From: Xiubo Li <Li.Xiubo@freescale.com>
To: <r65073@freescale.com>, <timur@tabi.org>, <lgirdwood@gmail.com>,
<broonie@kernel.org>
Cc: mark.rutland@arm.com, alsa-devel@alsa-project.org,
linux-doc@vger.kernel.org, tiwai@suse.de, b18965@freescale.com,
perex@perex.cz, LW@KARO-electronics.de, linux@arm.linux.org.uk,
b42378@freescale.com, oskar@scara.com, grant.likely@linaro.org,
devicetree@vger.kernel.org, ian.campbell@citrix.com,
pawel.moll@arm.com, swarren@wwwdotorg.org,
rob.herring@calxeda.com, linux-arm-kernel@lists.infradead.org,
fabio.estevam@freescale.com, linux-kernel@vger.kernel.org,
rob@landley.net, r64188@freescale.com, shawn.guo@linaro.org,
linuxppc-dev@lists.ozlabs.org
Subject: [PATCHv1 6/8] ASoC: fsl: add SGT15000 based audio machine driver.
Date: Thu, 17 Oct 2013 17:01:15 +0800 [thread overview]
Message-ID: <1382000477-17304-7-git-send-email-Li.Xiubo@freescale.com> (raw)
In-Reply-To: <1382000477-17304-1-git-send-email-Li.Xiubo@freescale.com>
This is the SGTl5000 codec based audio driver supported with both
playback and capture dai link implemention.
This implementation is only compatible with device tree definition.
Signed-off-by: Alison Wang <b18965@freescale.com
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
---
sound/soc/fsl/Kconfig | 10 +++
sound/soc/fsl/Makefile | 2 +
sound/soc/fsl/fsl-sgtl5000.c | 208 +++++++++++++++++++++++++++++++++++++++++++
3 files changed, 220 insertions(+)
create mode 100644 sound/soc/fsl/fsl-sgtl5000.c
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index a49b386..3fbbbf2 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -220,4 +220,14 @@ config SND_SOC_FSL_PCM
tristate
select SND_SOC_GENERIC_DMAENGINE_PCM
+config SND_SOC_FSL_SGTL5000
+ tristate "SoC Audio support for FSL boards with sgtl5000"
+ depends on OF && I2C
+ select SND_SOC_FSL_SAI
+ select SND_SOC_FSL_PCM
+ select SND_SOC_SGTL5000
+ help
+ Say Y if you want to add support for SoC audio on an FSL board with
+ a sgtl5000 codec.
+
endif # SND_FSL_SOC
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 865ac23..e8bf0bd 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -58,6 +58,8 @@ obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o
# FSL ARM SAI/SGT15000 Platform Support
snd-soc-fsl-sai-objs := fsl-sai.o
snd-soc-fsl-pcm-objs := fsl-pcm-dma.o
+snd-soc-fsl-sgtl5000-objs := fsl-sgtl5000.o
obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o
obj-$(CONFIG_SND_SOC_FSL_PCM) += snd-soc-fsl-pcm.o
+obj-$(CONFIG_SND_SOC_FSL_SGTL5000) += snd-soc-fsl-sgtl5000.o
diff --git a/sound/soc/fsl/fsl-sgtl5000.c b/sound/soc/fsl/fsl-sgtl5000.c
new file mode 100644
index 0000000..bab85ec
--- /dev/null
+++ b/sound/soc/fsl/fsl-sgtl5000.c
@@ -0,0 +1,208 @@
+/*
+ * Freeacale ALSA SoC Audio using SGT1500 as codec.
