From a61588553ba32a306ef187e929eeb29324ad7cac Mon Sep 17 00:00:00 2001 From: Russell King Date: Thu, 28 Feb 2019 15:30:34 +0000 Subject: ASoC: hdmi-codec: fix S/PDIF DAI [ Upstream commit 2e95f984aae4cf0608d0ba2189c756f2bd50b44a ] When using the S/PDIF DAI, there is no requirement to call snd_soc_dai_set_fmt() as there is no DAI format definition that defines S/PDIF. In any case, S/PDIF does not have separate clocks, this is embedded into the data stream. Consequently, when attempting to use TDA998x in S/PDIF mode, the attempt to configure TDA998x via the hw_params callback fails as the hdmi_codec_daifmt is left initialised to zero. Since the S/PDIF DAI will only be used by S/PDIF, prepare the hdmi_codec_daifmt structure for this format. Signed-off-by: Russell King Reviewed-by: Jyri Sarha Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/codecs/hdmi-codec.c | 118 +++++++++++++++++++++--------------------- 1 file changed, 59 insertions(+), 59 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index cf3b905b4ead..7406695ee5dc 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -536,73 +536,71 @@ static int hdmi_codec_set_fmt(struct snd_soc_dai *dai, { struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); struct hdmi_codec_daifmt cf = { 0 }; - int ret = 0; dev_dbg(dai->dev, "%s()\n", __func__); - if (dai->id == DAI_ID_SPDIF) { - cf.fmt = HDMI_SPDIF; - } else { - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - cf.bit_clk_master = 1; - cf.frame_clk_master = 1; - break; - case SND_SOC_DAIFMT_CBS_CFM: - cf.frame_clk_master = 1; - break; - case SND_SOC_DAIFMT_CBM_CFS: - cf.bit_clk_master = 1; - break; - case SND_SOC_DAIFMT_CBS_CFS: - break; - default: - return -EINVAL; - } + if (dai->id == DAI_ID_SPDIF) + return 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + cf.bit_clk_master = 1; + cf.frame_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFM: + cf.frame_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBM_CFS: + cf.bit_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - break; - case SND_SOC_DAIFMT_NB_IF: - cf.frame_clk_inv = 1; - break; - case SND_SOC_DAIFMT_IB_NF: - cf.bit_clk_inv = 1; - break; - case SND_SOC_DAIFMT_IB_IF: - cf.frame_clk_inv = 1; - cf.bit_clk_inv = 1; - break; - } + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + cf.frame_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_NF: + cf.bit_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + cf.frame_clk_inv = 1; + cf.bit_clk_inv = 1; + break; + } - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - cf.fmt = HDMI_I2S; - break; - case SND_SOC_DAIFMT_DSP_A: - cf.fmt = HDMI_DSP_A; - break; - case SND_SOC_DAIFMT_DSP_B: - cf.fmt = HDMI_DSP_B; - break; - case SND_SOC_DAIFMT_RIGHT_J: - cf.fmt = HDMI_RIGHT_J; - break; - case SND_SOC_DAIFMT_LEFT_J: - cf.fmt = HDMI_LEFT_J; - break; - case SND_SOC_DAIFMT_AC97: - cf.fmt = HDMI_AC97; - break; - default: - dev_err(dai->dev, "Invalid DAI interface format\n"); - return -EINVAL; - } + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cf.fmt = HDMI_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + cf.fmt = HDMI_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + cf.fmt = HDMI_DSP_B; + break; + case SND_SOC_DAIFMT_RIGHT_J: + cf.fmt = HDMI_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + cf.fmt = HDMI_LEFT_J; + break; + case SND_SOC_DAIFMT_AC97: + cf.fmt = HDMI_AC97; + break; + default: + dev_err(dai->dev, "Invalid DAI interface format\n"); + return -EINVAL; } hcp->daifmt[dai->id] = cf; - return ret; + return 0; } static int hdmi_codec_digital_mute(struct snd_soc_dai *dai, int mute) @@ -784,8 +782,10 @@ static int hdmi_codec_probe(struct platform_device *pdev) i++; } - if (hcd->spdif) + if (hcd->spdif) { hcp->daidrv[i] = hdmi_spdif_dai; + hcp->daifmt[DAI_ID_SPDIF].fmt = HDMI_SPDIF; + } ret = snd_soc_register_codec(dev, &hdmi_codec, hcp->daidrv, dai_count); -- cgit v1.2.3 From 82af2ff96f3a691812f8e8374d9df3130109c2c7 Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Fri, 8 Mar 2019 16:38:57 +0800 Subject: ASoC:soc-pcm:fix a codec fixup issue in TDM case [ Upstream commit 570f18b6a8d1f0e60e8caf30e66161b6438dcc91 ] On HDaudio platforms, if playback is started when capture is working, there is no audible output. This can