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

[alsa-devel] [PATCH v2 2/4] ASoC: cx20442: add bias control over a platform provided regulator
by Janusz Krzysztofik 02 Jan '12

02 Jan '12

Now that a regulator device for controlling the codec chip reset state over a platform agnostic regulator API is available on the only board using this driver so far, extend the driver with a bias control function which will request virtual power to the codec chip from that virtual regulator, and will supersede the present implementation existing at the sound card level. Thanks to the regulator sharing mechanism, both the old (the sound card) and the new (the codec) implementations should coexist smoothly until the sound card file is updated. For this to work as expected, update the sound card .set_bias_level callback to not touch codec->dapm.bias_level. While extending the cx20442 structure, drop unused control_type member. Created against linxu-3.2-rc6, tested on top of patch 1/4 "ARM: OMAP1: ams-delta: set up a regulator over the modem reset GPIO pin". Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Changes against initial version: * don't track the regulator enable/disable state, compare new bias level against the old one instead; thanks to Mark Brown for suggesting this change, * update the Amstrad Delta sound card file to not touch the codec bias level for the above change to work as expected, * display a warning if getting the regulator fails. sound/soc/codecs/cx20442.c | 48 ++++++++++++++++++++++++++++++++++++++++++- sound/soc/omap/ams-delta.c | 8 ++---- 2 files changed, 49 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index bc7067d..941c527 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -16,6 +16,7 @@ #include <linux/tty.h> #include <linux/slab.h> #include <linux/module.h> +#include <linux/regulator/consumer.h> #include <sound/core.h> #include <sound/initval.h> @@ -25,8 +26,8 @@ struct cx20442_priv { - enum snd_soc_control_type control_type; void *control_data; + struct regulator *por; }; #define CX20442_PM 0x0 @@ -324,6 +325,38 @@ static struct snd_soc_dai_driver cx20442_dai = { }, }; +static int cx20442_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct cx20442_priv *cx20442 = snd_soc_codec_get_drvdata(codec); + int err = 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (codec->dapm.bias_level != SND_SOC_BIAS_STANDBY) + break; + if (IS_ERR(cx20442->por)) + err = PTR_ERR(cx20442->por); + else + err = regulator_enable(cx20442->por); + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level != SND_SOC_BIAS_PREPARE) + break; + if (IS_ERR(cx20442->por)) + err = PTR_ERR(cx20442->por); + else + err = regulator_disable(cx20442->por); + break; + default: + break; + } + if (!err) + codec->dapm.bias_level = level; + + return err; +} + static int cx20442_codec_probe(struct snd_soc_codec *codec) { struct cx20442_priv *cx20442; @@ -331,9 +364,13 @@ static int cx20442_codec_probe(struct snd_soc_codec *codec) cx20442 = kzalloc(sizeof(struct cx20442_priv), GFP_KERNEL); if (cx20442 == NULL) return -ENOMEM; - snd_soc_codec_set_drvdata(codec, cx20442); + cx20442->por = regulator_get(codec->dev, "POR"); + if (IS_ERR(cx20442->por)) + dev_warn(codec->dev, "failed to get the regulator"); cx20442->control_data = NULL; + + snd_soc_codec_set_drvdata(codec, cx20442); codec->hw_write = NULL; codec->card->pop_time = 0; @@ -350,6 +387,12 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec) tty_hangup(tty); } + if (!IS_ERR(cx20442->por)) { + /* should be already in STANDBY, hence disabled */ + regulator_put(cx20442->por); + } + + snd_soc_codec_set_drvdata(codec, NULL); kfree(cx20442); return 0; } @@ -359,6 +402,7 @@ static const u8 cx20442_reg; static struct snd_soc_codec_driver cx20442_codec_dev = { .probe = cx20442_codec_probe, .remove = cx20442_codec_remove, + .set_bias_level = cx20442_set_bias_level, .reg_cache_default = &cx20442_reg, .reg_cache_size = 1, .reg_word_size = sizeof(u8), diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index be81bc7..0462c7e 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -431,22 +431,20 @@ static int ams_delta_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_codec *codec = card->rtd->codec; - switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + if (card->dapm.bias_level == SND_SOC_BIAS_OFF) ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, AMS_DELTA_LATCH2_MODEM_NRESET); break; case SND_SOC_BIAS_OFF: - if (codec->dapm.bias_level != SND_SOC_BIAS_OFF) + if (card->dapm.bias_level != SND_SOC_BIAS_OFF) ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, 0); } - codec->dapm.bias_level = level; + card->dapm.bias_level = level; return 0; } -- 1.7.3.4

