diff options
Diffstat (limited to 'sound/soc')
| -rw-r--r-- | sound/soc/davinci/Kconfig | 3 | ||||
| -rw-r--r-- | sound/soc/davinci/davinci-mcasp.c | 12 | ||||
| -rw-r--r-- | sound/soc/fsl/Kconfig | 4 | ||||
| -rw-r--r-- | sound/soc/fsl/fsl-asoc-card.c | 26 | ||||
| -rw-r--r-- | sound/soc/fsl/mpc5200_psc_ac97.c | 3 |
5 files changed, 40 insertions, 8 deletions
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 3736d9aabc56..50ca291cc225 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -5,7 +5,7 @@ config SND_DAVINCI_SOC config SND_EDMA_SOC tristate "SoC Audio for Texas Instruments chips using eDMA" - depends on SOC_AM33XX || SOC_AM43XX || ARCH_DAVINCI + depends on TI_EDMA select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M here if you want audio support for TI SoC which uses eDMA. @@ -13,6 +13,7 @@ config SND_EDMA_SOC - daVinci devices - AM335x - AM437x/AM438x + - DRA7xx family config SND_DAVINCI_SOC_I2S tristate diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 2ccb8bccc9d4..e1324989bd6b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -77,6 +77,7 @@ struct davinci_mcasp { u32 fifo_base; struct device *dev; struct snd_pcm_substream *substreams[2]; + unsigned int dai_fmt; /* McASP specific data */ int tdm_slots; @@ -398,6 +399,9 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, bool fs_pol_rising; bool inv_fs = false; + if (!fmt) + return 0; + pm_runtime_get_sync(mcasp->dev); switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: @@ -529,6 +533,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); } + + mcasp->dai_fmt = fmt; out: pm_runtime_put(mcasp->dev); return ret; @@ -1026,6 +1032,10 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int period_size = params_period_size(params); int ret; + ret = davinci_mcasp_set_dai_fmt(cpu_dai, mcasp->dai_fmt); + if (ret) + return ret; + /* * If mcasp is BCLK master, and a BCLK divider was not provided by * the machine driver, we need to calculate the ratio. @@ -1517,6 +1527,8 @@ static int mcasp_reparent_fck(struct platform_device *pdev) if (!parent_name) return 0; + dev_warn(&pdev->dev, "Update the bindings to use assigned-clocks!\n"); + gfclk = clk_get(&pdev->dev, "fck"); if (IS_ERR(gfclk)) { dev_err(&pdev->dev, "failed to get fck\n"); diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 14dfdee05fd5..35aabf9dc503 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -292,8 +292,8 @@ config SND_SOC_FSL_ASOC_CARD select SND_SOC_FSL_SSI help ALSA SoC Audio support with ASRC feature for Freescale SoCs that have - ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888 - and SGTL5000. + ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888, + CS4271, CS4272 and SGTL5000. Say Y if you want to add support for Freescale Generic ASoC Sound Card. endif # SND_IMX_SOC diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 562b3bd22d9a..dffd549a0e2a 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -28,6 +28,8 @@ #include "../codecs/wm8962.h" #include "../codecs/wm8960.h" +#define CS427x_SYSCLK_MCLK 0 + #define RX 0 #define TX 1 @@ -99,19 +101,26 @@ struct fsl_asoc_card_priv { /** * This dapm route map exsits for DPCM link only. * The other routes shall go through Device Tree. + * + * Note: keep all ASRC routes in the second half + * to drop them easily for non-ASRC cases. */ static const struct snd_soc_dapm_route audio_map[] = { - {"CPU-Playback", NULL, "ASRC-Playback"}, + /* 1st half -- Normal DAPM routes */ {"Playback", NULL, "CPU-Playback"}, - {"ASRC-Capture", NULL, "CPU-Capture"}, {"CPU-Capture", NULL, "Capture"}, + /* 2nd half -- ASRC DAPM routes */ + {"CPU-Playback", NULL, "ASRC-Playback"}, + {"ASRC-Capture", NULL, "CPU-Capture"}, }; static const struct snd_soc_dapm_route audio_map_ac97[] = { - {"AC97 Playback", NULL, "ASRC-Playback"}, + /* 1st half -- Normal DAPM routes */ {"Playback", NULL, "AC97 Playback"}, - {"ASRC-Capture", NULL, "AC97 Capture"}, {"AC97 Capture", NULL, "Capture"}, + /* 2nd half -- ASRC DAPM routes */ + {"AC97 Playback", NULL, "ASRC-Playback"}, + {"ASRC-Capture", NULL, "AC97 Capture"}, }; /* Add all possible widgets into here without being redundant */ @@ -528,6 +537,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; priv->cpu_priv.slot_width = 32; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } 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; } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { codec_dai_name = "sgtl5000"; priv->codec_priv.mclk_id = SGTL5000_SYSCLK; @@ -593,6 +606,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets; priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets); + /* Drop the second half of DAPM routes -- ASRC */ + if (!asrc_pdev) + priv->card.num_dapm_routes /= 2; + memcpy(priv->dai_link, fsl_asoc_card_dai, sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); @@ -681,6 +698,7 @@ fail: static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-ac97", }, { .compatible = "fsl,imx-audio-cs42888", }, + { .compatible = "fsl,imx-audio-cs427x", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", }, { .compatible = "fsl,imx-audio-wm8960", }, diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 0bab76051fd8..243700cc29e6 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -13,6 +13,7 @@ #include <linux/of_device.h> #include <linux/of_platform.h> #include <linux/delay.h> +#include <linux/time.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -127,7 +128,7 @@ static void psc_ac97_cold_reset(struct snd_ac97 *ac97) mutex_unlock(&psc_dma->mutex); - msleep(1); + usleep_range(1000, 2000); psc_ac97_warm_reset(ac97); } |
