November 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] HD-Audio: can msleep in hda_set_power_state() to D3 be optional?
by Lin, Mengdong 29 Nov '13

29 Nov '13

Hi Takashi, In hda_set_power_state(), there is a mandatory sleep up to 100ms before setting state to D3. /* this delay seems necessary to avoid click noise at power-down */ if (power_state == AC_PWRST_D3) { msleep(codec->epss ? 10 : 100); ... Could this be optional for specific codecs? flags = HDA_RW_NO_RESPONSE_FALLBACK; } Some codecs implement their own sleep in patch_ops.suspend() to avoid the noise. So is it okay to add a codec flag to decide whether the sleep is necessary here? Thanks Mengdong

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[alsa-devel] [PATCH] ASoC: adsp: Use async writes where possible
by Mark Brown 28 Nov '13

28 Nov '13

From: Mark Brown <broonie(a)linaro.org> This will allow a marginal speed improvement when used with a bus that supports async I/O by reducing the amount of context thrashing between writes, allowing the bus to be more fully utilised. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm_adsp.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 46ec0e9744d4..6b1c01c66e64 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1468,8 +1468,8 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) unsigned int val; int ret, count; - ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, - ADSP2_SYS_ENA, ADSP2_SYS_ENA); + ret = regmap_update_bits_async(dsp->regmap, dsp->base + ADSP2_CONTROL, + ADSP2_SYS_ENA, ADSP2_SYS_ENA); if (ret != 0) return ret; @@ -1521,9 +1521,9 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, val = (val & ARIZONA_SYSCLK_FREQ_MASK) >> ARIZONA_SYSCLK_FREQ_SHIFT; - ret = regmap_update_bits(dsp->regmap, - dsp->base + ADSP2_CLOCKING, - ADSP2_CLK_SEL_MASK, val); + ret = regmap_update_bits_async(dsp->regmap, + dsp->base + ADSP2_CLOCKING, + ADSP2_CLK_SEL_MASK, val); if (ret != 0) { adsp_err(dsp, "Failed to set clock rate: %d\n", ret); @@ -1586,10 +1586,10 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = regmap_update_bits(dsp->regmap, - dsp->base + ADSP2_CONTROL, - ADSP2_CORE_ENA | ADSP2_START, - ADSP2_CORE_ENA | ADSP2_START); + ret = regmap_update_bits_async(dsp->regmap, + dsp->base + ADSP2_CONTROL, + ADSP2_CORE_ENA | ADSP2_START, + ADSP2_CORE_ENA | ADSP2_START); if (ret != 0) goto err; -- 1.8.4.4

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[alsa-devel] [PATCH 1/3] ASoC: ak4642: Convert to module_i2c_driver()
by Mark Brown 28 Nov '13

28 Nov '13

From: Mark Brown <broonie(a)linaro.org> The device does not support anything other than I2C (at least with the current driver) so save code. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ak4642.c | 22 +--------------------- 1 file changed, 1 insertion(+), 21 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 2f861c9b1d69..5bb18e913fa5 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -511,7 +511,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4648 = { .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), }; -#if IS_ENABLED(CONFIG_I2C) static struct of_device_id ak4642_of_match[]; static int ak4642_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -571,27 +570,8 @@ static struct i2c_driver ak4642_i2c_driver = { .remove = ak4642_i2c_remove, .id_table = ak4642_i2c_id, }; -#endif -static int __init ak4642_modinit(void) -{ - int ret = 0; -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&ak4642_i2c_driver); -#endif - return ret; - -} -module_init(ak4642_modinit); - -static void __exit ak4642_exit(void) -{ -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&ak4642_i2c_driver); -#endif - -} -module_exit(ak4642_exit); +module_i2c_driver(ak4642_i2c_driver); MODULE_DESCRIPTION("Soc AK4642 driver"); MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); -- 1.8.4.4

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[alsa-devel] [PATCH 1/2] ALSA: hda - Initialize missing bass speaker pin for ASUS AIO ET2700
by Takashi Iwai 28 Nov '13

