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

Re: [alsa-devel] [PATCH 2/2]: ALSA: VIA HDA: Add support for codec VT180
by Takashi Iwai 10 Dec '12

10 Dec '12

At Mon, 10 Dec 2012 19:03:26 +0800, Lydia wrote: > > From: Lydia Wang <lydiawang(a)viatech.com.cn> > Subject: ALSA: VIA HDA: Add support for codec VT1808. > > Add support for new codec VT1808, which is similiar with VT1705CF. > > Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Thanks, applied both patches now. Takashi > Index: sound/sound/pci/hda/patch_via.c > =================================================================== > --- sound.orig/sound/pci/hda/patch_via.c 2012-12-10 15:09:52.000000000 +0800 > +++ sound/sound/pci/hda/patch_via.c 2012-12-10 15:12:50.000000000 +0800 > @@ -77,6 +77,7 @@ > VT1812, > VT1802, > VT1705CF, > + VT1808, > CODEC_TYPES, > }; > > @@ -300,6 +301,8 @@ > codec_type = VT1802; > else if (dev_id == 0x4760) > codec_type = VT1705CF; > + else if (dev_id == 0x4761 || dev_id == 0x4762) > + codec_type = VT1808; > else > codec_type = UNKNOWN; > return codec_type; > @@ -1112,6 +1115,7 @@ > parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ > break; > case VT1705CF: > + case VT1808: > verb = 0xf82; > parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ > break; > @@ -4061,6 +4065,10 @@ > .patch = patch_vt2002P}, > { .id = 0x11064760, .name = "VT1705CF", > .patch = patch_vt3476}, > + { .id = 0x11064761, .name = "VT1708SCE", > + .patch = patch_vt3476}, > + { .id = 0x11064762, .name = "VT1808", > + .patch = patch_vt3476}, > {} /* terminator */ > }; > > > >

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[alsa-devel] [PATCH 0/7] ALSA: CA0132: DSP Update
by Ian Minett 10 Dec '12

10 Dec '12

From: Ian Minett <ian_minett(a)creativelabs.com> Hi Takashi, thanks for the info about the endianness. If the controller natively swaps each element as it gets read/written to the DSP then it sounds like it should work as is... So with that in mind, we'd like to carry on with the DSP updates, if that's ok. The first batch is included in this patch series. 1: Add new definitions that will be used by the DSP routines 2: Add the new DSP mixer, switches and controls. This is quite large, but it's a little difficult to break it down further - hope it's ok. 3: Add code for handling data returned from the DSP. 4: Add support for tuning controls and data 5/6: Code tidy, split into two for a cleaner diff. 7: Updates to the mic sample rate, module description and memalloc error handling. Thanks very much, please let us know if we need to alter anything. - Ian Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com>

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Re: [alsa-devel] what is period_size, buffer_size and boundary size, i confuse them
by l9jj＠163.com 10 Dec '12

10 Dec '12

thank you very much for your help. i got another question in the interface snd_pcm_update_hw_ptr0(). The reference code is below. (1)i know that the result of representation (hdelta * HZ) / runtime->rate means the delta jiffies which spend on moving the data from the new_hw_ptr to the old_hw_ptr in the dma buffer. (2)and the jdelta represents the interval jiffies between now and last time enter here. (3)the question is why can we think there is more than one interrupt has lost across comparing the two variables size? (4)and if the compare idear setup that means one interrupt will take jdelta jiffies in time,so if someone is bigger it we can know that.is it true? (5)if the (4) is correct, why do not use the representation (runtime->period_size * HZ) / runtime->rate) as the interrupt interval, because it is fixed and the jdelta is not. (6)why plus the HZ/100 after jdelta and (runtime->period_size * HZ) / runtime->rate) ? for ensuring have enought time? HZ ==> 1 s HZ/1000 ===> 1 ms HZ/100 ===> 10 ms ? =================================reference code start ====================================> /* Skip the jiffies check for hardwares with BATCH flag. * Such hardware usually just increases the position at each IRQ, * thus it can't give any strange position. */if (runtime->hw.info & SNDRV_PCM_INFO_BATCH) goto no_jiffies_check; hdelta = delta; if (hdelta < runtime->delay) goto no_jiffies_check; hdelta -= runtime->delay; jdelta = jiffies - runtime->hw_ptr_jiffies; if (((hdelta * HZ) / runtime->rate) > jdelta + HZ/100) { delta = jdelta /(((runtime->period_size * HZ) / runtime->rate) + HZ/100); /* move new_hw_ptr according jiffies not pos variable */new_hw_ptr = old_hw_ptr; hw_base = delta; /* use loop to avoid checks for delta overflows *//* the delta value is small or zero in most cases */while (delta > 0) { new_hw_ptr += runtime->period_size; if (new_hw_ptr >= runtime->boundary) new_hw_ptr -= runtime->boundary; delta--; } /* align hw_base to buffer_size */hw_ptr_error(substream, "hw_ptr skipping! %s" "(pos=%ld, delta=%ld, period=%ld, " "jdelta=%lu/%lu/%lu, hw_ptr=%ld/%ld)\n", in_interrupt ? "[Q] " : "", (long)pos, (long)hdelta, (long)runtime->period_size, jdelta, ((hdelta * HZ) / runtime->rate), hw_base, (unsigned long)old_hw_ptr, (unsigned long)new_hw_ptr); /* reset values to proper state */delta = 0; hw_base = new_hw_ptr - (new_hw_ptr % runtime->buffer_size); } <======================================end =================================================== Best Regards. 在 2012-12-10 16:24:59, Clemens Ladisch <clemens(a)ladisch.de>写道: l9jj(a)163.com wrote: > does the variable buffer_size equal to DMA size? Yes. > (2)how the pos>buffer_size will happen in the interface snd_pcm_update_hw_ptr0()? It happens when your driver has a bug. The .pointer callback always must return value that is inside the DMA buffer. Regards, Clemens _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel l9jj(a)163.com ——网易邮箱手机客户端——

