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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[alsa-devel] [PATCH] ASoC: sigmadsp: Fix endianness conversion issue
by Lars-Peter Clausen 10 Dec '12

10 Dec '12

The 'addr' field of the sigma_action struct is stored as big endian in the firmware file. Cc: stable(a)vger.kernel.org Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/sigmadsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 5be42bf..4068f24 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -225,7 +225,7 @@ EXPORT_SYMBOL(process_sigma_firmware); static int sigma_action_write_regmap(void *control_data, const struct sigma_action *sa, size_t len) { - return regmap_raw_write(control_data, le16_to_cpu(sa->addr), + return regmap_raw_write(control_data, be16_to_cpu(sa->addr), sa->payload, len - 2); } -- 1.8.0

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[alsa-devel] [PATCH] ASoC: pcm: allow backend hardware to be freed in pause state
by Patrick Lai 10 Dec '12

10 Dec '12

When front-end PCM session is in paused state, back-end PCM session will be put in paused state as well if given front-end PCM session is the only client of given back-end. Then, application closes front-end PCM session, DPCM framework will not allow back-end enters HW_FREE state so back-end will never get shutdown completely. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ef22d0b..d340644 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1240,7 +1240,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) continue; -- The Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation 1.7.8.3

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[alsa-devel] what is period_size, buffer_size and boundary size, i confuse them
by l9jj 09 Dec '12

09 Dec '12

can anyone help me to figure out the concepts between period_size ,buffer_size and boundary size? n * period_size = buffer_size? m * buffer_size = boundary_size? and how the dma buffer mapped to the alsa buffer? thank you . br

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Re: [alsa-devel] [PATCH 00/25] ASoC: remove __dev* attributes for retirement of HOTPLUG
by Mark Brown 09 Dec '12

09 Dec '12

On Fri, Dec 07, 2012 at 09:26:12AM -0500, Bill Pemberton wrote: > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > and __devexit so that HOTPLUG can be retired. Applied all, thanks.

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[alsa-devel] ALSA sound/usb/mixer.c:415 cannot get current value for control 2 ch 1: err = -22
by Joe Cooper 09 Dec '12

09 Dec '12

On most HP Envy laptops the snd-usb-audio module causes the system to become unresponsive and Gnome Shell 3 to crash. This effects every kernel I've tried from 3.3.4 on up to 3.7rc8. It also produces these errors on the console: ALSA sound/usb/mixer.c:415 cannot get current value for control 2 ch 1: err = -22 ALSA sound/usb/clock.c:228 3:1:1: cannot set freq 44100 to ep 0x1 Blacklisting this module allows the system to function, but with most of its speakers not working. It also prevents use of other USB audio devices, making multitrack recording impossible. I have a ticket about it in the Fedora bug tracker, but was referred to alsa-devel. That ticket links to other discussions of this bug, as well. Here's that ticket: https://bugzilla.redhat.com/show_bug.cgi?id=884368 alsa-info is here: https://bugzilla.redhat.com/attachment.cgi?id=658989 Please let me know if you need further details, or if there's anything else I can do to help troubleshoot this.

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[alsa-devel] [FT C400,PATCH RFC,v4 00/10] M-Audio Fast Track C400
by Eldad Zack 09 Dec '12

09 Dec '12

Hi, The following patches adds support for the M-Audio Fast Track C400. This is version 4 of the patch series. * Incoroprated the feedback from Clemens Ladisch <clemens(a)ladisch.de>: Moved the quirks structure to the right place (also in other places, though not strictly required) and removed the "set quirks" log messages. Thanks, Clemens! * Added a patch to address the regression introduced by commit 947d299686aa9cc8aecf749d54e8475c6e498956 , "ALSA: snd-usb: properly initialize the sync endpoint". I believe there are no other devices that affected by this issue. Opening only the playback stream without first opening the capture stream now works with the C400. v3: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-November/057477.h… v2: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-November/056750.h… This series was tested with the latest mainline tree, 3.7-rc7 (HEAD e23739b4ade80a3a7f87198f008f6c44a7cbc9fd ) Also applies against current sound-unstable (not tested). (HEAD 900b0690ee7b2a227c5db5f1642ceb67d90c49fc ) * Everything works: mixer controls, effects control, playback and capture (with the correct sampling frequency set). Note that the device doesn't really have a master control or left/right pan. It's just "emulated" in the software. * To deal with the mixer channel IDs, limited by cmask bit length, I introduced an offset value. This allows to get rid of the shorehorning of the IDs into cmask and device-referencing in the low-level functions. Is this a good approach? * Audio in/out seems to work good in this series, mainly because the playback is an implicit feedback endpoint, and it is now configured with the capture endpoint as its sync-buddy. Also needed to correct the calculation, because of the different channel numbers between the two endpoints on the C400 (6 playback - implicit feedback, 4 capture). * The clock sources are still named "Unit 129" (0x81, internal) and "Unit 130" (0x82, SPDIF). I'll fix this later. I also plan to fix the channel names (i.e., instead of "AIn3", use "SPDIF In L"). * I've touched one FTU mixer creation function. I'd appreciate if someone can test this with the FTU to make sure I didn't break something. (I haven't gotten any feedback on that one yet) Cheers, Eldad Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> Eldad Zack (10): usb-audio: replace hardcoded value with const usb-audio: correct sync endpoint initialization usb-audio: use sender stride for implicit feedback usb-audio: add control index offset usb-audio: skip UAC2 EFFECT_UNIT usb-audio: parameterize FTU effect unit control usb-audio: M-Audio Fast Track C400 quirks table usb-audio: Fast Track C400 mixer ranges usb-audio: Fast Track C400 mixer controls usb-audio: FT C400 sync playback EP to capture EP sound/usb/endpoint.c | 9 ++- sound/usb/mixer.c | 44 +++++++++- sound/usb/mixer.h | 1 + sound/usb/mixer_quirks.c | 216 ++++++++++++++++++++++++++++++++++++++++++++-- sound/usb/pcm.c | 94 ++++++++++++++++++-- sound/usb/quirks-table.h | 71 +++++++++++++++ 6 files changed, 414 insertions(+), 21 deletions(-) -- 1.7.8.6

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[alsa-devel] [asoc:topic/arizona 34/34] ERROR: "arizona_out_vi_ramp" [sound/soc/codecs/snd-soc-wm5110.ko] undefined!
by kbuild test robot 09 Dec '12

09 Dec '12

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/arizona head: 8c3e248639a0a1b15b3b245f1e2ec05150f544d5 commit: 8c3e248639a0a1b15b3b245f1e2ec05150f544d5 [34/34] ASoC: wm5110: Enable volume ramp control config: x86_64-randconfig-h424 (attached as .config) All error/warnings: ERROR: "arizona_out_vi_ramp" [sound/soc/codecs/snd-soc-wm5110.ko] undefined! ERROR: "arizona_in_vi_ramp" [sound/soc/codecs/snd-soc-wm5110.ko] undefined! ERROR: "arizona_out_vd_ramp" [sound/soc/codecs/snd-soc-wm5110.ko] undefined! ERROR: "arizona_in_vd_ramp" [sound/soc/codecs/snd-soc-wm5110.ko] undefined! --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

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[alsa-devel] [PATCH 1/3] ASoC: arizona: Add volume ramp controls
by Mark Brown 09 Dec '12

09 Dec '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 25 +++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 6 ++++++ 2 files changed, 31 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index d5385dc..a377d1f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -274,6 +274,31 @@ EXPORT_SYMBOL_GPL(arizona_mixer_values); const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0); EXPORT_SYMBOL_GPL(arizona_mixer_tlv); +static const char *arizona_vol_ramp_text[] = { + "0ms/6dB", "0.5ms/6dB", "1ms/6dB", "2ms/6dB", "4ms/6dB", "8ms/6dB", + "15ms/6dB", "30ms/6dB", +}; + +const struct soc_enum arizona_in_vd_ramp = + SOC_ENUM_SINGLE(ARIZONA_INPUT_VOLUME_RAMP, + ARIZONA_IN_VD_RAMP_SHIFT, 7, arizona_vol_ramp_text); +EXPORT_SYMBOL_GPL(arizona_in_vd_ramp); + +const struct soc_enum arizona_in_vi_ramp = + SOC_ENUM_SINGLE(ARIZONA_INPUT_VOLUME_RAMP, + ARIZONA_IN_VI_RAMP_SHIFT, 7, arizona_vol_ramp_text); +EXPORT_SYMBOL_GPL(arizona_in_vi_ramp); + +const struct soc_enum arizona_out_vd_ramp = + SOC_ENUM_SINGLE(ARIZONA_OUTPUT_VOLUME_RAMP, + ARIZONA_OUT_VD_RAMP_SHIFT, 7, arizona_vol_ramp_text); +EXPORT_SYMBOL_GPL(arizona_out_vd_ramp); + +const struct soc_enum arizona_out_vi_ramp = + SOC_ENUM_SINGLE(ARIZONA_OUTPUT_VOLUME_RAMP, + ARIZONA_OUT_VI_RAMP_SHIFT, 7, arizona_vol_ramp_text); +EXPORT_SYMBOL_GPL(arizona_out_vi_ramp); + static const char *arizona_lhpf_mode_text[] = { "Low-pass", "High-pass" }; diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 24b9692..4f09bb5 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -165,6 +165,12 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; ARIZONA_MIXER_ROUTES(name, name "L"), \ ARIZONA_MIXER_ROUTES(name, name "R") +extern const struct soc_enum arizona_in_vi_ramp; +extern const struct soc_enum arizona_in_vd_ramp; + +extern const struct soc_enum arizona_out_vi_ramp; +extern const struct soc_enum arizona_out_vd_ramp; + extern const struct soc_enum arizona_lhpf1_mode; extern const struct soc_enum arizona_lhpf2_mode; extern const struct soc_enum arizona_lhpf3_mode; -- 1.7.10.4

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Re: [alsa-devel] [PATCH] ASoC: adsp: Set DSP clock less than SYSCLK at initialisation
by Mark Brown 09 Dec '12

09 Dec '12

On Fri, Dec 07, 2012 at 01:55:22PM +0000, Chris Rattray wrote: > In some configurations it is necessary to set the ADSP core clock > to less than the SYSCLK rate in order for the ADSP RAM to initialise > successfully. Once the firmware image is downloaded, the ADSP core > clock is then brought back up to the SYSCLK rate. As followed up offline this is a system-specific issue you were running into that this happened to work around. > Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> > Change-Id: I14412a301dcf1c0a6a728fa8c12d23715e944cd4 Don't include noise like Change-id that's specific to internal systems in upstream submissions.

