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MLK-24444-1: ASoC: fsl_dsp: Support PCM format for compress sound card 

For PCM format the supported format is mono/stereo, S16_LE/S32_LE,
and 8KHz-192kHz frequency.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
5.4-rM2-2.2.x-imx-squashed
Shengjiu Wang 2020-07-29 18:34:49 +08:00
parent edfae0f3a6
commit e17ee871b9
4 changed files with 69 additions and 3 deletions
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3
include/uapi/linux/mxc_dsp.h
View File

@ -44,7 +44,8 @@
#define CODEC_DRM_DEC 6
#define CODEC_SBC_DEC 7
#define CODEC_SBC_ENC 8
#define CODEC_DEMO_DEC 9
#define CODEC_PCM_DEC 9
#define CODEC_DEMO_DEC 10
#define RENDER_ESAI 0x10
#define RENDER_SAI 0x11

44
sound/soc/fsl/fsl_dsp_platform_compress.c
View File

@ -14,7 +14,7 @@
#include "fsl_dsp_platform.h"
#include "fsl_dsp_xaf_api.h"
#define NUM_CODEC 2
#define NUM_CODEC 3
#define MIN_FRAGMENT 1
#define MAX_FRAGMENT 1
#define MIN_FRAGMENT_SIZE (4 * 1024)
@ -137,6 +137,9 @@ static int dsp_platform_compr_set_params(struct snd_compr_stream *cstream,
int ret;
switch (params->codec.id) {
case SND_AUDIOCODEC_PCM:
drv->codec_type = CODEC_PCM_DEC;
break;
case SND_AUDIOCODEC_MP3:
drv->codec_type = CODEC_MP3_DEC;
break;
@ -210,6 +213,21 @@ static int dsp_platform_compr_set_params(struct snd_compr_stream *cstream,
drv->client->consume_bytes = 0;
drv->client->offset = 0;
if (drv->codec_type == CODEC_PCM_DEC) {
s_param.id = XA_PCM_CONFIG_PARAM_IN_PCM_WIDTH;
if (params->codec.format == SNDRV_PCM_FORMAT_S32_LE)
s_param.mixData.value = 32;
else
s_param.mixData.value = 16;
ret = xaf_comp_set_config(drv->client, &drv->component[0], 1, &s_param);
if (ret) {
dev_err(component->dev,
"set param[cmd:0x%x|val:0x%x] error, err = %d\n",
s_param.id, s_param.mixData.value, ret);
goto err_comp1_create;
}
}
s_param.id = XA_RENDERER_CONFIG_PARAM_SAMPLE_RATE;
s_param.mixData.value = params->codec.sample_rate;
ret = xaf_comp_set_config(drv->client, &drv->component[1], 1, &s_param);
@ -512,10 +530,32 @@ static int dsp_platform_compr_get_caps(struct snd_compr_stream *cstream,
caps->max_fragments = MAX_FRAGMENT;
caps->codecs[0] = SND_AUDIOCODEC_MP3;
caps->codecs[1] = SND_AUDIOCODEC_AAC;
caps->codecs[2] = SND_AUDIOCODEC_PCM;
return 0;
}
static struct snd_compr_codec_caps caps_pcm = {
.num_descriptors = 1,
.descriptor[0].max_ch = 2,
.descriptor[0].sample_rates[0] = 192000,
.descriptor[0].sample_rates[1] = 176400,
.descriptor[0].sample_rates[2] = 96000,
.descriptor[0].sample_rates[3] = 88200,
.descriptor[0].sample_rates[4] = 48000,
.descriptor[0].sample_rates[5] = 44100,
.descriptor[0].sample_rates[6] = 32000,
.descriptor[0].sample_rates[7] = 16000,
.descriptor[0].sample_rates[8] = 8000,
.descriptor[0].num_sample_rates = 9,
.descriptor[0].bit_rate[0] = 320,
.descriptor[0].bit_rate[1] = 192,
.descriptor[0].num_bitrates = 2,
.descriptor[0].profiles = SND_AUDIOPROFILE_PCM,
.descriptor[0].modes = 0,
.descriptor[0].formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
};
static struct snd_compr_codec_caps caps_mp3 = {
.num_descriptors = 1,
.descriptor[0].max_ch = 2,
@ -559,6 +599,8 @@ static int dsp_platform_compr_get_codec_caps(struct snd_compr_stream *cstream,
*codec = caps_mp3;
else if (codec->codec == SND_AUDIOCODEC_AAC)
*codec = caps_aac;
else if (codec->codec == SND_AUDIOCODEC_PCM)
*codec = caps_pcm;
else
return -EINVAL;

13
sound/soc/fsl/fsl_dsp_xaf_api.c
View File

@ -180,6 +180,7 @@ int xaf_comp_create(struct xf_client *client, struct xf_proxy *proxy,
struct xf_buffer *buf;
int ret = 0, size;
bool loadlib = true;
bool request_inbuf = true;
memset((void *)p_comp, 0, sizeof(struct xaf_comp));
@ -190,9 +191,14 @@ int xaf_comp_create(struct xf_client *client, struct xf_proxy *proxy,
p_comp->comp_type = comp_type;
if (comp_type == RENDER_ESAI || comp_type == RENDER_SAI)
/* No need to load library for PCM */
if (comp_type == RENDER_ESAI || comp_type == RENDER_SAI || comp_type == CODEC_PCM_DEC)
loadlib = false;
/* Need to allocate in buffer for PCM */
if (comp_type == RENDER_ESAI || comp_type == RENDER_SAI)
request_inbuf = false;
if (loadlib) {
p_comp->codec_lib.filename = lib_path;
p_comp->codec_wrap_lib.filename = lib_wrap_path;
@ -200,6 +206,9 @@ int xaf_comp_create(struct xf_client *client, struct xf_proxy *proxy,
}
switch (comp_type) {
case CODEC_PCM_DEC:
p_comp->dec_id = "audio-decoder/pcm";
break;
case CODEC_MP3_DEC:
p_comp->dec_id = "audio-decoder/mp3";
strcat(lib_path, "lib_dsp_mp3_dec.so");
@ -245,7 +254,9 @@ int xaf_comp_create(struct xf_client *client, struct xf_proxy *proxy,
dev_err(dsp_priv->dev, "load codec lib error\n");
goto err_codec_load;
}
}
if (request_inbuf) {
/* ...allocate input buffer */
if (dsp_priv->dsp_is_lpa)
size = INBUF_SIZE_LPA;

12
sound/soc/fsl/fsl_dsp_xaf_api.h
View File

@ -161,4 +161,16 @@ enum xa_config_param_renderer {
XA_RENDERER_CONFIG_PARAM_NUM = 7,
};
/* pcm codec configuration parameters */
enum xa_config_param_pcm {
XA_PCM_CONFIG_PARAM_SAMPLE_RATE = 0, /* not supported */
XA_PCM_CONFIG_PARAM_IN_PCM_WIDTH = 1,
XA_PCM_CONFIG_PARAM_IN_CHANNELS = 2, /* not supported */
XA_PCM_CONFIG_PARAM_OUT_PCM_WIDTH = 3, /* not supported */
XA_PCM_CONFIG_PARAM_OUT_CHANNELS = 4, /* not supported */
XA_PCM_CONFIG_PARAM_CHANROUTING = 5, /* not supported */
XA_PCM_CONFIG_PARAM_FUNC_PRINT = 13, /* not supported */
XA_PCM_CONFIG_PARAM_NUM = 14, /* not supported */
};
#endif /* FSL_DSP_XAF_API_H */