August 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: wm5102: Add AEC routing control
by Mark Brown 16 Aug '12

16 Aug '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a5aea03..e2fb07e 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -274,6 +274,27 @@ ARIZONA_MIXER_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE); + +static const char *wm5102_aec_loopback_texts[] = { + "HPOUT1L", "HPOUT1R", "HPOUT2L", "HPOUT2R", "EPOUT", + "SPKOUTL", "SPKOUTR", "SPKDAT1L", "SPKDAT1R", +}; + +static const unsigned int wm5102_aec_loopback_values[] = { + 0, 1, 2, 3, 4, 6, 7, 8, 9, +}; + +static const struct soc_enum wm5102_aec_loopback = + SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_SRC_SHIFT, + ARIZONA_AEC_LOOPBACK_SRC_MASK, + ARRAY_SIZE(wm5102_aec_loopback_texts), + wm5102_aec_loopback_texts, + wm5102_aec_loopback_values); + +static const struct snd_kcontrol_new wm5102_aec_loopback_mux = + SOC_DAPM_VALUE_ENUM("AEC Loopback", wm5102_aec_loopback); + static const struct snd_soc_dapm_widget wm5102_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, 0, NULL, 0), @@ -425,6 +446,9 @@ SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0, SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux), + SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), @@ -520,6 +544,7 @@ SND_SOC_DAPM_OUTPUT("SPKDAT1R"), { name, "Noise Generator", "Noise Generator" }, \ { name, "Tone Generator 1", "Tone Generator 1" }, \ { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "AEC", "AEC Loopback" }, \ { name, "IN1L", "IN1L PGA" }, \ { name, "IN1R", "IN1R PGA" }, \ { name, "IN2L", "IN2L PGA" }, \ @@ -685,21 +710,30 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("ASRC2L", "ASRC2L"), ARIZONA_MIXER_ROUTES("ASRC2R", "ASRC2R"), + { "AEC Loopback", "HPOUT1L", "OUT1L" }, + { "AEC Loopback", "HPOUT1R", "OUT1R" }, { "HPOUT1L", NULL, "OUT1L" }, { "HPOUT1R", NULL, "OUT1R" }, + { "AEC Loopback", "HPOUT2L", "OUT2L" }, + { "AEC Loopback", "HPOUT2R", "OUT2R" }, { "HPOUT2L", NULL, "OUT2L" }, { "HPOUT2R", NULL, "OUT2R" }, + { "AEC Loopback", "EPOUT", "OUT3L" }, { "EPOUTN", NULL, "OUT3L" }, { "EPOUTP", NULL, "OUT3L" }, + { "AEC Loopback", "SPKOUTL", "OUT4L" }, { "SPKOUTLN", NULL, "OUT4L" }, { "SPKOUTLP", NULL, "OUT4L" }, + { "AEC Loopback", "SPKOUTR", "OUT4R" }, { "SPKOUTRN", NULL, "OUT4R" }, { "SPKOUTRP", NULL, "OUT4R" }, + { "AEC Loopback", "SPKDAT1L", "OUT5L" }, + { "AEC Loopback", "SPKDAT1R", "OUT5R" }, { "SPKDAT1L", NULL, "OUT5L" }, { "SPKDAT1R", NULL, "OUT5R" }, }; -- 1.7.10.4

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Re: [alsa-devel] [PATCH 1/5] ARM: tegra: config: enable dmaengine based APB DMA driver
by Stephen Warren 16 Aug '12

16 Aug '12

On 08/16/2012 08:13 AM, Laxman Dewangan wrote: > Enable config for dmaengine based Tegra APB DMA driver and > disable the legacy APB DMA driver (SYSTEM_DMA). Laxman, if I apply this series to next-20120816 (plus a few patches in my local work branch plus the CPU hotplug patches from Joseph, although I suspect none of that matters), then audio playback on Tegra20 is broken; the pitch is far too high. Audio playback on Tegra30 works as expected.