28 Nov '13

Add a fixup entry for the missing bass speaker pin 0x16 on ASUS ET2700 AiO desktop. The channel map will be added in the next patch, so that this can be backported easily to stable kernels. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65961 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3a9a1b0c1dbe..fd835c52618e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1782,6 +1782,7 @@ enum { ALC889_FIXUP_IMAC91_VREF, ALC882_FIXUP_INV_DMIC, ALC882_FIXUP_NO_PRIMARY_HP, + ALC887_FIXUP_ASUS_BASS, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -2105,6 +2106,13 @@ static const struct hda_fixup alc882_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc882_fixup_no_primary_hp, }, + [ALC887_FIXUP_ASUS_BASS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x16, 0x99130130}, /* bass speaker */ + {} + }, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2138,6 +2146,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), + SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS), SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), -- 1.8.4.4

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[alsa-devel] [PATCH] Add nommu support for alsa-utils
by Aaron Wu 28 Nov '13

28 Nov '13

Signed-off-by: Aaron Wu <aaron.wu(a)analog.com> Add nommu support for alsa-utils mainly for Blackfin --- package/alsa-utils/alsa-utils-nommu.patch | 54 +++++++++++++++++++++++++++++ package/alsa-utils/alsa-utils.mk | 1 + 2 files changed, 55 insertions(+) create mode 100644 package/alsa-utils/alsa-utils-nommu.patch diff --git a/package/alsa-utils/alsa-utils-nommu.patch b/package/alsa-utils/alsa-utils-nommu.patch new file mode 100644 index 0000000..c0227f5 --- /dev/null +++ b/package/alsa-utils/alsa-utils-nommu.patch @@ -0,0 +1,54 @@ +diff -Nur alsa-utils-1.0.26/alsactl/init_utils_run.c alsa-utils-1.0.26-new/alsactl/init_utils_run.c +--- alsa-utils-1.0.26/alsactl/init_utils_run.c 2012-09-06 15:52:57.000000000 +0800 ++++ alsa-utils-1.0.26-new/alsactl/init_utils_run.c 2013-10-22 02:19:06.210649921 +0800 +@@ -88,8 +88,11 @@ + strlcat(program, argv[0], sizeof(program)); + argv[0] = program; + } +- ++#ifdef HAVE_FORK + pid = fork(); ++#else ++ pid = vfork(); ++#endif + switch(pid) { + case 0: + /* child closes parent ends of pipes */ +diff -Nur alsa-utils-1.0.26/alsaloop/alsaloop.c alsa-utils-1.0.26-new/alsaloop/alsaloop.c +--- alsa-utils-1.0.26/alsaloop/alsaloop.c 2012-09-06 15:52:57.000000000 +0800 ++++ alsa-utils-1.0.26-new/alsaloop/alsaloop.c 2013-10-22 02:21:26.994650695 +0800 +@@ -850,14 +850,22 @@ + logit(LOG_CRIT, "daemon() failed: %s\n", strerror(errno)); + exit(EXIT_FAILURE); + } ++ #ifdef HAVE_FORK + i = fork(); ++ #else ++ i = vfork(); ++ #endif + if (i < 0) { + logit(LOG_CRIT, "fork() failed: %s\n", strerror(errno)); + exit(EXIT_FAILURE); + } + if (i > 0) { + /* wait(&i); */ ++ #ifdef HAVE_FORK + exit(EXIT_SUCCESS); ++ #else ++ _exit(EXIT_SUCCESS); ++ #endif + } + } + +diff -Nur alsa-utils-1.0.26/configure.in alsa-utils-1.0.26-new/configure.in +--- alsa-utils-1.0.26/configure.in 2012-09-06 16:45:44.000000000 +0800 ++++ alsa-utils-1.0.26-new/configure.in 2013-10-23 02:25:40.391127379 +0800 +@@ -23,6 +23,8 @@ + AC_MSG_RESULT($CC) + fi + ++ ++AC_CHECK_FUNC([fork]) + AC_PROG_CC + dnl AC_PROG_CXX + AC_PROG_INSTALL diff --git a/package/alsa-utils/alsa-utils.mk b/package/alsa-utils/alsa-utils.mk index b951112..4956e04 100644 --- a/package/alsa-utils/alsa-utils.mk +++ b/package/alsa-utils/alsa-utils.mk @@ -9,6 +9,7 @@ ALSA_UTILS_SOURCE = alsa-utils-$(ALSA_UTILS_VERSION).tar.bz2 ALSA_UTILS_SITE = http://alsa.cybermirror.org/utils ALSA_UTILS_LICENSE = GPLv2 ALSA_UTILS_LICENSE_FILES = COPYING +ALSA_UTILS_AUTORECONF = YES ALSA_UTILS_INSTALL_STAGING = YES ALSA_UTILS_DEPENDENCIES = host-gettext alsa-lib \ $(if $(BR2_PACKAGE_NCURSES),ncurses) -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: hda - Add zenbook-boost model for ASUS Zenbook
by Takashi Iwai 28 Nov '13