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[alsa-devel] [PATCH 0/2]: ALSA: VIA HDA: patch series
by lydiawang 10 Dec '12

10 Dec '12

Following patch series mainly includes: -Support codec VT1705CF. -Support codec VT1808.

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Re: [alsa-devel] what is period_size, buffer_size and boundary size, i confuse them
by l9jj＠163.com 10 Dec '12

10 Dec '12

thank you very much. but, （1）what is the relationship between DMA buffer size and the boundary size？does the variable buffer_size equal to DMA size？ (2)how the pos>buffer_size will happen in the interface snd_pcm_update_hw_ptr0()？ the variable pos is in the range of DMA_size，it can not bigger than the buffer_size.it blocked me. thanks a lot. 在 2012-12-10 00:23:55, Clemens Ladisch <clemens(a)ladisch.de>写道: l9jj wrote: > can anyone help me to figure out the concepts between period_size ,buffer_size and boundary size? > n * period_size = buffer_size? Yes. On many devices, n is not required to be an integer. > m * buffer_size = boundary_size? Yes, where m is very big. In applications, the actual value of the boundary size does not matter (except as a special value for some sw params). > and how the dma buffer mapped to the alsa buffer? The DMA buffer _is_ the ALSA buffer. Regards, Clemens _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel l9jj(a)163.com ——网易邮箱手机客户端——

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[alsa-devel] [PATCH 2/2]: ALSA: VIA HDA: Add support for codec VT1808
by lydiawang 10 Dec '12

10 Dec '12

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Add support for codec VT1808. Add support for new codec VT1808, which is similiar with VT1705CF. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound/sound/pci/hda/patch_via.c =================================================================== --- sound.orig/sound/pci/hda/patch_via.c 2012-12-10 15:09:52.000000000 +0800 +++ sound/sound/pci/hda/patch_via.c 2012-12-10 15:12:50.000000000 +0800 @@ -77,6 +77,7 @@ VT1812, VT1802, VT1705CF, + VT1808, CODEC_TYPES, }; @@ -300,6 +301,8 @@ codec_type = VT1802; else if (dev_id == 0x4760) codec_type = VT1705CF; + else if (dev_id == 0x4761 || dev_id == 0x4762) + codec_type = VT1808; else codec_type = UNKNOWN; return codec_type; @@ -1112,6 +1115,7 @@ parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ break; case VT1705CF: + case VT1808: verb = 0xf82; parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ break; @@ -4061,6 +4065,10 @@ .patch = patch_vt2002P}, { .id = 0x11064760, .name = "VT1705CF", .patch = patch_vt3476}, + { .id = 0x11064761, .name = "VT1708SCE", + .patch = patch_vt3476}, + { .id = 0x11064762, .name = "VT1808", + .patch = patch_vt3476}, {} /* terminator */ };

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[alsa-devel] [PATCH 1/2]: ALSA: VIA HDA: Add support for codec VT1705CF
by lydiawang 10 Dec '12

