October 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: SOF: Intel: hda-stream: fix the CONFIG_ prefix missing
by Pierre-Louis Bossart 28 Oct '19

28 Oct '19

From: Keyon Jie <yang.jie(a)linux.intel.com> We are missing the 'CONFIG_' prefix when using the kernel configure item SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1, here correct them. Fixes: 43b2ab9009b13b ('ASoC: SOF: Intel: hda: Disable DMI L1 entry during capture') Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-stream.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 450f9c55785f..29ab43281670 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -190,7 +190,7 @@ hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction) * Workaround to address a known issue with host DMA that results * in xruns during pause/release in capture scenarios. */ - if (!IS_ENABLED(SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) + if (!IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) if (stream && direction == SNDRV_PCM_STREAM_CAPTURE) snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, HDA_VS_INTEL_EM2, @@ -228,7 +228,7 @@ int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag) spin_unlock_irq(&bus->reg_lock); /* Enable DMI L1 entry if there are no capture streams open */ - if (!IS_ENABLED(SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) + if (!IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) if (!active_capture_stream) snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, HDA_VS_INTEL_EM2, -- 2.20.1

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[alsa-devel] [PATCH] ASoC: ti: sdma-pcm: Add back the flags parameter for non standard dma names
by Peter Ujfalusi 28 Oct '19

28 Oct '19

When non standard names are used it is possible that one of the directions are not provided, thus the flags needs to be present to tell the core that we have half duplex setup. Fixes: 642aafea8889 ("ASoC: ti: remove compat dma probing") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/ti/sdma-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/ti/sdma-pcm.c b/sound/soc/ti/sdma-pcm.c index a236350beb10..2b0bc234e1b6 100644 --- a/sound/soc/ti/sdma-pcm.c +++ b/sound/soc/ti/sdma-pcm.c @@ -62,7 +62,7 @@ int sdma_pcm_platform_register(struct device *dev, config->chan_names[0] = txdmachan; config->chan_names[1] = rxdmachan; - return devm_snd_dmaengine_pcm_register(dev, config, 0); + return devm_snd_dmaengine_pcm_register(dev, config, flags); } EXPORT_SYMBOL_GPL(sdma_pcm_platform_register); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH] ASoC: rockchip: rockchip_max98090: Enable SHDN to fix headset detection
by Cheng-Yi Chiang 28 Oct '19

28 Oct '19

max98090 spec states that chip needs to be in turned-on state to supply mic bias. Enable SHDN dapm widget along with MICBIAS widget to actually turn on mic bias for proper headset button detection. This is similar to cht_ti_jack_event in sound/soc/intel/boards/cht_bsw_max98090_ti.c. Note that due to ts3a227e reports the jack event right away before the notifier is registered, if headset is plugged on boot, headset button will not get detected until headset is unplugged and plugged. This is still an issue to be fixed. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/rockchip/rockchip_max98090.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c index 0097df1fae66..e80b09143b63 100644 --- a/sound/soc/rockchip/rockchip_max98090.c +++ b/sound/soc/rockchip/rockchip_max98090.c @@ -66,10 +66,13 @@ static int rk_jack_event(struct notifier_block *nb, unsigned long event, struct snd_soc_jack *jack = (struct snd_soc_jack *)data; struct snd_soc_dapm_context *dapm = &jack->card->dapm; - if (event & SND_JACK_MICROPHONE) + if (event & SND_JACK_MICROPHONE) { snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); - else + snd_soc_dapm_force_enable_pin(dapm, "SHDN"); + } else { snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + snd_soc_dapm_disable_pin(dapm, "SHDN"); + } snd_soc_dapm_sync(dapm); -- 2.24.0.rc0.303.g954a862665-goog

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[alsa-devel] [PATCH] ASoC: fsl: fsl_dma: fix build failure
by Michael Ellerman 28 Oct '19