be root-caused to the use of the rx|tx_mask to store an HDaudio stream tag. If capture is stared before playback, rx_mask would be non-zero on HDaudio platform, then the channel number of playback, which is in the same codec dai with the capture, would be changed by soc_pcm_codec_params_fixup based on the tx_mask at first, then overwritten by this function based on rx_mask at last. According to the author of tx|rx_mask, tx_mask is for playback and rx_mask is for capture. And stream direction is checked at all other references of tx|rx_mask in ASoC, so here should be an error. This patch checks stream direction for tx|rx_mask for fixup function. This issue would affect not only HDaudio+ASoC, but also I2S codecs if the channel number based on rx_mask is not equal to the one for tx_mask. It could be rarely reproduecd because most drivers in kernel set the same channel number to tx|rx_mask or rx_mask is zero. Tested on all platforms using stream_tag & HDaudio and intel I2S platforms. Signed-off-by: Rander Wang Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/soc-pcm.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6fc85199ac73..584b7ffe78f5 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -894,10 +894,13 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, codec_params = *params; /* fixup params based on TDM slot masks */ - if (codec_dai->tx_mask) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + codec_dai->tx_mask) soc_pcm_codec_params_fixup(&codec_params, codec_dai->tx_mask); - if (codec_dai->rx_mask) + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && + codec_dai->rx_mask) soc_pcm_codec_params_fixup(&codec_params, codec_dai->rx_mask); -- cgit v1.2.3 From dc465d317403505ee72301e59e00a3252e6a43c5 Mon Sep 17 00:00:00 2001 From: John Hsu Date: Mon, 11 Mar 2019 09:36:45 +0800 Subject: ASoC: nau8824: fix the issue of the widget with prefix name [ Upstream commit 844a4a362dbec166b44d6b9b3dd45b08cb273703 ] The driver has two issues when machine add prefix name for codec. (1)The stream name of DAI can't find the AIF widgets. (2)The drivr can enable/disalbe the MICBIAS and SAR widgets. The patch will fix these issues caused by prefixed name added. Signed-off-by: John Hsu Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/codecs/nau8824.c | 46 ++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 38 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 0240759f951c..e8ea51247b17 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -634,8 +634,8 @@ static const struct snd_soc_dapm_widget nau8824_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADCR", NULL, NAU8824_REG_ANALOG_ADC_2, NAU8824_ADCR_EN_SFT, 0), - SND_SOC_DAPM_AIF_OUT("AIFTX", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_AIF_IN("AIFRX", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFRX", "Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DACL", NULL, NAU8824_REG_RDAC, NAU8824_DACL_EN_SFT, 0), @@ -784,6 +784,36 @@ static void nau8824_int_status_clear_all(struct regmap *regmap) } } +static void nau8824_dapm_disable_pin(struct nau8824 *nau8824, const char *pin) +{ + struct snd_soc_dapm_context *dapm = nau8824->dapm; + const char *prefix = dapm->component->name_prefix; + char prefixed_pin[80]; + + if (prefix) { + snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s", + prefix, pin); + snd_soc_dapm_disable_pin(dapm, prefixed_pin); + } else { + snd_soc_dapm_disable_pin(dapm, pin); + } +} + +static void nau8824_dapm_enable_pin(struct nau8824 *nau8824, const char *pin) +{ + struct snd_soc_dapm_context *dapm = nau8824->dapm; + const char *prefix = dapm->component->name_prefix; + char prefixed_pin[80]; + + if (prefix) { + snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s", + prefix, pin); + snd_soc_dapm_force_enable_pin(dapm, prefixed_pin); + } else { + snd_soc_dapm_force_enable_pin(dapm, pin); + } +} + static void nau8824_eject_jack(struct nau8824 *nau8824) { struct snd_soc_dapm_context *dapm = nau8824->dapm; @@ -792,8 +822,8 @@ static void nau8824_eject_jack(struct nau8824 *nau8824) /* Clear all interruption status */ nau8824_int_status_clear_all(regmap); - snd_soc_dapm_disable_pin(dapm, "SAR"); - snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + nau8824_dapm_disable_pin(nau8824, "SAR"); + nau8824_dapm_disable_pin(nau8824, "MICBIAS"); snd_soc_dapm_sync(dapm); /* Enable the insertion interruption, disable the ejection @@ -822,8 +852,8 @@ static void nau8824_jdet_work(struct work_struct *work) struct regmap *regmap = nau8824->regmap; int adc_value, event = 0, event_mask = 0; - snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); - snd_soc_dapm_force_enable_pin(dapm, "SAR"); + nau8824_dapm_enable_pin(nau8824, "MICBIAS"); + nau8824_dapm_enable_pin(nau8824, "SAR"); snd_soc_dapm_sync(dapm); msleep(100); @@ -834,8 +864,8 @@ static void nau8824_jdet_work(struct work_struct *work) if (adc_value < HEADSET_SARADC_THD) { event |= SND_JACK_HEADPHONE; - snd_soc_dapm_disable_pin(dapm, "SAR"); - snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + nau8824_dapm_disable_pin(nau8824, "SAR"); + nau8824_dapm_disable_pin(nau8824, "MICBIAS"); snd_soc_dapm_sync(dapm); } else { event |= SND_JACK_HEADSET; -- cgit v1.2.3 From 4e244b64fbdf4d7200f5dfcf31f44e0e0f971bf2 Mon Sep 17 00:00:00 2001 From: John Hsu Date: Wed, 13 Mar 2019 16:23:44 +0800 Subject: ASoC: nau8810: fix the issue of widget with prefixed name [ Upstream commit 54d1cf78b0f4ba348a7c7fb8b7d0708d71b6cc8a ] The driver changes the stream name of DAC and ADC to avoid the issue of widget with prefixed name. When the machine adds prefixed name for codec, the stream name of DAI may not find the widgets. Signed-off-by: John Hsu Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/codecs/nau8810.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index c8e2451ae0a3..193588eb9835 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -414,9 +414,9 @@ static const struct snd_soc_dapm_widget nau8810_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Mono Mixer", NAU8810_REG_POWER3, NAU8810_MOUTMX_EN_SFT, 0, &nau8810_mono_mixer_controls[0], ARRAY_SIZE(nau8810_mono_mixer_controls)), - SND_SOC_DAPM_DAC("DAC", "HiFi Playback", NAU8810_REG_POWER3, + SND_SOC_DAPM_DAC("DAC", "Playback", NAU8810_REG_POWER3, NAU8810_DAC_EN_SFT, 0), - SND_SOC_DAPM_ADC("ADC", "HiFi Capture", NAU8810_REG_POWER2, + SND_SOC_DAPM_ADC("ADC", "Capture", NAU8810_REG_POWER2, NAU8810_ADC_EN_SFT, 0), SND_SOC_DAPM_PGA("SpkN Out", NAU8810_REG_POWER3, NAU8810_NSPK_EN_SFT, 0, NULL, 0), -- cgit v1.2.3 From 97916fe0b8fa34954c7a5fa5b515e458d2b48043 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Tue, 12 Mar 2019 18:40:06 +0100 Subject: ASoC: samsung: odroid: Fix clock configuration for 44100 sample rate [ Upstream commit 2b13bee3884926cba22061efa75bd315e871de24 ] After commit fbeec965b8d1c ("ASoC: samsung: odroid: Fix 32000 sample rate handling") the audio root clock frequency is configured improperly for 44100 sample rate. Due to clock rate rounding it's 20070401 Hz instead of 22579000 Hz. This results in a too low value of the PSR clock divider in the CPU DAI driver and too fast actual sample rate for fs=44100. E.g. 1 kHz tone has actual 1780 Hz frequency (1 kHz * 20070401/22579000 * 2). Fix this by increasing the correction passed to clk_set_rate() to take into account inaccuracy of the EPLL frequency properly. Fixes: fbeec965b8d1c ("ASoC: samsung: odroid: Fix 32000 sample rate handling") Reported-by: JaeChul Lee Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/samsung/odroid.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 06a31a9585a0..32c9e197ca95 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -66,11 +66,11 @@ static int odroid_card_hw_params(struct snd_pcm_substream *substream, return ret; /* - * We add 1 to the rclk_freq value in order to avoid too low clock + * We add 2 to the rclk_freq value in order to avoid too low clock * frequency values due to the EPLL output frequency not being exact * multiple of the audio sampling rate. */ - rclk_freq = params_rate(params) * rfs + 1; + rclk_freq = params_rate(params) * rfs + 2; ret = clk_set_rate(priv->sclk_i2s, rclk_freq); if (ret < 0) -- cgit v1.2.3 From 97dac24e68ed587c44f23cbf99d76359ea6950f1 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 19 Mar 2019 11:52:06 +0000 Subject: ASoC: wm_adsp: Add locking to wm_adsp2_bus_error [ Upstream commit a2225a6d155fcb247fe4c6d87f7c91807462966d ] Best to lock across handling the bus error to ensure the DSP doesn't change power state as we are reading the status registers. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/codecs/wm_adsp.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 67330b6ab204..d632a0511d62 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3711,11 +3711,13 @@ irqreturn_t wm_adsp2_bus_error(struct wm_adsp *dsp) struct regmap *regmap = dsp->regmap; int ret = 0; + mutex_lock(&dsp->pwr_lock); + ret = regmap_read(regmap, dsp->base + ADSP2_LOCK_REGION_CTRL, &val); if (ret) { adsp_err(dsp, "Failed to read Region Lock Ctrl register: %d\n", ret); - return IRQ_HANDLED; + goto error; } if (val & ADSP2_WDT_TIMEOUT_STS_MASK) { @@ -3734,7 +3736,7 @@ irqreturn_t wm_adsp2_bus_error(struct wm_adsp *dsp) adsp_err(dsp, "Failed to read Bus Err Addr register: %d\n", ret); - return IRQ_HANDLED; + goto error; } adsp_err(dsp, "bus error address = 0x%x\n", @@ -3747,7 +3749,7 @@ irqreturn_t wm_adsp2_bus_error(struct wm_adsp *dsp) adsp_err(dsp, "Failed to read Pmem Xmem Err Addr register: %d\n", ret); - return IRQ_HANDLED; + goto error; } adsp_err(dsp, "xmem error address = 0x%x\n", @@ -3760,6 +3762,9 @@ irqreturn_t wm_adsp2_bus_error(struct wm_adsp *dsp) regmap_update_bits(regmap, dsp->base + ADSP2_LOCK_REGION_CTRL, ADSP2_CTRL_ERR_EINT, ADSP2_CTRL_ERR_EINT); +error: + mutex_unlock(&dsp->pwr_lock); + return IRQ_HANDLED; } EXPORT_SYMBOL_GPL(wm_adsp2_bus_error); -- cgit v1.2.3 From 10cc9e79fb097d31ac73165e1e91a48d9c949fda Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 20 Mar 2019 22:41:56 +0100 Subject: ASoC: cs4270: Set auto-increment bit for register writes [ Upstream commit f0f2338a9cfaf71db895fa989ea7234e8a9b471d ] The CS4270 does not by default increment the register address on consecutive writes. During normal operation it doesn't matter as all register accesses are done individually. At resume time after suspend, however, the regcache code gathers the biggest possible block of registers to sync and sends them one on one go. To fix this, set the INCR bit in all cases. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/codecs/cs4270.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 84f86745c30e..828bc615a190 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -643,6 +643,7 @@ static const struct regmap_config cs4270_regmap = { .reg_defaults = cs4270_reg_defaults, .num_reg_defaults = ARRAY_SIZE(cs4270_reg_defaults), .cache_type = REGCACHE_RBTREE, + .write_flag_mask = CS4270_I2C_INCR, .readable_reg = cs4270_reg_is_readable, .volatile_reg = cs4270_reg_is_volatile, -- cgit v1.2.3 From 906d79143057cbd78f35db2e1175417c6a2d8869 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Sat, 30 Mar 2019 09:02:02 -0700 Subject: ASoC: tlv320aic32x4: Fix Common Pins [ Upstream commit c63adb28f6d913310430f14c69f0a2ea55eed0cc ] The common pins were mistakenly not added to the DAPM graph. Adding these pins will allow valid graphs to be created. Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/codecs/tlv320aic32x4.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index e694f5f04eb9..628621fc3386 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -462,6 +462,8 @@ static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN2_R"), SND_SOC_DAPM_INPUT("IN3_L"), SND_SOC_DAPM_INPUT("IN3_R"), + SND_SOC_DAPM_INPUT("CM_L"), + SND_SOC_DAPM_INPUT("CM_R"), }; static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { -- cgit v1.2.3 From a8f82720673d41b85145f232c954fe3a365f72b5 Mon Sep 17 00:00:00 2001 From: Sugar Zhang Date: Wed, 3 Apr 2019 21:40:45 +0800 Subject: ASoC: rockchip: pdm: fix regmap_ops hang issue [ Upstream commit c85064435fe7a216ec0f0238ef2b8f7cd850a450 ] This is because set_fmt ops maybe called when PD is off, and in such case, regmap_ops will lead system hang. enale PD before doing regmap_ops. Signed-off-by: Sugar Zhang Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/rockchip/rockchip_pdm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 400e29edb1c9..8a2e3bbce3a1 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -208,7 +208,9 @@ static int rockchip_pdm_set_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } + pm_runtime_get_sync(cpu_dai->dev); regmap_update_bits(pdm->regmap, PDM_CLK_CTRL, mask, val); + pm_runtime_put(cpu_dai->dev); return 0; } -- cgit v1.2.3 From