2 1

[alsa-devel] [PATCH] ASoC: Fix return value of wm8903_gpio_direction_in() and wm8903_gpio_direction_out()
by Axel Lin 02 Jan '12

02 Jan '12

We can't just pass back the return value of snd_soc_update_bits() as it will be 1 if a bit changed rather than zero. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8903.c | 18 ++++++++++++++---- 1 files changed, 14 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index d88b727..c91fb2f 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1777,13 +1777,18 @@ static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); struct snd_soc_codec *codec = wm8903->codec; unsigned int mask, val; + int ret; mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK; val = (WM8903_GPn_FN_GPIO_INPUT << WM8903_GP1_FN_SHIFT) | WM8903_GP1_DIR; - return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - mask, val); + ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, + mask, val); + if (ret < 0) + return ret; + + return 0; } static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset) @@ -1803,13 +1808,18 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip, struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); struct snd_soc_codec *codec = wm8903->codec; unsigned int mask, val; + int ret; mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK | WM8903_GP1_LVL_MASK; val = (WM8903_GPn_FN_GPIO_OUTPUT << WM8903_GP1_FN_SHIFT) | (value << WM8903_GP2_LVL_SHIFT); - return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - mask, val); + ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, + mask, val); + if (ret < 0) + return ret; + + return 0; } static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value) -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/4] ASoC: Convert saarb to table based DAPM init
by Axel Lin 02 Jan '12

02 Jan '12

Also remove a unused ret variable to silence the build warning. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/saarb.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c index b2be225..c34146b 100644 --- a/sound/soc/pxa/saarb.c +++ b/sound/soc/pxa/saarb.c @@ -51,7 +51,7 @@ static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = { }; /* saarb machine audio map */ -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route saarb_audio_map[] = { {"Headset Stereophone", NULL, "HS1"}, {"Headset Stereophone", NULL, "HS2"}, @@ -121,17 +121,17 @@ static struct snd_soc_card snd_soc_card_saarb = { .owner = THIS_MODULE, .dai_link = saarb_dai, .num_links = ARRAY_SIZE(saarb_dai), + + .dapm_widgets = saarb_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(saarb_dapm_widgets), + .dapm_routes = saarb_audio_map, + .num_dapm_routes = ARRAY_SIZE(saarb_audio_map), }; static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - snd_soc_dapm_new_controls(dapm, saarb_dapm_widgets, - ARRAY_SIZE(saarb_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* connected pins */ snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); -- 1.7.5.4

2 4

[alsa-devel] [PATCH 1/4] ASoC: Use dai_fmt in saarb machine driver
by Axel Lin 02 Jan '12

02 Jan '12

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/saarb.c | 11 ++--------- 1 files changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c index 2e21712..b2be225 100644 --- a/sound/soc/pxa/saarb.c +++ b/sound/soc/pxa/saarb.c @@ -92,15 +92,6 @@ static int saarb_i2s_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); return ret; @@ -119,6 +110,8 @@ static struct snd_soc_dai_link saarb_dai[] = { .platform_name = "pxa-pcm-audio", .codec_name = "88pm860x-codec", .init = saarb_pm860x_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &saarb_i2s_ops, }, }; -- 1.7.5.4

3 8

[alsa-devel] [PATCH 1/3] ASoC: Fix return value of ak4641_pcm_set_dai_fmt()
by Axel Lin 02 Jan '12

02 Jan '12

We can't just pass back the return value of snd_soc_update_bits() as it will be 1 if a bit changed rather than zero. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ak4641.c | 7 ++++++- 1 files changed, 6 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 266ebea..c4d165a 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -339,6 +339,7 @@ static int ak4641_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; u8 btif; + int ret; /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -358,7 +359,11 @@ static int ak4641_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - return snd_soc_update_bits(codec, AK4641_BTIF, (0x3 << 5), btif); + ret = snd_soc_update_bits(codec, AK4641_BTIF, (0x3 << 5), btif); + if (ret < 0) + return ret; + + return 0; } static int ak4641_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, -- 1.7.5.4

2 3

Re: [alsa-devel] [PATCH 0/10] use devm_ functions
by Mark Brown 02 Jan '12

02 Jan '12

On Thu, Dec 29, 2011 at 05:51:19PM +0100, Julia Lawall wrote: > The following is a semantic patch (http://coccinelle.lip6.fr/) that > introduces uses of various devm_ functions. The main strategy is Applied all except the jz4740 patch which Axel already did, thanks.

