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

[alsa-devel] [PATCH] ASoC: ml26124: Remove duplicate code
by Axel Lin 15 Jun '15

15 Jun '15

Current code has duplicate code for 16000, 32000 and 48000 sample rates. get_srate() returns negative error code for unsupported rate, so we can remove the duplicate code in the swith cases by calling get_srate() first. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/ml26124.c | 58 +++++++++------------------------------------- 1 file changed, 11 insertions(+), 47 deletions(-) diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 62dda24..b74118e 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -341,6 +341,7 @@ static int ml26124_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); int i = get_coeff(priv->mclk, params_rate(hw_params)); + int srate; if (i < 0) return i; @@ -370,53 +371,16 @@ static int ml26124_hw_params(struct snd_pcm_substream *substream, BIT(0) | BIT(1), 0); } - switch (params_rate(hw_params)) { - case 16000: - snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, - get_srate(params_rate(hw_params))); - snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, - coeff_div[i].pllnl); - snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, - coeff_div[i].pllnh); - snd_soc_update_bits(codec, ML26124_PLLML, 0xff, - coeff_div[i].pllml); - snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, - coeff_div[i].pllmh); - snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, - coeff_div[i].plldiv); - break; - case 32000: - snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, - get_srate(params_rate(hw_params))); - snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, - coeff_div[i].pllnl); - snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, - coeff_div[i].pllnh); - snd_soc_update_bits(codec, ML26124_PLLML, 0xff, - coeff_div[i].pllml); - snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, - coeff_div[i].pllmh); - snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, - coeff_div[i].plldiv); - break; - case 48000: - snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, - get_srate(params_rate(hw_params))); - snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, - coeff_div[i].pllnl); - snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, - coeff_div[i].pllnh); - snd_soc_update_bits(codec, ML26124_PLLML, 0xff, - coeff_div[i].pllml); - snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, - coeff_div[i].pllmh); - snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, - coeff_div[i].plldiv); - break; - default: - pr_err("%s:this rate is no support for ml26124\n", __func__); - return -EINVAL; - } + srate = get_srate(params_rate(hw_params)); + if (srate < 0) + return srate; + + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, srate); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, coeff_div[i].plldiv); return 0; } -- 2.1.0

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[alsa-devel] [PATCH] ALSA: hda - adding a DAC/pin preference map for a HP Envy TS machine
by Hui Wang 15 Jun '15

15 Jun '15

On a HP Envy TouchSmart laptop, there are 2 speakers (main speaker and subwoofer speaker), 1 headphone and 2 DACs, without this fixup, the headphone will be assigned to a DAC and the 2 speakers will be assigned to another DAC, this assignment makes the surround-2.1 channels invalid. To fix it, here using a DAC/pin preference map to bind the main speaker to 1 DAC and the subwoofer speaker will be assigned to another DAC. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index bdcda6a..dcc7fe9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -100,6 +100,7 @@ enum { STAC_HP_ENVY_BASS, STAC_HP_BNB13_EQ, STAC_HP_ENVY_TS_BASS, + STAC_HP_ENVY_TS_DAC_BIND, STAC_92HD83XXX_GPIO10_EAPD, STAC_92HD83XXX_MODELS }; @@ -2171,6 +2172,22 @@ static void stac92hd83xxx_fixup_gpio10_eapd(struct hda_codec *codec, spec->eapd_switch = 0; } +static void hp_envy_ts_fixup_dac_bind(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct sigmatel_spec *spec = codec->spec; + static hda_nid_t preferred_pairs[] = { + 0xd, 0x13, + 0 + }; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + spec->gen.preferred_dacs = preferred_pairs; +} + static const struct hda_verb hp_bnb13_eq_verbs[] = { /* 44.1KHz base */ { 0x22, 0x7A6, 0x3E }, @@ -2686,6 +2703,12 @@ static const struct hda_fixup stac92hd83xxx_fixups[] = { {} }, }, + [STAC_HP_ENVY_TS_DAC_BIND] = { + .type = HDA_FIXUP_FUNC, + .v.func = hp_envy_ts_fixup_dac_bind, + .chained = true, + .chain_id = STAC_HP_ENVY_TS_BASS, + }, [STAC_92HD83XXX_GPIO10_EAPD] = { .type = HDA_FIXUP_FUNC, .v.func = stac92hd83xxx_fixup_gpio10_eapd, @@ -2764,6 +2787,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { "HP bNB13", STAC_HP_BNB13_EQ), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x190e, "HP ENVY TS", STAC_HP_ENVY_TS_BASS), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1967, + "HP ENVY TS", STAC_HP_ENVY_TS_DAC_BIND), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1940, "HP bNB13", STAC_HP_BNB13_EQ), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1941, -- 1.9.1