f81642e882b7f63bb8130e1c8fb9a13a523df039 Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Mon, 8 Apr 2019 17:08:58 +0800 Subject: ASoC: Intel: kbl: fix wrong number of channels [ Upstream commit d6ba3f815bc5f3c4249d15c8bc5fbb012651b4a4 ] Fix wrong setting on number of channels. The context wants to set constraint to 2 channels instead of 4. Signed-off-by: Tzung-Bi Shih Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 69ab55956492..41cb1fefbd42 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -405,7 +405,7 @@ static const struct snd_pcm_hw_constraint_list constraints_dmic_channels = { }; static const unsigned int dmic_2ch[] = { - 4, + 2, }; static const struct snd_pcm_hw_constraint_list constraints_dmic_2ch = { -- cgit v1.2.3 From 34f9130a457c0a4f9ec1b30a1311a3a02c6581f2 Mon Sep 17 00:00:00 2001 From: Ross Zwisler Date: Mon, 29 Apr 2019 12:25:17 -0600 Subject: ASoC: Intel: avoid Oops if DMA setup fails commit 0efa3334d65b7f421ba12382dfa58f6ff5bf83c4 upstream. Currently in sst_dsp_new() if we get an error return from sst_dma_new() we just print an error message and then still complete the function successfully. This means that we are trying to run without sst->dma properly set up, which will result in NULL pointer dereference when sst->dma is later used. This was happening for me in sst_dsp_dma_get_channel(): struct sst_dma *dma = dsp->dma; ... dma->ch = dma_request_channel(mask, dma_chan_filter, dsp); This resulted in: BUG: unable to handle kernel NULL pointer dereference at 0000000000000018 IP: sst_dsp_dma_get_channel+0x4f/0x125 [snd_soc_sst_firmware] Fix this by adding proper error handling for the case where we fail to set up DMA. This change only affects Haswell and Broadwell systems. Baytrail systems explicilty opt-out of DMA via sst->pdata->resindex_dma_base being set to -1. Signed-off-by: Ross Zwisler Cc: stable@vger.kernel.org Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown Signed-off-by: Greg Kroah-Hartman --- sound/soc/intel/common/sst-firmware.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 79a9fdf94d38..582b30a5118d 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -1252,11 +1252,15 @@ struct sst_dsp *sst_dsp_new(struct device *dev, goto irq_err; err = sst_dma_new(sst); - if (err) - dev_warn(dev, "sst_dma_new failed %d\n", err); + if (err) { + dev_err(dev, "sst_dma_new failed %d\n", err); + goto dma_err; + } return sst; +dma_err: + free_irq(sst->irq, sst); irq_err: if (sst->ops->free) sst->ops->free(sst); -- cgit v1.2.3 From b93d5632c8ba66cd7c0a163fa0b6149a95871505 Mon Sep 17 00:00:00 2001 From: Jon Hunter Date: Wed, 1 May 2019 15:29:38 +0100 Subject: ASoC: max98090: Fix restore of DAPM Muxes commit ecb2795c08bc825ebd604997e5be440b060c5b18 upstream. The max98090 driver defines 3 DAPM muxes; one for the right line output (LINMOD Mux), one for the left headphone mixer source (MIXHPLSEL Mux) and one for the right headphone mixer source (MIXHPRSEL Mux). The same bit is used for the mux as well as the DAPM enable, and although the mux can be correctly configured, after playback has completed, the mux will be reset during the disable phase. This is preventing the state of these muxes from being saved and restored correctly on system reboot. Fix this by marking these muxes as SND_SOC_NOPM. Note this has been verified this on the Tegra124 Nyan Big which features the MAX98090 codec. Signed-off-by: Jon Hunter Signed-off-by: Mark Brown Cc: stable@vger.kernel.org Signed-off-by: Greg Kroah-Hartman --- sound/soc/codecs/max98090.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 13bcfb1ef9b4..cc66ea5cc776 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1209,14 +1209,14 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { &max98090_right_rcv_mixer_controls[0], ARRAY_SIZE(max98090_right_rcv_mixer_controls)), - SND_SOC_DAPM_MUX("LINMOD Mux", M98090_REG_LOUTR_MIXER, - M98090_LINMOD_SHIFT, 0, &max98090_linmod_mux), + SND_SOC_DAPM_MUX("LINMOD Mux", SND_SOC_NOPM, 0, 0, + &max98090_linmod_mux), - SND_SOC_DAPM_MUX("MIXHPLSEL Mux", M98090_REG_HP_CONTROL, - M98090_MIXHPLSEL_SHIFT, 0, &max98090_mixhplsel_mux), + SND_SOC_DAPM_MUX("MIXHPLSEL Mux", SND_SOC_NOPM, 0, 0, + &max98090_mixhplsel_mux), - SND_SOC_DAPM_MUX("MIXHPRSEL Mux", M98090_REG_HP_CONTROL, - M98090_MIXHPRSEL_SHIFT, 