+ *
+ * Copyright 2012-2013 Freescale Semiconductor, Inc.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/i2c.h>
+#include <linux/clk.h>
+
+#include "../codecs/sgtl5000.h"
+#include "fsl-sai.h"
+
+static unsigned int sysclk_rate;
+
+static int fsl_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+ int ret;
+ struct device *dev = rtd->card->dev;
+
+ ret = snd_soc_dai_set_sysclk(rtd->codec_dai, SGTL5000_SYSCLK,
+ sysclk_rate, SND_SOC_CLOCK_IN);
+ if (ret) {
+ dev_err(dev, "could not set codec driver clock params :%d\n",
+ ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, FSL_SAI_CLK_BUS,
+ sysclk_rate, SND_SOC_CLOCK_OUT);
+ if (ret) {
+ dev_err(dev, "could not set cpu dai driver clock params :%d\n",
+ ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int sgtl5000_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ unsigned int channels = params_channels(params);
+
+ /* TODO: The SAI driver should figure this out for us */
+ switch (channels) {
+ case 2:
+ snd_soc_dai_set_tdm_slot(cpu_dai, 0xfffffffc, 0xfffffffc, 2, 0);
+ break;
+ case 1:
+ snd_soc_dai_set_tdm_slot(cpu_dai, 0xfffffffe, 0xfffffffe, 1, 0);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops fsl_sgtl5000_hifi_ops = {
+ .hw_params = sgtl5000_params,
+};
+
+static struct snd_soc_dai_link fsl_sgtl5000_dai = {
+ .name = "HiFi",
+ .stream_name = "HiFi",
+ .codec_dai_name = "sgtl5000",
+ .init = &fsl_sgtl5000_dai_init,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
+ .ops = &fsl_sgtl5000_hifi_ops,
+};
+
+static const struct snd_soc_dapm_widget fsl_sgtl5000_dapm_widgets[] = {
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_LINE("Line In Jack", NULL),
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Line Out Jack", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static struct snd_soc_card fsl_sgt1500_card = {
+ .owner = THIS_MODULE,
+ .num_links = 1,
+ .dai_link = &fsl_sgtl5000_dai,
+ .dapm_widgets = fsl_sgtl5000_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(fsl_sgtl5000_dapm_widgets),
+};
+
+static int fsl_sgtl5000_parse_dt(struct platform_device *pdev)
+{
+ int ret;
+ struct device_node *sai_np, *codec_np;
+ struct clk *codec_clk;
+ struct i2c_client *codec_dev;
+ struct device_node *np = pdev->dev.of_node;
+
+ ret = snd_soc_of_parse_card_name(&fsl_sgt1500_card, "model");
+ if (ret)
+ return ret;
+
+ ret = snd_soc_of_parse_audio_routing(&fsl_sgt1500_card,
+ "audio-routing");
+ if (ret)
+ return ret;
+
+ sai_np = of_parse_phandle(np, "saif-controller", 0);
+ if (!sai_np) {
+ dev_err(&pdev->dev, "\"saif-controller\" phandle missing or "
+ "invalid\n");
+ return -EINVAL;
+ }
+ fsl_sgtl5000_dai.cpu_of_node = sai_np;
+ fsl_sgtl5000_dai.platform_of_node = sai_np;
+
+ codec_np = of_parse_phandle(np, "audio-codec", 0);
+ if (!codec_np) {
+ dev_err(&pdev->dev, "\"audio-codec\" phandle missing or "
+ "invalid\n");
+ ret = -EINVAL;
+ goto sai_np_fail;
+ }
+ fsl_sgtl5000_dai.codec_of_node = codec_np;
+
+ codec_dev = of_find_i2c_device_by_node(codec_np);
+ if (!codec_dev) {
+ dev_err(&pdev->dev, "failed to find codec platform device\n");
+ ret = PTR_ERR(codec_dev);
+ goto codec_np_fail;
+ }
+
+ codec_clk = devm_clk_get(&codec_dev->dev, NULL);
+ if (IS_ERR(codec_clk)) {
+ dev_err(&pdev->dev, "failed to get codec clock\n");
+ ret = PTR_ERR(codec_clk);
+ goto codec_np_fail;
+ }
+
+ sysclk_rate = clk_get_rate(codec_clk);
+
+codec_np_fail:
+ of_node_put(codec_np);
+sai_np_fail:
+ of_node_put(sai_np);
+
+ return ret;
+}
+
+static int fsl_sgtl5000_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ fsl_sgt1500_card.dev = &pdev->dev;