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[alsa-devel] [PATCH 1/3] ASoC: max98090: read micbias from device property
by yang.a.fang＠intel.com 15 Jun '15

15 Jun '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> This patch reads max98090 micbias from acpi or dt Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/codecs/max98090.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index c230626..27df17e 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2419,6 +2419,8 @@ static int max98090_probe(struct snd_soc_codec *codec) struct max98090_cdata *cdata; enum max98090_type devtype; int ret = 0; + int err; + unsigned int micbias; dev_dbg(codec->dev, "max98090_probe\n"); @@ -2503,8 +2505,13 @@ static int max98090_probe(struct snd_soc_codec *codec) snd_soc_write(codec, M98090_REG_BIAS_CONTROL, M98090_VCM_MODE_MASK); + err = device_property_read_u32(codec->dev, "max,micbias", &micbias); + + if (err) + micbias = M98090_MBVSEL_2V8; + snd_soc_update_bits(codec, M98090_REG_MIC_BIAS_VOLTAGE, - M98090_MBVSEL_MASK, M98090_MBVSEL_2V8); + M98090_MBVSEL_MASK, micbias); max98090_add_widgets(codec); -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: Intel: mrfld: simplify sst_fill_widget_module_info
by Rasmus Villemoes 15 Jun '15

15 Jun '15

Some tiny improvements, cutting 180 bytes off the generated code. - use strchr() for single-character needle - compute index using pointer subtraction instead of two strlen() calls - factor out the common check for whether the initial part of kctl->id.name (before the space) is identical to w->name. Signed-off-by: Rasmus Villemoes <linux(a)rasmusvillemoes.dk> --- sound/soc/intel/atom/sst-atom-controls.c | 20 ++++++++------------ 1 file changed, 8 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 90aa5c0476f3..16670feeee6a 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -1280,36 +1280,32 @@ static int sst_fill_widget_module_info(struct snd_soc_dapm_widget *w, down_read(&card->controls_rwsem); list_for_each_entry(kctl, &card->controls, list) { - idx = strstr(kctl->id.name, " "); + idx = strchr(kctl->id.name, ' '); if (idx == NULL) continue; - index = strlen(kctl->id.name) - strlen(idx); + index = idx - (char*)kctl->id.name; + if (strncmp(kctl->id.name, w->name, index)) + continue; - if (strstr(kctl->id.name, "Volume") && - !strncmp(kctl->id.name, w->name, index)) + if (strstr(kctl->id.name, "Volume")) ret = sst_fill_module_list(kctl, w, SST_MODULE_GAIN); - else if (strstr(kctl->id.name, "params") && - !strncmp(kctl->id.name, w->name, index)) + else if (strstr(kctl->id.name, "params")) ret = sst_fill_module_list(kctl, w, SST_MODULE_ALGO); else if (strstr(kctl->id.name, "Switch") && - !strncmp(kctl->id.name, w->name, index) && strstr(kctl->id.name, "Gain")) { struct sst_gain_mixer_control *mc = (void *)kctl->private_value; mc->w = w; - } else if (strstr(kctl->id.name, "interleaver") && - !strncmp(kctl->id.name, w->name, index)) { + } else if (strstr(kctl->id.name, "interleaver")) { struct sst_enum *e = (void *)kctl->private_value; e->w = w; - } else if (strstr(kctl->id.name, "deinterleaver") && - !strncmp(kctl->id.name, w->name, index)) { - + } else if (strstr(kctl->id.name, "deinterleaver")) { struct sst_enum *e = (void *)kctl->private_value; e->w = w; -- 2.1.3

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[alsa-devel] [PATCH 0/3] ASoC: Mediatek: Add support for MT8173 SoC
by Koro Chen 15 Jun '15

15 Jun '15

This adds basic support for the Mediatek AFE (Audio Front End) unit for MT8173. This patch is based on Linux 4.1-rc1 and Sascha's SCPSYS power patch [1]. The AFE unit comprises several memory interfaces that communicate with CPU, a multi input multi output digital audio interconnect, and several external interfaces (e.g. I2S). [1] https://lkml.org/lkml/2015/6/9/172 changes since RFC: - Use DPCM to model AFE AFE memory interfaces such as DL1, VUL are modeled as frontend DAIs, and AFE IOs such as I2S, HDMI are backend DAIs. Which memif and IO to be used can be set in the machine driver. There is no need to indicate this in dts anymore so include/dt-bindings/sound/mtk-afe.h is removed. AFE connections are represented by DAPM widgets and routes, and set connection APIs/tables are no longer required. (mtk-afe-connection.c and mtk-afe-connection.h are removed) Instead, mixer controls are used to do connection/disconnection.

