[alsa-devel] [PATCH v6 07/10] ASoC: Intel: mrfld: add the DSP DAPM widgets
Subhransu S. Prusty
subhransu.s.prusty at intel.com
Tue Sep 9 11:41:30 CEST 2014
From: Vinod Koul <vinod.koul at intel.com>
This patch adds all DAPM widgets and the event handlers for DSP expect the
mixers. Since we are still discussing mixer update and is dependent upon
component series
Signed-off-by: Vinod Koul <vinod.koul at intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty at intel.com>
---
sound/soc/intel/sst-atom-controls.c | 230 +++++++++++++++++++++++++++++++++++-
sound/soc/intel/sst-mfld-platform.h | 4 +
2 files changed, 233 insertions(+), 1 deletion(-)
diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c
index 3be9ab1..7b01e04 100644
--- a/sound/soc/intel/sst-atom-controls.c
+++ b/sound/soc/intel/sst-atom-controls.c
@@ -492,8 +492,43 @@ static const uint swm_mixer_input_ids[SST_SWM_INPUT_COUNT] = {
[SST_IP_MEDIA2] = SST_SWM_IN_MEDIA2,
[SST_IP_MEDIA3] = SST_SWM_IN_MEDIA3,
};
+
+/**
+ * fill_swm_input - fill in the SWM input ids given the register
+ *
+ * The register value is a bit-field inicated which mixer inputs are ON. Use the
+ * lookup table to get the input-id and fill it in the structure.
+ */
+static int fill_swm_input(struct snd_soc_component *cmpnt,
+ struct swm_input_ids *swm_input, unsigned int reg)
+{
+ uint i, is_set, nb_inputs = 0;
+ u16 input_loc_id;
+
+ dev_dbg(cmpnt->dev, "reg: %#x\n", reg);
+ for (i = 0; i < SST_SWM_INPUT_COUNT; i++) {
+ is_set = reg & BIT(i);
+ if (!is_set)
+ continue;
+
+ input_loc_id = swm_mixer_input_ids[i];
+ SST_FILL_DESTINATION(2, swm_input->input_id,
+ input_loc_id, SST_DEFAULT_MODULE_ID);
+ nb_inputs++;
+ swm_input++;
+ dev_dbg(cmpnt->dev, "input id: %#x, nb_inputs: %d\n",
+ input_loc_id, nb_inputs);
+
+ if (nb_inputs == SST_CMD_SWM_MAX_INPUTS) {
+ dev_warn(cmpnt->dev, "SET_SWM cmd max inputs reached");
+ break;
+ }
+ }
+ return nb_inputs;
+}
+
static void sst_set_pipe_gain(struct sst_ids *ids,
- struct sst_data *drv, int mute)
+ struct sst_data *drv, int mute)
{
struct sst_gain_mixer_control *mc;
struct sst_gain_value *gv;
@@ -510,6 +545,114 @@ static void sst_set_pipe_gain(struct sst_ids *ids,
mc->pipe_id | mc->instance_id, mc->module_id, mute);
}
}
+
+static int sst_swm_mixer_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ struct sst_cmd_set_swm cmd;
+ struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm);
+ struct sst_data *drv = snd_soc_component_get_drvdata(cmpnt);
+ struct sst_ids *ids = w->priv;
+ bool set_mixer = false;
+ struct soc_mixer_control *mc;
+ int val = 0;
+ int i = 0;
+
+ dev_dbg(cmpnt->dev, "widget = %s\n", w->name);
+ /*
+ * Identify which mixer input is on and send the bitmap of the
+ * inputs as an IPC to the DSP.