0, &max98090_mixhprsel_mux), + SND_SOC_DAPM_MUX("MIXHPRSEL Mux", SND_SOC_NOPM, 0, 0, + &max98090_mixhprsel_mux), SND_SOC_DAPM_PGA("HP Left Out", M98090_REG_OUTPUT_ENABLE, M98090_HPLEN_SHIFT, 0, NULL, 0), -- cgit v1.2.3 From e047bab50092704805ca0d14fa85ab65ff4b56c0 Mon Sep 17 00:00:00 2001 From: Curtis Malainey Date: Fri, 3 May 2019 12:32:14 -0700 Subject: ASoC: RT5677-SPI: Disable 16Bit SPI Transfers commit a46eb523220e242affb9a6bc9bb8efc05f4f7459 upstream. The current algorithm allows 3 types of transfers, 16bit, 32bit and burst. According to Realtek, 16bit transfers have a special restriction in that it is restricted to the memory region of 0x18020000 ~ 0x18021000. This region is the memory location of the I2C registers. The current algorithm does not uphold this restriction and therefore fails to complete writes. Since this has been broken for some time it likely no one is using it. Better to simply disable the 16 bit writes. This will allow users to properly load firmware over SPI without data corruption. Signed-off-by: Curtis Malainey Reviewed-by: Ben Zhang Signed-off-by: Mark Brown Cc: stable@vger.kernel.org Signed-off-by: Greg Kroah-Hartman --- sound/soc/codecs/rt5677-spi.c | 35 ++++++++++++++++------------------- 1 file changed, 16 insertions(+), 19 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index bd51f3655ee3..06abcd017650 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -58,13 +58,15 @@ static DEFINE_MUTEX(spi_mutex); * RT5677_SPI_READ/WRITE_32: Transfer 4 bytes * RT5677_SPI_READ/WRITE_BURST: Transfer any multiples of 8 bytes * - * For example, reading 260 bytes at 0x60030002 uses the following commands: - * 0x60030002 RT5677_SPI_READ_16 2 bytes + * Note: + * 16 Bit writes and reads are restricted to the address range + * 0x18020000 ~ 0x18021000 + * + * For example, reading 256 bytes at 0x60030004 uses the following commands: * 0x60030004 RT5677_SPI_READ_32 4 bytes * 0x60030008 RT5677_SPI_READ_BURST 240 bytes * 0x600300F8 RT5677_SPI_READ_BURST 8 bytes * 0x60030100 RT5677_SPI_READ_32 4 bytes - * 0x60030104 RT5677_SPI_READ_16 2 bytes * * Input: * @read: true for read commands; false for write commands @@ -79,15 +81,13 @@ static u8 rt5677_spi_select_cmd(bool read, u32 align, u32 remain, u32 *len) { u8 cmd; - if (align == 2 || align == 6 || remain == 2) { - cmd = RT5677_SPI_READ_16; - *len = 2; - } else if (align == 4 || remain <= 6) { + if (align == 4 || remain <= 4) { cmd = RT5677_SPI_READ_32; *len = 4; } else { cmd = RT5677_SPI_READ_BURST; - *len = min_t(u32, remain & ~7, RT5677_SPI_BURST_LEN); + *len = (((remain - 1) >> 3) + 1) << 3; + *len = min_t(u32, *len, RT5677_SPI_BURST_LEN); } return read ? cmd : cmd + 1; } @@ -108,7 +108,7 @@ static void rt5677_spi_reverse(u8 *dst, u32 dstlen, const u8 *src, u32 srclen) } } -/* Read DSP address space using SPI. addr and len have to be 2-byte aligned. */ +/* Read DSP address space using SPI. addr and len have to be 4-byte aligned. */ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) { u32 offset; @@ -124,7 +124,7 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) if (!g_spi) return -ENODEV; - if ((addr & 1) || (len & 1)) { + if ((addr & 3) || (len & 3)) { dev_err(&g_spi->dev, "Bad read align 0x%x(%zu)\n", addr, len); return -EACCES; } @@ -159,13 +159,13 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) } EXPORT_SYMBOL_GPL(rt5677_spi_read); -/* Write DSP address space using SPI. addr has to be 2-byte aligned. - * If len is not 2-byte aligned, an extra byte of zero is written at the end +/* Write DSP address space using SPI. addr has to be 4-byte aligned. + * If len is not 4-byte aligned, then extra zeros are written at the end * as padding. */ int rt5677_spi_write(u32 addr, const void *txbuf, size_t len) { - u32 offset, len_with_pad = len; + u32 offset; int status = 0; struct spi_transfer t; struct spi_message m; @@ -178,22 +178,19 @@ int rt5677_spi_write(u32 addr, const void *txbuf, size_t len) if (!g_spi) return -ENODEV; - if (addr & 1) { + if (addr & 3) { dev_err(&g_spi->dev, "Bad write align 0x%x(%zu)\n", addr, len); return -EACCES; } - if (len & 1) - len_with_pad = len + 1; - memset(&t, 0, sizeof(t)); t.tx_buf = buf; t.speed_hz = RT5677_SPI_FREQ; spi_message_init_with_transfers(&m, &t, 