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[alsa-devel] [PATCH] ASoC: wm5102: Remove DRC2
by Mark Brown 16 Aug '12

16 Aug '12

It will be removed from future device revisions. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 16 ---------------- 1 file changed, 16 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a00901c..a5aea03 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -128,13 +128,9 @@ SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE), SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), -SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5, - ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA), ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), @@ -236,8 +232,6 @@ ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); -ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE); -ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); @@ -353,10 +347,6 @@ SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0, - NULL, 0), -SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0, - NULL, 0), SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, NULL, 0), @@ -470,8 +460,6 @@ ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"), ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), -ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"), -ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"), ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), @@ -557,8 +545,6 @@ SND_SOC_DAPM_OUTPUT("SPKDAT1R"), { name, "EQ4", "EQ4" }, \ { name, "DRC1L", "DRC1L" }, \ { name, "DRC1R", "DRC1R" }, \ - { name, "DRC2L", "DRC2L" }, \ - { name, "DRC2R", "DRC2R" }, \ { name, "LHPF1", "LHPF1" }, \ { name, "LHPF2", "LHPF2" }, \ { name, "LHPF3", "LHPF3" }, \ @@ -688,8 +674,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), - ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"), - ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"), ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), -- 1.7.10.4

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[alsa-devel] [PATCH] hda-emu: Add idt_set_gpio verb for stac92xx
by David Henningsson 16 Aug '12