+ */
+ for (i = 0; i < w->num_kcontrols; i++) {
+ if (dapm_kcontrol_get_value(w->kcontrols[i])) {
+ mc = (struct soc_mixer_control *)(w->kcontrols[i])->private_value;
+ val |= 1 << mc->shift;
+ }
+ }
+ dev_dbg(cmpnt->dev, "val = %#x\n", val);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ case SND_SOC_DAPM_POST_PMD:
+ set_mixer = true;
+ break;
+ case SND_SOC_DAPM_POST_REG:
+ if (w->power)
+ set_mixer = true;
+ break;
+ default:
+ set_mixer = false;
+ }
+
+ if (set_mixer == false)
+ return 0;
+
+ if (SND_SOC_DAPM_EVENT_ON(event) ||
+ event == SND_SOC_DAPM_POST_REG)
+ cmd.switch_state = SST_SWM_ON;
+ else
+ cmd.switch_state = SST_SWM_OFF;
+
+ SST_FILL_DEFAULT_DESTINATION(cmd.header.dst);
+ /*MMX_SET_SWM == SBA_SET_SWM*/
+ cmd.header.command_id = SBA_SET_SWM;
+
+ SST_FILL_DESTINATION(2, cmd.output_id,
+ ids->location_id, SST_DEFAULT_MODULE_ID);
+ cmd.nb_inputs = fill_swm_input(cmpnt, &cmd.input[0], val);
+ cmd.header.length = offsetof(struct sst_cmd_set_swm, input)
+ - sizeof(struct sst_dsp_header)
+ + (cmd.nb_inputs * sizeof(cmd.input[0]));
+
+ sst_fill_and_send_cmd(drv, SST_IPC_IA_CMD, SST_FLAG_BLOCKED,
+ ids->task_id, 0, &cmd,
+ sizeof(cmd.header) + cmd.header.length);
+ return 0;
+}
+
+/*SBA mixers - 16 inputs*/
+#define SST_SBA_DECLARE_MIX_CONTROLS(kctl_name) \
+ static const struct snd_kcontrol_new kctl_name[] = { \
+ SOC_DAPM_SINGLE("codec_in0", SND_SOC_NOPM, SST_IP_CODEC0, 1, 0), \
+ SOC_DAPM_SINGLE("codec_in1", SND_SOC_NOPM, SST_IP_CODEC1, 1, 0), \
+ SOC_DAPM_SINGLE("sprot_loop_in", SND_SOC_NOPM, SST_IP_LOOP0, 1, 0), \
+ SOC_DAPM_SINGLE("media_loop1_in", SND_SOC_NOPM, SST_IP_LOOP1, 1, 0), \
+ SOC_DAPM_SINGLE("media_loop2_in", SND_SOC_NOPM, SST_IP_LOOP2, 1, 0), \
+ SOC_DAPM_SINGLE("pcm0_in", SND_SOC_NOPM, SST_IP_PCM0, 1, 0), \
+ SOC_DAPM_SINGLE("pcm1_in", SND_SOC_NOPM, SST_IP_PCM1, 1, 0), \
+ }
+
+#define SST_SBA_MIXER_GRAPH_MAP(mix_name) \
+ { mix_name, "codec_in0", "codec_in0" }, \
+ { mix_name, "codec_in1", "codec_in1" }, \
+ { mix_name, "sprot_loop_in", "sprot_loop_in" }, \
+ { mix_name, "media_loop1_in", "media_loop1_in" }, \
+ { mix_name, "media_loop2_in", "media_loop2_in" }, \
+ { mix_name, "pcm0_in", "pcm0_in" }, \
+ { mix_name, "pcm1_in", "pcm1_in" }
+
+#define SST_MMX_DECLARE_MIX_CONTROLS(kctl_name) \
+ static const struct snd_kcontrol_new kctl_name[] = { \
+ SOC_DAPM_SINGLE("media0_in", SND_SOC_NOPM, SST_IP_MEDIA0, 1, 0), \
+ SOC_DAPM_SINGLE("media1_in", SND_SOC_NOPM, SST_IP_MEDIA1, 1, 0), \
+ SOC_DAPM_SINGLE("media2_in", SND_SOC_NOPM, SST_IP_MEDIA2, 1, 0), \
+ SOC_DAPM_SINGLE("media3_in", SND_SOC_NOPM, SST_IP_MEDIA3, 1, 0), \
+ }
+
+SST_MMX_DECLARE_MIX_CONTROLS(sst_mix_media0_controls);
+SST_MMX_DECLARE_MIX_CONTROLS(sst_mix_media1_controls);
+
+/*18 SBA mixers*/