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Re: [alsa-devel] [PATCH 03/20] ALSA: pxa2xx: remove __dev* attributes
by Haojian Zhuang 08 Dec '12

08 Dec '12

On Fri, Dec 7, 2012 at 1:35 AM, Bill Pemberton <wfp5p(a)virginia.edu> wrote: > CONFIG_HOTPLUG is going away as an option. As result the __dev* > markings will be going away. > > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > and __devexit. > > Signed-off-by: Bill Pemberton <wfp5p(a)virginia.edu> > Cc: Eric Miao <eric.y.miao(a)gmail.com> > Cc: Russell King <linux(a)arm.linux.org.uk> > Cc: Haojian Zhuang <haojian.zhuang(a)gmail.com> > Cc: linux-arm-kernel(a)lists.infradead.org > --- > sound/arm/pxa2xx-ac97-lib.c | 2 +- > sound/arm/pxa2xx-ac97.c | 6 +++--- > 2 files changed, 4 insertions(+), 4 deletions(-) > > diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c > index 48d7c0a..6fc0ae9 100644 > --- a/sound/arm/pxa2xx-ac97-lib.c > +++ b/sound/arm/pxa2xx-ac97-lib.c > @@ -314,7 +314,7 @@ int pxa2xx_ac97_hw_resume(void) > EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume); > #endif > > -int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) > +int pxa2xx_ac97_hw_probe(struct platform_device *dev) > { > int ret; > pxa2xx_audio_ops_t *pdata = dev->dev.platform_data; > diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c > index 4e1fda7..ec54be4 100644 > --- a/sound/arm/pxa2xx-ac97.c > +++ b/sound/arm/pxa2xx-ac97.c > @@ -163,7 +163,7 @@ static int pxa2xx_ac97_resume(struct device *dev) > static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, pxa2xx_ac97_suspend, pxa2xx_ac97_resume); > #endif > > -static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) > +static int pxa2xx_ac97_probe(struct platform_device *dev) > { > struct snd_card *card; > struct snd_ac97_bus *ac97_bus; > @@ -224,7 +224,7 @@ err_dev: > return ret; > } > > -static int __devexit pxa2xx_ac97_remove(struct platform_device *dev) > +static int pxa2xx_ac97_remove(struct platform_device *dev) > { > struct snd_card *card = platform_get_drvdata(dev); > > @@ -239,7 +239,7 @@ static int __devexit pxa2xx_ac97_remove(struct platform_device *dev) > > static struct platform_driver pxa2xx_ac97_driver = { > .probe = pxa2xx_ac97_probe, > - .remove = __devexit_p(pxa2xx_ac97_remove), > + .remove = pxa2xx_ac97_remove, > .driver = { > .name = "pxa2xx-ac97", > .owner = THIS_MODULE, > -- > 1.8.0.1 > Acked-by: Haojian Zhuang <haojian.zhuang(a)gmail.com>

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Re: [alsa-devel] [PATCH 05/25] ASoC: pxa: remove __dev* attributes
by Haojian Zhuang 08 Dec '12

08 Dec '12

On Fri, Dec 7, 2012 at 10:26 PM, Bill Pemberton <wfp5p(a)virginia.edu> wrote: > CONFIG_HOTPLUG is going away as an option. As result the __dev* > markings will be going away. > > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > and __devexit. > > Signed-off-by: Bill Pemberton <wfp5p(a)virginia.edu> > Cc: Eric Miao <eric.y.miao(a)gmail.com> > Cc: Russell King <linux(a)arm.linux.org.uk> > Cc: Haojian Zhuang <haojian.zhuang(a)gmail.com> > Cc: linux-arm-kernel(a)lists.infradead.org > --- > sound/soc/pxa/brownstone.c | 6 +++--- > sound/soc/pxa/corgi.c | 6 +++--- > sound/soc/pxa/e740_wm9705.c | 6 +++--- > sound/soc/pxa/e750_wm9705.c | 6 +++--- > sound/soc/pxa/e800_wm9712.c | 6 +++--- > sound/soc/pxa/imote2.c | 6 +++--- > sound/soc/pxa/mioa701_wm9713.c | 6 +++--- > sound/soc/pxa/mmp-pcm.c | 6 +++--- > sound/soc/pxa/mmp-sspa.c | 6 +++--- > sound/soc/pxa/palm27x.c | 4 ++-- > sound/soc/pxa/poodle.c | 6 +++--- > sound/soc/pxa/pxa-ssp.c | 6 +++--- > sound/soc/pxa/pxa2xx-ac97.c | 8 ++++---- > sound/soc/pxa/pxa2xx-i2s.c | 4 ++-- > sound/soc/pxa/pxa2xx-pcm.c | 6 +++--- > sound/soc/pxa/tosa.c | 6 +++--- > sound/soc/pxa/ttc-dkb.c | 6 +++--- > 17 files changed, 50 insertions(+), 50 deletions(-) > Acked-by: Haojian Zhuang <haojian.zhuang(a)gmail.com>

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[alsa-devel] [patch] usb6fire: prevent driver panic state when stopping
by Jurgen Kramer 07 Dec '12

07 Dec '12

Hi, The patch below prevents the 6fire usb driver going into panic state when stopping playing. On some systems the urb in handler (usb6fire_pcm_in_urb_handler) is being called while urbs are being killed off, this causes the driver to set panic state and can result in the kernel warning 'URB %p submitted while active'. Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> diff -uNrp linux/sound/usb/6fire/pcm.c linux.new/sound/usb/6fire/pcm.c --- linux/sound/usb/6fire/pcm.c 2012-11-23 17:01:41.754673597 +0100 +++ linux.new/sound/usb/6fire/pcm.c 2012-12-07 10:09:11.386666744 +0100 @@ -135,6 +135,9 @@ static void usb6fire_pcm_stream_stop(str struct control_runtime *ctrl_rt = rt->chip->control; if (rt->stream_state != STREAM_DISABLED) { + + rt->stream_state = STREAM_STOPPING; + for (i = 0; i < PCM_N_URBS; i++) { usb_kill_urb(&rt->in_urbs[i].instance); usb_kill_urb(&rt->out_urbs[i].instance); Jurgen

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[alsa-devel] M-Audio Audiophile 24/96 (snd-ice1712) lacks ADCs
by Pils Bert 07 Dec '12

07 Dec '12

Hi guys, I have a problem with my M-Audio Audiophile 24/96 soundcard: There are no ADC volume sliders present neither in alsamixer nor in envy24control. But instead there are also DAC sliders on the capture page in alsamixer?! But changing them doesn't have an effect on the H/W input gain. I read on website that some other users had the same problem (but it seams to be a rare problem). First of all my card has been correctly detected as Audiophile 24/96. But in /sys/module/snd_ice1712/parameters/model are only null values. I loaded the module with the model=audiophile parameter, but no ADCs appear. Second I skimmed over the module source code and found the lines case ICE1712_SUBDEVICE_AUDIOPHILE: ice->num_total_dacs = 2; ice->num_total_adcs = 2; break; in __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) in delta.c So the module says there have to be 2 ADCs... I also disabled my onboard soundcard (HDA Intel), but no ADCs appear. Maybe its some hardware configuration problem? Moreover, I had to edit my .asoundrc file in order to play and capture audio anyway (so no 'out of the box' experience here...): pcm.audiophile { type hw card 1 device 0 } pcm.analog { type plug slave.pcm audiophile ttable.0.0 1 # H/W 1 ttable.1.1 1 # H/W 2 } I'm using plain ALSA 1.0.25 (without pulseaudio) on Debian Testing (Linux Rowdy 3.2.0-4-amd64 #1 SMP Debian 3.2.32-1 x86_64 GNU/Linux) Maybe some of you guys have an good idea ;) Best regards, Jannik

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Re: [alsa-devel] [PATCH RESEND V2 1/2] ASoC: Samsung: Do not register samsung audio dma device as pdev
by Mark Brown 07 Dec '12

07 Dec '12

On Fri, Dec 07, 2012 at 01:59:21PM +0530, Padmavathi Venna wrote: > Previously, the ASoC 'platform' (PCM/DMA) object was instantiated via a > platform_device. This didn't represent the hardware well, since there > was no separate hardware associated with this platform_device; it was a > virtual device with sole purpose to call snd_soc_register_platform(). Applied, thanks.

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Re: [alsa-devel] [PATCH 00/20] ALSA: remove __dev* attributes for retirement of HOTPLUG
by Takashi Iwai 07 Dec '12

07 Dec '12

At Thu, 06 Dec 2012 12:45:56 -0500, Bill Pemberton wrote: > > Takashi Iwai writes: > > > > At Thu, 6 Dec 2012 12:35:09 -0500, > > Bill Pemberton wrote: > > > > > > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > > > and __devexit so that HOTPLUG can be retired. > > > > > > Bill Pemberton (20): > > > ALSA: pci: remove __dev* attributes > > > ALSA: AACI: remove __dev* attributes > > > ALSA: pxa2xx: remove __dev* attributes > > > ALSA: atmel: remove __dev* attributes > > > ALSA: firewire-speakers: remove __dev* attributes > > > ALSA: mips: remove __dev* attributes > > > ALSA: parisc/harmony: remove __dev* attributes > > > ALSA: oss: remove __dev* attributes > > > ALSA: ad1889: remove __dev* attributes > > > ALSA: bt87X: remove __dev* attributes > > > ALSA: oxygen: remove __dev* attributes > > > > Any reason the three above to be separate patches? > > > > They are separately listed in MAINTAINERS, so I broke them out for the > appropriate Cc:'s. Ah, understood. In general, split isn't needed for a case like this. Most of such trivial cleanup patches are applied at once to the relevant codes in the whole subsystem. In anyway, I applied all patches now. thanks, Takashi

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[alsa-devel] Compilation error for static build of alsa-lib
by Pankaj DEV 07 Dec '12

07 Dec '12

When building alsa (using buildroot) for static build, compilation errors occur. Don't know how to report bug, since the "bug tracker" link on web-page doesn't open The following patch solves the issue : --- a/modules/mixer/simple/sbasedl.c +++ b/modules/mixer/simple/sbasedl.c @@ -27,7 +27,11 @@ #include <fcntl.h> #include <sys/ioctl.h> #include <math.h> +#ifdef HAVE_LIBDL #include <dlfcn.h> +#else +#define RTLD_NOW 0 +#endif #include "config.h" #include "asoundlib.h" #include "mixer_abst.h" diff --git a/src/mixer/simple_abst.c b/src/mixer/simple_abst.c index 9e9aaf5..03070e0 100644 --- a/src/mixer/simple_abst.c +++ b/src/mixer/simple_abst.c @@ -34,7 +34,12 @@ #include <fcntl.h> #include <sys/ioctl.h> #include <math.h> +#ifdef HAVE_LIBDL #include <dlfcn.h> +#else +#define RTLD_NOW 0 +#define RTLD_GLOBAL 8 +#endif #include "config.h" #include "asoundlib.h" #include "mixer_simple.h"

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Re: [alsa-devel] [PATCH V2 1/2] ASoC: Samsung: Do not register samsung audio dma device as pdev
by Mark Brown 07 Dec '12

07 Dec '12

On Thu, Dec 06, 2012 at 09:20:36AM +0530, Padmavathi Venna wrote: > Previously, the ASoC 'platform' (PCM/DMA) object was instantiated via a > platform_device. This didn't represent the hardware well, since there > was no separate hardware associated with this platform_device; it was a > virtual device with sole purpose to call snd_soc_register_platform(). Looks good however this doesn't apply against my topic/samsung branch - can you please check if I need anything from Kukjin's tree? Though I guess the easiest thing is just to wait until the merge window is over, we're near to the next release.

