output/wav: Add 'scale' option.

Audio tools processing WAV failes generally need the samples to be in
the range -1 to +1. The scale option adds postprocessing to any samples
going into a WAV file, by dividing the sample values by the given factor.

diff --git a/src/output/wav.c b/src/output/wav.c
index 8aa787bd3f14e36f67fc7a58e870b20c98762fc2..61d184ccb5ce52333ce266605b7588022287f22c 100644 (file)
--- a/src/output/wav.c
+++ b/src/output/wav.c
@@ -27,6 +27,7 @@
 #define MIN_DATA_CHUNK_SAMPLES 10
 
 struct out_context {
+       double scale;
        gboolean header_done;
        uint64_t samplerate;
        int num_channels;
@@ -90,12 +91,29 @@ static int init(struct sr_output *o, GHashTable *options)
        struct out_context *outc;
        struct sr_channel *ch;
        GSList *l;
-
-       (void)options;
+       GHashTableIter iter;
+       gpointer key, value;
 
        outc = g_malloc0(sizeof(struct out_context));
        o->priv = outc;
 
+       outc->scale = 0.0;
+       if (options) {
+               g_hash_table_iter_init(&iter, options);
+               while (g_hash_table_iter_next(&iter, &key, &value)) {
+                       if (!strcmp(key, "scale")) {
+                               if (!g_variant_is_of_type(value, G_VARIANT_TYPE_DOUBLE)) {
+                                       sr_err("Invalid type for 'scale' option.");
+                                       return SR_ERR_ARG;
+                               }
+                               outc->scale = g_variant_get_double(value);
+                       } else {
+                               sr_err("Unknown option '%s'.", key);
+                               return SR_ERR_ARG;
+                       }
+               }
+       }
+
        for (l = o->sdi->channels; l; l = l->next) {
                ch = l->data;
                if (ch->type != SR_CHANNEL_ANALOG)
@@ -238,6 +256,7 @@ static int receive(const struct sr_output *o, const struct sr_datafeed_packet *p
        const struct sr_config *src;
        struct sr_channel *ch;
        GSList *l;
+       float f;
        int num_channels, size, *chan_idx, idx, i, j;
        uint8_t *buf;
 
@@ -289,7 +308,10 @@ static int receive(const struct sr_output *o, const struct sr_datafeed_packet *p
                        for (j = 0; j < num_channels; j++) {
                                idx = chan_idx[j];
                                buf = outc->chanbuf[idx] + outc->chanbuf_used[idx]++ * 4;
-                               float_to_le(buf, analog->data[i * num_channels + j]);
+                               f = analog->data[i * num_channels + j];
+                               if (outc->scale != 0.0)
+                                       f /= outc->scale;
+                               float_to_le(buf, f);
                        }
                }
                g_free(chan_idx);
@@ -329,10 +351,27 @@ static int cleanup(struct sr_output *o)
        return SR_OK;
 }
 
+static struct sr_option options[] = {
+       { "scale", "Scale", "Scale values by factor", NULL, NULL },
+       { 0 }
+};
+
+static struct sr_option *get_options(gboolean cached)
+{
+       if (cached)
+               return options;
+
+       options[0].def = g_variant_new_double(0);
+       g_variant_ref_sink(options[0].def);
+
+       return options;
+}
+
 SR_PRIV struct sr_output_module output_wav = {
        .id = "wav",
        .name = "WAV",
        .desc = "WAVE file format",
+       .options = get_options,
        .init = init,
        .receive = receive,
        .cleanup = cleanup,