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_pcm0_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_pcm1_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_pcm2_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_sprot_l0_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_media_l1_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_media_l2_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_voip_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec0_controls);
+SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec1_controls);
+
void sst_handle_vb_timer(struct snd_soc_dai *dai, bool enable)
{
struct sst_cmd_generic cmd;
@@ -690,6 +833,83 @@ static int sst_set_media_loop(struct snd_soc_dapm_widget *w,
return 0;
}
+static const struct snd_soc_dapm_widget sst_dapm_widgets[] = {
+ SST_AIF_IN("codec_in0", sst_set_be_modules),
+ SST_AIF_IN("codec_in1", sst_set_be_modules),
+ SST_AIF_OUT("codec_out0", sst_set_be_modules),
+ SST_AIF_OUT("codec_out1", sst_set_be_modules),
+
+ /*Media Paths*/
+ /*MediaX IN paths are set via ALLOC, so no SET_MEDIA_PATH command*/
+ SST_PATH_INPUT("media0_in", SST_TASK_MMX, SST_SWM_IN_MEDIA0, sst_generic_modules_event),
+ SST_PATH_INPUT("media1_in", SST_TASK_MMX, SST_SWM_IN_MEDIA1, NULL),
+ SST_PATH_INPUT("media2_in", SST_TASK_MMX, SST_SWM_IN_MEDIA2, sst_set_media_path),
+ SST_PATH_INPUT("media3_in", SST_TASK_MMX, SST_SWM_IN_MEDIA3, NULL),
+ SST_PATH_OUTPUT("media0_out", SST_TASK_MMX, SST_SWM_OUT_MEDIA0, sst_set_media_path),
+ SST_PATH_OUTPUT("media1_out", SST_TASK_MMX, SST_SWM_OUT_MEDIA1, sst_set_media_path),
+
+ /*SBA PCM Paths*/
+ SST_PATH_INPUT("pcm0_in", SST_TASK_SBA, SST_SWM_IN_PCM0, sst_set_media_path),
+ SST_PATH_INPUT("pcm1_in", SST_TASK_SBA, SST_SWM_IN_PCM1, sst_set_media_path),
+ SST_PATH_OUTPUT("pcm0_out", SST_TASK_SBA, SST_SWM_OUT_PCM0, sst_set_media_path),
+ SST_PATH_OUTPUT("pcm1_out", SST_TASK_SBA, SST_SWM_OUT_PCM1, sst_set_media_path),
+ SST_PATH_OUTPUT("pcm2_out", SST_TASK_SBA, SST_SWM_OUT_PCM2, sst_set_media_path),
+
+ /*SBA Loops*/
+ SST_PATH_INPUT("sprot_loop_in", SST_TASK_SBA, SST_SWM_IN_SPROT_LOOP, NULL),
+ SST_PATH_INPUT("media_loop1_in", SST_TASK_SBA, SST_SWM_IN_MEDIA_LOOP1, NULL),
+ SST_PATH_INPUT("media_loop2_in", SST_TASK_SBA, SST_SWM_IN_MEDIA_LOOP2, NULL),
+ SST_PATH_MEDIA_LOOP_OUTPUT("sprot_loop_out", SST_TASK_SBA, SST_SWM_OUT_SPROT_LOOP, SST_FMT_MONO, sst_set_media_loop),
+ SST_PATH_MEDIA_LOOP_OUTPUT("media_loop1_out", SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP1, SST_FMT_MONO, sst_set_media_loop),
+ SST_PATH_MEDIA_LOOP_OUTPUT("media_loop2_out", SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP2, SST_FMT_STEREO, sst_set_media_loop),
+
+ /*Media Mixers*/
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ {"media0_in", NULL, "Compress Playback"},
+ {"media1_in", NULL, "Headset Playback"},
+ {"media2_in", NULL, "pcm0_out"},
+
+ {"media0_out mix 0", "media0_in", "media0_in"},
+ {"media0_out mix 0", "media1_in", "media1_in"},
+ {"media0_out mix 0", "media2_in", "media2_in"},
+ {"media0_out mix 0", "media3_in", "media3_in"},
+ {"media1_out mix 0", "media0_in", "media0_in"},
+ {"media1_out mix 0", "media1_in", "media1_in"},
