October 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [patch] [ALSA] hwdep: silence integer overflow warning
by Dan Carpenter 20 Oct '11

20 Oct '11

Smatch complains that if device is INT_MAX then device + 1 can overflow. It just means we would have an annoying loop while we check all the devices from -2147483648 to SNDRV_MINOR_HWDEPS. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index c7ceb28..edfdf6d 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -273,7 +273,14 @@ static int snd_hwdep_control_ioctl(struct snd_card *card, if (get_user(device, (int __user *)arg)) return -EFAULT; mutex_lock(&register_mutex); - device = device < 0 ? 0 : device + 1; + + if (device < 0) + device = 0; + else if (device > SNDRV_MINOR_HWDEPS) + device = SNDRV_MINOR_HWDEPS; + else + device++; + while (device < SNDRV_MINOR_HWDEPS) { if (snd_hwdep_search(card, device)) break;

1 0

[alsa-devel] [PATCH 3/3] sound/soc/au1x: keep pointer to resource so it can be freed
by Julia Lawall 19 Oct '11

19 Oct '11

From: Julia Lawall <julia(a)diku.dk> Add a new variable for storing resources accessed subsequent to the one accessed using request_mem_region, so the one accessed using request_mem_region can be released if needed. This code is also missing some calls to iounmap. The semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @r@ expression E, E1; identifier f; statement S1,S2,S3; @@ if (E == NULL) { ... when != if (E == NULL || ...) S1 else S2 when != E = E1 *E->f ... when any return ...; } else S3 // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/au1x/ac97c.c | 22 ++++++++++++---------- sound/soc/au1x/i2sc.c | 22 ++++++++++++---------- 2 files changed, 24 insertions(+), 20 deletions(-) diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index 13802ff..0188643 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -226,7 +226,7 @@ static struct snd_soc_dai_driver au1xac97c_dai_driver = { static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) { int ret; - struct resource *r; + struct resource *r, *r1; struct au1xpsc_audio_data *ctx; ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); @@ -249,15 +249,15 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) if (!ctx->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + r1 = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r1) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r1->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + r1 = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r1) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r1->start; /* switch it on */ WR(ctx, AC97_ENABLE, EN_D | EN_CE); @@ -270,11 +270,13 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &au1xac97c_dai_driver); if (ret) - goto out1; + goto out2; ac97c_workdata = ctx; return 0; +out2: + iounmap(ctx->mmio); out1: release_mem_region(r->start, resource_size(r)); out0: diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 19e0d2a..3725c77 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -228,7 +228,7 @@ static struct snd_soc_dai_driver au1xi2s_dai_driver = { static int __devinit au1xi2s_drvprobe(struct platform_device *pdev) { int ret; - struct resource *r; + struct resource *r, *r1; struct au1xpsc_audio_data *ctx; ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); @@ -249,24 +249,26 @@ static int __devinit au1xi2s_drvprobe(struct platform_device *pdev) if (!ctx->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + r1 = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r1) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r1->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + r1 = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r1) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r1->start; platform_set_drvdata(pdev, ctx); ret = snd_soc_register_dai(&pdev->dev, &au1xi2s_dai_driver); if (ret) - goto out1; + goto out2; return 0; +out2: + iounmap(ctx->mmio); out1: release_mem_region(r->start, resource_size(r)); out0:

4 6

[alsa-devel] [PATCH] ASoC: ak4641: Use SND_SOC_DAPM_DAC for Voice Playback stream widget
by Axel Lin 19 Oct '11

19 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ak4641.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index f5125ae..7783858 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -225,7 +225,7 @@ static const struct snd_soc_dapm_widget ak4641_dapm_widgets[] = { SND_SOC_DAPM_PGA("Mono Out 2", AK4641_PM2, 3, 0, NULL, 0), SND_SOC_DAPM_ADC("Voice ADC", "Voice Capture", AK4641_BTIF, 0, 0), - SND_SOC_DAPM_ADC("Voice DAC", "Voice Playback", AK4641_BTIF, 1, 0), + SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AK4641_BTIF, 1, 0), SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4641_MIC, 3, 0), SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4641_MIC, 4, 0), -- 1.7.5.4

3 2

[alsa-devel] [PATCH] SoC: ak4641: Remove unused codec field from struct ak4641_priv
by Axel Lin 19 Oct '11

19 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ak4641.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 7a64e58..f5125ae 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -31,7 +31,6 @@ /* codec private data */ struct ak4641_priv { - struct snd_soc_codec *codec; unsigned int sysclk; int deemph; int playback_fs; -- 1.7.5.4