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Re: [alsa-devel] Compilation error for static build of alsa-lib
by Pankaj DEV 07 Dec '12

07 Dec '12

Forgot to mention the version 1.0.26 From: Pankaj DEV Sent: Friday, December 07, 2012 12:01 PM To: alsa-devel(a)alsa-project.org Subject: Compilation error for static build of alsa-lib When building alsa (using buildroot) for static build, compilation errors occur. Don't know how to report bug, since the "bug tracker" link on web-page doesn't open The following patch solves the issue : --- a/modules/mixer/simple/sbasedl.c +++ b/modules/mixer/simple/sbasedl.c @@ -27,7 +27,11 @@ #include <fcntl.h> #include <sys/ioctl.h> #include <math.h> +#ifdef HAVE_LIBDL #include <dlfcn.h> +#else +#define RTLD_NOW 0 +#endif #include "config.h" #include "asoundlib.h" #include "mixer_abst.h" diff --git a/src/mixer/simple_abst.c b/src/mixer/simple_abst.c index 9e9aaf5..03070e0 100644 --- a/src/mixer/simple_abst.c +++ b/src/mixer/simple_abst.c @@ -34,7 +34,12 @@ #include <fcntl.h> #include <sys/ioctl.h> #include <math.h> +#ifdef HAVE_LIBDL #include <dlfcn.h> +#else +#define RTLD_NOW 0 +#define RTLD_GLOBAL 8 +#endif #include "config.h" #include "asoundlib.h" #include "mixer_simple.h"

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[alsa-devel] Compilation error for static build of alsa-utils 1.0.26
by Pankaj DEV 07 Dec '12

07 Dec '12

When building alsa (using buildroot) for static build, compilation errors occur. Don't know how to report bug, since the "bug tracker" link on web-page doesn't open The following patch solves the issue : --- a/configure +++ b/configure @@ -6247,7 +6247,7 @@ if test "$alsa_prefix" != "" ; then LDFLAGS="$LDFLAGS $ALSA_LIBS" fi -ALSA_LIBS="$ALSA_LIBS -lasound -lm -ldl -lpthread" +ALSA_LIBS="$ALSA_LIBS -lasound -lm -lpthread" LIBS="$ALSA_LIBS $LIBS" { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ALSA_LIBS" >&5 $as_echo "$ALSA_LIBS" >&6; }

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[alsa-devel] [PATCH v2 1/3] ALSA: ASoC: McASP: remove unused variables
by Daniel Mack 07 Dec '12

07 Dec '12

codec_fmt and sample_rate variables are unused in both snd_platform_data and davinci_audio_dev, so drop them. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- include/linux/platform_data/davinci_asp.h | 1 - sound/soc/davinci/davinci-mcasp.c | 1 - sound/soc/davinci/davinci-mcasp.h | 2 -- 3 files changed, 4 deletions(-) diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index f3d6e4f..8db5ae0 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -23,7 +23,6 @@ struct snd_platform_data { u32 rx_dma_offset; int asp_chan_q; /* event queue number for ASP channel */ int ram_chan_q; /* event queue number for RAM channel */ - unsigned int codec_fmt; /* * Allowing this is more efficient and eliminates left and right swaps * caused by underruns, but will swap the left and right channels diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 46350b8..023a70d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1174,7 +1174,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->tdm_slots = pdata->tdm_slots; dev->num_serializer = pdata->num_serializer; dev->serial_dir = pdata->serial_dir; - dev->codec_fmt = pdata->codec_fmt; dev->version = pdata->version; dev->txnumevt = pdata->txnumevt; dev->rxnumevt = pdata->rxnumevt; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 156f15f..a42b5d9 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -40,9 +40,7 @@ enum { struct davinci_audio_dev { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; - int sample_rate; struct device *dev; - unsigned int codec_fmt; /* McASP specific data */ int tdm_slots; -- 1.7.11.7

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[alsa-devel] [PATCH] ASoC: omap-twl4030: Update the header file to support more boards
by Peter Ujfalusi 07 Dec '12

07 Dec '12

The common machine driver will be able to support new boards where the voice port is also in use. At the same time add possibility to fine tune the connections from twl4030 on the board: jack detection GPIO, input/output selection Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, Liam Tony, I just finalized a series which I intend to send for 3.9: It is going to update the omap-twl4030 ASoC machine driver to support more boards, thus allowing me to remove the driver for zoom2 and sdp3440. The issue is that in order to do it I need to make modification in arch/arm/mach-omap2/ and sound/soc/omap/ as well. One patch which changes the header file is the only compile time requirement for the changes. As a reference this is how my tree looks like regarding to this change (need to sort them still), git log --oneline : 1702213 ASoC: OMAP: Remove obsolete machine drivers for Zoom2 and SDP3403 41ff1a6 ARM: OMAP: sdp3430: Audio support via the common omap-twl4030 machine driver f51363ed ARM: OMAP: zoom: Audio support via the common omap-twl4030 machine driver 159e443 ASoC: omap-twl4030: Add support for routing, voice port and jack detect 77bda45 ARM: OMAP2+: twl-common: Allow boards to customize the twl4030 audio setup 0db92ac ARM: OMAP2+: twl-common: Add default twl4030 audio configuration 9005d5b ARM: OMAP: zoom: Zoom2 does not have extmute functionality c7ffd18 ASoC: sdp3430: No need to configure pin mux for extmute 43dc9d9 ASoC: twl4030: Configure extmute pinmux when the dedicated pin is in use db9207c ARM: OMAP: 3430sdp: Enable extmute functionality for audio e6aea6f ASoC: twl4030: Convert MICBIAS to SUPPLY widget 8eadb74 ASoC: sdp3430: No need to configure the Voice port anymore f11c1be ASoC: zoom2: No need to configure the Voice port anymore 63da45d ASoC: twl4030: Correct the support for Voice port a98e934 ASoC: omap-twl4030: Update the header file to support more boards I know it is quite late already, but Mark would you be able the squeeze in this patch for 3.8? If this is in, I can separate the arch/arm/mach-omap2/ and sound/soc/omap/ patches and queue them via the appropriate tree without creating cross tree merge issues (yes, again cross tree issue). All of these will come together for 3.9 and in linux-next. This header change does not affect the current driver since the new parameters are ignored by the current driver Mark, would you be able to do this? Thank you, Peter include/linux/platform_data/omap-twl4030.h | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/include/linux/platform_data/omap-twl4030.h b/include/linux/platform_data/omap-twl4030.h index c7bef78..ee60ef7 100644 --- a/include/linux/platform_data/omap-twl4030.h +++ b/include/linux/platform_data/omap-twl4030.h @@ -25,8 +25,34 @@ #ifndef _OMAP_TWL4030_H_ #define _OMAP_TWL4030_H_ +/* To select if only one channel is connected in a stereo port */ +#define OMAP_TWL4030_LEFT (1 << 0) +#define OMAP_TWL4030_RIGHT (1 << 1) + struct omap_tw4030_pdata { const char *card_name; + /* Voice port is connected to McBSP3 */ + bool voice_connected; + + /* The driver will parse the connection flags if this flag is set */ + bool custom_routing; + /* Flags to indicate connected audio ports. */ + u8 has_hs; + u8 has_hf; + u8 has_predriv; + u8 has_carkit; + bool has_ear; + + bool has_mainmic; + bool has_submic; + bool has_hsmic; + bool has_carkitmic; + bool has_digimic0; + bool has_digimic1; + u8 has_linein; + + /* Jack detect GPIO or <= 0 if it is not implemented */ + int jack_detect; }; #endif /* _OMAP_TWL4030_H_ */ -- 1.8.0

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[alsa-devel] [PATCH] ASoC: arizona: Log the clock we're setting the DAI to use
by Mark Brown 07 Dec '12

07 Dec '12

Useful for diagnostics. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a7eb553..d5385dc 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -811,6 +811,9 @@ static int arizona_dai_set_sysclk(struct snd_soc_dai *dai, return -EBUSY; } + dev_dbg(codec->dev, "Setting AIF%d to %s\n", dai->id + 1, + arizona_dai_clk_str(clk_id)); + memset(&routes, 0, sizeof(routes)); routes[0].sink = dai->driver->capture.stream_name; routes[1].sink = dai->driver->playback.stream_name; -- 1.7.10.4

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Re: [alsa-devel] I would like to resubmit an ALSA device driver for the Maxim 98090 Codec...
by Mark Brown 07 Dec '12

07 Dec '12

On Thu, Dec 06, 2012 at 05:14:50PM -0800, Jerry Wong wrote: > Here is the resubmission of the sound/soc/max98090.c driver with the changes you suggested including regmap I/O and some stylistic updates. Please fix your mailer to word wrap within paragraphs. Please also follow the submission procedure in SubmittingPatches, as previously mentioned this is essential especially the Signed-off-by bit. I really can't apply or even properly review something that doesn't follow that. Looking at the code you want to check the register defaults - defaults for volatile registers don't make a huge amount of sense. For the ranges based on bases use ... or if (else) rather than listing the individual registers by hand. For the custom get and sets I can't see why you're not able to use generic controls, if you aren't there's enough of these that you should add a generic type. Gains should always be volume controls with TLV, enable/disable stuff should be Switch and controls called Gain Volume don't make a lot of sense (just Volume). Prints that just log the build time aren't helpful, you'll notice no other mainline drivers do that... there's already a kernel level build time. Similarly other stuff like interrupt numbers that are getting logged are a bit noisy. Build time options for things like performance need to be made runtime or platform data based. Are you sure you want the bandgap reference running the entire time the system is powered up?

