alsa: Fix sample acquisition and send normalized values

[libsigrok.git] / hardware / alsa / protocol.c

diff --git a/hardware/alsa/protocol.c b/hardware/alsa/protocol.c
index 1c1d8849dd2a626d69bc752c106aedb60a95bf8a..a501ec8f97bb9b7d048169a9cce7c0e2ea2d0ff2 100644 (file)
--- a/hardware/alsa/protocol.c
+++ b/hardware/alsa/protocol.c
@@ -230,9 +230,10 @@ SR_PRIV int alsa_receive_data(int fd, int revents, void *cb_data)
        struct dev_context *devc;
        struct sr_datafeed_packet packet;
        struct sr_datafeed_analog analog;
-       char inbuf[4096];
+       int16_t inbuf[4096];
        int i, x, count, offset, samples_to_get;
-       uint16_t tmp16;
+       int16_t tmp16;
+       const float s16norm = 1 / (float)(1<<15);
 
        (void)fd;
        (void)revents;
@@ -262,11 +263,18 @@ SR_PRIV int alsa_receive_data(int fd, int revents, void *cb_data)
        }
 
        offset = 0;
-
-       for (i = 0; i < count; i++) {
+       /*
+        * It's impossible to know what voltage levels the soundcard handles.
+        * Some handle 0 dBV rms, some 0dBV peak-to-peak, +4dbmW (600 ohm), etc
+        * Each of these corresponds to a different voltage, and there is no
+        * mechanism to determine this voltage. The best solution is to send all
+        * audio data as a normalized float, and let the frontend or user worry
+        * about the calibration.
+        */
+       for (i = 0; i < count; i += devc->num_probes) {
                for (x = 0; x < devc->num_probes; x++) {
-                       tmp16 = *(uint16_t *)(inbuf + (i * 4) + (x * 2));
-                       analog.data[offset++] = (float)tmp16;
+                       tmp16 = inbuf[i+x];
+                       analog.data[offset++] = tmp16 * s16norm;
                }
        }