+
+ ret = fsl_sgtl5000_parse_dt(pdev);
+ if (ret) {
+ dev_err(&pdev->dev,
+ "parse sgtl5000 device tree failed :%d\n",
+ ret);
+ return ret;
+ }
+
+ ret = snd_soc_register_card(&fsl_sgt1500_card);
+ if (ret) {
+ dev_err(&pdev->dev, "register soc sound card failed :%d\n",
+ ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int fsl_sgtl5000_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&fsl_sgt1500_card);
+
+ return 0;
+}
+
+static const struct of_device_id fsl_sgtl5000_dt_ids[] = {
+ { .compatible = "fsl,vf610-sgtl5000", },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, fsl_sgtl5000_dt_ids);
+
+static struct platform_driver fsl_sgtl5000_driver = {
+ .driver = {
+ .name = "fsl-sgtl5000",
+ .owner = THIS_MODULE,
+ .of_match_table = fsl_sgtl5000_dt_ids,
+ },
+ .probe = fsl_sgtl5000_probe,
+ .remove = fsl_sgtl5000_remove,
+};
+module_platform_driver(fsl_sgtl5000_driver);
+
+MODULE_AUTHOR("Xiubo Li <Li.Xiubo@freescale.com>");
+MODULE_DESCRIPTION("Freescale SGTL5000 ASoC driver");
+MODULE_LICENSE("GPL");
--
1.8.0
next prev parent reply other threads:[~2013-10-17 9:11 UTC|newest]
Thread overview: 47+ messages / expand[flat|nested] mbox.gz Atom feed top
2013-10-17 9:01 [PATCHv1 0/8] ALSA: Add SAI driver and enable SGT15000 codec Xiubo Li
2013-10-17 9:01 ` [PATCHv1 1/8] ALSA: Add SAI SoC Digital Audio Interface driver Xiubo Li
2013-10-17 9:42 ` Lothar Waßmann
2013-10-18 3:19 ` Xiubo Li-B47053
2013-10-17 12:15 ` Timur Tabi
2013-10-17 12:21 ` [alsa-devel] " Lars-Peter Clausen
2013-10-17 13:22 ` Timur Tabi
2013-10-17 13:33 ` Lars-Peter Clausen
2013-10-17 13:37 ` Timur Tabi
2013-10-17 13:51 ` Lars-Peter Clausen
2013-10-17 14:10 ` Mark Brown
2013-10-18 3:42 ` Xiubo Li-B47053
2013-10-17 17:43 ` Lars-Peter Clausen
2013-10-21 6:59 ` Xiubo Li-B47053
2013-10-22 2:20 ` Xiubo Li-B47053
2013-10-28 5:58 ` Xiubo Li-B47053
2013-11-12 5:02 ` Vinod Koul
2013-11-12 7:35 ` Li Xiubo
2013-11-12 7:59 ` Lars-Peter Clausen
2013-10-24 11:05 ` Mark Brown
2013-10-28 7:15 ` Xiubo Li-B47053
2013-10-29 4:00 ` Xiubo Li-B47053
2013-10-29 4:02 ` Nicolin Chen
2013-10-29 9:31 ` Xiubo Li-B47053
2013-10-17 9:01 ` [PATCHv1 2/8] ARM: dts: Add Freescale SAI ALSA SoC Digital Audio Interface node for VF610 Xiubo Li
2013-10-17 9:01 ` [PATCHv1 3/8] ARM: dts: Enables SAI ALSA SoC DAI device for Vybrid VF610 TOWER board Xiubo Li
2013-10-17 9:01 ` [PATCHv1 4/8] Documentation: Add device tree bindings for Freescale SAI Xiubo Li
2013-10-17 9:01 ` [PATCHv1 5/8] ASoC: sgtl5000: Revise the bugs about the sgt15000 codec Xiubo Li
2013-10-17 9:56 ` Nicolin Chen
2013-10-21 4:07 ` Xiubo Li-B47053
2013-10-17 10:17 ` Lothar Waßmann
2013-10-21 4:15 ` Xiubo Li-B47053
2013-10-21 8:11 ` Lothar Waßmann
2013-10-21 11:21 ` Timur Tabi
2013-10-18 17:28 ` Mark Brown
2013-10-28 6:07 ` Xiubo Li-B47053
2013-10-17 9:01 ` Xiubo Li [this message]
2013-10-18 17:33 ` [PATCHv1 6/8] ASoC: fsl: add SGT15000 based audio machine driver Mark Brown
2013-10-21 7:50 ` Xiubo Li-B47053
2013-10-17 9:01 ` [PATCHv1 7/8] ARM: dts: Enable SGT15000 codec based audio driver node for VF610 Xiubo Li
2013-10-17 9:01 ` [PATCHv1 8/8] Documentation: Add device tree bindings for Freescale VF610 sound Xiubo Li
2013-10-17 9:46 ` Lucas Stach
2013-10-18 3:27 ` Xiubo Li-B47053
2013-10-18 17:31 ` Mark Brown
2013-10-21 7:24 ` Xiubo Li-B47053
2013-10-22 9:47 ` Mark Brown
2013-10-17 10:22 ` [PATCHv1 0/8] ALSA: Add SAI driver and enable SGT15000 codec Lothar Waßmann
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