28 Nov '13

There's been a complaint about the too soft output on ASUS Zenbook with ALC269VB in comparison with Windows. It turned out, however, that the actual hardware output level is same between Linux and Windows unless software plugin is used. Still, it'd be convenient to give a capability for increasing the output level via Realtek codec's D-class amp. As a compromise, here provides a new model option, zenbook-boost, to enable the COEFs for the output boost amp. It's not used unless the option is given explicitly just for safety reason. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=62651 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 2 files changed, 14 insertions(+) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 85c362d8ea34..6fbd108bcb27 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -32,6 +32,7 @@ ALC269/270/275/276/28x/29x lenovo-dock Enables docking station I/O for some Lenovos dell-headset-multi Headset jack, which can also be used as mic-in dell-headset-dock Headset jack (without mic-in), and also dock I/O + zenbook-boost Add +6dB amp to output of ASUS Zenbook ALC66x/67x/892 ============== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3a9a1b0c1dbe..284952c37929 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3799,6 +3799,7 @@ enum { ALC269_FIXUP_ACER_AC700, ALC269_FIXUP_LIMIT_INT_MIC_BOOST, ALC269VB_FIXUP_ASUS_ZENBOOK, + ALC269VB_FIXUP_ASUS_ZENBOOK_BOOST, ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED, ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, @@ -4082,6 +4083,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269VB_FIXUP_DMIC, }, + [ALC269VB_FIXUP_ASUS_ZENBOOK_BOOST] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* class-D output boost +6dB */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x12}, + {0x20, AC_VERB_SET_PROC_COEF, 0x3000}, + { } + }, + .chained = true, + .chain_id = ALC269VB_FIXUP_ASUS_ZENBOOK, + }, [ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_limit_int_mic_boost, @@ -4299,6 +4311,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, {.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"}, + {.id = ALC269VB_FIXUP_ASUS_ZENBOOK_BOOST, .name = "zenbook-boost"}, {} }; -- 1.8.4.4

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[alsa-devel] [PATCH] ALSA: hda - Check leaf nodes to find aamix amps
by Takashi Iwai 28 Nov '13

