help of snd-dice kernel driver) To help new developers, this library supports GObject Introspection for language bindings. The main logic of applications can be written with preferred languages such as Python, Ruby, Perl and so on. In the end of this patchset, some Python scripts with Gtk+, Qt4 and Qt5 are committed as samples. This is my first development with GNU Autotools, GLib/GObject and GObject Introspection. Furthermore, I'm a beginner of FireWire subsystem programming. I'm happy to receive your comments, especially: - the way to write Linux version dependency in configure.ac - the way of libtool versioning - the value of poll timeout in fw_unit/snd_unit - the value of thread condition timeout in fw_fcp/snd_dice/snd_efw - improvements of 'unit_query' object - any programming mistakes (threading and so on...) Regards Takashi Sakamoto (13): libhinawa: add build definitions libhinawa: add hinawa context libhinawa: support GTK-Doc to generate references libhinawa: add 'fw_unit' object as a listener for FireWire unit libhinawa: support GObject Introspection for language bindings libhinawa: add 'fw_resp' object as a responder for FireWire transaction libhinawa: add 'fw_req' object as requester for FireWire transaction libhinawa: add 'fw_fcp' object as a helper of FCP transaction libhinawa: add 'snd_unit' object as a listener for ALSA FireWire devices libhinawa: add 'snd_dice' object as a helper for Dice notification libhinawa: add 'snd_efw' object as a helper for EFW transaction libhinawa: add 'unit_query' as a query for ALSA FireWire devices libhinawa: add sample scripts Makefile | 2 +- libhinawa/AUTHORS | 1 + libhinawa/COPYING | 504 ++++++++++++++++++++++++++++ libhinawa/ChangeLog | 5 + libhinawa/Makefile.am | 6 + libhinawa/NEWS | 0 libhinawa/README | 30 ++ libhinawa/autogen.sh | 11 + libhinawa/configure.ac | 68 ++++ libhinawa/doc/Makefile.am | 2 + libhinawa/doc/reference/Makefile.am | 46 +++ libhinawa/doc/reference/hinawa-docs.sgml | 47 +++ libhinawa/doc/reference/version.xml.in | 1 + libhinawa/samples/gtk3.py | 190 +++++++++++ libhinawa/samples/qt4.py | 206 ++++++++++++ libhinawa/samples/qt5.py | 192 +++++++++++ libhinawa/samples/run.sh | 15 + libhinawa/src/Makefile.am | 97 ++++++ libhinawa/src/backport.h | 41 +++ libhinawa/src/fw_fcp.c | 282 ++++++++++++++++ libhinawa/src/fw_fcp.h | 58 ++++ libhinawa/src/fw_req.c | 265 +++++++++++++++ libhinawa/src/fw_req.h | 56 ++++ libhinawa/src/fw_resp.c | 232 +++++++++++++ libhinawa/src/fw_resp.h | 51 +++ libhinawa/src/fw_unit.c | 343 +++++++++++++++++++ libhinawa/src/fw_unit.h | 52 +++ libhinawa/src/hinawa_context.c | 60 ++++ libhinawa/src/hinawa_context.h | 10 + libhinawa/src/internal.h | 31 ++ libhinawa/src/snd_dice.c | 195 +++++++++++ libhinawa/src/snd_dice.h | 53 +++ libhinawa/src/snd_efw.c | 315 ++++++++++++++++++ libhinawa/src/snd_efw.h | 54 +++ libhinawa/src/snd_unit.c | 548 +++++++++++++++++++++++++++++++ libhinawa/src/snd_unit.h | 69 ++++ libhinawa/src/unit_query.c | 116 +++++++ libhinawa/src/unit_query.h | 48 +++ 38 files changed, 4301 insertions(+), 1 deletion(-) create mode 100644 libhinawa/AUTHORS create mode 100644 libhinawa/COPYING create mode 100644 libhinawa/ChangeLog create mode 100644 libhinawa/Makefile.am create mode 100644 libhinawa/NEWS create mode 100644 libhinawa/README create mode 100755 libhinawa/autogen.sh create mode 100644 libhinawa/configure.ac create mode 100644 libhinawa/doc/Makefile.am create mode 100644 libhinawa/doc/reference/Makefile.am create mode 100644 libhinawa/doc/reference/hinawa-docs.sgml create mode 100644 libhinawa/doc/reference/version.xml.in create mode 100755 libhinawa/samples/gtk3.py create mode 100755 libhinawa/samples/qt4.py create mode 100755 libhinawa/samples/qt5.py create mode 100755 libhinawa/samples/run.sh create mode 100644 libhinawa/src/Makefile.am create mode 100644 libhinawa/src/backport.h create mode 100644 libhinawa/src/fw_fcp.c create mode 100644 libhinawa/src/fw_fcp.h create mode 100644 libhinawa/src/fw_req.c create mode 100644 libhinawa/src/fw_req.h create mode 100644 libhinawa/src/fw_resp.c create mode 100644 libhinawa/src/fw_resp.h create mode 100644 libhinawa/src/fw_unit.c create mode 100644 libhinawa/src/fw_unit.h create mode 100644 libhinawa/src/hinawa_context.c create mode 100644 libhinawa/src/hinawa_context.h create mode 100644 libhinawa/src/internal.h create mode 100644 libhinawa/src/snd_dice.c create mode 100644 libhinawa/src/snd_dice.h create mode 100644 libhinawa/src/snd_efw.c create mode 100644 libhinawa/src/snd_efw.h create mode 100644 libhinawa/src/snd_unit.c create mode 100644 libhinawa/src/snd_unit.h create mode 100644 libhinawa/src/unit_query.c create mode 100644 libhinawa/src/unit_query.h -- 2.1.0

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Re: [alsa-devel] Patches to bind the SGTL5000 chip to AM33XX McASP
by Peter Ujfalusi 20 Mar '15

20 Mar '15

Greg, On 03/19/2015 07:48 PM, Greg Knight wrote: >> For a codec such as the SGTL5000 I would connect a clock, which is > running all >> the time. If you take a look at the driver, it enables the clock at > probe and >> leaves it running as long as the driver is loaded. > > Speaking briefly to my "electrical" colleague, he's a bit concerned > about the extra power load that a crystal oscillator might incur, while > I'm concerned about audio quality from a source like CLKOUT2. > > Is there a straightforward way we could enable the clock on-demand - > when we're communicating with the codec/mixer or actually using the > audio subsystem - while keeping it disabled when not in use? We don't > expect to be using the audio subsystem frequently. > > Worst-case we can set up a GPIO and enable/disable it as needed in > user-space - acceptable, if annoying, for our application. See my reply to Nikolay. -- Péter

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