From 47e039413cacee70229ebbf6de5a8e3b27e6f057 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 23 Jan 2015 16:21:36 +0100 Subject: ASoC: Add support for allocating AC'97 device before registering it In some cases it is necessary to before additional operations after the device has been initialized and before the device is registered. This can for example be resetting the device. This patch introduces a new function snd_soc_alloc_ac97_codec() which is similar to snd_soc_new_ac97_codec() except that it does not register the device. Any users of snd_soc_alloc_ac97_codec() are responsible for calling device_add() manually. Fixes: 6794f709b712 ("ASoC: ac97: Drop delayed device registration") Reported-by: Manuel Lauss Signed-off-by: Lars-Peter Clausen Tested-by: Manuel Lauss Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-ac97.c | 36 ++++++++++++++++++++++++++++++------ 1 file changed, 30 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 2e10e9a38376..08d7259bbaab 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -48,15 +48,18 @@ static void soc_ac97_device_release(struct device *dev) } /** - * snd_soc_new_ac97_codec - initailise AC97 device - * @codec: audio codec + * snd_soc_alloc_ac97_codec() - Allocate new a AC'97 device + * @codec: The CODEC for which to create the AC'97 device * - * Initialises AC97 codec resources for use by ad-hoc devices only. + * Allocated a new snd_ac97 device and intializes it, but does not yet register + * it. The caller is responsible to either call device_add(&ac97->dev) to + * register the device, or to call put_device(&ac97->dev) to free the device. + * + * Returns: A snd_ac97 device or a PTR_ERR in case of an error. */ -struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec) +struct snd_ac97 *snd_soc_alloc_ac97_codec(struct snd_soc_codec *codec) { struct snd_ac97 *ac97; - int ret; ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL); if (ac97 == NULL) @@ -73,7 +76,28 @@ struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec) codec->component.card->snd_card->number, 0, codec->component.name); - ret = device_register(&ac97->dev); + device_initialize(&ac97->dev); + + return ac97; +} +EXPORT_SYMBOL(snd_soc_alloc_ac97_codec); + +/** + * snd_soc_new_ac97_codec - initailise AC97 device + * @codec: audio codec + * + * Initialises AC97 codec resources for use by ad-hoc devices only. + */ +struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec) +{ + struct snd_ac97 *ac97; + int ret; + + ac97 = snd_soc_alloc_ac97_codec(codec); + if (IS_ERR(ac97)) + return ac97; + + ret = device_add(&ac97->dev); if (ret) { put_device(&ac97->dev); return ERR_PTR(ret); -- cgit v1.2.3 From f8d71be5553a3fbc20363243ecfc11dcdd1e18fe Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 23 Jan 2015 16:21:37 +0100 Subject: ASoC: wm97xx: Reset AC'97 device before registering it The wm97xx touchscreen driver binds itself to the snd_ac97 device that gets registered by the CODEC driver and expects that the device has already been reset. Before commit 6794f709b712 ("ASoC: ac97: Drop delayed device registration") the device was only registered after the probe function of the CODEC driver had finished running, but starting with the mentioned commit the device is registered as soon as snd_soc_new_ac97_codec() is called. This causes the touchscreen driver to no longer work. Modify the CODEC drivers to use snd_soc_alloc_ac97_codec() instead of snd_soc_new_ac97_codec() and make sure that the AC'97 device is reset before the snd_ac97 device gets registered. Fixes: 6794f709b712 ("ASoC: ac97: Drop delayed device registration") Reported-by: Manuel Lauss Signed-off-by: Lars-Peter Clausen Tested-by: Manuel Lauss Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm9705.c | 16 ++++++++++------ sound/soc/codecs/wm9712.c | 12 ++++++++---- sound/soc/codecs/wm9713.c | 12 ++++++++---- 3 files changed, 26 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 3eddb18fefd1..5cc457ef8894 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -344,23 +344,27 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) struct snd_ac97 *ac97; int ret = 0; - ac97 = snd_soc_new_ac97_codec(codec); + ac97 = snd_soc_alloc_ac97_codec(codec); if (IS_ERR(ac97)) { ret = PTR_ERR(ac97); dev_err(codec->dev, "Failed to register AC97 codec\n"); return ret; } - snd_soc_codec_set_drvdata(codec, ac97); - ret = wm9705_reset(codec); if (ret) - goto reset_err; + goto err_put_device; + + ret = device_add(&ac97->dev); + if (ret) + goto err_put_device; + + snd_soc_codec_set_drvdata(codec, ac97); return 0; -reset_err: - snd_soc_free_ac97_codec(ac97); +err_put_device: + put_device(&ac97->dev); return