1); - for (offset = 0; offset < len_with_pad;) { + for (offset = 0; offset < len;) { spi_cmd = rt5677_spi_select_cmd(false, (addr + offset) & 7, - len_with_pad - offset, &t.len); + len - offset, &t.len); /* Construct SPI message header */ buf[0] = spi_cmd; -- cgit v1.2.3 From 6d611134006ff5419b680d303b9fa05fa5138be5 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Mon, 29 Apr 2019 15:29:39 +0200 Subject: ASoC: hdmi-codec: unlock the device on startup errors [ Upstream commit 30180e8436046344b12813dc954b2e01dfdcd22d ] If the hdmi codec startup fails, it should clear the current_substream pointer to free the device. This is properly done for the audio_startup() callback but for snd_pcm_hw_constraint_eld(). Make sure the pointer cleared if an error is reported. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/codecs/hdmi-codec.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 7406695ee5dc..e00f5f49f21d 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -446,8 +446,12 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, if (!ret) { ret = snd_pcm_hw_constraint_eld(substream->runtime, hcp->eld); - if (ret) + if (ret) { + mutex_lock(&hcp->current_stream_lock); + hcp->current_stream = NULL; + mutex_unlock(&hcp->current_stream_lock); return ret; + } } /* Select chmap supported */ hdmi_codec_eld_chmap(hcp); -- cgit v1.2.3 From 48f4d9d550e9f7bcec838b72123b886fa006ec3c Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Tue, 16 Apr 2019 15:12:23 +0200 Subject: ASoC: imx: fix fiq dependencies [ Upstream commit ea751227c813ab833609afecfeedaf0aa26f327e ] During randconfig builds, I occasionally run into an invalid configuration of the freescale FIQ sound support: WARNING: unmet direct dependencies detected for SND_SOC_IMX_PCM_FIQ Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_IMX_SOC [=m] Selected by [y]: - SND_SOC_FSL_SPDIF [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_IMX_SOC [=m]!=n && (MXC_TZIC [=n] || MXC_AVIC [=y]) sound/soc/fsl/imx-ssi.o: In function `imx_ssi_remove': imx-ssi.c:(.text+0x28): undefined reference to `imx_pcm_fiq_exit' sound/soc/fsl/imx-ssi.o: In function `imx_ssi_probe': imx-ssi.c:(.text+0xa64): undefined reference to `imx_pcm_fiq_init' The Kconfig warning is a result of the symbol being defined inside of the "if SND_IMX_SOC" block, and is otherwise harmless. The link error is more tricky and happens with SND_SOC_IMX_SSI=y, which may or may not imply FIQ support. However, if SND_SOC_FSL_SSI is set to =m at the same time, that selects SND_SOC_IMX_PCM_FIQ as a loadable module dependency, which then causes a link failure from imx-ssi. The solution here is to make SND_SOC_IMX_PCM_FIQ built-in whenever one of its potential users is built-in. Fixes: ff40260f79dc ("ASoC: fsl: refine DMA/FIQ dependencies") Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/fsl/Kconfig | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 4087deeda7cf..2523b0065990 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -173,16 +173,17 @@ config SND_MPC52xx_SOC_EFIKA endif # SND_POWERPC_SOC +config SND_SOC_IMX_PCM_FIQ + tristate + default y if SND_SOC_IMX_SSI=y && (SND_SOC_FSL_SSI=m || SND_SOC_FSL_SPDIF=m) && (MXC_TZIC || MXC_AVIC) + select FIQ + if SND_IMX_SOC config SND_SOC_IMX_SSI tristate select SND_SOC_FSL_UTILS -config SND_SOC_IMX_PCM_FIQ - tristate - select FIQ - comment "SoC Audio support for Freescale i.MX boards:" config SND_MXC_SOC_WM1133_EV1 -- cgit v1.2.3 From 0d7325abe29ba50dc9822b1bc6bc9e47586504d6 Mon Sep 17 00:00:00 2001 From: Daniel Baluta Date: Sun, 21 Apr 2019 19:39:08 +0000 Subject: ASoC: fsl_sai: Update is_slave_mode with correct value [ Upstream commit ddb351145a967ee791a0fb0156852ec2fcb746ba ] is_slave_mode defaults to false because sai structure that contains it is kzalloc'ed. Anyhow, if we decide to set the following configuration SAI slave -> SAI master, is_slave_mode will remain set on true although SAI being master it should be set to false. Fix this by updating is_slave_mode for each call of fsl_sai_set_dai_fmt. Signed-off-by: Daniel Baluta Acked-by: Nicolin Chen Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/fsl/fsl_sai.