16 Aug '12

According to Takashi, this verb is valid and given by Sigmatel. Also merge two identical arrays and sort the table a little. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hda-spec.c | 66 ++++++++++++++++++++++++++++-------------------------------- 1 file changed, 31 insertions(+), 35 deletions(-) diff --git a/hda-spec.c b/hda-spec.c index dd43ee9..eebaa39 100644 --- a/hda-spec.c +++ b/hda-spec.c @@ -155,15 +155,10 @@ static struct xhda_verb_table idt_92hd8xx_verbs[] = { /* * STAC codecs */ -static struct xhda_verb_table stac927x_verbs[] = { +static struct xhda_verb_table stac92xx_verbs[] = { { 0xfeb, get_cached_verb, "get_analog_loop" }, /* undocumented */ { 0x7eb, set_cached_verb, "set_analog_loop" }, - { } /* terminator */ -}; - -static struct xhda_verb_table stac9205_verbs[] = { - { 0xfe0, get_cached_verb, "get_analog_loop" }, /* undocumented */ - { 0x7e0, set_cached_verb, "set_analog_loop" }, + { 0x7e7, set_cached_verb, "idt_set_gpio" }, { } /* terminator */ }; @@ -271,34 +266,35 @@ static struct verb_ext_list extensions[] = { { .id = 0x111d76e5, .verbs = idt_92hd8xx_verbs }, { .id = 0x111d76e7, .verbs = idt_92hd8xx_verbs }, - { .id = 0x83847618, .verbs = stac927x_verbs }, - { .id = 0x83847619, .verbs = stac927x_verbs }, - { .id = 0x83847616, .verbs = stac927x_verbs }, - { .id = 0x83847617, .verbs = stac927x_verbs }, - { .id = 0x83847614, .verbs = stac927x_verbs }, - { .id = 0x83847615, .verbs = stac927x_verbs }, - { .id = 0x83847620, .verbs = stac927x_verbs }, - { .id = 0x83847621, .verbs = stac927x_verbs }, - { .id = 0x83847622, .verbs = stac927x_verbs }, - { .id = 0x83847623, .verbs = stac927x_verbs }, - { .id = 0x83847624, .verbs = stac927x_verbs }, - { .id = 0x83847625, .verbs = stac927x_verbs }, - { .id = 0x83847626, .verbs = stac927x_verbs }, - { .id = 0x83847627, .verbs = stac927x_verbs }, - { .id = 0x83847628, .verbs = stac927x_verbs }, - { .id = 0x83847629, .verbs = stac927x_verbs }, - { .id = 0x83847645, .verbs = stac927x_verbs }, - { .id = 0x83847646, .verbs = stac927x_verbs }, - - { .id = 0x83847698, .verbs = stac9205_verbs }, - { .id = 0x838476a0, .verbs = stac9205_verbs }, - { .id = 0x838476a1, .verbs = stac9205_verbs }, - { .id = 0x838476a2, .verbs = stac9205_verbs }, - { .id = 0x838476a3, .verbs = stac9205_verbs }, - { .id = 0x838476a4, .verbs = stac9205_verbs }, - { .id = 0x838476a5, .verbs = stac9205_verbs }, - { .id = 0x838476a6, .verbs = stac9205_verbs }, - { .id = 0x838476a7, .verbs = stac9205_verbs }, + { .id = 0x83847614, .verbs = stac92xx_verbs }, + { .id = 0x83847615, .verbs = stac92xx_verbs }, + { .id = 0x83847616, .verbs = stac92xx_verbs }, + { .id = 0x83847617, .verbs = stac92xx_verbs }, + { .id = 0x83847618, .verbs = stac92xx_verbs }, + { .id = 0x83847619, .verbs = stac92xx_verbs }, + { .id = 0x83847620, .verbs = stac92xx_verbs }, + { .id = 0x83847621, .verbs = stac92xx_verbs }, + { .id = 0x83847622, .verbs = stac92xx_verbs }, + { .id = 0x83847623, .verbs = stac92xx_verbs }, + { .id = 0x83847624, .verbs = stac92xx_verbs }, + { .id = 0x83847625, .verbs = stac92xx_verbs }, + { .id = 0x83847626, .verbs = stac92xx_verbs }, + { .id = 0x83847627, .verbs = stac92xx_verbs }, + { .id = 0x83847628, .verbs = stac92xx_verbs }, + { .id = 0x83847629, .verbs = stac92xx_verbs }, + { .id = 0x83847645, .verbs = stac92xx_verbs }, + { .id = 0x83847646, .verbs = stac92xx_verbs }, + + { .id = 0x83847690, .verbs = stac92xx_verbs }, + { .id = 0x83847698, .verbs = stac92xx_verbs }, + { .id = 0x838476a0, .verbs = stac92xx_verbs }, + { .id = 0x838476a1, .verbs = stac92xx_verbs }, + { .id = 0x838476a2, .verbs = stac92xx_verbs }, + { .id = 0x838476a3, .verbs = stac92xx_verbs }, + { .id = 0x838476a4, .verbs = stac92xx_verbs }, + { .id = 0x838476a5, .verbs = stac92xx_verbs }, + { .id = 0x838476a6, .verbs = stac92xx_verbs }, + { .id = 0x838476a7, .verbs = stac92xx_verbs }, { .id = 0x11061708, .verbs = vt1708b_verbs }, { .id = 0x11061709, .verbs = vt1708b_verbs }, -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: wm8994: Add bytes controls for DRC
by Mark Brown 16 Aug '12

