December 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: simple-card: Add device's module clock selection.
by Xiubo Li 02 Jan '14

02 Jan '14

Try to get the device's module clock if the dt has no clocks and system-clock-frequency properties. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- a.out | 0 sound/soc/generic/simple-card.c | 21 ++++++++++++++++++--- 2 files changed, 18 insertions(+), 3 deletions(-) create mode 100644 a.out diff --git a/a.out b/a.out new file mode 100644 index 0000000..e69de29 diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 6ce9745..c0d92813 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -90,14 +90,29 @@ asoc_simple_card_sub_parse_of(struct device_node *np, * dai->sysclk come from * "clocks = <&xxx>" (if system has common clock) * or "system-clock-frequency = <xxx>" + * or device's module clock. */ - clk = of_clk_get(np, 0); - if (IS_ERR(clk)) + if (of_property_read_bool(np, "clocks")) { + clk = of_clk_get(np, 0); + if (IS_ERR(clk)) { + ret = PTR_ERR(clk); + goto parse_error; + } + + dai->sysclk = clk_get_rate(clk); + } else if (of_property_read_bool(np, "system-clock-frequency")) { of_property_read_u32(np, "system-clock-frequency", &dai->sysclk); - else + } else { + clk = of_clk_get(*node, 0); + if (IS_ERR(clk)) { + ret = PTR_ERR(clk); + goto parse_error; + } + dai->sysclk = clk_get_rate(clk); + } ret = 0; -- 1.8.4

4 5

[alsa-devel] [PATCH] ASoC: ad1836: Reject unsupported bit sizes
by Mark Brown 01 Jan '14

01 Jan '14

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ad1836.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 9a92b79..d7c9838 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -179,6 +179,8 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S32_LE: word_len = AD1836_WORD_LEN_24; break; + default: + return -EINVAL; } regmap_update_bits(ad1836->regmap, AD1836_DAC_CTRL1, -- 1.8.5.2

2 1

[alsa-devel] [PATCH] ASoC: make snd_soc_dai_link more symmetrical
by Jean-Francois Moine 01 Jan '14