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 18e5ce81527d..c1c733b573a7 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -274,12 +274,14 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_CBS_CFS: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; val_cr4 |= FSL_SAI_CR4_FSD_MSTR; + sai->is_slave_mode = false; break; case SND_SOC_DAIFMT_CBM_CFM: sai->is_slave_mode = true; break; case SND_SOC_DAIFMT_CBS_CFM: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; + sai->is_slave_mode = false; break; case SND_SOC_DAIFMT_CBM_CFS: val_cr4 |= FSL_SAI_CR4_FSD_MSTR; -- cgit v1.2.3 From b20f6ed77e68795130356806d40854b254cd70c3 Mon Sep 17 00:00:00 2001 From: Wen Yang Date: Tue, 26 Feb 2019 16:17:51 +0800 Subject: ASoC: eukrea-tlv320: fix a leaked reference by adding missing of_node_put [ Upstream commit b820d52e7eed7b30b2dfef5f4213a2bc3cbea6f3 ] The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/fsl/eukrea-tlv320.c:121:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function. ./sound/soc/fsl/eukrea-tlv320.c:127:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function. Signed-off-by: Wen Yang Cc: Liam Girdwood Cc: Mark Brown Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: alsa-devel@alsa-project.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/fsl/eukrea-tlv320.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 84ef6385736c..4c6f19ef98b2 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -119,13 +119,13 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "fsl,mux-int-port node missing or invalid.\n"); - return ret; + goto err; } ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port); if (ret) { dev_err(&pdev->dev, "fsl,mux-ext-port node missing or invalid.\n"); - return ret; + goto err; } /* -- cgit v1.2.3 From 91126ba6d6143c49f91e9d52ef52aa17f1c55061 Mon Sep 17 00:00:00 2001 From: Wen Yang Date: Tue, 26 Feb 2019 16:17:50 +0800 Subject: ASoC: fsl_utils: fix a leaked reference by adding missing of_node_put [ Upstream commit c705247136a523488eac806bd357c3e5d79a7acd ] The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/fsl/fsl_utils.c:74:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 38, but without a corresponding object release within this function. Signed-off-by: Wen Yang Cc: Timur Tabi Cc: Nicolin Chen Cc: Xiubo Li Cc: Fabio Estevam Cc: Liam Girdwood Cc: Mark Brown Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: alsa-devel@alsa-project.org Cc: linuxppc-dev@lists.ozlabs.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/fsl/fsl_utils.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index b9e42b503a37..4f8bdb7650e8 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -75,6 +75,7 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, iprop = of_get_property(dma_np, "cell-index", NULL); if (!iprop) { of_node_put(dma_np); + of_node_put(dma_channel_np); return -EINVAL; } *dma_id = be32_to_cpup(iprop); -- cgit v1.2.3 From 3e044426367cee282b26fac0d7fa32ec276fffbf Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Thu, 7 Mar 2019 11:11:30 +0100 Subject: ASoC: davinci-mcasp: Fix clang warning without CONFIG_PM [ Upstream commit 8ca5104715cfd14254ea5aecc390ae583b707607 ] Building with clang shows a variable that is only used by the suspend/resume functions but defined outside of their #ifdef block: sound/soc/ti/davinci-mcasp.c:48:12: error: variable 'context_regs' is not needed and will not be emitted We commonly fix these by marking the PM functions as __maybe_unused, but here that would grow the davinci_mcasp structure, so instead add another #ifdef here. Fixes: 1cc0c054f380 ("ASoC: davinci-mcasp: Convert the context save/restore to use array") Signed-off-by: Arnd Bergmann Acked-by: Peter Ujfalusi Reviewed-by: Nathan Chancellor Signed-off-by: Mark Brown Signed-off-by: Sasha Levin --- sound/soc/davinci/davinci-mcasp.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index f395bbc7c354..9aa741d27279 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -43,6 +43,7 @@ #define MCASP_MAX_AFIFO_DEPTH 64 +#ifdef CONFIG_PM static u32 context_regs[] = { DAVINCI_MCASP_TXFMCTL_REG, DAVINCI_MCASP_RXFMCTL_REG, @@ -65,6 +66,7 @@ struct davinci_mcasp_context { u32 *xrsr_regs; /* for serializer configuration */ bool pm_state; }; +#endif struct davinci_mcasp_ruledata { struct davinci_mcasp *mcasp; -- cgit v1.2.3