diff options
author | Takashi Iwai <tiwai@suse.de> | 2021-11-01 18:58:27 +0300 |
---|---|---|
committer | Takashi Iwai <tiwai@suse.de> | 2021-11-01 18:58:27 +0300 |
commit | a0292f3ebe63f8ed7ea28de57751f6bfb9416242 (patch) | |
tree | c1a9c859dbc4f9cd1c9dfcf255f58ade4d14177f /sound/soc/fsl | |
parent | 8f27b689066113a3e579d4df171c980c54368c4e (diff) | |
parent | 318a54c0ee4aaa3bfd69fdf505588510c7672c0c (diff) | |
download | linux-a0292f3ebe63f8ed7ea28de57751f6bfb9416242.tar.xz |
Merge tag 'asoc-v5.16' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.16
This is an unusually large set of updates, mostly a large crop of
unusually big drivers coupled with extensive overhauls of existing code.
There's a SH change here for the DAI format terminology, the change is
straightforward and the SH maintainers don't seem very active.
- A new version of the audio graph card which supports a wider range of
systems.
- Move of the Cirrus DSP framework into drivers/firmware to allow for
future use by non-audio DSPs.
- Several conversions to YAML DT bindings.
- Continuing cleanups to the SOF and Intel code.
- A very big overhaul of the cs42l42 driver, correcting many problems.
- Support for AMD Vangogh and Yelow Cap, Cirrus CS35L41, Maxim
MAX98520 and MAX98360A, Mediatek MT8195, Nuvoton NAU8821, nVidia
Tegra210, NXP i.MX8ULP, Qualcomm AudioReach, Realtek ALC5682I-VS,
RT5682S, and RT9120 and Rockchip RV1126 and RK3568
Diffstat (limited to 'sound/soc/fsl')
-rw-r--r-- | sound/soc/fsl/eukrea-tlv320.c | 2 | ||||
-rw-r--r-- | sound/soc/fsl/fsl-asoc-card.c | 54 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_audmix.c | 8 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_esai.c | 28 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_mqs.c | 4 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_rpmsg.c | 47 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_rpmsg.h | 12 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_sai.c | 34 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_sai.h | 2 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_spdif.c | 85 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_ssi.c | 38 | ||||
-rw-r--r-- | sound/soc/fsl/imx-audmix.c | 12 | ||||
-rw-r--r-- | sound/soc/fsl/imx-card.c | 6 | ||||
-rw-r--r-- | sound/soc/fsl/imx-es8328.c | 2 | ||||
-rw-r--r-- | sound/soc/fsl/imx-hdmi.c | 6 | ||||
-rw-r--r-- | sound/soc/fsl/imx-rpmsg.c | 2 | ||||
-rw-r--r-- | sound/soc/fsl/imx-sgtl5000.c | 2 | ||||
-rw-r--r-- | sound/soc/fsl/mpc8610_hpcd.c | 16 | ||||
-rw-r--r-- | sound/soc/fsl/p1022_ds.c | 16 | ||||
-rw-r--r-- | sound/soc/fsl/p1022_rdk.c | 2 |
20 files changed, 257 insertions, 121 deletions
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index e13271ea84de..8b61582753c8 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -70,7 +70,7 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, + SND_SOC_DAIFMT_CBP_CFP, .ops = &eukrea_tlv320_snd_ops, SND_SOC_DAILINK_REG(hifi), }; diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 06107ae46e20..6e6494f9f399 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -356,8 +356,8 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, * If only 4 wires are needed, just set SSI into * synchronous mode and enable 4 PADs in IOMUX. */ - switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (priv->dai_fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) | IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) | IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | @@ -367,7 +367,7 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, IMX_AUDMUX_V2_PTCR_TFSDIR | IMX_AUDMUX_V2_PTCR_TCLKDIR; break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBP_CFC: int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) | IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | IMX_AUDMUX_V2_PTCR_RCLKDIR | @@ -377,7 +377,7 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, IMX_AUDMUX_V2_PTCR_RFSDIR | IMX_AUDMUX_V2_PTCR_TFSDIR; break; - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBC_CFP: int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) | IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | IMX_AUDMUX_V2_PTCR_RFSDIR | @@ -387,7 +387,7 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, IMX_AUDMUX_V2_PTCR_RCLKDIR | IMX_AUDMUX_V2_PTCR_TCLKDIR; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) | IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) | IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | @@ -533,8 +533,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct device_node *cpu_np, *codec_np, *asrc_np; struct device_node *np = pdev->dev.of_node; struct platform_device *asrc_pdev = NULL; - struct device_node *bitclkmaster = NULL; - struct device_node *framemaster = NULL; + struct device_node *bitclkprovider = NULL; + struct device_node *frameprovider = NULL; struct platform_device *cpu_pdev; struct fsl_asoc_card_priv *priv; struct device *codec_dev = NULL; @@ -617,29 +617,29 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; priv->cpu_priv.slot_width = 32; - priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC; } else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) { codec_dai_name = "cs4271-hifi"; priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK; - priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { codec_dai_name = "sgtl5000"; priv->codec_priv.mclk_id = SGTL5000_SYSCLK; - priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; } else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic32x4")) { codec_dai_name = "tlv320aic32x4-hifi"; - priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { codec_dai_name = "wm8962"; priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; priv->codec_priv.pll_id = WM8962_FLL; - priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { codec_dai_name = "wm8960-hifi"; priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; - priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { codec_dai_name = "ac97-hifi"; priv->dai_fmt = SND_SOC_DAIFMT_AC97; @@ -648,7 +648,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) } else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) { codec_dai_name = "fsl-mqs-dai"; priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J | - SND_SOC_DAIFMT_CBS_CFS | + SND_SOC_DAIFMT_CBC_CFC | SND_SOC_DAIFMT_NB_NF; priv->dai_link[1].dpcm_capture = 0; priv->dai_link[2].dpcm_capture = 0; @@ -656,7 +656,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) { codec_dai_name = "wm8524-hifi"; - priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC; priv->dai_link[1].dpcm_capture = 0; priv->dai_link[2].dpcm_capture = 0; priv->cpu_priv.slot_width = 32; @@ -664,12 +664,12 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); } else if (of_device_is_compatible(np, "fsl,imx-audio-si476x")) { codec_dai_name = "si476x-codec"; - priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC; priv->card.dapm_routes = audio_map_rx; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx); } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8958")) { codec_dai_name = "wm8994-aif1"; - priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; priv->codec_priv.mclk_id = WM8994_FLL_SRC_MCLK1; priv->codec_priv.fll_id = WM8994_SYSCLK_FLL1; priv->codec_priv.pll_id = WM8994_FLL1; @@ -683,29 +683,29 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) } /* Format info from DT is optional. */ - snd_soc_daifmt_parse_clock_provider_as_phandle(np, NULL, &bitclkmaster, &framemaster); - if (bitclkmaster || framemaster) { + snd_soc_daifmt_parse_clock_provider_as_phandle(np, NULL, &bitclkprovider, &frameprovider); + if (bitclkprovider || frameprovider) { unsigned int daifmt = snd_soc_daifmt_parse_format(np, NULL); - if (codec_np == bitclkmaster) - daifmt |= (codec_np == framemaster) ? - SND_SOC_DAIFMT_CBM_CFM : SND_SOC_DAIFMT_CBM_CFS; + if (codec_np == bitclkprovider) + daifmt |= (codec_np == frameprovider) ? + SND_SOC_DAIFMT_CBP_CFP : SND_SOC_DAIFMT_CBP_CFC; else - daifmt |= (codec_np == framemaster) ? - SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS; + daifmt |= (codec_np == frameprovider) ? + SND_SOC_DAIFMT_CBC_CFP : SND_SOC_DAIFMT_CBC_CFC; /* Override dai_fmt with value from DT */ priv->dai_fmt = daifmt; } /* Change direction according to format */ - if (priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) { + if (priv->dai_fmt & SND_SOC_DAIFMT_CBP_CFP) { priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_IN; priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_IN; } - of_node_put(bitclkmaster); - of_node_put(framemaster); + of_node_put(bitclkprovider); + of_node_put(frameprovider); if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { dev_dbg(&pdev->dev, "failed to find codec device\n"); diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index f931288e256c..6dbb8c99f626 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -257,10 +257,10 @@ static int fsl_audmix_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - /* For playback the AUDMIX is slave, and for record is master */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - case SND_SOC_DAIFMT_CBS_CFS: + /* For playback the AUDMIX is consumer, and for record is provider */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index bda66b30e063..3a9e2df4e16f 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -52,7 +52,7 @@ struct fsl_esai_soc_data { * @sck_rate: clock rate of desired SCKx clock * @hck_dir: the direction of HCKx pads * @sck_div: if using PSR/PM dividers for SCKx clock - * @slave_mode: if fully using DAI slave mode + * @consumer_mode: if fully using DAI clock consumer mode * @synchronous: if using tx/rx synchronous mode * @name: driver name */ @@ -78,7 +78,7 @@ struct fsl_esai { u32 sck_rate[2]; bool hck_dir[2]; bool sck_div[2]; - bool slave_mode; + bool consumer_mode; bool synchronous; char name[32]; }; @@ -366,8 +366,8 @@ static int fsl_esai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) u32 sub, ratio = hck_rate / freq; int ret; - /* Don't apply for fully slave mode or unchanged bclk */ - if (esai_priv->slave_mode || esai_priv->sck_rate[tx] == freq) + /* Don't apply for fully consumer mode or unchanged bclk */ + if (esai_priv->consumer_mode || esai_priv->sck_rate[tx] == freq) return 0; if (ratio * freq > hck_rate) @@ -476,20 +476,20 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - esai_priv->slave_mode = false; + esai_priv->consumer_mode = false; - /* DAI clock master masks */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - esai_priv->slave_mode = true; + /* DAI clock provider masks */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + esai_priv->consumer_mode = true; break; - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBC_CFP: xccr |= ESAI_xCCR_xCKD; break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBP_CFC: xccr |= ESAI_xCCR_xFSD; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: xccr |= ESAI_xCCR_xFSD | ESAI_xCCR_xCKD; break; default: @@ -1016,8 +1016,8 @@ static int fsl_esai_probe(struct platform_device *pdev) /* Set a default slot number */ esai_priv->slots = 2; - /* Set a default master/slave state */ - esai_priv->slave_mode = true; + /* Set a default clock provider state */ + esai_priv->consumer_mode = true; /* Determine the FIFO depth */ iprop = of_get_property(np, "fsl,fifo-depth", NULL); diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index 69aeb0e71844..27b4536dce44 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -102,8 +102,8 @@ static int fsl_mqs_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; diff --git a/sound/soc/fsl/fsl_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c index d60f4dac6c1b..8508bc7f239d 100644 --- a/sound/soc/fsl/fsl_rpmsg.c +++ b/sound/soc/fsl/fsl_rpmsg.c @@ -138,11 +138,43 @@ static const struct snd_soc_component_driver fsl_component = { .name = "fsl-rpmsg", }; +static const struct fsl_rpmsg_soc_data imx7ulp_data = { + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, +}; + +static const struct fsl_rpmsg_soc_data imx8mm_data = { + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_DSD_U8 | + SNDRV_PCM_FMTBIT_DSD_U16_LE | SNDRV_PCM_FMTBIT_DSD_U32_LE, +}; + +static const struct fsl_rpmsg_soc_data imx8mn_data = { + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, +}; + +static const struct fsl_rpmsg_soc_data imx8mp_data = { + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, +}; + static const struct of_device_id fsl_rpmsg_ids[] = { - { .compatible = "fsl,imx7ulp-rpmsg-audio"}, - { .compatible = "fsl,imx8mm-rpmsg-audio"}, - { .compatible = "fsl,imx8mn-rpmsg-audio"}, - { .compatible = "fsl,imx8mp-rpmsg-audio"}, + { .compatible = "fsl,imx7ulp-rpmsg-audio", .data = &imx7ulp_data}, + { .compatible = "fsl,imx8mm-rpmsg-audio", .data = &imx8mm_data}, + { .compatible = "fsl,imx8mn-rpmsg-audio", .data = &imx8mn_data}, + { .compatible = "fsl,imx8mp-rpmsg-audio", .data = &imx8mp_data}, + { .compatible = "fsl,imx8ulp-rpmsg-audio", .data = &imx7ulp_data}, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_rpmsg_ids); @@ -157,6 +189,13 @@ static int fsl_rpmsg_probe(struct platform_device *pdev) if (!rpmsg) return -ENOMEM; + rpmsg->soc_data = of_device_get_match_data(&pdev->dev); + + fsl_rpmsg_dai.playback.rates = rpmsg->soc_data->rates; + fsl_rpmsg_dai.capture.rates = rpmsg->soc_data->rates; + fsl_rpmsg_dai.playback.formats = rpmsg->soc_data->formats; + fsl_rpmsg_dai.capture.formats = rpmsg->soc_data->formats; + if (of_property_read_bool(np, "fsl,enable-lpa")) { rpmsg->enable_lpa = 1; rpmsg->buffer_size = LPA_LARGE_BUFFER_SIZE; diff --git a/sound/soc/fsl/fsl_rpmsg.h b/sound/soc/fsl/fsl_rpmsg.h index 4f5b49eb18d8..b04086fbf828 100644 --- a/sound/soc/fsl/fsl_rpmsg.h +++ b/sound/soc/fsl/fsl_rpmsg.h @@ -7,6 +7,16 @@ #define __FSL_RPMSG_H /* + * struct fsl_rpmsg_soc_data + * @rates: supported rates + * @formats: supported formats + */ +struct fsl_rpmsg_soc_data { + int rates; + u64 formats; +}; + +/* * struct fsl_rpmsg - rpmsg private data * * @ipg: ipg clock for cpu dai (SAI) @@ -15,6 +25,7 @@ * @pll8k: parent clock for multiple of 8kHz frequency * @pll11k: parent clock for multiple of 11kHz frequency * @card_pdev: Platform_device pointer to register a sound card + * @soc_data: soc specific data * @mclk_streams: Active streams that are using baudclk * @force_lpa: force enable low power audio routine if condition satisfy * @enable_lpa: enable low power audio routine according to dts setting @@ -27,6 +38,7 @@ struct fsl_rpmsg { struct clk *pll8k; struct clk *pll11k; struct platform_device *card_pdev; + const struct fsl_rpmsg_soc_data *soc_data; unsigned int mclk_streams; int force_lpa; int enable_lpa; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 38f6362099d5..10544fa27dc0 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -297,23 +297,23 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, return -EINVAL; } - /* DAI clock master masks */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + /* DAI clock provider masks */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; val_cr4 |= FSL_SAI_CR4_FSD_MSTR; - sai->is_slave_mode = false; + sai->is_consumer_mode = false; break; - case SND_SOC_DAIFMT_CBM_CFM: - sai->is_slave_mode = true; + case SND_SOC_DAIFMT_CBP_CFP: + sai->is_consumer_mode = true; break; - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBC_CFP: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; - sai->is_slave_mode = false; + sai->is_consumer_mode = false; break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBP_CFC: val_cr4 |= FSL_SAI_CR4_FSD_MSTR; - sai->is_slave_mode = true; + sai->is_consumer_mode = true; break; default: return -EINVAL; @@ -356,8 +356,8 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) u32 id; int ret = 0; - /* Don't apply to slave mode */ - if (sai->is_slave_mode) + /* Don't apply to consumer mode */ + if (sai->is_consumer_mode) return 0; /* @@ -462,7 +462,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, pins = DIV_ROUND_UP(channels, slots); - if (!sai->is_slave_mode) { + if (!sai->is_consumer_mode) { if (sai->bclk_ratio) ret = fsl_sai_set_bclk(cpu_dai, tx, sai->bclk_ratio * @@ -502,12 +502,12 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr4 |= FSL_SAI_CR4_CHMOD; /* - * For SAI master mode, when Tx(Rx) sync with Rx(Tx) clock, Rx(Tx) will + * For SAI provider mode, when Tx(Rx) sync with Rx(Tx) clock, Rx(Tx) will * generate bclk and frame clock for Tx(Rx), we should set RCR4(TCR4), * RCR5(TCR5) for playback(capture), or there will be sync error. */ - if (!sai->is_slave_mode && fsl_sai_dir_is_synced(sai, adir)) { + if (!sai->is_consumer_mode && fsl_sai_dir_is_synced(sai, adir)) { regmap_update_bits(sai->regmap, FSL_SAI_xCR4(!tx, ofs), FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK | FSL_SAI_CR4_CHMOD_MASK, @@ -543,7 +543,7 @@ static int fsl_sai_hw_free(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), FSL_SAI_CR3_TRCE_MASK, 0); - if (!sai->is_slave_mode && + if (!sai->is_consumer_mode && sai->mclk_streams & BIT(substream->stream)) { clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[tx]]); sai->mclk_streams &= ~BIT(substream->stream); @@ -577,7 +577,7 @@ static void