3 2

[alsa-devel] [PATCH v6 2/2] ASoC: da7210: Add support for ALC and Noise suppression
by Ashish Chavan 19 Oct '11

19 Oct '11

This patch adds controls to set following ALC parameters, - Max gain, Min gain, Noise gain, Attack rate, Release rate and delay It also adds a switch to enable/disable noise suppression. As per DA7210 data sheet, ALC and noise suppression can be enabled only if certain conditions are met. This condition checks are handled by simply using "_EXT" version of controls to capture change events. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v5: - Changed error code from -EPERM to EINVAL - Used bit fields instead of magic numbers Changes since v1: - Reading from register instead of cache - Made debug message more specific about error - Changed "Gain" to "Volume" in names of ALC controls --- sound/soc/codecs/da7210.c | 110 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 110 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a31be70..f0a1999 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -26,11 +26,15 @@ #include <sound/tlv.h> /* DA7210 register space */ +#define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX1_L 0x09 +#define DA7210_AUX1_R 0x0A #define DA7210_AUX2 0x0B +#define DA7210_IN_GAIN 0x0C #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F @@ -59,6 +63,12 @@ #define DA7210_PLL_DIV2 0x2A #define DA7210_PLL_DIV3 0x2B #define DA7210_PLL 0x2C +#define DA7210_ALC_MAX 0x83 +#define DA7210_ALC_MIN 0x84 +#define DA7210_ALC_NOIS 0x85 +#define DA7210_ALC_ATT 0x86 +#define DA7210_ALC_REL 0x87 +#define DA7210_ALC_DEL 0x88 #define DA7210_A_HID_UNLOCK 0x8A #define DA7210_A_TEST_UNLOCK 0x8B #define DA7210_A_PLL1 0x90 @@ -81,6 +91,7 @@ #define DA7210_IN_R_EN (1 << 7) /* ADC bit fields */ +#define DA7210_ADC_ALC_EN (1 << 0) #define DA7210_ADC_L_EN (1 << 3) #define DA7210_ADC_R_EN (1 << 7) @@ -150,6 +161,29 @@ /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) +/* CONTROL bit fields */ +#define DA7210_NOISE_SUP_EN (1 << 3) + +/* IN_GAIN bit fields */ +#define DA7210_INPGA_L_VOL (0x0F << 0) +#define DA7210_INPGA_R_VOL (0xF0 << 0) + +/* ZERO_CROSS bit fields */ +#define DA7210_AUX1_L_ZC (1 << 0) +#define DA7210_AUX1_R_ZC (1 << 1) +#define DA7210_HP_L_ZC (1 << 6) +#define DA7210_HP_R_ZC (1 << 7) + +/* AUX1_L bit fields */ +#define DA7210_AUX1_L_VOL (0x3F << 0) + +/* AUX1_R bit fields */ +#define DA7210_AUX1_R_VOL (0x3F << 0) + +/* Minimum INPGA and AUX1 volume to enable noise suppression */ +#define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ +#define DA7210_AUX1_MIN_VOL_NS 0x35 /* 6dB */ + #define DA7210_VERSION "0.0.1" /* @@ -202,6 +236,69 @@ static const char *da7210_hp_mode_txt[] = { static const struct soc_enum da7210_hp_mode_sel = SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); +/* ALC can be enabled only if noise suppression is disabled */ +static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0]) { + /* Check if noise suppression is enabled */ + if (snd_soc_read(codec, DA7210_CONTROL) & DA7210_NOISE_SUP_EN) { + dev_dbg(codec->dev, + "Disable noise suppression to enable ALC\n"); + return -EINVAL; + } + } + /* If all conditions are met or we are actually disabling ALC */ + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +/* Noise suppression can be enabled only if following conditions are met + * ALC disabled + * ZC enabled for HP and AUX1 PGA + * INPGA_L_VOL and INPGA_R_VOL >= 10.5 dB + * AUX1_L_VOL and AUX1_R_VOL >= 6 dB + */ +static int da7210_put_noise_sup_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 val; + + if (ucontrol->value.integer.value[0]) { + /* Check if ALC is enabled */ + if (snd_soc_read(codec, DA7210_ADC) & DA7210_ADC_ALC_EN) + goto err; + + /* Check ZC for HP and AUX1 PGA */ + if ((snd_soc_read(codec, DA7210_ZERO_CROSS) & + (DA7210_AUX1_L_ZC | DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | + DA7210_HP_R_ZC)) != (DA7210_AUX1_L_ZC | + DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | DA7210_HP_R_ZC)) + goto err; + + /* Check INPGA_L_VOL and INPGA_R_VOL */ + val = snd_soc_read(codec, DA7210_IN_GAIN); + if (((val & DA7210_INPGA_L_VOL) < DA7210_INPGA_MIN_VOL_NS) || + (((val & DA7210_INPGA_R_VOL) >> 4) < + DA7210_INPGA_MIN_VOL_NS)) + goto err; + + /* Check AUX1_L_VOL and AUX1_R_VOL */ + if (((snd_soc_read(codec, DA7210_AUX1_L) & DA7210_AUX1_L_VOL) < + DA7210_AUX1_MIN_VOL_NS) || + ((snd_soc_read(codec, DA7210_AUX1_R) & DA7210_AUX1_R_VOL) < + DA7210_AUX1_MIN_VOL_NS)) + goto err; + } + /* If all conditions are met or we are actually disabling Noise sup */ + return snd_soc_put_volsw(kcontrol, ucontrol); + +err: + return -EINVAL; +} + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -260,6 +357,19 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), SOC_ENUM("Headphone Class", da7210_hp_mode_sel), + + /* ALC controls */ + SOC_SINGLE_EXT("ALC Enable Switch", DA7210_ADC, 0, 1, 0, + snd_soc_get_volsw, da7210_put_alc_sw), + SOC_SINGLE("ALC Capture Max Volume", DA7210_ALC_MAX, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Min Volume", DA7210_ALC_MIN, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Noise Volume", DA7210_ALC_NOIS, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Attack Rate", DA7210_ALC_ATT, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Rate", DA7210_ALC_REL, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Delay", DA7210_ALC_DEL, 0, 0xFF, 0), + + SOC_SINGLE_EXT("Noise Suppression Enable Switch", DA7210_CONTROL, 3, 1, + 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), }; /* Codec private data */ -- 1.7.1