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Re: [alsa-devel] M-Audio Midisport 2x2 Anniversary edition, latency problem
by Clemens Ladisch 14 Jun '15

14 Jun '15

Ilya Dikariev wrote: >>>>> I bought a MIDI-interface von M-Audio (see subject), but there are >>>>> strange things with it on my desktop (Arch, latest kernel). The >>>>> latency is about 5 seconds. >>>> >>>> How are you measuring the latency? >>> >>> I measured it only with my feeling of time. >> >> What are the specific events between which there is an interval of >> 5 seconds? > > Between I press any key on a MIDI-keyboard and I hear the sound from > the timidity-daemon (or any other MIDI-reciever in a computer). There are lots of components involved besides the USB/MIDI interface: Timidity, any sound server (if you're using one), and the sound card's drivers. It is extremely unlikely that the USB/MIDI interface could introduce latency, because the only way to do that would be to go to the extra effort of storing the MIDI events in some buffer, and then transmitting them later. > the MIDI events from the Pitch-wheel on the keyboard arrives the MIDI- > reciever immedietly (I saw it, when i tried to check out the output > from the /dev/midi). This proves that another component introduces the latency. (Note: you could also use amidi or aseqdump to do these tests.) How did you configure the audio output of Timidity? Regards, Clemens

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[alsa-devel] 2 speakers are assigned to the same DAC, this can't support 4.0/2.1 channles
by hwang4 14 Jun '15

14 Jun '15

I am trying to enable the subwoofer speaker on a HP laptop, on this machine, there are two speakers and one headphone, but the BIOS verb only enabled one speaker(nid 0xd) and one headphone(nid 0xb), I need to use quirk in the kernel driver to configure the second speaker (subwoofer speaker, nid 0x10). Under current alsa driver, the headphone will be assigned a dac (nid 0x13) and the 2 speakers will be assigned a dac (nid 0x14), this assignment is not good since 2 speakers share the same dac, this means 2 speakers can't work at the same time to support 4.0/2.1 channels. On another Dell machine with realtek codec, there are also 2 speakers, 1 headphone and 2 dacs, on this machine, 1 speaker and 1 headphone are assigned 1 dac, and the other speaker is assigned another dac, so there is no problem for this machine to support 4.0/2.1 channels. Through debugging, I found on Dell machine, the speaker nid only has one connection to dac (hardwired), so when driver assign dac to it, the map_single() can successfully assign the each dac to the 2 speakers respectively. But on that HP machine, the speaker has multiple connections for dac, the map_single() can't work for this machine. The alsa-info.txt for that HP machine is at http://pastebin.ubuntu.com/11667947/ The alsa-info.txt for that Dell machine is at http://pastebin.ubuntu.com/11667982/ To fix this problem, I can use the quirk in the driver to make the speaker have only one connection for dac or do some change in the hda_generic.c like below, in your opinion, which one is better or probably you have a different idea to fix this problem? diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index ac0db16..8194ff1 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1363,6 +1363,7 @@ static int try_assign_dacs(struct hda_codec *codec, int num_outs, dac = try_dac(codec, get_primary_out(codec, i), pin); if (!dac) dac = try_dac(codec, dacs[0], pin); if (!dac) dac = try_dac(codec, get_primary_out(codec, i), pin); if (dac) { @@ -1762,6 +1763,12 @@ static int fill_and_eval_dacs(struct hda_codec *codec, if (err < 0) return err; badness += err; + /* if there are 2 speakers and both of them are assigned to the same dac, + we need to increase the badness for this situation, because in this situation + the 2 speakers can't work together to support 4.0/2.1 channels */ + if (cfg->speaker_outs == 2 && err < spec->extra_out_badness->no_dac && + spec->multiout.extra_out_nid[1] == 0) + badness += spec->extra_out_badness->no_dac; } if (!spec->no_multi_io && cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index bdcda6a..717259c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2764,6 +2764,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { "HP bNB13", STAC_HP_BNB13_EQ), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x190e, "HP ENVY TS", STAC_HP_ENVY_TS_BASS), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1967, + "HP ENVY TS", STAC_HP_ENVY_TS_BASS), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1940, "HP bNB13", STAC_HP_BNB13_EQ), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1941,

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Re: [alsa-devel] Intel HDA & Realtek issues on ASUS G751JM
by Raymond Yau 13 Jun '15