ret; } diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index e04643d2bb24..9517571e820d 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -666,7 +666,7 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); int ret = 0; - wm9712->ac97 = snd_soc_new_ac97_codec(codec); + wm9712->ac97 = snd_soc_alloc_ac97_codec(codec); if (IS_ERR(wm9712->ac97)) { ret = PTR_ERR(wm9712->ac97); dev_err(codec->dev, "Failed to register AC97 codec: %d\n", ret); @@ -675,15 +675,19 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) ret = wm9712_reset(codec, 0); if (ret < 0) - goto reset_err; + goto err_put_device; + + ret = device_add(&wm9712->ac97->dev); + if (ret) + goto err_put_device; /* set alc mux to none */ ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); return 0; -reset_err: - snd_soc_free_ac97_codec(wm9712->ac97); +err_put_device: + put_device(&wm9712->ac97->dev); return ret; } diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 71b9d5b0734d..6ab1122a3872 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1225,7 +1225,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); int ret = 0, reg; - wm9713->ac97 = snd_soc_new_ac97_codec(codec); + wm9713->ac97 = snd_soc_alloc_ac97_codec(codec); if (IS_ERR(wm9713->ac97)) return PTR_ERR(wm9713->ac97); @@ -1234,7 +1234,11 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) wm9713_reset(codec, 0); ret = wm9713_reset(codec, 1); if (ret < 0) - goto reset_err; + goto err_put_device; + + ret = device_add(&wm9713->ac97->dev); + if (ret) + goto err_put_device; /* unmute the adc - move to kcontrol */ reg = ac97_read(codec, AC97_CD) & 0x7fff; @@ -1242,8 +1246,8 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) return 0; -reset_err: - snd_soc_free_ac97_codec(wm9713->ac97); +err_put_device: + put_device(&wm9713->ac97->dev); return ret; } -- cgit v1.2.3 From 54d96a40e0dfb5aa2eea0b010ddc1c7e8742e364 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 23 Jan 2015 14:51:09 +0800 Subject: ASoC: rt286: Fix capture volume setting issue The purpose of rt286_adc_event is to mute/numnte the ADC mixer. However, it will also set the capture volume to default value. As a result, "ADC0 Capture Volume" is not working if it is set before capture start. This patch remove rt286_adc_event and add "ADC0 Capture Switch" to mute/unmute ADC mixer. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 38 ++++++-------------------------------- 1 file changed, 6 insertions(+), 32 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 1d1c7f8a9af2..847cc4b9bee5 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -417,6 +417,8 @@ static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); static const struct snd_kcontrol_new rt286_snd_controls[] = { SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT286_DACL_GAIN, RT286_DACR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R("ADC0 Capture Switch", RT286_ADCL_GAIN, + RT286_ADCR_GAIN, 7, 1, 1), SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT286_ADCL_GAIN, RT286_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv), SOC_SINGLE_TLV("AMIC Volume", RT286_MIC_GAIN, @@ -538,32 +540,6 @@ static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, return 0; } -static int rt286_adc_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - unsigned int nid; - - nid = (w->reg >> 20) & 0xff; - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - snd_soc_update_bits(codec, - VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), - 0x7080, 0x7000); - break; - case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(codec, - VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), - 0x7080, 0x7080); - break; - default: - return 0; - } - - return 0; -} - static int rt286_vref_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -667,12 +643,10 @@ static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), /* ADC Mux */ - SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1, - &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | - SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1, - &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | - SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1, + &rt286_adc0_mux), + SND_SOC_DAPM_MUX("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1, + &rt286_adc1_mux), /* Audio Interface */ SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), -- cgit v1.2.3 From 3463667aa171ed5359d63a6195e65d457aa6eb2f Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 23 Jan 2015 14:15:30 +0200 Subject: ASoC: rt5640: Add RT5642 ACPI ID for Intel Baytrail Asus T100TAF uses ACPI ID "10EC5642" for its audio codec. I suppose it is updated ACPI ID for the RT5642 codec since some earlier platforms are using "10EC5640" with the RT5642 too. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index c3f2decd643c..1ff726c29249 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2124,6 +2124,7 @@ MODULE_DEVICE_TABLE(of, rt5640_of_match); static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, { "10EC5640", 0 }, + { "10EC5642", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); -- cgit v1.2.3 From b0402717c987c7e1150ae7ad800a9eb75664012f Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Thomas=20Niederpr=C3=BCm?= Date: Thu, 22 Jan 2015 00:02:01 +0100 Subject: ASoC: sta32x: correct bit shift value for IDE register MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The IDE bit in the CONFF register is the third bit not the fourth. Signed-off-by: Thomas Niederprüm Signed-off-by: Mark Brown --- sound/soc/codecs/sta32x.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h index d8e32a6262ee..d3191c983d71 100644 --- a/sound/soc/codecs/sta32x.h +++ b/sound/soc/codecs/sta32x.h @@ -131,7 +131,7 @@ #define STA32X_CONFF_OCFG_MASK 0x03 #define STA32X_CONFF_OCFG_SHIFT 0 #define STA32X_CONFF_IDE 0x04 -#define STA32X_CONFF_IDE_SHIFT 3 +#define STA32X_CONFF_IDE_SHIFT 2 #define STA32X_CONFF_BCLE 0x08 #define STA32X_CONFF_ECLE 0x20 #define STA32X_CONFF_PWDN 0x40 -- cgit v1.2.3 From a43bd7e125143b875caae6d4f9938855b440faaf Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Tue, 20 Jan 2015 15:43:16 +0800 Subject: ASoC: atmel_ssc_dai: fix start event for I2S mode According to the I2S specification information as following: - WS = 0, channel 1 (left) - WS = 1, channel 2 (right) So, the start event should be TF/RF falling edge. Reported-by: Songjun Wu Signed-off-by: Bo Shen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/atmel/atmel_ssc_dai.c | 18 ++++-------------- 1 file changed, 4 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 99ff35e2a25d..e691aab60761 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -348,7 +348,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, struct atmel_pcm_dma_params *dma_params; int dir, channels, bits; u32 tfmr, rfmr, tcmr, rcmr; - int start_event; int ret; int fslen, fslen_ext; @@ -457,19 +456,10 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, * The SSC transmit clock is obtained from the BCLK signal on * on the TK line, and the SSC receive clock is * generated from the transmit clock. - * - * For single channel data, one sample is transferred - * on the falling edge of the LRC clock. - * For two channel data, one sample is - * transferred on both edges of the LRC clock. */ - start_event = ((channels == 1) - ? SSC_START_FALLING_RF - : SSC_START_EDGE_RF); - rcmr = SSC_BF(RCMR_PERIOD, 0) | SSC_BF(RCMR_STTDLY, START_DELAY) - | SSC_BF(RCMR_START, start_event) + | SSC_BF(RCMR_START, SSC_START_FALLING_RF) | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? @@ -478,14 +468,14 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | SSC_BF(RFMR_FSLEN, 0) - | SSC_BF(RFMR_DATNB, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) | SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_LOOP, 0) | SSC_BF(RFMR_DATLEN, (bits - 1)); tcmr = SSC_BF(TCMR_PERIOD, 0) | SSC_BF(TCMR_STTDLY, START_DELAY) - | SSC_BF(TCMR_START, start_event) + | SSC_BF(TCMR_START, SSC_START_FALLING_RF) | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | SSC_BF(TCMR_CKO, SSC_CKO_NONE) | SSC_BF(TCMR_CKS, ssc->clk_from_rk_pin ? @@ -495,7 +485,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(TFMR_FSDEN, 0) | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | SSC_BF(TFMR_FSLEN, 0) - | SSC_BF(TFMR_DATNB, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) | SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATDEF, 0) | SSC_BF(TFMR_DATLEN, (bits - 1)); -- cgit v1.2.3 From 8a6cf30bf93df2c0f2637156e4a5070594bddebf Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Mon, 19 Jan 2015 08:23:43 +0100 Subject: ASoC: wm8731: init mutex in i2c init path The I2C init path forgot to init the mutex, leading to an oops when controls are accessed. Signed-off-by: Manuel Lauss Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8731.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index b9211b42f6e9..b115ed815db9 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -717,6 +717,8 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, if (wm8731 == NULL) return -ENOMEM; + mutex_init(&wm8731->lock); + wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); -- cgit v1.2.3 From 09a34aa582aec12c974b08c1ffedb9bd1940565a Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Wed, 21 Jan 2015 07:20:23 +0800 Subject: ASoC: Intel: Used lock version to update shim registers We need hold lock each time updating shirm registers, otherwise, we may set unexpected values to them when they are set in different thread at different time sequence. The notification work will be scheduled in global work queue, which won't hold this sst->spinlock itself, so here we need change to use the lock version to update shim registers. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 5bf14040c24a..8156cc1accb7 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -651,11 +651,11 @@ static void hsw_notification_work(struct work_struct *work) } /* tell DSP that notification has been handled */ - sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IPCD, + sst_dsp_shim_update_bits(hsw->dsp, SST_IPCD, SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE); /* unmask busy interrupt */ - sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IMRX, SST_IMRX_BUSY, 0); + sst_dsp_shim_update_bits(hsw->dsp, SST_IMRX, SST_IMRX_BUSY, 0); } static struct ipc_message *reply_find_msg(struct sst_hsw *hsw, u32 header) -- cgit v1.2.3 From 0b65ba9981d8fe80fd099f26dd96c60e07729aeb Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 30 Jan 2015 14:42:31 +0200 Subject: ASoC: tlv320aic3x: Fix data delay configuration Fix the issue introduced by: 368494093354 ASoC: tlv320aic3x: Add TDM support The CTRLC register were not receiving the correct delay configuration, which will corrupt DSP_A audio mode. Fixes: 368494093354 (ASoC: tlv320aic3x: Add TDM support) Reported-by: Pavel Machek Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b7ebce054b4e..dd222b10ce13 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1046,7 +1046,7 @@ static int aic3x_prepare(struct snd_pcm_substream *substream, delay += aic3x->tdm_delay; /* Configure data delay */ - snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, aic3x->tdm_delay); + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay); return 0; } -- cgit v1.2.3 From 9ee802ec5b1ae0ee468b6acf1bf489347997893b Mon Sep 17 00:00:00 2001 From: Filip Brozovic Date: Fri, 30 Jan 2015 12:58:24 +0100 Subject: ASoC: sgtl5000: Use shift mask when setting codec mode Shift the I2S mode value by the necessary amount before writing the registers. This makes Right Justified and PCM mode work in addition to the default Left Justified mode. Signed-off-by: Filip Brozovic Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 29cf7ce610f4..7665016a79ce 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -483,21 +483,21 @@ static int sgtl5000_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* setting i2s data format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: - i2sctl |= SGTL5000_I2S_MODE_PCM; + i2sctl |= SGTL5000_I2S_MODE_PCM << SGTL5000_I2S_MODE_SHIFT; break; case SND_SOC_DAIFMT_DSP_B: - i2sctl |= SGTL5000_I2S_MODE_PCM; + i2sctl |= SGTL5000_I2S_MODE_PCM << SGTL5000_I2S_MODE_SHIFT; i2sctl |= SGTL5000_I2S_LRALIGN; break; case SND_SOC_DAIFMT_I2S: - i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ << SGTL5000_I2S_MODE_SHIFT; break; case SND_SOC_DAIFMT_RIGHT_J: - i2sctl |= SGTL5000_I2S_MODE_RJ; + i2sctl |= SGTL5000_I2S_MODE_RJ << SGTL5000_I2S_MODE_SHIFT; i2sctl |= SGTL5000_I2S_LRPOL; break; case SND_SOC_DAIFMT_LEFT_J: - i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ << SGTL5000_I2S_MODE_SHIFT; i2sctl |= SGTL5000_I2S_LRALIGN; break; default: -- cgit v1.2.3 From 20cf2603b122bf71fb54def1de6a2ad73d5ddb0b Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Fri, 30 Jan 2015 17:38:42 +0800 Subject: ASoC: atmel_ssc_dai: fix the setting for DSP mode When SCC work in DSP A mode, the data outputs/inputs are shift out on falling edge, the frame sync are sample on the rising edge. Reported-by: Songjun Wu Signed-off-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_ssc_dai.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index e691aab60761..35e44e463cfe 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -502,7 +502,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | SSC_BF(RCMR_STTDLY, 1) | SSC_BF(RCMR_START, SSC_START_RISING_RF) - | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, SSC_CKS_DIV); @@ -517,7 +517,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | SSC_BF(TCMR_STTDLY, 1) | SSC_BF(TCMR_START, SSC_START_RISING_RF) - | SSC_BF(TCMR_CKI, SSC_CKI_RISING) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | SSC_BF(TCMR_CKS, SSC_CKS_DIV); @@ -546,7 +546,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rcmr = SSC_BF(RCMR_PERIOD, 0) | SSC_BF(RCMR_STTDLY, START_DELAY) | SSC_BF(RCMR_START, SSC_START_RISING_RF) - | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? SSC_CKS_PIN : SSC_CKS_CLOCK); -- cgit v1.2.3 From 58cc9c9a175885bbf6bae3acf18233d0a8229a84 Mon Sep 17 00:00:00 2001 From: Eric Nelson Date: Fri, 30 Jan 2015 14:07:55 -0700 Subject: ASoC: sgtl5000: add delay before first I2C access To quote from section 1.3.1 of the data sheet: The SGTL5000 has an internal reset that is deasserted 8 SYS_MCLK cycles after all power rails have been brought up. After this time, communication can start ... 