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Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/fsl-card' and 'asoc/topic/fsl-mpc5200' into asoc-next
This commit is contained in:
commit
6fee37df02
6 changed files with 49 additions and 8 deletions
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@ -24,6 +24,9 @@ The compatible list for this generic sound card currently:
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"fsl,imx-audio-cs42888"
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"fsl,imx-audio-cs427x"
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(compatible with CS4271 and CS4272)
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"fsl,imx-audio-wm8962"
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(compatible with Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt)
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@ -63,6 +66,12 @@ Optional properties:
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- audio-asrc : The phandle of ASRC. It can be absent if there's no
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need to add ASRC support via DPCM.
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Optional unless SSI is selected as a CPU DAI:
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- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX)
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- mux-ext-port : The external port of the i.MX audio muxer
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Example:
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sound-cs42888 {
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compatible = "fsl,imx-audio-cs42888";
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@ -5,7 +5,7 @@ config SND_DAVINCI_SOC
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config SND_EDMA_SOC
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tristate "SoC Audio for Texas Instruments chips using eDMA"
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depends on SOC_AM33XX || SOC_AM43XX || ARCH_DAVINCI
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depends on TI_EDMA
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select SND_SOC_GENERIC_DMAENGINE_PCM
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help
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Say Y or M here if you want audio support for TI SoC which uses eDMA.
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@ -13,6 +13,7 @@ config SND_EDMA_SOC
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- daVinci devices
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- AM335x
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- AM437x/AM438x
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- DRA7xx family
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config SND_DAVINCI_SOC_I2S
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tristate
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@ -77,6 +77,7 @@ struct davinci_mcasp {
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u32 fifo_base;
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struct device *dev;
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struct snd_pcm_substream *substreams[2];
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unsigned int dai_fmt;
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/* McASP specific data */
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int tdm_slots;
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@ -398,6 +399,9 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
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bool fs_pol_rising;
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bool inv_fs = false;
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if (!fmt)
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return 0;
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pm_runtime_get_sync(mcasp->dev);
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switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
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case SND_SOC_DAIFMT_DSP_A:
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@ -529,6 +533,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
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mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
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mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
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}
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mcasp->dai_fmt = fmt;
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out:
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pm_runtime_put(mcasp->dev);
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return ret;
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@ -1026,6 +1032,10 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
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int period_size = params_period_size(params);
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int ret;
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ret = davinci_mcasp_set_dai_fmt(cpu_dai, mcasp->dai_fmt);
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if (ret)
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return ret;
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/*
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* If mcasp is BCLK master, and a BCLK divider was not provided by
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* the machine driver, we need to calculate the ratio.
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@ -1517,6 +1527,8 @@ static int mcasp_reparent_fck(struct platform_device *pdev)
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if (!parent_name)
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return 0;
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dev_warn(&pdev->dev, "Update the bindings to use assigned-clocks!\n");
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gfclk = clk_get(&pdev->dev, "fck");
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if (IS_ERR(gfclk)) {
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dev_err(&pdev->dev, "failed to get fck\n");
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@ -292,8 +292,8 @@ config SND_SOC_FSL_ASOC_CARD
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select SND_SOC_FSL_SSI
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help
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ALSA SoC Audio support with ASRC feature for Freescale SoCs that have
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ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888
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and SGTL5000.
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ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888,
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CS4271, CS4272 and SGTL5000.
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Say Y if you want to add support for Freescale Generic ASoC Sound Card.
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endif # SND_IMX_SOC
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@ -28,6 +28,8 @@
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#include "../codecs/wm8962.h"
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#include "../codecs/wm8960.h"
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#define CS427x_SYSCLK_MCLK 0
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#define RX 0
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#define TX 1
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@ -99,19 +101,26 @@ struct fsl_asoc_card_priv {
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/**
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* This dapm route map exsits for DPCM link only.
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* The other routes shall go through Device Tree.
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*
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* Note: keep all ASRC routes in the second half
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* to drop them easily for non-ASRC cases.
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*/
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static const struct snd_soc_dapm_route audio_map[] = {
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{"CPU-Playback", NULL, "ASRC-Playback"},
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/* 1st half -- Normal DAPM routes */
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{"Playback", NULL, "CPU-Playback"},
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{"ASRC-Capture", NULL, "CPU-Capture"},
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{"CPU-Capture", NULL, "Capture"},
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/* 2nd half -- ASRC DAPM routes */
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{"CPU-Playback", NULL, "ASRC-Playback"},
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{"ASRC-Capture", NULL, "CPU-Capture"},
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};
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static const struct snd_soc_dapm_route audio_map_ac97[] = {
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{"AC97 Playback", NULL, "ASRC-Playback"},
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/* 1st half -- Normal DAPM routes */
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{"Playback", NULL, "AC97 Playback"},
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{"ASRC-Capture", NULL, "AC97 Capture"},
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{"AC97 Capture", NULL, "Capture"},
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/* 2nd half -- ASRC DAPM routes */
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{"AC97 Playback", NULL, "ASRC-Playback"},
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{"ASRC-Capture", NULL, "AC97 Capture"},
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};
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/* Add all possible widgets into here without being redundant */
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@ -528,6 +537,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
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priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
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priv->cpu_priv.slot_width = 32;
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priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
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} else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) {
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codec_dai_name = "cs4271-hifi";
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priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK;
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priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
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} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
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codec_dai_name = "sgtl5000";
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priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
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@ -593,6 +606,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
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priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
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priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
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/* Drop the second half of DAPM routes -- ASRC */
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if (!asrc_pdev)
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priv->card.num_dapm_routes /= 2;
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memcpy(priv->dai_link, fsl_asoc_card_dai,
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sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
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@ -681,6 +698,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
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static const struct of_device_id fsl_asoc_card_dt_ids[] = {
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{ .compatible = "fsl,imx-audio-ac97", },
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{ .compatible = "fsl,imx-audio-cs42888", },
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{ .compatible = "fsl,imx-audio-cs427x", },
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{ .compatible = "fsl,imx-audio-sgtl5000", },
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{ .compatible = "fsl,imx-audio-wm8962", },
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{ .compatible = "fsl,imx-audio-wm8960", },
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@ -13,6 +13,7 @@
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#include <linux/of_device.h>
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#include <linux/of_platform.h>
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#include <linux/delay.h>
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#include <linux/time.h>
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#include <sound/pcm.h>
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#include <sound/pcm_params.h>
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@ -127,7 +128,7 @@ static void psc_ac97_cold_reset(struct snd_ac97 *ac97)
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mutex_unlock(&psc_dma->mutex);
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msleep(1);
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usleep_range(1000, 2000);
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psc_ac97_warm_reset(ac97);
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}
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