+ {"media1_out mix 0", "media2_in", "media2_in"},
+ {"media1_out mix 0", "media3_in", "media3_in"},
+
+ {"media0_out", NULL, "media0_out mix 0"},
+ {"media1_out", NULL, "media1_out mix 0"},
+ {"pcm0_in", NULL, "media0_out"},
+ {"pcm1_in", NULL, "media1_out"},
+
+ {"Headset Capture", NULL, "pcm1_out"},
+ {"Headset Capture", NULL, "pcm2_out"},
+ {"pcm0_out", NULL, "pcm0_out mix 0"},
+ SST_SBA_MIXER_GRAPH_MAP("pcm0_out mix 0"),
+ {"pcm1_out", NULL, "pcm1_out mix 0"},
+ SST_SBA_MIXER_GRAPH_MAP("pcm1_out mix 0"),
+ {"pcm2_out", NULL, "pcm2_out mix 0"},
+ SST_SBA_MIXER_GRAPH_MAP("pcm2_out mix 0"),
+
+ {"media_loop1_in", NULL, "media_loop1_out"},
+ {"media_loop1_out", NULL, "media_loop1_out mix 0"},
+ SST_SBA_MIXER_GRAPH_MAP("media_loop1_out mix 0"),
+ {"media_loop2_in", NULL, "media_loop2_out"},
+ {"media_loop2_out", NULL, "media_loop2_out mix 0"},
+ SST_SBA_MIXER_GRAPH_MAP("media_loop2_out mix 0"),
+ {"sprot_loop_in", NULL, "sprot_loop_out"},
+ {"sprot_loop_out", NULL, "sprot_loop_out mix 0"},
+ SST_SBA_MIXER_GRAPH_MAP("sprot_loop_out mix 0"),
+
+ {"codec_out0", NULL, "codec_out0 mix 0"},
+ SST_SBA_MIXER_GRAPH_MAP("codec_out0 mix 0"),
+ {"codec_out1", NULL, "codec_out1 mix 0"},
+ SST_SBA_MIXER_GRAPH_MAP("codec_out1 mix 0"),
+
+};
static const char * const slot_names[] = {
"none",
"slot 0", "slot 1", "slot 2", "slot 3",
@@ -1006,6 +1226,8 @@ static int sst_map_modules_to_pipe(struct snd_soc_platform *platform)
int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform)
{
int i, ret = 0;
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_component_get_dapm(&platform->component);
struct sst_data *drv = snd_soc_platform_get_drvdata(platform);
drv->byte_stream = devm_kzalloc(platform->dev,
@@ -1013,6 +1235,12 @@ int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform)
if (!drv->byte_stream)
return -ENOMEM;
+ snd_soc_dapm_new_controls(dapm, sst_dapm_widgets,
+ ARRAY_SIZE(sst_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, intercon,
+ ARRAY_SIZE(intercon));
+ snd_soc_dapm_new_widgets(dapm->card);
+
for (i = 0; i < SST_NUM_GAINS; i++) {
sst_gains[i].mute = SST_GAIN_MUTE_DEFAULT;
sst_gains[i].l_gain = SST_GAIN_VOLUME_DEFAULT;
diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h
index bf2b6d4..c16d489 100644
--- a/sound/soc/intel/sst-mfld-platform.h
+++ b/sound/soc/intel/sst-mfld-platform.h
@@ -153,6 +153,10 @@ struct sst_device {
struct sst_data;
int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform);
+int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute);
+void send_ssp_cmd(struct snd_soc_dai *dai, const char *id, bool enable);
+void sst_handle_vb_timer(struct snd_soc_dai *dai, bool enable);
+
void sst_set_stream_status(struct sst_runtime_stream *stream, int state);
int sst_fill_stream_params(void *substream, const struct sst_data *ctx,
struct snd_sst_params *str_params, bool is_compress);
--
1.9.0
More information about the Alsa-devel
mailing list