3 2

[alsa-devel] [PATCH v6 1/2] ASoC: da7210: Add support for mute and zero cross controls
by Ashish Chavan 19 Oct '11

19 Oct '11

This patch adds support for below set of controls, (1) Mute controls for MIC, AUX and ADC (2) Zero cross controls for head phone, AUX, INPGA and line out (3) Head phone mode selection - class H or G It also adds digital_mute() call back. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v4: - Converted ADC mute controls to a single stereo control Changes since v2: - Added digital_mute() call back and removed control for "DAC Mute Switch" - Renamed mute control switches Changes since v1: - Changed Mic mute control to a double control - Fixed a typo --- sound/soc/codecs/da7210.c | 44 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 44 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 480fc6d..a31be70 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -30,6 +30,7 @@ #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX2 0x0B #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F @@ -41,6 +42,7 @@ #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_SOFTMUTE 0x18 #define DA7210_DAC_EQ1_2 0x19 #define DA7210_DAC_EQ3_4 0x1A #define DA7210_DAC_EQ5 0x1B @@ -49,6 +51,7 @@ #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 +#define DA7210_ZERO_CROSS 0x24 #define DA7210_DAI_SRC_SEL 0x25 #define DA7210_DAI_CFG1 0x26 #define DA7210_DAI_CFG3 0x28 @@ -144,6 +147,9 @@ #define DA7210_PLL_FS_96000 (0xF << 0) #define DA7210_PLL_EN (0x1 << 7) +/* SOFTMUTE bit fields */ +#define DA7210_RAMP_EN (1 << 6) + #define DA7210_VERSION "0.0.1" /* @@ -189,6 +195,13 @@ static const struct soc_enum da7210_dac_vf_cutoff = static const struct soc_enum da7210_adc_vf_cutoff = SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt); +static const char *da7210_hp_mode_txt[] = { + "Class H", "Class G" +}; + +static const struct soc_enum da7210_hp_mode_sel = + SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -232,6 +245,21 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_ENUM("ADC HPF Cutoff", da7210_adc_hpf_cutoff), SOC_SINGLE("ADC Voice Mode Switch", DA7210_ADC_HPF, 7, 1, 0), SOC_ENUM("ADC Voice Cutoff", da7210_adc_vf_cutoff), + + /* Mute controls */ + SOC_DOUBLE_R("Mic Capture Switch", DA7210_MIC_L, DA7210_MIC_R, 3, 1, 0), + SOC_SINGLE("Aux2 Capture Switch", DA7210_AUX2, 2, 1, 0), + SOC_DOUBLE("ADC Capture Switch", DA7210_ADC, 2, 6, 1, 0), + SOC_SINGLE("Digital Soft Mute Switch", DA7210_SOFTMUTE, 7, 1, 0), + SOC_SINGLE("Digital Soft Mute Rate", DA7210_SOFTMUTE, 0, 0x7, 0), + + /* Zero cross controls */ + SOC_DOUBLE("Aux1 ZC Switch", DA7210_ZERO_CROSS, 0, 1, 1, 0), + SOC_DOUBLE("In PGA ZC Switch", DA7210_ZERO_CROSS, 2, 3, 1, 0), + SOC_DOUBLE("Lineout ZC Switch", DA7210_ZERO_CROSS, 4, 5, 1, 0), + SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), + + SOC_ENUM("Headphone Class", da7210_hp_mode_sel), }; /* Codec private data */ @@ -448,6 +476,18 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return 0; } +static int da7210_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = snd_soc_read(codec, DA7210_DAC_HPF) & 0xFB; + + if (mute) + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg | 0x4); + else + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg); + return 0; +} + #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -456,6 +496,7 @@ static struct snd_soc_dai_ops da7210_dai_ops = { .startup = da7210_startup, .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .digital_mute = da7210_mute, }; static struct snd_soc_dai_driver da7210_dai = { @@ -545,6 +586,9 @@ static int da7210_probe(struct snd_soc_codec *codec) DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); + /* Enable ramp mode for DAC gain update */ + snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