28 Nov '13

The current generic parser assumes blindly that the volume and mute amps are found in the aamix node itself. But on some codecs, typically Analog Devices ones, the aamix amps are separately implemented in each leaf node of the aamix node, and the current driver can't establish the correct amp controls. This is a regression compared with the previous static quirks. This patch extends the search for the amps to the leaf nodes for allowing the aamix controls again on such codecs. In this implementation, I didn't code to loop through the whole paths, since usually one depth should suffice, and we can't search too deeply, as it may result in the conflicting control assignments. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65641 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 57 +++++++++++++++++++++++++++++++++++---------- 1 file changed, 45 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index faefff1d3e81..c4671d00babd 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -2808,6 +2808,42 @@ static int add_loopback_list(struct hda_gen_spec *spec, hda_nid_t mix, int idx) return 0; } +/* return true if either a volume or a mute amp is found for the given + * aamix path; the amp has to be either in the mixer node or its direct leaf + */ +static bool look_for_mix_leaf_ctls(struct hda_codec *codec, hda_nid_t mix_nid, + hda_nid_t pin, unsigned int *mix_val, + unsigned int *mute_val) +{ + int idx, num_conns; + const hda_nid_t *list; + hda_nid_t nid; + + idx = snd_hda_get_conn_index(codec, mix_nid, pin, true); + if (idx < 0) + return false; + + *mix_val = *mute_val = 0; + if (nid_has_volume(codec, mix_nid, HDA_INPUT)) + *mix_val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); + if (nid_has_mute(codec, mix_nid, HDA_INPUT)) + *mute_val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); + if (*mix_val && *mute_val) + return true; + + /* check leaf node */ + num_conns = snd_hda_get_conn_list(codec, mix_nid, &list); + if (num_conns < idx) + return false; + nid = list[idx]; + if (!*mix_val && nid_has_volume(codec, nid, HDA_OUTPUT)) + *mix_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); + if (!*mute_val && nid_has_mute(codec, nid, HDA_OUTPUT)) + *mute_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); + + return *mix_val || *mute_val; +} + /* create input playback/capture controls for the given pin */ static int new_analog_input(struct hda_codec *codec, int input_idx, hda_nid_t pin, const char *ctlname, int ctlidx, @@ -2815,12 +2851,11 @@ static int new_analog_input(struct hda_codec *codec, int input_idx, { struct hda_gen_spec *spec = codec->spec; struct nid_path *path; - unsigned int val; + unsigned int mix_val, mute_val; int err, idx; - if (!nid_has_volume(codec, mix_nid, HDA_INPUT) && - !nid_has_mute(codec, mix_nid, HDA_INPUT)) - return 0; /* no need for analog loopback */ + if (!look_for_mix_leaf_ctls(codec, mix_nid, pin, &mix_val, &mute_val)) + return 0; path = snd_hda_add_new_path(codec, pin, mix_nid, 0); if (!path) @@ -2829,20 +2864,18 @@ static int new_analog_input(struct hda_codec *codec, int input_idx, spec->loopback_paths[input_idx] = snd_hda_get_path_idx(codec, path); idx = path->idx[path->depth - 1]; - if (nid_has_volume(codec, mix_nid, HDA_INPUT)) { - val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); - err = __add_pb_vol_ctrl(spec, HDA_CTL_WIDGET_VOL, ctlname, ctlidx, val); + if (mix_val) { + err = __add_pb_vol_ctrl(spec, HDA_CTL_WIDGET_VOL, ctlname, ctlidx, mix_val); if (err < 0) return err; - path->ctls[NID_PATH_VOL_CTL] = val; + path->ctls[NID_PATH_VOL_CTL] = mix_val; } - if (nid_has_mute(codec, mix_nid, HDA_INPUT)) { - val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); - err = __add_pb_sw_ctrl(spec, HDA_CTL_WIDGET_MUTE, ctlname, ctlidx, val); + if (mute_val) { + err = __add_pb_sw_ctrl(spec, HDA_CTL_WIDGET_MUTE, ctlname, ctlidx, mute_val); if (err < 0) return err; - path->ctls[NID_PATH_MUTE_CTL] = val; + path->ctls[NID_PATH_MUTE_CTL] = mute_val; } path->active = true; -- 1.8.4.4

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[alsa-devel] [PATCH v7 0/4] Add dual-fifo mode support of i.MX ssi
by Nicolin Chen 28 Nov '13

28 Nov '13

* ! This series of patches has a direct dependency between them. When * ! applying them, we need to apply to one single branch. Otherwise, * ! it would break currect branches. Changelog v7: * Appended missing Acked-by to all four patches. * Sorry that I didn't add them at the first place. v6: * PATCH-1: Use goto err_firmware instead of return directly. * * Nothing changes for the other three ack-ed patches v5: * PATCH-3: Add period size constraint when using dual fifo mode * * Nothing changes for the other three patches v4: * PATCH-3: Drop useless register configuration. * * Nothing changes for the other three patches v3: * PATCH-1: Add comments to indicate the end of v1 and v2 array. * PATCH-3: Use better way to keep watermark as even number. * * Nothing changes for PATCH-2 and PATCH-4 v2: * Instead of adding rogue scripts to current SDMA driver based on firmware * V1, we define the new SDMA firmware as version 2 and bisect the PATCH-1 * to two patches: The first is to add version check code to the SDMA driver; * And the second is to add SSI dual FIFO DMATYPE. * * Nothing changes for the last two patches. v1: * SSI can reduce hardware overrun/underrun possibility when using dual * fifo mode. To support this mode, we need to first update sdma sciprt * list, and then enable dual fifo BIT in SSI driver, and last update DT * bindings of i.MX series. Nicolin Chen (4): dma: imx-sdma: Add sdma firmware version 2 support dma: imx-sdma: Add new dma type for ssi dual fifo script ASoC: fsl_ssi: Add dual fifo mode support ARM: dts: imx: use dual-fifo sdma script for ssi .../devicetree/bindings/dma/fsl-imx-sdma.txt | 1 + arch/arm/boot/dts/imx51.dtsi | 4 ++-- arch/arm/boot/dts/imx53.dtsi | 4 ++-- arch/arm/boot/dts/imx6qdl.dtsi | 12 +++++----- arch/arm/boot/dts/imx6sl.dtsi | 12 +++++----- drivers/dma/imx-sdma.c | 19 ++++++++++++++- include/linux/platform_data/dma-imx-sdma.h | 5 ++++ include/linux/platform_data/dma-imx.h | 1 + sound/soc/fsl/fsl_ssi.c | 27 +++++++++++++++++++++- 9 files changed, 67 insertions(+), 18 deletions(-) -- 1.8.4