16 Aug '12

If DRC coefficients are not configured via platform data then add bytes controls for them instead so they can be configured by applications. This is the normal means of controlling things like this for newer systems, we maintain compatibility with platform data to avoid disruption to existing systems. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 25 +++++++++++++++++++++---- 1 file changed, 21 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 7755ac9..2d60ed3 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -759,6 +759,18 @@ SOC_SINGLE_TLV("AIF2 EQ5 Volume", WM8994_AIF2_EQ_GAINS_2, 6, 31, 0, eq_tlv), }; +static const struct snd_kcontrol_new wm8994_drc_controls[] = { +SND_SOC_BYTES_MASK("AIF1.1 DRC", WM8994_AIF1_DRC1_1, 5, + WM8994_AIF1DAC1_DRC_ENA | WM8994_AIF1ADC1L_DRC_ENA | + WM8994_AIF1ADC1R_DRC_ENA), +SND_SOC_BYTES_MASK("AIF1.2 DRC", WM8994_AIF1_DRC2_1, 5, + WM8994_AIF1DAC2_DRC_ENA | WM8994_AIF1ADC2L_DRC_ENA | + WM8994_AIF1ADC2R_DRC_ENA), +SND_SOC_BYTES_MASK("AIF2 DRC", WM8994_AIF2_DRC_1, 5, + WM8994_AIF2DAC_DRC_ENA | WM8994_AIF2ADCL_DRC_ENA | + WM8994_AIF2ADCR_DRC_ENA), +}; + static const char *wm8958_ng_text[] = { "30ms", "125ms", "250ms", "500ms", }; @@ -3254,14 +3266,19 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) ret = snd_soc_add_codec_controls(wm8994->hubs.codec, controls, ARRAY_SIZE(controls)); - if (ret != 0) - dev_err(wm8994->hubs.codec->dev, - "Failed to add DRC mode controls: %d\n", ret); - for (i = 0; i < WM8994_NUM_DRC; i++) wm8994_set_drc(codec, i); + } else { + ret = snd_soc_add_codec_controls(wm8994->hubs.codec, + wm8994_drc_controls, + ARRAY_SIZE(wm8994_drc_controls)); } + if (ret != 0) + dev_err(wm8994->hubs.codec->dev, + "Failed to add DRC mode controls: %d\n", ret); + + dev_dbg(codec->dev, "%d ReTune Mobile configurations\n", pdata->num_retune_mobile_cfgs); -- 1.7.10.4

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[alsa-devel] [PATCH v2] ALSA: hda - Don't send invalid volume knob command on IDT 92hd75bxx
by David Henningsson 16 Aug '12

16 Aug '12

Instead of blindly initializing a volume knob widget, first check that there actually is a volume knob widget. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 94040cc..ea5775a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4272,7 +4272,8 @@ static int stac92xx_init(struct hda_codec *codec) unsigned int gpio; int i; - snd_hda_sequence_write(codec, spec->init); + if (spec->init) + snd_hda_sequence_write(codec, spec->init); /* power down adcs initially */ if (spec->powerdown_adcs) @@ -5748,7 +5749,6 @@ again: /* fallthru */ case 0x111d76b4: /* 6 Port without Analog Mixer */ case 0x111d76b5: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5773,7 +5773,6 @@ again: spec->stream_delay = 40; /* 40 milliseconds */ /* disable VSW */ - spec->init = stac92hd71bxx_core_init; unmute_init++; snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); @@ -5788,7 +5787,6 @@ again: /* fallthru */ default: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5796,6 +5794,9 @@ again: break; } + if (get_wcaps_type(get_wcaps(codec, 0x28)) == AC_WID_VOL_KNB) + spec->init = stac92hd71bxx_core_init; + if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) snd_hda_sequence_write_cache(codec, unmute_init); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] hda-emu: Fixup ALC268 input amp on Beep node
by David Henningsson 16 Aug '12

16 Aug '12

According to the ALC268 datasheet, there is actually an input amplifier on node 0x1d, even though this is not exposed in the codec proc file. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hda-spec.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) Relevant alsa info: alc268-toshiba-tecra-s11-m11--nvidia--ccert-201002-5346 diff --git a/hda-spec.c b/hda-spec.c index f1b03b4..dd43ee9 100644 --- a/hda-spec.c +++ b/hda-spec.c @@ -386,6 +386,20 @@ static void fixup_via_mixer(struct xhda_codec *codec) node->node[node->num_nodes++] = 0x08; } +static void fixup_alc268_beep(struct xhda_codec *codec) +{ + struct xhda_node *node = find_node(codec, 0x1d); + if (!node) + return; + node->wcaps |= (1 << 1); /* AMP-in present */ + + node->amp_in_caps.ofs = 0x0c; + node->amp_in_caps.nsteps = 0x0c; + node->amp_in_caps.stepsize = 0x07; + node->amp_in_caps.mute = 0; + node->amp_in_caps.override = 1; +} + struct fixup_list { unsigned int vendor_id; void (*func)(struct xhda_codec *); @@ -397,6 +411,7 @@ static struct fixup_list fixups[] = { { 0x11060438, fixup_via_mixer }, { 0x11064438, fixup_via_mixer }, { 0x11060448, fixup_via_mixer }, + { 0x10ec0268, fixup_alc268_beep }, { } }; -- 1.7.9.5