3 2

[alsa-devel] [PATCH RESEND] ASoC: ak4641: Remove unused codec field from struct ak4641_priv
by Axel Lin 19 Oct '11

19 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Sorry. a typo in subject line. here is a resend. sound/soc/codecs/ak4641.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 7a64e58..f5125ae 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -31,7 +31,6 @@ /* codec private data */ struct ak4641_priv { - struct snd_soc_codec *codec; unsigned int sysclk; int deemph; int playback_fs; -- 1.7.5.4

1 0

[alsa-devel] ASoC: Correct way to set private data in machine driver?
by Abdullah, Omair M 19 Oct '11

19 Oct '11

Hi all, Looking at the code of soc and any machine driver, soc-core uses dev_set_drvdata() and dev_get_drvdata() to pass a snd_soc_card structure. This needs to be set in the probe to the machine driver's snd_soc_card structure. Then how can the machine driver's private data be passed around (eg. in probe and remove)? regards, -- Omair

2 1

[PATCH] ASoC: au1x: fix error paths in ac97c/i2cs probe callbacks.
by Manuel Lauss 19 Oct '11

19 Oct '11

add iounmap() and release the correct mem resource. Signed-off-by: Manuel Lauss <manuel.lauss(a)googlemail.com> --- sound/soc/au1x/ac97c.c | 7 +++++-- sound/soc/au1x/i2sc.c | 7 +++++-- 2 files changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index 13802ff..4ec39aa 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -251,12 +251,12 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) r = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!r) - goto out1; + goto out2; ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; r = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!r) - goto out1; + goto out2; ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; /* switch it on */ @@ -275,7 +275,10 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) ac97c_workdata = ctx; return 0; +out2: + iounmap(ctx->mmio); out1: + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); release_mem_region(r->start, resource_size(r)); out0: kfree(ctx); diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 19e0d2a..59fb31a 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -251,12 +251,12 @@ static int __devinit au1xi2s_drvprobe(struct platform_device *pdev) r = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!r) - goto out1; + goto out2; ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; r = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!r) - goto out1; + goto out2; ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; platform_set_drvdata(pdev, ctx); @@ -267,7 +267,10 @@ static int __devinit au1xi2s_drvprobe(struct platform_device *pdev) return 0; +out2: + iounmap(ctx->mmio); out1: + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); release_mem_region(r->start, resource_size(r)); out0: kfree(ctx); -- 1.7.7

1 0

[alsa-devel] [PATCH 1/2] ASoC: sgtl5000: Fix define for SGTL5000_BIAS_R_MASK
by Axel Lin 19 Oct '11

19 Oct '11

According to the datasheet: CHIP_MIC_CTRL 0x002A BITS[9:8] BIAS_RESISTOR 0x0 = Powerd off 0x1 = 2.0 kohm 0x2 = 4.0 kohm 0x3 = 8.0 kohm Thus SGTL5000_BIAS_R_MASK should be defined as 0x0300 instead of 0x0200. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/sgtl5000.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index eec3ab3..8a9f435 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -280,7 +280,7 @@ /* * SGTL5000_CHIP_MIC_CTRL */ -#define SGTL5000_BIAS_R_MASK 0x0200 +#define SGTL5000_BIAS_R_MASK 0x0300 #define SGTL5000_BIAS_R_SHIFT 8 #define SGTL5000_BIAS_R_WIDTH 2 #define SGTL5000_BIAS_R_off 0x0 -- 1.7.5.4

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