fsl_sai_config_disable(struct fsl_sai *sai, int dir) * This is a hardware bug, and will be fix in the * next sai version. */ - if (!sai->is_slave_mode) { + if (!sai->is_consumer_mode) { /* Software Reset */ regmap_write(sai->regmap, FSL_SAI_xCSR(tx, ofs), FSL_SAI_CSR_SR); /* Clear SR bit to finish the reset */ diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index bc60030967dd..9aaf231bc024 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -259,7 +259,7 @@ struct fsl_sai { struct clk *bus_clk; struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; - bool is_slave_mode; + bool is_consumer_mode; bool is_lsb_first; bool is_dsp_mode; bool synchronous[2]; diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 1c53719bb61e..d178b479c8bd 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -111,6 +111,7 @@ struct spdif_mixer_control { * @dma_params_tx: DMA parameters for transmit channel * @dma_params_rx: DMA parameters for receive channel * @regcache_srpc: regcache for SRPC + * @bypass: status of bypass input to output */ struct fsl_spdif_priv { const struct fsl_spdif_soc_data *soc; @@ -133,6 +134,7 @@ struct fsl_spdif_priv { struct snd_dmaengine_dai_dma_data dma_params_rx; /* regcache for SRPC */ u32 regcache_srpc; + bool bypass; }; static struct fsl_spdif_soc_data fsl_spdif_vf610 = { @@ -186,6 +188,16 @@ static struct fsl_spdif_soc_data fsl_spdif_imx8mm = { .tx_formats = FSL_SPDIF_FORMATS_PLAYBACK, }; +static struct fsl_spdif_soc_data fsl_spdif_imx8ulp = { + .imx = true, + .shared_root_clock = true, + .raw_capture_mode = false, + .interrupts = 1, + .tx_burst = 2, /* Applied for EDMA */ + .rx_burst = 2, /* Applied for EDMA */ + .tx_formats = SNDRV_PCM_FMTBIT_S24_LE, /* Applied for EDMA */ +}; + /* Check if clk is a root clock that does not share clock source with others */ static inline bool fsl_spdif_can_set_clk_rate(struct fsl_spdif_priv *spdif, int clk) { @@ -895,6 +907,69 @@ static int fsl_spdif_rx_rcm_put(struct snd_kcontrol *kcontrol, return 0; } +static int fsl_spdif_bypass_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *priv = snd_soc_dai_get_drvdata(dai); + + ucontrol->value.integer.value[0] = priv->bypass ? 1 : 0; + + return 0; +} + +static int fsl_spdif_bypass_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *priv = snd_soc_dai_get_drvdata(dai); + struct snd_soc_card *card = dai->component->card; + bool set = (ucontrol->value.integer.value[0] != 0); + struct regmap *regmap = priv->regmap; + struct snd_soc_pcm_runtime *rtd; + u32 scr, mask; + int stream; + + rtd = snd_soc_get_pcm_runtime(card, card->dai_link); + + if (priv->bypass == set) + return 0; /* nothing to do */ + + if (snd_soc_dai_active(dai)) { + dev_err(dai->dev, "Cannot change BYPASS mode while stream is running.\n"); + return -EBUSY; + } + + pm_runtime_get_sync(dai->dev); + + if (set) { + /* Disable interrupts */ + regmap_update_bits(regmap, REG_SPDIF_SIE, 0xffffff, 0); + + /* Configure BYPASS mode */ + scr = SCR_TXSEL_RX | SCR_RXFIFO_OFF; + mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK | + SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK | SCR_TXSEL_MASK; + /* Power up SPDIF module */ + mask |= SCR_LOW_POWER; + } else { + /* Power down SPDIF module, disable TX */ + scr = SCR_LOW_POWER | SCR_TXSEL_OFF; + mask = SCR_LOW_POWER | SCR_TXSEL_MASK; + } + + regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr); + + /* Disable playback & capture if BYPASS mode is enabled, enable otherwise */ + for_each_pcm_streams(stream) + rtd->pcm->streams[stream].substream_count = (set ? 0 : 1); + + priv->bypass = set; + pm_runtime_put_sync(dai->dev); + + return 0; +} + /* DPLL lock information */ static int fsl_spdif_rxrate_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -1065,6 +1140,15 @@ static struct snd_kcontrol_new fsl_spdif_ctrls[] = { .info = fsl_spdif_rxrate_info, .get = fsl_spdif_rxrate_get, }, + /* RX bypass controller */ + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "Bypass Mode", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_mono_info, + .get = fsl_spdif_bypass_get, + .put = fsl_spdif_bypass_put, + }, /* User bit sync mode set/get controller */ { .