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>

4 files changed, 69 insertions, 3 deletions

diff --git a/include/uapi/linux/mxc_dsp.h b/include/uapi/linux/mxc_dsp.h
index 040681cc39c8..10c282f882c6 100644
--- a/include/uapi/linux/mxc_dsp.h
+++ b/include/uapi/linux/mxc_dsp.h
@@ -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
diff --git a/sound/soc/fsl/fsl_dsp_platform_compress.c b/sound/soc/fsl/fsl_dsp_platform_compress.c
index 38d375e315de..35bde5f7ef8e 100644
--- a/sound/soc/fsl/fsl_dsp_platform_compress.c
+++ b/sound/soc/fsl/fsl_dsp_platform_compress.c
@@ -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;
 
diff --git a/sound/soc/fsl/fsl_dsp_xaf_api.c b/sound/soc/fsl/fsl_dsp_xaf_api.c
index 2befc7c144b0..f721854ac67c 100644
--- a/sound/soc/fsl/fsl_dsp_xaf_api.c
+++ b/sound/soc/fsl/fsl_dsp_xaf_api.c
@@ -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;
diff --git a/sound/soc/fsl/fsl_dsp_xaf_api.h b/sound/soc/fsl/fsl_dsp_xaf_api.h
index 051a943f29e4..3b1175829a21 100644
--- a/sound/soc/fsl/fsl_dsp_xaf_api.h
+++ b/sound/soc/fsl/fsl_dsp_xaf_api.h
@@ -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 */