28 Oct '19

Commit 4ac85de9977e ("ASoC: fsl: fsl_dma: remove snd_pcm_ops") removed fsl_dma_ops but left a usage, leading to a build error for some configs, eg. mpc85xx_defconfig: sound/soc/fsl/fsl_dma.c: In function ‘fsl_soc_dma_probe’: sound/soc/fsl/fsl_dma.c:905:18: error: ‘fsl_dma_ops’ undeclared (first use in this function) dma->dai.ops = &fsl_dma_ops; ^~~~~~~~~~~ Remove the usage to fix the build. Fixes: 4ac85de9977e ("ASoC: fsl: fsl_dma: remove snd_pcm_ops") Signed-off-by: Michael Ellerman <mpe(a)ellerman.id.au> --- sound/soc/fsl/fsl_dma.c | 1 - 1 file changed, 1 deletion(-) This breakage is only in linux-next. diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index a092726510d4..2868c4f97cb2 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -901,7 +901,6 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) } dma->dai.name = DRV_NAME; - dma->dai.ops = &fsl_dma_ops; dma->dai.open = fsl_dma_open; dma->dai.close = fsl_dma_close; dma->dai.ioctl = snd_soc_pcm_lib_ioctl; -- 2.21.0

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[alsa-devel] [PATCH -next] ASoC: sunxi: sun4i-codec: remove unneeded semicolon
by YueHaibing 28 Oct '19

28 Oct '19

remove unneeded semicolon. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/sunxi/sun4i-codec.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index ee448d5..34f3e0b 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1442,7 +1442,7 @@ static struct snd_soc_card *sun8i_a23_codec_create_card(struct device *dev) if (!aux_dev.dlc.of_node) { dev_err(dev, "Can't find analog controls for codec.\n"); return ERR_PTR(-EINVAL); - }; + } card->dai_link = sun4i_codec_create_link(dev, &card->num_links); if (!card->dai_link) @@ -1480,7 +1480,7 @@ static struct snd_soc_card *sun8i_h3_codec_create_card(struct device *dev) if (!aux_dev.dlc.of_node) { dev_err(dev, "Can't find analog controls for codec.\n"); return ERR_PTR(-EINVAL); - }; + } card->dai_link = sun4i_codec_create_link(dev, &card->num_links); if (!card->dai_link) @@ -1518,7 +1518,7 @@ static struct snd_soc_card *sun8i_v3s_codec_create_card(struct device *dev) if (!aux_dev.dlc.of_node) { dev_err(dev, "Can't find analog controls for codec.\n"); return ERR_PTR(-EINVAL); - }; + } card->dai_link = sun4i_codec_create_link(dev, &card->num_links); if (!card->dai_link) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: eve: implement set_bias_level function for rt5514
by Brent Lu 28 Oct '19

28 Oct '19

The first DMIC capture always fail (zero sequence data from PCM port) after using DSP hotwording function (i.e. Google assistant). This rt5514 codec requires to control mclk directly in the set_bias_level function. Implement this function in machine driver to control the ssp1_mclk clock explicitly could fix this issue. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 50 ++++++++++++++++++++++ 1 file changed, 50 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index dc09a85..b546de8 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -653,6 +653,55 @@ static struct snd_soc_dai_link kabylake_dais[] = { }, }; +static int kabylake_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) +{ + struct snd_soc_component *component = dapm->component; + struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); + int ret = 0; + + if (!component || strcmp(component->name, RT5514_DEV_NAME)) + return 0; + + if (IS_ERR(priv->mclk)) + return 0; + + /* + * It's required to control mclk directly in the set_bias_level + * function for rt5514 codec or the recording function could + * break. + */ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_ON) { + dev_dbg(card->dev, "Disable mclk"); + clk_disable_unprepare(priv->mclk); + } else { + dev_dbg(card->dev, "Enable mclk"); + ret = clk_set_rate(priv->mclk, 24000000); + if (ret) { + dev_err(card->dev, "Can't set rate for mclk, err: %d\n", + ret); + return ret; + } + + ret = clk_prepare_enable(priv->mclk); + if (ret) { + dev_err(card->dev, "Can't enable mclk, err: %d\n", + ret); + + /* mclk is already enabled in FW */ + ret = 0; + } + } + break; + default: + break; + } + + return ret; +} + static int kabylake_card_late_probe(struct snd_soc_card *card) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); @@ -692,6 +741,7 @@ static struct snd_soc_card kabylake_audio_card = { .owner = THIS_MODULE, .dai_link = kabylake_dais, .num_links = ARRAY_SIZE(kabylake_dais), + .set_bias_level = kabylake_set_bias_level, .controls = kabylake_controls, .num_controls = ARRAY_SIZE(kabylake_controls), .dapm_widgets = kabylake_widgets, -- 2.7.4

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[alsa-devel] [PATCH V3] ASoC: fsl_esai: Add spin lock to protect reset, stop and start
by Shengjiu Wang 28 Oct '19

