alsa: Fix sample acquisition and send normalized values

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

diff --git a/hardware/alsa/api.c b/hardware/alsa/api.c
index dc762e06583abb0f683ba84cf25459f848046352..a8806fc2f6df044d6f8542fdfe3be6f7d03c26c6 100644 (file)
--- a/hardware/alsa/api.c
+++ b/hardware/alsa/api.c
@@ -209,9 +209,11 @@ static int hw_dev_acquisition_start(const struct sr_dev_inst *sdi,
        struct sr_datafeed_meta_analog meta;
        struct dev_context *devc;
        int count, ret;
+       char *endianess;
 
        devc = sdi->priv;
        devc->cb_data = cb_data;
+       devc->num_samples = 0;
 
        sr_dbg("Setting audio access type to RW/interleaved.");
        ret = snd_pcm_hw_params_set_access(devc->capture_handle,
@@ -222,9 +224,14 @@ static int hw_dev_acquisition_start(const struct sr_dev_inst *sdi,
        }
 
        /* FIXME: Hardcoded for 16bits. */
-       sr_dbg("Setting audio sample format to signed 16bit (little endian).");
+       if (SND_PCM_FORMAT_S16 == SND_PCM_FORMAT_S16_LE)
+               endianess = "lilltle endian";
+       else
+               endianess = "big endian";
+       sr_dbg("Setting audio sample format to signed 16bit (%s).", endianess);
        ret = snd_pcm_hw_params_set_format(devc->capture_handle,
-                       devc->hw_params, SND_PCM_FORMAT_S16_LE);
+                                          devc->hw_params,
+                                          SND_PCM_FORMAT_S16);
        if (ret < 0) {
                sr_err("Can't set audio sample format: %s.", snd_strerror(ret));
                return SR_ERR;