01 Jan '14

This patch renders optional the CODEC name / OF node on sound card register, as it is done for the CPU side, in the case the CODEC DAI name is unique. This simplifies the definition of CODECs with multiple DAIs, especially in DTs, while keeping compatibility. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- include/sound/soc.h | 6 ++++-- sound/soc/soc-core.c | 73 ++++++++++++++++++++++++++++++++++++++++---------------------- 2 files changed, 51 insertions(+), 28 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 1f741cb..67c382e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -852,8 +852,10 @@ struct snd_soc_dai_link { */ const char *cpu_dai_name; /* - * You MUST specify the link's codec, either by device name, or by - * DT/OF node, but not both. + * You MAY specify the link's codec, either by device name, or by + * DT/OF node, but not both. If this information is omitted, + * the CODEC DAI is matched using .codec_dai_name only, which hence + * must be globally unique. */ const char *codec_name; const struct device_node *codec_of_node; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4e53d87..11b729c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -883,28 +883,54 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) return -EPROBE_DEFER; } - /* Find CODEC from registered CODECs */ - list_for_each_entry(codec, &codec_list, list) { - if (dai_link->codec_of_node) { - if (codec->dev->of_node != dai_link->codec_of_node) - continue; - } else { - if (strcmp(codec->name, dai_link->codec_name)) - continue; + if (dai_link->codec_of_node || dai_link->codec_name) { + + /* Find CODEC from registered CODECs */ + list_for_each_entry(codec, &codec_list, list) { + if (dai_link->codec_of_node) { + if (codec->dev->of_node != dai_link->codec_of_node) + continue; + } else { + if (strcmp(codec->name, dai_link->codec_name)) + continue; + } + + rtd->codec = codec; + + /* + * CODEC found, so find CODEC DAI from registered DAIs from + * this CODEC + */ + list_for_each_entry(codec_dai, &dai_list, list) { + if (codec->dev == codec_dai->dev && + !strcmp(codec_dai->name, + dai_link->codec_dai_name)) { + + rtd->codec_dai = codec_dai; + } + } + + if (!rtd->codec_dai) { + dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", + dai_link->codec_dai_name); + return -EPROBE_DEFER; + } } - rtd->codec = codec; + if (!rtd->codec) { + dev_err(card->dev, "ASoC: CODEC %s not registered\n", + dai_link->codec_name); + return -EPROBE_DEFER; + } + } else { - /* - * CODEC found, so find CODEC DAI from registered DAIs from - * this CODEC - */ + /* find CODEC DAI from registered DAIs */ list_for_each_entry(codec_dai, &dai_list, list) { - if (codec->dev == codec_dai->dev && - !strcmp(codec_dai->name, + if (!strcmp(codec_dai->name, dai_link->codec_dai_name)) { - rtd->codec_dai = codec_dai; + rtd->codec = codec_dai->codec; + break; } } @@ -915,12 +941,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } } - if (!rtd->codec) { - dev_err(card->dev, "ASoC: CODEC %s not registered\n", - dai_link->codec_name); - return -EPROBE_DEFER; - } - /* if there's no platform we match on the empty platform */ platform_name = dai_link->platform_name; if (!platform_name && !dai_link->platform_of_node) @@ -3710,12 +3730,13 @@ int snd_soc_register_card(struct snd_soc_card *card) struct snd_soc_dai_link *link = &card->dai_link[i]; /* - * Codec must be specified by 1 of name or OF node, - * not both or neither. + * Codec may be specified by either name or OF node, + * but can be left unspecified + * and will be matched based on CODEC DAI name alone. */ - if (!!link->codec_name == !!link->codec_of_node) { + if (link->codec_name && link->codec_of_node) { dev_err(card->dev, - "ASoC: Neither/both codec name/of_node are set for %s\n", + "ASoC: Both codec name/of_node are set for %s\n", link->name); return -EINVAL; } -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 3

[alsa-devel] [PATCH] ASoC: imx-spdif: Use snd-soc-dummy CODEC driver to link card
by Nicolin Chen 01 Jan '14