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Re: [alsa-devel] [PATCH 05/20] ALSA: firewire-speakers: remove __dev* attributes
by Clemens Ladisch 06 Dec '12

06 Dec '12

Bill Pemberton wrote: > CONFIG_HOTPLUG is going away as an option. As result the __dev* > markings will be going away. > > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > and __devexit. > > Signed-off-by: Bill Pemberton <wfp5p(a)virginia.edu> > Cc: Clemens Ladisch <clemens(a)ladisch.de> Acked-by: Clemens Ladisch <clemens(a)ladisch.de> for all these (5,10,11): > sound/firewire/speakers.c | 8 ++++---- > sound/pci/bt87x.c | 22 +++++++++++----------- > sound/pci/oxygen/oxygen.c | 10 +++++----- > sound/pci/oxygen/virtuoso.c | 10 +++++----- > sound/pci/oxygen/xonar_cs43xx.c | 4 ++-- > sound/pci/oxygen/xonar_pcm179x.c | 4 ++-- > sound/pci/oxygen/xonar_wm87x6.c | 4 ++--

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[alsa-devel] How to mix two PCM channels with ALSA
by itxEpsilon 06 Dec '12

06 Dec '12

Dear Alsa developers: I'm trying to write a simple application for mixing two PCM sound channels in C++. Could anyone provide me a quick example about how to implement this operation? Thank you in advance. Itx

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Re: [alsa-devel] [PATCH 00/20] ALSA: remove __dev* attributes for retirement of HOTPLUG
by Takashi Iwai 06 Dec '12

06 Dec '12

At Thu, 6 Dec 2012 12:35:09 -0500, Bill Pemberton wrote: > > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > and __devexit so that HOTPLUG can be retired. > > Bill Pemberton (20): > ALSA: pci: remove __dev* attributes > ALSA: AACI: remove __dev* attributes > ALSA: pxa2xx: remove __dev* attributes > ALSA: atmel: remove __dev* attributes > ALSA: firewire-speakers: remove __dev* attributes > ALSA: mips: remove __dev* attributes > ALSA: parisc/harmony: remove __dev* attributes > ALSA: oss: remove __dev* attributes > ALSA: ad1889: remove __dev* attributes > ALSA: bt87X: remove __dev* attributes > ALSA: oxygen: remove __dev* attributes Any reason the three above to be separate patches? > ALSA: isa: remove __dev* attributes > ALSA: sound/ps3: remove __dev* attributes > ALSA: ppc: remove __dev* attributes > ALSA: sh: remove __dev* attributes > ALSA: sparc: remove __dev* attributes > ALSA: at73c213: remove __dev* attributes > ALSA: drivers: remove __dev* attributes > ALSA: snd-usb-6fire: remove __dev* attributes > ALSA: snd-usb-caiaq: remove __dev* attributes Also, these can be a single clean up for sound/usb/*... It's no big matter, but I just wonder since I don't see a logic... Takashi > sound/arm/aaci.c | 18 +++--- > sound/arm/pxa2xx-ac97-lib.c | 2 +- > sound/arm/pxa2xx-ac97.c | 6 +- > sound/atmel/abdac.c | 8 +-- > sound/atmel/ac97c.c | 10 ++-- > sound/drivers/aloop.c | 16 +++--- > sound/drivers/dummy.c | 14 ++--- > sound/drivers/ml403-ac97cr.c | 10 ++-- > sound/drivers/mpu401/mpu401.c | 18 +++--- > sound/drivers/mtpav.c | 14 ++--- > sound/drivers/mts64.c | 40 +++++++------- > sound/drivers/pcsp/pcsp.c | 14 ++--- > sound/drivers/pcsp/pcsp_input.c | 2 +- > sound/drivers/pcsp/pcsp_input.h | 2 +- > sound/drivers/pcsp/pcsp_lib.c | 2 +- > sound/drivers/pcsp/pcsp_mixer.c | 10 ++-- > sound/drivers/portman2x4.c | 18 +++--- > sound/drivers/serial-u16550.c | 32 +++++------ > sound/drivers/virmidi.c | 6 +- > sound/firewire/speakers.c | 8 +-- > sound/isa/ad1816a/ad1816a.c | 18 +++--- > sound/isa/ad1816a/ad1816a_lib.c | 19 ++++--- > sound/isa/ad1848/ad1848.c | 8 +-- > sound/isa/adlib.c | 8 +-- > sound/isa/als100.c | 22 ++++---- > sound/isa/azt2320.c | 26 ++++----- > sound/isa/cmi8328.c | 8 +-- > sound/isa/cmi8330.c | 42 +++++++------- > sound/isa/cs423x/cs4231.c | 8 +-- > sound/isa/cs423x/cs4236.c | 50 ++++++++--------- > sound/isa/es1688/es1688.c | 28 +++++----- > sound/isa/es18xx.c | 80 +++++++++++++-------------- > sound/isa/galaxy/galaxy.c | 26 ++++----- > sound/isa/gus/gusclassic.c | 15 ++--- > sound/isa/gus/gusextreme.c | 24 ++++---- > sound/isa/gus/gusmax.c | 16 +++--- > sound/isa/gus/interwave.c | 54 +++++++++--------- > sound/isa/msnd/msnd.h | 2 +- > sound/isa/msnd/msnd_pinnacle.c | 44 +++++++-------- > sound/isa/msnd/msnd_pinnacle_mixer.c | 2 +- > sound/isa/opl3sa2.c | 40 +++++++------- > sound/isa/opti9xx/miro.c | 68 +++++++++++------------ > sound/isa/opti9xx/opti92x-ad1848.c | 42 +++++++------- > sound/isa/sb/emu8000.c | 22 ++++---- > sound/isa/sb/jazz16.c | 18 +++--- > sound/isa/sb/sb16.c | 26 ++++----- > sound/isa/sb/sb8.c | 8 +-- > sound/isa/sc6000.c | 38 ++++++------- > sound/isa/sscape.c | 32 +++++------ > sound/isa/wavefront/wavefront.c | 53 +++++++++--------- > sound/isa/wavefront/wavefront_fx.c | 2 +- > sound/isa/wavefront/wavefront_midi.c | 2 +- > sound/isa/wavefront/wavefront_synth.c | 14 ++--- > sound/mips/au1x00.c | 4 +- > sound/mips/hal2.c | 14 ++--- > sound/mips/sgio2audio.c | 28 +++++----- > sound/oss/ad1848.c | 2 +- > sound/oss/kahlua.c | 10 ++-- > sound/parisc/harmony.c | 12 ++-- > sound/pci/ad1889.c | 18 +++--- > sound/pci/ak4531_codec.c | 10 ++-- > sound/pci/ali5451/ali5451.c | 32 +++++------ > sound/pci/als300.c | 14 ++--- > sound/pci/als4000.c | 12 ++-- > sound/pci/asihpi/asihpi.c | 55 +++++++++---------- > sound/pci/asihpi/hpioctl.c | 6 +- > sound/pci/asihpi/hpioctl.h | 6 +- > sound/pci/atiixp.c | 32 +++++------ > sound/pci/atiixp_modem.c | 20 +++---- > sound/pci/au88x0/au88x0.c | 10 ++-- > sound/pci/au88x0/au88x0_a3d.c | 6 +- > sound/pci/au88x0/au88x0_core.c | 2 +- > sound/pci/au88x0/au88x0_eq.c | 10 ++-- > sound/pci/au88x0/au88x0_game.c | 2 +- > sound/pci/au88x0/au88x0_mixer.c | 2 +- > sound/pci/au88x0/au88x0_mpu401.c | 2 +- > sound/pci/au88x0/au88x0_pcm.c | 6 +- > sound/pci/aw2/aw2-alsa.c | 28 +++++----- > sound/pci/azt3328.c | 22 ++++---- > sound/pci/bt87x.c | 22 ++++---- > sound/pci/ca0106/ca0106_main.c | 12 ++-- > sound/pci/ca0106/ca0106_mixer.c | 26 ++++----- > sound/pci/ca0106/ca0106_proc.c | 2 +- > sound/pci/ca0106/ca_midi.c | 2 +- > sound/pci/cmipci.c | 48 ++++++++-------- > sound/pci/cs4281.c | 30 +++++----- > sound/pci/cs46xx/cs46xx.c | 8 +-- > sound/pci/cs46xx/cs46xx_lib.c | 35 ++++++------ > sound/pci/cs5530.c | 16 +++--- > sound/pci/cs5535audio/cs5535audio.c | 18 +++--- > sound/pci/cs5535audio/cs5535audio.h | 10 ++-- > sound/pci/cs5535audio/cs5535audio_olpc.c | 10 ++-- > sound/pci/cs5535audio/cs5535audio_pcm.c | 2 +- > sound/pci/ctxfi/ctatc.c | 20 +++---- > sound/pci/ctxfi/ctatc.h | 8 +-- > sound/pci/ctxfi/cthardware.c | 4 +- > sound/pci/ctxfi/cthw20k1.c | 4 +- > sound/pci/ctxfi/cthw20k2.c | 4 +- > sound/pci/ctxfi/xfi.c | 6 +- > sound/pci/echoaudio/echoaudio.c | 46 ++++++++-------- > sound/pci/echoaudio/echoaudio.h | 4 +- > sound/pci/echoaudio/midi.c | 4 +- > sound/pci/emu10k1/emu10k1.c | 8 +-- > sound/pci/emu10k1/emu10k1_main.c | 4 +- > sound/pci/emu10k1/emu10k1x.c | 29 +++++----- > sound/pci/emu10k1/emufx.c | 25 +++++---- > sound/pci/emu10k1/emumixer.c | 22 ++++---- > sound/pci/emu10k1/emumpu401.c | 6 +- > sound/pci/emu10k1/emupcm.c | 11 ++-- > sound/pci/emu10k1/emuproc.c | 2 +- > sound/pci/emu10k1/p16v.c | 8 +-- > sound/pci/emu10k1/timer.c | 2 +- > sound/pci/ens1370.c | 52 +++++++++--------- > sound/pci/es1938.c | 20 +++---- > sound/pci/es1968.c | 42 +++++++------- > sound/pci/fm801.c | 26 ++++----- > sound/pci/hda/hda_intel.c | 48 ++++++++-------- > sound/pci/ice1712/amp.c | 6 +- > sound/pci/ice1712/aureon.c | 24 ++++---- > sound/pci/ice1712/delta.c | 42 +++++++------- > sound/pci/ice1712/ews.c | 28 +++++----- > sound/pci/ice1712/hoontech.c | 26 ++++----- > sound/pci/ice1712/ice1712.c | 94 ++++++++++++++++---------------- > sound/pci/ice1712/ice1724.c | 64 +++++++++++----------- > sound/pci/ice1712/juli.c | 26 ++++----- > sound/pci/ice1712/maya44.c | 16 +++--- > sound/pci/ice1712/phase.c | 22 ++++---- > sound/pci/ice1712/pontis.c | 10 ++-- > sound/pci/ice1712/prodigy192.c | 12 ++-- > sound/pci/ice1712/prodigy_hifi.c | 20 +++---- > sound/pci/ice1712/psc724.c | 8 +-- > sound/pci/ice1712/quartet.c | 24 ++++---- > sound/pci/ice1712/revo.c | 28 +++++----- > sound/pci/ice1712/se.c | 20 +++---- > sound/pci/ice1712/vt1720_mobo.c | 10 ++-- > sound/pci/ice1712/wtm.c | 10 ++-- > sound/pci/intel8x0.c | 56 +++++++++---------- > sound/pci/intel8x0m.c | 30 +++++----- > sound/pci/korg1212/korg1212.c | 12 ++-- > sound/pci/lola/lola.c | 14 ++--- > sound/pci/lola/lola_clock.c | 2 +- > sound/pci/lola/lola_mixer.c | 32 +++++------ > sound/pci/lola/lola_pcm.c | 4 +- > sound/pci/lola/lola_proc.c | 2 +- > sound/pci/lx6464es/lx6464es.c | 30 +++++----- > sound/pci/lx6464es/lx_core.c | 2 +- > sound/pci/lx6464es/lx_core.h | 2 +- > sound/pci/maestro3.c | 24 ++++---- > sound/pci/mixart/mixart.c | 12 ++-- > sound/pci/nm256/nm256.c | 18 +++--- > sound/pci/oxygen/oxygen.c | 10 ++-- > sound/pci/oxygen/virtuoso.c | 10 ++-- > sound/pci/oxygen/xonar_cs43xx.c | 4 +- > sound/pci/oxygen/xonar_pcm179x.c | 4 +- > sound/pci/oxygen/xonar_wm87x6.c | 4 +- > sound/pci/pcxhr/pcxhr.c | 14 ++--- > sound/pci/riptide/riptide.c | 20 +++---- > sound/pci/rme32.c | 10 ++-- > sound/pci/rme96.c | 14 ++--- > sound/pci/rme9652/hdsp.c | 14 ++--- > sound/pci/rme9652/hdspm.c | 41 +++++++------- > sound/pci/rme9652/rme9652.c | 22 ++++---- > sound/pci/sis7019.c | 16 +++--- > sound/pci/sonicvibes.c | 37 +++++++------ > sound/pci/trident/trident.c | 8 +-- > sound/pci/trident/trident_main.c | 46 ++++++++-------- > sound/pci/via82xx.c | 50 ++++++++--------- > sound/pci/via82xx_modem.c | 26 ++++----- > sound/pci/vx222/vx222.c | 14 ++--- > sound/pci/ymfpci/ymfpci.c | 12 ++-- > sound/pci/ymfpci/ymfpci_main.c | 44 ++++++++------- > sound/ppc/awacs.c | 54 +++++++++--------- > sound/ppc/beep.c | 2 +- > sound/ppc/burgundy.c | 22 ++++---- > sound/ppc/daca.c | 2 +- > sound/ppc/keywest.c | 4 +- > sound/ppc/pmac.c | 12 ++-- > sound/ppc/powermac.c | 6 +- > sound/ppc/snd_ps3.c | 12 ++-- > sound/ppc/tumbler.c | 16 +++--- > sound/sh/aica.c | 13 ++--- > sound/sh/sh_dac_audio.c | 10 ++-- > sound/sparc/amd7930.c | 16 +++--- > sound/sparc/cs4231.c | 38 ++++++------- > sound/sparc/dbri.c | 28 +++++----- > sound/spi/at73c213.c | 20 +++---- > sound/usb/6fire/chip.c | 4 +- > sound/usb/6fire/comm.c | 2 +- > sound/usb/6fire/comm.h | 2 +- > sound/usb/6fire/control.c | 8 +-- > sound/usb/6fire/control.h | 2 +- > sound/usb/6fire/firmware.h | 2 +- > sound/usb/6fire/midi.c | 2 +- > sound/usb/6fire/midi.h | 2 +- > sound/usb/6fire/pcm.c | 8 +-- > sound/usb/6fire/pcm.h | 2 +- > sound/usb/caiaq/control.c | 8 +-- > sound/usb/caiaq/device.c | 6 +- > 198 files changed, 1794 insertions(+), 1785 deletions(-) > > -- > 1.8.0.1 >