1 0

[alsa-devel] [PATCH 01/14] ASoC: Convert alc5623 to devm_kzalloc()
by Axel Lin 02 Jan '12

02 Jan '12

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/alc5623.c | 11 +++-------- 1 files changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 6a9b621..3feee56 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -1022,7 +1022,8 @@ static int alc5623_i2c_probe(struct i2c_client *client, dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2); - alc5623 = kzalloc(sizeof(struct alc5623_priv), GFP_KERNEL); + alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv), + GFP_KERNEL); if (alc5623 == NULL) return -ENOMEM; @@ -1044,7 +1045,6 @@ static int alc5623_i2c_probe(struct i2c_client *client, alc5623_dai.name = "alc5623-hifi"; break; default: - kfree(alc5623); return -EINVAL; } @@ -1053,20 +1053,15 @@ static int alc5623_i2c_probe(struct i2c_client *client, ret = snd_soc_register_codec(&client->dev, &soc_codec_device_alc5623, &alc5623_dai, 1); - if (ret != 0) { + if (ret != 0) dev_err(&client->dev, "Failed to register codec: %d\n", ret); - kfree(alc5623); - } return ret; } static int alc5623_i2c_remove(struct i2c_client *client) { - struct alc5623_priv *alc5623 = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - kfree(alc5623); return 0; } -- 1.7.5.4

3 17

[alsa-devel] Non-cacheable memory to alsa ring buffer
by Babu, Ramesh 02 Jan '12

02 Jan '12

Hi How can I allocate non-cacheable region for ring buffer? I have an issue, where my DMA is not fetching right data from Memory. I suspect that caching might be issue here. I am using the below api to allocate memory: retval = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_KERNEL), RING_MIN_BUFFER, RING_MAX_BUFFER); Where, RING_MIN_BUFFER = RING_MAX_BUFFER = (8*1024) Thanks Ramesh

1 0

[alsa-devel] [PATCH v8 3/3 RESEND] ASoC: da7210: Add support for line input and mic
by Ashish Chavan 02 Jan '12

02 Jan '12

DA7210 has three line inputs (AUX1 Left, AUX1 Right and AUX2) and a stereo MIC. This patch adds gain controls for MIC, AUX1, AUX2 as well as INPGA. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v7: - Added DAPM support for AUX1, AUX2 and INPGA (Moved here from DAPM patch) Changes since v2: - Removed static enable of mic and aux, as now DAPM will take care of that Changes since v1: - Removed explicit setting of default gains - Removed control to set mic bias voltage --- sound/soc/codecs/da7210.c | 77 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 77 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 0ebcbd5..ff084a6 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -181,9 +181,14 @@ /* AUX1_L bit fields */ #define DA7210_AUX1_L_VOL (0x3F << 0) +#define DA7210_AUX1_L_EN (1 << 7) /* AUX1_R bit fields */ #define DA7210_AUX1_R_VOL (0x3F << 0) +#define DA7210_AUX1_R_EN (1 << 7) + +/* AUX2 bit fields */ +#define DA7210_AUX2_EN (1 << 3) /* Minimum INPGA and AUX1 volume to enable noise suppression */ #define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ @@ -234,9 +239,19 @@ static const unsigned int mono_vol_tlv[] = { 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) }; +static const unsigned int aux1_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -48dB to 21dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) +}; + static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); /* ADC and DAC high pass filter f0 value */ static const char const *da7210_hpf_cutoff_txt[] = { @@ -344,6 +359,17 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, mono_vol_tlv), + SOC_DOUBLE_R_TLV("Mic Capture Volume", + DA7210_MIC_L, DA7210_MIC_R, + 0, 0x5, 0, mic_vol_tlv), + SOC_DOUBLE_R_TLV("Aux1 Capture Volume", + DA7210_AUX1_L, DA7210_AUX1_R, + 0, 0x3f, 0, aux1_vol_tlv), + SOC_SINGLE_TLV("Aux2 Capture Volume", DA7210_AUX2, 0, 0x3, 0, + aux2_vol_tlv), + SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, + inpga_gain_tlv), + /* DAC Equalizer controls */ SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1, @@ -421,26 +447,42 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = { SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0), SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_INMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_INMIX_L, 4, 1, 0), }; /* In Mixer Right */ static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = { SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0), SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_INMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_INMIX_R, 4, 1, 0), }; /* Out Mixer Left */ static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = { + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_OUTMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_L, 3, 1, 0), SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0), }; /* Out Mixer Right */ static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_OUTMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_R, 3, 1, 0), SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), }; /* Mono Mixer */ static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = { + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUT2, 3, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUT2, 4, 1, 0), SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0), SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0), }; @@ -451,14 +493,23 @@ static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { /* Input Lines */ SND_SOC_DAPM_INPUT("MICL"), SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("AUX1L"), + SND_SOC_DAPM_INPUT("AUX1R"), + SND_SOC_DAPM_INPUT("AUX2"), /* Input PGAs */ SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0), SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Left", DA7210_STARTUP3, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Right", DA7210_STARTUP3, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux2 Mono", DA7210_STARTUP3, 4, 1, NULL, 0), SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0), + /* MICBIAS */ + SND_SOC_DAPM_SUPPLY("Mic Bias", DA7210_MIC_L, 6, 0, NULL, 0), + /* Input Mixers */ SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, &da7210_dapm_inmixl_controls[0], @@ -514,12 +565,21 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Input path */ {"Mic Left", NULL, "MICL"}, {"Mic Right", NULL, "MICR"}, + {"Aux1 Left", NULL, "AUX1L"}, + {"Aux1 Right", NULL, "AUX1R"}, + {"Aux2 Mono", NULL, "AUX2"}, {"In Mixer Left", "Mic Left Switch", "Mic Left"}, {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + {"In Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"In Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Left", "Outmix Left Switch", "Out Mixer Left"}, {"In Mixer Right", "Mic Right Switch", "Mic Right"}, {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + {"In Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"In Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Right", "Outmix Right Switch", "Out Mixer Right"}, {"INPGA Left", NULL, "In Mixer Left"}, {"ADC Left", NULL, "INPGA Left"}, @@ -528,9 +588,20 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { {"ADC Right", NULL, "INPGA Right"}, /* Output path */ + {"Out Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"Out Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Left", "INPGA Left Switch", "INPGA Left"}, + {"Out Mixer Left", "INPGA Right Switch", "INPGA Right"}, {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + + {"Out Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"Out Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Right", "INPGA Right Switch", "INPGA Right"}, + {"Out Mixer Right", "INPGA Left Switch", "INPGA Left"}, {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + {"Mono Mixer", "INPGA Right Switch", "INPGA Right"}, + {"Mono Mixer", "INPGA Left Switch", "INPGA Left"}, {"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"}, {"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"}, @@ -887,6 +958,12 @@ static int da7210_probe(struct snd_soc_codec *codec) snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Enable Aux1 */ + snd_soc_write(codec, DA7210_AUX1_L, DA7210_AUX1_L_EN); + snd_soc_write(codec, DA7210_AUX1_R, DA7210_AUX1_R_EN); + /* Enable Aux2 */ + snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