01 Jan '14

This is a quick fix for the below two issues when building spdif as modules. 1) If modprobing modules in order: (Step 1) snd-soc-fsl-spdif -> (Step 2) snd-soc-imx-spdif -> (Step 3) snd-soc-spdif-tx/rx, we will fail to create imx-spdif card and dai link unless we rmmod snd-soc-imx-spdif and modprobe it again due to the execution platform_driver_unregister() in probe() when meeting -EPROBE_DEFER at Step 2. 2) After "imx-spdif sound-spdif.17: dit-hifi <-> 2004000.spdif mapping ok", 'rmmod snd-soc-imx-spdif' would cause kernel dump with warning: WARNING: CPU: 0 PID: 1301 at /home/rmk/git/linux-rmk/fs/sysfs/dir.c:915 sysfs_hash_and_remove+0x84/0x90() sysfs: can not remove 'dapm_widget', no directory This should be caused by disordered resourse releasing of the whole link. And trying to unregister the card and then CODEC dev can't fix this issue. Thus this patch just provides a simple fix to these two bugs by using the snd-soc-dummy in the core instead of seperate snd-soc-spdif-tx/rx so that there's no need to handle the registering and unregistering of CODEC or CODEC dai any more. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/Kconfig | 1 - sound/soc/fsl/imx-spdif.c | 84 ++++++++++++----------------------------------- 2 files changed, 21 insertions(+), 64 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index bb4ab96..1c7b3b7 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -201,7 +201,6 @@ config SND_SOC_IMX_SPDIF tristate "SoC Audio support for i.MX boards with S/PDIF" select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_SPDIF - select SND_SOC_SPDIF select REGMAP_MMIO help SoC Audio support for i.MX boards with S/PDIF diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index 980dd1f..e1dc401 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -14,17 +14,15 @@ #include <sound/soc.h> struct imx_spdif_data { - struct snd_soc_dai_link dai[2]; + struct snd_soc_dai_link dai; struct snd_soc_card card; - struct platform_device *txdev; - struct platform_device *rxdev; }; static int imx_spdif_audio_probe(struct platform_device *pdev) { struct device_node *spdif_np, *np = pdev->dev.of_node; struct imx_spdif_data *data; - int ret = 0, num_links = 0; + int ret = 0; spdif_np = of_parse_phandle(np, "spdif-controller", 0); if (!spdif_np) { @@ -39,69 +37,42 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) goto end; } - if (of_property_read_bool(np, "spdif-out")) { - data->dai[num_links].name = "S/PDIF TX"; - data->dai[num_links].stream_name = "S/PDIF PCM Playback"; - data->dai[num_links].codec_dai_name = "dit-hifi"; - data->dai[num_links].codec_name = "spdif-dit"; - data->dai[num_links].cpu_of_node = spdif_np; - data->dai[num_links].platform_of_node = spdif_np; - num_links++; - - data->txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0); - if (IS_ERR(data->txdev)) { - ret = PTR_ERR(data->txdev); - dev_err(&pdev->dev, "register dit failed: %d\n", ret); - goto end; - } - } + data->dai.name = "S/PDIF PCM"; + data->dai.stream_name = "S/PDIF PCM"; + data->dai.codec_dai_name = "snd-soc-dummy-dai"; + data->dai.codec_name = "snd-soc-dummy"; + data->dai.cpu_of_node = spdif_np; + data->dai.platform_of_node = spdif_np; + data->dai.playback_only = true; + data->dai.capture_only = true; - if (of_property_read_bool(np, "spdif-in")) { - data->dai[num_links].name = "S/PDIF RX"; - data->dai[num_links].stream_name = "S/PDIF PCM Capture"; - data->dai[num_links].codec_dai_name = "dir-hifi"; - data->dai[num_links].codec_name = "spdif-dir"; - data->dai[num_links].cpu_of_node = spdif_np; - data->dai[num_links].platform_of_node = spdif_np; - num_links++; - - data->rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0); - if (IS_ERR(data->rxdev)) { - ret = PTR_ERR(data->rxdev); - dev_err(&pdev->dev, "register dir failed: %d\n", ret); - goto error_dit; - } - } + if (of_property_read_bool(np, "spdif-out")) + data->dai.capture_only = false; + + if (of_property_read_bool(np, "spdif-in")) + data->dai.playback_only = false; - if (!num_links) { + if (data->dai.playback_only && data->dai.capture_only) { dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n"); - goto error_dir; + goto end; } data->card.dev = &pdev->dev; - data->card.num_links = num_links; - data->card.dai_link = data->dai; + data->card.dai_link = &data->dai; + data->card.num_links = 1; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) - goto error_dir; + goto end; ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret); - goto error_dir; + goto end; } platform_set_drvdata(pdev, data); - goto end; - -error_dir: - if (data->rxdev) - platform_device_unregister(data->rxdev); -error_dit: - if (data->txdev) - platform_device_unregister(data->txdev); end: if (spdif_np) of_node_put(spdif_np); @@ -109,18 +80,6 @@ end: return ret; } -static int imx_spdif_audio_remove(struct platform_device *pdev) -{ - struct imx_spdif_data *data = platform_get_drvdata(pdev); - - if (data->rxdev) - platform_device_unregister(data->rxdev); - if (data->txdev) - platform_device_unregister(data->txdev); - - return 0; -} - static const struct of_device_id imx_spdif_dt_ids[] = { { .compatible = "fsl,imx-audio-spdif", }, { /* sentinel */ } @@ -134,7 +93,6 @@ static struct platform_driver imx_spdif_driver = { .of_match_table = imx_spdif_dt_ids, }, .probe = imx_spdif_audio_probe, - .remove = imx_spdif_audio_remove, }; module_platform_driver(imx_spdif_driver); -- 1.8.4