28 Oct '19

xrun may happen at the end of stream, the trigger->fsl_esai_trigger_stop maybe called in the middle of fsl_esai_hw_reset, this may cause esai in wrong state after stop, and there may be endless xrun interrupt. This issue may also happen with trigger->fsl_esai_trigger_start. So Add spin lock to lock those functions. Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after xrun") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Change in v3 -refine comments Change in v2 -add lock for fsl_esai_trigger_start. sound/soc/fsl/fsl_esai.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 37b14c48b537..453392a9cd68 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -33,6 +33,7 @@ * @fsysclk: system clock source to derive HCK, SCK and FS * @spbaclk: SPBA clock (optional, depending on SoC design) * @task: tasklet to handle the reset operation + * @lock: spin lock between hw_reset() and trigger() * @fifo_depth: depth of tx/rx FIFO * @slot_width: width of each DAI slot * @slots: number of slots @@ -56,6 +57,7 @@ struct fsl_esai { struct clk *fsysclk; struct clk *spbaclk; struct tasklet_struct task; + spinlock_t lock; /* Protect hw_reset and trigger */ u32 fifo_depth; u32 slot_width; u32 slots; @@ -676,8 +678,10 @@ static void fsl_esai_hw_reset(unsigned long arg) { struct fsl_esai *esai_priv = (struct fsl_esai *)arg; bool tx = true, rx = false, enabled[2]; + unsigned long lock_flags; u32 tfcr, rfcr; + spin_lock_irqsave(&esai_priv->lock, lock_flags); /* Save the registers */ regmap_read(esai_priv->regmap, REG_ESAI_TFCR, &tfcr); regmap_read(esai_priv->regmap, REG_ESAI_RFCR, &rfcr); @@ -715,6 +719,8 @@ static void fsl_esai_hw_reset(unsigned long arg) fsl_esai_trigger_start(esai_priv, tx); if (enabled[rx]) fsl_esai_trigger_start(esai_priv, rx); + + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); } static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -722,6 +728,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned long lock_flags; esai_priv->channels[tx] = substream->runtime->channels; @@ -729,12 +736,16 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_start(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_stop(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; default: return -EINVAL; @@ -1002,6 +1013,7 @@ static int fsl_esai_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, esai_priv); + spin_lock_init(&esai_priv->lock); ret = fsl_esai_hw_init(esai_priv); if (ret) return ret; -- 2.21.0

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[alsa-devel] [PATCH V3] ASoC: fsl_asrc: refine the setting of internal clock divider
by Shengjiu Wang 28 Oct '19

28 Oct '19

The output divider should align with the output sample rate, if use ideal sample rate, there will be a lot of overload, which would cause underrun. The maximum divider of asrc clock is 1024, but there is no judgement for this limitation in driver, which may cause the divider setting not correct. For non-ideal ratio mode, the clock rate should divide the sample rate with no remainder, and the quotient should be less than 1024. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Change in v3 - refine comments Change in v2 - remove p2p/m2m word - use use_ideal_rate sound/soc/fsl/fsl_asrc.c | 45 ++++++++++++++++++++++++++++++---------- 1 file changed, 34 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 0bf91a6f54b9..a3cfceea7d2f 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -259,8 +259,15 @@ static int fsl_asrc_set_ideal_ratio(struct fsl_asrc_pair *pair, * It configures those ASRC registers according to a configuration instance * of struct asrc_config which includes in/output sample rate, width, channel * and clock settings. + * + * Note: + * The ideal ratio configuration can work with a flexible clock rate setting. + * Using IDEAL_RATIO_RATE gives a faster converting speed but overloads ASRC. + * For a regular audio playback, the clock rate should not be slower than an + * clock rate aligning with the output sample rate; For a use case requiring + * faster conversion, set use_ideal_rate to have the faster speed. */ -static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) +static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool use_ideal_rate) { struct asrc_config *config = pair->config; struct fsl_asrc *asrc_priv = pair->asrc_priv; @@ -268,7 +275,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) enum asrc_word_width input_word_width; enum asrc_word_width output_word_width; u32 inrate, outrate, indiv, outdiv; - u32 clk_index[2], div[2]; + u32 clk_index[2], div[2], rem[2]; + u64 clk_rate; int in, out, channels; int pre_proc, post_proc; struct clk *clk; @@ -351,27 +359,42 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) /* We only have output clock for ideal ratio mode */ clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; - div[IN] = clk_get_rate(clk) / inrate; - if (div[IN] == 0) { + clk_rate = clk_get_rate(clk); + rem[IN] = do_div(clk_rate, inrate); + div[IN] = (u32)clk_rate; + + /* + * The divider range is [1, 1024], defined by the hardware. For non- + * ideal ratio configuration, clock rate has to be strictly aligned + * with the sample rate. For ideal ratio configuration, clock rates + * only result in different converting speeds. So remainder does not + * matter, as long as we keep the divider within its valid range. + */ + if (div[IN] == 0 || (!ideal && (div[IN] > 1024 || rem[IN] != 0))) { pair_err("failed to support input sample rate %dHz by asrck_%x\n", inrate, clk_index[ideal ? OUT : IN]); return -EINVAL; } + div[IN] = min_t(u32, 1024, div[IN]); + clk = asrc_priv->asrck_clk[clk_index[OUT]]; - - /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ - if (ideal) - div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; + clk_rate = clk_get_rate(clk); + if (ideal && use_ideal_rate) + rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); else - div[OUT] = clk_get_rate(clk) / outrate; + rem[OUT] = do_div(clk_rate, outrate); + div[OUT] = clk_rate; - if (div[OUT] == 0) { + /* Output divider has the same limitation as the input one */ + if (div[OUT] == 0 || (!ideal && (div[OUT] > 1024 || rem[OUT] != 0))) { pair_err("failed to support output sample rate %dHz by asrck_%x\n", outrate, clk_index[OUT]); return -EINVAL; } + div[OUT] = min_t(u32, 1024, div[OUT]); + /* Set the channel number */ channels = config->channel_num; @@ -560,7 +583,7 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } - ret = fsl_asrc_config_pair(pair); + ret = fsl_asrc_config_pair(pair, false); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); return ret; -- 2.21.0