1.0us represents 8 SYS_MCLK cycles at the minimum 8.0 MHz SYS_MCLK. Signed-off-by: Eric Nelson Reviewed-by: Fabio Estevam Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/sgtl5000.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 7665016a79ce..aa98be32bb60 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1462,6 +1462,9 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, if (ret) return ret; + /* Need 8 clocks before I2C accesses */ + udelay(1); + /* read chip information */ ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, ®); if (ret) -- cgit v1.2.3 From 28d1ad09c50c758d3e295fa7ff90a4712e1254ea Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 5 Feb 2015 16:40:33 +0800 Subject: ASoC: rt286: Fix potencial crash in jd function We assign rt286->codec in rt286_probe. If rt286_jack_detect is invoked before rt286_probe, rt286->codec will be NULL and cause a kernel panic. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 847cc4b9bee5..f14d335b07b1 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -305,6 +305,8 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) *hp = false; *mic = false; + if (!rt286->codec) + return -EINVAL; if (rt286->pdata.cbj_en) { regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf); *hp = buf & 0x80000000; -- cgit v1.2.3 From 5c2b06369dafd796ebb4f17dab543d3da500245e Mon Sep 17 00:00:00 2001 From: Kevin Strasser Date: Thu, 5 Feb 2015 12:12:07 -0800 Subject: ASoC: Intel: fix sst firmware path for cht-bsw-rt5672 All sst firmware is provided under the intel directory of the linux-firmware tree. By default this directory structure is kept when installing on a target system. Change the path to expect a default linux-firmware installation. Signed-off-by: Kevin Strasser Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_acpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 2ac72eb5e75d..b3360139c41a 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -350,7 +350,7 @@ static struct sst_machines sst_acpi_bytcr[] = { /* Cherryview-based platforms: CherryTrail and Braswell */ static struct sst_machines sst_acpi_chv[] = { - {"10EC5670", "cht-bsw", "cht-bsw-rt5672", NULL, "fw_sst_22a8.bin", + {"10EC5670", "cht-bsw", "cht-bsw-rt5672", NULL, "intel/fw_sst_22a8.bin", &chv_platform_data }, {}, }; -- cgit v1.2.3 From 279e17ae81c17b40ae7a6c9e10f386a7aac7aa55 Mon Sep 17 00:00:00 2001 From: Christian Engelmayer Date: Sat, 7 Feb 2015 23:40:52 +0100 Subject: ASoC: Intel: sst: Fix firmware name size handling Function sst_acpi_probe() uses plain strcpy for setting member firmware_name of a struct intel_sst_drv from member firmware of a struct sst_machines. Thereby the destination array has got a length of 20 byte while the source may hold 32 byte. Since eg. commit 64b9c90b8600 ("ASoC: Intel: Fix BYTCR firmware name") increased strings from "fw_sst_0f28.bin" to "intel/fw_sst_0f28.bin" there is an actual possibility that the 20 byte array at the end of struct intel_sst_drv is overrun. Thus increase the size of the destination and use the same define for both structs. Detected by Coverity CID 1260087. Signed-off-by: Christian Engelmayer Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/intel/sst/sst.h | 3 ++- sound/soc/intel/sst/sst_acpi.c | 2 +- 2 files changed, 3 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index 7f4bbfcbc6f5..562bc483d6b7 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -58,6 +58,7 @@ enum sst_algo_ops { #define SST_BLOCK_TIMEOUT 1000 #define FW_SIGNATURE_SIZE 4 +#define FW_NAME_SIZE 32 /* stream states */ enum sst_stream_states { @@ -426,7 +427,7 @@ struct intel_sst_drv { * Holder for firmware name. Due to async call it needs to be * persistent till worker thread gets called */ - char firmware_name[20]; + char firmware_name[FW_NAME_SIZE]; }; /* misc definitions */ diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index b3360139c41a..51f83bad5319 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -47,7 +47,7 @@ struct sst_machines { char board[32]; char machine[32]; void (*machine_quirk)(void); - char firmware[32]; + char firmware[FW_NAME_SIZE]; struct sst_platform_info *pdata; }; -- cgit v1.2.3