2 2

[alsa-devel] [PATCH] ASoC: simple-card: Add cpu_dai and codec_dai names NULL check
by Xiubo Li 31 Dec '13

31 Dec '13

The name of cpu DAI maybe omitted, and then strlen() will lead kernel panic. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 3d190d0..1d87db8 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -142,6 +142,9 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (ret < 0) return ret; + if (!info->cpu_dai.name || !info->codec_dai.name) + return -EINVAL; + /* card name is created from CPU/CODEC dai name */ name = devm_kzalloc(dev, strlen(info->cpu_dai.name) + -- 1.8.4

2 3

[alsa-devel] [PATCH] ASoC: simple-card: Cleanup __asoc_simple_card_dai_init() ret check
by Xiubo Li 31 Dec '13

31 Dec '13

The ret parameter is always equal to zero till here. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 643c9ed..6ce9745 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -25,7 +25,7 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, daifmt |= set->fmt; - if (!ret && daifmt) + if (daifmt) ret = snd_soc_dai_set_fmt(dai, daifmt); if (ret == -ENOTSUPP) { -- 1.8.4

2 1

[alsa-devel] [Patch v2 1/2] ASoC: atmel: sam9x5_wm8731: fix oops when unload module
by Bo Shen 31 Dec '13

31 Dec '13

As the priv is not assigned to card->drvdata, it is NULL, so when unload module, it will cause NULL pointer oops. Assign priv to card->drvdata to fix this issue. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Change in v2: - using snd_soc_card_set_drvdata() --- sound/soc/atmel/sam9x5_wm8731.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index 992ae38..1b37228 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -97,6 +97,8 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) goto out; } + snd_soc_card_set_drvdata(card, priv); + card->dev = &pdev->dev; card->owner = THIS_MODULE; card->dai_link = dai; -- 1.7.9.5

3 6

[alsa-devel] [PATCH v2] ASoC: fsl_sai: Fix one bug for hardware limitation.
by Xiubo Li 31 Dec '13

31 Dec '13

This is maybe one bug or a limitation of the hardware that the {T,R}CR2's Synchronous Mode bits must be set as late as possible, or the SAI device maybe hanged up, and there has not any explaination about this limitation in the SAI Data Sheet. And the {T,R}CR2's Synchronous Mode bits must be set at the same time whether for Tx or Rx stream. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index af80246..2ece147 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -145,7 +145,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: val_cr4 |= FSL_SAI_CR4_FSE; - val_cr4 |= FSL_SAI_CR4_FSP; break; default: return -EINVAL; @@ -185,9 +184,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, return -EINVAL; } - if (fsl_dir == FSL_FMT_RECEIVER) - val_cr2 |= FSL_SAI_CR2_SYNC; - sai_writel(sai, val_cr2, sai->base + reg_cr2); sai_writel(sai, val_cr4, sai->base + reg_cr4); @@ -253,6 +249,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 |= FSL_SAI_CR5_WNW(word_width); val_cr5 |= FSL_SAI_CR5_W0W(word_width); + val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; if (sai->big_endian_data) val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); else @@ -272,7 +269,15 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 tcsr, rcsr, val_cr3, reg_cr3; + u32 tcsr, rcsr, val_cr2, val_cr3, reg_cr3; + + val_cr2 = sai_readl(sai, sai->base + FSL_SAI_TCR2); + val_cr2 &= ~FSL_SAI_CR2_SYNC; + sai_writel(sai, val_cr2, sai->base + FSL_SAI_TCR2); + + val_cr2 = sai_readl(sai, sai->base + FSL_SAI_RCR2); + val_cr2 |= FSL_SAI_CR2_SYNC; + sai_writel(sai, val_cr2, sai->base + FSL_SAI_RCR2); tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR); rcsr = sai_readl(sai, sai->base + FSL_SAI_RCSR); -- 1.8.4

