ASoC: msm: Remove unnecessary soc-dsp structures
After adopting new soc-dsp framework, list of back-ends
for a given front-end is no longer needed.
Signed-off-by: Patrick Lai <plai@codeaurora.org>
diff --git a/sound/soc/msm/msm8960.c b/sound/soc/msm/msm8960.c
index 8c0df5c..b10ed35 100644
--- a/sound/soc/msm/msm8960.c
+++ b/sound/soc/msm/msm8960.c
@@ -273,73 +273,35 @@
return 0;
}
-/*
- * LPA Needs only RX BE DAI links.
- * Hence define seperate BE list for lpa
- */
-
-static const char *lpa_mm_be[] = {
- LPASS_BE_SLIMBUS_0_RX,
-};
-
static struct snd_soc_dsp_link lpa_fe_media = {
- .supported_be = lpa_mm_be,
- .num_be = ARRAY_SIZE(lpa_mm_be),
- .fe_playback_channels = 2,
- .fe_capture_channels = 1,
+ .playback = true,
.trigger = {
SND_SOC_DSP_TRIGGER_POST,
SND_SOC_DSP_TRIGGER_POST
},
};
-static const char *mm_be[] = {
- LPASS_BE_SLIMBUS_0_RX,
- LPASS_BE_SLIMBUS_0_TX,
- LPASS_BE_HDMI,
- LPASS_BE_INT_BT_SCO_RX,
- LPASS_BE_INT_BT_SCO_TX,
- LPASS_BE_INT_FM_RX,
- LPASS_BE_INT_FM_TX,
-};
-
static struct snd_soc_dsp_link fe_media = {
- .supported_be = mm_be,
- .num_be = ARRAY_SIZE(mm_be),
- .fe_playback_channels = 2,
- .fe_capture_channels = 1,
+ .playback = true,
+ .capture = true,
.trigger = {
SND_SOC_DSP_TRIGGER_POST,
SND_SOC_DSP_TRIGGER_POST
},
};
-static const char *slimbus0_hl_be[] = {
- LPASS_BE_SLIMBUS_0_RX,
- LPASS_BE_SLIMBUS_0_TX,
-};
-
static struct snd_soc_dsp_link slimbus0_hl_media = {
- .supported_be = slimbus0_hl_be,
- .num_be = ARRAY_SIZE(slimbus0_hl_be),
- .fe_playback_channels = 2,
- .fe_capture_channels = 2,
+ .playback = true,
+ .capture = true,
.trigger = {
SND_SOC_DSP_TRIGGER_POST,
SND_SOC_DSP_TRIGGER_POST
},
};
-static const char *int_fm_hl_be[] = {
- LPASS_BE_INT_FM_RX,
- LPASS_BE_INT_FM_TX,
-};
-
static struct snd_soc_dsp_link int_fm_hl_media = {
- .supported_be = int_fm_hl_be,
- .num_be = ARRAY_SIZE(int_fm_hl_be),
- .fe_playback_channels = 2,
- .fe_capture_channels = 2,
+ .playback = true,
+ .capture = true,
.trigger = {
SND_SOC_DSP_TRIGGER_POST,
SND_SOC_DSP_TRIGGER_POST