10 Dec '12

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Add support for codec VT1705CF. Add support for new codec VT1705CF. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound/sound/pci/hda/patch_via.c =================================================================== --- sound.orig/sound/pci/hda/patch_via.c 2012-12-10 14:22:56.000000000 +0800 +++ sound/sound/pci/hda/patch_via.c 2012-12-10 15:02:02.000000000 +0800 @@ -76,6 +76,7 @@ VT2002P, VT1812, VT1802, + VT1705CF, CODEC_TYPES, }; @@ -220,6 +221,7 @@ int vt1708_hp_present; void (*set_widgets_power_state)(struct hda_codec *codec); + unsigned int dac_stream_tag[4]; struct hda_loopback_check loopback; int num_loopbacks; @@ -296,6 +298,8 @@ codec_type = VT1708S; else if ((dev_id & 0xfff) == 0x446) codec_type = VT1802; + else if (dev_id == 0x4760) + codec_type = VT1705CF; else codec_type = UNKNOWN; return codec_type; @@ -711,6 +715,28 @@ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); } +static void update_conv_power_state(struct hda_codec *codec, hda_nid_t nid, + unsigned int parm, unsigned int index) +{ + struct via_spec *spec = codec->spec; + unsigned int format; + if (snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_POWER_STATE, 0) == parm) + return; + format = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); + if (format && (spec->dac_stream_tag[index] != format)) + spec->dac_stream_tag[index] = format; + + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); + if (parm == AC_PWRST_D0) { + format = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); + if (!format && (spec->dac_stream_tag[index] != format)) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CHANNEL_STREAMID, + spec->dac_stream_tag[index]); + } +} + static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, unsigned int *affected_parm) { @@ -1085,6 +1111,10 @@ verb = 0xf93; parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ break; + case VT1705CF: + verb = 0xf82; + parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ + break; default: return; /* other codecs are not supported */ } @@ -3817,6 +3847,125 @@ return 0; } +/* patch for vt3476 */ + +static const struct hda_verb vt3476_init_verbs[] = { + /* Enable DMic 8/16/32K */ + {0x1, 0xF7B, 0x30}, + /* Enable Boost Volume backdoor */ + {0x1, 0xFB9, 0x20}, + /* Enable AOW-MW9 path */ + {0x1, 0xFB8, 0x10}, + { } +}; + +static void set_widgets_power_state_vt3476(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm, parm2; + /* MUX10 (1eh) = stereo mixer */ + imux_is_smixer = + snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 4; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ + update_power_state(codec, 0x1e, parm); + update_power_state(codec, 0x1f, parm); + update_power_state(codec, 0x10, parm); + update_power_state(codec, 0x11, parm); + + /* outputs */ + /* PW3 (27h), MW3(37h), AOW3 (bh) */ + if (spec->codec_type == VT1705CF) { + parm = AC_PWRST_D3; + update_power_state(codec, 0x27, parm); + update_power_state(codec, 0x37, parm); + } else { + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x27, &parm); + update_power_state(codec, 0x37, parm); + } + + /* PW2 (26h), MW2(36h), AOW2 (ah) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + update_power_state(codec, 0x36, parm); + if (spec->smart51_enabled) { + /* PW7(2bh), MW7(3bh), MUX7(1Bh) */ + set_pin_power_state(codec, 0x2b, &parm); + update_power_state(codec, 0x3b, parm); + update_power_state(codec, 0x1b, parm); + } + update_conv_power_state(codec, 0xa, parm, 2); + + /* PW1 (25h), MW1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + update_power_state(codec, 0x35, parm); + if (spec->smart51_enabled) { + /* PW6(2ah), MW6(3ah), MUX6(1ah) */ + set_pin_power_state(codec, 0x2a, &parm); + update_power_state(codec, 0x3a, parm); + update_power_state(codec, 0x1a, parm); + } + update_conv_power_state(codec, 0x9, parm, 1); + + /* PW4 (28h), MW4 (38h), MUX4(18h), AOW3(bh)/AOW0(8h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + update_power_state(codec, 0x38, parm); + update_power_state(codec, 0x18, parm); + if (spec->hp_independent_mode) + update_conv_power_state(codec, 0xb, parm, 3); + parm2 = parm; /* for pin 0x0b */ + + /* PW0 (24h), MW0(34h), MW9(3fh), AOW0 (8h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + update_power_state(codec, 0x34, parm); + if (!spec->hp_independent_mode && parm2 != AC_PWRST_D3) + parm = parm2; + update_conv_power_state(codec, 0x8, parm, 0); + /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ + update_power_state(codec, 0x3f, imux_is_smixer ? AC_PWRST_D0 : parm); +} + +static int patch_vt3476(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = via_new_spec(codec); + if (spec == NULL) + return -ENOMEM; + + spec->aa_mix_nid = 0x3f; + add_secret_dac_path(codec); + + /* automatic parse from the BIOS config */ + err = via_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } + + spec->init_verbs[spec->num_iverbs++] = vt3476_init_verbs; + + codec->patch_ops = via_patch_ops; + + spec->set_widgets_power_state = set_widgets_power_state_vt3476; + + return 0; +} + /* * patch entries */ @@ -3910,6 +4059,8 @@ .patch = patch_vt2002P}, { .id = 0x11068446, .name = "VT1802", .patch = patch_vt2002P}, + { .id = 0x11064760, .name = "VT1705CF", + .patch = patch_vt3476}, {} /* terminator */ };

