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-rw-r--r--src/output/alsa_plugin.c4
-rw-r--r--src/output/jack_plugin.c1
-rw-r--r--src/output/oss_plugin.c5
-rw-r--r--src/output/pulse_plugin.c6
4 files changed, 0 insertions, 16 deletions
diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c
index 874e4ac93..b3f20df42 100644
--- a/src/output/alsa_plugin.c
+++ b/src/output/alsa_plugin.c
@@ -408,10 +408,6 @@ alsa_open(void *data, struct audio_format *audio_format, GError **error)
ad->frame_size = audio_format_frame_size(audio_format);
- g_debug("ALSA device \"%s\" will be playing %i bit, %u channel audio at %u Hz\n",
- alsa_device(ad), audio_format->bits, audio_format->channels,
- audio_format->sample_rate);
-
return true;
}
diff --git a/src/output/jack_plugin.c b/src/output/jack_plugin.c
index 6b3dd9d03..16f592dc5 100644
--- a/src/output/jack_plugin.c
+++ b/src/output/jack_plugin.c
@@ -144,7 +144,6 @@ static void
set_audioformat(struct jack_data *jd, struct audio_format *audio_format)
{
audio_format->sample_rate = jack_get_sample_rate(jd->client);
- g_debug("samplerate = %u", audio_format->sample_rate);
audio_format->channels = 2;
if (audio_format->bits != 16 && audio_format->bits != 24)
diff --git a/src/output/oss_plugin.c b/src/output/oss_plugin.c
index d4ecf88f7..2921442eb 100644
--- a/src/output/oss_plugin.c
+++ b/src/output/oss_plugin.c
@@ -563,11 +563,6 @@ oss_output_open(void *data, struct audio_format *audio_format, GError **error)
*audio_format = od->audio_format;
- g_debug("device \"%s\" will be playing %u bit %u channel audio at "
- "%u Hz\n", od->device,
- od->audio_format.bits, od->audio_format.channels,
- od->audio_format.sample_rate);
-
mixer_open(od->mixer);
return ret;
diff --git a/src/output/pulse_plugin.c b/src/output/pulse_plugin.c
index dcdd427db..a03275a22 100644
--- a/src/output/pulse_plugin.c
+++ b/src/output/pulse_plugin.c
@@ -132,12 +132,6 @@ pulse_open(void *data, struct audio_format *audio_format, GError **error_r)
return false;
}
- g_debug("PulseAudio output \"%s\" connected and playing %i bit, %i "
- "channel audio at %i Hz\n",
- pd->name,
- audio_format->bits,
- audio_format->channels, audio_format->sample_rate);
-
return true;
}