sound/soc/fsl/imx-tlv320aic23.c
#include <linux/module.h>
#include <linux/of.h>
#include <linux/of_platform.h>
#include <linux/i2c.h>
#include <linux/clk.h>
#include <sound/soc.h>
#include "../codecs/tlv320aic23.h"
#include "imx-audmux.h"
#define DAI_NAME_SIZE 32
struct imx_tlv320aic23_data {
struct snd_soc_dai_link dai;
struct snd_soc_card card;
char codec_dai_name[DAI_NAME_SIZE];
char platform_name[DAI_NAME_SIZE];
struct clk *codec_clk;
unsigned int clk_frequency;
};
static int imx_tlv320aic23_dai_init(struct snd_soc_pcm_runtime *rtd)
{
struct imx_tlv320aic23_data *data = snd_soc_card_get_drvdata(rtd->card);
struct device *dev = rtd->card->dev;
int ret;
ret = snd_soc_dai_set_sysclk(rtd->codec_dai, TLV320AIC23_SYSCLK,
data->clk_frequency, SND_SOC_CLOCK_IN);
if (ret) {
dev_err(dev, "could not set codec driver clock params\n");
return ret;
}
return 0;
}
static const struct snd_soc_dapm_widget imx_tlv320aic23_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 int imx_tlv320aic23_audmux_config(struct platform_device *pdev)
{
struct device_node *np = pdev->dev.of_node;
int int_port, ext_port;
int ret;
ret = of_property_read_u32(np, "mux-int-port", &int_port);
if (ret) {
dev_err(&pdev->dev, "mux-int-port missing or invalid\n");
return ret;
}
ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
if (ret) {
dev_err(&pdev->dev, "mux-ext-port missing or invalid\n");
return ret;
}
/*
* The port numbering in the hardware manual starts at 1, while
* the audmux API expects it starts at 0.
*/
int_port--;
ext_port--;
ret = imx_audmux_v2_configure_port(int_port,
IMX_AUDMUX_V2_PTCR_SYN |
IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
IMX_AUDMUX_V2_PTCR_TFSDIR |
IMX_AUDMUX_V2_PTCR_TCLKDIR,
IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
if (ret) {
dev_err(&pdev->dev, "audmux internal port setup failed\n");
return ret;
}
ret = imx_audmux_v2_configure_port(ext_port,
IMX_AUDMUX_V2_PTCR_SYN,
IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
if (ret) {
dev_err(&pdev->dev, "audmux external port setup failed\n");
return ret;
}
return 0;
}
static int imx_tlv320aic23_probe(struct platform_device *pdev)
{
struct device_node *cpu_np, *codec_np;
struct platform_device *cpu_pdev;
struct i2c_client *codec_dev;
struct imx_tlv320aic23_data *data = NULL;
int ret;
cpu_np = of_parse_phandle(pdev->dev.of_node, "cpu-dai", 0);
codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0);
if (!cpu_np || !codec_np) {
dev_err(&pdev->dev, "phandle missing or invalid\n");
ret = -EINVAL;
goto fail;
}
if (strstr(cpu_np->name, "ssi")) {
ret = imx_tlv320aic23_audmux_config(pdev);
if (ret)
goto fail;
}
cpu_pdev = of_find_device_by_node(cpu_np);
if (!cpu_pdev) {
dev_err(&pdev->dev, "failed to find SSI platform device\n");
ret = -EPROBE_DEFER;
goto fail;
}
codec_dev = of_find_i2c_device_by_node(codec_np);
if (!codec_dev) {
dev_err(&pdev->dev, "failed to find codec platform device\n");
return -EPROBE_DEFER;
}
data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
if (!data) {
ret = -ENOMEM;
goto fail;
}
data->codec_clk = clk_get(&codec_dev->dev, NULL);
if (IS_ERR(data->codec_clk)) {
ret = PTR_ERR(data->codec_clk);
goto fail;
}
data->clk_frequency = clk_get_rate(data->codec_clk);