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[alsa-devel] USB-Audio and XHCI (USB 3)
by Aurélien Leblond 01 Jan '12

01 Jan '12

Hi everyone, I was wondering if anyone managed to run a USB 2 soundcard on a USB3 (XHCI) port? I'm running the kernel 3.2 rc7. With the M-Audio Fast Track Ultra (USB 2): The output of dmesg is this: [ 475.094595] usb 3-1: new high-speed USB device number 2 using xhci_hcd [ 475.108605] usb 3-1: config 1 interface 3 altsetting 0 bulk endpoint 0x7 has invalid maxpacket 8 [ 475.108615] usb 3-1: config 1 interface 3 altsetting 0 bulk endpoint 0x87 has invalid maxpacket 8 [ 475.108958] xhci_hcd 0000:04:00.0: WARN: short transfer on control ep [ 475.109328] xhci_hcd 0000:04:00.0: WARN: short transfer on control ep [ 475.109879] xhci_hcd 0000:04:00.0: WARN: short transfer on control ep [ 475.110564] xhci_hcd 0000:04:00.0: ERROR: unexpected command completion code 0x11. [ 475.110585] usb 3-1: can't set config #1, error -22 The snd-usb-audio module is not probed automatically, and if I modprobe it manually, no sound card is detected. The error message is described in this bug: https://bugzilla.novell.com/show_bug.cgi?id=679944 but if I understand well, kernel wise, everything is "ok" and the issue lies in Alsa The output of lsusb is: Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 001 Device 002: ID 8087:0020 Intel Corp. Integrated Rate Matching Hub Bus 002 Device 002: ID 8087:0020 Intel Corp. Integrated Rate Matching Hub Bus 001 Device 003: ID 050d:0307 Belkin Components Bus 002 Device 003: ID 0408:1af1 Quanta Computer, Inc. Bus 001 Device 004: ID 046d:c51b Logitech, Inc. V220 Cordless Optical Mouse for Notebooks Bus 003 Device 002: ID 0763:2080 Midiman M-Audio RunTime DFU The Edirol UA-25Ex (USB 1) works fine, I get a few warnings in dmesg but the sound card is detected and functional... Let me know if I should provide any more info, or outputs, anything that can help. Kind Regards, Aurélien

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