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[alsa-devel] [PATCH 3/3] sound/soc/soc-core.c: drop kfree of devm_kzalloc's data
by Julia Lawall 10 Dec '12

10 Dec '12

From: Julia Lawall <Julia.Lawall(a)lip6.fr> Using kfree to free data allocated with devm_kzalloc causes double frees. The semantic patch that fixes this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression x; @@ x = devm_kzalloc(...) ... ?-kfree(x); // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- The kfrees were introduced in b761c0ca. I sent this a few months ago, and I still think it should be applied... sound/soc/soc-core.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f81c597..86d7a36 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4136,7 +4136,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "Property '%s' index %d could not be read: %d\n", propname, 2 * i, ret); - kfree(routes); return -EINVAL; } ret = of_property_read_string_index(np, propname, @@ -4145,7 +4144,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "Property '%s' index %d could not be read: %d\n", propname, (2 * i) + 1, ret); - kfree(routes); return -EINVAL; } }

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[alsa-devel] [PATCH] ASoC: adsp: Correct version number masks
by Mark Brown 10 Dec '12

10 Dec '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3f0e69b..00caaa5 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -394,7 +394,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) algs = be32_to_cpu(adsp1_id.algs); adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", be32_to_cpu(adsp1_id.fw.id), - (be32_to_cpu(adsp1_id.fw.ver) & 0xff000) >> 16, + (be32_to_cpu(adsp1_id.fw.ver) & 0xff0000) >> 16, (be32_to_cpu(adsp1_id.fw.ver) & 0xff00) >> 8, be32_to_cpu(adsp1_id.fw.ver) & 0xff, algs); @@ -415,7 +415,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) algs = be32_to_cpu(adsp2_id.algs); adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", be32_to_cpu(adsp2_id.fw.id), - (be32_to_cpu(adsp2_id.fw.ver) & 0xff000) >> 16, + (be32_to_cpu(adsp2_id.fw.ver) & 0xff0000) >> 16, (be32_to_cpu(adsp2_id.fw.ver) & 0xff00) >> 8, be32_to_cpu(adsp2_id.fw.ver) & 0xff, algs); @@ -465,7 +465,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) case WMFW_ADSP1: adsp_info(dsp, "%d: ID %x v%d.%d.%d\n", i, be32_to_cpu(adsp1_alg[i].alg.id), - (be32_to_cpu(adsp1_alg[i].alg.ver) & 0xff000) >> 16, + (be32_to_cpu(adsp1_alg[i].alg.ver) & 0xff0000) >> 16, (be32_to_cpu(adsp1_alg[i].alg.ver) & 0xff00) >> 8, be32_to_cpu(adsp1_alg[i].alg.ver) & 0xff); break; @@ -473,7 +473,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) case WMFW_ADSP2: adsp_info(dsp, "%d: ID %x v%d.%d.%d\n", i, be32_to_cpu(adsp2_alg[i].alg.id), - (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff000) >> 16, + (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff0000) >> 16, (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff00) >> 8, be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff); break; -- 1.7.10.4

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[alsa-devel] [PATCH 1/3] ASoC: cs42l73: Add DMIC's as DAPM inputs.
by Paul Handrigan 10 Dec '12

10 Dec '12

Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- sound/soc/codecs/cs42l73.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 2c08c4c..4766795 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -589,6 +589,8 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = { }; static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("DMICA"), + SND_SOC_DAPM_INPUT("DMICB"), SND_SOC_DAPM_INPUT("LINEINA"), SND_SOC_DAPM_INPUT("LINEINB"), SND_SOC_DAPM_INPUT("MIC1"), @@ -795,6 +797,8 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = { {"ADC Left", NULL, "PGA Left"}, {"ADC Right", NULL, "PGA Right"}, + {"DMIC Left", NULL, "DMICA"}, + {"DMIC Right", NULL, "DMICB"}, {"Input Left Capture", "ADC Left Input", "ADC Left"}, {"Input Right Capture", "ADC Right Input", "ADC Right"}, -- 1.7.0.4

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