data->dai.name = "tlv320aic23";
data->dai.stream_name = "TLV320AIC23";
data->dai.codec_dai_name = "tlv320aic23-hifi";
data->dai.codec_of_node = codec_np;
data->dai.cpu_of_node = cpu_np;
data->dai.platform_of_node = cpu_np;
data->dai.init = &imx_tlv320aic23_dai_init;
data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM;
data->card.dev = &pdev->dev;
ret = snd_soc_of_parse_card_name(&data->card, "model");
if (ret)
goto fail;
ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
if (ret)
goto fail;
data->card.num_links = 1;
data->card.owner = THIS_MODULE;
data->card.dai_link = &data->dai;
data->card.dapm_widgets = imx_tlv320aic23_dapm_widgets;
data->card.num_dapm_widgets = ARRAY_SIZE(imx_tlv320aic23_dapm_widgets);
platform_set_drvdata(pdev, &data->card);
snd_soc_card_set_drvdata(&data->card, data);
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 fail;
}
of_node_put(cpu_np);
of_node_put(codec_np);
return 0;
fail:
if (data && !IS_ERR(data->codec_clk))
clk_put(data->codec_clk);
if (cpu_np)
of_node_put(cpu_np);
if (codec_np)
of_node_put(codec_np);
return ret;
}
static int imx_tlv320aic23_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
struct imx_tlv320aic23_data *data = snd_soc_card_get_drvdata(card);
clk_put(data->codec_clk);
return 0;
}
static const struct of_device_id imx_tlv320aic23_dt_ids[] = {
{ .compatible = "fsl,imx-audio-tlv320aic23", },
{ /* sentinel */ }
};
MODULE_DEVICE_TABLE(of, imx_tlv320aic23_dt_ids);
static struct platform_driver imx_tlv320aic23_driver = {
.driver = {
.name = "imx-tlv320aic23",
.owner = THIS_MODULE,
.pm = &snd_soc_pm_ops,
.of_match_table = imx_tlv320aic23_dt_ids,
},
.probe = imx_tlv320aic23_probe,
.remove = imx_tlv320aic23_remove,
};
module_platform_driver(imx_tlv320aic23_driver);
MODULE_AUTHOR("Freescale Semiconductor, Inc.");
MODULE_DESCRIPTION("Freescale i.MX TLV320AIC23 ASoC machine driver");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:imx-tlv320aic23");
arch/arm/boot/dts/imx6qdl-sabresd.dtsi
&i2c1 {
clock-frequency = <100000>;
pinctrl-names = "default";
pinctrl-0 = <&pinctrl_i2c1>;
status = "okay";
- codec: wm8962@1a {
- compatible = "wlf,wm8962";
+ codec: tlv320aic23@1a {
+ compatible = "fsl,tlv320aic23";
reg = <0x1a>;
clocks = <&clks 201>;
DCVDD-supply = <®_audio>;
sound {
- compatible = "fsl,imx6q-sabresd-wm8962",
- "fsl,imx-audio-wm8962";
- model = "wm8962-audio";
+ compatible = "fsl,imx6q-sabresd-tlv320aic23",
+ "fsl,imx-audio-tlv320aic23";
+ model = "tlv320aic23-audio";
cpu-dai = <&ssi2>;
audio-codec = <&codec>;
audio-routing =
- "Headphone Jack", "HPOUTL",
- "Headphone Jack", "HPOUTR"
- "Ext Spk", "SPKOUTL",
- "Ext Spk", "SPKOUTR",
- "MICBIAS", "AMIC",
- "IN3R", "MICBIAS",
- "DMIC", "MICBIAS",
- "DMICDAT", "DMIC";
+ "MICIN", "Mic Jack",
+ "Headphone Jack", "LHPOUT",
+ "Headphone Jack", "RHPOUT";
mux-int-port = <2>;
mux-ext-port = <3>;
hp-det-gpios = <&gpio1 29 1>;
mic-det-gpios = <&gpio3 20 1>;
};
reg_audio: regulator@2 {
compatible = "regulator-fixed";
reg = <2>;
- regulator-name = "wm8962-supply";
+ regulator-name = "tlv320aic23-supply";
gpio = <&gpio4 10 0>;
enable-active-high;
};
issue is:
Normal playback time of MP3 file is 3 minutes.