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[alsa-devel] [PATCH] ASoC: rsnd: core.c: add WARN_ON() on rsnd_channel_normalization()
by Kuninori Morimoto 28 Oct '19

28 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> chan > 8 or chan < 0 shouldn't happen. This patch uses WARN_ON() for such case. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index e9242a0..295faad 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -302,7 +302,7 @@ int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io, int rsnd_channel_normalization(int chan) { - if ((chan > 8) || (chan < 0)) + if (WARN_ON((chan > 8) || (chan < 0))) return 0; /* TDM Extend Mode needs 8ch */ -- 2.7.4

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[alsa-devel] [PATCH] ALSA: pci: Fix memory leak in snd_korg1212_create
by Navid Emamdoost 28 Oct '19

28 Oct '19

In the implementation of snd_korg1212_create() the allocated memory for korg1212 is leaked in cases of error. Release korg1212 via snd_korg1212_free() if either of these calls fail: snd_korg1212_downloadDSPCode(), snd_pcm_new(), or snd_ctl_add(). Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/pci/korg1212/korg1212.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 0d81eac0a478..e976e857d915 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -2367,8 +2367,10 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, mdelay(CARD_BOOT_DELAY_IN_MS); - if (snd_korg1212_downloadDSPCode(korg1212)) + if (snd_korg1212_downloadDSPCode(korg1212)) { + snd_korg1212_free(korg1212); return -EBUSY; + } K1212_DEBUG_PRINTK("korg1212: dspMemPhy = %08x U[%08x], " "PlayDataPhy = %08x L[%08x]\n" @@ -2383,8 +2385,11 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, korg1212->RoutingTablePhy, LowerWordSwap(korg1212->RoutingTablePhy), korg1212->AdatTimeCodePhy, LowerWordSwap(korg1212->AdatTimeCodePhy)); - if ((err = snd_pcm_new(korg1212->card, "korg1212", 0, 1, 1, &korg1212->pcm)) < 0) + err = snd_pcm_new(korg1212->card, "korg1212", 0, 1, 1, &korg1212->pcm); + if (err < 0) { + snd_korg1212_free(korg1212); return err; + } korg1212->pcm->private_data = korg1212; korg1212->pcm->private_free = snd_korg1212_free_pcm; @@ -2398,8 +2403,10 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, for (i = 0; i < ARRAY_SIZE(snd_korg1212_controls); i++) { err = snd_ctl_add(korg1212->card, snd_ctl_new1(&snd_korg1212_controls[i], korg1212)); - if (err < 0) + if (err < 0) { + snd_korg1212_free(korg1212); return err; + } } snd_korg1212_proc_init(korg1212); -- 2.17.1

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