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Re: [alsa-devel] [PATCH 20/20] ALSA: snd-usb-caiaq: remove __dev* attributes
by Daniel Mack 06 Dec '12

06 Dec '12

On 06.12.2012 18:35, Bill Pemberton wrote: > CONFIG_HOTPLUG is going away as an option. As result the __dev* > markings will be going away. > > Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, > and __devexit. > > Signed-off-by: Bill Pemberton <wfp5p(a)virginia.edu> > Cc: Daniel Mack <zonque(a)gmail.com> Acked-by: Daniel Mack <zonque(a)gmail.com> Thanks! Daniel > --- > sound/usb/caiaq/control.c | 8 ++++---- > sound/usb/caiaq/device.c | 6 +++--- > 2 files changed, 7 insertions(+), 7 deletions(-) > > diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c > index 00e5d0a..adb8d03 100644 > --- a/sound/usb/caiaq/control.c > +++ b/sound/usb/caiaq/control.c > @@ -137,7 +137,7 @@ static int control_put(struct snd_kcontrol *kcontrol, > return 1; > } > > -static struct snd_kcontrol_new kcontrol_template __devinitdata = { > +static struct snd_kcontrol_new kcontrol_template = { > .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, > .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, > .index = 0, > @@ -489,8 +489,8 @@ static struct caiaq_controller kontrols4_controller[] = { > { "LED: FX2: Mode", 133 | CNT_INTVAL }, > }; > > -static int __devinit add_controls(struct caiaq_controller *c, int num, > - struct snd_usb_caiaqdev *dev) > +static int add_controls(struct caiaq_controller *c, int num, > + struct snd_usb_caiaqdev *dev) > { > int i, ret; > struct snd_kcontrol *kc; > @@ -507,7 +507,7 @@ static int __devinit add_controls(struct caiaq_controller *c, int num, > return 0; > } > > -int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) > +int snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) > { > int ret = 0; > > diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c > index 7da0d0a..c828f81 100644 > --- a/sound/usb/caiaq/device.c > +++ b/sound/usb/caiaq/device.c > @@ -289,7 +289,7 @@ int snd_usb_caiaq_set_auto_msg(struct snd_usb_caiaqdev *dev, > tmp, sizeof(tmp)); > } > > -static void __devinit setup_card(struct snd_usb_caiaqdev *dev) > +static void setup_card(struct snd_usb_caiaqdev *dev) > { > int ret; > char val[4]; > @@ -407,7 +407,7 @@ static int create_card(struct usb_device *usb_dev, > return 0; > } > > -static int __devinit init_card(struct snd_usb_caiaqdev *dev) > +static int init_card(struct snd_usb_caiaqdev *dev) > { > char *c, usbpath[32]; > struct usb_device *usb_dev = dev->chip.dev; > @@ -481,7 +481,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) > return 0; > } > > -static int __devinit snd_probe(struct usb_interface *intf, > +static int snd_probe(struct usb_interface *intf, > const struct usb_device_id *id) > { > int ret; >

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[alsa-devel] [PATCH] ASoC: lm49453: Fix analog and digital volume ranges
by MR.Swami.Reddy＠ti.com 06 Dec '12

06 Dec '12

From: MR Swami Reddy <MR.Swami.Reddy(a)ti.com> [PATCH] ASoC: lm49453: Fix analog and digital volume ranges Signed-off-by: M R Swami Reddy <mr.swami.reddy(a)ti.com> -- sound/soc/codecs/lm49453.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 808e50a..afbefa2 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -495,7 +495,8 @@ SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0), /* TLV Declarations */ static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1); -static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(analog_tlv, 0, 200, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0); static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = { /* Sidetone supports mono only */ @@ -516,9 +517,9 @@ SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, static const struct snd_kcontrol_new lm49453_snd_controls[] = { /* mic1 and mic2 supports mono only */ SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6, - 0, digital_tlv), + 0, analog_tlv), SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6, - 0, digital_tlv), + 0, analog_tlv), SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv),

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[alsa-devel] [PATCH 1/3] ALSA: ASoC: McASP: remove unused variables
by Daniel Mack 06 Dec '12

06 Dec '12

codec_fmt and sample_rate variables are unused in both snd_platform_data and davinci_audio_dev, so drop them. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- include/linux/platform_data/davinci_asp.h | 1 - sound/soc/davinci/davinci-mcasp.c | 1 - sound/soc/davinci/davinci-mcasp.h | 2 -- 3 files changed, 4 deletions(-) diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index f3d6e4f..8db5ae0 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -23,7 +23,6 @@ struct snd_platform_data { u32 rx_dma_offset; int asp_chan_q; /* event queue number for ASP channel */ int ram_chan_q; /* event queue number for RAM channel */ - unsigned int codec_fmt; /* * Allowing this is more efficient and eliminates left and right swaps * caused by underruns, but will swap the left and right channels diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 46350b8..023a70d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1174,7 +1174,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->tdm_slots = pdata->tdm_slots; dev->num_serializer = pdata->num_serializer; dev->serial_dir = pdata->serial_dir; - dev->codec_fmt = pdata->codec_fmt; dev->version = pdata->version; dev->txnumevt = pdata->txnumevt; dev->rxnumevt = pdata->rxnumevt; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 156f15f..a42b5d9 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -40,9 +40,7 @@ enum { struct davinci_audio_dev { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; - int sample_rate; struct device *dev; - unsigned int codec_fmt; /* McASP specific data */ int tdm_slots; -- 1.7.11.7

2 5

[alsa-devel] [PATCH] ASoC: lm49453: Fix adc, mic and sidetone volume ranges
by MR.Swami.Reddy＠ti.com 06 Dec '12

06 Dec '12

From: MR Swami Reddy <MR.Swami.Reddy(a)ti.com> [PATCH] ASoC: lm49453: Fix adc, mic and sidetone volume ranges Signed-off-by: MR Swami Reddy <mr.swami.reddy(a)ti.com> -- sound/soc/codecs/lm49453.c | 43 ++++++++++++++++++++++++------------------- 1 files changed, 24 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 808e50a..379fcdf 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -494,36 +494,41 @@ SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0), }; /* TLV Declarations */ -static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1); -static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(adc_dac_tlv, -7650, 150, 1); +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 200, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0); +static const DECLARE_TLV_DB_SCALE(stn_tlv, -7200, 150, 0); static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = { /* Sidetone supports mono only */ SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), }; static const struct snd_kcontrol_new lm49453_snd_controls[] = { /* mic1 and mic2 supports mono only */ - SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6, - 0, digital_tlv), - SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6, - 0, digital_tlv), + SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_MICL_REG, 0, 15, 0, mic_tlv), + SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_MICR_REG, 0, 15, 0, mic_tlv), + + SOC_SINGLE_TLV("ADCL Volume", LM49453_P0_ADC_LEVELL_REG, 0, 63, + 0, adc_dac_tlv), + SOC_SINGLE_TLV("ADCR Volume", LM49453_P0_ADC_LEVELR_REG, 0, 63, + 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, - LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DMIC1_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG, - LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DMIC2_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum), SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum), @@ -538,16 +543,16 @@ static const struct snd_kcontrol_new lm49453_snd_controls[] = { 2, 1, 0), SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG, - LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_HP_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG, - LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_LO_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG, - LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_LS_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG, - LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_HA_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG, - 0, 6, 0, digital_tlv), + 0, 63, 0, adc_dac_tlv), SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, 0, 3, 0, port_tlv),