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -1560,6 +1644,7 @@ static const struct of_device_id fsl_spdif_dt_ids[] = { { .compatible = "fsl,imx6sx-spdif", .data = &fsl_spdif_imx6sx, }, { .compatible = "fsl,imx8qm-spdif", .data = &fsl_spdif_imx8qm, }, { .compatible = "fsl,imx8mm-spdif", .data = &fsl_spdif_imx8mm, }, + { .compatible = "fsl,imx8ulp-spdif", .data = &fsl_spdif_imx8ulp, }, {} }; MODULE_DEVICE_TABLE(of, fsl_spdif_dt_ids); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ecbc1c365d5b..1169d1104b9e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -350,16 +350,16 @@ static bool fsl_ssi_is_ac97(struct fsl_ssi *ssi) SND_SOC_DAIFMT_AC97; } -static bool fsl_ssi_is_i2s_master(struct fsl_ssi *ssi) +static bool fsl_ssi_is_i2s_clock_provider(struct fsl_ssi *ssi) { - return (ssi->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) == - SND_SOC_DAIFMT_CBS_CFS; + return (ssi->dai_fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) == + SND_SOC_DAIFMT_CBC_CFC; } -static bool fsl_ssi_is_i2s_cbm_cfs(struct fsl_ssi *ssi) +static bool fsl_ssi_is_i2s_cbp_cfc(struct fsl_ssi *ssi) { - return (ssi->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) == - SND_SOC_DAIFMT_CBM_CFS; + return (ssi->dai_fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) == + SND_SOC_DAIFMT_CBP_CFC; } /** @@ -808,7 +808,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, u32 wl = SSI_SxCCR_WL(sample_size); int ret; - if (fsl_ssi_is_i2s_master(ssi)) { + if (fsl_ssi_is_i2s_clock_provider(ssi)) { ret = fsl_ssi_set_bclk(substream, dai, hw_params); if (ret) return ret; @@ -841,7 +841,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, u8 i2s_net = ssi->i2s_net; /* Normal + Network mode to send 16-bit data in 32-bit frames */ - if (fsl_ssi_is_i2s_cbm_cfs(ssi) && sample_size == 16) + if (fsl_ssi_is_i2s_cbp_cfc(ssi) && sample_size == 16) i2s_net = SSI_SCR_I2S_MODE_NORMAL | SSI_SCR_NET; /* Use Normal mode to send mono data at 1st slot of 2 slots */ @@ -865,7 +865,7 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); - if (fsl_ssi_is_i2s_master(ssi) && + if (fsl_ssi_is_i2s_clock_provider(ssi) && ssi->baudclk_streams & BIT(substream->stream)) { clk_disable_unprepare(ssi->baudclk); ssi->baudclk_streams &= ~BIT(substream->stream); @@ -891,18 +891,18 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt) ssi->i2s_net = SSI_SCR_NET; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: if (IS_ERR(ssi->baudclk)) { dev_err(ssi->dev, "missing baudclk for master mode\n"); return -EINVAL; } fallthrough; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBP_CFC: ssi->i2s_net |= SSI_SCR_I2S_MODE_MASTER; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: ssi->i2s_net |= SSI_SCR_I2S_MODE_SLAVE; break; default: @@ -962,17 +962,17 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt) return -EINVAL; } - /* DAI clock master masks */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + /* DAI clock provider masks */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: /* Output bit and frame sync clocks */ strcr |= SSI_STCR_TFDIR | SSI_STCR_TXDIR; scr |= SSI_SCR_SYS_CLK_EN; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: /* Input bit or frame sync clocks */ break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBP_CFC: /* Input bit clock but output frame sync clock */ strcr |= SSI_STCR_TFDIR; break; @@ -1341,7 +1341,7 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, } } - /* Do not error out for slave cases that live without a baud clock */ + /* Do not error out for consumer cases that live without a baud clock */ ssi->baudclk = devm_clk_get(dev, "baud"); if (IS_ERR(ssi->baudclk)) dev_dbg(dev, "failed to get baud clock: %ld\n", diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index a364e2415de0..502fe1b522ab 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -80,8 +80,8 @@ static int imx_audmix_fe_hw_params(struct snd_pcm_substream *substream, u32 channels = params_channels(params); int ret, dir; - /* For playback the AUDMIX is slave, and for record is master */ - fmt |= tx ? SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBM_CFM; + /* For playback the AUDMIX is consumer, and for record is provider */ + fmt |= tx ? SND_SOC_DAIFMT_CBC_CFC : SND_SOC_DAIFMT_CBP_CFP; dir = tx ? SND_SOC_CLOCK_OUT : SND_SOC_CLOCK_IN; /* set DAI configuration */ @@ -121,8 +121,8 @@ static int imx_audmix_be_hw_params(struct snd_pcm_substream *substream, if (!tx) return 0; - /* For