2 3

[alsa-devel] HD audio problem (fwd)
by Russ Whitaker 31 Dec '13

31 Dec '13

Have not heard back from you. What should I do? ---------- Forwarded message ---------- Date: Thu, 19 Dec 2013 23:48:19 -0800 (PST) From: Russ Whitaker <russ(a)ashlandhome.net> To: alsa-devel(a)alsa-project.org Subject: HD audio problem Hi Bought a new mini-itx mb: ECS KBN-I On board sound chip: VT1705CF Installed kernel linux-3.10.24 Otherwise distro is Slackware64 14.1 mplayer plays .mp3 files just fine. No sound from video clips that use Adobe flashplayer, and I get this error message: ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave I think the problem is: card0 and card1 are interchanged as card0 is listed as the HDMI video output. How to fixit, please. Thank you, Russ

1 0

[alsa-devel] [PATCH] ASoC: fsl_sai: Fix one bug for hardware limitation.
by Xiubo Li 31 Dec '13

31 Dec '13

This is maybe one bug or a limitation of the hardware that the {T,R}CR2's Synchronous Mode bits must be set as late as possible, or the SAI device maybe hanged up, and there has not any explaination about this limitation in the SAI Data Sheet. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 31 ++++++++++++++++++++++--------- 1 file changed, 22 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index af80246..59228a10 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -62,6 +62,7 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, reg_cr2 = FSL_SAI_RCR2; val_cr2 = sai_readl(sai, sai->base + reg_cr2); + switch (clk_id) { case FSL_SAI_CLK_BUS: val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; @@ -82,6 +83,7 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, default: return -EINVAL; } + sai_writel(sai, val_cr2, sai->base + reg_cr2); return 0; @@ -138,14 +140,13 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, val_cr4 = sai_readl(sai, sai->base + reg_cr4); if (sai->big_endian_data) - val_cr4 |= FSL_SAI_CR4_MF; - else val_cr4 &= ~FSL_SAI_CR4_MF; + else + val_cr4 |= FSL_SAI_CR4_MF; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: val_cr4 |= FSL_SAI_CR4_FSE; - val_cr4 |= FSL_SAI_CR4_FSP; break; default: return -EINVAL; @@ -185,9 +186,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, return -EINVAL; } - if (fsl_dir == FSL_FMT_RECEIVER) - val_cr2 |= FSL_SAI_CR2_SYNC; - sai_writel(sai, val_cr2, sai->base + reg_cr2); sai_writel(sai, val_cr4, sai->base + reg_cr4); @@ -253,10 +251,11 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 |= FSL_SAI_CR5_WNW(word_width); val_cr5 |= FSL_SAI_CR5_W0W(word_width); + val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; if (sai->big_endian_data) - val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); - else val_cr5 |= FSL_SAI_CR5_FBT(0); + else + val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); val_cr4 |= FSL_SAI_CR4_FRSZ(channels); val_mr = ~0UL - ((1 << channels) - 1); @@ -327,8 +326,22 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + u32 val_cr2; + int ret; + + ret = clk_prepare_enable(sai->clk); + if (ret) + return ret; + + val_cr2 = sai_readl(sai, sai->base + FSL_SAI_TCR2); + val_cr2 &= ~FSL_SAI_CR2_SYNC; + sai_writel(sai, val_cr2, sai->base + FSL_SAI_TCR2); - return clk_prepare_enable(sai->clk); + val_cr2 = sai_readl(sai, sai->base + FSL_SAI_RCR2); + val_cr2 |= FSL_SAI_CR2_SYNC; + sai_writel(sai, val_cr2, sai->base + FSL_SAI_RCR2); + + return 0; } static void fsl_sai_shutdown(struct snd_pcm_substream *substream, -- 1.8.4

3 2