1 0

[alsa-devel] [PATCH] ASoC: arizona: Store the DAI clock ID when setting
by Mark Brown 06 Dec '12

06 Dec '12

So the code to suppress duplicate changes is effective. Reported-by: Kyung Kwee Ryu <Kyung-Kwee.Ryu@wolfsonmicro.comyu@wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index ef68f7a..dfaeca9 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1,3 +1,4 @@ +#define DEBUG /* * arizona.c - Wolfson Arizona class device shared support * @@ -679,6 +680,9 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; + int base = dai->driver->base; int i, sr_val; /* @@ -699,14 +703,16 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream, case ARIZONA_CLK_SYSCLK: snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, ARIZONA_SAMPLE_RATE_1_MASK, sr_val); - snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_RATE_MASK, 0); + if (base) + snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, 0); break; case ARIZONA_CLK_ASYNCCLK: snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1, ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val); - snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_RATE_MASK, 8); + if (base) + snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, 8); break; default: arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk); @@ -721,12 +727,10 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); - struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; int base = dai->driver->base; const int *rates; int i, ret; - int bclk, lrclk, wl, frame, sr_val; + int bclk, lrclk, wl, frame; if (params_rate(params) % 8000) rates = &arizona_44k1_bclk_rates[0]; @@ -754,7 +758,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, wl = snd_pcm_format_width(params_format(params)); frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; - ret = arizona_hw_params_rate(codec, params, dai); + ret = arizona_hw_params_rate(substream, params, dai); if (ret != 0) return ret; @@ -823,6 +827,8 @@ static int arizona_dai_set_sysclk(struct snd_soc_dai *dai, routes[1].source = arizona_dai_clk_str(clk_id); snd_soc_dapm_add_routes(&codec->dapm, routes, ARRAY_SIZE(routes)); + dai_priv->clk = clk_id; + return snd_soc_dapm_sync(&codec->dapm); } -- 1.7.10.4

1 0

[alsa-devel] [RFC][PATCH 0/2] ASoC: add DT support on simple-card
by Kuninori Morimoto 06 Dec '12

06 Dec '12

Hi Mark These patches add DT support for soc-simple-card driver. Kuninori Morimoto (2): ASoC: add snd_soc_of_parse_daifmt() ASoC: simple-card: add DT support include/sound/soc.h | 2 + sound/soc/generic/simple-card.c | 90 +++++++++++++++++++++++++++++++++++++-- sound/soc/soc-core.c | 72 +++++++++++++++++++++++++++++++ 3 files changed, 161 insertions(+), 3 deletions(-) 1st patch adds snd_soc_of_parse_daifmt() to select SND_SOC_DAIFMT_xxx, but I'm not sure whether this is good idea. So these are [RFC] patches. I thought that it will be trouble if... 1) platform/driver DT used snd_fmt = <0xYYYY> style for SND_SOC_DAIFMT_xxx 2) someone might update SND_SOC_DAIFMT_xxx value for some reasons. 3) bootloader should update DT value according to kernel version if 2) happened I put snd_soc_of_parse_daifmt() on soc-core.c because I thought that other driver can reuse it. I need your comment for it. Best regards --- Kuninori Morimoto

4 13

[alsa-devel] [PATCH] ASoC: arizona: Make FLL lock timeout very high
by Mark Brown 06 Dec '12

06 Dec '12

Provide robustness against low quality FLL sync clocks by increasing the timeout for lock to an absurdly high point; we should never get anywhere near hitting the timeout in a real system unless it is failing. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 5fc11bc..ef68f7a 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1083,7 +1083,7 @@ int arizona_set_fll(struct arizona_fll *fll, int source, ARIZONA_FLL1_SYNC_ENA); ret = wait_for_completion_timeout(&fll->ok, - msecs_to_jiffies(25)); + msecs_to_jiffies(250)); if (ret == 0) arizona_fll_warn(fll, "Timed out waiting for lock\n"); -- 1.7.10.4

1 0

[alsa-devel] [PATCH 00/14] ASoC: Use devm_* APIs
by Sachin Kamat 06 Dec '12

06 Dec '12

This series is build tested against the for-next branch of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tree. Sachin Kamat (14): ASoC: wm8350: Use devm_regulator_bulk_get ASoC: wm8962: Use devm_regulator_bulk_get ASoC: wm8400: Use devm_regulator_bulk_get ASoC: wm8993: Use devm_regulator_bulk_get ASoC: ak4535: Use devm_regmap_init_i2c() ASoC: da7210: Use devm_regmap_init_i2c() ASoC: lm49453: Use devm_regmap_init_i2c() ASoC: max9768: Use devm_regmap_init_i2c() ASoC: wm8955: Use devm_regmap_init_i2c() ASoC: wm8960: Use devm_regmap_init_i2c() ASoC: wm8962: Use devm_regmap_init_i2c() ASoC: wm8978: Use devm_regmap_init_i2c() ASoC: wm8993: Use devm_regmap_init_i2c() ASoC: wm9081: Use devm_regmap_init_i2c() sound/soc/codecs/ak4535.c | 7 +------ sound/soc/codecs/da7210.c | 13 ++----------- sound/soc/codecs/lm49453.c | 10 ++-------- sound/soc/codecs/max9768.c | 7 ++----- sound/soc/codecs/wm8350.c | 4 +--- sound/soc/codecs/wm8400.c | 14 +++----------- sound/soc/codecs/wm8955.c | 11 +---------- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8962.c | 24 ++++++++---------------- sound/soc/codecs/wm8978.c | 13 +++---------- sound/soc/codecs/wm8993.c | 14 ++++---------- sound/soc/codecs/wm9081.c | 22 ++++++---------------- 12 files changed, 34 insertions(+), 107 deletions(-) -- 1.7.4.1

2 30

[alsa-devel] [PATCH v3] HDA: Add PCI device prefix for clarity
by Daniel J Blueman 05 Dec '12