playback the AUDMIX is slave */ - fmt |= SND_SOC_DAIFMT_CBM_CFM; + /* For playback the AUDMIX is consumer */ + fmt |= SND_SOC_DAIFMT_CBP_CFP; /* set AUDMIX DAI configuration */ ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), fmt); @@ -132,12 +132,12 @@ static int imx_audmix_be_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops imx_audmix_fe_ops = { +static const struct snd_soc_ops imx_audmix_fe_ops = { .startup = imx_audmix_fe_startup, .hw_params = imx_audmix_fe_hw_params, }; -static struct snd_soc_ops imx_audmix_be_ops = { +static const struct snd_soc_ops imx_audmix_be_ops = { .hw_params = imx_audmix_be_hw_params, }; diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c index 58fd0639a069..6f06afd23b16 100644 --- a/sound/soc/fsl/imx-card.c +++ b/sound/soc/fsl/imx-card.c @@ -442,12 +442,12 @@ static int imx_aif_startup(struct snd_pcm_substream *substream) return ret; } -static struct snd_soc_ops imx_aif_ops = { +static const struct snd_soc_ops imx_aif_ops = { .hw_params = imx_aif_hw_params, .startup = imx_aif_startup, }; -static struct snd_soc_ops imx_aif_ops_be = { +static const struct snd_soc_ops imx_aif_ops_be = { .hw_params = imx_aif_hw_params, }; @@ -652,7 +652,7 @@ static int imx_card_parse_of(struct imx_card_data *data) NULL, &link->dai_fmt); if (ret) link->dai_fmt = SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS | + SND_SOC_DAIFMT_CBC_CFC | SND_SOC_DAIFMT_I2S; /* Get tdm slot */ diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index 1981dcd7e930..09c674ee79f1 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.c @@ -174,7 +174,7 @@ static int imx_es8328_probe(struct platform_device *pdev) data->dai.platforms->of_node = ssi_np; data->dai.init = &imx_es8328_dai_init; data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; + SND_SOC_DAIFMT_CBP_CFP; data->card.dev = dev; data->card.dapm_widgets = imx_es8328_dapm_widgets; diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index 34a0dceae621..f10359a28800 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -59,7 +59,7 @@ static int imx_hdmi_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops imx_hdmi_ops = { +static const struct snd_soc_ops imx_hdmi_ops = { .hw_params = imx_hdmi_hw_params, }; @@ -171,7 +171,7 @@ static int imx_hdmi_probe(struct platform_device *pdev) data->dai.codecs->name = "hdmi-audio-codec.1"; data->dai.dai_fmt = data->dai_fmt | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS; + SND_SOC_DAIFMT_CBC_CFC; } if (hdmi_in) { @@ -181,7 +181,7 @@ static int imx_hdmi_probe(struct platform_device *pdev) data->dai.codecs->name = "hdmi-audio-codec.2"; data->dai.dai_fmt = data->dai_fmt | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; + SND_SOC_DAIFMT_CBP_CFP; } data->card.dapm_widgets = imx_hdmi_widgets; diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c index f96fe4ff8425..2e117311e582 100644 --- a/sound/soc/fsl/imx-rpmsg.c +++ b/sound/soc/fsl/imx-rpmsg.c @@ -64,7 +64,7 @@ static int imx_rpmsg_probe(struct platform_device *pdev) data->dai.stream_name = "rpmsg hifi"; data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS; + SND_SOC_DAIFMT_CBC_CFC; /* Optional codec node */ ret = of_parse_phandle_with_fixed_args(np, "audio-codec", 0, 0, &args); diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index f45cb4bbb6c4..2f1acd011042 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -153,7 +153,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->dai.platforms->of_node = ssi_np; data->dai.init = &imx_sgtl5000_dai_init; data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; + SND_SOC_DAIFMT_CBP_CFP; data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 58b9ca3c4da0..e71a992fbf93 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -264,7 +264,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) if (strcasecmp(sprop, "i2s-slave") == 0) { machine_data->dai_format = - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM; + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBP_CFP; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; @@ -282,37 +282,37 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) machine_data->clk_frequency = be32_to_cpup(iprop); } else if (strcasecmp(sprop, "i2s-master") == 0) { machine_data->dai_format = - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS; + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBC_CFC; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "lj-slave") == 0) { machine_data->dai_format = - SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM; + SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBP_CFP; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "lj-master") == 0) { machine_data->dai_format = - SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS; + SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBC_CFC; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "rj-slave") == 0) { machine_data->dai_format = - SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBM_CFM; + SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBP_CFP; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "rj-master") == 0) { machine_data->dai_format = - SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBS_CFS; + SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBC_CFC; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "ac97-slave") == 0) { machine_data->dai_format = - SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBM_CFM; + SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBP_CFP; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "ac97-master") == 0) { machine_data->dai_format = - SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBS_CFS; + SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBC_CFC; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else { diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 317c767b0099..b45742931b0d 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -275,7 +275,7 @@ static int p1022_ds_probe(struct platform_device *pdev) if (strcasecmp(sprop, "i2s-slave") == 0) { mdata->dai_format = SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM; + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBP_CFP; mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; @@ -293,37 +293,37 @@ static int p1022_ds_probe(struct platform_device *pdev) mdata->clk_frequency = be32_to_cpup(iprop); } else if (strcasecmp(sprop, "i2s-master") == 0) { mdata->dai_format = SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS; + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBC_CFC; mdata->codec_clk_direction = SND_SOC_CLOCK_IN; mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "lj-slave") == 0) { mdata->dai_format = SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM; + SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBP_CFP; mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "lj-master") == 0) { mdata->dai_format = SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS; + SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBC_CFC; mdata->codec_clk_direction = SND_SOC_CLOCK_IN; mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "rj-slave") == 0) { mdata->dai_format = SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBM_CFM; + SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBP_CFP; mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "rj-master") == 0) { mdata->dai_format = SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBS_CFS; + SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBC_CFC; mdata->codec_clk_direction = SND_SOC_CLOCK_IN; mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "ac97-slave") == 0) { mdata->dai_format = SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBM_CFM; + SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBP_CFP; mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "ac97-master") == 0) { mdata->dai_format = SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBS_CFS; + SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBC_CFC; mdata->codec_clk_direction = SND_SOC_CLOCK_IN; mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else { diff --git a/sound/soc/fsl/p1022_rdk.c b/sound/soc/fsl/p1022_rdk.c index 714515b8081f..b395adabe823 100644 --- a/sound/soc/fsl/p1022_rdk.c +++ b/sound/soc/fsl/p1022_rdk.c @@ -265,7 +265,7 @@ static int p1022_rdk_probe(struct platform_device *pdev) * only one way to configure the SSI. */ mdata->dai_format = SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM; + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBP_CFP; mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; |