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[alsa-devel] Intermittent cut out on the video
by Dolevo Jay 28 Nov '13

28 Nov '13

Hi experts, In my system, I use two headphone jacks to connect my speakers, one is to the jack on the pc itself and the other one is to the jack located on the monitor which I connect to the same pc with hdmi cable. I have two sound cards on my system and I use the following asound.conf to have audio from two jacks simultaneously. http://pastebin.com/2yd6EW8C As you see, I also use dmix because I have multiple of audio sources coming and I mix them. However, I now have the following problem. I connect two speakers to the each jack and start hearing an audio stream. The audio coming from the jack on the pc is cutting out intermittently. It just cuts out for a fraction of a second and continues as normal and after some time it happens again. The frequency of the occurrence increases over time. I have absolutely no problem with the audio from the jack on the display. Could anyone shed some light on this issue? In my opinion, this is a sort of configuration issue since I have no problem at one source while having at the other source. Thanks in advance, Regards, Dolevo

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Re: [alsa-devel] [PATCHv2 3/8] ARM: dts: Enables SAI ALSA SoC DAI device for Vybrid VF610 TOWER board.
by Bill Pringlemeir 28 Nov '13

28 Nov '13

[sorry, I should have copied alsa-devel(a)alsa-project.org previously]. On 19 Nov 2013, Li.Xiubo(a)freescale.com wrote: >>> This patch add and enable SAI device. >>> >>> arch/arm/boot/dts/vf610-twr.dts | 6 ++++++ >>> 1 file changed, 6 insertions(+) >>> >>> diff --git a/arch/arm/boot/dts/vf610-twr.dts >>> b/arch/arm/boot/dts/vf610-twr.dts index 1a58678..e4106dd 100644 >>> --- a/arch/arm/boot/dts/vf610-twr.dts >>> +++ b/arch/arm/boot/dts/vf610-twr.dts >>>>> -57,6 +57,12 @@ >>> status = "okay"; >>> }; >> >> Should we put a comment here that a TWR-AUDIO-SGTL board is needed? Or >> is this better in a menuconfig? I guess TWR-AUDIO-SGTL is needed as this >> codec is not on the VF610-TWR main board? >> > > Yes, the SGTL5000 codec is not on the VF610-TWR main board, and needs an extra > TWR-AUDIO-SGTL board. > > IMO, puting this comment in the menuconfig is much better. I agree. I guess the patch set 6 makes more sense. Also, you might consider adding a printk in the 'fsl_sgtl5000_probe()'. I guess that devm_snd_soc_register_card() will fail if the board is not present? As the board is not 'standard' people may come asking why the audio fails on the 'VF610-TWR'. + ret = devm_snd_soc_register_card(&pdev->dev, &fsl_sgt1500_card); + if (ret) { + dev_err(&pdev->dev, "register soc sound card failed :%d\n" + "TWR-AUDIO-SGTL board required.\n", + ret); + return ret; + } Especially, you can make one kernel that handles both cases w/wo TWR-AUDIO-SGTL board attached. Thanks for your work. Bill Pringlemeir.

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