05 Dec '12

When printing, use a prefix of the PCI domain, bus, device and function as in other drivers, to differentiate multiple devices. Important for reporting and debugging. A future step is to tidy this up with dev_printk et al. v2: Move conversion specifier into call site, preventing build issues v3: Refactor for Takashi's for-next branch Signed-off-by: Daniel J Blueman <daniel(a)quora.org> --- sound/pci/hda/hda_intel.c | 125 +++++++++++++++++++++++---------------------- 1 file changed, 64 insertions(+), 61 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 22ecadc..eb48109 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -192,7 +192,7 @@ MODULE_DESCRIPTION("Intel HDA driver"); #ifdef CONFIG_SND_VERBOSE_PRINTK #define SFX /* nop */ #else -#define SFX "hda-intel: " +#define SFX "hda-intel " #endif #if defined(CONFIG_PM) && defined(CONFIG_VGA_SWITCHEROO) @@ -717,7 +717,7 @@ static int azx_alloc_cmd_io(struct azx *chip) snd_dma_pci_data(chip->pci), PAGE_SIZE, &chip->rb); if (err < 0) { - snd_printk(KERN_ERR SFX "cannot allocate CORB/RIRB\n"); + snd_printk(KERN_ERR SFX "%s: cannot allocate CORB/RIRB\n", pci_name(chip->pci)); return err; } mark_pages_wc(chip, &chip->rb, true); @@ -894,9 +894,9 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, } if (!chip->polling_mode && chip->poll_count < 2) { - snd_printdd(SFX "azx_get_response timeout, " + snd_printdd(SFX "%s: azx_get_response timeout, " "polling the codec once: last cmd=0x%08x\n", - chip->last_cmd[addr]); + pci_name(chip->pci), chip->last_cmd[addr]); do_poll = 1; chip->poll_count++; goto again; @@ -904,17 +904,17 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, if (!chip->polling_mode) { - snd_printk(KERN_WARNING SFX "azx_get_response timeout, " + snd_printk(KERN_WARNING SFX "%s: azx_get_response timeout, " "switching to polling mode: last cmd=0x%08x\n", - chip->last_cmd[addr]); + pci_name(chip->pci), chip->last_cmd[addr]); chip->polling_mode = 1; goto again; } if (chip->msi) { - snd_printk(KERN_WARNING SFX "No response from codec, " + snd_printk(KERN_WARNING SFX "%s: No response from codec, " "disabling MSI: last cmd=0x%08x\n", - chip->last_cmd[addr]); + pci_name(chip->pci), chip->last_cmd[addr]); free_irq(chip->irq, chip); chip->irq = -1; pci_disable_msi(chip->pci); @@ -980,8 +980,8 @@ static int azx_single_wait_for_response(struct azx *chip, unsigned int addr) udelay(1); } if (printk_ratelimit()) - snd_printd(SFX "get_response timeout: IRS=0x%x\n", - azx_readw(chip, IRS)); + snd_printd(SFX "%s: get_response timeout: IRS=0x%x\n", + pci_name(chip->pci), azx_readw(chip, IRS)); chip->rirb.res[addr] = -1; return -EIO; } @@ -1008,8 +1008,8 @@ static int azx_single_send_cmd(struct hda_bus *bus, u32 val) udelay(1); } if (printk_ratelimit()) - snd_printd(SFX "send_cmd timeout: IRS=0x%x, val=0x%x\n", - azx_readw(chip, IRS), val); + snd_printd(SFX "%s: send_cmd timeout: IRS=0x%x, val=0x%x\n", + pci_name(chip->pci), azx_readw(chip, IRS), val); return -EIO; } @@ -1095,7 +1095,7 @@ static int azx_reset(struct azx *chip, int full_reset) __skip: /* check to see if controller is ready */ if (!azx_readb(chip, GCTL)) { - snd_printd(SFX "azx_reset: controller not ready!\n"); + snd_printd(SFX "%s: azx_reset: controller not ready!\n", pci_name(chip->pci)); return -EBUSY; } @@ -1107,7 +1107,7 @@ static int azx_reset(struct azx *chip, int full_reset) /* detect codecs */ if (!chip->codec_mask) { chip->codec_mask = azx_readw(chip, STATESTS); - snd_printdd(SFX "codec_mask = 0x%x\n", chip->codec_mask); + snd_printdd(SFX "%s: codec_mask = 0x%x\n", pci_name(chip->pci), chip->codec_mask); } return 0; @@ -1251,7 +1251,7 @@ static void azx_init_pci(struct azx *chip) * The PCI register TCSEL is defined in the Intel manuals. */ if (!(chip->driver_caps & AZX_DCAPS_NO_TCSEL)) { - snd_printdd(SFX "Clearing TCSEL\n"); + snd_printdd(SFX "%s: Clearing TCSEL\n", pci_name(chip->pci)); update_pci_byte(chip->pci, ICH6_PCIREG_TCSEL, 0x07, 0); } @@ -1259,7 +1259,7 @@ static void azx_init_pci(struct azx *chip) * we need to enable snoop. */ if (chip->driver_caps & AZX_DCAPS_ATI_SNOOP) { - snd_printdd(SFX "Setting ATI snoop: %d\n", azx_snoop(chip)); + snd_printdd(SFX "%s: Setting ATI snoop: %d\n", pci_name(chip->pci), azx_snoop(chip)); update_pci_byte(chip->pci, ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, 0x07, azx_snoop(chip) ? ATI_SB450_HDAUDIO_ENABLE_SNOOP : 0); @@ -1267,7 +1267,7 @@ static void azx_init_pci(struct azx *chip) /* For NVIDIA HDA, enable snoop */ if (chip->driver_caps & AZX_DCAPS_NVIDIA_SNOOP) { - snd_printdd(SFX "Setting Nvidia snoop: %d\n", azx_snoop(chip)); + snd_printdd(SFX "%s: Setting Nvidia snoop: %d\n", pci_name(chip->pci), azx_snoop(chip)); update_pci_byte(chip->pci, NVIDIA_HDA_TRANSREG_ADDR, 0x0f, NVIDIA_HDA_ENABLE_COHBITS); @@ -1292,8 +1292,8 @@ static void azx_init_pci(struct azx *chip) pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop); } - snd_printdd(SFX "SCH snoop: %s\n", - (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) + snd_printdd(SFX "%s: SCH snoop: %s\n", + pci_name(chip->pci), (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) ? "Disabled" : "Enabled"); } } @@ -1453,8 +1453,8 @@ static int azx_setup_periods(struct azx *chip, pos_align; pos_adj = frames_to_bytes(runtime, pos_adj); if (pos_adj >= period_bytes) { - snd_printk(KERN_WARNING SFX "Too big adjustment %d\n", - bdl_pos_adj[chip->dev_index]); + snd_printk(KERN_WARNING SFX "%s: Too big adjustment %d\n", + pci_name(chip->pci), bdl_pos_adj[chip->dev_index]); pos_adj = 0; } else { ofs = setup_bdle(chip, substream, azx_dev, @@ -1478,8 +1478,8 @@ static int azx_setup_periods(struct azx *chip, return 0; error: - snd_printk(KERN_ERR SFX "Too many BDL entries: buffer=%d, period=%d\n", - azx_dev->bufsize, period_bytes); + snd_printk(KERN_ERR SFX "%s: Too many BDL entries: buffer=%d, period=%d\n", + pci_name(chip->pci), azx_dev->bufsize, period_bytes); return -EINVAL; } @@ -1576,7 +1576,7 @@ static int probe_codec(struct azx *chip, int addr) mutex_unlock(&chip->bus->cmd_mutex); if (res == -1) return -EIO; - snd_printdd(SFX "codec #%d probed OK\n", addr); + snd_printdd(SFX "%s: codec #%d probed OK\n", pci_name(chip->pci), addr); return 0; } @@ -1653,7 +1653,7 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode return err; if (chip->driver_caps & AZX_DCAPS_RIRB_DELAY) { - snd_printd(SFX "Enable delay in RIRB handling\n"); + snd_printd(SFX "%s: Enable delay in RIRB handling\n", pci_name(chip->pci)); chip->bus->needs_damn_long_delay = 1; } @@ -1670,8 +1670,8 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode * that don't exist */ snd_printk(KERN_WARNING SFX - "Codec #%d probe error; " - "disabling it...\n", c); + "%s: Codec #%d probe error; " + "disabling it...\n", pci_name(chip->pci), c); chip->codec_mask &= ~(1 << c); /* More badly, accessing to a non-existing * codec often screws up the controller chip, @@ -1691,7 +1691,8 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode * access works around the stall. Grrr... */ if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) { - snd_printd(SFX "Enable sync_write for stable communication\n"); + snd_printd(SFX "%s: Enable sync_write for stable communication\n", + pci_name(chip->pci)); chip->bus->sync_write = 1; chip->bus->allow_bus_reset = 1; } @@ -1709,7 +1710,7 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode } } if (!codecs) { - snd_printk(KERN_ERR SFX "no codecs initialized\n"); + snd_printk(KERN_ERR SFX "%s: no codecs initialized\n", pci_name(chip->pci)); return -ENXIO; } return 0; @@ -2019,16 +2020,16 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) ctls); if (!format_val) { snd_printk(KERN_ERR SFX - "invalid format_val, rate=%d, ch=%d, format=%d\n", - runtime->rate, runtime->channels, runtime->format); + "%s: invalid format_val, rate=%d, ch=%d, format=%d\n", + pci_name(chip->pci), runtime->rate, runtime->channels, runtime->format); return -EINVAL; } bufsize = snd_pcm_lib_buffer_bytes(substream); period_bytes = snd_pcm_lib_period_bytes(substream); - snd_printdd(SFX "azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", - bufsize, format_val); + snd_printdd(SFX "%s: azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", + pci_name(chip->pci), bufsize, format_val); if (bufsize != azx_dev->bufsize || period_bytes != azx_dev->period_bytes || @@ -2287,9 +2288,9 @@ static unsigned int azx_get_position(struct azx *chip, delay += azx_dev->bufsize; if (delay >= azx_dev->period_bytes) { snd_printk(KERN_WARNING SFX - "Unstable LPIB (%d >= %d); " + "%s: Unstable LPIB (%d >= %d); " "disabling LPIB delay counting\n", - delay, azx_dev->period_bytes); + pci_name(chip->pci), delay, azx_dev->period_bytes); delay = 0; chip->driver_caps &= ~AZX_DCAPS_COUNT_LPIB_DELAY; } @@ -2446,7 +2447,8 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, list_for_each_entry(apcm, &chip->pcm_list, list) { if (apcm->pcm->device == pcm_dev) { - snd_printk(KERN_ERR SFX "PCM %d already exists\n", pcm_dev); + snd_printk(KERN_ERR SFX "%s: PCM %d already exists\n", + pci_name(chip->pci), pcm_dev); return -EBUSY; } } @@ -2771,15 +2773,14 @@ static void azx_vs_set_state(struct pci_dev *pci, } } else { snd_printk(KERN_INFO SFX - "%s %s via VGA-switcheroo\n", - disabled ? "Disabling" : "Enabling", - pci_name(chip->pci)); + "%s: %s via VGA-switcheroo\n", pci_name(chip->pci), + disabled ? "Disabling" : "Enabling"); if (disabled) { azx_suspend(&pci->dev); chip->disabled = true; if (snd_hda_lock_devices(chip->bus)) - snd_printk(KERN_WARNING SFX - "Cannot lock devices!\n"); + snd_printk(KERN_WARNING SFX "%s: Cannot lock devices!\n", + pci_name(chip->pci)); } else { snd_hda_unlock_devices(chip->bus); chip->disabled = false; @@ -2998,11 +2999,11 @@ static int __devinit check_position_fix(struct azx *chip, int fix) /* Check VIA/ATI HD Audio Controller exist */ if (chip->driver_caps & AZX_DCAPS_POSFIX_VIA) { - snd_printd(SFX "Using VIACOMBO position fix\n"); + snd_printd(SFX "%s: Using VIACOMBO position fix\n", pci_name(chip->pci)); return POS_FIX_VIACOMBO; } if (chip->driver_caps & AZX_DCAPS_POSFIX_LPIB) { - snd_printd(SFX "Using LPIB position fix\n"); + snd_printd(SFX "%s: Using LPIB position fix\n", pci_name(chip->pci)); return POS_FIX_LPIB; } return POS_FIX_AUTO; @@ -3118,8 +3119,8 @@ static void __devinit azx_check_snoop_available(struct azx *chip) } if (snoop != chip->snoop) { - snd_printk(KERN_INFO SFX "Force to %s mode\n", - snoop ? "snoop" : "non-snoop"); + snd_printk(KERN_INFO SFX "%s: Force to %s mode\n", + pci_name(chip->pci), snoop ? "snoop" : "non-snoop"); chip->snoop = snoop; } } @@ -3145,7 +3146,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip = kzalloc(sizeof(*chip), GFP_KERNEL); if (!chip) { - snd_printk(KERN_ERR SFX "cannot allocate chip\n"); + snd_printk(KERN_ERR SFX "%s: Cannot allocate chip\n", pci_name(pci)); pci_disable_device(pci); return -ENOMEM; } @@ -3193,7 +3194,8 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { - snd_printk(KERN_ERR SFX "Error creating device [card]!\n"); + snd_printk(KERN_ERR SFX "%s: Error creating device [card]!\n", + pci_name(chip->pci)); azx_free(chip); return err; } @@ -3228,7 +3230,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) chip->addr = pci_resource_start(pci, 0); chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { - snd_printk(KERN_ERR SFX "ioremap error\n"); + snd_printk(KERN_ERR SFX "%s: ioremap error\n", pci_name(chip->pci)); return -ENXIO; } @@ -3243,7 +3245,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) synchronize_irq(chip->irq); gcap = azx_readw(chip, GCAP); - snd_printdd(SFX "chipset global capabilities = 0x%x\n", gcap); + snd_printdd(SFX "%s: chipset global capabilities = 0x%x\n", pci_name(chip->pci), gcap); /* disable SB600 64bit support for safety */ if (chip->pci->vendor == PCI_VENDOR_ID_ATI) { @@ -3260,7 +3262,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) /* disable 64bit DMA address on some devices */ if (chip->driver_caps & AZX_DCAPS_NO_64BIT) { - snd_printd(SFX "Disabling 64bit DMA\n"); + snd_printd(SFX "%s: Disabling 64bit DMA\n", pci_name(chip->pci)); gcap &= ~ICH6_GCAP_64OK; } @@ -3315,7 +3317,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev), GFP_KERNEL); if (!chip->azx_dev) { - snd_printk(KERN_ERR SFX "cannot malloc azx_dev\n"); + snd_printk(KERN_ERR SFX "%s: cannot malloc azx_dev\n", pci_name(chip->pci)); return -ENOMEM; } @@ -3325,7 +3327,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) snd_dma_pci_data(chip->pci), BDL_SIZE, &chip->azx_dev[i].bdl); if (err < 0) { - snd_printk(KERN_ERR SFX "cannot allocate BDL\n"); + snd_printk(KERN_ERR SFX "%s: cannot allocate BDL\n", pci_name(chip->pci)); return -ENOMEM; } mark_pages_wc(chip, &chip->azx_dev[i].bdl, true); @@ -3335,7 +3337,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) snd_dma_pci_data(chip->pci), chip->num_streams * 8, &chip->posbuf); if (err < 0) { - snd_printk(KERN_ERR SFX "cannot allocate posbuf\n"); + snd_printk(KERN_ERR SFX "%s: cannot allocate posbuf\n", pci_name(chip->pci)); return -ENOMEM; } mark_pages_wc(chip, &chip->posbuf, true); @@ -3353,7 +3355,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) /* codec detection */ if (!chip->codec_mask) { - snd_printk(KERN_ERR SFX "no codecs found!\n"); + snd_printk(KERN_ERR SFX "%s: no codecs found!\n", pci_name(chip->pci)); return -ENODEV; } @@ -3389,7 +3391,8 @@ static void azx_firmware_cb(const struct firmware *fw, void *context) struct pci_dev *pci = chip->pci; if (!fw) { - snd_printk(KERN_ERR SFX "Cannot load firmware, aborting\n"); + snd_printk(KERN_ERR SFX "%s: Cannot load firmware, aborting\n", + pci_name(chip->pci)); goto error; } @@ -3425,7 +3428,7 @@ static int __devinit azx_probe(struct pci_dev *pci, err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { - snd_printk(KERN_ERR SFX "Error creating card!\n"); + snd_printk(KERN_ERR "hda-intel: Error creating card!\n"); return err; } @@ -3441,14 +3444,14 @@ static int __devinit azx_probe(struct pci_dev *pci, err = register_vga_switcheroo(chip); if (err < 0) { snd_printk(KERN_ERR SFX - "Error registering VGA-switcheroo client\n"); + "%s: Error registering VGA-switcheroo client\n", pci_name(pci)); goto out_free; } if (check_hdmi_disabled(pci)) { - snd_printk(KERN_INFO SFX "VGA controller for %s is disabled\n", + snd_printk(KERN_INFO SFX "%s: VGA controller is disabled\n", pci_name(pci)); - snd_printk(KERN_INFO SFX "Delaying initialization\n"); + snd_printk(KERN_INFO SFX "%s: Delaying initialization\n", pci_name(pci)); chip->disabled = true; } @@ -3461,8 +3464,8 @@ static int __devinit azx_probe(struct pci_dev *pci, #ifdef CONFIG_SND_HDA_PATCH_LOADER if (patch[dev] && *patch[dev]) { - snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n", - patch[dev]); + snd_printk(KERN_ERR SFX "%s: Applying patch firmware '%s'\n", + pci_name(pci), patch[dev]); err = request_firmware_nowait(THIS_MODULE, true, patch[dev], &pci->dev, GFP_KERNEL, card, azx_firmware_cb); -- 1.7.10.4