Now playing time is 6 minutes.
Sound not normal
thanks
TLV320AIC23 provides an external clock for 12M
Hi haibo
I believe one needs to pay attention to codec reference clock,
seems it is twice slow as needed. Also seems Sabre board
uses 24MHz at AUD_MCLK/CLKO for codec.
Best regards
igor
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root@imx6solosabresd:~# gplay AsLongasYouLoveMe.mp3
playbin2 is employed!
fsl_player_init(): Successfully initialize!
fsl_player_set_media_location(): filename=AsLongasYouLoveMe.mp3
[Stopped ][Vol=01][00:00:00/00:00:00][fps:0]Home directory not accessible: Permission denied
FOUND GST_MESSAGE_TAG!
ID3v2 frame: buffer of 21 bytes, type: application/x-gst-id3v2-tpe1-frame, version=(int)3
: buffer of 25 bytes, type: application/x-gst-id3v2-talb-frame, version=(int)3
: buffer of 57 bytes, type: application/x-gst-id3v2-tit2-frame, version=(int)3
: buffer of 33 bytes, type: application/x-gst-id3v2-tyer-frame, version=(int)3
container format: ID3 tag
Home directory not accessible: Permission denied
Home directory not accessible: Permission denied
FOUND GST_MESSAGE_TAG!
audio codec: MPEG 1 Audio, Layer 3 (MP3)
nominal bitrate: 128000
FOUND GST_MESSAGE_TAG!
Beep: 4.0.3
Core: MP3 decoder Wrapper build on Mar 21 2014 15:04:50
mime: audio/mpeg, mpegversion = (int)1
file: /usr/lib/imx-mm/audio-codec/wrap/lib_mp3d_wrap_arm12_elinux.so.3
has crc: FALSE
channel mode: joint-stereo
CODEC: BLN_MAD-MMCODECS_MP3D_ARM_02.13.00_CORTEX-A8 build on Apr 10 2014 15:26:15.
FOUND GST_MESSAGE_TAG!
number of channels: 2
bitrate: 128000
sampling frequency (Hz): 44100
audio codec: MPEG-1 Layer 3
FOUND GST_MESSAGE_TAG!
minimum bitrate: 128012
bitrate: 127981
maximum bitrate: 128012
FOUND GST_MESSAGE_TAG!
minimum bitrate: 127706
get GST_MESSAGE_ELEMENT playbin2-stream-changed, uri=(string)file:///home/root/AsLongasYouLoveMe.mp3;
fsl_player_play()
FSL_PLAYER_01.00_LINUX build on Jul 22 2015 12:53:22
[h]display the operation Help
[p]Play
[s]Stop
[e]Seek
[a]Pause when playing, play when paused
[v]Volume
[m]Switch to mute or not
[>]Play next file
[<]Play previous file
[r]Switch to repeated mode or not
[f]Set full screen or not
[z]resize the width and height
[t]Rotate
[c]Setting play rate
[i]Display the metadata
[x]eXit
[Playing ][Vol=01][00:00:02/00:03:31][fps:0]v
Set volume[0-1.0]:9
[Playing ][Vol=09][00:00:15/00:03:31][fps:0]0
[Playing ][Vol=09][00:00:22/00:03:31][fps:0]0.
[Playing ][Vol=09][00:00:27/00:03:31][fps:0]8
[Playing ][Vol=09][00:00:30/00:03:31][fps:0]v
Set volume[0-1.0]:8
[Playing ][Vol=08][00:00:37/00:03:31][fps:0]
Normal playback audio file
new issue is:
The volume is very small, how to set up the volume of small
please recheck if word size of codec and i.MX6 SSI are adjusted
correctly.
~igor