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[alsa-devel] command 0x7e7 sent on STAC9200 / STAC9228
by David Henningsson 16 Aug '12

16 Aug '12

Hi Takashi, While working with fixing the stuff that my automated test tool has discovered, here's one I'm not sure what to do with. If you run hda-emu on any of these three: codecs/canonical/stac9200-dell-inspiron-6400-ccert-200705-83 codecs/canonical/stac9200-dell-inspiron-6400-ccert-200712-207 codecs/canonical/stac9228-dell-inspiron-1420-ccert-200708-95 Hda-emu prints out: invalid command: NID=0x1, verb=0x7e7, parm=0x0 In this case, I'm not sure if the command is invalid (and thus should be removed), or if it's a secret IDT verb that we need to set. From patch_sigmatel.c: /* Configure GPIOx as CMOS */ snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0); But in hda-spec.c this verb is only in the verb table of IDT codecs, not STAC codecs. Can you offer some enlightement? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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[alsa-devel] [PATCH] ALSA: hda - Don't set invalid volume knob command on IDT 92hd75bxx
by David Henningsson 16 Aug '12

16 Aug '12

Instead of blindly initializing a volume knob widget, first check that there actually is a volume knob widget. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) Relevant alsa info: 92hd75b3x5-hp-probook-6550b-ccert-201101-6954 diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 94040cc..1018324 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4272,7 +4272,8 @@ static int stac92xx_init(struct hda_codec *codec) unsigned int gpio; int i; - snd_hda_sequence_write(codec, spec->init); + if (spec->init) + snd_hda_sequence_write(codec, spec->init); /* power down adcs initially */ if (spec->powerdown_adcs) @@ -5748,7 +5749,6 @@ again: /* fallthru */ case 0x111d76b4: /* 6 Port without Analog Mixer */ case 0x111d76b5: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5773,7 +5773,6 @@ again: spec->stream_delay = 40; /* 40 milliseconds */ /* disable VSW */ - spec->init = stac92hd71bxx_core_init; unmute_init++; snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); @@ -5788,7 +5787,6 @@ again: /* fallthru */ default: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5796,6 +5794,10 @@ again: break; } + if (codec->start_nid + codec->num_nodes >= 0x28 && + get_wcaps_type(get_wcaps(codec, 0x28)) == AC_WID_VOL_KNB) + spec->init = stac92hd71bxx_core_init; + if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) snd_hda_sequence_write_cache(codec, unmute_init); -- 1.7.9.5

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Re: [alsa-devel] Question about Conversion to S24_LE from S24_3LE
by Takashi Iwai 16 Aug '12

16 Aug '12

At Thu, 16 Aug 2012 16:55:55 +0900 (JST), etsuko suzuki wrote: > > Thank you for your kindness reply. > > > Did you subscribe to the ML? > > Yes I did. > I subscribed ML and I accepted the mail 'Welcome to the "Alsa-devel" mailing l > ist'. > And I can receive mails of this ML. > But I can't post it to ML. > (The mails were NOT rejected.) > > I had sent e-mail to 'alsa-devel-owner(a)alsa-project.org' several times. > But there was no answer. > (The mails were NOT rejected.) > > Is something done else? > I am sorry about like this question... Hm, I guess your message didn't reach properly, or was filtered by some spam filter. Better to contact with Jaroslav (Cc'ed), who maintains alsa-devel ML. Takashi > > Regards. > > Etsuko >

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