2 1

[alsa-devel] [PATCH 1/3] ALSA: ASoC: cs4271: fix sparse warning
by Daniel Mack 05 Dec '12

05 Dec '12

Make the flag in the pdata of type bool to fix a sparse warning. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> --- include/sound/cs4271.h | 2 +- sound/soc/codecs/cs4271.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h index 6d9e15e..dd8c48d 100644 --- a/include/sound/cs4271.h +++ b/include/sound/cs4271.h @@ -19,7 +19,7 @@ struct cs4271_platform_data { int gpio_nreset; /* GPIO driving Reset pin, if any */ - int amutec_eq_bmutec:1; /* flag to enable AMUTEC=BMUTEC */ + bool amutec_eq_bmutec; /* flag to enable AMUTEC=BMUTEC */ }; #endif /* __CS4271_H */ diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 6ad3878..70de900 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -474,7 +474,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; int ret; int gpio_nreset = -EINVAL; - int amutec_eq_bmutec = 0; + bool amutec_eq_bmutec = false; #ifdef CONFIG_OF if (of_match_device(cs4271_dt_ids, codec->dev)) { @@ -483,7 +483,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) if (!of_get_property(codec->dev->of_node, "cirrus,amutec-eq-bmutec", NULL)) - amutec_eq_bmutec = 1; + amutec_eq_bmutec = true; } #endif -- 1.7.11.7

2 6

[alsa-devel] [PATCH] ASoC:lm49453: Update lm49453_reg_defs values as per LM49453 HW version-RevB
by MR.Swami.Reddy＠ti.com 05 Dec '12

05 Dec '12

From: MR Swami Reddy <MR.Swami.Reddy(a)ti.com> [PATCH] ASoC:lm49453: Update lm49453_reg_defs values as per LM49453 HW version-RevB Signed-off-by: M R Swami Reddy <mr.swami.reddy(a)ti.com> --- sound/soc/codecs/lm49453.c | 61 +++++++++++--------------------------------- 1 files changed, 15 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 99b0a9d..808e50a 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -111,9 +111,9 @@ static struct reg_default lm49453_reg_defs[] = { { 101, 0x00 }, { 102, 0x00 }, { 103, 0x01 }, - { 105, 0x01 }, - { 106, 0x00 }, - { 107, 0x01 }, + { 104, 0x01 }, + { 105, 0x00 }, + { 106, 0x01 }, { 107, 0x00 }, { 108, 0x00 }, { 109, 0x00 }, @@ -163,56 +163,25 @@ static struct reg_default lm49453_reg_defs[] = { { 184, 0x00 }, { 185, 0x00 }, { 186, 0x00 }, - { 189, 0x00 }, + { 187, 0x00 }, { 188, 0x00 }, - { 194, 0x00 }, - { 195, 0x00 }, - { 196, 0x00 }, - { 197, 0x00 }, - { 200, 0x00 }, - { 201, 0x00 }, - { 202, 0x00 }, - { 203, 0x00 }, - { 204, 0x00 }, - { 205, 0x00 }, - { 208, 0x00 }, + { 189, 0x00 }, + { 208, 0x06 }, { 209, 0x00 }, - { 210, 0x00 }, - { 211, 0x00 }, - { 213, 0x00 }, - { 214, 0x00 }, - { 215, 0x00 }, - { 216, 0x00 }, - { 217, 0x00 }, - { 218, 0x00 }, - { 219, 0x00 }, + { 210, 0x08 }, + { 211, 0x54 }, + { 212, 0x14 }, + { 213, 0x0d }, + { 214, 0x0d }, + { 215, 0x14 }, + { 216, 0x60 }, { 221, 0x00 }, { 222, 0x00 }, + { 223, 0x00 }, { 224, 0x00 }, - { 225, 0x00 }, - { 226, 0x00 }, - { 227, 0x00 }, - { 228, 0x00 }, - { 229, 0x00 }, - { 230, 0x13 }, - { 231, 0x00 }, - { 232, 0x80 }, - { 233, 0x0C }, - { 234, 0xDD }, - { 235, 0x00 }, - { 236, 0x04 }, - { 237, 0x00 }, - { 238, 0x00 }, - { 239, 0x00 }, - { 240, 0x00 }, - { 241, 0x00 }, - { 242, 0x00 }, - { 243, 0x00 }, - { 244, 0x00 }, - { 245, 0x00 }, { 248, 0x00 }, { 249, 0x00 }, - { 254, 0x00 }, + { 250, 0x00 }, { 255, 0x00 }, };

1 0

[alsa-devel] [query] codec suspend not called when bias level is SND_SOC_BIAS_ON
by Shiraz Hashim 05 Dec '12

05 Dec '12

Hi Mark, I was going through the suspend procedure for soc-core and I could see that codec suspend is called only when its bias level is SND_SOC_BIAS_OFF. if (!codec->suspended && codec->driver->suspend) { switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: /* * If the CODEC is capable of idle * bias off then being in STANDBY * means it's doing something, * otherwise fall through. */ if (codec->dapm.idle_bias_off) { dev_dbg(codec->dev, "idle_bias_off CODEC on over suspend\n"); break; } case SND_SOC_BIAS_OFF: codec->driver->suspend(codec); codec->suspended = 1; codec->cache_sync = 1; break; default: dev_dbg(codec->dev, "CODEC is on over suspend\n"); break; } } My question is, during on-going audio play when system is put into suspend, codec->driver->suspend would not be called as its bias level is still SND_SOC_BIAS_ON, thus not giving a chance to codec to save its state. Is my understanding correct ? How to deal with this ? -- regards Shiraz

2 1

[alsa-devel] [PATCH 0/5] Realtek auto-mic enhancements / cleanups
by Takashi Iwai 05 Dec '12

05 Dec '12

Hi, this is a series of patches to enhance and clean up the automatic input selection via mic jack on HD-audio Realtek drivers. [PATCH 1/5] ALSA: hda - Rearrange INPUT_PIN_ATTR_* [PATCH 2/5] ALSA: hda - More generic auto-mic switching for Realtek [PATCH 3/5] ALSA: hda - Use standard sort function in [PATCH 4/5] ALSA: hda - Implement shared front mic/hp jack for HP [PATCH 5/5] ALSA: hda - Refactor alc_kcontrol_new() usages The first motivation was to generalize the code a bit better. Then, I got a machine that should be able to switch front mic and the secondary headphone. The machine is suitable for the new auto-mic test, so I included this hack in 4th path as well. Takashi

3 36

[alsa-devel] [PATCH] ASoC: lm49453: Fix analog and digital volume ranges
by MR.Swami.Reddy＠ti.com 05 Dec '12

05 Dec '12

From: MR Swami Reddy <MR.Swami.Reddy(a)ti.com> [PATCH] ASoC: lm49453: Fix analog and digital volume ranges" Signed-off-by: M R Swami Reddy <mr.swami.reddy(a)ti.com> -- sound/soc/codecs/lm49453.c | 7 +++++---- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 808e50a..ba1aec4 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -494,8 +494,9 @@ SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0), }; /* TLV Declarations */ -static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(analog_tlv, 0, 200, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0); static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = { /* Sidetone supports mono only */ @@ -516,9 +517,9 @@ SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, static const struct snd_kcontrol_new lm49453_snd_controls[] = { /* mic1 and mic2 supports mono only */ SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6, - 0, digital_tlv), + 0, analog_tlv), SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6, - 0, digital_tlv), + 0, analog_tlv), SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv),

2 1