13 Jun '15

> > Do you mean hdajackretask cannot shutdown pulseaudio or you still have other > > applicaion using the sound card which prevent dynamic reconfiguration of > > snd-hda-intel ? > > > I was testing using a browser on a youtube video. When I apply the > changes, I lose the sound (pulseaudio gets killed), that message > appears and I have to refresh the page to get sound again - but > nothing changes. > > > > > Can you post the output of alsa-info.sh after you patched the line out jacks > > ? > http://www.alsa-project.org/db/?f=82a057b76a781edb53217b2cc096bce35459ab0a > > > Which node is the headphone spdif combo jack ? > > > Using hdajacksensetest I found out the mapping: > SPDIF port (jack 1): Pin 0x16 (Green Line Out, Rear side) > jack 2: Pin 0x18 (Green Line Out, Rear side) > jack 3: Pin 0x15 (Green Line Out, Rear side) > > > Do both headphone jacks have headphone , headset or line out icon ? > You can see the icons here [0] at page 20 > > > Do you mean switch to 5.1 profile in sound preference is not enough ? > By listening on Youtube I can't hear the center and subwoofer without > modifying the pulseaudio conf > > [0] http://dlcdnet.asus.com/pub/ASUS/nb/G751JM/0409_E9481.pdf?_ga=1.206896403.1… If your specifcaion and user guide does not mention the support of surround 5.1 You need to find out the internal subwoofer use both channel or just left or right speaker-test -c4 -t wav -D hw:0,0 If you hear high frequency from subwoofer, this mean you need software low pass filtter Two headphone jacks, internal speaker and subwoofer There are only three playback volume controls http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093668.html Your case are A. 3 DACs, 2 HP, 2 speakers Node 0x1a [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Bass Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Surround Phantom Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x90170130: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 3 0x0c 0x0d* 0x0e

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[alsa-devel] ASoC: How to declare volume TLV dB scale?
by Sergej Sawazki 13 Jun '15

13 Jun '15

Dear all, could someone please help me to understand the mixer controls TLV scale? What would be the DECLARE_TLV_DB_SCALE values for the following example? reg. value: 0 .. 0x3ff phys. value: 0dB .. -128dB volume: 100% .. 0%(mute) What does TLV actually stand for? Many thanks in advance, Sergej Sawazki

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[alsa-devel] Intel HDA & Realtek issues on ASUS G751JM
by Nicolae Rosia 13 Jun '15

13 Jun '15

Hello, I can't get 4.0/5.1 sound working on my laptop [0] [1] with Fedora 22. With Linux 4.0 there is no output on jacks but with 4.1-rc6 I'm getting sound only on one output. Switching between pacmd set-card-profile alsa_card.pci-0000_00_1b.0 output:analog-stereo and pacmd set-card-profile alsa_card.pci-0000_00_1b.0 output:analog-surround-40 only boosts the volume on the first output. Attached is dmesg [2], lsmod [3], result of lspci -vv [4] and pacmd list-cards and list-sinks [5] and output of various information from /proc/asound while runnig 4.1-rc6. Let me know if you need more information or how I can help. Best regards, Nicolae Rosia [rosia@localhost ~]$ cat /proc/asound/devices 1: : sequencer 2: [ 0] : control 3: [ 0- 3]: digital audio playback 4: [ 0- 7]: digital audio playback 5: [ 0- 8]: digital audio playback 6: [ 0- 0]: hardware dependent 7: [ 1] : control 8: [ 1- 0]: digital audio playback 9: [ 1- 0]: digital audio capture 10: [ 1- 1]: digital audio playback 11: [ 1- 0]: hardware dependent 33: : timer [rosia@localhost ~]$ cat /proc/asound/cards 0 [HDMI ]: HDA-Intel - HDA Intel HDMI HDA Intel HDMI at 0xeda14000 irq 33 1 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xeda10000 irq 32 [rosia@localhost ~]$ cat /proc/asound/modules 0 snd_hda_intel 1 snd_hda_intel [rosia@localhost ~]$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version k4.1.0-0.rc6.git2.1.fc23.x86_64. [0] http://www.emag.ro/laptop-gaming-asus-cu-procesor-intelr-coretm-i7-4710hq-2… [1] https://www.asus.com/Notebooks_Ultrabooks/ASUS_ROG_G751JM/specifications/ [2] http://pastebin.com/rm3dPPYm [3] http://pastebin.com/gHZWjgBx [4] http://pastebin.com/UUf91Q31 [5] http://pastebin.com/ca5dcxy8

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